Adding upstream version 1:115.7.0.upstream/1%115.7.0upstream

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

503 files changed, 53333 insertions, 0 deletions

diff --git a/third_party/libwebrtc/sdk/BUILD.gn b/third_party/libwebrtc/sdk/BUILD.gn
new file mode 100644
index 0000000000..65d1b10124
--- /dev/null
+++ b/third_party/libwebrtc/sdk/BUILD.gn
@@ -0,0 +1,1757 @@
+# Copyright 2016 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+import("//third_party/libaom/options.gni")
+import("../webrtc.gni")
+if (is_ios) {
+  import("//build/config/ios/ios_sdk.gni")
+  import("//build/config/ios/rules.gni")
+}
+if (is_mac) {
+  import("//build/config/mac/rules.gni")
+}
+
+group("sdk") {
+  public_deps = []
+  if (!build_with_chromium) {
+    if (is_android) {
+      public_deps += [ "android" ]
+    }
+    if (is_ios) {
+      public_deps += [ ":framework_objc" ]
+    }
+  }
+}
+
+rtc_library("media_constraints") {
+  sources = [
+    "media_constraints.cc",
+    "media_constraints.h",
+  ]
+  deps = [
+    "../api:audio_options_api",
+    "../api:libjingle_peerconnection_api",
+  ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
+}
+
+rtc_library("sdk_tests") {
+  testonly = true
+  sources = [ "media_constraints_unittest.cc" ]
+  deps = [
+    ":media_constraints",
+    "../test:test_support",
+  ]
+}
+
+if (is_ios || is_mac) {
+  config("common_config_objc") {
+    include_dirs = [
+      "objc",
+
+      # This is needed so that framework headers can include base headers
+      # without pathname (so it works from within the framework module).
+      "objc/base",
+    ]
+    cflags = [
+      "-Wimplicit-retain-self",
+      "-Wstrict-overflow",
+      "-Wmissing-field-initializers",
+    ]
+
+    if (use_clang_coverage) {
+      configs = [ "//build/config/coverage:default_coverage" ]
+    }
+  }
+
+  config("used_from_extension") {
+    if (is_ios && rtc_apprtcmobile_broadcast_extension) {
+      cflags = [ "-fapplication-extension" ]
+    }
+  }
+
+  # TODO(bugs.webrtc.org/9627): Remove this when unused. Targets should depend on base_objc
+  # or helpers_objc directly instead.
+  rtc_library("common_objc") {
+    visibility = [ "*" ]
+
+    sources = [ "objc/helpers/noop.mm" ]
+
+    public_configs = [ ":common_config_objc" ]
+
+    deps = [
+      ":base_objc",
+      ":helpers_objc",
+    ]
+  }
+
+  rtc_library("base_objc") {
+    visibility = [ "*" ]
+    sources = [
+      "objc/base/RTCCodecSpecificInfo.h",
+      "objc/base/RTCEncodedImage.h",
+      "objc/base/RTCEncodedImage.m",
+      "objc/base/RTCI420Buffer.h",
+      "objc/base/RTCLogging.h",
+      "objc/base/RTCLogging.mm",
+      "objc/base/RTCMacros.h",
+      "objc/base/RTCMutableI420Buffer.h",
+      "objc/base/RTCMutableYUVPlanarBuffer.h",
+      "objc/base/RTCSSLCertificateVerifier.h",
+      "objc/base/RTCVideoCapturer.h",
+      "objc/base/RTCVideoCapturer.m",
+      "objc/base/RTCVideoCodecInfo.h",
+      "objc/base/RTCVideoCodecInfo.m",
+      "objc/base/RTCVideoDecoder.h",
+      "objc/base/RTCVideoDecoderFactory.h",
+      "objc/base/RTCVideoEncoder.h",
+      "objc/base/RTCVideoEncoderFactory.h",
+      "objc/base/RTCVideoEncoderQpThresholds.h",
+      "objc/base/RTCVideoEncoderQpThresholds.m",
+      "objc/base/RTCVideoEncoderSettings.h",
+      "objc/base/RTCVideoEncoderSettings.m",
+      "objc/base/RTCVideoFrame.h",
+      "objc/base/RTCVideoFrame.mm",
+      "objc/base/RTCVideoFrameBuffer.h",
+      "objc/base/RTCVideoRenderer.h",
+      "objc/base/RTCYUVPlanarBuffer.h",
+    ]
+
+    deps = [
+      "../rtc_base:checks",
+      "../rtc_base:logging",
+    ]
+    configs += [
+      "..:common_objc",
+      ":used_from_extension",
+    ]
+
+    public_configs = [ ":common_config_objc" ]
+  }
+
+  rtc_library("helpers_objc") {
+    sources = [
+      "objc/helpers/AVCaptureSession+DevicePosition.h",
+      "objc/helpers/AVCaptureSession+DevicePosition.mm",
+      "objc/helpers/NSString+StdString.h",
+      "objc/helpers/NSString+StdString.mm",
+      "objc/helpers/RTCDispatcher+Private.h",
+      "objc/helpers/RTCDispatcher.h",
+      "objc/helpers/RTCDispatcher.m",
+      "objc/helpers/scoped_cftyperef.h",
+    ]
+
+    deps = [
+      ":base_objc",
+      "../rtc_base:checks",
+    ]
+
+    absl_deps = [ "//third_party/abseil-cpp/absl/strings" ]
+
+    frameworks = [
+      "AVFoundation.framework",
+      "CoreMedia.framework",
+      "Foundation.framework",
+    ]
+
+    configs += [
+      "..:common_objc",
+      ":used_from_extension",
+    ]
+
+    public_configs = [ ":common_config_objc" ]
+
+    if (is_ios) {
+      sources += [
+        "objc/helpers/RTCCameraPreviewView.h",
+        "objc/helpers/RTCCameraPreviewView.m",
+        "objc/helpers/UIDevice+RTCDevice.h",
+        "objc/helpers/UIDevice+RTCDevice.mm",
+      ]
+      frameworks += [ "UIKit.framework" ]
+    }
+  }
+
+  if (!build_with_chromium) {
+    rtc_library("callback_logger_objc") {
+      sources = [
+        "objc/api/logging/RTCCallbackLogger.h",
+        "objc/api/logging/RTCCallbackLogger.mm",
+      ]
+
+      deps = [
+        ":base_objc",
+        ":helpers_objc",
+        "../rtc_base:checks",
+        "../rtc_base:log_sinks",
+        "../rtc_base:logging",
+      ]
+
+      configs += [
+        "..:common_objc",
+        ":used_from_extension",
+      ]
+
+      absl_deps = [ "//third_party/abseil-cpp/absl/strings" ]
+    }
+
+    rtc_library("file_logger_objc") {
+      sources = [
+        "objc/api/peerconnection/RTCFileLogger.h",
+        "objc/api/peerconnection/RTCFileLogger.mm",
+      ]
+
+      deps = [
+        ":base_objc",
+        "../rtc_base:checks",
+        "../rtc_base:file_rotating_stream",
+        "../rtc_base:log_sinks",
+        "../rtc_base:logging",
+      ]
+
+      configs += [
+        "..:common_objc",
+        ":used_from_extension",
+      ]
+    }
+  }
+
+  if (!build_with_chromium) {
+    if (is_ios) {
+      rtc_library("native_api_audio_device_module") {
+        visibility = [ "*" ]
+
+        sources = [
+          "objc/native/api/audio_device_module.h",
+          "objc/native/api/audio_device_module.mm",
+        ]
+
+        deps = [
+          ":audio_device",
+          "../api:make_ref_counted",
+          "../modules/audio_device:audio_device_api",
+          "../modules/audio_device:audio_device_generic",
+          "../rtc_base:checks",
+          "../rtc_base:logging",
+          "../system_wrappers",
+        ]
+      }
+
+      rtc_source_set("audio_session_observer") {
+        visibility = [ ":*" ]
+
+        sources = [ "objc/native/src/audio/audio_session_observer.h" ]
+
+        deps = [ "../rtc_base:threading" ]
+      }
+
+      rtc_library("audio_device") {
+        visibility = [ "*" ]
+
+        sources = [
+          "objc/native/src/audio/audio_device_ios.h",
+          "objc/native/src/audio/audio_device_ios.mm",
+          "objc/native/src/audio/audio_device_module_ios.h",
+          "objc/native/src/audio/audio_device_module_ios.mm",
+          "objc/native/src/audio/helpers.h",
+          "objc/native/src/audio/helpers.mm",
+          "objc/native/src/audio/voice_processing_audio_unit.h",
+          "objc/native/src/audio/voice_processing_audio_unit.mm",
+        ]
+
+        deps = [
+          ":audio_objc",
+          ":audio_session_observer",
+          ":base_objc",
+          "../api:array_view",
+          "../api:scoped_refptr",
+          "../api:sequence_checker",
+          "../api/task_queue",
+          "../api/task_queue:default_task_queue_factory",
+          "../api/task_queue:pending_task_safety_flag",
+          "../modules/audio_device:audio_device_api",
+          "../modules/audio_device:audio_device_buffer",
+          "../modules/audio_device:audio_device_generic",
+          "../rtc_base:buffer",
+          "../rtc_base:checks",
+          "../rtc_base:logging",
+          "../rtc_base:macromagic",
+          "../rtc_base:refcount",
+          "../rtc_base:threading",
+          "../rtc_base:timeutils",
+          "../system_wrappers:field_trial",
+          "../system_wrappers:metrics",
+        ]
+        absl_deps = [ "//third_party/abseil-cpp/absl/base:core_headers" ]
+
+        frameworks = [ "AudioToolbox.framework" ]
+      }
+
+      # This target exists to expose :audio_session_objc and
+      # :audio_session_delegate_adapter_objc for backward compatibility,
+      # and should be deprecated.
+      group("audio_objc") {
+        public_deps = [  # no-presubmit-check TODO(webrtc:11238)
+          ":audio_session_delegate_adapter_objc",
+          ":audio_session_objc",
+        ]
+      }
+
+      rtc_library("audio_session_delegate_adapter_objc") {
+        sources = [
+          "objc/components/audio/RTCNativeAudioSessionDelegateAdapter.h",
+          "objc/components/audio/RTCNativeAudioSessionDelegateAdapter.mm",
+        ]
+
+        configs += [
+          "..:common_objc",
+          ":used_from_extension",
+        ]
+
+        public_configs = [ ":common_config_objc" ]
+
+        deps = [
+          ":audio_session_objc",
+          ":audio_session_observer",
+          ":base_objc",
+        ]
+      }
+
+      rtc_library("audio_session_objc") {
+        visibility = [ "*" ]
+
+        sources = [
+          "objc/components/audio/RTCAudioSession+Configuration.mm",
+          "objc/components/audio/RTCAudioSession+Private.h",
+          "objc/components/audio/RTCAudioSession.h",
+          "objc/components/audio/RTCAudioSession.mm",
+          "objc/components/audio/RTCAudioSessionConfiguration.h",
+          "objc/components/audio/RTCAudioSessionConfiguration.m",
+        ]
+
+        configs += [
+          "..:common_objc",
+          ":used_from_extension",
+        ]
+
+        public_configs = [ ":common_config_objc" ]
+
+        frameworks = [ "AVFoundation.framework" ]
+
+        deps = [
+          ":base_objc",
+          ":helpers_objc",
+          "../rtc_base:checks",
+          "../rtc_base/synchronization:mutex",
+        ]
+
+        absl_deps = [ "//third_party/abseil-cpp/absl/base:core_headers" ]
+      }
+
+      rtc_source_set("network_monitor_observer") {
+        visibility = [ ":*" ]
+
+        sources = [ "objc/native/src/network_monitor_observer.h" ]
+
+        deps = [
+          "../rtc_base:network_constants",
+          "../rtc_base:stringutils",
+          "../rtc_base:threading",
+        ]
+
+        absl_deps = [ "//third_party/abseil-cpp/absl/strings" ]
+      }
+
+      rtc_library("network_monitor_objc") {
+        visibility = [ "*" ]
+
+        sources = [
+          "objc/components/network/RTCNetworkMonitor+Private.h",
+          "objc/components/network/RTCNetworkMonitor.h",
+          "objc/components/network/RTCNetworkMonitor.mm",
+        ]
+
+        configs += [ ":used_from_extension" ]
+
+        frameworks = [ "Network.framework" ]
+
+        deps = [
+          ":base_objc",
+          ":helpers_objc",
+          ":network_monitor_observer",
+          "../rtc_base:stringutils",
+          "../rtc_base/system:gcd_helpers",
+        ]
+      }
+    }
+
+    rtc_source_set("audio_device_api_objc") {
+      visibility = [ "*" ]
+
+      sources = [ "objc/components/audio/RTCAudioDevice.h" ]
+
+      public_configs = [ ":common_config_objc" ]
+
+      frameworks = [ "AudioToolbox.framework" ]
+
+      deps = [ ":base_objc" ]
+    }
+
+    rtc_library("audio_device_objc") {
+      visibility = [ "*" ]
+      allow_poison = [ "default_task_queue" ]
+      sources = [
+        "objc/native/src/objc_audio_device.h",
+        "objc/native/src/objc_audio_device.mm",
+        "objc/native/src/objc_audio_device_delegate.h",
+        "objc/native/src/objc_audio_device_delegate.mm",
+      ]
+
+      deps = [
+        ":audio_device_api_objc",
+        "../api:array_view",
+        "../api:make_ref_counted",
+        "../api:refcountedbase",
+        "../api:scoped_refptr",
+        "../api:sequence_checker",
+        "../api/task_queue",
+        "../api/task_queue:default_task_queue_factory",
+        "../modules/audio_device:audio_device_api",
+        "../modules/audio_device:audio_device_buffer",
+        "../rtc_base:buffer",
+        "../rtc_base:checks",
+        "../rtc_base:logging",
+        "../rtc_base:safe_minmax",
+        "../rtc_base:threading",
+        "../rtc_base:timeutils",
+      ]
+    }
+
+    rtc_library("objc_audio_device_module") {
+      visibility = [ "*" ]
+      allow_poison = [ "default_task_queue" ]
+      sources = [
+        "objc/native/api/objc_audio_device_module.h",
+        "objc/native/api/objc_audio_device_module.mm",
+      ]
+
+      deps = [
+        ":audio_device_api_objc",
+        ":audio_device_objc",
+        "../api:make_ref_counted",
+        "../modules/audio_device:audio_device_api",
+        "../rtc_base:logging",
+      ]
+    }
+
+    if (!build_with_mozilla) {
+    rtc_library("videosource_objc") {
+      sources = [
+        "objc/api/peerconnection/RTCVideoSource+Private.h",
+        "objc/api/peerconnection/RTCVideoSource.h",
+        "objc/api/peerconnection/RTCVideoSource.mm",
+      ]
+
+      deps = [
+        ":base_objc",
+        ":mediasource_objc",
+        ":native_video",
+        ":videoframebuffer_objc",
+        "../api:libjingle_peerconnection_api",
+        "../api:media_stream_interface",
+        "../api/video:video_frame",
+        "../api/video:video_rtp_headers",
+        "../common_video",
+        "../media:rtc_media_base",
+        "../pc:video_track_source_proxy",
+        "../rtc_base:checks",
+        "../rtc_base:threading",
+        "//third_party/libyuv",
+      ]
+
+      configs += [
+        "..:common_objc",
+        ":used_from_extension",
+      ]
+    }
+    }
+
+    rtc_library("videoframebuffer_objc") {
+      visibility = [ "*" ]
+      sources = [
+        "objc/api/video_frame_buffer/RTCNativeI420Buffer+Private.h",
+        "objc/api/video_frame_buffer/RTCNativeI420Buffer.h",
+        "objc/api/video_frame_buffer/RTCNativeI420Buffer.mm",
+        "objc/api/video_frame_buffer/RTCNativeMutableI420Buffer.h",
+        "objc/api/video_frame_buffer/RTCNativeMutableI420Buffer.mm",
+        "objc/components/video_frame_buffer/RTCCVPixelBuffer.h",
+        "objc/components/video_frame_buffer/RTCCVPixelBuffer.mm",
+      ]
+      deps = [
+        ":base_objc",
+        "../rtc_base:logging",
+        "//api/video:video_frame",
+        "//api/video:video_rtp_headers",
+        "//common_video",
+        "//rtc_base:checks",
+        "//third_party/libyuv",
+      ]
+      configs += [
+        "..:common_objc",
+        ":used_from_extension",
+      ]
+      frameworks = [
+        "VideoToolbox.framework",
+        "CoreGraphics.framework",
+        "CoreVideo.framework",
+      ]
+    }
+
+    if (!build_with_mozilla) {
+    rtc_library("opengl_objc") {
+      sources = [
+        "objc/components/renderer/opengl/RTCDefaultShader.h",
+        "objc/components/renderer/opengl/RTCDefaultShader.mm",
+        "objc/components/renderer/opengl/RTCI420TextureCache.h",
+        "objc/components/renderer/opengl/RTCI420TextureCache.mm",
+        "objc/components/renderer/opengl/RTCOpenGLDefines.h",
+        "objc/components/renderer/opengl/RTCShader.h",
+        "objc/components/renderer/opengl/RTCShader.mm",
+        "objc/components/renderer/opengl/RTCVideoViewShading.h",
+      ]
+      frameworks = [ "CoreVideo.framework" ]
+      if (is_ios) {
+        sources += [
+          "objc/components/renderer/opengl/RTCNV12TextureCache.h",
+          "objc/components/renderer/opengl/RTCNV12TextureCache.m",
+        ]
+        frameworks += [
+          "GLKit.framework",
+          "OpenGLES.framework",
+          "QuartzCore.framework",
+        ]
+      } else if (is_mac) {
+        frameworks += [
+          "CoreMedia.framework",
+          "OpenGL.framework",
+        ]
+      }
+
+      # TODO(bugs.webrtc.org/12937): Remove OpenGL deprecation warning
+      # workaround.
+      defines = [ "GLES_SILENCE_DEPRECATION" ]
+
+      deps = [
+        ":base_objc",
+        ":helpers_objc",
+        ":mediaconstraints_objc",
+        ":native_video",
+        ":videoframebuffer_objc",
+        ":videosource_objc",
+        "../api:libjingle_peerconnection_api",
+        "../api/video:video_frame",
+        "../api/video:video_rtp_headers",
+        "../common_video",
+        "../media:rtc_media_base",
+        "../rtc_base:checks",
+        "../rtc_base:logging",
+      ]
+      absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
+
+      configs += [
+        "..:common_objc",
+        ":used_from_extension",
+      ]
+    }
+
+    rtc_library("opengl_ui_objc") {
+      visibility = [ "*" ]
+      allow_poison = [
+        "audio_codecs",  # TODO(bugs.webrtc.org/8396): Remove.
+        "default_task_queue",
+      ]
+      if (is_ios) {
+        sources = [
+          "objc/components/renderer/opengl/RTCDisplayLinkTimer.h",
+          "objc/components/renderer/opengl/RTCDisplayLinkTimer.m",
+          "objc/components/renderer/opengl/RTCEAGLVideoView.h",
+          "objc/components/renderer/opengl/RTCEAGLVideoView.m",
+        ]
+
+        # TODO(bugs.webrtc.org/12937): Remove OpenGL deprecation warning
+        # workaround.
+        defines = [ "GLES_SILENCE_DEPRECATION" ]
+      }
+      if (is_mac) {
+        sources = [
+          "objc/components/renderer/opengl/RTCNSGLVideoView.h",
+          "objc/components/renderer/opengl/RTCNSGLVideoView.m",
+        ]
+      }
+      configs += [ "..:common_objc" ]
+      deps = [
+        ":base_objc",
+        ":helpers_objc",
+        ":metal_objc",
+        ":opengl_objc",
+        ":videocapture_objc",
+        ":videoframebuffer_objc",
+      ]
+    }
+
+    rtc_library("metal_objc") {
+      visibility = [ "*" ]
+      allow_poison = [
+        "audio_codecs",  # TODO(bugs.webrtc.org/8396): Remove.
+        "default_task_queue",
+      ]
+      sources = [
+        "objc/components/renderer/metal/RTCMTLI420Renderer.h",
+        "objc/components/renderer/metal/RTCMTLI420Renderer.mm",
+        "objc/components/renderer/metal/RTCMTLNV12Renderer.h",
+        "objc/components/renderer/metal/RTCMTLNV12Renderer.mm",
+        "objc/components/renderer/metal/RTCMTLRGBRenderer.h",
+        "objc/components/renderer/metal/RTCMTLRGBRenderer.mm",
+        "objc/components/renderer/metal/RTCMTLRenderer+Private.h",
+        "objc/components/renderer/metal/RTCMTLRenderer.h",
+        "objc/components/renderer/metal/RTCMTLRenderer.mm",
+      ]
+      frameworks = [
+        "CoreVideo.framework",
+        "Metal.framework",
+        "MetalKit.framework",
+      ]
+      if (is_ios) {
+        sources += [
+          "objc/components/renderer/metal/RTCMTLVideoView.h",
+          "objc/components/renderer/metal/RTCMTLVideoView.m",
+        ]
+      }
+      if (is_mac) {
+        sources += [
+          "objc/components/renderer/metal/RTCMTLNSVideoView.h",
+          "objc/components/renderer/metal/RTCMTLNSVideoView.m",
+        ]
+        frameworks += [ "AppKit.framework" ]
+      }
+      deps = [
+        ":base_objc",
+        ":peerconnectionfactory_base_objc",
+        ":videoframebuffer_objc",
+        "../api/video:video_frame",
+        "../api/video:video_rtp_headers",
+        "../rtc_base:checks",
+      ]
+      configs += [ "..:common_objc" ]
+      public_configs = [ ":common_config_objc" ]
+    }
+
+    # TODO(bugs.webrtc.org/9627): Remove this target.
+    rtc_library("videocapturebase_objc") {
+      visibility = [ "*" ]
+      sources = [ "objc/helpers/noop.mm" ]
+
+      configs += [ "..:common_objc" ]
+
+      public_configs = [ ":common_config_objc" ]
+
+      deps = [
+        ":base_objc",
+        ":videoframebuffer_objc",
+      ]
+    }
+    }
+
+    rtc_library("videocapture_objc") {
+      visibility = [ "*" ]
+      allow_poison = [ "audio_codecs" ]  # TODO(bugs.webrtc.org/8396): Remove.
+      sources = [
+        "objc/components/capturer/RTCCameraVideoCapturer.h",
+        "objc/components/capturer/RTCCameraVideoCapturer.m",
+        "objc/components/capturer/RTCFileVideoCapturer.h",
+        "objc/components/capturer/RTCFileVideoCapturer.m",
+      ]
+      frameworks = [
+        "AVFoundation.framework",
+        "CoreVideo.framework",
+        "QuartzCore.framework",
+      ]
+
+      configs += [ "..:common_objc" ]
+
+      public_configs = [ ":common_config_objc" ]
+
+      deps = [
+        ":base_objc",
+        ":helpers_objc",
+        ":videoframebuffer_objc",
+        "../rtc_base/system:gcd_helpers",
+      ]
+    }
+
+    if (!build_with_mozilla) {
+    rtc_library("videocodec_objc") {
+      visibility = [ "*" ]
+      configs += [ "..:no_global_constructors" ]
+      sources = [
+        "objc/components/video_codec/RTCCodecSpecificInfoH264+Private.h",
+        "objc/components/video_codec/RTCCodecSpecificInfoH264.h",
+        "objc/components/video_codec/RTCCodecSpecificInfoH264.mm",
+        "objc/components/video_codec/RTCH264ProfileLevelId.h",
+        "objc/components/video_codec/RTCH264ProfileLevelId.mm",
+      ]
+      if (is_ios) {
+        sources += [
+          "objc/components/video_codec/UIDevice+H264Profile.h",
+          "objc/components/video_codec/UIDevice+H264Profile.mm",
+        ]
+      }
+
+      public_configs = [ ":common_config_objc" ]
+      deps = [
+        ":base_objc",
+        ":helpers_objc",
+        "../api/video_codecs:video_codecs_api",
+        "../common_video",
+        "../media:media_constants",
+        "../media:rtc_media_base",
+        "../modules/video_coding:video_codec_interface",
+        "../rtc_base:checks",
+      ]
+    }
+
+    rtc_library("default_codec_factory_objc") {
+      sources = [
+        "objc/components/video_codec/RTCDefaultVideoDecoderFactory.h",
+        "objc/components/video_codec/RTCDefaultVideoDecoderFactory.m",
+        "objc/components/video_codec/RTCDefaultVideoEncoderFactory.h",
+        "objc/components/video_codec/RTCDefaultVideoEncoderFactory.m",
+      ]
+
+      deps = [
+        ":base_objc",
+        ":native_video",
+        ":videocodec_objc",
+        ":videotoolbox_objc",
+        ":vp8",
+        ":vp9",
+        ":vpx_codec_constants",
+      ]
+
+      defines = []
+      if (enable_libaom) {
+        defines += [ "RTC_USE_LIBAOM_AV1_ENCODER" ]
+        deps += [ ":libaom_av1_encoder" ]
+      }
+
+      if (rtc_include_dav1d_in_internal_decoder_factory) {
+        deps += [ ":dav1d_decoder" ]
+      }
+    }
+
+    rtc_library("vpx_codec_constants") {
+      configs += [ "..:no_global_constructors" ]
+      sources = [
+        "objc/api/video_codec/RTCVideoCodecConstants.h",
+        "objc/api/video_codec/RTCVideoCodecConstants.mm",
+      ]
+
+      deps = [
+        ":base_objc",
+        "../media:media_constants",
+        "../media:rtc_media_base",
+      ]
+    }
+
+    rtc_library("vp8") {
+      visibility = [ "*" ]
+      allow_poison = [ "software_video_codecs" ]
+      sources = [
+        "objc/api/video_codec/RTCVideoDecoderVP8.h",
+        "objc/api/video_codec/RTCVideoDecoderVP8.mm",
+        "objc/api/video_codec/RTCVideoEncoderVP8.h",
+        "objc/api/video_codec/RTCVideoEncoderVP8.mm",
+      ]
+
+      deps = [
+        ":base_objc",
+        ":wrapped_native_codec_objc",
+        "../modules/video_coding:webrtc_vp8",
+      ]
+    }
+
+    rtc_library("vp9") {
+      visibility = [ "*" ]
+      allow_poison = [ "software_video_codecs" ]
+      sources = [
+        "objc/api/video_codec/RTCVideoDecoderVP9.h",
+        "objc/api/video_codec/RTCVideoDecoderVP9.mm",
+        "objc/api/video_codec/RTCVideoEncoderVP9.h",
+        "objc/api/video_codec/RTCVideoEncoderVP9.mm",
+      ]
+
+      deps = [
+        ":base_objc",
+        ":wrapped_native_codec_objc",
+        "../media:rtc_media_base",
+        "../modules/video_coding:webrtc_vp9",
+      ]
+    }
+
+    rtc_library("dav1d_decoder") {
+      visibility = [ "*" ]
+      allow_poison = [ "software_video_codecs" ]
+      sources = [
+        "objc/api/video_codec/RTCVideoDecoderAV1.h",
+        "objc/api/video_codec/RTCVideoDecoderAV1.mm",
+      ]
+
+      deps = [
+        ":base_objc",
+        ":wrapped_native_codec_objc",
+        "../media:rtc_media_base",
+        "../modules/video_coding/codecs/av1:dav1d_decoder",
+      ]
+    }
+
+    rtc_library("libaom_av1_encoder") {
+      visibility = [ "*" ]
+      allow_poison = [ "software_video_codecs" ]
+      sources = [
+        "objc/api/video_codec/RTCVideoEncoderAV1.h",
+        "objc/api/video_codec/RTCVideoEncoderAV1.mm",
+      ]
+
+      deps = [
+        ":base_objc",
+        ":wrapped_native_codec_objc",
+        "../media:rtc_media_base",
+        "../modules/video_coding/codecs/av1:libaom_av1_encoder",
+      ]
+    }
+
+    rtc_library("mediaconstraints_objc") {
+      configs += [ "..:no_global_constructors" ]
+      sources = [
+        "objc/api/peerconnection/RTCMediaConstraints+Private.h",
+        "objc/api/peerconnection/RTCMediaConstraints.h",
+        "objc/api/peerconnection/RTCMediaConstraints.mm",
+      ]
+
+      public_configs = [ ":common_config_objc" ]
+      deps = [
+        ":base_objc",
+        ":helpers_objc",
+        ":media_constraints",
+      ]
+    }
+
+    # TODO(bugs.webrtc.org/9627): Remove, targets should depend on base_objc.
+    rtc_library("videorenderer_objc") {
+      visibility = [ "*" ]
+      sources = [ "objc/helpers/noop.mm" ]
+
+      configs += [ "..:common_objc" ]
+      public_configs = [ ":common_config_objc" ]
+
+      deps = [ ":base_objc" ]
+    }
+
+    rtc_library("videorendereradapter_objc") {
+      visibility = [ "*" ]
+      allow_poison = [ "audio_codecs" ]  # TODO(bugs.webrtc.org/8396): Remove.
+      sources = [
+        "objc/api/RTCVideoRendererAdapter+Private.h",
+        "objc/api/RTCVideoRendererAdapter.h",
+        "objc/api/RTCVideoRendererAdapter.mm",
+      ]
+
+      configs += [ "..:common_objc" ]
+      public_configs = [ ":common_config_objc" ]
+
+      deps = [
+        ":base_objc",
+        ":native_api",
+        ":videoframebuffer_objc",
+        "../api:libjingle_peerconnection_api",
+        "../api:media_stream_interface",
+      ]
+    }
+
+    rtc_library("mediasource_objc") {
+      sources = [
+        "objc/api/peerconnection/RTCMediaSource+Private.h",
+        "objc/api/peerconnection/RTCMediaSource.h",
+        "objc/api/peerconnection/RTCMediaSource.mm",
+      ]
+
+      configs += [
+        "..:common_objc",
+        ":used_from_extension",
+      ]
+      public_configs = [ ":common_config_objc" ]
+
+      deps = [
+        ":base_objc",
+        "../api:media_stream_interface",
+        "../rtc_base:checks",
+      ]
+    }
+
+    rtc_library("base_native_additions_objc") {
+      sources = [
+        "objc/api/peerconnection/RTCEncodedImage+Private.h",
+        "objc/api/peerconnection/RTCEncodedImage+Private.mm",
+        "objc/api/peerconnection/RTCVideoCodecInfo+Private.h",
+        "objc/api/peerconnection/RTCVideoCodecInfo+Private.mm",
+        "objc/api/peerconnection/RTCVideoEncoderSettings+Private.h",
+        "objc/api/peerconnection/RTCVideoEncoderSettings+Private.mm",
+      ]
+
+      configs += [ "..:common_objc" ]
+
+      public_configs = [ ":common_config_objc" ]
+
+      deps = [
+        ":base_objc",
+        ":helpers_objc",
+        "../api/video:encoded_image",
+        "../api/video_codecs:video_codecs_api",
+        "../modules/video_coding:video_codec_interface",
+        "../rtc_base:refcount",
+        "../rtc_base:safe_conversions",
+      ]
+    }
+
+    rtc_library("peerconnectionfactory_base_objc") {
+      visibility = [ "*" ]
+      allow_poison = [
+        "audio_codecs",  # TODO(bugs.webrtc.org/8396): Remove.
+        "default_task_queue",
+      ]
+      configs += [ "..:no_global_constructors" ]
+      sources = [
+        "objc/api/peerconnection/RTCAudioSource+Private.h",
+        "objc/api/peerconnection/RTCAudioSource.h",
+        "objc/api/peerconnection/RTCAudioSource.mm",
+        "objc/api/peerconnection/RTCAudioTrack+Private.h",
+        "objc/api/peerconnection/RTCAudioTrack.h",
+        "objc/api/peerconnection/RTCAudioTrack.mm",
+        "objc/api/peerconnection/RTCCertificate.h",
+        "objc/api/peerconnection/RTCCertificate.mm",
+        "objc/api/peerconnection/RTCConfiguration+Native.h",
+        "objc/api/peerconnection/RTCConfiguration+Private.h",
+        "objc/api/peerconnection/RTCConfiguration.h",
+        "objc/api/peerconnection/RTCConfiguration.mm",
+        "objc/api/peerconnection/RTCCryptoOptions.h",
+        "objc/api/peerconnection/RTCCryptoOptions.mm",
+        "objc/api/peerconnection/RTCDataChannel+Private.h",
+        "objc/api/peerconnection/RTCDataChannel.h",
+        "objc/api/peerconnection/RTCDataChannel.mm",
+        "objc/api/peerconnection/RTCDataChannelConfiguration+Private.h",
+        "objc/api/peerconnection/RTCDataChannelConfiguration.h",
+        "objc/api/peerconnection/RTCDataChannelConfiguration.mm",
+        "objc/api/peerconnection/RTCDtmfSender+Private.h",
+        "objc/api/peerconnection/RTCDtmfSender.h",
+        "objc/api/peerconnection/RTCDtmfSender.mm",
+        "objc/api/peerconnection/RTCFieldTrials.h",
+        "objc/api/peerconnection/RTCFieldTrials.mm",
+        "objc/api/peerconnection/RTCIceCandidate+Private.h",
+        "objc/api/peerconnection/RTCIceCandidate.h",
+        "objc/api/peerconnection/RTCIceCandidate.mm",
+        "objc/api/peerconnection/RTCIceCandidateErrorEvent+Private.h",
+        "objc/api/peerconnection/RTCIceCandidateErrorEvent.h",
+        "objc/api/peerconnection/RTCIceCandidateErrorEvent.mm",
+        "objc/api/peerconnection/RTCIceServer+Private.h",
+        "objc/api/peerconnection/RTCIceServer.h",
+        "objc/api/peerconnection/RTCIceServer.mm",
+        "objc/api/peerconnection/RTCLegacyStatsReport+Private.h",
+        "objc/api/peerconnection/RTCLegacyStatsReport.h",
+        "objc/api/peerconnection/RTCLegacyStatsReport.mm",
+        "objc/api/peerconnection/RTCMediaStream+Private.h",
+        "objc/api/peerconnection/RTCMediaStream.h",
+        "objc/api/peerconnection/RTCMediaStream.mm",
+        "objc/api/peerconnection/RTCMediaStreamTrack+Private.h",
+        "objc/api/peerconnection/RTCMediaStreamTrack.h",
+        "objc/api/peerconnection/RTCMediaStreamTrack.mm",
+        "objc/api/peerconnection/RTCMetrics.h",
+        "objc/api/peerconnection/RTCMetrics.mm",
+        "objc/api/peerconnection/RTCMetricsSampleInfo+Private.h",
+        "objc/api/peerconnection/RTCMetricsSampleInfo.h",
+        "objc/api/peerconnection/RTCMetricsSampleInfo.mm",
+        "objc/api/peerconnection/RTCPeerConnection+DataChannel.mm",
+        "objc/api/peerconnection/RTCPeerConnection+Private.h",
+        "objc/api/peerconnection/RTCPeerConnection+Stats.mm",
+        "objc/api/peerconnection/RTCPeerConnection.h",
+        "objc/api/peerconnection/RTCPeerConnection.mm",
+        "objc/api/peerconnection/RTCPeerConnectionFactory+Native.h",
+        "objc/api/peerconnection/RTCPeerConnectionFactory+Private.h",
+        "objc/api/peerconnection/RTCPeerConnectionFactory.h",
+        "objc/api/peerconnection/RTCPeerConnectionFactory.mm",
+        "objc/api/peerconnection/RTCPeerConnectionFactoryBuilder+DefaultComponents.h",
+        "objc/api/peerconnection/RTCPeerConnectionFactoryBuilder+DefaultComponents.mm",
+        "objc/api/peerconnection/RTCPeerConnectionFactoryBuilder.h",
+        "objc/api/peerconnection/RTCPeerConnectionFactoryBuilder.mm",
+        "objc/api/peerconnection/RTCPeerConnectionFactoryOptions+Private.h",
+        "objc/api/peerconnection/RTCPeerConnectionFactoryOptions.h",
+        "objc/api/peerconnection/RTCPeerConnectionFactoryOptions.mm",
+        "objc/api/peerconnection/RTCRtcpParameters+Private.h",
+        "objc/api/peerconnection/RTCRtcpParameters.h",
+        "objc/api/peerconnection/RTCRtcpParameters.mm",
+        "objc/api/peerconnection/RTCRtpCodecParameters+Private.h",
+        "objc/api/peerconnection/RTCRtpCodecParameters.h",
+        "objc/api/peerconnection/RTCRtpCodecParameters.mm",
+        "objc/api/peerconnection/RTCRtpEncodingParameters+Private.h",
+        "objc/api/peerconnection/RTCRtpEncodingParameters.h",
+        "objc/api/peerconnection/RTCRtpEncodingParameters.mm",
+        "objc/api/peerconnection/RTCRtpHeaderExtension+Private.h",
+        "objc/api/peerconnection/RTCRtpHeaderExtension.h",
+        "objc/api/peerconnection/RTCRtpHeaderExtension.mm",
+        "objc/api/peerconnection/RTCRtpParameters+Private.h",
+        "objc/api/peerconnection/RTCRtpParameters.h",
+        "objc/api/peerconnection/RTCRtpParameters.mm",
+        "objc/api/peerconnection/RTCRtpReceiver+Native.h",
+        "objc/api/peerconnection/RTCRtpReceiver+Private.h",
+        "objc/api/peerconnection/RTCRtpReceiver.h",
+        "objc/api/peerconnection/RTCRtpReceiver.mm",
+        "objc/api/peerconnection/RTCRtpSender+Native.h",
+        "objc/api/peerconnection/RTCRtpSender+Private.h",
+        "objc/api/peerconnection/RTCRtpSender.h",
+        "objc/api/peerconnection/RTCRtpSender.mm",
+        "objc/api/peerconnection/RTCRtpTransceiver+Private.h",
+        "objc/api/peerconnection/RTCRtpTransceiver.h",
+        "objc/api/peerconnection/RTCRtpTransceiver.mm",
+        "objc/api/peerconnection/RTCSSLAdapter.h",
+        "objc/api/peerconnection/RTCSSLAdapter.mm",
+        "objc/api/peerconnection/RTCSessionDescription+Private.h",
+        "objc/api/peerconnection/RTCSessionDescription.h",
+        "objc/api/peerconnection/RTCSessionDescription.mm",
+        "objc/api/peerconnection/RTCStatisticsReport+Private.h",
+        "objc/api/peerconnection/RTCStatisticsReport.h",
+        "objc/api/peerconnection/RTCStatisticsReport.mm",
+        "objc/api/peerconnection/RTCTracing.h",
+        "objc/api/peerconnection/RTCTracing.mm",
+        "objc/api/peerconnection/RTCVideoTrack+Private.h",
+        "objc/api/peerconnection/RTCVideoTrack.h",
+        "objc/api/peerconnection/RTCVideoTrack.mm",
+      ]
+
+      configs += [
+        "..:common_objc",
+        ":used_from_extension",
+      ]
+      public_configs = [ ":common_config_objc" ]
+
+      deps = [
+        ":audio_device_api_objc",
+        ":base_native_additions_objc",
+        ":base_objc",
+        ":file_logger_objc",
+        ":helpers_objc",
+        ":mediaconstraints_objc",
+        ":mediasource_objc",
+        ":native_api",
+        ":native_video",
+        ":objc_audio_device_module",
+        ":videoframebuffer_objc",
+        ":videorendereradapter_objc",
+        ":videosource_objc",
+        ":videotoolbox_objc",
+        "../api:dtmf_sender_interface",
+        "../api:libjingle_peerconnection_api",
+        "../api:media_stream_interface",
+        "../api:rtc_event_log_output_file",
+        "../api:rtc_stats_api",
+        "../api:rtp_parameters",
+        "../api:rtp_sender_interface",
+        "../api:scoped_refptr",
+        "../api/audio_codecs:audio_codecs_api",
+        "../api/audio_codecs:builtin_audio_decoder_factory",
+        "../api/audio_codecs:builtin_audio_encoder_factory",
+        "../api/crypto:frame_decryptor_interface",
+        "../api/crypto:frame_encryptor_interface",
+        "../api/rtc_event_log:rtc_event_log_factory",
+        "../api/task_queue:default_task_queue_factory",
+        "../api/transport:field_trial_based_config",
+        "../api/video:video_frame",
+        "../api/video:video_rtp_headers",
+        "../api/video_codecs:video_codecs_api",
+        "../common_video",
+        "../media:media_constants",
+        "../media:rtc_audio_video",
+        "../media:rtc_media_base",
+        "../modules/audio_device:audio_device_api",
+        "../modules/audio_processing",
+        "../modules/audio_processing:api",
+        "../modules/video_coding:video_codec_interface",
+        "../pc:peer_connection_factory",
+        "../pc:webrtc_sdp",
+        "../rtc_base:checks",
+        "../rtc_base:event_tracer",
+        "../rtc_base:logging",
+        "../rtc_base:network_constants",
+        "../rtc_base:rtc_certificate_generator",
+        "../rtc_base:safe_conversions",
+        "../rtc_base:ssl",
+        "../rtc_base:stringutils",
+        "../rtc_base:threading",
+        "../rtc_base:timeutils",
+        "../stats:rtc_stats",
+        "../system_wrappers:field_trial",
+        "../system_wrappers:metrics",
+      ]
+
+      if (is_ios) {
+        deps += [ ":native_api_audio_device_module" ]
+      }
+    }
+
+    if (rtc_include_tests) {
+      if (is_ios) {
+        rtc_library("sdk_unittests_sources") {
+          testonly = true
+          include_dirs = [ "objc/" ]
+
+          sources = [
+            "objc/unittests/ObjCVideoTrackSource_xctest.mm",
+            "objc/unittests/RTCAudioSessionTest.mm",
+            "objc/unittests/RTCCVPixelBuffer_xctest.mm",
+            "objc/unittests/RTCCallbackLogger_xctest.m",
+            "objc/unittests/RTCCameraVideoCapturerTests.mm",
+            "objc/unittests/RTCCertificateTest.mm",
+            "objc/unittests/RTCConfigurationTest.mm",
+            "objc/unittests/RTCDataChannelConfigurationTest.mm",
+            "objc/unittests/RTCEncodedImage_xctest.mm",
+            "objc/unittests/RTCFileVideoCapturer_xctest.mm",
+            "objc/unittests/RTCH264ProfileLevelId_xctest.m",
+            "objc/unittests/RTCIceCandidateTest.mm",
+            "objc/unittests/RTCIceServerTest.mm",
+            "objc/unittests/RTCMTLVideoView_xctest.m",
+            "objc/unittests/RTCMediaConstraintsTest.mm",
+            "objc/unittests/RTCNV12TextureCache_xctest.m",
+            "objc/unittests/RTCPeerConnectionFactoryBuilderTest.mm",
+            "objc/unittests/RTCPeerConnectionFactory_xctest.m",
+            "objc/unittests/RTCPeerConnectionTest.mm",
+            "objc/unittests/RTCSessionDescriptionTest.mm",
+            "objc/unittests/RTCTracingTest.mm",
+            "objc/unittests/frame_buffer_helpers.h",
+            "objc/unittests/frame_buffer_helpers.mm",
+            "objc/unittests/nalu_rewriter_xctest.mm",
+            "objc/unittests/objc_video_decoder_factory_tests.mm",
+            "objc/unittests/objc_video_encoder_factory_tests.mm",
+            "objc/unittests/scoped_cftyperef_tests.mm",
+          ]
+
+          # TODO(bugs.webrtc.org/12937): Remove OpenGL deprecation warning
+          # workaround.
+          defines = [ "GLES_SILENCE_DEPRECATION" ]
+
+          # TODO(peterhanspers): Reenable these tests on simulator.
+          # See bugs.webrtc.org/7812
+          if (target_environment != "simulator") {
+            sources += [
+              "objc/unittests/RTCAudioDeviceModule_xctest.mm",
+              "objc/unittests/RTCAudioDevice_xctest.mm",
+            ]
+          }
+
+          deps = [
+            ":audio_device",
+            ":audio_session_objc",
+            ":base_native_additions_objc",
+            ":base_objc",
+            ":callback_logger_objc",
+            ":framework_objc",
+            ":helpers_objc",
+            ":mediaconstraints_objc",
+            ":metal_objc",
+            ":native_api",
+            ":native_api_audio_device_module",
+            ":native_video",
+            ":peerconnectionfactory_base_objc",
+            ":video_toolbox_cc",
+            ":videocapture_objc",
+            ":videocodec_objc",
+            ":videoframebuffer_objc",
+            ":videosource_objc",
+            ":videotoolbox_objc",
+            "../api:scoped_refptr",
+            "../api/audio_codecs:builtin_audio_decoder_factory",
+            "../api/audio_codecs:builtin_audio_encoder_factory",
+            "../api/task_queue:default_task_queue_factory",
+            "../api/video:video_frame",
+            "../api/video_codecs:video_codecs_api",
+            "../common_video",
+            "../media:codec",
+            "../media:rtc_media_base",
+            "../media:rtc_media_tests_utils",
+            "../modules/audio_device:audio_device_api",
+            "../modules/audio_processing:api",
+            "../modules/video_coding:video_codec_interface",
+            "../rtc_base:gunit_helpers",
+            "../rtc_base:macromagic",
+            "../rtc_base:refcount",
+            "../rtc_base:rtc_event",
+            "../rtc_base/system:unused",
+            "../system_wrappers",
+            "//third_party/libyuv",
+          ]
+
+          if (rtc_ios_macos_use_opengl_rendering) {
+            deps += [ ":opengl_objc" ]
+          }
+
+          public_deps = [
+            "//build/config/ios:xctest",
+            "//third_party/ocmock",
+          ]
+        }
+
+        bundle_data("sdk_unittests_bundle_data") {
+          sources = [
+            "objc/unittests/audio_short16.pcm",
+            "objc/unittests/audio_short44.pcm",
+            "objc/unittests/audio_short48.pcm",
+
+            # Sample video taken from https://media.xiph.org/video/derf/
+            "objc/unittests/foreman.mp4",
+          ]
+          outputs = [ "{{bundle_resources_dir}}/{{source_file_part}}" ]
+        }
+
+        # These tests use static linking.
+        rtc_test("sdk_unittests") {
+          is_xctest = true
+          info_plist = "//test/ios/Info.plist"
+          sources = [ "objc/unittests/main.mm" ]
+
+          extra_substitutions = [ "GTEST_BUNDLE_ID_SUFFIX=generic-unit-test" ]
+          deps = [
+            ":peerconnectionfactory_base_objc",
+            ":sdk_unittests_bundle_data",
+            ":sdk_unittests_sources",
+            "../rtc_base:threading",
+            "//test:test_support",
+          ]
+          ldflags = [ "-all_load" ]
+        }
+
+        # These tests link to the framework.
+        rtc_test("sdk_framework_unittests") {
+          is_xctest = true
+          info_plist = "//test/ios/Info.plist"
+          sources = [
+            "objc/unittests/RTCDoNotPutCPlusPlusInFrameworkHeaders_xctest.m",
+            "objc/unittests/main.mm",
+          ]
+
+          extra_substitutions = [ "GTEST_BUNDLE_ID_SUFFIX=generic-unit-test" ]
+          deps = [
+            ":framework_objc+link",
+            ":ios_framework_bundle",
+            "../rtc_base:threading",
+            "//test:test_support",
+          ]
+        }
+      }
+    }
+
+    if (is_ios) {
+      apple_framework_bundle_with_umbrella_header("framework_objc") {
+        info_plist = "objc/Info.plist"
+        output_name = "WebRTC"
+
+        common_objc_headers = [
+          "objc/base/RTCCodecSpecificInfo.h",
+          "objc/base/RTCEncodedImage.h",
+          "objc/base/RTCI420Buffer.h",
+          "objc/base/RTCLogging.h",
+          "objc/base/RTCMacros.h",
+          "objc/base/RTCMutableI420Buffer.h",
+          "objc/base/RTCMutableYUVPlanarBuffer.h",
+          "objc/base/RTCSSLCertificateVerifier.h",
+          "objc/base/RTCVideoCapturer.h",
+          "objc/base/RTCVideoCodecInfo.h",
+          "objc/base/RTCVideoDecoder.h",
+          "objc/base/RTCVideoDecoderFactory.h",
+          "objc/base/RTCVideoEncoder.h",
+          "objc/base/RTCVideoEncoderFactory.h",
+          "objc/base/RTCVideoEncoderQpThresholds.h",
+          "objc/base/RTCVideoEncoderSettings.h",
+          "objc/base/RTCVideoFrame.h",
+          "objc/base/RTCVideoFrameBuffer.h",
+          "objc/base/RTCVideoRenderer.h",
+          "objc/base/RTCYUVPlanarBuffer.h",
+          "objc/components/audio/RTCAudioDevice.h",
+          "objc/components/audio/RTCAudioSession.h",
+          "objc/components/audio/RTCAudioSessionConfiguration.h",
+          "objc/components/capturer/RTCCameraVideoCapturer.h",
+          "objc/components/capturer/RTCFileVideoCapturer.h",
+          "objc/components/network/RTCNetworkMonitor.h",
+          "objc/components/renderer/metal/RTCMTLVideoView.h",
+          "objc/components/renderer/opengl/RTCEAGLVideoView.h",
+          "objc/components/renderer/opengl/RTCVideoViewShading.h",
+          "objc/components/video_codec/RTCCodecSpecificInfoH264.h",
+          "objc/components/video_codec/RTCDefaultVideoDecoderFactory.h",
+          "objc/components/video_codec/RTCDefaultVideoEncoderFactory.h",
+          "objc/components/video_codec/RTCH264ProfileLevelId.h",
+          "objc/components/video_codec/RTCVideoDecoderFactoryH264.h",
+          "objc/components/video_codec/RTCVideoDecoderH264.h",
+          "objc/components/video_codec/RTCVideoEncoderFactoryH264.h",
+          "objc/components/video_codec/RTCVideoEncoderH264.h",
+          "objc/components/video_frame_buffer/RTCCVPixelBuffer.h",
+          "objc/helpers/RTCCameraPreviewView.h",
+          "objc/helpers/RTCDispatcher.h",
+          "objc/helpers/UIDevice+RTCDevice.h",
+          "objc/api/peerconnection/RTCAudioSource.h",
+          "objc/api/peerconnection/RTCAudioTrack.h",
+          "objc/api/peerconnection/RTCConfiguration.h",
+          "objc/api/peerconnection/RTCDataChannel.h",
+          "objc/api/peerconnection/RTCDataChannelConfiguration.h",
+          "objc/api/peerconnection/RTCFieldTrials.h",
+          "objc/api/peerconnection/RTCIceCandidate.h",
+          "objc/api/peerconnection/RTCIceCandidateErrorEvent.h",
+          "objc/api/peerconnection/RTCIceServer.h",
+          "objc/api/peerconnection/RTCLegacyStatsReport.h",
+          "objc/api/peerconnection/RTCMediaConstraints.h",
+          "objc/api/peerconnection/RTCMediaSource.h",
+          "objc/api/peerconnection/RTCMediaStream.h",
+          "objc/api/peerconnection/RTCMediaStreamTrack.h",
+          "objc/api/peerconnection/RTCMetrics.h",
+          "objc/api/peerconnection/RTCMetricsSampleInfo.h",
+          "objc/api/peerconnection/RTCPeerConnection.h",
+          "objc/api/peerconnection/RTCPeerConnectionFactory.h",
+          "objc/api/peerconnection/RTCPeerConnectionFactoryOptions.h",
+          "objc/api/peerconnection/RTCRtcpParameters.h",
+          "objc/api/peerconnection/RTCRtpCodecParameters.h",
+          "objc/api/peerconnection/RTCRtpEncodingParameters.h",
+          "objc/api/peerconnection/RTCRtpHeaderExtension.h",
+          "objc/api/peerconnection/RTCRtpParameters.h",
+          "objc/api/peerconnection/RTCRtpReceiver.h",
+          "objc/api/peerconnection/RTCRtpSender.h",
+          "objc/api/peerconnection/RTCRtpTransceiver.h",
+          "objc/api/peerconnection/RTCDtmfSender.h",
+          "objc/api/peerconnection/RTCSSLAdapter.h",
+          "objc/api/peerconnection/RTCSessionDescription.h",
+          "objc/api/peerconnection/RTCStatisticsReport.h",
+          "objc/api/peerconnection/RTCTracing.h",
+          "objc/api/peerconnection/RTCCertificate.h",
+          "objc/api/peerconnection/RTCCryptoOptions.h",
+          "objc/api/peerconnection/RTCVideoSource.h",
+          "objc/api/peerconnection/RTCVideoTrack.h",
+          "objc/api/video_codec/RTCVideoCodecConstants.h",
+          "objc/api/video_codec/RTCVideoDecoderVP8.h",
+          "objc/api/video_codec/RTCVideoDecoderVP9.h",
+          "objc/api/video_codec/RTCVideoDecoderAV1.h",
+          "objc/api/video_codec/RTCVideoEncoderVP8.h",
+          "objc/api/video_codec/RTCVideoEncoderVP9.h",
+          "objc/api/video_codec/RTCVideoEncoderAV1.h",
+          "objc/api/video_frame_buffer/RTCNativeI420Buffer.h",
+          "objc/api/video_frame_buffer/RTCNativeMutableI420Buffer.h",
+        ]
+
+        if (!build_with_chromium) {
+          common_objc_headers += [
+            "objc/api/logging/RTCCallbackLogger.h",
+            "objc/api/peerconnection/RTCFileLogger.h",
+          ]
+        }
+
+        sources = common_objc_headers
+        public_headers = common_objc_headers
+
+        ldflags = [
+          "-all_load",
+          "-install_name",
+          "@rpath/$output_name.framework/$output_name",
+        ]
+
+        deps = [
+          ":audio_objc",
+          ":base_objc",
+          ":default_codec_factory_objc",
+          ":metal_objc",
+          ":native_api",
+          ":native_video",
+          ":peerconnectionfactory_base_objc",
+          ":videocapture_objc",
+          ":videocodec_objc",
+          ":videotoolbox_objc",
+        ]
+        if (rtc_ios_macos_use_opengl_rendering) {
+          deps += [ ":opengl_ui_objc" ]
+        }
+        if (!build_with_chromium) {
+          deps += [
+            ":callback_logger_objc",
+            ":file_logger_objc",
+          ]
+        }
+
+        frameworks = [
+          "AVFoundation.framework",
+          "CoreGraphics.framework",
+          "CoreMedia.framework",
+          "Foundation.framework",
+          "UIKit.framework",
+        ]
+
+        configs = [
+          "..:common_objc",
+          ":used_from_extension",
+        ]
+
+        public_configs = [ ":common_config_objc" ]
+      }
+
+      bundle_data("ios_framework_bundle") {
+        deps = [ "../sdk:framework_objc" ]
+        sources = [ "$root_build_dir/WebRTC.framework" ]
+        outputs = [ "{{bundle_resources_dir}}/Frameworks/{{source_file_part}}" ]
+      }
+    }
+
+    if (is_mac) {
+      apple_framework_bundle_with_umbrella_header("mac_framework_objc") {
+        info_plist = "objc/Info.plist"
+        output_name = "WebRTC"
+
+        sources = [
+          "objc/api/peerconnection/RTCAudioSource.h",
+          "objc/api/peerconnection/RTCAudioTrack.h",
+          "objc/api/peerconnection/RTCCertificate.h",
+          "objc/api/peerconnection/RTCConfiguration.h",
+          "objc/api/peerconnection/RTCCryptoOptions.h",
+          "objc/api/peerconnection/RTCDataChannel.h",
+          "objc/api/peerconnection/RTCDataChannelConfiguration.h",
+          "objc/api/peerconnection/RTCDtmfSender.h",
+          "objc/api/peerconnection/RTCFieldTrials.h",
+          "objc/api/peerconnection/RTCIceCandidate.h",
+          "objc/api/peerconnection/RTCIceCandidateErrorEvent.h",
+          "objc/api/peerconnection/RTCIceServer.h",
+          "objc/api/peerconnection/RTCLegacyStatsReport.h",
+          "objc/api/peerconnection/RTCMediaConstraints.h",
+          "objc/api/peerconnection/RTCMediaSource.h",
+          "objc/api/peerconnection/RTCMediaStream.h",
+          "objc/api/peerconnection/RTCMediaStreamTrack.h",
+          "objc/api/peerconnection/RTCMetrics.h",
+          "objc/api/peerconnection/RTCMetricsSampleInfo.h",
+          "objc/api/peerconnection/RTCPeerConnection.h",
+          "objc/api/peerconnection/RTCPeerConnectionFactory.h",
+          "objc/api/peerconnection/RTCPeerConnectionFactoryOptions.h",
+          "objc/api/peerconnection/RTCRtcpParameters.h",
+          "objc/api/peerconnection/RTCRtpCodecParameters.h",
+          "objc/api/peerconnection/RTCRtpEncodingParameters.h",
+          "objc/api/peerconnection/RTCRtpHeaderExtension.h",
+          "objc/api/peerconnection/RTCRtpParameters.h",
+          "objc/api/peerconnection/RTCRtpReceiver.h",
+          "objc/api/peerconnection/RTCRtpSender.h",
+          "objc/api/peerconnection/RTCRtpTransceiver.h",
+          "objc/api/peerconnection/RTCSSLAdapter.h",
+          "objc/api/peerconnection/RTCSessionDescription.h",
+          "objc/api/peerconnection/RTCStatisticsReport.h",
+          "objc/api/peerconnection/RTCTracing.h",
+          "objc/api/peerconnection/RTCVideoSource.h",
+          "objc/api/peerconnection/RTCVideoTrack.h",
+          "objc/api/video_codec/RTCVideoDecoderAV1.h",
+          "objc/api/video_codec/RTCVideoDecoderVP8.h",
+          "objc/api/video_codec/RTCVideoDecoderVP9.h",
+          "objc/api/video_codec/RTCVideoEncoderAV1.h",
+          "objc/api/video_codec/RTCVideoEncoderVP8.h",
+          "objc/api/video_codec/RTCVideoEncoderVP9.h",
+          "objc/api/video_frame_buffer/RTCNativeI420Buffer.h",
+          "objc/api/video_frame_buffer/RTCNativeMutableI420Buffer.h",
+          "objc/base/RTCCodecSpecificInfo.h",
+          "objc/base/RTCEncodedImage.h",
+          "objc/base/RTCI420Buffer.h",
+          "objc/base/RTCLogging.h",
+          "objc/base/RTCMacros.h",
+          "objc/base/RTCMutableI420Buffer.h",
+          "objc/base/RTCMutableYUVPlanarBuffer.h",
+          "objc/base/RTCSSLCertificateVerifier.h",
+          "objc/base/RTCVideoCapturer.h",
+          "objc/base/RTCVideoCodecInfo.h",
+          "objc/base/RTCVideoDecoder.h",
+          "objc/base/RTCVideoDecoderFactory.h",
+          "objc/base/RTCVideoEncoder.h",
+          "objc/base/RTCVideoEncoderFactory.h",
+          "objc/base/RTCVideoEncoderQpThresholds.h",
+          "objc/base/RTCVideoEncoderSettings.h",
+          "objc/base/RTCVideoFrame.h",
+          "objc/base/RTCVideoFrameBuffer.h",
+          "objc/base/RTCVideoRenderer.h",
+          "objc/base/RTCYUVPlanarBuffer.h",
+          "objc/components/capturer/RTCCameraVideoCapturer.h",
+          "objc/components/capturer/RTCFileVideoCapturer.h",
+          "objc/components/renderer/metal/RTCMTLNSVideoView.h",
+          "objc/components/renderer/opengl/RTCNSGLVideoView.h",
+          "objc/components/renderer/opengl/RTCVideoViewShading.h",
+          "objc/components/video_codec/RTCCodecSpecificInfoH264.h",
+          "objc/components/video_codec/RTCDefaultVideoDecoderFactory.h",
+          "objc/components/video_codec/RTCDefaultVideoEncoderFactory.h",
+          "objc/components/video_codec/RTCH264ProfileLevelId.h",
+          "objc/components/video_codec/RTCVideoDecoderFactoryH264.h",
+          "objc/components/video_codec/RTCVideoDecoderH264.h",
+          "objc/components/video_codec/RTCVideoEncoderFactoryH264.h",
+          "objc/components/video_codec/RTCVideoEncoderH264.h",
+          "objc/components/video_frame_buffer/RTCCVPixelBuffer.h",
+          "objc/helpers/RTCDispatcher.h",
+        ]
+        if (!build_with_chromium) {
+          sources += [
+            "objc/api/logging/RTCCallbackLogger.h",
+            "objc/api/peerconnection/RTCFileLogger.h",
+          ]
+        }
+
+        deps = [
+          ":base_objc",
+          ":default_codec_factory_objc",
+          ":native_api",
+          ":native_video",
+          ":opengl_ui_objc",
+          ":peerconnectionfactory_base_objc",
+          ":videocapture_objc",
+          ":videocodec_objc",
+          ":videotoolbox_objc",
+        ]
+        if (!build_with_chromium) {
+          deps += [
+            ":callback_logger_objc",
+            ":file_logger_objc",
+          ]
+        }
+
+        frameworks = [
+          "AVFoundation.framework",
+          "CoreGraphics.framework",
+          "CoreMedia.framework",
+          "OpenGL.framework",
+        ]
+
+        configs = [ "..:common_objc" ]
+
+        public_configs = [ ":common_config_objc" ]
+      }
+
+      bundle_data("mac_framework_bundle") {
+        deps = [ "../sdk:mac_framework_objc" ]
+        sources = [ "$root_build_dir/WebRTC.framework" ]
+        outputs = [ "{{bundle_contents_dir}}/Frameworks/{{source_file_part}}" ]
+      }
+    }
+
+    rtc_library("wrapped_native_codec_objc") {
+      sources = [
+        "objc/api/video_codec/RTCWrappedNativeVideoDecoder.h",
+        "objc/api/video_codec/RTCWrappedNativeVideoDecoder.mm",
+        "objc/api/video_codec/RTCWrappedNativeVideoEncoder.h",
+        "objc/api/video_codec/RTCWrappedNativeVideoEncoder.mm",
+      ]
+
+      configs += [ "..:common_objc" ]
+      public_configs = [ ":common_config_objc" ]
+
+      deps = [
+        ":base_objc",
+        ":helpers_objc",
+        "../api/video_codecs:video_codecs_api",
+        "../media:codec",
+        "../media:rtc_media_base",
+      ]
+    }
+
+    # The native API is currently experimental and may change without notice.
+    rtc_library("native_api") {
+      visibility = [ "*" ]
+      allow_poison = [ "audio_codecs" ]  # TODO(bugs.webrtc.org/8396): Remove.
+      sources = [
+        "objc/native/api/network_monitor_factory.h",
+        "objc/native/api/network_monitor_factory.mm",
+        "objc/native/api/ssl_certificate_verifier.h",
+        "objc/native/api/ssl_certificate_verifier.mm",
+        "objc/native/api/video_capturer.h",
+        "objc/native/api/video_capturer.mm",
+        "objc/native/api/video_decoder_factory.h",
+        "objc/native/api/video_decoder_factory.mm",
+        "objc/native/api/video_encoder_factory.h",
+        "objc/native/api/video_encoder_factory.mm",
+        "objc/native/api/video_frame.h",
+        "objc/native/api/video_frame.mm",
+        "objc/native/api/video_frame_buffer.h",
+        "objc/native/api/video_frame_buffer.mm",
+        "objc/native/api/video_renderer.h",
+        "objc/native/api/video_renderer.mm",
+      ]
+
+      configs += [ "..:common_objc" ]
+
+      public_configs = [ ":common_config_objc" ]
+
+      deps = [
+        ":base_objc",
+        ":native_video",
+        ":videoframebuffer_objc",
+        "../api:libjingle_peerconnection_api",
+        "../api:make_ref_counted",
+        "../api:media_stream_interface",
+        "../api:scoped_refptr",
+        "../api/video:video_frame",
+        "../api/video:video_rtp_headers",
+        "../api/video_codecs:video_codecs_api",
+        "../common_video",
+        "../rtc_base:buffer",
+        "../rtc_base:logging",
+        "../rtc_base:ssl",
+        "../rtc_base:threading",
+      ]
+      if (is_ios) {
+        deps += [ ":native_network_monitor" ]
+      }
+      absl_deps = [ "//third_party/abseil-cpp/absl/memory" ]
+    }
+
+    if (is_ios) {
+      rtc_library("native_network_monitor") {
+        visibility = [ "*" ]
+
+        sources = [
+          "objc/native/src/objc_network_monitor.h",
+          "objc/native/src/objc_network_monitor.mm",
+        ]
+
+        deps = [
+          ":network_monitor_objc",
+          ":network_monitor_observer",
+          "../api:field_trials_view",
+          "../api:sequence_checker",
+          "../api/task_queue:pending_task_safety_flag",
+          "../rtc_base:logging",
+          "../rtc_base:macromagic",
+          "../rtc_base:stringutils",
+          "../rtc_base:threading",
+        ]
+
+        absl_deps = [ "//third_party/abseil-cpp/absl/strings" ]
+      }
+    }
+
+    rtc_library("native_video") {
+      sources = [
+        "objc/native/src/objc_frame_buffer.h",
+        "objc/native/src/objc_frame_buffer.mm",
+        "objc/native/src/objc_video_decoder_factory.h",
+        "objc/native/src/objc_video_decoder_factory.mm",
+        "objc/native/src/objc_video_encoder_factory.h",
+        "objc/native/src/objc_video_encoder_factory.mm",
+        "objc/native/src/objc_video_frame.h",
+        "objc/native/src/objc_video_frame.mm",
+        "objc/native/src/objc_video_renderer.h",
+        "objc/native/src/objc_video_renderer.mm",
+        "objc/native/src/objc_video_track_source.h",
+        "objc/native/src/objc_video_track_source.mm",
+      ]
+
+      configs += [ "..:common_objc" ]
+
+      public_configs = [ ":common_config_objc" ]
+
+      deps = [
+        ":base_native_additions_objc",
+        ":base_objc",
+        ":helpers_objc",
+        ":videocodec_objc",
+        ":videoframebuffer_objc",
+        ":vpx_codec_constants",
+        ":wrapped_native_codec_objc",
+        "../api:make_ref_counted",
+        "../api/video:video_frame",
+        "../api/video:video_rtp_headers",
+        "../api/video_codecs:video_codecs_api",
+        "../common_video",
+        "../media:codec",
+        "../media:rtc_audio_video",
+        "../media:rtc_media_base",
+        "../modules/video_coding:video_codec_interface",
+        "../rtc_base:checks",
+        "../rtc_base:logging",
+        "../rtc_base:timestamp_aligner",
+        "../rtc_base:timeutils",
+      ]
+    }
+
+    rtc_library("video_toolbox_cc") {
+      visibility = [
+        ":sdk_unittests_sources",
+        ":videotoolbox_objc",
+      ]
+      sources = [
+        "objc/components/video_codec/helpers.cc",
+        "objc/components/video_codec/helpers.h",
+        "objc/components/video_codec/nalu_rewriter.cc",
+        "objc/components/video_codec/nalu_rewriter.h",
+      ]
+      deps = [
+        "../common_video",
+        "../modules/video_coding:webrtc_h264",
+        "../rtc_base:buffer",
+        "../rtc_base:checks",
+        "../rtc_base:logging",
+      ]
+    }
+
+    rtc_library("videotoolbox_objc") {
+      visibility = [ "*" ]
+      allow_poison = [ "audio_codecs" ]  # TODO(bugs.webrtc.org/8396): Remove.
+      sources = [
+        "objc/components/video_codec/RTCVideoDecoderFactoryH264.h",
+        "objc/components/video_codec/RTCVideoDecoderFactoryH264.m",
+        "objc/components/video_codec/RTCVideoDecoderH264.h",
+        "objc/components/video_codec/RTCVideoDecoderH264.mm",
+        "objc/components/video_codec/RTCVideoEncoderFactoryH264.h",
+        "objc/components/video_codec/RTCVideoEncoderFactoryH264.m",
+        "objc/components/video_codec/RTCVideoEncoderH264.h",
+        "objc/components/video_codec/RTCVideoEncoderH264.mm",
+      ]
+
+      configs += [
+        "..:common_objc",
+        ":used_from_extension",
+      ]
+
+      if (is_ios && rtc_apprtcmobile_broadcast_extension) {
+        defines = [ "RTC_APPRTCMOBILE_BROADCAST_EXTENSION" ]
+      }
+
+      deps = [
+        ":base_native_additions_objc",
+        ":base_objc",
+        ":helpers_objc",
+        ":video_toolbox_cc",
+        ":videocodec_objc",
+        ":videoframebuffer_objc",
+        "../api/video_codecs:video_codecs_api",
+        "../common_video",
+        "../modules/video_coding:video_codec_interface",
+        "../rtc_base:buffer",
+        "../rtc_base:checks",
+        "../rtc_base:logging",
+        "../rtc_base:timeutils",
+        "//third_party/libyuv",
+      ]
+
+      frameworks = [
+        "CoreFoundation.framework",
+        "CoreMedia.framework",
+        "CoreVideo.framework",
+        "VideoToolbox.framework",
+      ]
+    }
+    }
+  }
+}
diff --git a/third_party/libwebrtc/sdk/OWNERS b/third_party/libwebrtc/sdk/OWNERS
new file mode 100644
index 0000000000..4d31ffb663
--- /dev/null
+++ b/third_party/libwebrtc/sdk/OWNERS
@@ -0,0 +1 @@
+magjed@webrtc.org
diff --git a/third_party/libwebrtc/sdk/android/api/org/webrtc/AddIceObserver.java b/third_party/libwebrtc/sdk/android/api/org/webrtc/AddIceObserver.java
new file mode 100644
index 0000000000..ff2c690029
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/api/org/webrtc/AddIceObserver.java
@@ -0,0 +1,20 @@
+/*
+ *  Copyright 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+/** Interface to handle completion of addIceCandidate  */
+public interface AddIceObserver {
+  /** Called when ICE candidate added successfully.*/
+  @CalledByNative public void onAddSuccess();
+
+  /** Called when ICE candidate addition failed.*/
+  @CalledByNative public void onAddFailure(String error);
+}
diff --git a/third_party/libwebrtc/sdk/android/api/org/webrtc/AudioDecoderFactoryFactory.java b/third_party/libwebrtc/sdk/android/api/org/webrtc/AudioDecoderFactoryFactory.java
new file mode 100644
index 0000000000..dd3e262896
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/api/org/webrtc/AudioDecoderFactoryFactory.java
@@ -0,0 +1,21 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+/**
+ * Implementations of this interface can create a native {@code webrtc::AudioDecoderFactory}.
+ */
+public interface AudioDecoderFactoryFactory {
+  /**
+   * Returns a pointer to a {@code webrtc::AudioDecoderFactory}. The caller takes ownership.
+   */
+  long createNativeAudioDecoderFactory();
+}
diff --git a/third_party/libwebrtc/sdk/android/api/org/webrtc/AudioEncoderFactoryFactory.java b/third_party/libwebrtc/sdk/android/api/org/webrtc/AudioEncoderFactoryFactory.java
new file mode 100644
index 0000000000..814b71aba1
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/api/org/webrtc/AudioEncoderFactoryFactory.java
@@ -0,0 +1,21 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+/**
+ * Implementations of this interface can create a native {@code webrtc::AudioEncoderFactory}.
+ */
+public interface AudioEncoderFactoryFactory {
+  /**
+   * Returns a pointer to a {@code webrtc::AudioEncoderFactory}. The caller takes ownership.
+   */
+  long createNativeAudioEncoderFactory();
+}
diff --git a/third_party/libwebrtc/sdk/android/api/org/webrtc/AudioProcessingFactory.java b/third_party/libwebrtc/sdk/android/api/org/webrtc/AudioProcessingFactory.java
new file mode 100644
index 0000000000..bd8fdb8989
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/api/org/webrtc/AudioProcessingFactory.java
@@ -0,0 +1,20 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+/** Factory for creating webrtc::AudioProcessing instances. */
+public interface AudioProcessingFactory {
+  /**
+   * Dynamically allocates a webrtc::AudioProcessing instance and returns a pointer to it.
+   * The caller takes ownership of the object.
+   */
+  public long createNative();
+}
diff --git a/third_party/libwebrtc/sdk/android/api/org/webrtc/AudioSource.java b/third_party/libwebrtc/sdk/android/api/org/webrtc/AudioSource.java
new file mode 100644
index 0000000000..f8104e5904
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/api/org/webrtc/AudioSource.java
@@ -0,0 +1,26 @@
+/*
+ *  Copyright 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+/**
+ * Java wrapper for a C++ AudioSourceInterface.  Used as the source for one or
+ * more {@code AudioTrack} objects.
+ */
+public class AudioSource extends MediaSource {
+  public AudioSource(long nativeSource) {
+    super(nativeSource);
+  }
+
+  /** Returns a pointer to webrtc::AudioSourceInterface. */
+  long getNativeAudioSource() {
+    return getNativeMediaSource();
+  }
+}
diff --git a/third_party/libwebrtc/sdk/android/api/org/webrtc/AudioTrack.java b/third_party/libwebrtc/sdk/android/api/org/webrtc/AudioTrack.java
new file mode 100644
index 0000000000..ca745db634
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/api/org/webrtc/AudioTrack.java
@@ -0,0 +1,32 @@
+/*
+ *  Copyright 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+/** Java wrapper for a C++ AudioTrackInterface */
+public class AudioTrack extends MediaStreamTrack {
+  public AudioTrack(long nativeTrack) {
+    super(nativeTrack);
+  }
+
+  /** Sets the volume for the underlying MediaSource. Volume is a gain value in the range
+   *  0 to 10.
+   */
+  public void setVolume(double volume) {
+    nativeSetVolume(getNativeAudioTrack(), volume);
+  }
+
+  /** Returns a pointer to webrtc::AudioTrackInterface. */
+  long getNativeAudioTrack() {
+    return getNativeMediaStreamTrack();
+  }
+
+  private static native void nativeSetVolume(long track, double volume);
+}
diff --git a/third_party/libwebrtc/sdk/android/api/org/webrtc/BuiltinAudioDecoderFactoryFactory.java b/third_party/libwebrtc/sdk/android/api/org/webrtc/BuiltinAudioDecoderFactoryFactory.java
new file mode 100644
index 0000000000..5ebc19f25d
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/api/org/webrtc/BuiltinAudioDecoderFactoryFactory.java
@@ -0,0 +1,23 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+/**
+ * Creates a native {@code webrtc::AudioDecoderFactory} with the builtin audio decoders.
+ */
+public class BuiltinAudioDecoderFactoryFactory implements AudioDecoderFactoryFactory {
+  @Override
+  public long createNativeAudioDecoderFactory() {
+    return nativeCreateBuiltinAudioDecoderFactory();
+  }
+
+  private static native long nativeCreateBuiltinAudioDecoderFactory();
+}
diff --git a/third_party/libwebrtc/sdk/android/api/org/webrtc/BuiltinAudioEncoderFactoryFactory.java b/third_party/libwebrtc/sdk/android/api/org/webrtc/BuiltinAudioEncoderFactoryFactory.java
new file mode 100644
index 0000000000..e884d4c3b9
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/api/org/webrtc/BuiltinAudioEncoderFactoryFactory.java
@@ -0,0 +1,23 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+/**
+ * This class creates a native {@code webrtc::AudioEncoderFactory} with the builtin audio encoders.
+ */
+public class BuiltinAudioEncoderFactoryFactory implements AudioEncoderFactoryFactory {
+  @Override
+  public long createNativeAudioEncoderFactory() {
+    return nativeCreateBuiltinAudioEncoderFactory();
+  }
+
+  private static native long nativeCreateBuiltinAudioEncoderFactory();
+}
diff --git a/third_party/libwebrtc/sdk/android/api/org/webrtc/CallSessionFileRotatingLogSink.java b/third_party/libwebrtc/sdk/android/api/org/webrtc/CallSessionFileRotatingLogSink.java
new file mode 100644
index 0000000000..f4edb58847
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/api/org/webrtc/CallSessionFileRotatingLogSink.java
@@ -0,0 +1,41 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+public class CallSessionFileRotatingLogSink {
+  private long nativeSink;
+
+  public static byte[] getLogData(String dirPath) {
+    if (dirPath == null) {
+      throw new IllegalArgumentException("dirPath may not be null.");
+    }
+    return nativeGetLogData(dirPath);
+  }
+
+  public CallSessionFileRotatingLogSink(
+      String dirPath, int maxFileSize, Logging.Severity severity) {
+    if (dirPath == null) {
+      throw new IllegalArgumentException("dirPath may not be null.");
+    }
+    nativeSink = nativeAddSink(dirPath, maxFileSize, severity.ordinal());
+  }
+
+  public void dispose() {
+    if (nativeSink != 0) {
+      nativeDeleteSink(nativeSink);
+      nativeSink = 0;
+    }
+  }
+
+  private static native long nativeAddSink(String dirPath, int maxFileSize, int severity);
+  private static native void nativeDeleteSink(long sink);
+  private static native byte[] nativeGetLogData(String dirPath);
+}
diff --git a/third_party/libwebrtc/sdk/android/api/org/webrtc/Camera1Capturer.java b/third_party/libwebrtc/sdk/android/api/org/webrtc/Camera1Capturer.java
new file mode 100644
index 0000000000..de172aa1d7
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/api/org/webrtc/Camera1Capturer.java
@@ -0,0 +1,33 @@
+/*
+ *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+import android.content.Context;
+
+public class Camera1Capturer extends CameraCapturer {
+  private final boolean captureToTexture;
+
+  public Camera1Capturer(
+      String cameraName, CameraEventsHandler eventsHandler, boolean captureToTexture) {
+    super(cameraName, eventsHandler, new Camera1Enumerator(captureToTexture));
+
+    this.captureToTexture = captureToTexture;
+  }
+
+  @Override
+  protected void createCameraSession(CameraSession.CreateSessionCallback createSessionCallback,
+      CameraSession.Events events, Context applicationContext,
+      SurfaceTextureHelper surfaceTextureHelper, String cameraName, int width, int height,
+      int framerate) {
+    Camera1Session.create(createSessionCallback, events, captureToTexture, applicationContext,
+        surfaceTextureHelper, cameraName, width, height, framerate);
+  }
+}
diff --git a/third_party/libwebrtc/sdk/android/api/org/webrtc/Camera1Enumerator.java b/third_party/libwebrtc/sdk/android/api/org/webrtc/Camera1Enumerator.java
new file mode 100644
index 0000000000..4a1aacdb05
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/api/org/webrtc/Camera1Enumerator.java
@@ -0,0 +1,190 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+import android.os.SystemClock;
+import androidx.annotation.Nullable;
+import java.util.ArrayList;
+import java.util.List;
+import org.webrtc.CameraEnumerationAndroid.CaptureFormat;
+
+@SuppressWarnings("deprecation")
+public class Camera1Enumerator implements CameraEnumerator {
+  private final static String TAG = "Camera1Enumerator";
+  // Each entry contains the supported formats for corresponding camera index. The formats for all
+  // cameras are enumerated on the first call to getSupportedFormats(), and cached for future
+  // reference.
+  private static List<List<CaptureFormat>> cachedSupportedFormats;
+
+  private final boolean captureToTexture;
+
+  public Camera1Enumerator() {
+    this(true /* captureToTexture */);
+  }
+
+  public Camera1Enumerator(boolean captureToTexture) {
+    this.captureToTexture = captureToTexture;
+  }
+
+  // Returns device names that can be used to create a new VideoCapturerAndroid.
+  @Override
+  public String[] getDeviceNames() {
+    ArrayList<String> namesList = new ArrayList<>();
+    for (int i = 0; i < android.hardware.Camera.getNumberOfCameras(); ++i) {
+      String name = getDeviceName(i);
+      if (name != null) {
+        namesList.add(name);
+        Logging.d(TAG, "Index: " + i + ". " + name);
+      } else {
+        Logging.e(TAG, "Index: " + i + ". Failed to query camera name.");
+      }
+    }
+    String[] namesArray = new String[namesList.size()];
+    return namesList.toArray(namesArray);
+  }
+
+  @Override
+  public boolean isFrontFacing(String deviceName) {
+    android.hardware.Camera.CameraInfo info = getCameraInfo(getCameraIndex(deviceName));
+    return info != null && info.facing == android.hardware.Camera.CameraInfo.CAMERA_FACING_FRONT;
+  }
+
+  @Override
+  public boolean isBackFacing(String deviceName) {
+    android.hardware.Camera.CameraInfo info = getCameraInfo(getCameraIndex(deviceName));
+    return info != null && info.facing == android.hardware.Camera.CameraInfo.CAMERA_FACING_BACK;
+  }
+
+  @Override
+  public boolean isInfrared(String deviceName) {
+    return false;
+  }
+
+  @Override
+  public List<CaptureFormat> getSupportedFormats(String deviceName) {
+    return getSupportedFormats(getCameraIndex(deviceName));
+  }
+
+  @Override
+  public CameraVideoCapturer createCapturer(
+      String deviceName, CameraVideoCapturer.CameraEventsHandler eventsHandler) {
+    return new Camera1Capturer(deviceName, eventsHandler, captureToTexture);
+  }
+
+  private static @Nullable android.hardware.Camera.CameraInfo getCameraInfo(int index) {
+    android.hardware.Camera.CameraInfo info = new android.hardware.Camera.CameraInfo();
+    try {
+      android.hardware.Camera.getCameraInfo(index, info);
+    } catch (Exception e) {
+      Logging.e(TAG, "getCameraInfo failed on index " + index, e);
+      return null;
+    }
+    return info;
+  }
+
+  static synchronized List<CaptureFormat> getSupportedFormats(int cameraId) {
+    if (cachedSupportedFormats == null) {
+      cachedSupportedFormats = new ArrayList<List<CaptureFormat>>();
+      for (int i = 0; i < android.hardware.Camera.getNumberOfCameras(); ++i) {
+        cachedSupportedFormats.add(enumerateFormats(i));
+      }
+    }
+    return cachedSupportedFormats.get(cameraId);
+  }
+
+  private static List<CaptureFormat> enumerateFormats(int cameraId) {
+    Logging.d(TAG, "Get supported formats for camera index " + cameraId + ".");
+    final long startTimeMs = SystemClock.elapsedRealtime();
+    final android.hardware.Camera.Parameters parameters;
+    android.hardware.Camera camera = null;
+    try {
+      Logging.d(TAG, "Opening camera with index " + cameraId);
+      camera = android.hardware.Camera.open(cameraId);
+      parameters = camera.getParameters();
+    } catch (RuntimeException e) {
+      Logging.e(TAG, "Open camera failed on camera index " + cameraId, e);
+      return new ArrayList<CaptureFormat>();
+    } finally {
+      if (camera != null) {
+        camera.release();
+      }
+    }
+
+    final List<CaptureFormat> formatList = new ArrayList<CaptureFormat>();
+    try {
+      int minFps = 0;
+      int maxFps = 0;
+      final List<int[]> listFpsRange = parameters.getSupportedPreviewFpsRange();
+      if (listFpsRange != null) {
+        // getSupportedPreviewFpsRange() returns a sorted list. Take the fps range
+        // corresponding to the highest fps.
+        final int[] range = listFpsRange.get(listFpsRange.size() - 1);
+        minFps = range[android.hardware.Camera.Parameters.PREVIEW_FPS_MIN_INDEX];
+        maxFps = range[android.hardware.Camera.Parameters.PREVIEW_FPS_MAX_INDEX];
+      }
+      for (android.hardware.Camera.Size size : parameters.getSupportedPreviewSizes()) {
+        formatList.add(new CaptureFormat(size.width, size.height, minFps, maxFps));
+      }
+    } catch (Exception e) {
+      Logging.e(TAG, "getSupportedFormats() failed on camera index " + cameraId, e);
+    }
+
+    final long endTimeMs = SystemClock.elapsedRealtime();
+    Logging.d(TAG, "Get supported formats for camera index " + cameraId + " done."
+            + " Time spent: " + (endTimeMs - startTimeMs) + " ms.");
+    return formatList;
+  }
+
+  // Convert from android.hardware.Camera.Size to Size.
+  static List<Size> convertSizes(List<android.hardware.Camera.Size> cameraSizes) {
+    final List<Size> sizes = new ArrayList<Size>();
+    for (android.hardware.Camera.Size size : cameraSizes) {
+      sizes.add(new Size(size.width, size.height));
+    }
+    return sizes;
+  }
+
+  // Convert from int[2] to CaptureFormat.FramerateRange.
+  static List<CaptureFormat.FramerateRange> convertFramerates(List<int[]> arrayRanges) {
+    final List<CaptureFormat.FramerateRange> ranges = new ArrayList<CaptureFormat.FramerateRange>();
+    for (int[] range : arrayRanges) {
+      ranges.add(new CaptureFormat.FramerateRange(
+          range[android.hardware.Camera.Parameters.PREVIEW_FPS_MIN_INDEX],
+          range[android.hardware.Camera.Parameters.PREVIEW_FPS_MAX_INDEX]));
+    }
+    return ranges;
+  }
+
+  // Returns the camera index for camera with name `deviceName`, or throws IllegalArgumentException
+  // if no such camera can be found.
+  static int getCameraIndex(String deviceName) {
+    Logging.d(TAG, "getCameraIndex: " + deviceName);
+    for (int i = 0; i < android.hardware.Camera.getNumberOfCameras(); ++i) {
+      if (deviceName.equals(getDeviceName(i))) {
+        return i;
+      }
+    }
+    throw new IllegalArgumentException("No such camera: " + deviceName);
+  }
+
+  // Returns the name of the camera with camera index. Returns null if the
+  // camera can not be used.
+  static @Nullable String getDeviceName(int index) {
+    android.hardware.Camera.CameraInfo info = getCameraInfo(index);
+    if (info == null) {
+      return null;
+    }
+
+    String facing =
+        (info.facing == android.hardware.Camera.CameraInfo.CAMERA_FACING_FRONT) ? "front" : "back";
+    return "Camera " + index + ", Facing " + facing + ", Orientation " + info.orientation;
+  }
+}
diff --git a/third_party/libwebrtc/sdk/android/api/org/webrtc/Camera2Capturer.java b/third_party/libwebrtc/sdk/android/api/org/webrtc/Camera2Capturer.java
new file mode 100644
index 0000000000..c4becf4819
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/api/org/webrtc/Camera2Capturer.java
@@ -0,0 +1,36 @@
+/*
+ *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+import android.content.Context;
+import android.hardware.camera2.CameraManager;
+import androidx.annotation.Nullable;
+
+public class Camera2Capturer extends CameraCapturer {
+  private final Context context;
+  @Nullable private final CameraManager cameraManager;
+
+  public Camera2Capturer(Context context, String cameraName, CameraEventsHandler eventsHandler) {
+    super(cameraName, eventsHandler, new Camera2Enumerator(context));
+
+    this.context = context;
+    cameraManager = (CameraManager) context.getSystemService(Context.CAMERA_SERVICE);
+  }
+
+  @Override
+  protected void createCameraSession(CameraSession.CreateSessionCallback createSessionCallback,
+      CameraSession.Events events, Context applicationContext,
+      SurfaceTextureHelper surfaceTextureHelper, String cameraName, int width, int height,
+      int framerate) {
+    Camera2Session.create(createSessionCallback, events, applicationContext, cameraManager,
+        surfaceTextureHelper, cameraName, width, height, framerate);
+  }
+}
diff --git a/third_party/libwebrtc/sdk/android/api/org/webrtc/Camera2Enumerator.java b/third_party/libwebrtc/sdk/android/api/org/webrtc/Camera2Enumerator.java
new file mode 100644
index 0000000000..44e239ad8e
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/api/org/webrtc/Camera2Enumerator.java
@@ -0,0 +1,251 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+import android.content.Context;
+import android.graphics.Rect;
+import android.graphics.SurfaceTexture;
+import android.hardware.camera2.CameraAccessException;
+import android.hardware.camera2.CameraCharacteristics;
+import android.hardware.camera2.CameraManager;
+import android.hardware.camera2.CameraMetadata;
+import android.hardware.camera2.params.StreamConfigurationMap;
+import android.os.Build;
+import android.os.SystemClock;
+import android.util.Range;
+import androidx.annotation.Nullable;
+import java.util.ArrayList;
+import java.util.Collections;
+import java.util.HashMap;
+import java.util.List;
+import java.util.Map;
+import org.webrtc.CameraEnumerationAndroid.CaptureFormat;
+
+public class Camera2Enumerator implements CameraEnumerator {
+  private final static String TAG = "Camera2Enumerator";
+  private final static double NANO_SECONDS_PER_SECOND = 1.0e9;
+
+  // Each entry contains the supported formats for a given camera index. The formats are enumerated
+  // lazily in getSupportedFormats(), and cached for future reference.
+  private static final Map<String, List<CaptureFormat>> cachedSupportedFormats =
+      new HashMap<String, List<CaptureFormat>>();
+
+  final Context context;
+  @Nullable final CameraManager cameraManager;
+
+  public Camera2Enumerator(Context context) {
+    this.context = context;
+    this.cameraManager = (CameraManager) context.getSystemService(Context.CAMERA_SERVICE);
+  }
+
+  @Override
+  public String[] getDeviceNames() {
+    try {
+      return cameraManager.getCameraIdList();
+    } catch (CameraAccessException e) {
+      Logging.e(TAG, "Camera access exception", e);
+      return new String[] {};
+    }
+  }
+
+  @Override
+  public boolean isFrontFacing(String deviceName) {
+    CameraCharacteristics characteristics = getCameraCharacteristics(deviceName);
+
+    return characteristics != null
+        && characteristics.get(CameraCharacteristics.LENS_FACING)
+        == CameraMetadata.LENS_FACING_FRONT;
+  }
+
+  @Override
+  public boolean isBackFacing(String deviceName) {
+    CameraCharacteristics characteristics = getCameraCharacteristics(deviceName);
+
+    return characteristics != null
+        && characteristics.get(CameraCharacteristics.LENS_FACING)
+        == CameraMetadata.LENS_FACING_BACK;
+  }
+
+  @Override
+  public boolean isInfrared(String deviceName) {
+    CameraCharacteristics characteristics = getCameraCharacteristics(deviceName);
+
+    if (Build.VERSION.SDK_INT >= Build.VERSION_CODES.Q) {
+      Integer colors = characteristics.get(CameraCharacteristics.SENSOR_INFO_COLOR_FILTER_ARRANGEMENT);
+      return colors != null && colors.equals(CameraCharacteristics.SENSOR_INFO_COLOR_FILTER_ARRANGEMENT_NIR);
+    }
+
+    return false;
+  }
+
+  @Nullable
+  @Override
+  public List<CaptureFormat> getSupportedFormats(String deviceName) {
+    return getSupportedFormats(context, deviceName);
+  }
+
+  @Override
+  public CameraVideoCapturer createCapturer(
+      String deviceName, CameraVideoCapturer.CameraEventsHandler eventsHandler) {
+    return new Camera2Capturer(context, deviceName, eventsHandler);
+  }
+
+  private @Nullable CameraCharacteristics getCameraCharacteristics(String deviceName) {
+    try {
+      return cameraManager.getCameraCharacteristics(deviceName);
+    } catch (CameraAccessException | RuntimeException e) {
+      Logging.e(TAG, "Camera access exception", e);
+      return null;
+    }
+  }
+
+  /**
+   * Checks if API is supported and all cameras have better than legacy support.
+   */
+  public static boolean isSupported(Context context) {
+    CameraManager cameraManager = (CameraManager) context.getSystemService(Context.CAMERA_SERVICE);
+    try {
+      String[] cameraIds = cameraManager.getCameraIdList();
+      for (String id : cameraIds) {
+        CameraCharacteristics characteristics = cameraManager.getCameraCharacteristics(id);
+        if (characteristics.get(CameraCharacteristics.INFO_SUPPORTED_HARDWARE_LEVEL)
+            == CameraCharacteristics.INFO_SUPPORTED_HARDWARE_LEVEL_LEGACY) {
+          return false;
+        }
+      }
+    } catch (CameraAccessException | RuntimeException e) {
+      Logging.e(TAG, "Failed to check if camera2 is supported", e);
+      return false;
+    }
+    return true;
+  }
+
+  static int getFpsUnitFactor(Range<Integer>[] fpsRanges) {
+    if (fpsRanges.length == 0) {
+      return 1000;
+    }
+    return fpsRanges[0].getUpper() < 1000 ? 1000 : 1;
+  }
+
+  static List<Size> getSupportedSizes(CameraCharacteristics cameraCharacteristics) {
+    final StreamConfigurationMap streamMap =
+        cameraCharacteristics.get(CameraCharacteristics.SCALER_STREAM_CONFIGURATION_MAP);
+    final int supportLevel =
+        cameraCharacteristics.get(CameraCharacteristics.INFO_SUPPORTED_HARDWARE_LEVEL);
+
+    final android.util.Size[] nativeSizes = streamMap.getOutputSizes(SurfaceTexture.class);
+    final List<Size> sizes = convertSizes(nativeSizes);
+
+    // Video may be stretched pre LMR1 on legacy implementations.
+    // Filter out formats that have different aspect ratio than the sensor array.
+    if (Build.VERSION.SDK_INT < Build.VERSION_CODES.LOLLIPOP_MR1
+        && supportLevel == CameraCharacteristics.INFO_SUPPORTED_HARDWARE_LEVEL_LEGACY) {
+      final Rect activeArraySize =
+          cameraCharacteristics.get(CameraCharacteristics.SENSOR_INFO_ACTIVE_ARRAY_SIZE);
+      final ArrayList<Size> filteredSizes = new ArrayList<Size>();
+
+      for (Size size : sizes) {
+        if (activeArraySize.width() * size.height == activeArraySize.height() * size.width) {
+          filteredSizes.add(size);
+        }
+      }
+
+      return filteredSizes;
+    } else {
+      return sizes;
+    }
+  }
+
+  @Nullable
+  static List<CaptureFormat> getSupportedFormats(Context context, String cameraId) {
+    return getSupportedFormats(
+        (CameraManager) context.getSystemService(Context.CAMERA_SERVICE), cameraId);
+  }
+
+  @Nullable
+  static List<CaptureFormat> getSupportedFormats(CameraManager cameraManager, String cameraId) {
+    synchronized (cachedSupportedFormats) {
+      if (cachedSupportedFormats.containsKey(cameraId)) {
+        return cachedSupportedFormats.get(cameraId);
+      }
+
+      Logging.d(TAG, "Get supported formats for camera index " + cameraId + ".");
+      final long startTimeMs = SystemClock.elapsedRealtime();
+
+      final CameraCharacteristics cameraCharacteristics;
+      try {
+        cameraCharacteristics = cameraManager.getCameraCharacteristics(cameraId);
+      } catch (Exception ex) {
+        Logging.e(TAG, "getCameraCharacteristics()", ex);
+        return new ArrayList<CaptureFormat>();
+      }
+
+      final StreamConfigurationMap streamMap =
+          cameraCharacteristics.get(CameraCharacteristics.SCALER_STREAM_CONFIGURATION_MAP);
+
+      Range<Integer>[] fpsRanges =
+          cameraCharacteristics.get(CameraCharacteristics.CONTROL_AE_AVAILABLE_TARGET_FPS_RANGES);
+      List<CaptureFormat.FramerateRange> framerateRanges =
+          convertFramerates(fpsRanges, getFpsUnitFactor(fpsRanges));
+      List<Size> sizes = getSupportedSizes(cameraCharacteristics);
+
+      int defaultMaxFps = 0;
+      for (CaptureFormat.FramerateRange framerateRange : framerateRanges) {
+        defaultMaxFps = Math.max(defaultMaxFps, framerateRange.max);
+      }
+
+      final List<CaptureFormat> formatList = new ArrayList<CaptureFormat>();
+      for (Size size : sizes) {
+        long minFrameDurationNs = 0;
+        try {
+          minFrameDurationNs = streamMap.getOutputMinFrameDuration(
+              SurfaceTexture.class, new android.util.Size(size.width, size.height));
+        } catch (Exception e) {
+          // getOutputMinFrameDuration() is not supported on all devices. Ignore silently.
+        }
+        final int maxFps = (minFrameDurationNs == 0)
+            ? defaultMaxFps
+            : (int) Math.round(NANO_SECONDS_PER_SECOND / minFrameDurationNs) * 1000;
+        formatList.add(new CaptureFormat(size.width, size.height, 0, maxFps));
+        Logging.d(TAG, "Format: " + size.width + "x" + size.height + "@" + maxFps);
+      }
+
+      cachedSupportedFormats.put(cameraId, formatList);
+      final long endTimeMs = SystemClock.elapsedRealtime();
+      Logging.d(TAG, "Get supported formats for camera index " + cameraId + " done."
+              + " Time spent: " + (endTimeMs - startTimeMs) + " ms.");
+      return formatList;
+    }
+  }
+
+  // Convert from android.util.Size to Size.
+  private static List<Size> convertSizes(android.util.Size[] cameraSizes) {
+    if (cameraSizes == null || cameraSizes.length == 0) {
+      return Collections.emptyList();
+    }
+    final List<Size> sizes = new ArrayList<>(cameraSizes.length);
+    for (android.util.Size size : cameraSizes) {
+      sizes.add(new Size(size.getWidth(), size.getHeight()));
+    }
+    return sizes;
+  }
+
+  // Convert from android.util.Range<Integer> to CaptureFormat.FramerateRange.
+  static List<CaptureFormat.FramerateRange> convertFramerates(
+      Range<Integer>[] arrayRanges, int unitFactor) {
+    final List<CaptureFormat.FramerateRange> ranges = new ArrayList<CaptureFormat.FramerateRange>();
+    for (Range<Integer> range : arrayRanges) {
+      ranges.add(new CaptureFormat.FramerateRange(
+          range.getLower() * unitFactor, range.getUpper() * unitFactor));
+    }
+    return ranges;
+  }
+}
diff --git a/third_party/libwebrtc/sdk/android/api/org/webrtc/CameraEnumerationAndroid.java b/third_party/libwebrtc/sdk/android/api/org/webrtc/CameraEnumerationAndroid.java
new file mode 100644
index 0000000000..0c3188fffe
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/api/org/webrtc/CameraEnumerationAndroid.java
@@ -0,0 +1,206 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+import static java.lang.Math.abs;
+
+import android.graphics.ImageFormat;
+import java.util.ArrayList;
+import java.util.Arrays;
+import java.util.Collections;
+import java.util.Comparator;
+import java.util.List;
+
+@SuppressWarnings("deprecation")
+public class CameraEnumerationAndroid {
+  private final static String TAG = "CameraEnumerationAndroid";
+
+  static final ArrayList<Size> COMMON_RESOLUTIONS = new ArrayList<Size>(Arrays.asList(
+      // 0, Unknown resolution
+      new Size(160, 120), // 1, QQVGA
+      new Size(240, 160), // 2, HQVGA
+      new Size(320, 240), // 3, QVGA
+      new Size(400, 240), // 4, WQVGA
+      new Size(480, 320), // 5, HVGA
+      new Size(640, 360), // 6, nHD
+      new Size(640, 480), // 7, VGA
+      new Size(768, 480), // 8, WVGA
+      new Size(854, 480), // 9, FWVGA
+      new Size(800, 600), // 10, SVGA
+      new Size(960, 540), // 11, qHD
+      new Size(960, 640), // 12, DVGA
+      new Size(1024, 576), // 13, WSVGA
+      new Size(1024, 600), // 14, WVSGA
+      new Size(1280, 720), // 15, HD
+      new Size(1280, 1024), // 16, SXGA
+      new Size(1920, 1080), // 17, Full HD
+      new Size(1920, 1440), // 18, Full HD 4:3
+      new Size(2560, 1440), // 19, QHD
+      new Size(3840, 2160) // 20, UHD
+      ));
+
+  public static class CaptureFormat {
+    // Class to represent a framerate range. The framerate varies because of lightning conditions.
+    // The values are multiplied by 1000, so 1000 represents one frame per second.
+    public static class FramerateRange {
+      public int min;
+      public int max;
+
+      public FramerateRange(int min, int max) {
+        this.min = min;
+        this.max = max;
+      }
+
+      @Override
+      public String toString() {
+        return "[" + (min / 1000.0f) + ":" + (max / 1000.0f) + "]";
+      }
+
+      @Override
+      public boolean equals(Object other) {
+        if (!(other instanceof FramerateRange)) {
+          return false;
+        }
+        final FramerateRange otherFramerate = (FramerateRange) other;
+        return min == otherFramerate.min && max == otherFramerate.max;
+      }
+
+      @Override
+      public int hashCode() {
+        // Use prime close to 2^16 to avoid collisions for normal values less than 2^16.
+        return 1 + 65537 * min + max;
+      }
+    }
+
+    public final int width;
+    public final int height;
+    public final FramerateRange framerate;
+
+    // TODO(hbos): If VideoCapturer.startCapture is updated to support other image formats then this
+    // needs to be updated and VideoCapturer.getSupportedFormats need to return CaptureFormats of
+    // all imageFormats.
+    public final int imageFormat = ImageFormat.NV21;
+
+    public CaptureFormat(int width, int height, int minFramerate, int maxFramerate) {
+      this.width = width;
+      this.height = height;
+      this.framerate = new FramerateRange(minFramerate, maxFramerate);
+    }
+
+    public CaptureFormat(int width, int height, FramerateRange framerate) {
+      this.width = width;
+      this.height = height;
+      this.framerate = framerate;
+    }
+
+    // Calculates the frame size of this capture format.
+    public int frameSize() {
+      return frameSize(width, height, imageFormat);
+    }
+
+    // Calculates the frame size of the specified image format. Currently only
+    // supporting ImageFormat.NV21.
+    // The size is width * height * number of bytes per pixel.
+    // http://developer.android.com/reference/android/hardware/Camera.html#addCallbackBuffer(byte[])
+    public static int frameSize(int width, int height, int imageFormat) {
+      if (imageFormat != ImageFormat.NV21) {
+        throw new UnsupportedOperationException("Don't know how to calculate "
+            + "the frame size of non-NV21 image formats.");
+      }
+      return (width * height * ImageFormat.getBitsPerPixel(imageFormat)) / 8;
+    }
+
+    @Override
+    public String toString() {
+      return width + "x" + height + "@" + framerate;
+    }
+
+    @Override
+    public boolean equals(Object other) {
+      if (!(other instanceof CaptureFormat)) {
+        return false;
+      }
+      final CaptureFormat otherFormat = (CaptureFormat) other;
+      return width == otherFormat.width && height == otherFormat.height
+          && framerate.equals(otherFormat.framerate);
+    }
+
+    @Override
+    public int hashCode() {
+      return 1 + (width * 65497 + height) * 251 + framerate.hashCode();
+    }
+  }
+
+  // Helper class for finding the closest supported format for the two functions below. It creates a
+  // comparator based on the difference to some requested parameters, where the element with the
+  // minimum difference is the element that is closest to the requested parameters.
+  private static abstract class ClosestComparator<T> implements Comparator<T> {
+    // Difference between supported and requested parameter.
+    abstract int diff(T supportedParameter);
+
+    @Override
+    public int compare(T t1, T t2) {
+      return diff(t1) - diff(t2);
+    }
+  }
+
+  // Prefer a fps range with an upper bound close to `framerate`. Also prefer a fps range with a low
+  // lower bound, to allow the framerate to fluctuate based on lightning conditions.
+  public static CaptureFormat.FramerateRange getClosestSupportedFramerateRange(
+      List<CaptureFormat.FramerateRange> supportedFramerates, final int requestedFps) {
+    return Collections.min(
+        supportedFramerates, new ClosestComparator<CaptureFormat.FramerateRange>() {
+          // Progressive penalty if the upper bound is further away than `MAX_FPS_DIFF_THRESHOLD`
+          // from requested.
+          private static final int MAX_FPS_DIFF_THRESHOLD = 5000;
+          private static final int MAX_FPS_LOW_DIFF_WEIGHT = 1;
+          private static final int MAX_FPS_HIGH_DIFF_WEIGHT = 3;
+
+          // Progressive penalty if the lower bound is bigger than `MIN_FPS_THRESHOLD`.
+          private static final int MIN_FPS_THRESHOLD = 8000;
+          private static final int MIN_FPS_LOW_VALUE_WEIGHT = 1;
+          private static final int MIN_FPS_HIGH_VALUE_WEIGHT = 4;
+
+          // Use one weight for small `value` less than `threshold`, and another weight above.
+          private int progressivePenalty(int value, int threshold, int lowWeight, int highWeight) {
+            return (value < threshold) ? value * lowWeight
+                                       : threshold * lowWeight + (value - threshold) * highWeight;
+          }
+
+          @Override
+          int diff(CaptureFormat.FramerateRange range) {
+            final int minFpsError = progressivePenalty(
+                range.min, MIN_FPS_THRESHOLD, MIN_FPS_LOW_VALUE_WEIGHT, MIN_FPS_HIGH_VALUE_WEIGHT);
+            final int maxFpsError = progressivePenalty(Math.abs(requestedFps * 1000 - range.max),
+                MAX_FPS_DIFF_THRESHOLD, MAX_FPS_LOW_DIFF_WEIGHT, MAX_FPS_HIGH_DIFF_WEIGHT);
+            return minFpsError + maxFpsError;
+          }
+        });
+  }
+
+  public static Size getClosestSupportedSize(
+      List<Size> supportedSizes, final int requestedWidth, final int requestedHeight) {
+    return Collections.min(supportedSizes, new ClosestComparator<Size>() {
+      @Override
+      int diff(Size size) {
+        return abs(requestedWidth - size.width) + abs(requestedHeight - size.height);
+      }
+    });
+  }
+
+  // Helper method for camera classes.
+  static void reportCameraResolution(Histogram histogram, Size resolution) {
+    int index = COMMON_RESOLUTIONS.indexOf(resolution);
+    // 0 is reserved for unknown resolution, so add 1.
+    // indexOf returns -1 for unknown resolutions so it becomes 0 automatically.
+    histogram.addSample(index + 1);
+  }
+}
diff --git a/third_party/libwebrtc/sdk/android/api/org/webrtc/CameraEnumerator.java b/third_party/libwebrtc/sdk/android/api/org/webrtc/CameraEnumerator.java
new file mode 100644
index 0000000000..db34d542c8
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/api/org/webrtc/CameraEnumerator.java
@@ -0,0 +1,26 @@
+/*
+ *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+import org.webrtc.CameraEnumerationAndroid.CaptureFormat;
+
+import java.util.List;
+
+public interface CameraEnumerator {
+  public String[] getDeviceNames();
+  public boolean isFrontFacing(String deviceName);
+  public boolean isBackFacing(String deviceName);
+  public boolean isInfrared(String deviceName);
+  public List<CaptureFormat> getSupportedFormats(String deviceName);
+
+  public CameraVideoCapturer createCapturer(
+      String deviceName, CameraVideoCapturer.CameraEventsHandler eventsHandler);
+}
diff --git a/third_party/libwebrtc/sdk/android/api/org/webrtc/CameraVideoCapturer.java b/third_party/libwebrtc/sdk/android/api/org/webrtc/CameraVideoCapturer.java
new file mode 100644
index 0000000000..ec26868b5c
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/api/org/webrtc/CameraVideoCapturer.java
@@ -0,0 +1,172 @@
+/*
+ *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+import android.media.MediaRecorder;
+
+/**
+ * Base interface for camera1 and camera2 implementations. Extends VideoCapturer with a
+ * switchCamera() function. Also provides subinterfaces for handling camera events, and a helper
+ * class for detecting camera freezes.
+ */
+public interface CameraVideoCapturer extends VideoCapturer {
+  /**
+   * Camera events handler - can be used to be notifed about camera events. The callbacks are
+   * executed from an arbitrary thread.
+   */
+  public interface CameraEventsHandler {
+    // Camera error handler - invoked when camera can not be opened
+    // or any camera exception happens on camera thread.
+    void onCameraError(String errorDescription);
+
+    // Called when camera is disconnected.
+    void onCameraDisconnected();
+
+    // Invoked when camera stops receiving frames.
+    void onCameraFreezed(String errorDescription);
+
+    // Callback invoked when camera is opening.
+    void onCameraOpening(String cameraName);
+
+    // Callback invoked when first camera frame is available after camera is started.
+    void onFirstFrameAvailable();
+
+    // Callback invoked when camera is closed.
+    void onCameraClosed();
+  }
+
+  /**
+   * Camera switch handler - one of these functions are invoked with the result of switchCamera().
+   * The callback may be called on an arbitrary thread.
+   */
+  public interface CameraSwitchHandler {
+    // Invoked on success. `isFrontCamera` is true if the new camera is front facing.
+    void onCameraSwitchDone(boolean isFrontCamera);
+
+    // Invoked on failure, e.g. camera is stopped or only one camera available.
+    void onCameraSwitchError(String errorDescription);
+  }
+
+  /**
+   * Switch camera to the next valid camera id. This can only be called while the camera is running.
+   * This function can be called from any thread.
+   */
+  void switchCamera(CameraSwitchHandler switchEventsHandler);
+
+  /**
+   * Switch camera to the specified camera id. This can only be called while the camera is running.
+   * This function can be called from any thread.
+   */
+  void switchCamera(CameraSwitchHandler switchEventsHandler, String cameraName);
+
+  /**
+   * MediaRecorder add/remove handler - one of these functions are invoked with the result of
+   * addMediaRecorderToCamera() or removeMediaRecorderFromCamera calls.
+   * The callback may be called on an arbitrary thread.
+   */
+  @Deprecated
+  public interface MediaRecorderHandler {
+    // Invoked on success.
+    void onMediaRecorderSuccess();
+
+    // Invoked on failure, e.g. camera is stopped or any exception happens.
+    void onMediaRecorderError(String errorDescription);
+  }
+
+  /**
+   * Add MediaRecorder to camera pipeline. This can only be called while the camera is running.
+   * Once MediaRecorder is added to camera pipeline camera switch is not allowed.
+   * This function can be called from any thread.
+   */
+  @Deprecated
+  default void addMediaRecorderToCamera(
+      MediaRecorder mediaRecorder, MediaRecorderHandler resultHandler) {
+    throw new UnsupportedOperationException("Deprecated and not implemented.");
+  }
+
+  /**
+   * Remove MediaRecorder from camera pipeline. This can only be called while the camera is running.
+   * This function can be called from any thread.
+   */
+  @Deprecated
+  default void removeMediaRecorderFromCamera(MediaRecorderHandler resultHandler) {
+    throw new UnsupportedOperationException("Deprecated and not implemented.");
+  }
+
+  /**
+   * Helper class to log framerate and detect if the camera freezes. It will run periodic callbacks
+   * on the SurfaceTextureHelper thread passed in the ctor, and should only be operated from that
+   * thread.
+   */
+  public static class CameraStatistics {
+    private final static String TAG = "CameraStatistics";
+    private final static int CAMERA_OBSERVER_PERIOD_MS = 2000;
+    private final static int CAMERA_FREEZE_REPORT_TIMOUT_MS = 4000;
+
+    private final SurfaceTextureHelper surfaceTextureHelper;
+    private final CameraEventsHandler eventsHandler;
+    private int frameCount;
+    private int freezePeriodCount;
+    // Camera observer - monitors camera framerate. Observer is executed on camera thread.
+    private final Runnable cameraObserver = new Runnable() {
+      @Override
+      public void run() {
+        final int cameraFps = Math.round(frameCount * 1000.0f / CAMERA_OBSERVER_PERIOD_MS);
+        Logging.d(TAG, "Camera fps: " + cameraFps + ".");
+        if (frameCount == 0) {
+          ++freezePeriodCount;
+          if (CAMERA_OBSERVER_PERIOD_MS * freezePeriodCount >= CAMERA_FREEZE_REPORT_TIMOUT_MS
+              && eventsHandler != null) {
+            Logging.e(TAG, "Camera freezed.");
+            if (surfaceTextureHelper.isTextureInUse()) {
+              // This can only happen if we are capturing to textures.
+              eventsHandler.onCameraFreezed("Camera failure. Client must return video buffers.");
+            } else {
+              eventsHandler.onCameraFreezed("Camera failure.");
+            }
+            return;
+          }
+        } else {
+          freezePeriodCount = 0;
+        }
+        frameCount = 0;
+        surfaceTextureHelper.getHandler().postDelayed(this, CAMERA_OBSERVER_PERIOD_MS);
+      }
+    };
+
+    public CameraStatistics(
+        SurfaceTextureHelper surfaceTextureHelper, CameraEventsHandler eventsHandler) {
+      if (surfaceTextureHelper == null) {
+        throw new IllegalArgumentException("SurfaceTextureHelper is null");
+      }
+      this.surfaceTextureHelper = surfaceTextureHelper;
+      this.eventsHandler = eventsHandler;
+      this.frameCount = 0;
+      this.freezePeriodCount = 0;
+      surfaceTextureHelper.getHandler().postDelayed(cameraObserver, CAMERA_OBSERVER_PERIOD_MS);
+    }
+
+    private void checkThread() {
+      if (Thread.currentThread() != surfaceTextureHelper.getHandler().getLooper().getThread()) {
+        throw new IllegalStateException("Wrong thread");
+      }
+    }
+
+    public void addFrame() {
+      checkThread();
+      ++frameCount;
+    }
+
+    public void release() {
+      surfaceTextureHelper.getHandler().removeCallbacks(cameraObserver);
+    }
+  }
+}
diff --git a/third_party/libwebrtc/sdk/android/api/org/webrtc/CapturerObserver.java b/third_party/libwebrtc/sdk/android/api/org/webrtc/CapturerObserver.java
new file mode 100644
index 0000000000..382dc15b3a
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/api/org/webrtc/CapturerObserver.java
@@ -0,0 +1,27 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+/**
+ * Interface for observering a capturer. Passed to {@link VideoCapturer#initialize}. Provided by
+ * {@link VideoSource#getCapturerObserver}.
+ *
+ * All callbacks must be executed on a single thread.
+ */
+public interface CapturerObserver {
+  /** Notify if the capturer have been started successfully or not. */
+  void onCapturerStarted(boolean success);
+  /** Notify that the capturer has been stopped. */
+  void onCapturerStopped();
+
+  /** Delivers a captured frame. */
+  void onFrameCaptured(VideoFrame frame);
+}
diff --git a/third_party/libwebrtc/sdk/android/api/org/webrtc/CryptoOptions.java b/third_party/libwebrtc/sdk/android/api/org/webrtc/CryptoOptions.java
new file mode 100644
index 0000000000..6e06bc6426
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/api/org/webrtc/CryptoOptions.java
@@ -0,0 +1,145 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+/**
+ * CryptoOptions defines advanced cryptographic settings for native WebRTC.
+ * These settings must be passed into RTCConfiguration. WebRTC is secur by
+ * default and you should not need to set any of these options unless you are
+ * specifically looking for an additional crypto feature such as AES_GCM
+ * support. This class is the Java binding of native api/crypto/cryptooptions.h
+ */
+public final class CryptoOptions {
+  /**
+   * SRTP Related Peer Connection Options.
+   */
+  public final class Srtp {
+    /**
+     * Enable GCM crypto suites from RFC 7714 for SRTP. GCM will only be used
+     * if both sides enable it
+     */
+    private final boolean enableGcmCryptoSuites;
+    /**
+     * If set to true, the (potentially insecure) crypto cipher
+     * kSrtpAes128CmSha1_32 will be included in the list of supported ciphers
+     * during negotiation. It will only be used if both peers support it and no
+     * other ciphers get preferred.
+     */
+    private final boolean enableAes128Sha1_32CryptoCipher;
+    /**
+     * If set to true, encrypted RTP header extensions as defined in RFC 6904
+     * will be negotiated. They will only be used if both peers support them.
+     */
+    private final boolean enableEncryptedRtpHeaderExtensions;
+
+    private Srtp(boolean enableGcmCryptoSuites, boolean enableAes128Sha1_32CryptoCipher,
+        boolean enableEncryptedRtpHeaderExtensions) {
+      this.enableGcmCryptoSuites = enableGcmCryptoSuites;
+      this.enableAes128Sha1_32CryptoCipher = enableAes128Sha1_32CryptoCipher;
+      this.enableEncryptedRtpHeaderExtensions = enableEncryptedRtpHeaderExtensions;
+    }
+
+    @CalledByNative("Srtp")
+    public boolean getEnableGcmCryptoSuites() {
+      return enableGcmCryptoSuites;
+    }
+
+    @CalledByNative("Srtp")
+    public boolean getEnableAes128Sha1_32CryptoCipher() {
+      return enableAes128Sha1_32CryptoCipher;
+    }
+
+    @CalledByNative("Srtp")
+    public boolean getEnableEncryptedRtpHeaderExtensions() {
+      return enableEncryptedRtpHeaderExtensions;
+    }
+  }
+
+  /**
+   * Options to be used when the FrameEncryptor / FrameDecryptor APIs are used.
+   */
+  public final class SFrame {
+    /**
+     * If set all RtpSenders must have an FrameEncryptor attached to them before
+     * they are allowed to send packets. All RtpReceivers must have a
+     * FrameDecryptor attached to them before they are able to receive packets.
+     */
+    private final boolean requireFrameEncryption;
+
+    private SFrame(boolean requireFrameEncryption) {
+      this.requireFrameEncryption = requireFrameEncryption;
+    }
+
+    @CalledByNative("SFrame")
+    public boolean getRequireFrameEncryption() {
+      return requireFrameEncryption;
+    }
+  }
+
+  private final Srtp srtp;
+  private final SFrame sframe;
+
+  private CryptoOptions(boolean enableGcmCryptoSuites, boolean enableAes128Sha1_32CryptoCipher,
+      boolean enableEncryptedRtpHeaderExtensions, boolean requireFrameEncryption) {
+    this.srtp = new Srtp(
+        enableGcmCryptoSuites, enableAes128Sha1_32CryptoCipher, enableEncryptedRtpHeaderExtensions);
+    this.sframe = new SFrame(requireFrameEncryption);
+  }
+
+  public static Builder builder() {
+    return new Builder();
+  }
+
+  @CalledByNative
+  public Srtp getSrtp() {
+    return srtp;
+  }
+
+  @CalledByNative
+  public SFrame getSFrame() {
+    return sframe;
+  }
+
+  public static class Builder {
+    private boolean enableGcmCryptoSuites;
+    private boolean enableAes128Sha1_32CryptoCipher;
+    private boolean enableEncryptedRtpHeaderExtensions;
+    private boolean requireFrameEncryption;
+
+    private Builder() {}
+
+    public Builder setEnableGcmCryptoSuites(boolean enableGcmCryptoSuites) {
+      this.enableGcmCryptoSuites = enableGcmCryptoSuites;
+      return this;
+    }
+
+    public Builder setEnableAes128Sha1_32CryptoCipher(boolean enableAes128Sha1_32CryptoCipher) {
+      this.enableAes128Sha1_32CryptoCipher = enableAes128Sha1_32CryptoCipher;
+      return this;
+    }
+
+    public Builder setEnableEncryptedRtpHeaderExtensions(
+        boolean enableEncryptedRtpHeaderExtensions) {
+      this.enableEncryptedRtpHeaderExtensions = enableEncryptedRtpHeaderExtensions;
+      return this;
+    }
+
+    public Builder setRequireFrameEncryption(boolean requireFrameEncryption) {
+      this.requireFrameEncryption = requireFrameEncryption;
+      return this;
+    }
+
+    public CryptoOptions createCryptoOptions() {
+      return new CryptoOptions(enableGcmCryptoSuites, enableAes128Sha1_32CryptoCipher,
+          enableEncryptedRtpHeaderExtensions, requireFrameEncryption);
+    }
+  }
+}
diff --git a/third_party/libwebrtc/sdk/android/api/org/webrtc/DataChannel.java b/third_party/libwebrtc/sdk/android/api/org/webrtc/DataChannel.java
new file mode 100644
index 0000000000..b9301f1faa
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/api/org/webrtc/DataChannel.java
@@ -0,0 +1,196 @@
+/*
+ *  Copyright 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+import java.nio.ByteBuffer;
+
+/** Java wrapper for a C++ DataChannelInterface. */
+public class DataChannel {
+  /** Java wrapper for WebIDL RTCDataChannel. */
+  public static class Init {
+    public boolean ordered = true;
+    // Optional unsigned short in WebIDL, -1 means unspecified.
+    public int maxRetransmitTimeMs = -1;
+    // Optional unsigned short in WebIDL, -1 means unspecified.
+    public int maxRetransmits = -1;
+    public String protocol = "";
+    public boolean negotiated;
+    // Optional unsigned short in WebIDL, -1 means unspecified.
+    public int id = -1;
+
+    @CalledByNative("Init")
+    boolean getOrdered() {
+      return ordered;
+    }
+
+    @CalledByNative("Init")
+    int getMaxRetransmitTimeMs() {
+      return maxRetransmitTimeMs;
+    }
+
+    @CalledByNative("Init")
+    int getMaxRetransmits() {
+      return maxRetransmits;
+    }
+
+    @CalledByNative("Init")
+    String getProtocol() {
+      return protocol;
+    }
+
+    @CalledByNative("Init")
+    boolean getNegotiated() {
+      return negotiated;
+    }
+
+    @CalledByNative("Init")
+    int getId() {
+      return id;
+    }
+  }
+
+  /** Java version of C++ DataBuffer.  The atom of data in a DataChannel. */
+  public static class Buffer {
+    /** The underlying data. */
+    public final ByteBuffer data;
+
+    /**
+     * Indicates whether `data` contains UTF-8 text or "binary data"
+     * (i.e. anything else).
+     */
+    public final boolean binary;
+
+    @CalledByNative("Buffer")
+    public Buffer(ByteBuffer data, boolean binary) {
+      this.data = data;
+      this.binary = binary;
+    }
+  }
+
+  /** Java version of C++ DataChannelObserver. */
+  public interface Observer {
+    /** The data channel's bufferedAmount has changed. */
+    @CalledByNative("Observer") public void onBufferedAmountChange(long previousAmount);
+    /** The data channel state has changed. */
+    @CalledByNative("Observer") public void onStateChange();
+    /**
+     * A data buffer was successfully received.  NOTE: `buffer.data` will be
+     * freed once this function returns so callers who want to use the data
+     * asynchronously must make sure to copy it first.
+     */
+    @CalledByNative("Observer") public void onMessage(Buffer buffer);
+  }
+
+  /** Keep in sync with DataChannelInterface::DataState. */
+  public enum State {
+    CONNECTING,
+    OPEN,
+    CLOSING,
+    CLOSED;
+
+    @CalledByNative("State")
+    static State fromNativeIndex(int nativeIndex) {
+      return values()[nativeIndex];
+    }
+  }
+
+  private long nativeDataChannel;
+  private long nativeObserver;
+
+  @CalledByNative
+  public DataChannel(long nativeDataChannel) {
+    this.nativeDataChannel = nativeDataChannel;
+  }
+
+  /** Register `observer`, replacing any previously-registered observer. */
+  public void registerObserver(Observer observer) {
+    checkDataChannelExists();
+    if (nativeObserver != 0) {
+      nativeUnregisterObserver(nativeObserver);
+    }
+    nativeObserver = nativeRegisterObserver(observer);
+  }
+
+  /** Unregister the (only) observer. */
+  public void unregisterObserver() {
+    checkDataChannelExists();
+    nativeUnregisterObserver(nativeObserver);
+    nativeObserver = 0;
+  }
+
+  public String label() {
+    checkDataChannelExists();
+    return nativeLabel();
+  }
+
+  public int id() {
+    checkDataChannelExists();
+    return nativeId();
+  }
+
+  public State state() {
+    checkDataChannelExists();
+    return nativeState();
+  }
+
+  /**
+   * Return the number of bytes of application data (UTF-8 text and binary data)
+   * that have been queued using SendBuffer but have not yet been transmitted
+   * to the network.
+   */
+  public long bufferedAmount() {
+    checkDataChannelExists();
+    return nativeBufferedAmount();
+  }
+
+  /** Close the channel. */
+  public void close() {
+    checkDataChannelExists();
+    nativeClose();
+  }
+
+  /** Send `data` to the remote peer; return success. */
+  public boolean send(Buffer buffer) {
+    checkDataChannelExists();
+    // TODO(fischman): this could be cleverer about avoiding copies if the
+    // ByteBuffer is direct and/or is backed by an array.
+    byte[] data = new byte[buffer.data.remaining()];
+    buffer.data.get(data);
+    return nativeSend(data, buffer.binary);
+  }
+
+  /** Dispose of native resources attached to this channel. */
+  public void dispose() {
+    checkDataChannelExists();
+    JniCommon.nativeReleaseRef(nativeDataChannel);
+    nativeDataChannel = 0;
+  }
+
+  @CalledByNative
+  long getNativeDataChannel() {
+    return nativeDataChannel;
+  }
+
+  private void checkDataChannelExists() {
+    if (nativeDataChannel == 0) {
+      throw new IllegalStateException("DataChannel has been disposed.");
+    }
+  }
+
+  private native long nativeRegisterObserver(Observer observer);
+  private native void nativeUnregisterObserver(long observer);
+  private native String nativeLabel();
+  private native int nativeId();
+  private native State nativeState();
+  private native long nativeBufferedAmount();
+  private native void nativeClose();
+  private native boolean nativeSend(byte[] data, boolean binary);
+};
diff --git a/third_party/libwebrtc/sdk/android/api/org/webrtc/Dav1dDecoder.java b/third_party/libwebrtc/sdk/android/api/org/webrtc/Dav1dDecoder.java
new file mode 100644
index 0000000000..ecb16bc3a1
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/api/org/webrtc/Dav1dDecoder.java
@@ -0,0 +1,20 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+public class Dav1dDecoder extends WrappedNativeVideoDecoder {
+  @Override
+  public long createNativeVideoDecoder() {
+    return nativeCreateDecoder();
+  }
+
+  static native long nativeCreateDecoder();
+}
diff --git a/third_party/libwebrtc/sdk/android/api/org/webrtc/DefaultVideoDecoderFactory.java b/third_party/libwebrtc/sdk/android/api/org/webrtc/DefaultVideoDecoderFactory.java
new file mode 100644
index 0000000000..d7a8694d3d
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/api/org/webrtc/DefaultVideoDecoderFactory.java
@@ -0,0 +1,69 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+import androidx.annotation.Nullable;
+import java.util.Arrays;
+import java.util.LinkedHashSet;
+
+/**
+ * Helper class that combines HW and SW decoders.
+ */
+public class DefaultVideoDecoderFactory implements VideoDecoderFactory {
+  private final VideoDecoderFactory hardwareVideoDecoderFactory;
+  private final VideoDecoderFactory softwareVideoDecoderFactory = new SoftwareVideoDecoderFactory();
+  private final @Nullable VideoDecoderFactory platformSoftwareVideoDecoderFactory;
+
+  /**
+   * Create decoder factory using default hardware decoder factory.
+   */
+  public DefaultVideoDecoderFactory(@Nullable EglBase.Context eglContext) {
+    this.hardwareVideoDecoderFactory = new HardwareVideoDecoderFactory(eglContext);
+    this.platformSoftwareVideoDecoderFactory = new PlatformSoftwareVideoDecoderFactory(eglContext);
+  }
+
+  /**
+   * Create decoder factory using explicit hardware decoder factory.
+   */
+  DefaultVideoDecoderFactory(VideoDecoderFactory hardwareVideoDecoderFactory) {
+    this.hardwareVideoDecoderFactory = hardwareVideoDecoderFactory;
+    this.platformSoftwareVideoDecoderFactory = null;
+  }
+
+  @Override
+  public @Nullable VideoDecoder createDecoder(VideoCodecInfo codecType) {
+    VideoDecoder softwareDecoder = softwareVideoDecoderFactory.createDecoder(codecType);
+    final VideoDecoder hardwareDecoder = hardwareVideoDecoderFactory.createDecoder(codecType);
+    if (softwareDecoder == null && platformSoftwareVideoDecoderFactory != null) {
+      softwareDecoder = platformSoftwareVideoDecoderFactory.createDecoder(codecType);
+    }
+    if (hardwareDecoder != null && softwareDecoder != null) {
+      // Both hardware and software supported, wrap it in a software fallback
+      return new VideoDecoderFallback(
+          /* fallback= */ softwareDecoder, /* primary= */ hardwareDecoder);
+    }
+    return hardwareDecoder != null ? hardwareDecoder : softwareDecoder;
+  }
+
+  @Override
+  public VideoCodecInfo[] getSupportedCodecs() {
+    LinkedHashSet<VideoCodecInfo> supportedCodecInfos = new LinkedHashSet<VideoCodecInfo>();
+
+    supportedCodecInfos.addAll(Arrays.asList(softwareVideoDecoderFactory.getSupportedCodecs()));
+    supportedCodecInfos.addAll(Arrays.asList(hardwareVideoDecoderFactory.getSupportedCodecs()));
+    if (platformSoftwareVideoDecoderFactory != null) {
+      supportedCodecInfos.addAll(
+          Arrays.asList(platformSoftwareVideoDecoderFactory.getSupportedCodecs()));
+    }
+
+    return supportedCodecInfos.toArray(new VideoCodecInfo[supportedCodecInfos.size()]);
+  }
+}
diff --git a/third_party/libwebrtc/sdk/android/api/org/webrtc/DtmfSender.java b/third_party/libwebrtc/sdk/android/api/org/webrtc/DtmfSender.java
new file mode 100644
index 0000000000..6549823089
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/api/org/webrtc/DtmfSender.java
@@ -0,0 +1,96 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+/** Java wrapper for a C++ DtmfSenderInterface. */
+public class DtmfSender {
+  private long nativeDtmfSender;
+
+  public DtmfSender(long nativeDtmfSender) {
+    this.nativeDtmfSender = nativeDtmfSender;
+  }
+
+  /**
+   * @return true if this DtmfSender is capable of sending DTMF. Otherwise false.
+   */
+  public boolean canInsertDtmf() {
+    checkDtmfSenderExists();
+    return nativeCanInsertDtmf(nativeDtmfSender);
+  }
+
+  /**
+   * Queues a task that sends the provided DTMF tones.
+   * <p>
+   * If insertDtmf is called on the same object while an existing task for this
+   * object to generate DTMF is still running, the previous task is canceled.
+   *
+   * @param tones        This parameter is treated as a series of characters. The characters 0
+   *                     through 9, A through D, #, and * generate the associated DTMF tones. The
+   *                     characters a to d are equivalent to A to D. The character ',' indicates a
+   *                     delay of 2 seconds before processing the next character in the tones
+   *                     parameter. Unrecognized characters are ignored.
+   * @param duration     Indicates the duration in ms to use for each character passed in the tones
+   *                     parameter. The duration cannot be more than 6000 or less than 70.
+   * @param interToneGap Indicates the gap between tones in ms. Must be at least 50 ms but should be
+   *                     as short as possible.
+   * @return             true on success and false on failure.
+   */
+  public boolean insertDtmf(String tones, int duration, int interToneGap) {
+    checkDtmfSenderExists();
+    return nativeInsertDtmf(nativeDtmfSender, tones, duration, interToneGap);
+  }
+
+  /**
+   * @return The tones remaining to be played out
+   */
+  public String tones() {
+    checkDtmfSenderExists();
+    return nativeTones(nativeDtmfSender);
+  }
+
+  /**
+   * @return The current tone duration value in ms. This value will be the value last set via the
+   *         insertDtmf() method, or the default value of 100 ms if insertDtmf() was never called.
+   */
+  public int duration() {
+    checkDtmfSenderExists();
+    return nativeDuration(nativeDtmfSender);
+  }
+
+  /**
+   * @return The current value of the between-tone gap in ms. This value will be the value last set
+   *         via the insertDtmf() method, or the default value of 50 ms if insertDtmf() was never
+   *         called.
+   */
+  public int interToneGap() {
+    checkDtmfSenderExists();
+    return nativeInterToneGap(nativeDtmfSender);
+  }
+
+  public void dispose() {
+    checkDtmfSenderExists();
+    JniCommon.nativeReleaseRef(nativeDtmfSender);
+    nativeDtmfSender = 0;
+  }
+
+  private void checkDtmfSenderExists() {
+    if (nativeDtmfSender == 0) {
+      throw new IllegalStateException("DtmfSender has been disposed.");
+    }
+  }
+
+  private static native boolean nativeCanInsertDtmf(long dtmfSender);
+  private static native boolean nativeInsertDtmf(
+      long dtmfSender, String tones, int duration, int interToneGap);
+  private static native String nativeTones(long dtmfSender);
+  private static native int nativeDuration(long dtmfSender);
+  private static native int nativeInterToneGap(long dtmfSender);
+};
diff --git a/third_party/libwebrtc/sdk/android/api/org/webrtc/EglBase.java b/third_party/libwebrtc/sdk/android/api/org/webrtc/EglBase.java
new file mode 100644
index 0000000000..64771d004a
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/api/org/webrtc/EglBase.java
@@ -0,0 +1,255 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+import android.graphics.SurfaceTexture;
+import android.view.Surface;
+import androidx.annotation.Nullable;
+import java.util.ArrayList;
+import javax.microedition.khronos.egl.EGL10;
+
+/**
+ * Holds EGL state and utility methods for handling an egl 1.0 EGLContext, an EGLDisplay,
+ * and an EGLSurface.
+ */
+public interface EglBase {
+  // EGL wrapper for an actual EGLContext.
+  public interface Context {
+    public final static long NO_CONTEXT = 0;
+
+    /**
+     * Returns an EGL context that can be used by native code. Returns NO_CONTEXT if the method is
+     * unsupported.
+     *
+     * @note This is currently only supported for EGL 1.4 and not for EGL 1.0.
+     */
+    long getNativeEglContext();
+  }
+
+  // According to the documentation, EGL can be used from multiple threads at the same time if each
+  // thread has its own EGLContext, but in practice it deadlocks on some devices when doing this.
+  // Therefore, synchronize on this global lock before calling dangerous EGL functions that might
+  // deadlock. See https://bugs.chromium.org/p/webrtc/issues/detail?id=5702 for more info.
+  public static final Object lock = new Object();
+
+  // These constants are taken from EGL14.EGL_OPENGL_ES2_BIT and EGL14.EGL_CONTEXT_CLIENT_VERSION.
+  // https://android.googlesource.com/platform/frameworks/base/+/master/opengl/java/android/opengl/EGL14.java
+  // This is similar to how GlSurfaceView does:
+  // http://grepcode.com/file/repository.grepcode.com/java/ext/com.google.android/android/5.1.1_r1/android/opengl/GLSurfaceView.java#760
+  public static final int EGL_OPENGL_ES2_BIT = 4;
+  public static final int EGL_OPENGL_ES3_BIT = 0x40;
+  // Android-specific extension.
+  public static final int EGL_RECORDABLE_ANDROID = 0x3142;
+
+  public static ConfigBuilder configBuilder() {
+    return new ConfigBuilder();
+  }
+
+  public static class ConfigBuilder {
+    private int openGlesVersion = 2;
+    private boolean hasAlphaChannel;
+    private boolean supportsPixelBuffer;
+    private boolean isRecordable;
+
+    public ConfigBuilder setOpenGlesVersion(int version) {
+      if (version < 1 || version > 3) {
+        throw new IllegalArgumentException("OpenGL ES version " + version + " not supported");
+      }
+      this.openGlesVersion = version;
+      return this;
+    }
+
+    public ConfigBuilder setHasAlphaChannel(boolean hasAlphaChannel) {
+      this.hasAlphaChannel = hasAlphaChannel;
+      return this;
+    }
+
+    public ConfigBuilder setSupportsPixelBuffer(boolean supportsPixelBuffer) {
+      this.supportsPixelBuffer = supportsPixelBuffer;
+      return this;
+    }
+
+    public ConfigBuilder setIsRecordable(boolean isRecordable) {
+      this.isRecordable = isRecordable;
+      return this;
+    }
+
+    public int[] createConfigAttributes() {
+      ArrayList<Integer> list = new ArrayList<>();
+      list.add(EGL10.EGL_RED_SIZE);
+      list.add(8);
+      list.add(EGL10.EGL_GREEN_SIZE);
+      list.add(8);
+      list.add(EGL10.EGL_BLUE_SIZE);
+      list.add(8);
+      if (hasAlphaChannel) {
+        list.add(EGL10.EGL_ALPHA_SIZE);
+        list.add(8);
+      }
+      if (openGlesVersion == 2 || openGlesVersion == 3) {
+        list.add(EGL10.EGL_RENDERABLE_TYPE);
+        list.add(openGlesVersion == 3 ? EGL_OPENGL_ES3_BIT : EGL_OPENGL_ES2_BIT);
+      }
+      if (supportsPixelBuffer) {
+        list.add(EGL10.EGL_SURFACE_TYPE);
+        list.add(EGL10.EGL_PBUFFER_BIT);
+      }
+      if (isRecordable) {
+        list.add(EGL_RECORDABLE_ANDROID);
+        list.add(1);
+      }
+      list.add(EGL10.EGL_NONE);
+
+      final int[] res = new int[list.size()];
+      for (int i = 0; i < list.size(); ++i) {
+        res[i] = list.get(i);
+      }
+      return res;
+    }
+  }
+
+  public static final int[] CONFIG_PLAIN = configBuilder().createConfigAttributes();
+  public static final int[] CONFIG_RGBA =
+      configBuilder().setHasAlphaChannel(true).createConfigAttributes();
+  public static final int[] CONFIG_PIXEL_BUFFER =
+      configBuilder().setSupportsPixelBuffer(true).createConfigAttributes();
+  public static final int[] CONFIG_PIXEL_RGBA_BUFFER = configBuilder()
+                                                           .setHasAlphaChannel(true)
+                                                           .setSupportsPixelBuffer(true)
+                                                           .createConfigAttributes();
+  public static final int[] CONFIG_RECORDABLE =
+      configBuilder().setIsRecordable(true).createConfigAttributes();
+
+  static int getOpenGlesVersionFromConfig(int[] configAttributes) {
+    for (int i = 0; i < configAttributes.length - 1; ++i) {
+      if (configAttributes[i] == EGL10.EGL_RENDERABLE_TYPE) {
+        switch (configAttributes[i + 1]) {
+          case EGL_OPENGL_ES2_BIT:
+            return 2;
+          case EGL_OPENGL_ES3_BIT:
+            return 3;
+          default:
+            return 1;
+        }
+      }
+    }
+    // Default to V1 if no renderable type is specified.
+    return 1;
+  }
+
+  /**
+   * Create a new context with the specified config attributes, sharing data with `sharedContext`.
+   * If `sharedContext` is null, a root EGL 1.4 context is created.
+   */
+  public static EglBase create(@Nullable Context sharedContext, int[] configAttributes) {
+    if (sharedContext == null) {
+      return createEgl14(configAttributes);
+    } else if (sharedContext instanceof EglBase14.Context) {
+      return createEgl14((EglBase14.Context) sharedContext, configAttributes);
+    } else if (sharedContext instanceof EglBase10.Context) {
+      return createEgl10((EglBase10.Context) sharedContext, configAttributes);
+    }
+    throw new IllegalArgumentException("Unrecognized Context");
+  }
+
+  /**
+   * Helper function for creating a plain root context. This function will try to create an EGL 1.4
+   * context if possible, and an EGL 1.0 context otherwise.
+   */
+  public static EglBase create() {
+    return create(null /* shaderContext */, CONFIG_PLAIN);
+  }
+
+  /**
+   * Helper function for creating a plain context, sharing data with `sharedContext`. This function
+   * will try to create an EGL 1.4 context if possible, and an EGL 1.0 context otherwise.
+   */
+  public static EglBase create(Context sharedContext) {
+    return create(sharedContext, CONFIG_PLAIN);
+  }
+
+  /** Explicitly create a root EGl 1.0 context with the specified config attributes. */
+  public static EglBase10 createEgl10(int[] configAttributes) {
+    return new EglBase10Impl(/* sharedContext= */ null, configAttributes);
+  }
+
+  /**
+   * Explicitly create a root EGl 1.0 context with the specified config attributes and shared
+   * context.
+   */
+  public static EglBase10 createEgl10(EglBase10.Context sharedContext, int[] configAttributes) {
+    return new EglBase10Impl(
+        sharedContext == null ? null : sharedContext.getRawContext(), configAttributes);
+  }
+
+  /**
+   * Explicitly create a root EGl 1.0 context with the specified config attributes
+   * and shared context.
+   */
+  public static EglBase10 createEgl10(
+      javax.microedition.khronos.egl.EGLContext sharedContext, int[] configAttributes) {
+    return new EglBase10Impl(sharedContext, configAttributes);
+  }
+
+  /** Explicitly create a root EGl 1.4 context with the specified config attributes. */
+  public static EglBase14 createEgl14(int[] configAttributes) {
+    return new EglBase14Impl(/* sharedContext= */ null, configAttributes);
+  }
+
+  /**
+   * Explicitly create a root EGl 1.4 context with the specified config attributes and shared
+   * context.
+   */
+  public static EglBase14 createEgl14(EglBase14.Context sharedContext, int[] configAttributes) {
+    return new EglBase14Impl(
+        sharedContext == null ? null : sharedContext.getRawContext(), configAttributes);
+  }
+
+  /**
+   * Explicitly create a root EGl 1.4 context with the specified config attributes
+   * and shared context.
+   */
+  public static EglBase14 createEgl14(
+      android.opengl.EGLContext sharedContext, int[] configAttributes) {
+    return new EglBase14Impl(sharedContext, configAttributes);
+  }
+
+  void createSurface(Surface surface);
+
+  // Create EGLSurface from the Android SurfaceTexture.
+  void createSurface(SurfaceTexture surfaceTexture);
+
+  // Create dummy 1x1 pixel buffer surface so the context can be made current.
+  void createDummyPbufferSurface();
+
+  void createPbufferSurface(int width, int height);
+
+  Context getEglBaseContext();
+
+  boolean hasSurface();
+
+  int surfaceWidth();
+
+  int surfaceHeight();
+
+  void releaseSurface();
+
+  void release();
+
+  void makeCurrent();
+
+  // Detach the current EGL context, so that it can be made current on another thread.
+  void detachCurrent();
+
+  void swapBuffers();
+
+  void swapBuffers(long presentationTimeStampNs);
+}
diff --git a/third_party/libwebrtc/sdk/android/api/org/webrtc/EglBase10.java b/third_party/libwebrtc/sdk/android/api/org/webrtc/EglBase10.java
new file mode 100644
index 0000000000..f8b0a3c0d0
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/api/org/webrtc/EglBase10.java
@@ -0,0 +1,20 @@
+/*
+ *  Copyright 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+import javax.microedition.khronos.egl.EGLContext;
+
+/** EGL 1.0 implementation of EglBase. */
+public interface EglBase10 extends EglBase {
+  interface Context extends EglBase.Context {
+    EGLContext getRawContext();
+  }
+}
diff --git a/third_party/libwebrtc/sdk/android/api/org/webrtc/EglBase14.java b/third_party/libwebrtc/sdk/android/api/org/webrtc/EglBase14.java
new file mode 100644
index 0000000000..69c89c44dc
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/api/org/webrtc/EglBase14.java
@@ -0,0 +1,20 @@
+/*
+ *  Copyright 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+import android.opengl.EGLContext;
+
+/** EGL 1.4 implementation of EglBase. */
+public interface EglBase14 extends EglBase {
+  interface Context extends EglBase.Context {
+    EGLContext getRawContext();
+  }
+}
diff --git a/third_party/libwebrtc/sdk/android/api/org/webrtc/EglRenderer.java b/third_party/libwebrtc/sdk/android/api/org/webrtc/EglRenderer.java
new file mode 100644
index 0000000000..5ab0868ef3
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/api/org/webrtc/EglRenderer.java
@@ -0,0 +1,787 @@
+/*
+ *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+import android.graphics.Bitmap;
+import android.graphics.Matrix;
+import android.graphics.SurfaceTexture;
+import android.opengl.GLES20;
+import android.os.Handler;
+import android.os.HandlerThread;
+import android.os.Looper;
+import android.os.Message;
+import android.view.Surface;
+import androidx.annotation.Nullable;
+import java.nio.ByteBuffer;
+import java.text.DecimalFormat;
+import java.util.ArrayList;
+import java.util.Iterator;
+import java.util.concurrent.CountDownLatch;
+import java.util.concurrent.TimeUnit;
+
+/**
+ * Implements VideoSink by displaying the video stream on an EGL Surface. This class is intended to
+ * be used as a helper class for rendering on SurfaceViews and TextureViews.
+ */
+public class EglRenderer implements VideoSink {
+  private static final String TAG = "EglRenderer";
+  private static final long LOG_INTERVAL_SEC = 4;
+
+  public interface FrameListener { void onFrame(Bitmap frame); }
+
+  /** Callback for clients to be notified about errors encountered during rendering. */
+  public static interface ErrorCallback {
+    /** Called if GLES20.GL_OUT_OF_MEMORY is encountered during rendering. */
+    void onGlOutOfMemory();
+  }
+
+  private static class FrameListenerAndParams {
+    public final FrameListener listener;
+    public final float scale;
+    public final RendererCommon.GlDrawer drawer;
+    public final boolean applyFpsReduction;
+
+    public FrameListenerAndParams(FrameListener listener, float scale,
+        RendererCommon.GlDrawer drawer, boolean applyFpsReduction) {
+      this.listener = listener;
+      this.scale = scale;
+      this.drawer = drawer;
+      this.applyFpsReduction = applyFpsReduction;
+    }
+  }
+
+  private class EglSurfaceCreation implements Runnable {
+    private Object surface;
+
+    // TODO(bugs.webrtc.org/8491): Remove NoSynchronizedMethodCheck suppression.
+    @SuppressWarnings("NoSynchronizedMethodCheck")
+    public synchronized void setSurface(Object surface) {
+      this.surface = surface;
+    }
+
+    @Override
+    // TODO(bugs.webrtc.org/8491): Remove NoSynchronizedMethodCheck suppression.
+    @SuppressWarnings("NoSynchronizedMethodCheck")
+    public synchronized void run() {
+      if (surface != null && eglBase != null && !eglBase.hasSurface()) {
+        if (surface instanceof Surface) {
+          eglBase.createSurface((Surface) surface);
+        } else if (surface instanceof SurfaceTexture) {
+          eglBase.createSurface((SurfaceTexture) surface);
+        } else {
+          throw new IllegalStateException("Invalid surface: " + surface);
+        }
+        eglBase.makeCurrent();
+        // Necessary for YUV frames with odd width.
+        GLES20.glPixelStorei(GLES20.GL_UNPACK_ALIGNMENT, 1);
+      }
+    }
+  }
+
+  /**
+   * Handler that triggers a callback when an uncaught exception happens when handling a message.
+   */
+  private static class HandlerWithExceptionCallback extends Handler {
+    private final Runnable exceptionCallback;
+
+    public HandlerWithExceptionCallback(Looper looper, Runnable exceptionCallback) {
+      super(looper);
+      this.exceptionCallback = exceptionCallback;
+    }
+
+    @Override
+    public void dispatchMessage(Message msg) {
+      try {
+        super.dispatchMessage(msg);
+      } catch (Exception e) {
+        Logging.e(TAG, "Exception on EglRenderer thread", e);
+        exceptionCallback.run();
+        throw e;
+      }
+    }
+  }
+
+  protected final String name;
+
+  // `renderThreadHandler` is a handler for communicating with `renderThread`, and is synchronized
+  // on `handlerLock`.
+  private final Object handlerLock = new Object();
+  @Nullable private Handler renderThreadHandler;
+
+  private final ArrayList<FrameListenerAndParams> frameListeners = new ArrayList<>();
+
+  private volatile ErrorCallback errorCallback;
+
+  // Variables for fps reduction.
+  private final Object fpsReductionLock = new Object();
+  // Time for when next frame should be rendered.
+  private long nextFrameTimeNs;
+  // Minimum duration between frames when fps reduction is active, or -1 if video is completely
+  // paused.
+  private long minRenderPeriodNs;
+
+  // EGL and GL resources for drawing YUV/OES textures. After initialization, these are only
+  // accessed from the render thread.
+  @Nullable private EglBase eglBase;
+  private final VideoFrameDrawer frameDrawer;
+  @Nullable private RendererCommon.GlDrawer drawer;
+  private boolean usePresentationTimeStamp;
+  private final Matrix drawMatrix = new Matrix();
+
+  // Pending frame to render. Serves as a queue with size 1. Synchronized on `frameLock`.
+  private final Object frameLock = new Object();
+  @Nullable private VideoFrame pendingFrame;
+
+  // These variables are synchronized on `layoutLock`.
+  private final Object layoutLock = new Object();
+  private float layoutAspectRatio;
+  // If true, mirrors the video stream horizontally.
+  private boolean mirrorHorizontally;
+  // If true, mirrors the video stream vertically.
+  private boolean mirrorVertically;
+
+  // These variables are synchronized on `statisticsLock`.
+  private final Object statisticsLock = new Object();
+  // Total number of video frames received in renderFrame() call.
+  private int framesReceived;
+  // Number of video frames dropped by renderFrame() because previous frame has not been rendered
+  // yet.
+  private int framesDropped;
+  // Number of rendered video frames.
+  private int framesRendered;
+  // Start time for counting these statistics, or 0 if we haven't started measuring yet.
+  private long statisticsStartTimeNs;
+  // Time in ns spent in renderFrameOnRenderThread() function.
+  private long renderTimeNs;
+  // Time in ns spent by the render thread in the swapBuffers() function.
+  private long renderSwapBufferTimeNs;
+
+  // Used for bitmap capturing.
+  private final GlTextureFrameBuffer bitmapTextureFramebuffer =
+      new GlTextureFrameBuffer(GLES20.GL_RGBA);
+
+  private final Runnable logStatisticsRunnable = new Runnable() {
+    @Override
+    public void run() {
+      logStatistics();
+      synchronized (handlerLock) {
+        if (renderThreadHandler != null) {
+          renderThreadHandler.removeCallbacks(logStatisticsRunnable);
+          renderThreadHandler.postDelayed(
+              logStatisticsRunnable, TimeUnit.SECONDS.toMillis(LOG_INTERVAL_SEC));
+        }
+      }
+    }
+  };
+
+  private final EglSurfaceCreation eglSurfaceCreationRunnable = new EglSurfaceCreation();
+
+  /**
+   * Standard constructor. The name will be used for the render thread name and included when
+   * logging. In order to render something, you must first call init() and createEglSurface.
+   */
+  public EglRenderer(String name) {
+    this(name, new VideoFrameDrawer());
+  }
+
+  public EglRenderer(String name, VideoFrameDrawer videoFrameDrawer) {
+    this.name = name;
+    this.frameDrawer = videoFrameDrawer;
+  }
+
+  /**
+   * Initialize this class, sharing resources with `sharedContext`. The custom `drawer` will be used
+   * for drawing frames on the EGLSurface. This class is responsible for calling release() on
+   * `drawer`. It is allowed to call init() to reinitialize the renderer after a previous
+   * init()/release() cycle. If usePresentationTimeStamp is true, eglPresentationTimeANDROID will be
+   * set with the frame timestamps, which specifies desired presentation time and might be useful
+   * for e.g. syncing audio and video.
+   */
+  public void init(@Nullable final EglBase.Context sharedContext, final int[] configAttributes,
+      RendererCommon.GlDrawer drawer, boolean usePresentationTimeStamp) {
+    synchronized (handlerLock) {
+      if (renderThreadHandler != null) {
+        throw new IllegalStateException(name + "Already initialized");
+      }
+      logD("Initializing EglRenderer");
+      this.drawer = drawer;
+      this.usePresentationTimeStamp = usePresentationTimeStamp;
+
+      final HandlerThread renderThread = new HandlerThread(name + "EglRenderer");
+      renderThread.start();
+      renderThreadHandler =
+          new HandlerWithExceptionCallback(renderThread.getLooper(), new Runnable() {
+            @Override
+            public void run() {
+              synchronized (handlerLock) {
+                renderThreadHandler = null;
+              }
+            }
+          });
+      // Create EGL context on the newly created render thread. It should be possibly to create the
+      // context on this thread and make it current on the render thread, but this causes failure on
+      // some Marvel based JB devices. https://bugs.chromium.org/p/webrtc/issues/detail?id=6350.
+      ThreadUtils.invokeAtFrontUninterruptibly(renderThreadHandler, () -> {
+        // If sharedContext is null, then texture frames are disabled. This is typically for old
+        // devices that might not be fully spec compliant, so force EGL 1.0 since EGL 1.4 has
+        // caused trouble on some weird devices.
+        if (sharedContext == null) {
+          logD("EglBase10.create context");
+          eglBase = EglBase.createEgl10(configAttributes);
+        } else {
+          logD("EglBase.create shared context");
+          eglBase = EglBase.create(sharedContext, configAttributes);
+        }
+      });
+      renderThreadHandler.post(eglSurfaceCreationRunnable);
+      final long currentTimeNs = System.nanoTime();
+      resetStatistics(currentTimeNs);
+      renderThreadHandler.postDelayed(
+          logStatisticsRunnable, TimeUnit.SECONDS.toMillis(LOG_INTERVAL_SEC));
+    }
+  }
+
+  /**
+   * Same as above with usePresentationTimeStamp set to false.
+   *
+   * @see #init(EglBase.Context, int[], RendererCommon.GlDrawer, boolean)
+   */
+  public void init(@Nullable final EglBase.Context sharedContext, final int[] configAttributes,
+      RendererCommon.GlDrawer drawer) {
+    init(sharedContext, configAttributes, drawer, /* usePresentationTimeStamp= */ false);
+  }
+
+  public void createEglSurface(Surface surface) {
+    createEglSurfaceInternal(surface);
+  }
+
+  public void createEglSurface(SurfaceTexture surfaceTexture) {
+    createEglSurfaceInternal(surfaceTexture);
+  }
+
+  private void createEglSurfaceInternal(Object surface) {
+    eglSurfaceCreationRunnable.setSurface(surface);
+    postToRenderThread(eglSurfaceCreationRunnable);
+  }
+
+  /**
+   * Block until any pending frame is returned and all GL resources released, even if an interrupt
+   * occurs. If an interrupt occurs during release(), the interrupt flag will be set. This function
+   * should be called before the Activity is destroyed and the EGLContext is still valid. If you
+   * don't call this function, the GL resources might leak.
+   */
+  public void release() {
+    logD("Releasing.");
+    final CountDownLatch eglCleanupBarrier = new CountDownLatch(1);
+    synchronized (handlerLock) {
+      if (renderThreadHandler == null) {
+        logD("Already released");
+        return;
+      }
+      renderThreadHandler.removeCallbacks(logStatisticsRunnable);
+      // Release EGL and GL resources on render thread.
+      renderThreadHandler.postAtFrontOfQueue(() -> {
+        // Detach current shader program.
+        synchronized (EglBase.lock) {
+          GLES20.glUseProgram(/* program= */ 0);
+        }
+        if (drawer != null) {
+          drawer.release();
+          drawer = null;
+        }
+        frameDrawer.release();
+        bitmapTextureFramebuffer.release();
+        if (eglBase != null) {
+          logD("eglBase detach and release.");
+          eglBase.detachCurrent();
+          eglBase.release();
+          eglBase = null;
+        }
+        frameListeners.clear();
+        eglCleanupBarrier.countDown();
+      });
+      final Looper renderLooper = renderThreadHandler.getLooper();
+      // TODO(magjed): Replace this post() with renderLooper.quitSafely() when API support >= 18.
+      renderThreadHandler.post(() -> {
+        logD("Quitting render thread.");
+        renderLooper.quit();
+      });
+      // Don't accept any more frames or messages to the render thread.
+      renderThreadHandler = null;
+    }
+    // Make sure the EGL/GL cleanup posted above is executed.
+    ThreadUtils.awaitUninterruptibly(eglCleanupBarrier);
+    synchronized (frameLock) {
+      if (pendingFrame != null) {
+        pendingFrame.release();
+        pendingFrame = null;
+      }
+    }
+    logD("Releasing done.");
+  }
+
+  /**
+   * Reset the statistics logged in logStatistics().
+   */
+  private void resetStatistics(long currentTimeNs) {
+    synchronized (statisticsLock) {
+      statisticsStartTimeNs = currentTimeNs;
+      framesReceived = 0;
+      framesDropped = 0;
+      framesRendered = 0;
+      renderTimeNs = 0;
+      renderSwapBufferTimeNs = 0;
+    }
+  }
+
+  public void printStackTrace() {
+    synchronized (handlerLock) {
+      final Thread renderThread =
+          (renderThreadHandler == null) ? null : renderThreadHandler.getLooper().getThread();
+      if (renderThread != null) {
+        final StackTraceElement[] renderStackTrace = renderThread.getStackTrace();
+        if (renderStackTrace.length > 0) {
+          logW("EglRenderer stack trace:");
+          for (StackTraceElement traceElem : renderStackTrace) {
+            logW(traceElem.toString());
+          }
+        }
+      }
+    }
+  }
+
+  /**
+   * Set if the video stream should be mirrored horizontally or not.
+   */
+  public void setMirror(final boolean mirror) {
+    logD("setMirrorHorizontally: " + mirror);
+    synchronized (layoutLock) {
+      this.mirrorHorizontally = mirror;
+    }
+  }
+
+  /**
+   * Set if the video stream should be mirrored vertically or not.
+   */
+  public void setMirrorVertically(final boolean mirrorVertically) {
+    logD("setMirrorVertically: " + mirrorVertically);
+    synchronized (layoutLock) {
+      this.mirrorVertically = mirrorVertically;
+    }
+  }
+
+  /**
+   * Set layout aspect ratio. This is used to crop frames when rendering to avoid stretched video.
+   * Set this to 0 to disable cropping.
+   */
+  public void setLayoutAspectRatio(float layoutAspectRatio) {
+    logD("setLayoutAspectRatio: " + layoutAspectRatio);
+    synchronized (layoutLock) {
+      this.layoutAspectRatio = layoutAspectRatio;
+    }
+  }
+
+  /**
+   * Limit render framerate.
+   *
+   * @param fps Limit render framerate to this value, or use Float.POSITIVE_INFINITY to disable fps
+   *            reduction.
+   */
+  public void setFpsReduction(float fps) {
+    logD("setFpsReduction: " + fps);
+    synchronized (fpsReductionLock) {
+      final long previousRenderPeriodNs = minRenderPeriodNs;
+      if (fps <= 0) {
+        minRenderPeriodNs = Long.MAX_VALUE;
+      } else {
+        minRenderPeriodNs = (long) (TimeUnit.SECONDS.toNanos(1) / fps);
+      }
+      if (minRenderPeriodNs != previousRenderPeriodNs) {
+        // Fps reduction changed - reset frame time.
+        nextFrameTimeNs = System.nanoTime();
+      }
+    }
+  }
+
+  public void disableFpsReduction() {
+    setFpsReduction(Float.POSITIVE_INFINITY /* fps */);
+  }
+
+  public void pauseVideo() {
+    setFpsReduction(0 /* fps */);
+  }
+
+  /**
+   * Register a callback to be invoked when a new video frame has been received. This version uses
+   * the drawer of the EglRenderer that was passed in init.
+   *
+   * @param listener The callback to be invoked. The callback will be invoked on the render thread.
+   *                 It should be lightweight and must not call removeFrameListener.
+   * @param scale    The scale of the Bitmap passed to the callback, or 0 if no Bitmap is
+   *                 required.
+   */
+  public void addFrameListener(final FrameListener listener, final float scale) {
+    addFrameListener(listener, scale, null, false /* applyFpsReduction */);
+  }
+
+  /**
+   * Register a callback to be invoked when a new video frame has been received.
+   *
+   * @param listener The callback to be invoked. The callback will be invoked on the render thread.
+   *                 It should be lightweight and must not call removeFrameListener.
+   * @param scale    The scale of the Bitmap passed to the callback, or 0 if no Bitmap is
+   *                 required.
+   * @param drawer   Custom drawer to use for this frame listener or null to use the default one.
+   */
+  public void addFrameListener(
+      final FrameListener listener, final float scale, final RendererCommon.GlDrawer drawerParam) {
+    addFrameListener(listener, scale, drawerParam, false /* applyFpsReduction */);
+  }
+
+  /**
+   * Register a callback to be invoked when a new video frame has been received.
+   *
+   * @param listener The callback to be invoked. The callback will be invoked on the render thread.
+   *                 It should be lightweight and must not call removeFrameListener.
+   * @param scale    The scale of the Bitmap passed to the callback, or 0 if no Bitmap is
+   *                 required.
+   * @param drawer   Custom drawer to use for this frame listener or null to use the default one.
+   * @param applyFpsReduction This callback will not be called for frames that have been dropped by
+   *                          FPS reduction.
+   */
+  public void addFrameListener(final FrameListener listener, final float scale,
+      @Nullable final RendererCommon.GlDrawer drawerParam, final boolean applyFpsReduction) {
+    postToRenderThread(() -> {
+      final RendererCommon.GlDrawer listenerDrawer = drawerParam == null ? drawer : drawerParam;
+      frameListeners.add(
+          new FrameListenerAndParams(listener, scale, listenerDrawer, applyFpsReduction));
+    });
+  }
+
+  /**
+   * Remove any pending callback that was added with addFrameListener. If the callback is not in
+   * the queue, nothing happens. It is ensured that callback won't be called after this method
+   * returns.
+   *
+   * @param runnable The callback to remove.
+   */
+  public void removeFrameListener(final FrameListener listener) {
+    final CountDownLatch latch = new CountDownLatch(1);
+    synchronized (handlerLock) {
+      if (renderThreadHandler == null) {
+        return;
+      }
+      if (Thread.currentThread() == renderThreadHandler.getLooper().getThread()) {
+        throw new RuntimeException("removeFrameListener must not be called on the render thread.");
+      }
+      postToRenderThread(() -> {
+        latch.countDown();
+        final Iterator<FrameListenerAndParams> iter = frameListeners.iterator();
+        while (iter.hasNext()) {
+          if (iter.next().listener == listener) {
+            iter.remove();
+          }
+        }
+      });
+    }
+    ThreadUtils.awaitUninterruptibly(latch);
+  }
+
+  /** Can be set in order to be notified about errors encountered during rendering. */
+  public void setErrorCallback(ErrorCallback errorCallback) {
+    this.errorCallback = errorCallback;
+  }
+
+  // VideoSink interface.
+  @Override
+  public void onFrame(VideoFrame frame) {
+    synchronized (statisticsLock) {
+      ++framesReceived;
+    }
+    final boolean dropOldFrame;
+    synchronized (handlerLock) {
+      if (renderThreadHandler == null) {
+        logD("Dropping frame - Not initialized or already released.");
+        return;
+      }
+      synchronized (frameLock) {
+        dropOldFrame = (pendingFrame != null);
+        if (dropOldFrame) {
+          pendingFrame.release();
+        }
+        pendingFrame = frame;
+        pendingFrame.retain();
+        renderThreadHandler.post(this ::renderFrameOnRenderThread);
+      }
+    }
+    if (dropOldFrame) {
+      synchronized (statisticsLock) {
+        ++framesDropped;
+      }
+    }
+  }
+
+  /**
+   * Release EGL surface. This function will block until the EGL surface is released.
+   */
+  public void releaseEglSurface(final Runnable completionCallback) {
+    // Ensure that the render thread is no longer touching the Surface before returning from this
+    // function.
+    eglSurfaceCreationRunnable.setSurface(null /* surface */);
+    synchronized (handlerLock) {
+      if (renderThreadHandler != null) {
+        renderThreadHandler.removeCallbacks(eglSurfaceCreationRunnable);
+        renderThreadHandler.postAtFrontOfQueue(() -> {
+          if (eglBase != null) {
+            eglBase.detachCurrent();
+            eglBase.releaseSurface();
+          }
+          completionCallback.run();
+        });
+        return;
+      }
+    }
+    completionCallback.run();
+  }
+
+  /**
+   * Private helper function to post tasks safely.
+   */
+  private void postToRenderThread(Runnable runnable) {
+    synchronized (handlerLock) {
+      if (renderThreadHandler != null) {
+        renderThreadHandler.post(runnable);
+      }
+    }
+  }
+
+  private void clearSurfaceOnRenderThread(float r, float g, float b, float a) {
+    if (eglBase != null && eglBase.hasSurface()) {
+      logD("clearSurface");
+      GLES20.glClearColor(r, g, b, a);
+      GLES20.glClear(GLES20.GL_COLOR_BUFFER_BIT);
+      eglBase.swapBuffers();
+    }
+  }
+
+  /**
+   * Post a task to clear the surface to a transparent uniform color.
+   */
+  public void clearImage() {
+    clearImage(0 /* red */, 0 /* green */, 0 /* blue */, 0 /* alpha */);
+  }
+
+  /**
+   * Post a task to clear the surface to a specific color.
+   */
+  public void clearImage(final float r, final float g, final float b, final float a) {
+    synchronized (handlerLock) {
+      if (renderThreadHandler == null) {
+        return;
+      }
+      renderThreadHandler.postAtFrontOfQueue(() -> clearSurfaceOnRenderThread(r, g, b, a));
+    }
+  }
+
+  /**
+   * Renders and releases `pendingFrame`.
+   */
+  private void renderFrameOnRenderThread() {
+    // Fetch and render `pendingFrame`.
+    final VideoFrame frame;
+    synchronized (frameLock) {
+      if (pendingFrame == null) {
+        return;
+      }
+      frame = pendingFrame;
+      pendingFrame = null;
+    }
+    if (eglBase == null || !eglBase.hasSurface()) {
+      logD("Dropping frame - No surface");
+      frame.release();
+      return;
+    }
+    // Check if fps reduction is active.
+    final boolean shouldRenderFrame;
+    synchronized (fpsReductionLock) {
+      if (minRenderPeriodNs == Long.MAX_VALUE) {
+        // Rendering is paused.
+        shouldRenderFrame = false;
+      } else if (minRenderPeriodNs <= 0) {
+        // FPS reduction is disabled.
+        shouldRenderFrame = true;
+      } else {
+        final long currentTimeNs = System.nanoTime();
+        if (currentTimeNs < nextFrameTimeNs) {
+          logD("Skipping frame rendering - fps reduction is active.");
+          shouldRenderFrame = false;
+        } else {
+          nextFrameTimeNs += minRenderPeriodNs;
+          // The time for the next frame should always be in the future.
+          nextFrameTimeNs = Math.max(nextFrameTimeNs, currentTimeNs);
+          shouldRenderFrame = true;
+        }
+      }
+    }
+
+    final long startTimeNs = System.nanoTime();
+
+    final float frameAspectRatio = frame.getRotatedWidth() / (float) frame.getRotatedHeight();
+    final float drawnAspectRatio;
+    synchronized (layoutLock) {
+      drawnAspectRatio = layoutAspectRatio != 0f ? layoutAspectRatio : frameAspectRatio;
+    }
+
+    final float scaleX;
+    final float scaleY;
+
+    if (frameAspectRatio > drawnAspectRatio) {
+      scaleX = drawnAspectRatio / frameAspectRatio;
+      scaleY = 1f;
+    } else {
+      scaleX = 1f;
+      scaleY = frameAspectRatio / drawnAspectRatio;
+    }
+
+    drawMatrix.reset();
+    drawMatrix.preTranslate(0.5f, 0.5f);
+    drawMatrix.preScale(mirrorHorizontally ? -1f : 1f, mirrorVertically ? -1f : 1f);
+    drawMatrix.preScale(scaleX, scaleY);
+    drawMatrix.preTranslate(-0.5f, -0.5f);
+
+    try {
+      if (shouldRenderFrame) {
+        GLES20.glClearColor(0 /* red */, 0 /* green */, 0 /* blue */, 0 /* alpha */);
+        GLES20.glClear(GLES20.GL_COLOR_BUFFER_BIT);
+        frameDrawer.drawFrame(frame, drawer, drawMatrix, 0 /* viewportX */, 0 /* viewportY */,
+            eglBase.surfaceWidth(), eglBase.surfaceHeight());
+
+        final long swapBuffersStartTimeNs = System.nanoTime();
+        if (usePresentationTimeStamp) {
+          eglBase.swapBuffers(frame.getTimestampNs());
+        } else {
+          eglBase.swapBuffers();
+        }
+
+        final long currentTimeNs = System.nanoTime();
+        synchronized (statisticsLock) {
+          ++framesRendered;
+          renderTimeNs += (currentTimeNs - startTimeNs);
+          renderSwapBufferTimeNs += (currentTimeNs - swapBuffersStartTimeNs);
+        }
+      }
+
+      notifyCallbacks(frame, shouldRenderFrame);
+    } catch (GlUtil.GlOutOfMemoryException e) {
+      logE("Error while drawing frame", e);
+      final ErrorCallback errorCallback = this.errorCallback;
+      if (errorCallback != null) {
+        errorCallback.onGlOutOfMemory();
+      }
+      // Attempt to free up some resources.
+      drawer.release();
+      frameDrawer.release();
+      bitmapTextureFramebuffer.release();
+      // Continue here on purpose and retry again for next frame. In worst case, this is a continous
+      // problem and no more frames will be drawn.
+    } finally {
+      frame.release();
+    }
+  }
+
+  private void notifyCallbacks(VideoFrame frame, boolean wasRendered) {
+    if (frameListeners.isEmpty())
+      return;
+
+    drawMatrix.reset();
+    drawMatrix.preTranslate(0.5f, 0.5f);
+    drawMatrix.preScale(mirrorHorizontally ? -1f : 1f, mirrorVertically ? -1f : 1f);
+    drawMatrix.preScale(1f, -1f); // We want the output to be upside down for Bitmap.
+    drawMatrix.preTranslate(-0.5f, -0.5f);
+
+    Iterator<FrameListenerAndParams> it = frameListeners.iterator();
+    while (it.hasNext()) {
+      FrameListenerAndParams listenerAndParams = it.next();
+      if (!wasRendered && listenerAndParams.applyFpsReduction) {
+        continue;
+      }
+      it.remove();
+
+      final int scaledWidth = (int) (listenerAndParams.scale * frame.getRotatedWidth());
+      final int scaledHeight = (int) (listenerAndParams.scale * frame.getRotatedHeight());
+
+      if (scaledWidth == 0 || scaledHeight == 0) {
+        listenerAndParams.listener.onFrame(null);
+        continue;
+      }
+
+      bitmapTextureFramebuffer.setSize(scaledWidth, scaledHeight);
+
+      GLES20.glBindFramebuffer(GLES20.GL_FRAMEBUFFER, bitmapTextureFramebuffer.getFrameBufferId());
+      GLES20.glFramebufferTexture2D(GLES20.GL_FRAMEBUFFER, GLES20.GL_COLOR_ATTACHMENT0,
+          GLES20.GL_TEXTURE_2D, bitmapTextureFramebuffer.getTextureId(), 0);
+
+      GLES20.glClearColor(0 /* red */, 0 /* green */, 0 /* blue */, 0 /* alpha */);
+      GLES20.glClear(GLES20.GL_COLOR_BUFFER_BIT);
+      frameDrawer.drawFrame(frame, listenerAndParams.drawer, drawMatrix, 0 /* viewportX */,
+          0 /* viewportY */, scaledWidth, scaledHeight);
+
+      final ByteBuffer bitmapBuffer = ByteBuffer.allocateDirect(scaledWidth * scaledHeight * 4);
+      GLES20.glViewport(0, 0, scaledWidth, scaledHeight);
+      GLES20.glReadPixels(
+          0, 0, scaledWidth, scaledHeight, GLES20.GL_RGBA, GLES20.GL_UNSIGNED_BYTE, bitmapBuffer);
+
+      GLES20.glBindFramebuffer(GLES20.GL_FRAMEBUFFER, 0);
+      GlUtil.checkNoGLES2Error("EglRenderer.notifyCallbacks");
+
+      final Bitmap bitmap = Bitmap.createBitmap(scaledWidth, scaledHeight, Bitmap.Config.ARGB_8888);
+      bitmap.copyPixelsFromBuffer(bitmapBuffer);
+      listenerAndParams.listener.onFrame(bitmap);
+    }
+  }
+
+  private String averageTimeAsString(long sumTimeNs, int count) {
+    return (count <= 0) ? "NA" : TimeUnit.NANOSECONDS.toMicros(sumTimeNs / count) + " us";
+  }
+
+  private void logStatistics() {
+    final DecimalFormat fpsFormat = new DecimalFormat("#.0");
+    final long currentTimeNs = System.nanoTime();
+    synchronized (statisticsLock) {
+      final long elapsedTimeNs = currentTimeNs - statisticsStartTimeNs;
+      if (elapsedTimeNs <= 0 || (minRenderPeriodNs == Long.MAX_VALUE && framesReceived == 0)) {
+        return;
+      }
+      final float renderFps = framesRendered * TimeUnit.SECONDS.toNanos(1) / (float) elapsedTimeNs;
+      logD("Duration: " + TimeUnit.NANOSECONDS.toMillis(elapsedTimeNs) + " ms."
+          + " Frames received: " + framesReceived + "."
+          + " Dropped: " + framesDropped + "."
+          + " Rendered: " + framesRendered + "."
+          + " Render fps: " + fpsFormat.format(renderFps) + "."
+          + " Average render time: " + averageTimeAsString(renderTimeNs, framesRendered) + "."
+          + " Average swapBuffer time: "
+          + averageTimeAsString(renderSwapBufferTimeNs, framesRendered) + ".");
+      resetStatistics(currentTimeNs);
+    }
+  }
+
+  private void logE(String string, Throwable e) {
+    Logging.e(TAG, name + string, e);
+  }
+
+  private void logD(String string) {
+    Logging.d(TAG, name + string);
+  }
+
+  private void logW(String string) {
+    Logging.w(TAG, name + string);
+  }
+}
diff --git a/third_party/libwebrtc/sdk/android/api/org/webrtc/EncodedImage.java b/third_party/libwebrtc/sdk/android/api/org/webrtc/EncodedImage.java
new file mode 100644
index 0000000000..a6eef67da8
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/api/org/webrtc/EncodedImage.java
@@ -0,0 +1,183 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+import androidx.annotation.Nullable;
+import java.nio.ByteBuffer;
+import java.util.concurrent.TimeUnit;
+
+/**
+ * An encoded frame from a video stream. Used as an input for decoders and as an output for
+ * encoders.
+ */
+public class EncodedImage implements RefCounted {
+  // Must be kept in sync with common_types.h FrameType.
+  public enum FrameType {
+    EmptyFrame(0),
+    VideoFrameKey(3),
+    VideoFrameDelta(4);
+
+    private final int nativeIndex;
+
+    private FrameType(int nativeIndex) {
+      this.nativeIndex = nativeIndex;
+    }
+
+    public int getNative() {
+      return nativeIndex;
+    }
+
+    @CalledByNative("FrameType")
+    static FrameType fromNativeIndex(int nativeIndex) {
+      for (FrameType type : FrameType.values()) {
+        if (type.getNative() == nativeIndex) {
+          return type;
+        }
+      }
+      throw new IllegalArgumentException("Unknown native frame type: " + nativeIndex);
+    }
+  }
+
+  private final RefCountDelegate refCountDelegate;
+  public final ByteBuffer buffer;
+  public final int encodedWidth;
+  public final int encodedHeight;
+  public final long captureTimeMs; // Deprecated
+  public final long captureTimeNs;
+  public final FrameType frameType;
+  public final int rotation;
+  public final @Nullable Integer qp;
+
+  // TODO(bugs.webrtc.org/9378): Use retain and release from jni code.
+  @Override
+  public void retain() {
+    refCountDelegate.retain();
+  }
+
+  @Override
+  public void release() {
+    refCountDelegate.release();
+  }
+
+  @CalledByNative
+  private EncodedImage(ByteBuffer buffer, @Nullable Runnable releaseCallback, int encodedWidth,
+      int encodedHeight, long captureTimeNs, FrameType frameType, int rotation,
+      @Nullable Integer qp) {
+    this.buffer = buffer;
+    this.encodedWidth = encodedWidth;
+    this.encodedHeight = encodedHeight;
+    this.captureTimeMs = TimeUnit.NANOSECONDS.toMillis(captureTimeNs);
+    this.captureTimeNs = captureTimeNs;
+    this.frameType = frameType;
+    this.rotation = rotation;
+    this.qp = qp;
+    this.refCountDelegate = new RefCountDelegate(releaseCallback);
+  }
+
+  @CalledByNative
+  private ByteBuffer getBuffer() {
+    return buffer;
+  }
+
+  @CalledByNative
+  private int getEncodedWidth() {
+    return encodedWidth;
+  }
+
+  @CalledByNative
+  private int getEncodedHeight() {
+    return encodedHeight;
+  }
+
+  @CalledByNative
+  private long getCaptureTimeNs() {
+    return captureTimeNs;
+  }
+
+  @CalledByNative
+  private int getFrameType() {
+    return frameType.getNative();
+  }
+
+  @CalledByNative
+  private int getRotation() {
+    return rotation;
+  }
+
+  @CalledByNative
+  private @Nullable Integer getQp() {
+    return qp;
+  }
+
+  public static Builder builder() {
+    return new Builder();
+  }
+
+  public static class Builder {
+    private ByteBuffer buffer;
+    private @Nullable Runnable releaseCallback;
+    private int encodedWidth;
+    private int encodedHeight;
+    private long captureTimeNs;
+    private EncodedImage.FrameType frameType;
+    private int rotation;
+    private @Nullable Integer qp;
+
+    private Builder() {}
+
+    public Builder setBuffer(ByteBuffer buffer, @Nullable Runnable releaseCallback) {
+      this.buffer = buffer;
+      this.releaseCallback = releaseCallback;
+      return this;
+    }
+
+    public Builder setEncodedWidth(int encodedWidth) {
+      this.encodedWidth = encodedWidth;
+      return this;
+    }
+
+    public Builder setEncodedHeight(int encodedHeight) {
+      this.encodedHeight = encodedHeight;
+      return this;
+    }
+
+    @Deprecated
+    public Builder setCaptureTimeMs(long captureTimeMs) {
+      this.captureTimeNs = TimeUnit.MILLISECONDS.toNanos(captureTimeMs);
+      return this;
+    }
+
+    public Builder setCaptureTimeNs(long captureTimeNs) {
+      this.captureTimeNs = captureTimeNs;
+      return this;
+    }
+
+    public Builder setFrameType(EncodedImage.FrameType frameType) {
+      this.frameType = frameType;
+      return this;
+    }
+
+    public Builder setRotation(int rotation) {
+      this.rotation = rotation;
+      return this;
+    }
+
+    public Builder setQp(@Nullable Integer qp) {
+      this.qp = qp;
+      return this;
+    }
+
+    public EncodedImage createEncodedImage() {
+      return new EncodedImage(buffer, releaseCallback, encodedWidth, encodedHeight, captureTimeNs,
+          frameType, rotation, qp);
+    }
+  }
+}
diff --git a/third_party/libwebrtc/sdk/android/api/org/webrtc/FecControllerFactoryFactoryInterface.java b/third_party/libwebrtc/sdk/android/api/org/webrtc/FecControllerFactoryFactoryInterface.java
new file mode 100644
index 0000000000..6d39390f72
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/api/org/webrtc/FecControllerFactoryFactoryInterface.java
@@ -0,0 +1,22 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+/**
+ * Factory for creating webrtc::FecControllerFactory instances.
+ */
+public interface FecControllerFactoryFactoryInterface {
+  /**
+   * Dynamically allocates a webrtc::FecControllerFactory instance and returns a pointer to it.
+   * The caller takes ownership of the object.
+   */
+  public long createNative();
+}
diff --git a/third_party/libwebrtc/sdk/android/api/org/webrtc/FileVideoCapturer.java b/third_party/libwebrtc/sdk/android/api/org/webrtc/FileVideoCapturer.java
new file mode 100644
index 0000000000..8270367970
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/api/org/webrtc/FileVideoCapturer.java
@@ -0,0 +1,201 @@
+/*
+ *  Copyright 2016 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+import android.content.Context;
+import android.os.SystemClock;
+import java.io.IOException;
+import java.io.RandomAccessFile;
+import java.nio.ByteBuffer;
+import java.nio.channels.FileChannel;
+import java.nio.charset.Charset;
+import java.util.Timer;
+import java.util.TimerTask;
+import java.util.concurrent.TimeUnit;
+
+public class FileVideoCapturer implements VideoCapturer {
+  private interface VideoReader {
+    VideoFrame getNextFrame();
+    void close();
+  }
+
+  /**
+   * Read video data from file for the .y4m container.
+   */
+  @SuppressWarnings("StringSplitter")
+  private static class VideoReaderY4M implements VideoReader {
+    private static final String TAG = "VideoReaderY4M";
+    private static final String Y4M_FRAME_DELIMETER = "FRAME";
+    private static final int FRAME_DELIMETER_LENGTH = Y4M_FRAME_DELIMETER.length() + 1;
+
+    private final int frameWidth;
+    private final int frameHeight;
+    // First char after header
+    private final long videoStart;
+    private final RandomAccessFile mediaFile;
+    private final FileChannel mediaFileChannel;
+
+    public VideoReaderY4M(String file) throws IOException {
+      mediaFile = new RandomAccessFile(file, "r");
+      mediaFileChannel = mediaFile.getChannel();
+      StringBuilder builder = new StringBuilder();
+      for (;;) {
+        int c = mediaFile.read();
+        if (c == -1) {
+          // End of file reached.
+          throw new RuntimeException("Found end of file before end of header for file: " + file);
+        }
+        if (c == '\n') {
+          // End of header found.
+          break;
+        }
+        builder.append((char) c);
+      }
+      videoStart = mediaFileChannel.position();
+      String header = builder.toString();
+      String[] headerTokens = header.split("[ ]");
+      int w = 0;
+      int h = 0;
+      String colorSpace = "";
+      for (String tok : headerTokens) {
+        char c = tok.charAt(0);
+        switch (c) {
+          case 'W':
+            w = Integer.parseInt(tok.substring(1));
+            break;
+          case 'H':
+            h = Integer.parseInt(tok.substring(1));
+            break;
+          case 'C':
+            colorSpace = tok.substring(1);
+            break;
+        }
+      }
+      Logging.d(TAG, "Color space: " + colorSpace);
+      if (!colorSpace.equals("420") && !colorSpace.equals("420mpeg2")) {
+        throw new IllegalArgumentException(
+            "Does not support any other color space than I420 or I420mpeg2");
+      }
+      if ((w % 2) == 1 || (h % 2) == 1) {
+        throw new IllegalArgumentException("Does not support odd width or height");
+      }
+      frameWidth = w;
+      frameHeight = h;
+      Logging.d(TAG, "frame dim: (" + w + ", " + h + ")");
+    }
+
+    @Override
+    public VideoFrame getNextFrame() {
+      final long captureTimeNs = TimeUnit.MILLISECONDS.toNanos(SystemClock.elapsedRealtime());
+      final JavaI420Buffer buffer = JavaI420Buffer.allocate(frameWidth, frameHeight);
+      final ByteBuffer dataY = buffer.getDataY();
+      final ByteBuffer dataU = buffer.getDataU();
+      final ByteBuffer dataV = buffer.getDataV();
+      final int chromaHeight = (frameHeight + 1) / 2;
+      final int sizeY = frameHeight * buffer.getStrideY();
+      final int sizeU = chromaHeight * buffer.getStrideU();
+      final int sizeV = chromaHeight * buffer.getStrideV();
+
+      try {
+        ByteBuffer frameDelim = ByteBuffer.allocate(FRAME_DELIMETER_LENGTH);
+        if (mediaFileChannel.read(frameDelim) < FRAME_DELIMETER_LENGTH) {
+          // We reach end of file, loop
+          mediaFileChannel.position(videoStart);
+          if (mediaFileChannel.read(frameDelim) < FRAME_DELIMETER_LENGTH) {
+            throw new RuntimeException("Error looping video");
+          }
+        }
+        String frameDelimStr = new String(frameDelim.array(), Charset.forName("US-ASCII"));
+        if (!frameDelimStr.equals(Y4M_FRAME_DELIMETER + "\n")) {
+          throw new RuntimeException(
+              "Frames should be delimited by FRAME plus newline, found delimter was: '"
+              + frameDelimStr + "'");
+        }
+
+        mediaFileChannel.read(dataY);
+        mediaFileChannel.read(dataU);
+        mediaFileChannel.read(dataV);
+      } catch (IOException e) {
+        throw new RuntimeException(e);
+      }
+
+      return new VideoFrame(buffer, 0 /* rotation */, captureTimeNs);
+    }
+
+    @Override
+    public void close() {
+      try {
+        // Closing a file also closes the channel.
+        mediaFile.close();
+      } catch (IOException e) {
+        Logging.e(TAG, "Problem closing file", e);
+      }
+    }
+  }
+
+  private final static String TAG = "FileVideoCapturer";
+  private final VideoReader videoReader;
+  private CapturerObserver capturerObserver;
+  private final Timer timer = new Timer();
+
+  private final TimerTask tickTask = new TimerTask() {
+    @Override
+    public void run() {
+      tick();
+    }
+  };
+
+  public FileVideoCapturer(String inputFile) throws IOException {
+    try {
+      videoReader = new VideoReaderY4M(inputFile);
+    } catch (IOException e) {
+      Logging.d(TAG, "Could not open video file: " + inputFile);
+      throw e;
+    }
+  }
+
+  public void tick() {
+    VideoFrame videoFrame = videoReader.getNextFrame();
+    capturerObserver.onFrameCaptured(videoFrame);
+    videoFrame.release();
+  }
+
+  @Override
+  public void initialize(SurfaceTextureHelper surfaceTextureHelper, Context applicationContext,
+      CapturerObserver capturerObserver) {
+    this.capturerObserver = capturerObserver;
+  }
+
+  @Override
+  public void startCapture(int width, int height, int framerate) {
+    timer.schedule(tickTask, 0, 1000 / framerate);
+  }
+
+  @Override
+  public void stopCapture() throws InterruptedException {
+    timer.cancel();
+  }
+
+  @Override
+  public void changeCaptureFormat(int width, int height, int framerate) {
+    // Empty on purpose
+  }
+
+  @Override
+  public void dispose() {
+    videoReader.close();
+  }
+
+  @Override
+  public boolean isScreencast() {
+    return false;
+  }
+}
diff --git a/third_party/libwebrtc/sdk/android/api/org/webrtc/FrameDecryptor.java b/third_party/libwebrtc/sdk/android/api/org/webrtc/FrameDecryptor.java
new file mode 100644
index 0000000000..2932f3d94a
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/api/org/webrtc/FrameDecryptor.java
@@ -0,0 +1,26 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+/**
+ * The FrameDecryptor interface allows Java API users to provide a
+ * pointer to their native implementation of the FrameDecryptorInterface.
+ * FrameDecryptors are extremely performance sensitive as they must process all
+ * incoming video and audio frames. Due to this reason they should always be
+ * backed by a native implementation
+ * @note Not ready for production use.
+ */
+public interface FrameDecryptor {
+  /**
+   * @return A FrameDecryptorInterface pointer.
+   */
+  long getNativeFrameDecryptor();
+}
diff --git a/third_party/libwebrtc/sdk/android/api/org/webrtc/FrameEncryptor.java b/third_party/libwebrtc/sdk/android/api/org/webrtc/FrameEncryptor.java
new file mode 100644
index 0000000000..bc81223f21
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/api/org/webrtc/FrameEncryptor.java
@@ -0,0 +1,26 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+/**
+ * The FrameEncryptor interface allows Java API users to provide a pointer to
+ * their native implementation of the FrameEncryptorInterface.
+ * FrameEncyptors are extremely performance sensitive as they must process all
+ * outgoing video and audio frames. Due to this reason they should always be
+ * backed by a native implementation.
+ * @note Not ready for production use.
+ */
+public interface FrameEncryptor {
+  /**
+   * @return A FrameEncryptorInterface pointer.
+   */
+  long getNativeFrameEncryptor();
+}
diff --git a/third_party/libwebrtc/sdk/android/api/org/webrtc/GlRectDrawer.java b/third_party/libwebrtc/sdk/android/api/org/webrtc/GlRectDrawer.java
new file mode 100644
index 0000000000..d1fbd1b7bc
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/api/org/webrtc/GlRectDrawer.java
@@ -0,0 +1,31 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+/** Simplest possible GL shader that just draws frames as opaque quads. */
+public class GlRectDrawer extends GlGenericDrawer {
+  private static final String FRAGMENT_SHADER = "void main() {\n"
+      + "  gl_FragColor = sample(tc);\n"
+      + "}\n";
+
+  private static class ShaderCallbacks implements GlGenericDrawer.ShaderCallbacks {
+    @Override
+    public void onNewShader(GlShader shader) {}
+
+    @Override
+    public void onPrepareShader(GlShader shader, float[] texMatrix, int frameWidth, int frameHeight,
+        int viewportWidth, int viewportHeight) {}
+  }
+
+  public GlRectDrawer() {
+    super(FRAGMENT_SHADER, new ShaderCallbacks());
+  }
+}
diff --git a/third_party/libwebrtc/sdk/android/api/org/webrtc/GlShader.java b/third_party/libwebrtc/sdk/android/api/org/webrtc/GlShader.java
new file mode 100644
index 0000000000..7efd8d3a95
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/api/org/webrtc/GlShader.java
@@ -0,0 +1,131 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+import android.opengl.GLES20;
+
+import java.nio.FloatBuffer;
+
+// Helper class for handling OpenGL shaders and shader programs.
+public class GlShader {
+  private static final String TAG = "GlShader";
+
+  private static int compileShader(int shaderType, String source) {
+    final int shader = GLES20.glCreateShader(shaderType);
+    if (shader == 0) {
+      throw new RuntimeException("glCreateShader() failed. GLES20 error: " + GLES20.glGetError());
+    }
+    GLES20.glShaderSource(shader, source);
+    GLES20.glCompileShader(shader);
+    int[] compileStatus = new int[] {GLES20.GL_FALSE};
+    GLES20.glGetShaderiv(shader, GLES20.GL_COMPILE_STATUS, compileStatus, 0);
+    if (compileStatus[0] != GLES20.GL_TRUE) {
+      Logging.e(
+          TAG, "Compile error " + GLES20.glGetShaderInfoLog(shader) + " in shader:\n" + source);
+      throw new RuntimeException(GLES20.glGetShaderInfoLog(shader));
+    }
+    GlUtil.checkNoGLES2Error("compileShader");
+    return shader;
+  }
+
+  private int program;
+
+  public GlShader(String vertexSource, String fragmentSource) {
+    final int vertexShader = compileShader(GLES20.GL_VERTEX_SHADER, vertexSource);
+    final int fragmentShader = compileShader(GLES20.GL_FRAGMENT_SHADER, fragmentSource);
+    program = GLES20.glCreateProgram();
+    if (program == 0) {
+      throw new RuntimeException("glCreateProgram() failed. GLES20 error: " + GLES20.glGetError());
+    }
+    GLES20.glAttachShader(program, vertexShader);
+    GLES20.glAttachShader(program, fragmentShader);
+    GLES20.glLinkProgram(program);
+    int[] linkStatus = new int[] {GLES20.GL_FALSE};
+    GLES20.glGetProgramiv(program, GLES20.GL_LINK_STATUS, linkStatus, 0);
+    if (linkStatus[0] != GLES20.GL_TRUE) {
+      Logging.e(TAG, "Could not link program: " + GLES20.glGetProgramInfoLog(program));
+      throw new RuntimeException(GLES20.glGetProgramInfoLog(program));
+    }
+    // According to the documentation of glLinkProgram():
+    // "After the link operation, applications are free to modify attached shader objects, compile
+    // attached shader objects, detach shader objects, delete shader objects, and attach additional
+    // shader objects. None of these operations affects the information log or the program that is
+    // part of the program object."
+    // But in practice, detaching shaders from the program seems to break some devices. Deleting the
+    // shaders are fine however - it will delete them when they are no longer attached to a program.
+    GLES20.glDeleteShader(vertexShader);
+    GLES20.glDeleteShader(fragmentShader);
+    GlUtil.checkNoGLES2Error("Creating GlShader");
+  }
+
+  public int getAttribLocation(String label) {
+    if (program == -1) {
+      throw new RuntimeException("The program has been released");
+    }
+    int location = GLES20.glGetAttribLocation(program, label);
+    if (location < 0) {
+      throw new RuntimeException("Could not locate '" + label + "' in program");
+    }
+    return location;
+  }
+
+  /**
+   * Enable and upload a vertex array for attribute `label`. The vertex data is specified in
+   * `buffer` with `dimension` number of components per vertex.
+   */
+  public void setVertexAttribArray(String label, int dimension, FloatBuffer buffer) {
+    setVertexAttribArray(label, dimension, 0 /* stride */, buffer);
+  }
+
+  /**
+   * Enable and upload a vertex array for attribute `label`. The vertex data is specified in
+   * `buffer` with `dimension` number of components per vertex and specified `stride`.
+   */
+  public void setVertexAttribArray(String label, int dimension, int stride, FloatBuffer buffer) {
+    if (program == -1) {
+      throw new RuntimeException("The program has been released");
+    }
+    int location = getAttribLocation(label);
+    GLES20.glEnableVertexAttribArray(location);
+    GLES20.glVertexAttribPointer(location, dimension, GLES20.GL_FLOAT, false, stride, buffer);
+    GlUtil.checkNoGLES2Error("setVertexAttribArray");
+  }
+
+  public int getUniformLocation(String label) {
+    if (program == -1) {
+      throw new RuntimeException("The program has been released");
+    }
+    int location = GLES20.glGetUniformLocation(program, label);
+    if (location < 0) {
+      throw new RuntimeException("Could not locate uniform '" + label + "' in program");
+    }
+    return location;
+  }
+
+  public void useProgram() {
+    if (program == -1) {
+      throw new RuntimeException("The program has been released");
+    }
+    synchronized (EglBase.lock) {
+      GLES20.glUseProgram(program);
+    }
+    GlUtil.checkNoGLES2Error("glUseProgram");
+  }
+
+  public void release() {
+    Logging.d(TAG, "Deleting shader.");
+    // Delete program, automatically detaching any shaders from it.
+    if (program != -1) {
+      GLES20.glDeleteProgram(program);
+      program = -1;
+    }
+  }
+}
diff --git a/third_party/libwebrtc/sdk/android/api/org/webrtc/GlTextureFrameBuffer.java b/third_party/libwebrtc/sdk/android/api/org/webrtc/GlTextureFrameBuffer.java
new file mode 100644
index 0000000000..b906fe56e0
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/api/org/webrtc/GlTextureFrameBuffer.java
@@ -0,0 +1,122 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+import android.opengl.GLES20;
+
+/**
+ * Helper class for handling OpenGL framebuffer with only color attachment and no depth or stencil
+ * buffer. Intended for simple tasks such as texture copy, texture downscaling, and texture color
+ * conversion. This class is not thread safe and must be used by a thread with an active GL context.
+ */
+// TODO(magjed): Add unittests for this class.
+public class GlTextureFrameBuffer {
+  private final int pixelFormat;
+  private int frameBufferId;
+  private int textureId;
+  private int width;
+  private int height;
+
+  /**
+   * Generate texture and framebuffer resources. An EGLContext must be bound on the current thread
+   * when calling this function. The framebuffer is not complete until setSize() is called.
+   */
+  public GlTextureFrameBuffer(int pixelFormat) {
+    switch (pixelFormat) {
+      case GLES20.GL_LUMINANCE:
+      case GLES20.GL_RGB:
+      case GLES20.GL_RGBA:
+        this.pixelFormat = pixelFormat;
+        break;
+      default:
+        throw new IllegalArgumentException("Invalid pixel format: " + pixelFormat);
+    }
+    this.width = 0;
+    this.height = 0;
+  }
+
+  /**
+   * (Re)allocate texture. Will do nothing if the requested size equals the current size. An
+   * EGLContext must be bound on the current thread when calling this function. Must be called at
+   * least once before using the framebuffer. May be called multiple times to change size.
+   */
+  public void setSize(int width, int height) {
+    if (width <= 0 || height <= 0) {
+      throw new IllegalArgumentException("Invalid size: " + width + "x" + height);
+    }
+    if (width == this.width && height == this.height) {
+      return;
+    }
+    this.width = width;
+    this.height = height;
+    // Lazy allocation the first time setSize() is called.
+    if (textureId == 0) {
+      textureId = GlUtil.generateTexture(GLES20.GL_TEXTURE_2D);
+    }
+    if (frameBufferId == 0) {
+      final int frameBuffers[] = new int[1];
+      GLES20.glGenFramebuffers(1, frameBuffers, 0);
+      frameBufferId = frameBuffers[0];
+    }
+
+    // Allocate texture.
+    GLES20.glActiveTexture(GLES20.GL_TEXTURE0);
+    GLES20.glBindTexture(GLES20.GL_TEXTURE_2D, textureId);
+    GLES20.glTexImage2D(GLES20.GL_TEXTURE_2D, 0, pixelFormat, width, height, 0, pixelFormat,
+        GLES20.GL_UNSIGNED_BYTE, null);
+    GLES20.glBindTexture(GLES20.GL_TEXTURE_2D, 0);
+    GlUtil.checkNoGLES2Error("GlTextureFrameBuffer setSize");
+
+    // Attach the texture to the framebuffer as color attachment.
+    GLES20.glBindFramebuffer(GLES20.GL_FRAMEBUFFER, frameBufferId);
+    GLES20.glFramebufferTexture2D(
+        GLES20.GL_FRAMEBUFFER, GLES20.GL_COLOR_ATTACHMENT0, GLES20.GL_TEXTURE_2D, textureId, 0);
+
+    // Check that the framebuffer is in a good state.
+    final int status = GLES20.glCheckFramebufferStatus(GLES20.GL_FRAMEBUFFER);
+    if (status != GLES20.GL_FRAMEBUFFER_COMPLETE) {
+      throw new IllegalStateException("Framebuffer not complete, status: " + status);
+    }
+
+    GLES20.glBindFramebuffer(GLES20.GL_FRAMEBUFFER, 0);
+  }
+
+  public int getWidth() {
+    return width;
+  }
+
+  public int getHeight() {
+    return height;
+  }
+
+  /** Gets the OpenGL frame buffer id. This value is only valid after setSize() has been called. */
+  public int getFrameBufferId() {
+    return frameBufferId;
+  }
+
+  /** Gets the OpenGL texture id. This value is only valid after setSize() has been called. */
+  public int getTextureId() {
+    return textureId;
+  }
+
+  /**
+   * Release texture and framebuffer. An EGLContext must be bound on the current thread when calling
+   * this function. This object should not be used after this call.
+   */
+  public void release() {
+    GLES20.glDeleteTextures(1, new int[] {textureId}, 0);
+    textureId = 0;
+    GLES20.glDeleteFramebuffers(1, new int[] {frameBufferId}, 0);
+    frameBufferId = 0;
+    width = 0;
+    height = 0;
+  }
+}
diff --git a/third_party/libwebrtc/sdk/android/api/org/webrtc/GlUtil.java b/third_party/libwebrtc/sdk/android/api/org/webrtc/GlUtil.java
new file mode 100644
index 0000000000..e2dd0c56d6
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/api/org/webrtc/GlUtil.java
@@ -0,0 +1,66 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+import android.opengl.GLES20;
+import android.opengl.GLException;
+import java.nio.ByteBuffer;
+import java.nio.ByteOrder;
+import java.nio.FloatBuffer;
+
+/**
+ * Some OpenGL static utility functions.
+ */
+public class GlUtil {
+  private GlUtil() {}
+
+  public static class GlOutOfMemoryException extends GLException {
+    public GlOutOfMemoryException(int error, String msg) {
+      super(error, msg);
+    }
+  }
+
+  // Assert that no OpenGL ES 2.0 error has been raised.
+  public static void checkNoGLES2Error(String msg) {
+    int error = GLES20.glGetError();
+    if (error != GLES20.GL_NO_ERROR) {
+      throw error == GLES20.GL_OUT_OF_MEMORY
+          ? new GlOutOfMemoryException(error, msg)
+          : new GLException(error, msg + ": GLES20 error: " + error);
+    }
+  }
+
+  public static FloatBuffer createFloatBuffer(float[] coords) {
+    // Allocate a direct ByteBuffer, using 4 bytes per float, and copy coords into it.
+    ByteBuffer bb = ByteBuffer.allocateDirect(coords.length * 4);
+    bb.order(ByteOrder.nativeOrder());
+    FloatBuffer fb = bb.asFloatBuffer();
+    fb.put(coords);
+    fb.position(0);
+    return fb;
+  }
+
+  /**
+   * Generate texture with standard parameters.
+   */
+  public static int generateTexture(int target) {
+    final int textureArray[] = new int[1];
+    GLES20.glGenTextures(1, textureArray, 0);
+    final int textureId = textureArray[0];
+    GLES20.glBindTexture(target, textureId);
+    GLES20.glTexParameterf(target, GLES20.GL_TEXTURE_MIN_FILTER, GLES20.GL_LINEAR);
+    GLES20.glTexParameterf(target, GLES20.GL_TEXTURE_MAG_FILTER, GLES20.GL_LINEAR);
+    GLES20.glTexParameterf(target, GLES20.GL_TEXTURE_WRAP_S, GLES20.GL_CLAMP_TO_EDGE);
+    GLES20.glTexParameterf(target, GLES20.GL_TEXTURE_WRAP_T, GLES20.GL_CLAMP_TO_EDGE);
+    checkNoGLES2Error("generateTexture");
+    return textureId;
+  }
+}
diff --git a/third_party/libwebrtc/sdk/android/api/org/webrtc/HardwareVideoDecoderFactory.java b/third_party/libwebrtc/sdk/android/api/org/webrtc/HardwareVideoDecoderFactory.java
new file mode 100644
index 0000000000..215598a85d
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/api/org/webrtc/HardwareVideoDecoderFactory.java
@@ -0,0 +1,57 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+import android.media.MediaCodecInfo;
+import androidx.annotation.Nullable;
+import java.util.Arrays;
+
+/** Factory for Android hardware VideoDecoders. */
+public class HardwareVideoDecoderFactory extends MediaCodecVideoDecoderFactory {
+  private final static Predicate<MediaCodecInfo> defaultAllowedPredicate =
+      new Predicate<MediaCodecInfo>() {
+        @Override
+        public boolean test(MediaCodecInfo arg) {
+          return MediaCodecUtils.isHardwareAccelerated(arg);
+        }
+      };
+
+  /** Creates a HardwareVideoDecoderFactory that does not use surface textures. */
+  @Deprecated // Not removed yet to avoid breaking callers.
+  public HardwareVideoDecoderFactory() {
+    this(null);
+  }
+
+  /**
+   * Creates a HardwareVideoDecoderFactory that supports surface texture rendering.
+   *
+   * @param sharedContext The textures generated will be accessible from this context. May be null,
+   *                      this disables texture support.
+   */
+  public HardwareVideoDecoderFactory(@Nullable EglBase.Context sharedContext) {
+    this(sharedContext, /* codecAllowedPredicate= */ null);
+  }
+
+  /**
+   * Creates a HardwareVideoDecoderFactory that supports surface texture rendering.
+   *
+   * @param sharedContext The textures generated will be accessible from this context. May be null,
+   *                      this disables texture support.
+   * @param codecAllowedPredicate predicate to filter codecs. It is combined with the default
+   *                              predicate that only allows hardware codecs.
+   */
+  public HardwareVideoDecoderFactory(@Nullable EglBase.Context sharedContext,
+      @Nullable Predicate<MediaCodecInfo> codecAllowedPredicate) {
+    super(sharedContext,
+        (codecAllowedPredicate == null ? defaultAllowedPredicate
+                                       : codecAllowedPredicate.and(defaultAllowedPredicate)));
+  }
+}
diff --git a/third_party/libwebrtc/sdk/android/api/org/webrtc/IceCandidateErrorEvent.java b/third_party/libwebrtc/sdk/android/api/org/webrtc/IceCandidateErrorEvent.java
new file mode 100644
index 0000000000..aae9da7061
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/api/org/webrtc/IceCandidateErrorEvent.java
@@ -0,0 +1,43 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+public final class IceCandidateErrorEvent {
+  /** The local IP address used to communicate with the STUN or TURN server. */
+  public final String address;
+  /** The port used to communicate with the STUN or TURN server. */
+  public final int port;
+  /**
+   * The STUN or TURN URL that identifies the STUN or TURN server for which the failure occurred.
+   */
+  public final String url;
+  /**
+   * The numeric STUN error code returned by the STUN or TURN server. If no host candidate can reach
+   * the server, errorCode will be set to the value 701 which is outside the STUN error code range.
+   * This error is only fired once per server URL while in the RTCIceGatheringState of "gathering".
+   */
+  public final int errorCode;
+  /**
+   * The STUN reason text returned by the STUN or TURN server. If the server could not be reached,
+   * errorText will be set to an implementation-specific value providing details about the error.
+   */
+  public final String errorText;
+
+  @CalledByNative
+  public IceCandidateErrorEvent(
+      String address, int port, String url, int errorCode, String errorText) {
+    this.address = address;
+    this.port = port;
+    this.url = url;
+    this.errorCode = errorCode;
+    this.errorText = errorText;
+  }
+}
diff --git a/third_party/libwebrtc/sdk/android/api/org/webrtc/JavaI420Buffer.java b/third_party/libwebrtc/sdk/android/api/org/webrtc/JavaI420Buffer.java
new file mode 100644
index 0000000000..322b8f38c9
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/api/org/webrtc/JavaI420Buffer.java
@@ -0,0 +1,200 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+import androidx.annotation.Nullable;
+import java.nio.ByteBuffer;
+import org.webrtc.VideoFrame.I420Buffer;
+
+/** Implementation of VideoFrame.I420Buffer backed by Java direct byte buffers. */
+public class JavaI420Buffer implements VideoFrame.I420Buffer {
+  private final int width;
+  private final int height;
+  private final ByteBuffer dataY;
+  private final ByteBuffer dataU;
+  private final ByteBuffer dataV;
+  private final int strideY;
+  private final int strideU;
+  private final int strideV;
+  private final RefCountDelegate refCountDelegate;
+
+  private JavaI420Buffer(int width, int height, ByteBuffer dataY, int strideY, ByteBuffer dataU,
+      int strideU, ByteBuffer dataV, int strideV, @Nullable Runnable releaseCallback) {
+    this.width = width;
+    this.height = height;
+    this.dataY = dataY;
+    this.dataU = dataU;
+    this.dataV = dataV;
+    this.strideY = strideY;
+    this.strideU = strideU;
+    this.strideV = strideV;
+    this.refCountDelegate = new RefCountDelegate(releaseCallback);
+  }
+
+  private static void checkCapacity(ByteBuffer data, int width, int height, int stride) {
+    // The last row does not necessarily need padding.
+    final int minCapacity = stride * (height - 1) + width;
+    if (data.capacity() < minCapacity) {
+      throw new IllegalArgumentException(
+          "Buffer must be at least " + minCapacity + " bytes, but was " + data.capacity());
+    }
+  }
+
+  /** Wraps existing ByteBuffers into JavaI420Buffer object without copying the contents. */
+  public static JavaI420Buffer wrap(int width, int height, ByteBuffer dataY, int strideY,
+      ByteBuffer dataU, int strideU, ByteBuffer dataV, int strideV,
+      @Nullable Runnable releaseCallback) {
+    if (dataY == null || dataU == null || dataV == null) {
+      throw new IllegalArgumentException("Data buffers cannot be null.");
+    }
+    if (!dataY.isDirect() || !dataU.isDirect() || !dataV.isDirect()) {
+      throw new IllegalArgumentException("Data buffers must be direct byte buffers.");
+    }
+
+    // Slice the buffers to prevent external modifications to the position / limit of the buffer.
+    // Note that this doesn't protect the contents of the buffers from modifications.
+    dataY = dataY.slice();
+    dataU = dataU.slice();
+    dataV = dataV.slice();
+
+    final int chromaWidth = (width + 1) / 2;
+    final int chromaHeight = (height + 1) / 2;
+    checkCapacity(dataY, width, height, strideY);
+    checkCapacity(dataU, chromaWidth, chromaHeight, strideU);
+    checkCapacity(dataV, chromaWidth, chromaHeight, strideV);
+
+    return new JavaI420Buffer(
+        width, height, dataY, strideY, dataU, strideU, dataV, strideV, releaseCallback);
+  }
+
+  /** Allocates an empty I420Buffer suitable for an image of the given dimensions. */
+  public static JavaI420Buffer allocate(int width, int height) {
+    int chromaHeight = (height + 1) / 2;
+    int strideUV = (width + 1) / 2;
+    int yPos = 0;
+    int uPos = yPos + width * height;
+    int vPos = uPos + strideUV * chromaHeight;
+
+    ByteBuffer buffer =
+        JniCommon.nativeAllocateByteBuffer(width * height + 2 * strideUV * chromaHeight);
+
+    buffer.position(yPos);
+    buffer.limit(uPos);
+    ByteBuffer dataY = buffer.slice();
+
+    buffer.position(uPos);
+    buffer.limit(vPos);
+    ByteBuffer dataU = buffer.slice();
+
+    buffer.position(vPos);
+    buffer.limit(vPos + strideUV * chromaHeight);
+    ByteBuffer dataV = buffer.slice();
+
+    return new JavaI420Buffer(width, height, dataY, width, dataU, strideUV, dataV, strideUV,
+        () -> { JniCommon.nativeFreeByteBuffer(buffer); });
+  }
+
+  @Override
+  public int getWidth() {
+    return width;
+  }
+
+  @Override
+  public int getHeight() {
+    return height;
+  }
+
+  @Override
+  public ByteBuffer getDataY() {
+    // Return a slice to prevent relative reads from changing the position.
+    return dataY.slice();
+  }
+
+  @Override
+  public ByteBuffer getDataU() {
+    // Return a slice to prevent relative reads from changing the position.
+    return dataU.slice();
+  }
+
+  @Override
+  public ByteBuffer getDataV() {
+    // Return a slice to prevent relative reads from changing the position.
+    return dataV.slice();
+  }
+
+  @Override
+  public int getStrideY() {
+    return strideY;
+  }
+
+  @Override
+  public int getStrideU() {
+    return strideU;
+  }
+
+  @Override
+  public int getStrideV() {
+    return strideV;
+  }
+
+  @Override
+  public I420Buffer toI420() {
+    retain();
+    return this;
+  }
+
+  @Override
+  public void retain() {
+    refCountDelegate.retain();
+  }
+
+  @Override
+  public void release() {
+    refCountDelegate.release();
+  }
+
+  @Override
+  public VideoFrame.Buffer cropAndScale(
+      int cropX, int cropY, int cropWidth, int cropHeight, int scaleWidth, int scaleHeight) {
+    return cropAndScaleI420(this, cropX, cropY, cropWidth, cropHeight, scaleWidth, scaleHeight);
+  }
+
+  public static VideoFrame.Buffer cropAndScaleI420(final I420Buffer buffer, int cropX, int cropY,
+      int cropWidth, int cropHeight, int scaleWidth, int scaleHeight) {
+    if (cropWidth == scaleWidth && cropHeight == scaleHeight) {
+      // No scaling.
+      ByteBuffer dataY = buffer.getDataY();
+      ByteBuffer dataU = buffer.getDataU();
+      ByteBuffer dataV = buffer.getDataV();
+
+      dataY.position(cropX + cropY * buffer.getStrideY());
+      dataU.position(cropX / 2 + cropY / 2 * buffer.getStrideU());
+      dataV.position(cropX / 2 + cropY / 2 * buffer.getStrideV());
+
+      buffer.retain();
+      return JavaI420Buffer.wrap(scaleWidth, scaleHeight, dataY.slice(), buffer.getStrideY(),
+          dataU.slice(), buffer.getStrideU(), dataV.slice(), buffer.getStrideV(), buffer::release);
+    }
+
+    JavaI420Buffer newBuffer = JavaI420Buffer.allocate(scaleWidth, scaleHeight);
+    nativeCropAndScaleI420(buffer.getDataY(), buffer.getStrideY(), buffer.getDataU(),
+        buffer.getStrideU(), buffer.getDataV(), buffer.getStrideV(), cropX, cropY, cropWidth,
+        cropHeight, newBuffer.getDataY(), newBuffer.getStrideY(), newBuffer.getDataU(),
+        newBuffer.getStrideU(), newBuffer.getDataV(), newBuffer.getStrideV(), scaleWidth,
+        scaleHeight);
+    return newBuffer;
+  }
+
+  private static native void nativeCropAndScaleI420(ByteBuffer srcY, int srcStrideY,
+      ByteBuffer srcU, int srcStrideU, ByteBuffer srcV, int srcStrideV, int cropX, int cropY,
+      int cropWidth, int cropHeight, ByteBuffer dstY, int dstStrideY, ByteBuffer dstU,
+      int dstStrideU, ByteBuffer dstV, int dstStrideV, int scaleWidth, int scaleHeight);
+}
diff --git a/third_party/libwebrtc/sdk/android/api/org/webrtc/LibaomAv1Encoder.java b/third_party/libwebrtc/sdk/android/api/org/webrtc/LibaomAv1Encoder.java
new file mode 100644
index 0000000000..569a719f44
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/api/org/webrtc/LibaomAv1Encoder.java
@@ -0,0 +1,25 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+public class LibaomAv1Encoder extends WrappedNativeVideoEncoder {
+  @Override
+  public long createNativeVideoEncoder() {
+    return nativeCreateEncoder();
+  }
+
+  static native long nativeCreateEncoder();
+
+  @Override
+  public boolean isHardwareEncoder() {
+    return false;
+  }
+}
diff --git a/third_party/libwebrtc/sdk/android/api/org/webrtc/LibvpxVp8Decoder.java b/third_party/libwebrtc/sdk/android/api/org/webrtc/LibvpxVp8Decoder.java
new file mode 100644
index 0000000000..54ad0aa137
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/api/org/webrtc/LibvpxVp8Decoder.java
@@ -0,0 +1,20 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+public class LibvpxVp8Decoder extends WrappedNativeVideoDecoder {
+  @Override
+  public long createNativeVideoDecoder() {
+    return nativeCreateDecoder();
+  }
+
+  static native long nativeCreateDecoder();
+}
diff --git a/third_party/libwebrtc/sdk/android/api/org/webrtc/LibvpxVp8Encoder.java b/third_party/libwebrtc/sdk/android/api/org/webrtc/LibvpxVp8Encoder.java
new file mode 100644
index 0000000000..4be9e52c14
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/api/org/webrtc/LibvpxVp8Encoder.java
@@ -0,0 +1,25 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+public class LibvpxVp8Encoder extends WrappedNativeVideoEncoder {
+  @Override
+  public long createNativeVideoEncoder() {
+    return nativeCreateEncoder();
+  }
+
+  static native long nativeCreateEncoder();
+
+  @Override
+  public boolean isHardwareEncoder() {
+    return false;
+  }
+}
diff --git a/third_party/libwebrtc/sdk/android/api/org/webrtc/LibvpxVp9Decoder.java b/third_party/libwebrtc/sdk/android/api/org/webrtc/LibvpxVp9Decoder.java
new file mode 100644
index 0000000000..90a24433a3
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/api/org/webrtc/LibvpxVp9Decoder.java
@@ -0,0 +1,22 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+public class LibvpxVp9Decoder extends WrappedNativeVideoDecoder {
+  @Override
+  public long createNativeVideoDecoder() {
+    return nativeCreateDecoder();
+  }
+
+  static native long nativeCreateDecoder();
+
+  static native boolean nativeIsSupported();
+}
diff --git a/third_party/libwebrtc/sdk/android/api/org/webrtc/LibvpxVp9Encoder.java b/third_party/libwebrtc/sdk/android/api/org/webrtc/LibvpxVp9Encoder.java
new file mode 100644
index 0000000000..1211ae93fb
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/api/org/webrtc/LibvpxVp9Encoder.java
@@ -0,0 +1,27 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+public class LibvpxVp9Encoder extends WrappedNativeVideoEncoder {
+  @Override
+  public long createNativeVideoEncoder() {
+    return nativeCreateEncoder();
+  }
+
+  static native long nativeCreateEncoder();
+
+  @Override
+  public boolean isHardwareEncoder() {
+    return false;
+  }
+
+  static native boolean nativeIsSupported();
+}
diff --git a/third_party/libwebrtc/sdk/android/api/org/webrtc/MediaConstraints.java b/third_party/libwebrtc/sdk/android/api/org/webrtc/MediaConstraints.java
new file mode 100644
index 0000000000..bae04e532c
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/api/org/webrtc/MediaConstraints.java
@@ -0,0 +1,99 @@
+/*
+ *  Copyright 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+import androidx.annotation.Nullable;
+import java.util.ArrayList;
+import java.util.List;
+
+/**
+ * Description of media constraints for {@code MediaStream} and
+ * {@code PeerConnection}.
+ */
+public class MediaConstraints {
+  /** Simple String key/value pair. */
+  public static class KeyValuePair {
+    private final String key;
+    private final String value;
+
+    public KeyValuePair(String key, String value) {
+      this.key = key;
+      this.value = value;
+    }
+
+    @CalledByNative("KeyValuePair")
+    public String getKey() {
+      return key;
+    }
+
+    @CalledByNative("KeyValuePair")
+    public String getValue() {
+      return value;
+    }
+
+    @Override
+    public String toString() {
+      return key + ": " + value;
+    }
+
+    @Override
+    public boolean equals(@Nullable Object other) {
+      if (this == other) {
+        return true;
+      }
+      if (other == null || getClass() != other.getClass()) {
+        return false;
+      }
+      KeyValuePair that = (KeyValuePair) other;
+      return key.equals(that.key) && value.equals(that.value);
+    }
+
+    @Override
+    public int hashCode() {
+      return key.hashCode() + value.hashCode();
+    }
+  }
+
+  public final List<KeyValuePair> mandatory;
+  public final List<KeyValuePair> optional;
+
+  public MediaConstraints() {
+    mandatory = new ArrayList<KeyValuePair>();
+    optional = new ArrayList<KeyValuePair>();
+  }
+
+  private static String stringifyKeyValuePairList(List<KeyValuePair> list) {
+    StringBuilder builder = new StringBuilder("[");
+    for (KeyValuePair pair : list) {
+      if (builder.length() > 1) {
+        builder.append(", ");
+      }
+      builder.append(pair.toString());
+    }
+    return builder.append("]").toString();
+  }
+
+  @Override
+  public String toString() {
+    return "mandatory: " + stringifyKeyValuePairList(mandatory) + ", optional: "
+        + stringifyKeyValuePairList(optional);
+  }
+
+  @CalledByNative
+  List<KeyValuePair> getMandatory() {
+    return mandatory;
+  }
+
+  @CalledByNative
+  List<KeyValuePair> getOptional() {
+    return optional;
+  }
+}
diff --git a/third_party/libwebrtc/sdk/android/api/org/webrtc/MediaSource.java b/third_party/libwebrtc/sdk/android/api/org/webrtc/MediaSource.java
new file mode 100644
index 0000000000..9245e3e2eb
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/api/org/webrtc/MediaSource.java
@@ -0,0 +1,74 @@
+/*
+ *  Copyright 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+/** Java wrapper for a C++ MediaSourceInterface. */
+public class MediaSource {
+  /** Tracks MediaSourceInterface.SourceState */
+  public enum State {
+    INITIALIZING,
+    LIVE,
+    ENDED,
+    MUTED;
+
+    @CalledByNative("State")
+    static State fromNativeIndex(int nativeIndex) {
+      return values()[nativeIndex];
+    }
+  }
+
+  private final RefCountDelegate refCountDelegate;
+  private long nativeSource;
+
+  public MediaSource(long nativeSource) {
+    refCountDelegate = new RefCountDelegate(() -> JniCommon.nativeReleaseRef(nativeSource));
+    this.nativeSource = nativeSource;
+  }
+
+  public State state() {
+    checkMediaSourceExists();
+    return nativeGetState(nativeSource);
+  }
+
+  public void dispose() {
+    checkMediaSourceExists();
+    refCountDelegate.release();
+    nativeSource = 0;
+  }
+
+  /** Returns a pointer to webrtc::MediaSourceInterface. */
+  protected long getNativeMediaSource() {
+    checkMediaSourceExists();
+    return nativeSource;
+  }
+
+  /**
+   * Runs code in {@code runnable} holding a reference to the media source. If the object has
+   * already been released, does nothing.
+   */
+  void runWithReference(Runnable runnable) {
+    if (refCountDelegate.safeRetain()) {
+      try {
+        runnable.run();
+      } finally {
+        refCountDelegate.release();
+      }
+    }
+  }
+
+  private void checkMediaSourceExists() {
+    if (nativeSource == 0) {
+      throw new IllegalStateException("MediaSource has been disposed.");
+    }
+  }
+
+  private static native State nativeGetState(long pointer);
+}
diff --git a/third_party/libwebrtc/sdk/android/api/org/webrtc/MediaStreamTrack.java b/third_party/libwebrtc/sdk/android/api/org/webrtc/MediaStreamTrack.java
new file mode 100644
index 0000000000..2e4c3e18f7
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/api/org/webrtc/MediaStreamTrack.java
@@ -0,0 +1,129 @@
+/*
+ *  Copyright 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+import androidx.annotation.Nullable;
+
+/** Java wrapper for a C++ MediaStreamTrackInterface. */
+public class MediaStreamTrack {
+  public static final String AUDIO_TRACK_KIND = "audio";
+  public static final String VIDEO_TRACK_KIND = "video";
+
+  /** Tracks MediaStreamTrackInterface.TrackState */
+  public enum State {
+    LIVE,
+    ENDED;
+
+    @CalledByNative("State")
+    static State fromNativeIndex(int nativeIndex) {
+      return values()[nativeIndex];
+    }
+  }
+
+  // Must be kept in sync with cricket::MediaType.
+  public enum MediaType {
+    MEDIA_TYPE_AUDIO(0),
+    MEDIA_TYPE_VIDEO(1);
+
+    private final int nativeIndex;
+
+    private MediaType(int nativeIndex) {
+      this.nativeIndex = nativeIndex;
+    }
+
+    @CalledByNative("MediaType")
+    int getNative() {
+      return nativeIndex;
+    }
+
+    @CalledByNative("MediaType")
+    static MediaType fromNativeIndex(int nativeIndex) {
+      for (MediaType type : MediaType.values()) {
+        if (type.getNative() == nativeIndex) {
+          return type;
+        }
+      }
+      throw new IllegalArgumentException("Unknown native media type: " + nativeIndex);
+    }
+  }
+
+  /** Factory method to create an AudioTrack or VideoTrack subclass. */
+  static @Nullable MediaStreamTrack createMediaStreamTrack(long nativeTrack) {
+    if (nativeTrack == 0) {
+      return null;
+    }
+    String trackKind = nativeGetKind(nativeTrack);
+    if (trackKind.equals(AUDIO_TRACK_KIND)) {
+      return new AudioTrack(nativeTrack);
+    } else if (trackKind.equals(VIDEO_TRACK_KIND)) {
+      return new VideoTrack(nativeTrack);
+    } else {
+      return null;
+    }
+  }
+
+  private long nativeTrack;
+
+  public MediaStreamTrack(long nativeTrack) {
+    if (nativeTrack == 0) {
+      throw new IllegalArgumentException("nativeTrack may not be null");
+    }
+    this.nativeTrack = nativeTrack;
+  }
+
+  public String id() {
+    checkMediaStreamTrackExists();
+    return nativeGetId(nativeTrack);
+  }
+
+  public String kind() {
+    checkMediaStreamTrackExists();
+    return nativeGetKind(nativeTrack);
+  }
+
+  public boolean enabled() {
+    checkMediaStreamTrackExists();
+    return nativeGetEnabled(nativeTrack);
+  }
+
+  public boolean setEnabled(boolean enable) {
+    checkMediaStreamTrackExists();
+    return nativeSetEnabled(nativeTrack, enable);
+  }
+
+  public State state() {
+    checkMediaStreamTrackExists();
+    return nativeGetState(nativeTrack);
+  }
+
+  public void dispose() {
+    checkMediaStreamTrackExists();
+    JniCommon.nativeReleaseRef(nativeTrack);
+    nativeTrack = 0;
+  }
+
+  long getNativeMediaStreamTrack() {
+    checkMediaStreamTrackExists();
+    return nativeTrack;
+  }
+
+  private void checkMediaStreamTrackExists() {
+    if (nativeTrack == 0) {
+      throw new IllegalStateException("MediaStreamTrack has been disposed.");
+    }
+  }
+
+  private static native String nativeGetId(long track);
+  private static native String nativeGetKind(long track);
+  private static native boolean nativeGetEnabled(long track);
+  private static native boolean nativeSetEnabled(long track, boolean enabled);
+  private static native State nativeGetState(long track);
+}
diff --git a/third_party/libwebrtc/sdk/android/api/org/webrtc/Metrics.java b/third_party/libwebrtc/sdk/android/api/org/webrtc/Metrics.java
new file mode 100644
index 0000000000..253376831c
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/api/org/webrtc/Metrics.java
@@ -0,0 +1,81 @@
+/*
+ *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+import java.util.HashMap;
+import java.util.Map;
+
+// Java-side of androidmetrics.cc
+//
+// Rtc histograms can be queried through the API, getAndReset().
+// The returned map holds the name of a histogram and its samples.
+//
+// Example of `map` with one histogram:
+// `name`: "WebRTC.Video.InputFramesPerSecond"
+//     `min`: 1
+//     `max`: 100
+//     `bucketCount`: 50
+//     `samples`: [30]:1
+//
+// Most histograms are not updated frequently (e.g. most video metrics are an
+// average over the call and recorded when a stream is removed).
+// The metrics can for example be retrieved when a peer connection is closed.
+public class Metrics {
+  private static final String TAG = "Metrics";
+
+  public final Map<String, HistogramInfo> map =
+      new HashMap<String, HistogramInfo>(); // <name, HistogramInfo>
+
+  @CalledByNative
+  Metrics() {}
+
+  /**
+   * Class holding histogram information.
+   */
+  public static class HistogramInfo {
+    public final int min;
+    public final int max;
+    public final int bucketCount;
+    public final Map<Integer, Integer> samples =
+        new HashMap<Integer, Integer>(); // <value, # of events>
+
+    @CalledByNative("HistogramInfo")
+    public HistogramInfo(int min, int max, int bucketCount) {
+      this.min = min;
+      this.max = max;
+      this.bucketCount = bucketCount;
+    }
+
+    @CalledByNative("HistogramInfo")
+    public void addSample(int value, int numEvents) {
+      samples.put(value, numEvents);
+    }
+  }
+
+  @CalledByNative
+  private void add(String name, HistogramInfo info) {
+    map.put(name, info);
+  }
+
+  // Enables gathering of metrics (which can be fetched with getAndReset()).
+  // Must be called before PeerConnectionFactory is created.
+  public static void enable() {
+    nativeEnable();
+  }
+
+  // Gets and clears native histograms.
+  public static Metrics getAndReset() {
+    return nativeGetAndReset();
+  }
+
+  private static native void nativeEnable();
+  private static native Metrics nativeGetAndReset();
+}
diff --git a/third_party/libwebrtc/sdk/android/api/org/webrtc/NativeLibraryLoader.java b/third_party/libwebrtc/sdk/android/api/org/webrtc/NativeLibraryLoader.java
new file mode 100644
index 0000000000..8bd7b3b250
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/api/org/webrtc/NativeLibraryLoader.java
@@ -0,0 +1,24 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+/**
+ * Interface for loading native libraries. A custom loader can be passed to
+ * PeerConnectionFactory.initialize.
+ */
+public interface NativeLibraryLoader {
+  /**
+   * Loads a native library with the given name.
+   *
+   * @return True on success
+   */
+  boolean load(String name);
+}
diff --git a/third_party/libwebrtc/sdk/android/api/org/webrtc/NativePeerConnectionFactory.java b/third_party/libwebrtc/sdk/android/api/org/webrtc/NativePeerConnectionFactory.java
new file mode 100644
index 0000000000..aeb91e1750
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/api/org/webrtc/NativePeerConnectionFactory.java
@@ -0,0 +1,20 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+/** Factory for creating webrtc::jni::OwnedPeerConnection instances. */
+public interface NativePeerConnectionFactory {
+  /**
+   * Create a new webrtc::jni::OwnedPeerConnection instance and returns a pointer to it.
+   * The caller takes ownership of the object.
+   */
+  long createNativePeerConnection();
+}
diff --git a/third_party/libwebrtc/sdk/android/api/org/webrtc/NetEqFactoryFactory.java b/third_party/libwebrtc/sdk/android/api/org/webrtc/NetEqFactoryFactory.java
new file mode 100644
index 0000000000..8464324cbc
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/api/org/webrtc/NetEqFactoryFactory.java
@@ -0,0 +1,21 @@
+/*
+ *  Copyright 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+/**
+ * Implementations of this interface can create a native {@code webrtc::NetEqFactory}.
+ */
+public interface NetEqFactoryFactory {
+  /**
+   * Returns a pointer to a {@code webrtc::NetEqFactory}. The caller takes ownership.
+   */
+  long createNativeNetEqFactory();
+}
diff --git a/third_party/libwebrtc/sdk/android/api/org/webrtc/PlatformSoftwareVideoDecoderFactory.java b/third_party/libwebrtc/sdk/android/api/org/webrtc/PlatformSoftwareVideoDecoderFactory.java
new file mode 100644
index 0000000000..caca5e5889
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/api/org/webrtc/PlatformSoftwareVideoDecoderFactory.java
@@ -0,0 +1,39 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+import android.media.MediaCodecInfo;
+import androidx.annotation.Nullable;
+import java.util.Arrays;
+
+/** Factory for Android platform software VideoDecoders. */
+public class PlatformSoftwareVideoDecoderFactory extends MediaCodecVideoDecoderFactory {
+  /**
+   * Default allowed predicate.
+   */
+  private static final Predicate<MediaCodecInfo> defaultAllowedPredicate =
+      new Predicate<MediaCodecInfo>() {
+        @Override
+        public boolean test(MediaCodecInfo arg) {
+          return MediaCodecUtils.isSoftwareOnly(arg);
+        }
+      };
+
+  /**
+   * Creates a PlatformSoftwareVideoDecoderFactory that supports surface texture rendering.
+   *
+   * @param sharedContext The textures generated will be accessible from this context. May be null,
+   *                      this disables texture support.
+   */
+  public PlatformSoftwareVideoDecoderFactory(@Nullable EglBase.Context sharedContext) {
+    super(sharedContext, defaultAllowedPredicate);
+  }
+}
diff --git a/third_party/libwebrtc/sdk/android/api/org/webrtc/Predicate.java b/third_party/libwebrtc/sdk/android/api/org/webrtc/Predicate.java
new file mode 100644
index 0000000000..50e6975000
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/api/org/webrtc/Predicate.java
@@ -0,0 +1,73 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+/**
+ * Represents a predicate (boolean-valued function) of one argument.
+ */
+public interface Predicate<T> {
+  /**
+   * Evaluates this predicate on the given argument.
+   *
+   * @param arg the input argument
+   * @return true if the input argument matches the predicate, otherwise false
+   */
+  boolean test(T arg);
+
+  /**
+   * Returns a composed predicate that represents a short-circuiting logical OR of this predicate
+   * and another. When evaluating the composed predicate, if this predicate is true, then the other
+   * predicate is not evaluated.
+   *
+   * @param other a predicate that will be logically-ORed with this predicate
+   * @return a composed predicate that represents the short-circuiting logical OR of this predicate
+   *     and the other predicate
+   */
+  default Predicate<T> or(Predicate<? super T> other) {
+    return new Predicate<T>() {
+      @Override
+      public boolean test(T arg) {
+        return Predicate.this.test(arg) || other.test(arg);
+      }
+    };
+  }
+
+  /**
+   * Returns a composed predicate that represents a short-circuiting logical AND of this predicate
+   * and another.
+   *
+   * @param other a predicate that will be logically-ANDed with this predicate
+   * @return a composed predicate that represents the short-circuiting logical AND of this predicate
+   *     and the other predicate
+   */
+  default Predicate<T> and(Predicate<? super T> other) {
+    return new Predicate<T>() {
+      @Override
+      public boolean test(T arg) {
+        return Predicate.this.test(arg) && other.test(arg);
+      }
+    };
+  }
+
+  /**
+   * Returns a predicate that represents the logical negation of this predicate.
+   *
+   * @return a predicate that represents the logical negation of this predicate
+   */
+  default Predicate<T> negate() {
+    return new Predicate<T>() {
+      @Override
+      public boolean test(T arg) {
+        return !Predicate.this.test(arg);
+      }
+    };
+  }
+}
\ No newline at end of file
diff --git a/third_party/libwebrtc/sdk/android/api/org/webrtc/RefCounted.java b/third_party/libwebrtc/sdk/android/api/org/webrtc/RefCounted.java
new file mode 100644
index 0000000000..0c1c3bf1f9
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/api/org/webrtc/RefCounted.java
@@ -0,0 +1,28 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+/**
+ * Interface for ref counted objects in WebRTC. These objects have significant resources that need
+ * to be freed when they are no longer in use. Each objects starts with ref count of one when
+ * created. If a reference is passed as a parameter to a method, the caller has ownesrship of the
+ * object by default - calling release is not necessary unless retain is called.
+ */
+public interface RefCounted {
+  /** Increases ref count by one. */
+  @CalledByNative void retain();
+
+  /**
+   * Decreases ref count by one. When the ref count reaches zero, resources related to the object
+   * will be freed.
+   */
+  @CalledByNative void release();
+}
diff --git a/third_party/libwebrtc/sdk/android/api/org/webrtc/RendererCommon.java b/third_party/libwebrtc/sdk/android/api/org/webrtc/RendererCommon.java
new file mode 100644
index 0000000000..b97901c634
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/api/org/webrtc/RendererCommon.java
@@ -0,0 +1,259 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+import android.graphics.Point;
+import android.opengl.Matrix;
+import android.view.View;
+
+/**
+ * Static helper functions for renderer implementations.
+ */
+public class RendererCommon {
+  /** Interface for reporting rendering events. */
+  public static interface RendererEvents {
+    /**
+     * Callback fired once first frame is rendered.
+     */
+    public void onFirstFrameRendered();
+
+    /**
+     * Callback fired when rendered frame resolution or rotation has changed.
+     */
+    public void onFrameResolutionChanged(int videoWidth, int videoHeight, int rotation);
+  }
+
+  /**
+   * Interface for rendering frames on an EGLSurface with specified viewport location. Rotation,
+   * mirror, and cropping is specified using a 4x4 texture coordinate transform matrix. The frame
+   * input can either be an OES texture, RGB texture, or YUV textures in I420 format. The function
+   * release() must be called manually to free the resources held by this object.
+   */
+  public static interface GlDrawer {
+    /**
+     * Functions for drawing frames with different sources. The rendering surface target is
+     * implied by the current EGL context of the calling thread and requires no explicit argument.
+     * The coordinates specify the viewport location on the surface target.
+     */
+    void drawOes(int oesTextureId, float[] texMatrix, int frameWidth, int frameHeight,
+        int viewportX, int viewportY, int viewportWidth, int viewportHeight);
+    void drawRgb(int textureId, float[] texMatrix, int frameWidth, int frameHeight, int viewportX,
+        int viewportY, int viewportWidth, int viewportHeight);
+    void drawYuv(int[] yuvTextures, float[] texMatrix, int frameWidth, int frameHeight,
+        int viewportX, int viewportY, int viewportWidth, int viewportHeight);
+
+    /**
+     * Release all GL resources. This needs to be done manually, otherwise resources may leak.
+     */
+    void release();
+  }
+
+  /**
+   * Helper class for determining layout size based on layout requirements, scaling type, and video
+   * aspect ratio.
+   */
+  public static class VideoLayoutMeasure {
+    // The scaling type determines how the video will fill the allowed layout area in measure(). It
+    // can be specified separately for the case when video has matched orientation with layout size
+    // and when there is an orientation mismatch.
+    private float visibleFractionMatchOrientation =
+        convertScalingTypeToVisibleFraction(ScalingType.SCALE_ASPECT_BALANCED);
+    private float visibleFractionMismatchOrientation =
+        convertScalingTypeToVisibleFraction(ScalingType.SCALE_ASPECT_BALANCED);
+
+    public void setScalingType(ScalingType scalingType) {
+      setScalingType(/* scalingTypeMatchOrientation= */ scalingType,
+          /* scalingTypeMismatchOrientation= */ scalingType);
+    }
+
+    public void setScalingType(
+        ScalingType scalingTypeMatchOrientation, ScalingType scalingTypeMismatchOrientation) {
+      this.visibleFractionMatchOrientation =
+          convertScalingTypeToVisibleFraction(scalingTypeMatchOrientation);
+      this.visibleFractionMismatchOrientation =
+          convertScalingTypeToVisibleFraction(scalingTypeMismatchOrientation);
+    }
+
+    public void setVisibleFraction(
+        float visibleFractionMatchOrientation, float visibleFractionMismatchOrientation) {
+      this.visibleFractionMatchOrientation = visibleFractionMatchOrientation;
+      this.visibleFractionMismatchOrientation = visibleFractionMismatchOrientation;
+    }
+
+    public Point measure(int widthSpec, int heightSpec, int frameWidth, int frameHeight) {
+      // Calculate max allowed layout size.
+      final int maxWidth = View.getDefaultSize(Integer.MAX_VALUE, widthSpec);
+      final int maxHeight = View.getDefaultSize(Integer.MAX_VALUE, heightSpec);
+      if (frameWidth == 0 || frameHeight == 0 || maxWidth == 0 || maxHeight == 0) {
+        return new Point(maxWidth, maxHeight);
+      }
+      // Calculate desired display size based on scaling type, video aspect ratio,
+      // and maximum layout size.
+      final float frameAspect = frameWidth / (float) frameHeight;
+      final float displayAspect = maxWidth / (float) maxHeight;
+      final float visibleFraction = (frameAspect > 1.0f) == (displayAspect > 1.0f)
+          ? visibleFractionMatchOrientation
+          : visibleFractionMismatchOrientation;
+      final Point layoutSize = getDisplaySize(visibleFraction, frameAspect, maxWidth, maxHeight);
+
+      // If the measure specification is forcing a specific size - yield.
+      if (View.MeasureSpec.getMode(widthSpec) == View.MeasureSpec.EXACTLY) {
+        layoutSize.x = maxWidth;
+      }
+      if (View.MeasureSpec.getMode(heightSpec) == View.MeasureSpec.EXACTLY) {
+        layoutSize.y = maxHeight;
+      }
+      return layoutSize;
+    }
+  }
+
+  // Types of video scaling:
+  // SCALE_ASPECT_FIT - video frame is scaled to fit the size of the view by
+  //    maintaining the aspect ratio (black borders may be displayed).
+  // SCALE_ASPECT_FILL - video frame is scaled to fill the size of the view by
+  //    maintaining the aspect ratio. Some portion of the video frame may be
+  //    clipped.
+  // SCALE_ASPECT_BALANCED - Compromise between FIT and FILL. Video frame will fill as much as
+  // possible of the view while maintaining aspect ratio, under the constraint that at least
+  // `BALANCED_VISIBLE_FRACTION` of the frame content will be shown.
+  public static enum ScalingType { SCALE_ASPECT_FIT, SCALE_ASPECT_FILL, SCALE_ASPECT_BALANCED }
+  // The minimum fraction of the frame content that will be shown for `SCALE_ASPECT_BALANCED`.
+  // This limits excessive cropping when adjusting display size.
+  private static float BALANCED_VISIBLE_FRACTION = 0.5625f;
+
+  /**
+   * Returns layout transformation matrix that applies an optional mirror effect and compensates
+   * for video vs display aspect ratio.
+   */
+  public static float[] getLayoutMatrix(
+      boolean mirror, float videoAspectRatio, float displayAspectRatio) {
+    float scaleX = 1;
+    float scaleY = 1;
+    // Scale X or Y dimension so that video and display size have same aspect ratio.
+    if (displayAspectRatio > videoAspectRatio) {
+      scaleY = videoAspectRatio / displayAspectRatio;
+    } else {
+      scaleX = displayAspectRatio / videoAspectRatio;
+    }
+    // Apply optional horizontal flip.
+    if (mirror) {
+      scaleX *= -1;
+    }
+    final float matrix[] = new float[16];
+    Matrix.setIdentityM(matrix, 0);
+    Matrix.scaleM(matrix, 0, scaleX, scaleY, 1);
+    adjustOrigin(matrix);
+    return matrix;
+  }
+
+  /** Converts a float[16] matrix array to android.graphics.Matrix. */
+  public static android.graphics.Matrix convertMatrixToAndroidGraphicsMatrix(float[] matrix4x4) {
+    // clang-format off
+    float[] values = {
+        matrix4x4[0 * 4 + 0], matrix4x4[1 * 4 + 0], matrix4x4[3 * 4 + 0],
+        matrix4x4[0 * 4 + 1], matrix4x4[1 * 4 + 1], matrix4x4[3 * 4 + 1],
+        matrix4x4[0 * 4 + 3], matrix4x4[1 * 4 + 3], matrix4x4[3 * 4 + 3],
+    };
+    // clang-format on
+
+    android.graphics.Matrix matrix = new android.graphics.Matrix();
+    matrix.setValues(values);
+    return matrix;
+  }
+
+  /** Converts android.graphics.Matrix to a float[16] matrix array. */
+  public static float[] convertMatrixFromAndroidGraphicsMatrix(android.graphics.Matrix matrix) {
+    float[] values = new float[9];
+    matrix.getValues(values);
+
+    // The android.graphics.Matrix looks like this:
+    // [x1 y1 w1]
+    // [x2 y2 w2]
+    // [x3 y3 w3]
+    // We want to contruct a matrix that looks like this:
+    // [x1 y1  0 w1]
+    // [x2 y2  0 w2]
+    // [ 0  0  1  0]
+    // [x3 y3  0 w3]
+    // Since it is stored in column-major order, it looks like this:
+    // [x1 x2 0 x3
+    //  y1 y2 0 y3
+    //   0  0 1  0
+    //  w1 w2 0 w3]
+    // clang-format off
+    float[] matrix4x4 = {
+        values[0 * 3 + 0],  values[1 * 3 + 0], 0,  values[2 * 3 + 0],
+        values[0 * 3 + 1],  values[1 * 3 + 1], 0,  values[2 * 3 + 1],
+        0,                  0,                 1,  0,
+        values[0 * 3 + 2],  values[1 * 3 + 2], 0,  values[2 * 3 + 2],
+    };
+    // clang-format on
+    return matrix4x4;
+  }
+
+  /**
+   * Calculate display size based on scaling type, video aspect ratio, and maximum display size.
+   */
+  public static Point getDisplaySize(
+      ScalingType scalingType, float videoAspectRatio, int maxDisplayWidth, int maxDisplayHeight) {
+    return getDisplaySize(convertScalingTypeToVisibleFraction(scalingType), videoAspectRatio,
+        maxDisplayWidth, maxDisplayHeight);
+  }
+
+  /**
+   * Move `matrix` transformation origin to (0.5, 0.5). This is the origin for texture coordinates
+   * that are in the range 0 to 1.
+   */
+  private static void adjustOrigin(float[] matrix) {
+    // Note that OpenGL is using column-major order.
+    // Pre translate with -0.5 to move coordinates to range [-0.5, 0.5].
+    matrix[12] -= 0.5f * (matrix[0] + matrix[4]);
+    matrix[13] -= 0.5f * (matrix[1] + matrix[5]);
+    // Post translate with 0.5 to move coordinates to range [0, 1].
+    matrix[12] += 0.5f;
+    matrix[13] += 0.5f;
+  }
+
+  /**
+   * Each scaling type has a one-to-one correspondence to a numeric minimum fraction of the video
+   * that must remain visible.
+   */
+  private static float convertScalingTypeToVisibleFraction(ScalingType scalingType) {
+    switch (scalingType) {
+      case SCALE_ASPECT_FIT:
+        return 1.0f;
+      case SCALE_ASPECT_FILL:
+        return 0.0f;
+      case SCALE_ASPECT_BALANCED:
+        return BALANCED_VISIBLE_FRACTION;
+      default:
+        throw new IllegalArgumentException();
+    }
+  }
+
+  /**
+   * Calculate display size based on minimum fraction of the video that must remain visible,
+   * video aspect ratio, and maximum display size.
+   */
+  public static Point getDisplaySize(
+      float minVisibleFraction, float videoAspectRatio, int maxDisplayWidth, int maxDisplayHeight) {
+    // If there is no constraint on the amount of cropping, fill the allowed display area.
+    if (minVisibleFraction == 0 || videoAspectRatio == 0) {
+      return new Point(maxDisplayWidth, maxDisplayHeight);
+    }
+    // Each dimension is constrained on max display size and how much we are allowed to crop.
+    final int width = Math.min(
+        maxDisplayWidth, Math.round(maxDisplayHeight / minVisibleFraction * videoAspectRatio));
+    final int height = Math.min(
+        maxDisplayHeight, Math.round(maxDisplayWidth / minVisibleFraction / videoAspectRatio));
+    return new Point(width, height);
+  }
+}
diff --git a/third_party/libwebrtc/sdk/android/api/org/webrtc/SSLCertificateVerifier.java b/third_party/libwebrtc/sdk/android/api/org/webrtc/SSLCertificateVerifier.java
new file mode 100644
index 0000000000..461cd3b143
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/api/org/webrtc/SSLCertificateVerifier.java
@@ -0,0 +1,27 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+/**
+ * The SSLCertificateVerifier interface allows API users to provide custom
+ * logic to verify certificates.
+ */
+public interface SSLCertificateVerifier {
+  /**
+   * Implementations of verify allow applications to provide custom logic for
+   * verifying certificates. This is not required by default and should be used
+   * with care.
+   *
+   * @param certificate A byte array containing a DER encoded X509 certificate.
+   * @return True if the certificate is verified and trusted else false.
+   */
+  @CalledByNative boolean verify(byte[] certificate);
+}
diff --git a/third_party/libwebrtc/sdk/android/api/org/webrtc/ScreenCapturerAndroid.java b/third_party/libwebrtc/sdk/android/api/org/webrtc/ScreenCapturerAndroid.java
new file mode 100644
index 0000000000..231d507155
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/api/org/webrtc/ScreenCapturerAndroid.java
@@ -0,0 +1,212 @@
+/*
+ *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+import android.app.Activity;
+import android.content.Context;
+import android.content.Intent;
+import android.hardware.display.DisplayManager;
+import android.hardware.display.VirtualDisplay;
+import android.media.projection.MediaProjection;
+import android.media.projection.MediaProjectionManager;
+import android.view.Surface;
+import androidx.annotation.Nullable;
+
+/**
+ * An implementation of VideoCapturer to capture the screen content as a video stream.
+ * Capturing is done by {@code MediaProjection} on a {@code SurfaceTexture}. We interact with this
+ * {@code SurfaceTexture} using a {@code SurfaceTextureHelper}.
+ * The {@code SurfaceTextureHelper} is created by the native code and passed to this capturer in
+ * {@code VideoCapturer.initialize()}. On receiving a new frame, this capturer passes it
+ * as a texture to the native code via {@code CapturerObserver.onFrameCaptured()}. This takes
+ * place on the HandlerThread of the given {@code SurfaceTextureHelper}. When done with each frame,
+ * the native code returns the buffer to the  {@code SurfaceTextureHelper} to be used for new
+ * frames. At any time, at most one frame is being processed.
+ */
+public class ScreenCapturerAndroid implements VideoCapturer, VideoSink {
+  private static final int DISPLAY_FLAGS =
+      DisplayManager.VIRTUAL_DISPLAY_FLAG_PUBLIC | DisplayManager.VIRTUAL_DISPLAY_FLAG_PRESENTATION;
+  // DPI for VirtualDisplay, does not seem to matter for us.
+  private static final int VIRTUAL_DISPLAY_DPI = 400;
+
+  private final Intent mediaProjectionPermissionResultData;
+  private final MediaProjection.Callback mediaProjectionCallback;
+
+  private int width;
+  private int height;
+  @Nullable private VirtualDisplay virtualDisplay;
+  @Nullable private SurfaceTextureHelper surfaceTextureHelper;
+  @Nullable private CapturerObserver capturerObserver;
+  private long numCapturedFrames;
+  @Nullable private MediaProjection mediaProjection;
+  private boolean isDisposed;
+  @Nullable private MediaProjectionManager mediaProjectionManager;
+
+  /**
+   * Constructs a new Screen Capturer.
+   *
+   * @param mediaProjectionPermissionResultData the result data of MediaProjection permission
+   *     activity; the calling app must validate that result code is Activity.RESULT_OK before
+   *     calling this method.
+   * @param mediaProjectionCallback MediaProjection callback to implement application specific
+   *     logic in events such as when the user revokes a previously granted capture permission.
+  **/
+  public ScreenCapturerAndroid(Intent mediaProjectionPermissionResultData,
+      MediaProjection.Callback mediaProjectionCallback) {
+    this.mediaProjectionPermissionResultData = mediaProjectionPermissionResultData;
+    this.mediaProjectionCallback = mediaProjectionCallback;
+  }
+
+  private void checkNotDisposed() {
+    if (isDisposed) {
+      throw new RuntimeException("capturer is disposed.");
+    }
+  }
+
+  @Nullable
+  public MediaProjection getMediaProjection() {
+    return mediaProjection;
+  }
+
+  @Override
+  // TODO(bugs.webrtc.org/8491): Remove NoSynchronizedMethodCheck suppression.
+  @SuppressWarnings("NoSynchronizedMethodCheck")
+  public synchronized void initialize(final SurfaceTextureHelper surfaceTextureHelper,
+      final Context applicationContext, final CapturerObserver capturerObserver) {
+    checkNotDisposed();
+
+    if (capturerObserver == null) {
+      throw new RuntimeException("capturerObserver not set.");
+    }
+    this.capturerObserver = capturerObserver;
+
+    if (surfaceTextureHelper == null) {
+      throw new RuntimeException("surfaceTextureHelper not set.");
+    }
+    this.surfaceTextureHelper = surfaceTextureHelper;
+
+    mediaProjectionManager = (MediaProjectionManager) applicationContext.getSystemService(
+        Context.MEDIA_PROJECTION_SERVICE);
+  }
+
+  @Override
+  // TODO(bugs.webrtc.org/8491): Remove NoSynchronizedMethodCheck suppression.
+  @SuppressWarnings("NoSynchronizedMethodCheck")
+  public synchronized void startCapture(
+      final int width, final int height, final int ignoredFramerate) {
+    checkNotDisposed();
+
+    this.width = width;
+    this.height = height;
+
+    mediaProjection = mediaProjectionManager.getMediaProjection(
+        Activity.RESULT_OK, mediaProjectionPermissionResultData);
+
+    // Let MediaProjection callback use the SurfaceTextureHelper thread.
+    mediaProjection.registerCallback(mediaProjectionCallback, surfaceTextureHelper.getHandler());
+
+    createVirtualDisplay();
+    capturerObserver.onCapturerStarted(true);
+    surfaceTextureHelper.startListening(ScreenCapturerAndroid.this);
+  }
+
+  @Override
+  // TODO(bugs.webrtc.org/8491): Remove NoSynchronizedMethodCheck suppression.
+  @SuppressWarnings("NoSynchronizedMethodCheck")
+  public synchronized void stopCapture() {
+    checkNotDisposed();
+    ThreadUtils.invokeAtFrontUninterruptibly(surfaceTextureHelper.getHandler(), new Runnable() {
+      @Override
+      public void run() {
+        surfaceTextureHelper.stopListening();
+        capturerObserver.onCapturerStopped();
+
+        if (virtualDisplay != null) {
+          virtualDisplay.release();
+          virtualDisplay = null;
+        }
+
+        if (mediaProjection != null) {
+          // Unregister the callback before stopping, otherwise the callback recursively
+          // calls this method.
+          mediaProjection.unregisterCallback(mediaProjectionCallback);
+          mediaProjection.stop();
+          mediaProjection = null;
+        }
+      }
+    });
+  }
+
+  @Override
+  // TODO(bugs.webrtc.org/8491): Remove NoSynchronizedMethodCheck suppression.
+  @SuppressWarnings("NoSynchronizedMethodCheck")
+  public synchronized void dispose() {
+    isDisposed = true;
+  }
+
+  /**
+   * Changes output video format. This method can be used to scale the output
+   * video, or to change orientation when the captured screen is rotated for example.
+   *
+   * @param width new output video width
+   * @param height new output video height
+   * @param ignoredFramerate ignored
+   */
+  @Override
+  // TODO(bugs.webrtc.org/8491): Remove NoSynchronizedMethodCheck suppression.
+  @SuppressWarnings("NoSynchronizedMethodCheck")
+  public synchronized void changeCaptureFormat(
+      final int width, final int height, final int ignoredFramerate) {
+    checkNotDisposed();
+
+    this.width = width;
+    this.height = height;
+
+    if (virtualDisplay == null) {
+      // Capturer is stopped, the virtual display will be created in startCaptuer().
+      return;
+    }
+
+    // Create a new virtual display on the surfaceTextureHelper thread to avoid interference
+    // with frame processing, which happens on the same thread (we serialize events by running
+    // them on the same thread).
+    ThreadUtils.invokeAtFrontUninterruptibly(surfaceTextureHelper.getHandler(), new Runnable() {
+      @Override
+      public void run() {
+        virtualDisplay.release();
+        createVirtualDisplay();
+      }
+    });
+  }
+
+  private void createVirtualDisplay() {
+    surfaceTextureHelper.setTextureSize(width, height);
+    virtualDisplay = mediaProjection.createVirtualDisplay("WebRTC_ScreenCapture", width, height,
+        VIRTUAL_DISPLAY_DPI, DISPLAY_FLAGS, new Surface(surfaceTextureHelper.getSurfaceTexture()),
+        null /* callback */, null /* callback handler */);
+  }
+
+  // This is called on the internal looper thread of {@Code SurfaceTextureHelper}.
+  @Override
+  public void onFrame(VideoFrame frame) {
+    numCapturedFrames++;
+    capturerObserver.onFrameCaptured(frame);
+  }
+
+  @Override
+  public boolean isScreencast() {
+    return true;
+  }
+
+  public long getNumCapturedFrames() {
+    return numCapturedFrames;
+  }
+}
diff --git a/third_party/libwebrtc/sdk/android/api/org/webrtc/SdpObserver.java b/third_party/libwebrtc/sdk/android/api/org/webrtc/SdpObserver.java
new file mode 100644
index 0000000000..afa99bc552
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/api/org/webrtc/SdpObserver.java
@@ -0,0 +1,26 @@
+/*
+ *  Copyright 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+/** Interface for observing SDP-related events. */
+public interface SdpObserver {
+  /** Called on success of Create{Offer,Answer}(). */
+  @CalledByNative void onCreateSuccess(SessionDescription sdp);
+
+  /** Called on success of Set{Local,Remote}Description(). */
+  @CalledByNative void onSetSuccess();
+
+  /** Called on error of Create{Offer,Answer}(). */
+  @CalledByNative void onCreateFailure(String error);
+
+  /** Called on error of Set{Local,Remote}Description(). */
+  @CalledByNative void onSetFailure(String error);
+}
diff --git a/third_party/libwebrtc/sdk/android/api/org/webrtc/SessionDescription.java b/third_party/libwebrtc/sdk/android/api/org/webrtc/SessionDescription.java
new file mode 100644
index 0000000000..be89599a5f
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/api/org/webrtc/SessionDescription.java
@@ -0,0 +1,56 @@
+/*
+ *  Copyright 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+import java.util.Locale;
+
+/**
+ * Description of an RFC 4566 Session.
+ * SDPs are passed as serialized Strings in Java-land and are materialized
+ * to SessionDescriptionInterface as appropriate in the JNI layer.
+ */
+public class SessionDescription {
+  /** Java-land enum version of SessionDescriptionInterface's type() string. */
+  public static enum Type {
+    OFFER,
+    PRANSWER,
+    ANSWER,
+    ROLLBACK;
+
+    public String canonicalForm() {
+      return name().toLowerCase(Locale.US);
+    }
+
+    @CalledByNative("Type")
+    public static Type fromCanonicalForm(String canonical) {
+      return Type.valueOf(Type.class, canonical.toUpperCase(Locale.US));
+    }
+  }
+
+  public final Type type;
+  public final String description;
+
+  @CalledByNative
+  public SessionDescription(Type type, String description) {
+    this.type = type;
+    this.description = description;
+  }
+
+  @CalledByNative
+  String getDescription() {
+    return description;
+  }
+
+  @CalledByNative
+  String getTypeInCanonicalForm() {
+    return type.canonicalForm();
+  }
+}
diff --git a/third_party/libwebrtc/sdk/android/api/org/webrtc/SoftwareVideoDecoderFactory.java b/third_party/libwebrtc/sdk/android/api/org/webrtc/SoftwareVideoDecoderFactory.java
new file mode 100644
index 0000000000..2ac42e834e
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/api/org/webrtc/SoftwareVideoDecoderFactory.java
@@ -0,0 +1,53 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+import androidx.annotation.Nullable;
+import java.util.Arrays;
+import java.util.List;
+
+public class SoftwareVideoDecoderFactory implements VideoDecoderFactory {
+  private static final String TAG = "SoftwareVideoDecoderFactory";
+
+  private final long nativeFactory;
+
+  public SoftwareVideoDecoderFactory() {
+    this.nativeFactory = nativeCreateFactory();
+  }
+
+  @Nullable
+  @Override
+  public VideoDecoder createDecoder(VideoCodecInfo info) {
+    long nativeDecoder = nativeCreateDecoder(nativeFactory, info);
+    if (nativeDecoder == 0) {
+      Logging.w(TAG, "Trying to create decoder for unsupported format. " + info);
+      return null;
+    }
+
+    return new WrappedNativeVideoDecoder() {
+      @Override
+      public long createNativeVideoDecoder() {
+        return nativeDecoder;
+      }
+    };
+  }
+
+  @Override
+  public VideoCodecInfo[] getSupportedCodecs() {
+    return nativeGetSupportedCodecs(nativeFactory).toArray(new VideoCodecInfo[0]);
+  }
+
+  private static native long nativeCreateFactory();
+
+  private static native long nativeCreateDecoder(long factory, VideoCodecInfo videoCodecInfo);
+
+  private static native List<VideoCodecInfo> nativeGetSupportedCodecs(long factory);
+}
diff --git a/third_party/libwebrtc/sdk/android/api/org/webrtc/SoftwareVideoEncoderFactory.java b/third_party/libwebrtc/sdk/android/api/org/webrtc/SoftwareVideoEncoderFactory.java
new file mode 100644
index 0000000000..7f4c457b97
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/api/org/webrtc/SoftwareVideoEncoderFactory.java
@@ -0,0 +1,58 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+import androidx.annotation.Nullable;
+import java.util.Arrays;
+import java.util.List;
+
+public class SoftwareVideoEncoderFactory implements VideoEncoderFactory {
+  private static final String TAG = "SoftwareVideoEncoderFactory";
+
+  private final long nativeFactory;
+
+  public SoftwareVideoEncoderFactory() {
+    this.nativeFactory = nativeCreateFactory();
+  }
+
+  @Nullable
+  @Override
+  public VideoEncoder createEncoder(VideoCodecInfo info) {
+    long nativeEncoder = nativeCreateEncoder(nativeFactory, info);
+    if (nativeEncoder == 0) {
+      Logging.w(TAG, "Trying to create encoder for unsupported format. " + info);
+      return null;
+    }
+
+    return new WrappedNativeVideoEncoder() {
+      @Override
+      public long createNativeVideoEncoder() {
+        return nativeEncoder;
+      }
+
+      @Override
+      public boolean isHardwareEncoder() {
+        return false;
+      }
+    };
+  }
+
+  @Override
+  public VideoCodecInfo[] getSupportedCodecs() {
+    return nativeGetSupportedCodecs(nativeFactory).toArray(new VideoCodecInfo[0]);
+  }
+
+  private static native long nativeCreateFactory();
+
+  private static native long nativeCreateEncoder(long factory, VideoCodecInfo videoCodecInfo);
+
+  private static native List<VideoCodecInfo> nativeGetSupportedCodecs(long factory);
+}
diff --git a/third_party/libwebrtc/sdk/android/api/org/webrtc/StatsObserver.java b/third_party/libwebrtc/sdk/android/api/org/webrtc/StatsObserver.java
new file mode 100644
index 0000000000..b9984c18db
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/api/org/webrtc/StatsObserver.java
@@ -0,0 +1,17 @@
+/*
+ *  Copyright 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+/** Interface for observing Stats reports (see webrtc::StatsObservers). */
+public interface StatsObserver {
+  /** Called when the reports are ready.*/
+  @CalledByNative public void onComplete(StatsReport[] reports);
+}
diff --git a/third_party/libwebrtc/sdk/android/api/org/webrtc/StatsReport.java b/third_party/libwebrtc/sdk/android/api/org/webrtc/StatsReport.java
new file mode 100644
index 0000000000..b8f1cf87fe
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/api/org/webrtc/StatsReport.java
@@ -0,0 +1,63 @@
+/*
+ *  Copyright 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+/** Java version of webrtc::StatsReport. */
+public class StatsReport {
+  /** Java version of webrtc::StatsReport::Value. */
+  public static class Value {
+    public final String name;
+    public final String value;
+
+    @CalledByNative("Value")
+    public Value(String name, String value) {
+      this.name = name;
+      this.value = value;
+    }
+
+    @Override
+    public String toString() {
+      StringBuilder builder = new StringBuilder();
+      builder.append("[").append(name).append(": ").append(value).append("]");
+      return builder.toString();
+    }
+  }
+
+  public final String id;
+  public final String type;
+  // Time since 1970-01-01T00:00:00Z in milliseconds.
+  public final double timestamp;
+  public final Value[] values;
+
+  @CalledByNative
+  public StatsReport(String id, String type, double timestamp, Value[] values) {
+    this.id = id;
+    this.type = type;
+    this.timestamp = timestamp;
+    this.values = values;
+  }
+
+  @Override
+  public String toString() {
+    StringBuilder builder = new StringBuilder();
+    builder.append("id: ")
+        .append(id)
+        .append(", type: ")
+        .append(type)
+        .append(", timestamp: ")
+        .append(timestamp)
+        .append(", values: ");
+    for (int i = 0; i < values.length; ++i) {
+      builder.append(values[i].toString()).append(", ");
+    }
+    return builder.toString();
+  }
+}
diff --git a/third_party/libwebrtc/sdk/android/api/org/webrtc/SurfaceEglRenderer.java b/third_party/libwebrtc/sdk/android/api/org/webrtc/SurfaceEglRenderer.java
new file mode 100644
index 0000000000..6cba3f473b
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/api/org/webrtc/SurfaceEglRenderer.java
@@ -0,0 +1,160 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+import android.view.SurfaceHolder;
+import java.util.concurrent.CountDownLatch;
+
+/**
+ * Display the video stream on a Surface.
+ * renderFrame() is asynchronous to avoid blocking the calling thread.
+ * This class is thread safe and handles access from potentially three different threads:
+ * Interaction from the main app in init, release and setMirror.
+ * Interaction from C++ rtc::VideoSinkInterface in renderFrame.
+ * Interaction from SurfaceHolder lifecycle in surfaceCreated, surfaceChanged, and surfaceDestroyed.
+ */
+public class SurfaceEglRenderer extends EglRenderer implements SurfaceHolder.Callback {
+  private static final String TAG = "SurfaceEglRenderer";
+
+  // Callback for reporting renderer events. Read-only after initialization so no lock required.
+  private RendererCommon.RendererEvents rendererEvents;
+
+  private final Object layoutLock = new Object();
+  private boolean isRenderingPaused;
+  private boolean isFirstFrameRendered;
+  private int rotatedFrameWidth;
+  private int rotatedFrameHeight;
+  private int frameRotation;
+
+  /**
+   * In order to render something, you must first call init().
+   */
+  public SurfaceEglRenderer(String name) {
+    super(name);
+  }
+
+  /**
+   * Initialize this class, sharing resources with `sharedContext`. The custom `drawer` will be used
+   * for drawing frames on the EGLSurface. This class is responsible for calling release() on
+   * `drawer`. It is allowed to call init() to reinitialize the renderer after a previous
+   * init()/release() cycle.
+   */
+  public void init(final EglBase.Context sharedContext,
+      RendererCommon.RendererEvents rendererEvents, final int[] configAttributes,
+      RendererCommon.GlDrawer drawer) {
+    ThreadUtils.checkIsOnMainThread();
+    this.rendererEvents = rendererEvents;
+    synchronized (layoutLock) {
+      isFirstFrameRendered = false;
+      rotatedFrameWidth = 0;
+      rotatedFrameHeight = 0;
+      frameRotation = 0;
+    }
+    super.init(sharedContext, configAttributes, drawer);
+  }
+
+  @Override
+  public void init(final EglBase.Context sharedContext, final int[] configAttributes,
+      RendererCommon.GlDrawer drawer) {
+    init(sharedContext, null /* rendererEvents */, configAttributes, drawer);
+  }
+
+  /**
+   * Limit render framerate.
+   *
+   * @param fps Limit render framerate to this value, or use Float.POSITIVE_INFINITY to disable fps
+   *            reduction.
+   */
+  @Override
+  public void setFpsReduction(float fps) {
+    synchronized (layoutLock) {
+      isRenderingPaused = fps == 0f;
+    }
+    super.setFpsReduction(fps);
+  }
+
+  @Override
+  public void disableFpsReduction() {
+    synchronized (layoutLock) {
+      isRenderingPaused = false;
+    }
+    super.disableFpsReduction();
+  }
+
+  @Override
+  public void pauseVideo() {
+    synchronized (layoutLock) {
+      isRenderingPaused = true;
+    }
+    super.pauseVideo();
+  }
+
+  // VideoSink interface.
+  @Override
+  public void onFrame(VideoFrame frame) {
+    updateFrameDimensionsAndReportEvents(frame);
+    super.onFrame(frame);
+  }
+
+  // SurfaceHolder.Callback interface.
+  @Override
+  public void surfaceCreated(final SurfaceHolder holder) {
+    ThreadUtils.checkIsOnMainThread();
+    createEglSurface(holder.getSurface());
+  }
+
+  @Override
+  public void surfaceDestroyed(SurfaceHolder holder) {
+    ThreadUtils.checkIsOnMainThread();
+    final CountDownLatch completionLatch = new CountDownLatch(1);
+    releaseEglSurface(completionLatch::countDown);
+    ThreadUtils.awaitUninterruptibly(completionLatch);
+  }
+
+  @Override
+  public void surfaceChanged(SurfaceHolder holder, int format, int width, int height) {
+    ThreadUtils.checkIsOnMainThread();
+    logD("surfaceChanged: format: " + format + " size: " + width + "x" + height);
+  }
+
+  // Update frame dimensions and report any changes to `rendererEvents`.
+  private void updateFrameDimensionsAndReportEvents(VideoFrame frame) {
+    synchronized (layoutLock) {
+      if (isRenderingPaused) {
+        return;
+      }
+      if (!isFirstFrameRendered) {
+        isFirstFrameRendered = true;
+        logD("Reporting first rendered frame.");
+        if (rendererEvents != null) {
+          rendererEvents.onFirstFrameRendered();
+        }
+      }
+      if (rotatedFrameWidth != frame.getRotatedWidth()
+          || rotatedFrameHeight != frame.getRotatedHeight()
+          || frameRotation != frame.getRotation()) {
+        logD("Reporting frame resolution changed to " + frame.getBuffer().getWidth() + "x"
+            + frame.getBuffer().getHeight() + " with rotation " + frame.getRotation());
+        if (rendererEvents != null) {
+          rendererEvents.onFrameResolutionChanged(
+              frame.getBuffer().getWidth(), frame.getBuffer().getHeight(), frame.getRotation());
+        }
+        rotatedFrameWidth = frame.getRotatedWidth();
+        rotatedFrameHeight = frame.getRotatedHeight();
+        frameRotation = frame.getRotation();
+      }
+    }
+  }
+
+  private void logD(String string) {
+    Logging.d(TAG, name + ": " + string);
+  }
+}
diff --git a/third_party/libwebrtc/sdk/android/api/org/webrtc/SurfaceTextureHelper.java b/third_party/libwebrtc/sdk/android/api/org/webrtc/SurfaceTextureHelper.java
new file mode 100644
index 0000000000..3ea22736ea
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/api/org/webrtc/SurfaceTextureHelper.java
@@ -0,0 +1,390 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+import android.annotation.TargetApi;
+import android.graphics.SurfaceTexture;
+import android.opengl.GLES11Ext;
+import android.opengl.GLES20;
+import android.os.Build;
+import android.os.Handler;
+import android.os.HandlerThread;
+import androidx.annotation.Nullable;
+import java.util.concurrent.Callable;
+import org.webrtc.EglBase.Context;
+import org.webrtc.TextureBufferImpl.RefCountMonitor;
+import org.webrtc.VideoFrame.TextureBuffer;
+
+/**
+ * Helper class for using a SurfaceTexture to create WebRTC VideoFrames. In order to create WebRTC
+ * VideoFrames, render onto the SurfaceTexture. The frames will be delivered to the listener. Only
+ * one texture frame can be in flight at once, so the frame must be released in order to receive a
+ * new frame. Call stopListening() to stop receiveing new frames. Call dispose to release all
+ * resources once the texture frame is released.
+ */
+public class SurfaceTextureHelper {
+  /**
+   * Interface for monitoring texture buffers created from this SurfaceTexture. Since only one
+   * texture buffer can exist at a time, this can be used to monitor for stuck frames.
+   */
+  public interface FrameRefMonitor {
+    /** A new frame was created. New frames start with ref count of 1. */
+    void onNewBuffer(TextureBuffer textureBuffer);
+    /** Ref count of the frame was incremented by the calling thread. */
+    void onRetainBuffer(TextureBuffer textureBuffer);
+    /** Ref count of the frame was decremented by the calling thread. */
+    void onReleaseBuffer(TextureBuffer textureBuffer);
+    /** Frame was destroyed (ref count reached 0). */
+    void onDestroyBuffer(TextureBuffer textureBuffer);
+  }
+
+  private static final String TAG = "SurfaceTextureHelper";
+  /**
+   * Construct a new SurfaceTextureHelper sharing OpenGL resources with `sharedContext`. A dedicated
+   * thread and handler is created for handling the SurfaceTexture. May return null if EGL fails to
+   * initialize a pixel buffer surface and make it current. If alignTimestamps is true, the frame
+   * timestamps will be aligned to rtc::TimeNanos(). If frame timestamps are aligned to
+   * rtc::TimeNanos() there is no need for aligning timestamps again in
+   * PeerConnectionFactory.createVideoSource(). This makes the timestamps more accurate and
+   * closer to actual creation time.
+   */
+  public static SurfaceTextureHelper create(final String threadName,
+      final EglBase.Context sharedContext, boolean alignTimestamps, final YuvConverter yuvConverter,
+      FrameRefMonitor frameRefMonitor) {
+    final HandlerThread thread = new HandlerThread(threadName);
+    thread.start();
+    final Handler handler = new Handler(thread.getLooper());
+
+    // The onFrameAvailable() callback will be executed on the SurfaceTexture ctor thread. See:
+    // http://grepcode.com/file/repository.grepcode.com/java/ext/com.google.android/android/5.1.1_r1/android/graphics/SurfaceTexture.java#195.
+    // Therefore, in order to control the callback thread on API lvl < 21, the SurfaceTextureHelper
+    // is constructed on the `handler` thread.
+    return ThreadUtils.invokeAtFrontUninterruptibly(handler, new Callable<SurfaceTextureHelper>() {
+      @Nullable
+      @Override
+      public SurfaceTextureHelper call() {
+        try {
+          return new SurfaceTextureHelper(
+              sharedContext, handler, alignTimestamps, yuvConverter, frameRefMonitor);
+        } catch (RuntimeException e) {
+          Logging.e(TAG, threadName + " create failure", e);
+          return null;
+        }
+      }
+    });
+  }
+
+  /**
+   * Same as above with alignTimestamps set to false and yuvConverter set to new YuvConverter.
+   *
+   * @see #create(String, EglBase.Context, boolean, YuvConverter, FrameRefMonitor)
+   */
+  public static SurfaceTextureHelper create(
+      final String threadName, final EglBase.Context sharedContext) {
+    return create(threadName, sharedContext, /* alignTimestamps= */ false, new YuvConverter(),
+        /*frameRefMonitor=*/null);
+  }
+
+  /**
+   * Same as above with yuvConverter set to new YuvConverter.
+   *
+   * @see #create(String, EglBase.Context, boolean, YuvConverter, FrameRefMonitor)
+   */
+  public static SurfaceTextureHelper create(
+      final String threadName, final EglBase.Context sharedContext, boolean alignTimestamps) {
+    return create(
+        threadName, sharedContext, alignTimestamps, new YuvConverter(), /*frameRefMonitor=*/null);
+  }
+
+  /**
+   * Create a SurfaceTextureHelper without frame ref monitor.
+   *
+   * @see #create(String, EglBase.Context, boolean, YuvConverter, FrameRefMonitor)
+   */
+  public static SurfaceTextureHelper create(final String threadName,
+      final EglBase.Context sharedContext, boolean alignTimestamps, YuvConverter yuvConverter) {
+    return create(
+        threadName, sharedContext, alignTimestamps, yuvConverter, /*frameRefMonitor=*/null);
+  }
+
+  private final RefCountMonitor textureRefCountMonitor = new RefCountMonitor() {
+    @Override
+    public void onRetain(TextureBufferImpl textureBuffer) {
+      if (frameRefMonitor != null) {
+        frameRefMonitor.onRetainBuffer(textureBuffer);
+      }
+    }
+
+    @Override
+    public void onRelease(TextureBufferImpl textureBuffer) {
+      if (frameRefMonitor != null) {
+        frameRefMonitor.onReleaseBuffer(textureBuffer);
+      }
+    }
+
+    @Override
+    public void onDestroy(TextureBufferImpl textureBuffer) {
+      returnTextureFrame();
+      if (frameRefMonitor != null) {
+        frameRefMonitor.onDestroyBuffer(textureBuffer);
+      }
+    }
+  };
+
+  private final Handler handler;
+  private final EglBase eglBase;
+  private final SurfaceTexture surfaceTexture;
+  private final int oesTextureId;
+  private final YuvConverter yuvConverter;
+  @Nullable private final TimestampAligner timestampAligner;
+  private final FrameRefMonitor frameRefMonitor;
+
+  // These variables are only accessed from the `handler` thread.
+  @Nullable private VideoSink listener;
+  // The possible states of this class.
+  private boolean hasPendingTexture;
+  private volatile boolean isTextureInUse;
+  private boolean isQuitting;
+  private int frameRotation;
+  private int textureWidth;
+  private int textureHeight;
+  // `pendingListener` is set in setListener() and the runnable is posted to the handler thread.
+  // setListener() is not allowed to be called again before stopListening(), so this is thread safe.
+  @Nullable private VideoSink pendingListener;
+  final Runnable setListenerRunnable = new Runnable() {
+    @Override
+    public void run() {
+      Logging.d(TAG, "Setting listener to " + pendingListener);
+      listener = pendingListener;
+      pendingListener = null;
+      // May have a pending frame from the previous capture session - drop it.
+      if (hasPendingTexture) {
+        // Calling updateTexImage() is neccessary in order to receive new frames.
+        updateTexImage();
+        hasPendingTexture = false;
+      }
+    }
+  };
+
+  private SurfaceTextureHelper(Context sharedContext, Handler handler, boolean alignTimestamps,
+      YuvConverter yuvConverter, FrameRefMonitor frameRefMonitor) {
+    if (handler.getLooper().getThread() != Thread.currentThread()) {
+      throw new IllegalStateException("SurfaceTextureHelper must be created on the handler thread");
+    }
+    this.handler = handler;
+    this.timestampAligner = alignTimestamps ? new TimestampAligner() : null;
+    this.yuvConverter = yuvConverter;
+    this.frameRefMonitor = frameRefMonitor;
+
+    eglBase = EglBase.create(sharedContext, EglBase.CONFIG_PIXEL_BUFFER);
+    try {
+      // Both these statements have been observed to fail on rare occasions, see BUG=webrtc:5682.
+      eglBase.createDummyPbufferSurface();
+      eglBase.makeCurrent();
+    } catch (RuntimeException e) {
+      // Clean up before rethrowing the exception.
+      eglBase.release();
+      handler.getLooper().quit();
+      throw e;
+    }
+
+    oesTextureId = GlUtil.generateTexture(GLES11Ext.GL_TEXTURE_EXTERNAL_OES);
+    surfaceTexture = new SurfaceTexture(oesTextureId);
+    surfaceTexture.setOnFrameAvailableListener(st -> {
+      if (hasPendingTexture) {
+        Logging.d(TAG, "A frame is already pending, dropping frame.");
+      }
+
+      hasPendingTexture = true;
+      tryDeliverTextureFrame();
+    }, handler);
+  }
+
+  /**
+   * Start to stream textures to the given `listener`. If you need to change listener, you need to
+   * call stopListening() first.
+   */
+  public void startListening(final VideoSink listener) {
+    if (this.listener != null || this.pendingListener != null) {
+      throw new IllegalStateException("SurfaceTextureHelper listener has already been set.");
+    }
+    this.pendingListener = listener;
+    handler.post(setListenerRunnable);
+  }
+
+  /**
+   * Stop listening. The listener set in startListening() is guaranteded to not receive any more
+   * onFrame() callbacks after this function returns.
+   */
+  public void stopListening() {
+    Logging.d(TAG, "stopListening()");
+    handler.removeCallbacks(setListenerRunnable);
+    ThreadUtils.invokeAtFrontUninterruptibly(handler, () -> {
+      listener = null;
+      pendingListener = null;
+    });
+  }
+
+  /**
+   * Use this function to set the texture size. Note, do not call setDefaultBufferSize() yourself
+   * since this class needs to be aware of the texture size.
+   */
+  public void setTextureSize(int textureWidth, int textureHeight) {
+    if (textureWidth <= 0) {
+      throw new IllegalArgumentException("Texture width must be positive, but was " + textureWidth);
+    }
+    if (textureHeight <= 0) {
+      throw new IllegalArgumentException(
+          "Texture height must be positive, but was " + textureHeight);
+    }
+    surfaceTexture.setDefaultBufferSize(textureWidth, textureHeight);
+    handler.post(() -> {
+      this.textureWidth = textureWidth;
+      this.textureHeight = textureHeight;
+      tryDeliverTextureFrame();
+    });
+  }
+
+  /**
+   * Forces a frame to be produced. If no new frame is available, the last frame is sent to the
+   * listener again.
+   */
+  public void forceFrame() {
+    handler.post(() -> {
+      hasPendingTexture = true;
+      tryDeliverTextureFrame();
+    });
+  }
+
+  /** Set the rotation of the delivered frames. */
+  public void setFrameRotation(int rotation) {
+    handler.post(() -> this.frameRotation = rotation);
+  }
+
+  /**
+   * Retrieve the underlying SurfaceTexture. The SurfaceTexture should be passed in to a video
+   * producer such as a camera or decoder.
+   */
+  public SurfaceTexture getSurfaceTexture() {
+    return surfaceTexture;
+  }
+
+  /** Retrieve the handler that calls onFrame(). This handler is valid until dispose() is called. */
+  public Handler getHandler() {
+    return handler;
+  }
+
+  /**
+   * This function is called when the texture frame is released. Only one texture frame can be in
+   * flight at once, so this function must be called before a new frame is delivered.
+   */
+  private void returnTextureFrame() {
+    handler.post(() -> {
+      isTextureInUse = false;
+      if (isQuitting) {
+        release();
+      } else {
+        tryDeliverTextureFrame();
+      }
+    });
+  }
+
+  public boolean isTextureInUse() {
+    return isTextureInUse;
+  }
+
+  /**
+   * Call disconnect() to stop receiving frames. OpenGL resources are released and the handler is
+   * stopped when the texture frame has been released. You are guaranteed to not receive any more
+   * onFrame() after this function returns.
+   */
+  public void dispose() {
+    Logging.d(TAG, "dispose()");
+    ThreadUtils.invokeAtFrontUninterruptibly(handler, () -> {
+      isQuitting = true;
+      if (!isTextureInUse) {
+        release();
+      }
+    });
+  }
+
+  /**
+   * Posts to the correct thread to convert `textureBuffer` to I420.
+   *
+   * @deprecated Use toI420() instead.
+   */
+  @Deprecated
+  public VideoFrame.I420Buffer textureToYuv(final TextureBuffer textureBuffer) {
+    return textureBuffer.toI420();
+  }
+
+  private void updateTexImage() {
+    // SurfaceTexture.updateTexImage apparently can compete and deadlock with eglSwapBuffers,
+    // as observed on Nexus 5. Therefore, synchronize it with the EGL functions.
+    // See https://bugs.chromium.org/p/webrtc/issues/detail?id=5702 for more info.
+    synchronized (EglBase.lock) {
+      surfaceTexture.updateTexImage();
+    }
+  }
+
+  private void tryDeliverTextureFrame() {
+    if (handler.getLooper().getThread() != Thread.currentThread()) {
+      throw new IllegalStateException("Wrong thread.");
+    }
+    if (isQuitting || !hasPendingTexture || isTextureInUse || listener == null) {
+      return;
+    }
+    if (textureWidth == 0 || textureHeight == 0) {
+      // Information about the resolution needs to be provided by a call to setTextureSize() before
+      // frames are produced.
+      Logging.w(TAG, "Texture size has not been set.");
+      return;
+    }
+    isTextureInUse = true;
+    hasPendingTexture = false;
+
+    updateTexImage();
+
+    final float[] transformMatrix = new float[16];
+    surfaceTexture.getTransformMatrix(transformMatrix);
+    long timestampNs = surfaceTexture.getTimestamp();
+    if (timestampAligner != null) {
+      timestampNs = timestampAligner.translateTimestamp(timestampNs);
+    }
+    final VideoFrame.TextureBuffer buffer =
+        new TextureBufferImpl(textureWidth, textureHeight, TextureBuffer.Type.OES, oesTextureId,
+            RendererCommon.convertMatrixToAndroidGraphicsMatrix(transformMatrix), handler,
+            yuvConverter, textureRefCountMonitor);
+    if (frameRefMonitor != null) {
+      frameRefMonitor.onNewBuffer(buffer);
+    }
+    final VideoFrame frame = new VideoFrame(buffer, frameRotation, timestampNs);
+    listener.onFrame(frame);
+    frame.release();
+  }
+
+  private void release() {
+    if (handler.getLooper().getThread() != Thread.currentThread()) {
+      throw new IllegalStateException("Wrong thread.");
+    }
+    if (isTextureInUse || !isQuitting) {
+      throw new IllegalStateException("Unexpected release.");
+    }
+    yuvConverter.release();
+    GLES20.glDeleteTextures(1, new int[] {oesTextureId}, 0);
+    surfaceTexture.release();
+    eglBase.release();
+    handler.getLooper().quit();
+    if (timestampAligner != null) {
+      timestampAligner.dispose();
+    }
+  }
+}
diff --git a/third_party/libwebrtc/sdk/android/api/org/webrtc/SurfaceViewRenderer.java b/third_party/libwebrtc/sdk/android/api/org/webrtc/SurfaceViewRenderer.java
new file mode 100644
index 0000000000..6c9140abbd
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/api/org/webrtc/SurfaceViewRenderer.java
@@ -0,0 +1,300 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+import android.content.Context;
+import android.content.res.Resources.NotFoundException;
+import android.graphics.Point;
+import android.os.Looper;
+import android.util.AttributeSet;
+import android.view.SurfaceHolder;
+import android.view.SurfaceView;
+
+/**
+ * Display the video stream on a SurfaceView.
+ */
+public class SurfaceViewRenderer extends SurfaceView
+    implements SurfaceHolder.Callback, VideoSink, RendererCommon.RendererEvents {
+  private static final String TAG = "SurfaceViewRenderer";
+
+  // Cached resource name.
+  private final String resourceName;
+  private final RendererCommon.VideoLayoutMeasure videoLayoutMeasure =
+      new RendererCommon.VideoLayoutMeasure();
+  private final SurfaceEglRenderer eglRenderer;
+
+  // Callback for reporting renderer events. Read-only after initialization so no lock required.
+  private RendererCommon.RendererEvents rendererEvents;
+
+  // Accessed only on the main thread.
+  private int rotatedFrameWidth;
+  private int rotatedFrameHeight;
+  private boolean enableFixedSize;
+  private int surfaceWidth;
+  private int surfaceHeight;
+
+  /**
+   * Standard View constructor. In order to render something, you must first call init().
+   */
+  public SurfaceViewRenderer(Context context) {
+    super(context);
+    this.resourceName = getResourceName();
+    eglRenderer = new SurfaceEglRenderer(resourceName);
+    getHolder().addCallback(this);
+    getHolder().addCallback(eglRenderer);
+  }
+
+  /**
+   * Standard View constructor. In order to render something, you must first call init().
+   */
+  public SurfaceViewRenderer(Context context, AttributeSet attrs) {
+    super(context, attrs);
+    this.resourceName = getResourceName();
+    eglRenderer = new SurfaceEglRenderer(resourceName);
+    getHolder().addCallback(this);
+    getHolder().addCallback(eglRenderer);
+  }
+
+  /**
+   * Initialize this class, sharing resources with `sharedContext`. It is allowed to call init() to
+   * reinitialize the renderer after a previous init()/release() cycle.
+   */
+  public void init(EglBase.Context sharedContext, RendererCommon.RendererEvents rendererEvents) {
+    init(sharedContext, rendererEvents, EglBase.CONFIG_PLAIN, new GlRectDrawer());
+  }
+
+  /**
+   * Initialize this class, sharing resources with `sharedContext`. The custom `drawer` will be used
+   * for drawing frames on the EGLSurface. This class is responsible for calling release() on
+   * `drawer`. It is allowed to call init() to reinitialize the renderer after a previous
+   * init()/release() cycle.
+   */
+  public void init(final EglBase.Context sharedContext,
+      RendererCommon.RendererEvents rendererEvents, final int[] configAttributes,
+      RendererCommon.GlDrawer drawer) {
+    ThreadUtils.checkIsOnMainThread();
+    this.rendererEvents = rendererEvents;
+    rotatedFrameWidth = 0;
+    rotatedFrameHeight = 0;
+    eglRenderer.init(sharedContext, this /* rendererEvents */, configAttributes, drawer);
+  }
+
+  /**
+   * Block until any pending frame is returned and all GL resources released, even if an interrupt
+   * occurs. If an interrupt occurs during release(), the interrupt flag will be set. This function
+   * should be called before the Activity is destroyed and the EGLContext is still valid. If you
+   * don't call this function, the GL resources might leak.
+   */
+  public void release() {
+    eglRenderer.release();
+  }
+
+  /**
+   * Register a callback to be invoked when a new video frame has been received.
+   *
+   * @param listener The callback to be invoked. The callback will be invoked on the render thread.
+   *                 It should be lightweight and must not call removeFrameListener.
+   * @param scale    The scale of the Bitmap passed to the callback, or 0 if no Bitmap is
+   *                 required.
+   * @param drawer   Custom drawer to use for this frame listener.
+   */
+  public void addFrameListener(
+      EglRenderer.FrameListener listener, float scale, RendererCommon.GlDrawer drawerParam) {
+    eglRenderer.addFrameListener(listener, scale, drawerParam);
+  }
+
+  /**
+   * Register a callback to be invoked when a new video frame has been received. This version uses
+   * the drawer of the EglRenderer that was passed in init.
+   *
+   * @param listener The callback to be invoked. The callback will be invoked on the render thread.
+   *                 It should be lightweight and must not call removeFrameListener.
+   * @param scale    The scale of the Bitmap passed to the callback, or 0 if no Bitmap is
+   *                 required.
+   */
+  public void addFrameListener(EglRenderer.FrameListener listener, float scale) {
+    eglRenderer.addFrameListener(listener, scale);
+  }
+
+  public void removeFrameListener(EglRenderer.FrameListener listener) {
+    eglRenderer.removeFrameListener(listener);
+  }
+
+  /**
+   * Enables fixed size for the surface. This provides better performance but might be buggy on some
+   * devices. By default this is turned off.
+   */
+  public void setEnableHardwareScaler(boolean enabled) {
+    ThreadUtils.checkIsOnMainThread();
+    enableFixedSize = enabled;
+    updateSurfaceSize();
+  }
+
+  /**
+   * Set if the video stream should be mirrored or not.
+   */
+  public void setMirror(final boolean mirror) {
+    eglRenderer.setMirror(mirror);
+  }
+
+  /**
+   * Set how the video will fill the allowed layout area.
+   */
+  public void setScalingType(RendererCommon.ScalingType scalingType) {
+    ThreadUtils.checkIsOnMainThread();
+    videoLayoutMeasure.setScalingType(scalingType);
+    requestLayout();
+  }
+
+  public void setScalingType(RendererCommon.ScalingType scalingTypeMatchOrientation,
+      RendererCommon.ScalingType scalingTypeMismatchOrientation) {
+    ThreadUtils.checkIsOnMainThread();
+    videoLayoutMeasure.setScalingType(scalingTypeMatchOrientation, scalingTypeMismatchOrientation);
+    requestLayout();
+  }
+
+  /**
+   * Limit render framerate.
+   *
+   * @param fps Limit render framerate to this value, or use Float.POSITIVE_INFINITY to disable fps
+   *            reduction.
+   */
+  public void setFpsReduction(float fps) {
+    eglRenderer.setFpsReduction(fps);
+  }
+
+  public void disableFpsReduction() {
+    eglRenderer.disableFpsReduction();
+  }
+
+  public void pauseVideo() {
+    eglRenderer.pauseVideo();
+  }
+
+  // VideoSink interface.
+  @Override
+  public void onFrame(VideoFrame frame) {
+    eglRenderer.onFrame(frame);
+  }
+
+  // View layout interface.
+  @Override
+  protected void onMeasure(int widthSpec, int heightSpec) {
+    ThreadUtils.checkIsOnMainThread();
+    Point size =
+        videoLayoutMeasure.measure(widthSpec, heightSpec, rotatedFrameWidth, rotatedFrameHeight);
+    setMeasuredDimension(size.x, size.y);
+    logD("onMeasure(). New size: " + size.x + "x" + size.y);
+  }
+
+  @Override
+  protected void onLayout(boolean changed, int left, int top, int right, int bottom) {
+    ThreadUtils.checkIsOnMainThread();
+    eglRenderer.setLayoutAspectRatio((right - left) / (float) (bottom - top));
+    updateSurfaceSize();
+  }
+
+  private void updateSurfaceSize() {
+    ThreadUtils.checkIsOnMainThread();
+    if (enableFixedSize && rotatedFrameWidth != 0 && rotatedFrameHeight != 0 && getWidth() != 0
+        && getHeight() != 0) {
+      final float layoutAspectRatio = getWidth() / (float) getHeight();
+      final float frameAspectRatio = rotatedFrameWidth / (float) rotatedFrameHeight;
+      final int drawnFrameWidth;
+      final int drawnFrameHeight;
+      if (frameAspectRatio > layoutAspectRatio) {
+        drawnFrameWidth = (int) (rotatedFrameHeight * layoutAspectRatio);
+        drawnFrameHeight = rotatedFrameHeight;
+      } else {
+        drawnFrameWidth = rotatedFrameWidth;
+        drawnFrameHeight = (int) (rotatedFrameWidth / layoutAspectRatio);
+      }
+      // Aspect ratio of the drawn frame and the view is the same.
+      final int width = Math.min(getWidth(), drawnFrameWidth);
+      final int height = Math.min(getHeight(), drawnFrameHeight);
+      logD("updateSurfaceSize. Layout size: " + getWidth() + "x" + getHeight() + ", frame size: "
+          + rotatedFrameWidth + "x" + rotatedFrameHeight + ", requested surface size: " + width
+          + "x" + height + ", old surface size: " + surfaceWidth + "x" + surfaceHeight);
+      if (width != surfaceWidth || height != surfaceHeight) {
+        surfaceWidth = width;
+        surfaceHeight = height;
+        getHolder().setFixedSize(width, height);
+      }
+    } else {
+      surfaceWidth = surfaceHeight = 0;
+      getHolder().setSizeFromLayout();
+    }
+  }
+
+  // SurfaceHolder.Callback interface.
+  @Override
+  public void surfaceCreated(final SurfaceHolder holder) {
+    ThreadUtils.checkIsOnMainThread();
+    surfaceWidth = surfaceHeight = 0;
+    updateSurfaceSize();
+  }
+
+  @Override
+  public void surfaceDestroyed(SurfaceHolder holder) {}
+
+  @Override
+  public void surfaceChanged(SurfaceHolder holder, int format, int width, int height) {}
+
+  private String getResourceName() {
+    try {
+      return getResources().getResourceEntryName(getId());
+    } catch (NotFoundException e) {
+      return "";
+    }
+  }
+
+  /**
+   * Post a task to clear the SurfaceView to a transparent uniform color.
+   */
+  public void clearImage() {
+    eglRenderer.clearImage();
+  }
+
+  @Override
+  public void onFirstFrameRendered() {
+    if (rendererEvents != null) {
+      rendererEvents.onFirstFrameRendered();
+    }
+  }
+
+  @Override
+  public void onFrameResolutionChanged(int videoWidth, int videoHeight, int rotation) {
+    if (rendererEvents != null) {
+      rendererEvents.onFrameResolutionChanged(videoWidth, videoHeight, rotation);
+    }
+    int rotatedWidth = rotation == 0 || rotation == 180 ? videoWidth : videoHeight;
+    int rotatedHeight = rotation == 0 || rotation == 180 ? videoHeight : videoWidth;
+    // run immediately if possible for ui thread tests
+    postOrRun(() -> {
+      rotatedFrameWidth = rotatedWidth;
+      rotatedFrameHeight = rotatedHeight;
+      updateSurfaceSize();
+      requestLayout();
+    });
+  }
+
+  private void postOrRun(Runnable r) {
+    if (Thread.currentThread() == Looper.getMainLooper().getThread()) {
+      r.run();
+    } else {
+      post(r);
+    }
+  }
+
+  private void logD(String string) {
+    Logging.d(TAG, resourceName + ": " + string);
+  }
+}
diff --git a/third_party/libwebrtc/sdk/android/api/org/webrtc/TextureBufferImpl.java b/third_party/libwebrtc/sdk/android/api/org/webrtc/TextureBufferImpl.java
new file mode 100644
index 0000000000..8e0e40ef70
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/api/org/webrtc/TextureBufferImpl.java
@@ -0,0 +1,203 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+import android.graphics.Matrix;
+import android.os.Handler;
+import androidx.annotation.Nullable;
+
+/**
+ * Android texture buffer that glues together the necessary information together with a generic
+ * release callback. ToI420() is implemented by providing a Handler and a YuvConverter.
+ */
+public class TextureBufferImpl implements VideoFrame.TextureBuffer {
+  interface RefCountMonitor {
+    void onRetain(TextureBufferImpl textureBuffer);
+    void onRelease(TextureBufferImpl textureBuffer);
+    void onDestroy(TextureBufferImpl textureBuffer);
+  }
+
+  // This is the full resolution the texture has in memory after applying the transformation matrix
+  // that might include cropping. This resolution is useful to know when sampling the texture to
+  // avoid downscaling artifacts.
+  private final int unscaledWidth;
+  private final int unscaledHeight;
+  // This is the resolution that has been applied after cropAndScale().
+  private final int width;
+  private final int height;
+  private final Type type;
+  private final int id;
+  private final Matrix transformMatrix;
+  private final Handler toI420Handler;
+  private final YuvConverter yuvConverter;
+  private final RefCountDelegate refCountDelegate;
+  private final RefCountMonitor refCountMonitor;
+
+  public TextureBufferImpl(int width, int height, Type type, int id, Matrix transformMatrix,
+      Handler toI420Handler, YuvConverter yuvConverter, @Nullable Runnable releaseCallback) {
+    this(width, height, width, height, type, id, transformMatrix, toI420Handler, yuvConverter,
+        new RefCountMonitor() {
+          @Override
+          public void onRetain(TextureBufferImpl textureBuffer) {}
+
+          @Override
+          public void onRelease(TextureBufferImpl textureBuffer) {}
+
+          @Override
+          public void onDestroy(TextureBufferImpl textureBuffer) {
+            if (releaseCallback != null) {
+              releaseCallback.run();
+            }
+          }
+        });
+  }
+
+  TextureBufferImpl(int width, int height, Type type, int id, Matrix transformMatrix,
+      Handler toI420Handler, YuvConverter yuvConverter, RefCountMonitor refCountMonitor) {
+    this(width, height, width, height, type, id, transformMatrix, toI420Handler, yuvConverter,
+        refCountMonitor);
+  }
+
+  private TextureBufferImpl(int unscaledWidth, int unscaledHeight, int width, int height, Type type,
+      int id, Matrix transformMatrix, Handler toI420Handler, YuvConverter yuvConverter,
+      RefCountMonitor refCountMonitor) {
+    this.unscaledWidth = unscaledWidth;
+    this.unscaledHeight = unscaledHeight;
+    this.width = width;
+    this.height = height;
+    this.type = type;
+    this.id = id;
+    this.transformMatrix = transformMatrix;
+    this.toI420Handler = toI420Handler;
+    this.yuvConverter = yuvConverter;
+    this.refCountDelegate = new RefCountDelegate(() -> refCountMonitor.onDestroy(this));
+    this.refCountMonitor = refCountMonitor;
+  }
+
+  @Override
+  public VideoFrame.TextureBuffer.Type getType() {
+    return type;
+  }
+
+  @Override
+  public int getTextureId() {
+    return id;
+  }
+
+  @Override
+  public Matrix getTransformMatrix() {
+    return transformMatrix;
+  }
+
+  @Override
+  public int getWidth() {
+    return width;
+  }
+
+  @Override
+  public int getHeight() {
+    return height;
+  }
+
+  @Override
+  public VideoFrame.I420Buffer toI420() {
+    return ThreadUtils.invokeAtFrontUninterruptibly(
+        toI420Handler, () -> yuvConverter.convert(this));
+  }
+
+  @Override
+  public void retain() {
+    refCountMonitor.onRetain(this);
+    refCountDelegate.retain();
+  }
+
+  @Override
+  public void release() {
+    refCountMonitor.onRelease(this);
+    refCountDelegate.release();
+  }
+
+  @Override
+  public VideoFrame.Buffer cropAndScale(
+      int cropX, int cropY, int cropWidth, int cropHeight, int scaleWidth, int scaleHeight) {
+    final Matrix cropAndScaleMatrix = new Matrix();
+    // In WebRTC, Y=0 is the top row, while in OpenGL Y=0 is the bottom row. This means that the Y
+    // direction is effectively reversed.
+    final int cropYFromBottom = height - (cropY + cropHeight);
+    cropAndScaleMatrix.preTranslate(cropX / (float) width, cropYFromBottom / (float) height);
+    cropAndScaleMatrix.preScale(cropWidth / (float) width, cropHeight / (float) height);
+
+    return applyTransformMatrix(cropAndScaleMatrix,
+        Math.round(unscaledWidth * cropWidth / (float) width),
+        Math.round(unscaledHeight * cropHeight / (float) height), scaleWidth, scaleHeight);
+  }
+
+  /**
+   * Returns the width of the texture in memory. This should only be used for downscaling, and you
+   * should still respect the width from getWidth().
+   */
+  public int getUnscaledWidth() {
+    return unscaledWidth;
+  }
+
+  /**
+   * Returns the height of the texture in memory. This should only be used for downscaling, and you
+   * should still respect the height from getHeight().
+   */
+  public int getUnscaledHeight() {
+    return unscaledHeight;
+  }
+
+  public Handler getToI420Handler() {
+    return toI420Handler;
+  }
+
+  public YuvConverter getYuvConverter() {
+    return yuvConverter;
+  }
+
+  /**
+   * Create a new TextureBufferImpl with an applied transform matrix and a new size. The
+   * existing buffer is unchanged. The given transform matrix is applied first when texture
+   * coordinates are still in the unmodified [0, 1] range.
+   */
+  @Override
+  public TextureBufferImpl applyTransformMatrix(
+      Matrix transformMatrix, int newWidth, int newHeight) {
+    return applyTransformMatrix(transformMatrix, /* unscaledWidth= */ newWidth,
+        /* unscaledHeight= */ newHeight, /* scaledWidth= */ newWidth,
+        /* scaledHeight= */ newHeight);
+  }
+
+  private TextureBufferImpl applyTransformMatrix(Matrix transformMatrix, int unscaledWidth,
+      int unscaledHeight, int scaledWidth, int scaledHeight) {
+    final Matrix newMatrix = new Matrix(this.transformMatrix);
+    newMatrix.preConcat(transformMatrix);
+    retain();
+    return new TextureBufferImpl(unscaledWidth, unscaledHeight, scaledWidth, scaledHeight, type, id,
+        newMatrix, toI420Handler, yuvConverter, new RefCountMonitor() {
+          @Override
+          public void onRetain(TextureBufferImpl textureBuffer) {
+            refCountMonitor.onRetain(TextureBufferImpl.this);
+          }
+
+          @Override
+          public void onRelease(TextureBufferImpl textureBuffer) {
+            refCountMonitor.onRelease(TextureBufferImpl.this);
+          }
+
+          @Override
+          public void onDestroy(TextureBufferImpl textureBuffer) {
+            release();
+          }
+        });
+  }
+}
diff --git a/third_party/libwebrtc/sdk/android/api/org/webrtc/TimestampAligner.java b/third_party/libwebrtc/sdk/android/api/org/webrtc/TimestampAligner.java
new file mode 100644
index 0000000000..d96c939595
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/api/org/webrtc/TimestampAligner.java
@@ -0,0 +1,59 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+/**
+ * The TimestampAligner class helps translating camera timestamps into the same timescale as is
+ * used by rtc::TimeNanos(). Some cameras have built in timestamping which is more accurate than
+ * reading the system clock, but using a different epoch and unknown clock drift. Frame timestamps
+ * in webrtc should use rtc::TimeNanos (system monotonic time), and this class provides a filter
+ * which lets us use the rtc::TimeNanos timescale, and at the same time take advantage of higher
+ * accuracy of the camera clock. This class is a wrapper on top of rtc::TimestampAligner.
+ */
+public class TimestampAligner {
+  /**
+   * Wrapper around rtc::TimeNanos(). This is normally same as System.nanoTime(), but call this
+   * function to be safe.
+   */
+  public static long getRtcTimeNanos() {
+    return nativeRtcTimeNanos();
+  }
+
+  private volatile long nativeTimestampAligner = nativeCreateTimestampAligner();
+
+  /**
+   * Translates camera timestamps to the same timescale as is used by rtc::TimeNanos().
+   * `cameraTimeNs` is assumed to be accurate, but with an unknown epoch and clock drift. Returns
+   * the translated timestamp.
+   */
+  public long translateTimestamp(long cameraTimeNs) {
+    checkNativeAlignerExists();
+    return nativeTranslateTimestamp(nativeTimestampAligner, cameraTimeNs);
+  }
+
+  /** Dispose native timestamp aligner. */
+  public void dispose() {
+    checkNativeAlignerExists();
+    nativeReleaseTimestampAligner(nativeTimestampAligner);
+    nativeTimestampAligner = 0;
+  }
+
+  private void checkNativeAlignerExists() {
+    if (nativeTimestampAligner == 0) {
+      throw new IllegalStateException("TimestampAligner has been disposed.");
+    }
+  }
+
+  private static native long nativeRtcTimeNanos();
+  private static native long nativeCreateTimestampAligner();
+  private static native void nativeReleaseTimestampAligner(long timestampAligner);
+  private static native long nativeTranslateTimestamp(long timestampAligner, long cameraTimeNs);
+}
diff --git a/third_party/libwebrtc/sdk/android/api/org/webrtc/TurnCustomizer.java b/third_party/libwebrtc/sdk/android/api/org/webrtc/TurnCustomizer.java
new file mode 100644
index 0000000000..41bedb7dcb
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/api/org/webrtc/TurnCustomizer.java
@@ -0,0 +1,41 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+/** Java wrapper for a C++ TurnCustomizer. */
+public class TurnCustomizer {
+  private long nativeTurnCustomizer;
+
+  public TurnCustomizer(long nativeTurnCustomizer) {
+    this.nativeTurnCustomizer = nativeTurnCustomizer;
+  }
+
+  public void dispose() {
+    checkTurnCustomizerExists();
+    nativeFreeTurnCustomizer(nativeTurnCustomizer);
+    nativeTurnCustomizer = 0;
+  }
+
+  private static native void nativeFreeTurnCustomizer(long turnCustomizer);
+
+  /** Return a pointer to webrtc::TurnCustomizer. */
+  @CalledByNative
+  long getNativeTurnCustomizer() {
+    checkTurnCustomizerExists();
+    return nativeTurnCustomizer;
+  }
+
+  private void checkTurnCustomizerExists() {
+    if (nativeTurnCustomizer == 0) {
+      throw new IllegalStateException("TurnCustomizer has been disposed.");
+    }
+  }
+}
diff --git a/third_party/libwebrtc/sdk/android/api/org/webrtc/VideoCapturer.java b/third_party/libwebrtc/sdk/android/api/org/webrtc/VideoCapturer.java
new file mode 100644
index 0000000000..67eb7ab086
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/api/org/webrtc/VideoCapturer.java
@@ -0,0 +1,53 @@
+/*
+ *  Copyright 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+import android.content.Context;
+
+// Base interface for all VideoCapturers to implement.
+public interface VideoCapturer {
+  /**
+   * This function is used to initialize the camera thread, the android application context, and the
+   * capture observer. It will be called only once and before any startCapture() request. The
+   * camera thread is guaranteed to be valid until dispose() is called. If the VideoCapturer wants
+   * to deliver texture frames, it should do this by rendering on the SurfaceTexture in
+   * {@code surfaceTextureHelper}, register itself as a listener, and forward the frames to
+   * CapturerObserver.onFrameCaptured(). The caller still has ownership of {@code
+   * surfaceTextureHelper} and is responsible for making sure surfaceTextureHelper.dispose() is
+   * called. This also means that the caller can reuse the SurfaceTextureHelper to initialize a new
+   * VideoCapturer once the previous VideoCapturer has been disposed.
+   */
+  void initialize(SurfaceTextureHelper surfaceTextureHelper, Context applicationContext,
+      CapturerObserver capturerObserver);
+
+  /**
+   * Start capturing frames in a format that is as close as possible to {@code width x height} and
+   * {@code framerate}.
+   */
+  void startCapture(int width, int height, int framerate);
+
+  /**
+   * Stop capturing. This function should block until capture is actually stopped.
+   */
+  void stopCapture() throws InterruptedException;
+
+  void changeCaptureFormat(int width, int height, int framerate);
+
+  /**
+   * Perform any final cleanup here. No more capturing will be done after this call.
+   */
+  void dispose();
+
+  /**
+   * @return true if-and-only-if this is a screen capturer.
+   */
+  boolean isScreencast();
+}
diff --git a/third_party/libwebrtc/sdk/android/api/org/webrtc/VideoCodecInfo.java b/third_party/libwebrtc/sdk/android/api/org/webrtc/VideoCodecInfo.java
new file mode 100644
index 0000000000..363be347b5
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/api/org/webrtc/VideoCodecInfo.java
@@ -0,0 +1,86 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+import androidx.annotation.Nullable;
+import java.util.Arrays;
+import java.util.Locale;
+import java.util.Map;
+
+/**
+ * Represent a video codec as encoded in SDP.
+ */
+public class VideoCodecInfo {
+  // Keys for H264 VideoCodecInfo properties.
+  public static final String H264_FMTP_PROFILE_LEVEL_ID = "profile-level-id";
+  public static final String H264_FMTP_LEVEL_ASYMMETRY_ALLOWED = "level-asymmetry-allowed";
+  public static final String H264_FMTP_PACKETIZATION_MODE = "packetization-mode";
+
+  public static final String H264_PROFILE_CONSTRAINED_BASELINE = "42e0";
+  public static final String H264_PROFILE_CONSTRAINED_HIGH = "640c";
+  public static final String H264_LEVEL_3_1 = "1f"; // 31 in hex.
+  public static final String H264_CONSTRAINED_HIGH_3_1 =
+      H264_PROFILE_CONSTRAINED_HIGH + H264_LEVEL_3_1;
+  public static final String H264_CONSTRAINED_BASELINE_3_1 =
+      H264_PROFILE_CONSTRAINED_BASELINE + H264_LEVEL_3_1;
+
+  public final String name;
+  public final Map<String, String> params;
+  @Deprecated public final int payload;
+
+  @CalledByNative
+  public VideoCodecInfo(String name, Map<String, String> params) {
+    this.payload = 0;
+    this.name = name;
+    this.params = params;
+  }
+
+  @Deprecated
+  public VideoCodecInfo(int payload, String name, Map<String, String> params) {
+    this.payload = payload;
+    this.name = name;
+    this.params = params;
+  }
+
+  @Override
+  public boolean equals(@Nullable Object obj) {
+    if (obj == null)
+      return false;
+    if (obj == this)
+      return true;
+    if (!(obj instanceof VideoCodecInfo))
+      return false;
+
+    VideoCodecInfo otherInfo = (VideoCodecInfo) obj;
+    return name.equalsIgnoreCase(otherInfo.name) && params.equals(otherInfo.params);
+  }
+
+  @Override
+  public int hashCode() {
+    Object[] values = {name.toUpperCase(Locale.ROOT), params};
+    return Arrays.hashCode(values);
+  }
+
+  @Override
+  public String toString() {
+    return "VideoCodec{" + name + " " + params + "}";
+  }
+
+  @CalledByNative
+  String getName() {
+    return name;
+  }
+
+  @CalledByNative
+  Map<String, String> getParams() {
+    return params;
+  }
+}
diff --git a/third_party/libwebrtc/sdk/android/api/org/webrtc/VideoCodecStatus.java b/third_party/libwebrtc/sdk/android/api/org/webrtc/VideoCodecStatus.java
new file mode 100644
index 0000000000..670d255880
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/api/org/webrtc/VideoCodecStatus.java
@@ -0,0 +1,42 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+/**
+ * Status codes reported by video encoding/decoding components. This should be kept in sync with
+ * video_error_codes.h.
+ */
+public enum VideoCodecStatus {
+  TARGET_BITRATE_OVERSHOOT(5),
+  REQUEST_SLI(2),
+  NO_OUTPUT(1),
+  OK(0),
+  ERROR(-1),
+  LEVEL_EXCEEDED(-2),
+  MEMORY(-3),
+  ERR_PARAMETER(-4),
+  ERR_SIZE(-5),
+  TIMEOUT(-6),
+  UNINITIALIZED(-7),
+  ERR_REQUEST_SLI(-12),
+  FALLBACK_SOFTWARE(-13);
+
+  private final int number;
+
+  private VideoCodecStatus(int number) {
+    this.number = number;
+  }
+
+  @CalledByNative
+  public int getNumber() {
+    return number;
+  }
+}
diff --git a/third_party/libwebrtc/sdk/android/api/org/webrtc/VideoDecoder.java b/third_party/libwebrtc/sdk/android/api/org/webrtc/VideoDecoder.java
new file mode 100644
index 0000000000..a80fa4fef2
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/api/org/webrtc/VideoDecoder.java
@@ -0,0 +1,94 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+/**
+ * Interface for a video decoder that can be used in WebRTC. All calls to the class will be made on
+ * a single decoding thread.
+ */
+public interface VideoDecoder {
+  /** Settings passed to the decoder by WebRTC. */
+  public class Settings {
+    public final int numberOfCores;
+    public final int width;
+    public final int height;
+
+    @CalledByNative("Settings")
+    public Settings(int numberOfCores, int width, int height) {
+      this.numberOfCores = numberOfCores;
+      this.width = width;
+      this.height = height;
+    }
+  }
+
+  /** Additional info for decoding. */
+  public class DecodeInfo {
+    public final boolean isMissingFrames;
+    public final long renderTimeMs;
+
+    public DecodeInfo(boolean isMissingFrames, long renderTimeMs) {
+      this.isMissingFrames = isMissingFrames;
+      this.renderTimeMs = renderTimeMs;
+    }
+  }
+
+  public interface Callback {
+    /**
+     * Call to return a decoded frame. Can be called on any thread.
+     *
+     * @param frame Decoded frame
+     * @param decodeTimeMs Time it took to decode the frame in milliseconds or null if not available
+     * @param qp QP value of the decoded frame or null if not available
+     */
+    void onDecodedFrame(VideoFrame frame, Integer decodeTimeMs, Integer qp);
+  }
+
+  /**
+   * The decoder implementation backing this interface is either 1) a Java
+   * decoder (e.g., an Android platform decoder), or alternatively 2) a native
+   * decoder (e.g., a software decoder or a C++ decoder adapter).
+   *
+   * For case 1), createNativeVideoDecoder() should return zero.
+   * In this case, we expect the native library to call the decoder through
+   * JNI using the Java interface declared below.
+   *
+   * For case 2), createNativeVideoDecoder() should return a non-zero value.
+   * In this case, we expect the native library to treat the returned value as
+   * a raw pointer of type webrtc::VideoDecoder* (ownership is transferred to
+   * the caller). The native library should then directly call the
+   * webrtc::VideoDecoder interface without going through JNI. All calls to
+   * the Java interface methods declared below should thus throw an
+   * UnsupportedOperationException.
+   */
+  @CalledByNative
+  default long createNativeVideoDecoder() {
+    return 0;
+  }
+
+  /**
+   * Initializes the decoding process with specified settings. Will be called on the decoding thread
+   * before any decode calls.
+   */
+  @CalledByNative VideoCodecStatus initDecode(Settings settings, Callback decodeCallback);
+  /**
+   * Called when the decoder is no longer needed. Any more calls to decode will not be made.
+   */
+  @CalledByNative VideoCodecStatus release();
+  /**
+   * Request the decoder to decode a frame.
+   */
+  @CalledByNative VideoCodecStatus decode(EncodedImage frame, DecodeInfo info);
+  /**
+   * Should return a descriptive name for the implementation. Gets called once and cached. May be
+   * called from arbitrary thread.
+   */
+  @CalledByNative String getImplementationName();
+}
diff --git a/third_party/libwebrtc/sdk/android/api/org/webrtc/VideoDecoderFactory.java b/third_party/libwebrtc/sdk/android/api/org/webrtc/VideoDecoderFactory.java
new file mode 100644
index 0000000000..8b25516e99
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/api/org/webrtc/VideoDecoderFactory.java
@@ -0,0 +1,30 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+import androidx.annotation.Nullable;
+
+/** Factory for creating VideoDecoders. */
+public interface VideoDecoderFactory {
+  /**
+   * Creates a VideoDecoder for the given codec. Supports the same codecs supported by
+   * VideoEncoderFactory.
+   */
+  @Nullable @CalledByNative VideoDecoder createDecoder(VideoCodecInfo info);
+
+  /**
+   * Enumerates the list of supported video codecs.
+   */
+  @CalledByNative
+  default VideoCodecInfo[] getSupportedCodecs() {
+    return new VideoCodecInfo[0];
+  }
+}
diff --git a/third_party/libwebrtc/sdk/android/api/org/webrtc/VideoDecoderFallback.java b/third_party/libwebrtc/sdk/android/api/org/webrtc/VideoDecoderFallback.java
new file mode 100644
index 0000000000..ddfa3ecd40
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/api/org/webrtc/VideoDecoderFallback.java
@@ -0,0 +1,31 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+/**
+ * A combined video decoder that falls back on a secondary decoder if the primary decoder fails.
+ */
+public class VideoDecoderFallback extends WrappedNativeVideoDecoder {
+  private final VideoDecoder fallback;
+  private final VideoDecoder primary;
+
+  public VideoDecoderFallback(VideoDecoder fallback, VideoDecoder primary) {
+    this.fallback = fallback;
+    this.primary = primary;
+  }
+
+  @Override
+  public long createNativeVideoDecoder() {
+    return nativeCreateDecoder(fallback, primary);
+  }
+
+  private static native long nativeCreateDecoder(VideoDecoder fallback, VideoDecoder primary);
+}
diff --git a/third_party/libwebrtc/sdk/android/api/org/webrtc/VideoEncoder.java b/third_party/libwebrtc/sdk/android/api/org/webrtc/VideoEncoder.java
new file mode 100644
index 0000000000..0d8cf830ae
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/api/org/webrtc/VideoEncoder.java
@@ -0,0 +1,385 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+import androidx.annotation.Nullable;
+import org.webrtc.EncodedImage;
+
+/**
+ * Interface for a video encoder that can be used with WebRTC. All calls will be made on the
+ * encoding thread. The encoder may be constructed on a different thread and changing thread after
+ * calling release is allowed.
+ */
+public interface VideoEncoder {
+  /** Settings passed to the encoder by WebRTC. */
+  public class Settings {
+    public final int numberOfCores;
+    public final int width;
+    public final int height;
+    public final int startBitrate; // Kilobits per second.
+    public final int maxFramerate;
+    public final int numberOfSimulcastStreams;
+    public final boolean automaticResizeOn;
+    public final Capabilities capabilities;
+
+    // TODO(bugs.webrtc.org/10720): Remove.
+    @Deprecated
+    public Settings(int numberOfCores, int width, int height, int startBitrate, int maxFramerate,
+        int numberOfSimulcastStreams, boolean automaticResizeOn) {
+      this(numberOfCores, width, height, startBitrate, maxFramerate, numberOfSimulcastStreams,
+          automaticResizeOn, new VideoEncoder.Capabilities(false /* lossNotification */));
+    }
+
+    @CalledByNative("Settings")
+    public Settings(int numberOfCores, int width, int height, int startBitrate, int maxFramerate,
+        int numberOfSimulcastStreams, boolean automaticResizeOn, Capabilities capabilities) {
+      this.numberOfCores = numberOfCores;
+      this.width = width;
+      this.height = height;
+      this.startBitrate = startBitrate;
+      this.maxFramerate = maxFramerate;
+      this.numberOfSimulcastStreams = numberOfSimulcastStreams;
+      this.automaticResizeOn = automaticResizeOn;
+      this.capabilities = capabilities;
+    }
+  }
+
+  /** Capabilities (loss notification, etc.) passed to the encoder by WebRTC. */
+  public class Capabilities {
+    /**
+     * The remote side has support for the loss notification RTCP feedback message format, and will
+     * be sending these feedback messages if necessary.
+     */
+    public final boolean lossNotification;
+
+    @CalledByNative("Capabilities")
+    public Capabilities(boolean lossNotification) {
+      this.lossNotification = lossNotification;
+    }
+  }
+
+  /** Additional info for encoding. */
+  public class EncodeInfo {
+    public final EncodedImage.FrameType[] frameTypes;
+
+    @CalledByNative("EncodeInfo")
+    public EncodeInfo(EncodedImage.FrameType[] frameTypes) {
+      this.frameTypes = frameTypes;
+    }
+  }
+
+  // TODO(sakal): Add values to these classes as necessary.
+  /** Codec specific information about the encoded frame. */
+  public class CodecSpecificInfo {}
+
+  public class CodecSpecificInfoVP8 extends CodecSpecificInfo {}
+
+  public class CodecSpecificInfoVP9 extends CodecSpecificInfo {}
+
+  public class CodecSpecificInfoH264 extends CodecSpecificInfo {}
+
+  public class CodecSpecificInfoAV1 extends CodecSpecificInfo {}
+
+  /**
+   * Represents bitrate allocated for an encoder to produce frames. Bitrate can be divided between
+   * spatial and temporal layers.
+   */
+  public class BitrateAllocation {
+    // First index is the spatial layer and second the temporal layer.
+    public final int[][] bitratesBbs;
+
+    /**
+     * Initializes the allocation with a two dimensional array of bitrates. The first index of the
+     * array is the spatial layer and the second index in the temporal layer.
+     */
+    @CalledByNative("BitrateAllocation")
+    public BitrateAllocation(int[][] bitratesBbs) {
+      this.bitratesBbs = bitratesBbs;
+    }
+
+    /**
+     * Gets the total bitrate allocated for all layers.
+     */
+    public int getSum() {
+      int sum = 0;
+      for (int[] spatialLayer : bitratesBbs) {
+        for (int bitrate : spatialLayer) {
+          sum += bitrate;
+        }
+      }
+      return sum;
+    }
+  }
+
+  /** Settings for WebRTC quality based scaling. */
+  public class ScalingSettings {
+    public final boolean on;
+    @Nullable public final Integer low;
+    @Nullable public final Integer high;
+
+    /**
+     * Settings to disable quality based scaling.
+     */
+    public static final ScalingSettings OFF = new ScalingSettings();
+
+    /**
+     * Creates settings to enable quality based scaling.
+     *
+     * @param low Average QP at which to scale up the resolution.
+     * @param high Average QP at which to scale down the resolution.
+     */
+    public ScalingSettings(int low, int high) {
+      this.on = true;
+      this.low = low;
+      this.high = high;
+    }
+
+    private ScalingSettings() {
+      this.on = false;
+      this.low = null;
+      this.high = null;
+    }
+
+    // TODO(bugs.webrtc.org/8830): Below constructors are deprecated.
+    // Default thresholds are going away, so thresholds have to be set
+    // when scaling is on.
+    /**
+     * Creates quality based scaling setting.
+     *
+     * @param on True if quality scaling is turned on.
+     */
+    @Deprecated
+    public ScalingSettings(boolean on) {
+      this.on = on;
+      this.low = null;
+      this.high = null;
+    }
+
+    /**
+     * Creates quality based scaling settings with custom thresholds.
+     *
+     * @param on True if quality scaling is turned on.
+     * @param low Average QP at which to scale up the resolution.
+     * @param high Average QP at which to scale down the resolution.
+     */
+    @Deprecated
+    public ScalingSettings(boolean on, int low, int high) {
+      this.on = on;
+      this.low = low;
+      this.high = high;
+    }
+
+    @Override
+    public String toString() {
+      return on ? "[ " + low + ", " + high + " ]" : "OFF";
+    }
+  }
+
+  /**
+   * Bitrate limits for resolution.
+   */
+  public class ResolutionBitrateLimits {
+    /**
+     * Maximum size of video frame, in pixels, the bitrate limits are intended for.
+     */
+    public final int frameSizePixels;
+
+    /**
+     * Recommended minimum bitrate to start encoding.
+     */
+    public final int minStartBitrateBps;
+
+    /**
+     * Recommended minimum bitrate.
+     */
+    public final int minBitrateBps;
+
+    /**
+     * Recommended maximum bitrate.
+     */
+    public final int maxBitrateBps;
+
+    public ResolutionBitrateLimits(
+        int frameSizePixels, int minStartBitrateBps, int minBitrateBps, int maxBitrateBps) {
+      this.frameSizePixels = frameSizePixels;
+      this.minStartBitrateBps = minStartBitrateBps;
+      this.minBitrateBps = minBitrateBps;
+      this.maxBitrateBps = maxBitrateBps;
+    }
+
+    @CalledByNative("ResolutionBitrateLimits")
+    public int getFrameSizePixels() {
+      return frameSizePixels;
+    }
+
+    @CalledByNative("ResolutionBitrateLimits")
+    public int getMinStartBitrateBps() {
+      return minStartBitrateBps;
+    }
+
+    @CalledByNative("ResolutionBitrateLimits")
+    public int getMinBitrateBps() {
+      return minBitrateBps;
+    }
+
+    @CalledByNative("ResolutionBitrateLimits")
+    public int getMaxBitrateBps() {
+      return maxBitrateBps;
+    }
+  }
+
+  /** Rate control parameters. */
+  public class RateControlParameters {
+    /**
+     * Adjusted target bitrate, per spatial/temporal layer. May be lower or higher than the target
+     * depending on encoder behaviour.
+     */
+    public final BitrateAllocation bitrate;
+
+    /**
+     * Target framerate, in fps. A value <= 0.0 is invalid and should be interpreted as framerate
+     * target not available. In this case the encoder should fall back to the max framerate
+     * specified in `codec_settings` of the last InitEncode() call.
+     */
+    public final double framerateFps;
+
+    @CalledByNative("RateControlParameters")
+    public RateControlParameters(BitrateAllocation bitrate, double framerateFps) {
+      this.bitrate = bitrate;
+      this.framerateFps = framerateFps;
+    }
+  }
+
+  /**
+   * Metadata about the Encoder.
+   */
+  public class EncoderInfo {
+    /**
+     * The width and height of the incoming video frames should be divisible by
+     * |requested_resolution_alignment|
+     */
+    public final int requestedResolutionAlignment;
+
+    /**
+     * Same as above but if true, each simulcast layer should also be divisible by
+     * |requested_resolution_alignment|.
+     */
+    public final boolean applyAlignmentToAllSimulcastLayers;
+
+    public EncoderInfo(
+        int requestedResolutionAlignment, boolean applyAlignmentToAllSimulcastLayers) {
+      this.requestedResolutionAlignment = requestedResolutionAlignment;
+      this.applyAlignmentToAllSimulcastLayers = applyAlignmentToAllSimulcastLayers;
+    }
+
+    @CalledByNative("EncoderInfo")
+    public int getRequestedResolutionAlignment() {
+      return requestedResolutionAlignment;
+    }
+
+    @CalledByNative("EncoderInfo")
+    public boolean getApplyAlignmentToAllSimulcastLayers() {
+      return applyAlignmentToAllSimulcastLayers;
+    }
+  }
+
+  public interface Callback {
+    /**
+     * Old encoders assume that the byte buffer held by `frame` is not accessed after the call to
+     * this method returns. If the pipeline downstream needs to hold on to the buffer, it then has
+     * to make its own copy. We want to move to a model where no copying is needed, and instead use
+     * retain()/release() to signal to the encoder when it is safe to reuse the buffer.
+     *
+     * Over the transition, implementations of this class should use the maybeRetain() method if
+     * they want to keep a reference to the buffer, and fall back to copying if that method returns
+     * false.
+     */
+    void onEncodedFrame(EncodedImage frame, CodecSpecificInfo info);
+  }
+
+  /**
+   * The encoder implementation backing this interface is either 1) a Java
+   * encoder (e.g., an Android platform encoder), or alternatively 2) a native
+   * encoder (e.g., a software encoder or a C++ encoder adapter).
+   *
+   * For case 1), createNativeVideoEncoder() should return zero.
+   * In this case, we expect the native library to call the encoder through
+   * JNI using the Java interface declared below.
+   *
+   * For case 2), createNativeVideoEncoder() should return a non-zero value.
+   * In this case, we expect the native library to treat the returned value as
+   * a raw pointer of type webrtc::VideoEncoder* (ownership is transferred to
+   * the caller). The native library should then directly call the
+   * webrtc::VideoEncoder interface without going through JNI. All calls to
+   * the Java interface methods declared below should thus throw an
+   * UnsupportedOperationException.
+   */
+  @CalledByNative
+  default long createNativeVideoEncoder() {
+    return 0;
+  }
+
+  /**
+   * Returns true if the encoder is backed by hardware.
+   */
+  @CalledByNative
+  default boolean isHardwareEncoder() {
+    return true;
+  }
+
+  /**
+   * Initializes the encoding process. Call before any calls to encode.
+   */
+  @CalledByNative VideoCodecStatus initEncode(Settings settings, Callback encodeCallback);
+
+  /**
+   * Releases the encoder. No more calls to encode will be made after this call.
+   */
+  @CalledByNative VideoCodecStatus release();
+
+  /**
+   * Requests the encoder to encode a frame.
+   */
+  @CalledByNative VideoCodecStatus encode(VideoFrame frame, EncodeInfo info);
+
+  /** Sets the bitrate allocation and the target framerate for the encoder. */
+  VideoCodecStatus setRateAllocation(BitrateAllocation allocation, int framerate);
+
+  /** Sets the bitrate allocation and the target framerate for the encoder. */
+  default @CalledByNative VideoCodecStatus setRates(RateControlParameters rcParameters) {
+    // Round frame rate up to avoid overshoots.
+    int framerateFps = (int) Math.ceil(rcParameters.framerateFps);
+    return setRateAllocation(rcParameters.bitrate, framerateFps);
+  }
+
+  /** Any encoder that wants to use WebRTC provided quality scaler must implement this method. */
+  @CalledByNative ScalingSettings getScalingSettings();
+
+  /** Returns the list of bitrate limits. */
+  @CalledByNative
+  default ResolutionBitrateLimits[] getResolutionBitrateLimits() {
+    // TODO(ssilkin): Update downstream projects and remove default implementation.
+    ResolutionBitrateLimits bitrate_limits[] = {};
+    return bitrate_limits;
+  }
+
+  /**
+   * Should return a descriptive name for the implementation. Gets called once and cached. May be
+   * called from arbitrary thread.
+   */
+  @CalledByNative String getImplementationName();
+
+  @CalledByNative
+  default EncoderInfo getEncoderInfo() {
+    return new EncoderInfo(
+        /* requestedResolutionAlignment= */ 1, /* applyAlignmentToAllSimulcastLayers= */ false);
+  }
+}
diff --git a/third_party/libwebrtc/sdk/android/api/org/webrtc/VideoEncoderFactory.java b/third_party/libwebrtc/sdk/android/api/org/webrtc/VideoEncoderFactory.java
new file mode 100644
index 0000000000..2a46662d14
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/api/org/webrtc/VideoEncoderFactory.java
@@ -0,0 +1,72 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+import androidx.annotation.Nullable;
+
+/** Factory for creating VideoEncoders. */
+public interface VideoEncoderFactory {
+  public interface VideoEncoderSelector {
+    /** Called with the VideoCodecInfo of the currently used encoder. */
+    @CalledByNative("VideoEncoderSelector") void onCurrentEncoder(VideoCodecInfo info);
+
+    /**
+     * Called with the current available bitrate. Returns null if the encoder selector prefers to
+     * keep the current encoder or a VideoCodecInfo if a new encoder is preferred.
+     */
+    @Nullable @CalledByNative("VideoEncoderSelector") VideoCodecInfo onAvailableBitrate(int kbps);
+
+    /**
+     * Called every time the encoder input resolution change. Returns null if the encoder selector
+     * prefers to keep the current encoder or a VideoCodecInfo if a new encoder is preferred.
+     */
+    @Nullable
+    @CalledByNative("VideoEncoderSelector")
+    default VideoCodecInfo onResolutionChange(int widht, int height) {
+      return null;
+    }
+
+    /**
+     * Called when the currently used encoder signal itself as broken. Returns null if the encoder
+     * selector prefers to keep the current encoder or a VideoCodecInfo if a new encoder is
+     * preferred.
+     */
+    @Nullable @CalledByNative("VideoEncoderSelector") VideoCodecInfo onEncoderBroken();
+  }
+
+  /** Creates an encoder for the given video codec. */
+  @Nullable @CalledByNative VideoEncoder createEncoder(VideoCodecInfo info);
+
+  /**
+   * Enumerates the list of supported video codecs. This method will only be called once and the
+   * result will be cached.
+   */
+  @CalledByNative VideoCodecInfo[] getSupportedCodecs();
+
+  /**
+   * Enumerates the list of supported video codecs that can also be tagged with
+   * implementation information. This method will only be called once and the
+   * result will be cached.
+   */
+  @CalledByNative
+  default VideoCodecInfo[] getImplementations() {
+    return getSupportedCodecs();
+  }
+
+  /**
+   * Returns a VideoEncoderSelector if implemented by the VideoEncoderFactory,
+   * null otherwise.
+   */
+  @CalledByNative
+  default VideoEncoderSelector getEncoderSelector() {
+    return null;
+  }
+}
diff --git a/third_party/libwebrtc/sdk/android/api/org/webrtc/VideoEncoderFallback.java b/third_party/libwebrtc/sdk/android/api/org/webrtc/VideoEncoderFallback.java
new file mode 100644
index 0000000000..fa36b7c989
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/api/org/webrtc/VideoEncoderFallback.java
@@ -0,0 +1,36 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+/**
+ * A combined video encoder that falls back on a secondary encoder if the primary encoder fails.
+ */
+public class VideoEncoderFallback extends WrappedNativeVideoEncoder {
+  private final VideoEncoder fallback;
+  private final VideoEncoder primary;
+
+  public VideoEncoderFallback(VideoEncoder fallback, VideoEncoder primary) {
+    this.fallback = fallback;
+    this.primary = primary;
+  }
+
+  @Override
+  public long createNativeVideoEncoder() {
+    return nativeCreateEncoder(fallback, primary);
+  }
+
+  @Override
+  public boolean isHardwareEncoder() {
+    return primary.isHardwareEncoder();
+  }
+
+  private static native long nativeCreateEncoder(VideoEncoder fallback, VideoEncoder primary);
+}
diff --git a/third_party/libwebrtc/sdk/android/api/org/webrtc/VideoFileRenderer.java b/third_party/libwebrtc/sdk/android/api/org/webrtc/VideoFileRenderer.java
new file mode 100644
index 0000000000..aef8030459
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/api/org/webrtc/VideoFileRenderer.java
@@ -0,0 +1,162 @@
+/*
+ *  Copyright 2016 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+import android.os.Handler;
+import android.os.HandlerThread;
+import java.io.FileOutputStream;
+import java.io.IOException;
+import java.nio.ByteBuffer;
+import java.nio.charset.Charset;
+import java.util.concurrent.CountDownLatch;
+
+/**
+ * Can be used to save the video frames to file.
+ */
+public class VideoFileRenderer implements VideoSink {
+  private static final String TAG = "VideoFileRenderer";
+
+  private final HandlerThread renderThread;
+  private final Handler renderThreadHandler;
+  private final HandlerThread fileThread;
+  private final Handler fileThreadHandler;
+  private final FileOutputStream videoOutFile;
+  private final String outputFileName;
+  private final int outputFileWidth;
+  private final int outputFileHeight;
+  private final int outputFrameSize;
+  private final ByteBuffer outputFrameBuffer;
+  private EglBase eglBase;
+  private YuvConverter yuvConverter;
+  private int frameCount;
+
+  public VideoFileRenderer(String outputFile, int outputFileWidth, int outputFileHeight,
+      final EglBase.Context sharedContext) throws IOException {
+    if ((outputFileWidth % 2) == 1 || (outputFileHeight % 2) == 1) {
+      throw new IllegalArgumentException("Does not support uneven width or height");
+    }
+
+    this.outputFileName = outputFile;
+    this.outputFileWidth = outputFileWidth;
+    this.outputFileHeight = outputFileHeight;
+
+    outputFrameSize = outputFileWidth * outputFileHeight * 3 / 2;
+    outputFrameBuffer = ByteBuffer.allocateDirect(outputFrameSize);
+
+    videoOutFile = new FileOutputStream(outputFile);
+    videoOutFile.write(
+        ("YUV4MPEG2 C420 W" + outputFileWidth + " H" + outputFileHeight + " Ip F30:1 A1:1\n")
+            .getBytes(Charset.forName("US-ASCII")));
+
+    renderThread = new HandlerThread(TAG + "RenderThread");
+    renderThread.start();
+    renderThreadHandler = new Handler(renderThread.getLooper());
+
+    fileThread = new HandlerThread(TAG + "FileThread");
+    fileThread.start();
+    fileThreadHandler = new Handler(fileThread.getLooper());
+
+    ThreadUtils.invokeAtFrontUninterruptibly(renderThreadHandler, new Runnable() {
+      @Override
+      public void run() {
+        eglBase = EglBase.create(sharedContext, EglBase.CONFIG_PIXEL_BUFFER);
+        eglBase.createDummyPbufferSurface();
+        eglBase.makeCurrent();
+        yuvConverter = new YuvConverter();
+      }
+    });
+  }
+
+  @Override
+  public void onFrame(VideoFrame frame) {
+    frame.retain();
+    renderThreadHandler.post(() -> renderFrameOnRenderThread(frame));
+  }
+
+  private void renderFrameOnRenderThread(VideoFrame frame) {
+    final VideoFrame.Buffer buffer = frame.getBuffer();
+
+    // If the frame is rotated, it will be applied after cropAndScale. Therefore, if the frame is
+    // rotated by 90 degrees, swap width and height.
+    final int targetWidth = frame.getRotation() % 180 == 0 ? outputFileWidth : outputFileHeight;
+    final int targetHeight = frame.getRotation() % 180 == 0 ? outputFileHeight : outputFileWidth;
+
+    final float frameAspectRatio = (float) buffer.getWidth() / (float) buffer.getHeight();
+    final float fileAspectRatio = (float) targetWidth / (float) targetHeight;
+
+    // Calculate cropping to equalize the aspect ratio.
+    int cropWidth = buffer.getWidth();
+    int cropHeight = buffer.getHeight();
+    if (fileAspectRatio > frameAspectRatio) {
+      cropHeight = (int) (cropHeight * (frameAspectRatio / fileAspectRatio));
+    } else {
+      cropWidth = (int) (cropWidth * (fileAspectRatio / frameAspectRatio));
+    }
+
+    final int cropX = (buffer.getWidth() - cropWidth) / 2;
+    final int cropY = (buffer.getHeight() - cropHeight) / 2;
+
+    final VideoFrame.Buffer scaledBuffer =
+        buffer.cropAndScale(cropX, cropY, cropWidth, cropHeight, targetWidth, targetHeight);
+    frame.release();
+
+    final VideoFrame.I420Buffer i420 = scaledBuffer.toI420();
+    scaledBuffer.release();
+
+    fileThreadHandler.post(() -> {
+      YuvHelper.I420Rotate(i420.getDataY(), i420.getStrideY(), i420.getDataU(), i420.getStrideU(),
+          i420.getDataV(), i420.getStrideV(), outputFrameBuffer, i420.getWidth(), i420.getHeight(),
+          frame.getRotation());
+      i420.release();
+
+      try {
+        videoOutFile.write("FRAME\n".getBytes(Charset.forName("US-ASCII")));
+        videoOutFile.write(
+            outputFrameBuffer.array(), outputFrameBuffer.arrayOffset(), outputFrameSize);
+      } catch (IOException e) {
+        throw new RuntimeException("Error writing video to disk", e);
+      }
+      frameCount++;
+    });
+  }
+
+  /**
+   * Release all resources. All already posted frames will be rendered first.
+   */
+  public void release() {
+    final CountDownLatch cleanupBarrier = new CountDownLatch(1);
+    renderThreadHandler.post(() -> {
+      yuvConverter.release();
+      eglBase.release();
+      renderThread.quit();
+      cleanupBarrier.countDown();
+    });
+    ThreadUtils.awaitUninterruptibly(cleanupBarrier);
+    fileThreadHandler.post(() -> {
+      try {
+        videoOutFile.close();
+        Logging.d(TAG,
+            "Video written to disk as " + outputFileName + ". The number of frames is " + frameCount
+                + " and the dimensions of the frames are " + outputFileWidth + "x"
+                + outputFileHeight + ".");
+      } catch (IOException e) {
+        throw new RuntimeException("Error closing output file", e);
+      }
+      fileThread.quit();
+    });
+    try {
+      fileThread.join();
+    } catch (InterruptedException e) {
+      Thread.currentThread().interrupt();
+      Logging.e(TAG, "Interrupted while waiting for the write to disk to complete.", e);
+    }
+  }
+}
diff --git a/third_party/libwebrtc/sdk/android/api/org/webrtc/VideoFrame.java b/third_party/libwebrtc/sdk/android/api/org/webrtc/VideoFrame.java
new file mode 100644
index 0000000000..52a378b8f8
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/api/org/webrtc/VideoFrame.java
@@ -0,0 +1,218 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+import android.graphics.Matrix;
+import android.opengl.GLES11Ext;
+import android.opengl.GLES20;
+import androidx.annotation.Nullable;
+import java.nio.ByteBuffer;
+
+/**
+ * Java version of webrtc::VideoFrame and webrtc::VideoFrameBuffer. A difference from the C++
+ * version is that no explicit tag is used, and clients are expected to use 'instanceof' to find the
+ * right subclass of the buffer. This allows clients to create custom VideoFrame.Buffer in
+ * arbitrary format in their custom VideoSources, and then cast it back to the correct subclass in
+ * their custom VideoSinks. All implementations must also implement the toI420() function,
+ * converting from the underlying representation if necessary. I420 is the most widely accepted
+ * format and serves as a fallback for video sinks that can only handle I420, e.g. the internal
+ * WebRTC software encoders.
+ */
+public class VideoFrame implements RefCounted {
+  /**
+   * Implements image storage medium. Might be for example an OpenGL texture or a memory region
+   * containing I420-data.
+   *
+   * <p>Reference counting is needed since a video buffer can be shared between multiple VideoSinks,
+   * and the buffer needs to be returned to the VideoSource as soon as all references are gone.
+   */
+  public interface Buffer extends RefCounted {
+    /**
+     * Representation of the underlying buffer. Currently, only NATIVE and I420 are supported.
+     */
+    @CalledByNative("Buffer")
+    @VideoFrameBufferType
+    default int getBufferType() {
+      return VideoFrameBufferType.NATIVE;
+    }
+
+    /**
+     * Resolution of the buffer in pixels.
+     */
+    @CalledByNative("Buffer") int getWidth();
+    @CalledByNative("Buffer") int getHeight();
+
+    /**
+     * Returns a memory-backed frame in I420 format. If the pixel data is in another format, a
+     * conversion will take place. All implementations must provide a fallback to I420 for
+     * compatibility with e.g. the internal WebRTC software encoders.
+     *
+     * <p> Conversion may fail, for example if reading the pixel data from a texture fails. If the
+     * conversion fails, null is returned.
+     */
+    @Nullable @CalledByNative("Buffer") I420Buffer toI420();
+
+    @Override @CalledByNative("Buffer") void retain();
+    @Override @CalledByNative("Buffer") void release();
+
+    /**
+     * Crops a region defined by `cropx`, `cropY`, `cropWidth` and `cropHeight`. Scales it to size
+     * `scaleWidth` x `scaleHeight`.
+     */
+    @CalledByNative("Buffer")
+    Buffer cropAndScale(
+        int cropX, int cropY, int cropWidth, int cropHeight, int scaleWidth, int scaleHeight);
+  }
+
+  /**
+   * Interface for I420 buffers.
+   */
+  public interface I420Buffer extends Buffer {
+    @Override
+    default int getBufferType() {
+      return VideoFrameBufferType.I420;
+    }
+
+    /**
+     * Returns a direct ByteBuffer containing Y-plane data. The buffer capacity is at least
+     * getStrideY() * getHeight() bytes. The position of the returned buffer is ignored and must
+     * be 0. Callers may mutate the ByteBuffer (eg. through relative-read operations), so
+     * implementations must return a new ByteBuffer or slice for each call.
+     */
+    @CalledByNative("I420Buffer") ByteBuffer getDataY();
+    /**
+     * Returns a direct ByteBuffer containing U-plane data. The buffer capacity is at least
+     * getStrideU() * ((getHeight() + 1) / 2) bytes. The position of the returned buffer is ignored
+     * and must be 0. Callers may mutate the ByteBuffer (eg. through relative-read operations), so
+     * implementations must return a new ByteBuffer or slice for each call.
+     */
+    @CalledByNative("I420Buffer") ByteBuffer getDataU();
+    /**
+     * Returns a direct ByteBuffer containing V-plane data. The buffer capacity is at least
+     * getStrideV() * ((getHeight() + 1) / 2) bytes. The position of the returned buffer is ignored
+     * and must be 0. Callers may mutate the ByteBuffer (eg. through relative-read operations), so
+     * implementations must return a new ByteBuffer or slice for each call.
+     */
+    @CalledByNative("I420Buffer") ByteBuffer getDataV();
+
+    @CalledByNative("I420Buffer") int getStrideY();
+    @CalledByNative("I420Buffer") int getStrideU();
+    @CalledByNative("I420Buffer") int getStrideV();
+  }
+
+  /**
+   * Interface for buffers that are stored as a single texture, either in OES or RGB format.
+   */
+  public interface TextureBuffer extends Buffer {
+    enum Type {
+      OES(GLES11Ext.GL_TEXTURE_EXTERNAL_OES),
+      RGB(GLES20.GL_TEXTURE_2D);
+
+      private final int glTarget;
+
+      private Type(final int glTarget) {
+        this.glTarget = glTarget;
+      }
+
+      public int getGlTarget() {
+        return glTarget;
+      }
+    }
+
+    Type getType();
+    int getTextureId();
+
+    /**
+     * Retrieve the transform matrix associated with the frame. This transform matrix maps 2D
+     * homogeneous coordinates of the form (s, t, 1) with s and t in the inclusive range [0, 1] to
+     * the coordinate that should be used to sample that location from the buffer.
+     */
+    Matrix getTransformMatrix();
+
+    /**
+     * Create a new TextureBufferImpl with an applied transform matrix and a new size. The existing
+     * buffer is unchanged. The given transform matrix is applied first when texture coordinates are
+     * still in the unmodified [0, 1] range.
+     */
+    default TextureBuffer applyTransformMatrix(
+        Matrix transformMatrix, int newWidth, int newHeight) {
+      throw new UnsupportedOperationException("Not implemented");
+    }
+  }
+
+  private final Buffer buffer;
+  private final int rotation;
+  private final long timestampNs;
+
+  /**
+   * Constructs a new VideoFrame backed by the given {@code buffer}.
+   *
+   * @note Ownership of the buffer object is tranferred to the new VideoFrame.
+   */
+  @CalledByNative
+  public VideoFrame(Buffer buffer, int rotation, long timestampNs) {
+    if (buffer == null) {
+      throw new IllegalArgumentException("buffer not allowed to be null");
+    }
+    if (rotation % 90 != 0) {
+      throw new IllegalArgumentException("rotation must be a multiple of 90");
+    }
+    this.buffer = buffer;
+    this.rotation = rotation;
+    this.timestampNs = timestampNs;
+  }
+
+  @CalledByNative
+  public Buffer getBuffer() {
+    return buffer;
+  }
+
+  /**
+   * Rotation of the frame in degrees.
+   */
+  @CalledByNative
+  public int getRotation() {
+    return rotation;
+  }
+
+  /**
+   * Timestamp of the frame in nano seconds.
+   */
+  @CalledByNative
+  public long getTimestampNs() {
+    return timestampNs;
+  }
+
+  public int getRotatedWidth() {
+    if (rotation % 180 == 0) {
+      return buffer.getWidth();
+    }
+    return buffer.getHeight();
+  }
+
+  public int getRotatedHeight() {
+    if (rotation % 180 == 0) {
+      return buffer.getHeight();
+    }
+    return buffer.getWidth();
+  }
+
+  @Override
+  public void retain() {
+    buffer.retain();
+  }
+
+  @Override
+  @CalledByNative
+  public void release() {
+    buffer.release();
+  }
+}
diff --git a/third_party/libwebrtc/sdk/android/api/org/webrtc/VideoFrameBufferType.java b/third_party/libwebrtc/sdk/android/api/org/webrtc/VideoFrameBufferType.java
new file mode 100644
index 0000000000..7b05b88cba
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/api/org/webrtc/VideoFrameBufferType.java
@@ -0,0 +1,33 @@
+
+// Copyright 2022 The Chromium Authors. All rights reserved.
+// Use of this source code is governed by a BSD-style license that can be
+// found in the LICENSE file.
+
+// This file is autogenerated by
+//     java_cpp_enum.py
+// From
+//     ../../api/video/video_frame_buffer.h
+
+package org.webrtc;
+
+import androidx.annotation.IntDef;
+
+import java.lang.annotation.Retention;
+import java.lang.annotation.RetentionPolicy;
+
+@IntDef({
+    VideoFrameBufferType.NATIVE, VideoFrameBufferType.I420, VideoFrameBufferType.I420A,
+    VideoFrameBufferType.I422, VideoFrameBufferType.I444, VideoFrameBufferType.I010,
+    VideoFrameBufferType.I210, VideoFrameBufferType.NV12
+})
+@Retention(RetentionPolicy.SOURCE)
+public @interface VideoFrameBufferType {
+  int NATIVE = 0;
+  int I420 = 1;
+  int I420A = 2;
+  int I422 = 3;
+  int I444 = 4;
+  int I010 = 5;
+  int I210 = 6;
+  int NV12 = 7;
+}
diff --git a/third_party/libwebrtc/sdk/android/api/org/webrtc/VideoFrameDrawer.java b/third_party/libwebrtc/sdk/android/api/org/webrtc/VideoFrameDrawer.java
new file mode 100644
index 0000000000..af32587886
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/api/org/webrtc/VideoFrameDrawer.java
@@ -0,0 +1,241 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+import android.graphics.Matrix;
+import android.graphics.Point;
+import android.opengl.GLES20;
+import androidx.annotation.Nullable;
+import java.nio.ByteBuffer;
+
+/**
+ * Helper class to draw VideoFrames. Calls either drawer.drawOes, drawer.drawRgb, or
+ * drawer.drawYuv depending on the type of the buffer. The frame will be rendered with rotation
+ * taken into account. You can supply an additional render matrix for custom transformations.
+ */
+public class VideoFrameDrawer {
+  public static final String TAG = "VideoFrameDrawer";
+  /**
+   * Draws a VideoFrame.TextureBuffer. Calls either drawer.drawOes or drawer.drawRgb
+   * depending on the type of the buffer. You can supply an additional render matrix. This is
+   * used multiplied together with the transformation matrix of the frame. (M = renderMatrix *
+   * transformationMatrix)
+   */
+  public static void drawTexture(RendererCommon.GlDrawer drawer, VideoFrame.TextureBuffer buffer,
+      Matrix renderMatrix, int frameWidth, int frameHeight, int viewportX, int viewportY,
+      int viewportWidth, int viewportHeight) {
+    Matrix finalMatrix = new Matrix(buffer.getTransformMatrix());
+    finalMatrix.preConcat(renderMatrix);
+    float[] finalGlMatrix = RendererCommon.convertMatrixFromAndroidGraphicsMatrix(finalMatrix);
+    switch (buffer.getType()) {
+      case OES:
+        drawer.drawOes(buffer.getTextureId(), finalGlMatrix, frameWidth, frameHeight, viewportX,
+            viewportY, viewportWidth, viewportHeight);
+        break;
+      case RGB:
+        drawer.drawRgb(buffer.getTextureId(), finalGlMatrix, frameWidth, frameHeight, viewportX,
+            viewportY, viewportWidth, viewportHeight);
+        break;
+      default:
+        throw new RuntimeException("Unknown texture type.");
+    }
+  }
+
+  /**
+   * Helper class for uploading YUV bytebuffer frames to textures that handles stride > width. This
+   * class keeps an internal ByteBuffer to avoid unnecessary allocations for intermediate copies.
+   */
+  private static class YuvUploader {
+    // Intermediate copy buffer for uploading yuv frames that are not packed, i.e. stride > width.
+    // TODO(magjed): Investigate when GL_UNPACK_ROW_LENGTH is available, or make a custom shader
+    // that handles stride and compare performance with intermediate copy.
+    @Nullable private ByteBuffer copyBuffer;
+    @Nullable private int[] yuvTextures;
+
+    /**
+     * Upload `planes` into OpenGL textures, taking stride into consideration.
+     *
+     * @return Array of three texture indices corresponding to Y-, U-, and V-plane respectively.
+     */
+    @Nullable
+    public int[] uploadYuvData(int width, int height, int[] strides, ByteBuffer[] planes) {
+      final int[] planeWidths = new int[] {width, width / 2, width / 2};
+      final int[] planeHeights = new int[] {height, height / 2, height / 2};
+      // Make a first pass to see if we need a temporary copy buffer.
+      int copyCapacityNeeded = 0;
+      for (int i = 0; i < 3; ++i) {
+        if (strides[i] > planeWidths[i]) {
+          copyCapacityNeeded = Math.max(copyCapacityNeeded, planeWidths[i] * planeHeights[i]);
+        }
+      }
+      // Allocate copy buffer if necessary.
+      if (copyCapacityNeeded > 0
+          && (copyBuffer == null || copyBuffer.capacity() < copyCapacityNeeded)) {
+        copyBuffer = ByteBuffer.allocateDirect(copyCapacityNeeded);
+      }
+      // Make sure YUV textures are allocated.
+      if (yuvTextures == null) {
+        yuvTextures = new int[3];
+        for (int i = 0; i < 3; i++) {
+          yuvTextures[i] = GlUtil.generateTexture(GLES20.GL_TEXTURE_2D);
+        }
+      }
+      // Upload each plane.
+      for (int i = 0; i < 3; ++i) {
+        GLES20.glActiveTexture(GLES20.GL_TEXTURE0 + i);
+        GLES20.glBindTexture(GLES20.GL_TEXTURE_2D, yuvTextures[i]);
+        // GLES only accepts packed data, i.e. stride == planeWidth.
+        final ByteBuffer packedByteBuffer;
+        if (strides[i] == planeWidths[i]) {
+          // Input is packed already.
+          packedByteBuffer = planes[i];
+        } else {
+          YuvHelper.copyPlane(
+              planes[i], strides[i], copyBuffer, planeWidths[i], planeWidths[i], planeHeights[i]);
+          packedByteBuffer = copyBuffer;
+        }
+        GLES20.glTexImage2D(GLES20.GL_TEXTURE_2D, 0, GLES20.GL_LUMINANCE, planeWidths[i],
+            planeHeights[i], 0, GLES20.GL_LUMINANCE, GLES20.GL_UNSIGNED_BYTE, packedByteBuffer);
+      }
+      return yuvTextures;
+    }
+
+    @Nullable
+    public int[] uploadFromBuffer(VideoFrame.I420Buffer buffer) {
+      int[] strides = {buffer.getStrideY(), buffer.getStrideU(), buffer.getStrideV()};
+      ByteBuffer[] planes = {buffer.getDataY(), buffer.getDataU(), buffer.getDataV()};
+      return uploadYuvData(buffer.getWidth(), buffer.getHeight(), strides, planes);
+    }
+
+    @Nullable
+    public int[] getYuvTextures() {
+      return yuvTextures;
+    }
+
+    /**
+     * Releases cached resources. Uploader can still be used and the resources will be reallocated
+     * on first use.
+     */
+    public void release() {
+      copyBuffer = null;
+      if (yuvTextures != null) {
+        GLES20.glDeleteTextures(3, yuvTextures, 0);
+        yuvTextures = null;
+      }
+    }
+  }
+
+  private static int distance(float x0, float y0, float x1, float y1) {
+    return (int) Math.round(Math.hypot(x1 - x0, y1 - y0));
+  }
+
+  // These points are used to calculate the size of the part of the frame we are rendering.
+  final static float[] srcPoints =
+      new float[] {0f /* x0 */, 0f /* y0 */, 1f /* x1 */, 0f /* y1 */, 0f /* x2 */, 1f /* y2 */};
+  private final float[] dstPoints = new float[6];
+  private final Point renderSize = new Point();
+  private int renderWidth;
+  private int renderHeight;
+
+  // Calculate the frame size after `renderMatrix` is applied. Stores the output in member variables
+  // `renderWidth` and `renderHeight` to avoid allocations since this function is called for every
+  // frame.
+  private void calculateTransformedRenderSize(
+      int frameWidth, int frameHeight, @Nullable Matrix renderMatrix) {
+    if (renderMatrix == null) {
+      renderWidth = frameWidth;
+      renderHeight = frameHeight;
+      return;
+    }
+    // Transform the texture coordinates (in the range [0, 1]) according to `renderMatrix`.
+    renderMatrix.mapPoints(dstPoints, srcPoints);
+
+    // Multiply with the width and height to get the positions in terms of pixels.
+    for (int i = 0; i < 3; ++i) {
+      dstPoints[i * 2 + 0] *= frameWidth;
+      dstPoints[i * 2 + 1] *= frameHeight;
+    }
+
+    // Get the length of the sides of the transformed rectangle in terms of pixels.
+    renderWidth = distance(dstPoints[0], dstPoints[1], dstPoints[2], dstPoints[3]);
+    renderHeight = distance(dstPoints[0], dstPoints[1], dstPoints[4], dstPoints[5]);
+  }
+
+  private final YuvUploader yuvUploader = new YuvUploader();
+  // This variable will only be used for checking reference equality and is used for caching I420
+  // textures.
+  @Nullable private VideoFrame lastI420Frame;
+  private final Matrix renderMatrix = new Matrix();
+
+  public void drawFrame(VideoFrame frame, RendererCommon.GlDrawer drawer) {
+    drawFrame(frame, drawer, null /* additionalRenderMatrix */);
+  }
+
+  public void drawFrame(
+      VideoFrame frame, RendererCommon.GlDrawer drawer, Matrix additionalRenderMatrix) {
+    drawFrame(frame, drawer, additionalRenderMatrix, 0 /* viewportX */, 0 /* viewportY */,
+        frame.getRotatedWidth(), frame.getRotatedHeight());
+  }
+
+  public void drawFrame(VideoFrame frame, RendererCommon.GlDrawer drawer,
+      @Nullable Matrix additionalRenderMatrix, int viewportX, int viewportY, int viewportWidth,
+      int viewportHeight) {
+    final int width = frame.getRotatedWidth();
+    final int height = frame.getRotatedHeight();
+    calculateTransformedRenderSize(width, height, additionalRenderMatrix);
+    if (renderWidth <= 0 || renderHeight <= 0) {
+      Logging.w(TAG, "Illegal frame size: " + renderWidth + "x" + renderHeight);
+      return;
+    }
+
+    final boolean isTextureFrame = frame.getBuffer() instanceof VideoFrame.TextureBuffer;
+    renderMatrix.reset();
+    renderMatrix.preTranslate(0.5f, 0.5f);
+    if (!isTextureFrame) {
+      renderMatrix.preScale(1f, -1f); // I420-frames are upside down
+    }
+    renderMatrix.preRotate(frame.getRotation());
+    renderMatrix.preTranslate(-0.5f, -0.5f);
+    if (additionalRenderMatrix != null) {
+      renderMatrix.preConcat(additionalRenderMatrix);
+    }
+
+    if (isTextureFrame) {
+      lastI420Frame = null;
+      drawTexture(drawer, (VideoFrame.TextureBuffer) frame.getBuffer(), renderMatrix, renderWidth,
+          renderHeight, viewportX, viewportY, viewportWidth, viewportHeight);
+    } else {
+      // Only upload the I420 data to textures once per frame, if we are called multiple times
+      // with the same frame.
+      if (frame != lastI420Frame) {
+        lastI420Frame = frame;
+        final VideoFrame.I420Buffer i420Buffer = frame.getBuffer().toI420();
+        yuvUploader.uploadFromBuffer(i420Buffer);
+        i420Buffer.release();
+      }
+
+      drawer.drawYuv(yuvUploader.getYuvTextures(),
+          RendererCommon.convertMatrixFromAndroidGraphicsMatrix(renderMatrix), renderWidth,
+          renderHeight, viewportX, viewportY, viewportWidth, viewportHeight);
+    }
+  }
+
+  public VideoFrame.Buffer prepareBufferForViewportSize(
+      VideoFrame.Buffer buffer, int width, int height) {
+    buffer.retain();
+    return buffer;
+  }
+
+  public void release() {
+    yuvUploader.release();
+    lastI420Frame = null;
+  }
+}
diff --git a/third_party/libwebrtc/sdk/android/api/org/webrtc/VideoProcessor.java b/third_party/libwebrtc/sdk/android/api/org/webrtc/VideoProcessor.java
new file mode 100644
index 0000000000..c39a55c27e
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/api/org/webrtc/VideoProcessor.java
@@ -0,0 +1,76 @@
+/*
+ *  Copyright 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+import androidx.annotation.Nullable;
+
+/**
+ * Lightweight abstraction for an object that can receive video frames, process them, and pass them
+ * on to another object. This object is also allowed to observe capturer start/stop.
+ */
+public interface VideoProcessor extends CapturerObserver {
+  public static class FrameAdaptationParameters {
+    public final int cropX;
+    public final int cropY;
+    public final int cropWidth;
+    public final int cropHeight;
+    public final int scaleWidth;
+    public final int scaleHeight;
+    public final long timestampNs;
+    public final boolean drop;
+
+    public FrameAdaptationParameters(int cropX, int cropY, int cropWidth, int cropHeight,
+        int scaleWidth, int scaleHeight, long timestampNs, boolean drop) {
+      this.cropX = cropX;
+      this.cropY = cropY;
+      this.cropWidth = cropWidth;
+      this.cropHeight = cropHeight;
+      this.scaleWidth = scaleWidth;
+      this.scaleHeight = scaleHeight;
+      this.timestampNs = timestampNs;
+      this.drop = drop;
+    }
+  }
+
+  /**
+   * This is a chance to access an unadapted frame. The default implementation applies the
+   * adaptation and forwards the frame to {@link #onFrameCaptured(VideoFrame)}.
+   */
+  default void onFrameCaptured(VideoFrame frame, FrameAdaptationParameters parameters) {
+    VideoFrame adaptedFrame = applyFrameAdaptationParameters(frame, parameters);
+    if (adaptedFrame != null) {
+      onFrameCaptured(adaptedFrame);
+      adaptedFrame.release();
+    }
+  }
+
+  /**
+   * Set the sink that receives the output from this processor. Null can be passed in to unregister
+   * a sink.
+   */
+  void setSink(@Nullable VideoSink sink);
+
+  /**
+   * Applies the frame adaptation parameters to a frame. Returns null if the frame is meant to be
+   * dropped. Returns a new frame. The caller is responsible for releasing the returned frame.
+   */
+  public static @Nullable VideoFrame applyFrameAdaptationParameters(
+      VideoFrame frame, FrameAdaptationParameters parameters) {
+    if (parameters.drop) {
+      return null;
+    }
+
+    final VideoFrame.Buffer adaptedBuffer =
+        frame.getBuffer().cropAndScale(parameters.cropX, parameters.cropY, parameters.cropWidth,
+            parameters.cropHeight, parameters.scaleWidth, parameters.scaleHeight);
+    return new VideoFrame(adaptedBuffer, frame.getRotation(), parameters.timestampNs);
+  }
+}
diff --git a/third_party/libwebrtc/sdk/android/api/org/webrtc/VideoSink.java b/third_party/libwebrtc/sdk/android/api/org/webrtc/VideoSink.java
new file mode 100644
index 0000000000..5a0a6c719c
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/api/org/webrtc/VideoSink.java
@@ -0,0 +1,23 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+/**
+ * Java version of rtc::VideoSinkInterface.
+ */
+public interface VideoSink {
+  /**
+   * Implementations should call frame.retain() if they need to hold a reference to the frame after
+   * this function returns. Each call to retain() should be followed by a call to frame.release()
+   * when the reference is no longer needed.
+   */
+  @CalledByNative void onFrame(VideoFrame frame);
+}
diff --git a/third_party/libwebrtc/sdk/android/api/org/webrtc/VideoSource.java b/third_party/libwebrtc/sdk/android/api/org/webrtc/VideoSource.java
new file mode 100644
index 0000000000..2e22d1a2db
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/api/org/webrtc/VideoSource.java
@@ -0,0 +1,162 @@
+/*
+ *  Copyright 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+import androidx.annotation.Nullable;
+
+/**
+ * Java wrapper of native AndroidVideoTrackSource.
+ */
+public class VideoSource extends MediaSource {
+  /** Simple aspect ratio clas for use in constraining output format. */
+  public static class AspectRatio {
+    public static final AspectRatio UNDEFINED = new AspectRatio(/* width= */ 0, /* height= */ 0);
+
+    public final int width;
+    public final int height;
+
+    public AspectRatio(int width, int height) {
+      this.width = width;
+      this.height = height;
+    }
+  }
+
+  private final NativeAndroidVideoTrackSource nativeAndroidVideoTrackSource;
+  private final Object videoProcessorLock = new Object();
+  @Nullable private VideoProcessor videoProcessor;
+  private boolean isCapturerRunning;
+
+  private final CapturerObserver capturerObserver = new CapturerObserver() {
+    @Override
+    public void onCapturerStarted(boolean success) {
+      nativeAndroidVideoTrackSource.setState(success);
+      synchronized (videoProcessorLock) {
+        isCapturerRunning = success;
+        if (videoProcessor != null) {
+          videoProcessor.onCapturerStarted(success);
+        }
+      }
+    }
+
+    @Override
+    public void onCapturerStopped() {
+      nativeAndroidVideoTrackSource.setState(/* isLive= */ false);
+      synchronized (videoProcessorLock) {
+        isCapturerRunning = false;
+        if (videoProcessor != null) {
+          videoProcessor.onCapturerStopped();
+        }
+      }
+    }
+
+    @Override
+    public void onFrameCaptured(VideoFrame frame) {
+      final VideoProcessor.FrameAdaptationParameters parameters =
+          nativeAndroidVideoTrackSource.adaptFrame(frame);
+      synchronized (videoProcessorLock) {
+        if (videoProcessor != null) {
+          videoProcessor.onFrameCaptured(frame, parameters);
+          return;
+        }
+      }
+
+      VideoFrame adaptedFrame = VideoProcessor.applyFrameAdaptationParameters(frame, parameters);
+      if (adaptedFrame != null) {
+        nativeAndroidVideoTrackSource.onFrameCaptured(adaptedFrame);
+        adaptedFrame.release();
+      }
+    }
+  };
+
+  public VideoSource(long nativeSource) {
+    super(nativeSource);
+    this.nativeAndroidVideoTrackSource = new NativeAndroidVideoTrackSource(nativeSource);
+  }
+
+  /**
+   * Calling this function will cause frames to be scaled down to the requested resolution. Also,
+   * frames will be cropped to match the requested aspect ratio, and frames will be dropped to match
+   * the requested fps. The requested aspect ratio is orientation agnostic and will be adjusted to
+   * maintain the input orientation, so it doesn't matter if e.g. 1280x720 or 720x1280 is requested.
+   */
+  public void adaptOutputFormat(int width, int height, int fps) {
+    final int maxSide = Math.max(width, height);
+    final int minSide = Math.min(width, height);
+    adaptOutputFormat(maxSide, minSide, minSide, maxSide, fps);
+  }
+
+  /**
+   * Same as above, but allows setting two different target resolutions depending on incoming
+   * frame orientation. This gives more fine-grained control and can e.g. be used to force landscape
+   * video to be cropped to portrait video.
+   */
+  public void adaptOutputFormat(
+      int landscapeWidth, int landscapeHeight, int portraitWidth, int portraitHeight, int fps) {
+    adaptOutputFormat(new AspectRatio(landscapeWidth, landscapeHeight),
+        /* maxLandscapePixelCount= */ landscapeWidth * landscapeHeight,
+        new AspectRatio(portraitWidth, portraitHeight),
+        /* maxPortraitPixelCount= */ portraitWidth * portraitHeight, fps);
+  }
+
+  /** Same as above, with even more control as each constraint is optional. */
+  public void adaptOutputFormat(AspectRatio targetLandscapeAspectRatio,
+      @Nullable Integer maxLandscapePixelCount, AspectRatio targetPortraitAspectRatio,
+      @Nullable Integer maxPortraitPixelCount, @Nullable Integer maxFps) {
+    nativeAndroidVideoTrackSource.adaptOutputFormat(targetLandscapeAspectRatio,
+        maxLandscapePixelCount, targetPortraitAspectRatio, maxPortraitPixelCount, maxFps);
+  }
+
+  public void setIsScreencast(boolean isScreencast) {
+    nativeAndroidVideoTrackSource.setIsScreencast(isScreencast);
+  }
+
+  /**
+   * Hook for injecting a custom video processor before frames are passed onto WebRTC. The frames
+   * will be cropped and scaled depending on CPU and network conditions before they are passed to
+   * the video processor. Frames will be delivered to the video processor on the same thread they
+   * are passed to this object. The video processor is allowed to deliver the processed frames
+   * back on any thread.
+   */
+  public void setVideoProcessor(@Nullable VideoProcessor newVideoProcessor) {
+    synchronized (videoProcessorLock) {
+      if (videoProcessor != null) {
+        videoProcessor.setSink(/* sink= */ null);
+        if (isCapturerRunning) {
+          videoProcessor.onCapturerStopped();
+        }
+      }
+      videoProcessor = newVideoProcessor;
+      if (newVideoProcessor != null) {
+        newVideoProcessor.setSink(
+            (frame)
+                -> runWithReference(() -> nativeAndroidVideoTrackSource.onFrameCaptured(frame)));
+        if (isCapturerRunning) {
+          newVideoProcessor.onCapturerStarted(/* success= */ true);
+        }
+      }
+    }
+  }
+
+  public CapturerObserver getCapturerObserver() {
+    return capturerObserver;
+  }
+
+  /** Returns a pointer to webrtc::VideoTrackSourceInterface. */
+  long getNativeVideoTrackSource() {
+    return getNativeMediaSource();
+  }
+
+  @Override
+  public void dispose() {
+    setVideoProcessor(/* newVideoProcessor= */ null);
+    super.dispose();
+  }
+}
diff --git a/third_party/libwebrtc/sdk/android/api/org/webrtc/VideoTrack.java b/third_party/libwebrtc/sdk/android/api/org/webrtc/VideoTrack.java
new file mode 100644
index 0000000000..5593d424f3
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/api/org/webrtc/VideoTrack.java
@@ -0,0 +1,76 @@
+/*
+ *  Copyright 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+import java.util.IdentityHashMap;
+
+/** Java version of VideoTrackInterface. */
+public class VideoTrack extends MediaStreamTrack {
+  private final IdentityHashMap<VideoSink, Long> sinks = new IdentityHashMap<VideoSink, Long>();
+
+  public VideoTrack(long nativeTrack) {
+    super(nativeTrack);
+  }
+
+  /**
+   * Adds a VideoSink to the track.
+   *
+   * A track can have any number of VideoSinks. VideoSinks will replace
+   * renderers. However, converting old style texture frames will involve costly
+   * conversion to I420 so it is not recommended to upgrade before all your
+   * sources produce VideoFrames.
+   */
+  public void addSink(VideoSink sink) {
+    if (sink == null) {
+      throw new IllegalArgumentException("The VideoSink is not allowed to be null");
+    }
+    // We allow calling addSink() with the same sink multiple times. This is similar to the C++
+    // VideoTrack::AddOrUpdateSink().
+    if (!sinks.containsKey(sink)) {
+      final long nativeSink = nativeWrapSink(sink);
+      sinks.put(sink, nativeSink);
+      nativeAddSink(getNativeMediaStreamTrack(), nativeSink);
+    }
+  }
+
+  /**
+   * Removes a VideoSink from the track.
+   *
+   * If the VideoSink was not attached to the track, this is a no-op.
+   */
+  public void removeSink(VideoSink sink) {
+    final Long nativeSink = sinks.remove(sink);
+    if (nativeSink != null) {
+      nativeRemoveSink(getNativeMediaStreamTrack(), nativeSink);
+      nativeFreeSink(nativeSink);
+    }
+  }
+
+  @Override
+  public void dispose() {
+    for (long nativeSink : sinks.values()) {
+      nativeRemoveSink(getNativeMediaStreamTrack(), nativeSink);
+      nativeFreeSink(nativeSink);
+    }
+    sinks.clear();
+    super.dispose();
+  }
+
+  /** Returns a pointer to webrtc::VideoTrackInterface. */
+  long getNativeVideoTrack() {
+    return getNativeMediaStreamTrack();
+  }
+
+  private static native void nativeAddSink(long track, long nativeSink);
+  private static native void nativeRemoveSink(long track, long nativeSink);
+  private static native long nativeWrapSink(VideoSink sink);
+  private static native void nativeFreeSink(long sink);
+}
diff --git a/third_party/libwebrtc/sdk/android/api/org/webrtc/WrappedNativeVideoDecoder.java b/third_party/libwebrtc/sdk/android/api/org/webrtc/WrappedNativeVideoDecoder.java
new file mode 100644
index 0000000000..027120e48e
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/api/org/webrtc/WrappedNativeVideoDecoder.java
@@ -0,0 +1,38 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+/**
+ * Wraps a native webrtc::VideoDecoder.
+ */
+public abstract class WrappedNativeVideoDecoder implements VideoDecoder {
+  @Override public abstract long createNativeVideoDecoder();
+
+  @Override
+  public final VideoCodecStatus initDecode(Settings settings, Callback decodeCallback) {
+    throw new UnsupportedOperationException("Not implemented.");
+  }
+
+  @Override
+  public final VideoCodecStatus release() {
+    throw new UnsupportedOperationException("Not implemented.");
+  }
+
+  @Override
+  public final VideoCodecStatus decode(EncodedImage frame, DecodeInfo info) {
+    throw new UnsupportedOperationException("Not implemented.");
+  }
+
+  @Override
+  public final String getImplementationName() {
+    throw new UnsupportedOperationException("Not implemented.");
+  }
+}
diff --git a/third_party/libwebrtc/sdk/android/api/org/webrtc/WrappedNativeVideoEncoder.java b/third_party/libwebrtc/sdk/android/api/org/webrtc/WrappedNativeVideoEncoder.java
new file mode 100644
index 0000000000..7d0908a6ac
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/api/org/webrtc/WrappedNativeVideoEncoder.java
@@ -0,0 +1,49 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+/**
+ * Wraps a native webrtc::VideoEncoder.
+ */
+public abstract class WrappedNativeVideoEncoder implements VideoEncoder {
+  @Override public abstract long createNativeVideoEncoder();
+  @Override public abstract boolean isHardwareEncoder();
+
+  @Override
+  public final VideoCodecStatus initEncode(Settings settings, Callback encodeCallback) {
+    throw new UnsupportedOperationException("Not implemented.");
+  }
+
+  @Override
+  public final VideoCodecStatus release() {
+    throw new UnsupportedOperationException("Not implemented.");
+  }
+
+  @Override
+  public final VideoCodecStatus encode(VideoFrame frame, EncodeInfo info) {
+    throw new UnsupportedOperationException("Not implemented.");
+  }
+
+  @Override
+  public final VideoCodecStatus setRateAllocation(BitrateAllocation allocation, int framerate) {
+    throw new UnsupportedOperationException("Not implemented.");
+  }
+
+  @Override
+  public final ScalingSettings getScalingSettings() {
+    throw new UnsupportedOperationException("Not implemented.");
+  }
+
+  @Override
+  public final String getImplementationName() {
+    throw new UnsupportedOperationException("Not implemented.");
+  }
+}
diff --git a/third_party/libwebrtc/sdk/android/api/org/webrtc/YuvConverter.java b/third_party/libwebrtc/sdk/android/api/org/webrtc/YuvConverter.java
new file mode 100644
index 0000000000..c855d4be41
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/api/org/webrtc/YuvConverter.java
@@ -0,0 +1,252 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+import android.graphics.Matrix;
+import android.opengl.GLES20;
+import android.opengl.GLException;
+import androidx.annotation.Nullable;
+import java.nio.ByteBuffer;
+import org.webrtc.VideoFrame.I420Buffer;
+import org.webrtc.VideoFrame.TextureBuffer;
+
+/**
+ * Class for converting OES textures to a YUV ByteBuffer. It can be constructed on any thread, but
+ * should only be operated from a single thread with an active EGL context.
+ */
+public final class YuvConverter {
+  private static final String TAG = "YuvConverter";
+
+  private static final String FRAGMENT_SHADER =
+      // Difference in texture coordinate corresponding to one
+      // sub-pixel in the x direction.
+      "uniform vec2 xUnit;\n"
+      // Color conversion coefficients, including constant term
+      + "uniform vec4 coeffs;\n"
+      + "\n"
+      + "void main() {\n"
+      // Since the alpha read from the texture is always 1, this could
+      // be written as a mat4 x vec4 multiply. However, that seems to
+      // give a worse framerate, possibly because the additional
+      // multiplies by 1.0 consume resources.
+      + "  gl_FragColor.r = coeffs.a + dot(coeffs.rgb,\n"
+      + "      sample(tc - 1.5 * xUnit).rgb);\n"
+      + "  gl_FragColor.g = coeffs.a + dot(coeffs.rgb,\n"
+      + "      sample(tc - 0.5 * xUnit).rgb);\n"
+      + "  gl_FragColor.b = coeffs.a + dot(coeffs.rgb,\n"
+      + "      sample(tc + 0.5 * xUnit).rgb);\n"
+      + "  gl_FragColor.a = coeffs.a + dot(coeffs.rgb,\n"
+      + "      sample(tc + 1.5 * xUnit).rgb);\n"
+      + "}\n";
+
+  private static class ShaderCallbacks implements GlGenericDrawer.ShaderCallbacks {
+    // Y'UV444 to RGB888, see https://en.wikipedia.org/wiki/YUV#Y%E2%80%B2UV444_to_RGB888_conversion
+    // We use the ITU-R BT.601 coefficients for Y, U and V.
+    // The values in Wikipedia are inaccurate, the accurate values derived from the spec are:
+    // Y = 0.299 * R + 0.587 * G + 0.114 * B
+    // U = -0.168736 * R - 0.331264 * G + 0.5 * B + 0.5
+    // V = 0.5 * R - 0.418688 * G - 0.0813124 * B + 0.5
+    // To map the Y-values to range [16-235] and U- and V-values to range [16-240], the matrix has
+    // been multiplied with matrix:
+    // {{219 / 255, 0, 0, 16 / 255},
+    // {0, 224 / 255, 0, 16 / 255},
+    // {0, 0, 224 / 255, 16 / 255},
+    // {0, 0, 0, 1}}
+    private static final float[] yCoeffs =
+        new float[] {0.256788f, 0.504129f, 0.0979059f, 0.0627451f};
+    private static final float[] uCoeffs =
+        new float[] {-0.148223f, -0.290993f, 0.439216f, 0.501961f};
+    private static final float[] vCoeffs =
+        new float[] {0.439216f, -0.367788f, -0.0714274f, 0.501961f};
+
+    private int xUnitLoc;
+    private int coeffsLoc;
+
+    private float[] coeffs;
+    private float stepSize;
+
+    public void setPlaneY() {
+      coeffs = yCoeffs;
+      stepSize = 1.0f;
+    }
+
+    public void setPlaneU() {
+      coeffs = uCoeffs;
+      stepSize = 2.0f;
+    }
+
+    public void setPlaneV() {
+      coeffs = vCoeffs;
+      stepSize = 2.0f;
+    }
+
+    @Override
+    public void onNewShader(GlShader shader) {
+      xUnitLoc = shader.getUniformLocation("xUnit");
+      coeffsLoc = shader.getUniformLocation("coeffs");
+    }
+
+    @Override
+    public void onPrepareShader(GlShader shader, float[] texMatrix, int frameWidth, int frameHeight,
+        int viewportWidth, int viewportHeight) {
+      GLES20.glUniform4fv(coeffsLoc, /* count= */ 1, coeffs, /* offset= */ 0);
+      // Matrix * (1;0;0;0) / (width / stepSize). Note that OpenGL uses column major order.
+      GLES20.glUniform2f(
+          xUnitLoc, stepSize * texMatrix[0] / frameWidth, stepSize * texMatrix[1] / frameWidth);
+    }
+  }
+
+  private final ThreadUtils.ThreadChecker threadChecker = new ThreadUtils.ThreadChecker();
+  private final GlTextureFrameBuffer i420TextureFrameBuffer =
+      new GlTextureFrameBuffer(GLES20.GL_RGBA);
+  private final ShaderCallbacks shaderCallbacks = new ShaderCallbacks();
+  private final GlGenericDrawer drawer = new GlGenericDrawer(FRAGMENT_SHADER, shaderCallbacks);
+  private final VideoFrameDrawer videoFrameDrawer;
+
+  /**
+   * This class should be constructed on a thread that has an active EGL context.
+   */
+  public YuvConverter() {
+    this(new VideoFrameDrawer());
+  }
+
+  public YuvConverter(VideoFrameDrawer videoFrameDrawer) {
+    this.videoFrameDrawer = videoFrameDrawer;
+    threadChecker.detachThread();
+  }
+
+  /** Converts the texture buffer to I420. */
+  @Nullable
+  public I420Buffer convert(TextureBuffer inputTextureBuffer) {
+    try {
+      return convertInternal(inputTextureBuffer);
+    } catch (GLException e) {
+      Logging.w(TAG, "Failed to convert TextureBuffer", e);
+    }
+    return null;
+  }
+
+  private I420Buffer convertInternal(TextureBuffer inputTextureBuffer) {
+    TextureBuffer preparedBuffer = (TextureBuffer) videoFrameDrawer.prepareBufferForViewportSize(
+        inputTextureBuffer, inputTextureBuffer.getWidth(), inputTextureBuffer.getHeight());
+
+    // We draw into a buffer laid out like
+    //
+    //    +---------+
+    //    |         |
+    //    |  Y      |
+    //    |         |
+    //    |         |
+    //    +----+----+
+    //    | U  | V  |
+    //    |    |    |
+    //    +----+----+
+    //
+    // In memory, we use the same stride for all of Y, U and V. The
+    // U data starts at offset `height` * `stride` from the Y data,
+    // and the V data starts at at offset |stride/2| from the U
+    // data, with rows of U and V data alternating.
+    //
+    // Now, it would have made sense to allocate a pixel buffer with
+    // a single byte per pixel (EGL10.EGL_COLOR_BUFFER_TYPE,
+    // EGL10.EGL_LUMINANCE_BUFFER,), but that seems to be
+    // unsupported by devices. So do the following hack: Allocate an
+    // RGBA buffer, of width `stride`/4. To render each of these
+    // large pixels, sample the texture at 4 different x coordinates
+    // and store the results in the four components.
+    //
+    // Since the V data needs to start on a boundary of such a
+    // larger pixel, it is not sufficient that `stride` is even, it
+    // has to be a multiple of 8 pixels.
+    final int frameWidth = preparedBuffer.getWidth();
+    final int frameHeight = preparedBuffer.getHeight();
+    final int stride = ((frameWidth + 7) / 8) * 8;
+    final int uvHeight = (frameHeight + 1) / 2;
+    // Total height of the combined memory layout.
+    final int totalHeight = frameHeight + uvHeight;
+    final ByteBuffer i420ByteBuffer = JniCommon.nativeAllocateByteBuffer(stride * totalHeight);
+    // Viewport width is divided by four since we are squeezing in four color bytes in each RGBA
+    // pixel.
+    final int viewportWidth = stride / 4;
+
+    // Produce a frame buffer starting at top-left corner, not bottom-left.
+    final Matrix renderMatrix = new Matrix();
+    renderMatrix.preTranslate(0.5f, 0.5f);
+    renderMatrix.preScale(1f, -1f);
+    renderMatrix.preTranslate(-0.5f, -0.5f);
+
+    i420TextureFrameBuffer.setSize(viewportWidth, totalHeight);
+
+    // Bind our framebuffer.
+    GLES20.glBindFramebuffer(GLES20.GL_FRAMEBUFFER, i420TextureFrameBuffer.getFrameBufferId());
+    GlUtil.checkNoGLES2Error("glBindFramebuffer");
+
+    // Draw Y.
+    shaderCallbacks.setPlaneY();
+    VideoFrameDrawer.drawTexture(drawer, preparedBuffer, renderMatrix, frameWidth, frameHeight,
+        /* viewportX= */ 0, /* viewportY= */ 0, viewportWidth,
+        /* viewportHeight= */ frameHeight);
+
+    // Draw U.
+    shaderCallbacks.setPlaneU();
+    VideoFrameDrawer.drawTexture(drawer, preparedBuffer, renderMatrix, frameWidth, frameHeight,
+        /* viewportX= */ 0, /* viewportY= */ frameHeight, viewportWidth / 2,
+        /* viewportHeight= */ uvHeight);
+
+    // Draw V.
+    shaderCallbacks.setPlaneV();
+    VideoFrameDrawer.drawTexture(drawer, preparedBuffer, renderMatrix, frameWidth, frameHeight,
+        /* viewportX= */ viewportWidth / 2, /* viewportY= */ frameHeight, viewportWidth / 2,
+        /* viewportHeight= */ uvHeight);
+
+    GLES20.glReadPixels(0, 0, i420TextureFrameBuffer.getWidth(), i420TextureFrameBuffer.getHeight(),
+        GLES20.GL_RGBA, GLES20.GL_UNSIGNED_BYTE, i420ByteBuffer);
+
+    GlUtil.checkNoGLES2Error("YuvConverter.convert");
+
+    // Restore normal framebuffer.
+    GLES20.glBindFramebuffer(GLES20.GL_FRAMEBUFFER, 0);
+
+    // Prepare Y, U, and V ByteBuffer slices.
+    final int yPos = 0;
+    final int uPos = yPos + stride * frameHeight;
+    // Rows of U and V alternate in the buffer, so V data starts after the first row of U.
+    final int vPos = uPos + stride / 2;
+
+    i420ByteBuffer.position(yPos);
+    i420ByteBuffer.limit(yPos + stride * frameHeight);
+    final ByteBuffer dataY = i420ByteBuffer.slice();
+
+    i420ByteBuffer.position(uPos);
+    // The last row does not have padding.
+    final int uvSize = stride * (uvHeight - 1) + stride / 2;
+    i420ByteBuffer.limit(uPos + uvSize);
+    final ByteBuffer dataU = i420ByteBuffer.slice();
+
+    i420ByteBuffer.position(vPos);
+    i420ByteBuffer.limit(vPos + uvSize);
+    final ByteBuffer dataV = i420ByteBuffer.slice();
+
+    preparedBuffer.release();
+
+    return JavaI420Buffer.wrap(frameWidth, frameHeight, dataY, stride, dataU, stride, dataV, stride,
+        () -> { JniCommon.nativeFreeByteBuffer(i420ByteBuffer); });
+  }
+
+  public void release() {
+    threadChecker.checkIsOnValidThread();
+    drawer.release();
+    i420TextureFrameBuffer.release();
+    videoFrameDrawer.release();
+    // Allow this class to be reused.
+    threadChecker.detachThread();
+  }
+}
diff --git a/third_party/libwebrtc/sdk/android/api/org/webrtc/YuvHelper.java b/third_party/libwebrtc/sdk/android/api/org/webrtc/YuvHelper.java
new file mode 100644
index 0000000000..afb8e837d1
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/api/org/webrtc/YuvHelper.java
@@ -0,0 +1,200 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+import java.nio.ByteBuffer;
+
+/** Wraps libyuv methods to Java. All passed byte buffers must be direct byte buffers. */
+public class YuvHelper {
+  /**
+   * Copy I420 Buffer to a contiguously allocated buffer.
+   * <p> In Android, MediaCodec can request a buffer of a specific layout with the stride and
+   * slice-height (or plane height), and this function is used in this case.
+   * <p> For more information, see
+   * https://cs.android.com/android/platform/superproject/+/64fea7e5726daebc40f46890100837c01091100d:frameworks/base/media/java/android/media/MediaFormat.java;l=568
+   * @param dstStrideY the stride of output buffers' Y plane.
+   * @param dstSliceHeightY the slice-height of output buffer's Y plane.
+   * @param dstStrideU the stride of output buffers' U (and V) plane.
+   * @param dstSliceHeightU the slice-height of output buffer's U (and V) plane
+   */
+  public static void I420Copy(ByteBuffer srcY, int srcStrideY, ByteBuffer srcU, int srcStrideU,
+      ByteBuffer srcV, int srcStrideV, ByteBuffer dst, int dstWidth, int dstHeight, int dstStrideY,
+      int dstSliceHeightY, int dstStrideU, int dstSliceHeightU) {
+    final int chromaWidth = (dstWidth + 1) / 2;
+    final int chromaHeight = (dstHeight + 1) / 2;
+
+    final int dstStartY = 0;
+    final int dstEndY = dstStartY + dstStrideY * dstHeight;
+    final int dstStartU = dstStartY + dstStrideY * dstSliceHeightY;
+    final int dstEndU = dstStartU + dstStrideU * chromaHeight;
+    final int dstStartV = dstStartU + dstStrideU * dstSliceHeightU;
+    // The last line doesn't need any padding, so  use chromaWidth
+    // to calculate the exact end position.
+    final int dstEndV = dstStartV + dstStrideU * (chromaHeight - 1) + chromaWidth;
+    if (dst.capacity() < dstEndV) {
+      throw new IllegalArgumentException("Expected destination buffer capacity to be at least "
+          + dstEndV + " was " + dst.capacity());
+    }
+
+    dst.limit(dstEndY);
+    dst.position(dstStartY);
+    final ByteBuffer dstY = dst.slice();
+    dst.limit(dstEndU);
+    dst.position(dstStartU);
+    final ByteBuffer dstU = dst.slice();
+    dst.limit(dstEndV);
+    dst.position(dstStartV);
+    final ByteBuffer dstV = dst.slice();
+
+    I420Copy(srcY, srcStrideY, srcU, srcStrideU, srcV, srcStrideV, dstY, dstStrideY, dstU,
+        dstStrideU, dstV, dstStrideU, dstWidth, dstHeight);
+  }
+
+  /** Helper method for copying I420 to tightly packed destination buffer. */
+  public static void I420Copy(ByteBuffer srcY, int srcStrideY, ByteBuffer srcU, int srcStrideU,
+      ByteBuffer srcV, int srcStrideV, ByteBuffer dst, int dstWidth, int dstHeight) {
+    I420Copy(srcY, srcStrideY, srcU, srcStrideU, srcV, srcStrideV, dst, dstWidth, dstHeight,
+        dstWidth, dstHeight, (dstWidth + 1) / 2, (dstHeight + 1) / 2);
+  }
+
+  /**
+   * Copy I420 Buffer to a contiguously allocated buffer.
+   * @param dstStrideY the stride of output buffers' Y plane.
+   * @param dstSliceHeightY the slice-height of output buffer's Y plane.
+   */
+  public static void I420ToNV12(ByteBuffer srcY, int srcStrideY, ByteBuffer srcU, int srcStrideU,
+      ByteBuffer srcV, int srcStrideV, ByteBuffer dst, int dstWidth, int dstHeight, int dstStrideY,
+      int dstSliceHeightY) {
+    final int chromaHeight = (dstHeight + 1) / 2;
+    final int chromaWidth = (dstWidth + 1) / 2;
+
+    final int dstStartY = 0;
+    final int dstEndY = dstStartY + dstStrideY * dstHeight;
+    final int dstStartUV = dstStartY + dstStrideY * dstSliceHeightY;
+    final int dstEndUV = dstStartUV + chromaWidth * chromaHeight * 2;
+    if (dst.capacity() < dstEndUV) {
+      throw new IllegalArgumentException("Expected destination buffer capacity to be at least "
+          + dstEndUV + " was " + dst.capacity());
+    }
+
+    dst.limit(dstEndY);
+    dst.position(dstStartY);
+    final ByteBuffer dstY = dst.slice();
+    dst.limit(dstEndUV);
+    dst.position(dstStartUV);
+    final ByteBuffer dstUV = dst.slice();
+
+    I420ToNV12(srcY, srcStrideY, srcU, srcStrideU, srcV, srcStrideV, dstY, dstStrideY, dstUV,
+        chromaWidth * 2, dstWidth, dstHeight);
+  }
+
+  /** Helper method for copying I420 to tightly packed NV12 destination buffer. */
+  public static void I420ToNV12(ByteBuffer srcY, int srcStrideY, ByteBuffer srcU, int srcStrideU,
+      ByteBuffer srcV, int srcStrideV, ByteBuffer dst, int dstWidth, int dstHeight) {
+    I420ToNV12(srcY, srcStrideY, srcU, srcStrideU, srcV, srcStrideV, dst, dstWidth, dstHeight,
+        dstWidth, dstHeight);
+  }
+
+  /** Helper method for rotating I420 to tightly packed destination buffer. */
+  public static void I420Rotate(ByteBuffer srcY, int srcStrideY, ByteBuffer srcU, int srcStrideU,
+      ByteBuffer srcV, int srcStrideV, ByteBuffer dst, int srcWidth, int srcHeight,
+      int rotationMode) {
+    final int dstWidth = rotationMode % 180 == 0 ? srcWidth : srcHeight;
+    final int dstHeight = rotationMode % 180 == 0 ? srcHeight : srcWidth;
+
+    final int dstChromaHeight = (dstHeight + 1) / 2;
+    final int dstChromaWidth = (dstWidth + 1) / 2;
+
+    final int minSize = dstWidth * dstHeight + dstChromaWidth * dstChromaHeight * 2;
+    if (dst.capacity() < minSize) {
+      throw new IllegalArgumentException("Expected destination buffer capacity to be at least "
+          + minSize + " was " + dst.capacity());
+    }
+
+    final int startY = 0;
+    final int startU = dstHeight * dstWidth;
+    final int startV = startU + dstChromaHeight * dstChromaWidth;
+
+    dst.position(startY);
+    final ByteBuffer dstY = dst.slice();
+    dst.position(startU);
+    final ByteBuffer dstU = dst.slice();
+    dst.position(startV);
+    final ByteBuffer dstV = dst.slice();
+
+    nativeI420Rotate(srcY, srcStrideY, srcU, srcStrideU, srcV, srcStrideV, dstY, dstWidth, dstU,
+        dstChromaWidth, dstV, dstChromaWidth, srcWidth, srcHeight, rotationMode);
+  }
+
+  /** Helper method for copying a single colour plane. */
+  public static void copyPlane(
+      ByteBuffer src, int srcStride, ByteBuffer dst, int dstStride, int width, int height) {
+    nativeCopyPlane(src, srcStride, dst, dstStride, width, height);
+  }
+
+  /** Converts ABGR little endian (rgba in memory) to I420. */
+  public static void ABGRToI420(ByteBuffer src, int srcStride, ByteBuffer dstY, int dstStrideY,
+      ByteBuffer dstU, int dstStrideU, ByteBuffer dstV, int dstStrideV, int width, int height) {
+    nativeABGRToI420(
+        src, srcStride, dstY, dstStrideY, dstU, dstStrideU, dstV, dstStrideV, width, height);
+  }
+
+  /**
+   * Copies I420 to the I420 dst buffer.
+   * <p> Unlike `libyuv::I420Copy`, this function checks if the height <= 0, so flipping is not
+   * supported.
+   */
+  public static void I420Copy(ByteBuffer srcY, int srcStrideY, ByteBuffer srcU, int srcStrideU,
+      ByteBuffer srcV, int srcStrideV, ByteBuffer dstY, int dstStrideY, ByteBuffer dstU,
+      int dstStrideU, ByteBuffer dstV, int dstStrideV, int width, int height) {
+    if (srcY == null || srcU == null || srcV == null || dstY == null || dstU == null || dstV == null
+        || width <= 0 || height <= 0) {
+      throw new IllegalArgumentException("Invalid I420Copy input arguments");
+    }
+    nativeI420Copy(srcY, srcStrideY, srcU, srcStrideU, srcV, srcStrideV, dstY, dstStrideY, dstU,
+        dstStrideU, dstV, dstStrideV, width, height);
+  }
+
+  public static void I420ToNV12(ByteBuffer srcY, int srcStrideY, ByteBuffer srcU, int srcStrideU,
+      ByteBuffer srcV, int srcStrideV, ByteBuffer dstY, int dstStrideY, ByteBuffer dstUV,
+      int dstStrideUV, int width, int height) {
+    if (srcY == null || srcU == null || srcV == null || dstY == null || dstUV == null || width <= 0
+        || height <= 0) {
+      throw new IllegalArgumentException("Invalid I420ToNV12 input arguments");
+    }
+    nativeI420ToNV12(srcY, srcStrideY, srcU, srcStrideU, srcV, srcStrideV, dstY, dstStrideY, dstUV,
+        dstStrideUV, width, height);
+  }
+
+  public static void I420Rotate(ByteBuffer srcY, int srcStrideY, ByteBuffer srcU, int srcStrideU,
+      ByteBuffer srcV, int srcStrideV, ByteBuffer dstY, int dstStrideY, ByteBuffer dstU,
+      int dstStrideU, ByteBuffer dstV, int dstStrideV, int srcWidth, int srcHeight,
+      int rotationMode) {
+    nativeI420Rotate(srcY, srcStrideY, srcU, srcStrideU, srcV, srcStrideV, dstY, dstStrideY, dstU,
+        dstStrideU, dstV, dstStrideV, srcWidth, srcHeight, rotationMode);
+  }
+
+  private static native void nativeCopyPlane(
+      ByteBuffer src, int srcStride, ByteBuffer dst, int dstStride, int width, int height);
+  private static native void nativeI420Copy(ByteBuffer srcY, int srcStrideY, ByteBuffer srcU,
+      int srcStrideU, ByteBuffer srcV, int srcStrideV, ByteBuffer dstY, int dstStrideY,
+      ByteBuffer dstU, int dstStrideU, ByteBuffer dstV, int dstStrideV, int width, int height);
+  private static native void nativeI420ToNV12(ByteBuffer srcY, int srcStrideY, ByteBuffer srcU,
+      int srcStrideU, ByteBuffer srcV, int srcStrideV, ByteBuffer dstY, int dstStrideY,
+      ByteBuffer dstUV, int dstStrideUV, int width, int height);
+  private static native void nativeI420Rotate(ByteBuffer srcY, int srcStrideY, ByteBuffer srcU,
+      int srcStrideU, ByteBuffer srcV, int srcStrideV, ByteBuffer dstY, int dstStrideY,
+      ByteBuffer dstU, int dstStrideU, ByteBuffer dstV, int dstStrideV, int srcWidth, int srcHeight,
+      int rotationMode);
+  private static native void nativeABGRToI420(ByteBuffer src, int srcStride, ByteBuffer dstY,
+      int dstStrideY, ByteBuffer dstU, int dstStrideU, ByteBuffer dstV, int dstStrideV, int width,
+      int height);
+}
diff --git a/third_party/libwebrtc/sdk/android/api/org/webrtc/audio/AudioDeviceModule.java b/third_party/libwebrtc/sdk/android/api/org/webrtc/audio/AudioDeviceModule.java
new file mode 100644
index 0000000000..502c68cc9a
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/api/org/webrtc/audio/AudioDeviceModule.java
@@ -0,0 +1,38 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc.audio;
+
+/**
+ * This interface is a thin wrapper on top of a native C++ webrtc::AudioDeviceModule (ADM). The
+ * reason for basing it on a native ADM instead of a pure Java interface is that we have two native
+ * Android implementations (OpenSLES and AAudio) that does not make sense to wrap through JNI.
+ *
+ * <p>Note: This class is still under development and may change without notice.
+ */
+public interface AudioDeviceModule {
+  /**
+   * Returns a C++ pointer to a webrtc::AudioDeviceModule. Caller does _not_ take ownership and
+   * lifetime is handled through the release() call.
+   */
+  long getNativeAudioDeviceModulePointer();
+
+  /**
+   * Release resources for this AudioDeviceModule, including native resources. The object should not
+   * be used after this call.
+   */
+  void release();
+
+  /** Control muting/unmuting the speaker. */
+  void setSpeakerMute(boolean mute);
+
+  /** Control muting/unmuting the microphone. */
+  void setMicrophoneMute(boolean mute);
+}
diff --git a/third_party/libwebrtc/sdk/android/api/org/webrtc/audio/JavaAudioDeviceModule.java b/third_party/libwebrtc/sdk/android/api/org/webrtc/audio/JavaAudioDeviceModule.java
new file mode 100644
index 0000000000..d3d57602a8
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/api/org/webrtc/audio/JavaAudioDeviceModule.java
@@ -0,0 +1,436 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc.audio;
+
+import android.content.Context;
+import android.media.AudioAttributes;
+import android.media.AudioDeviceInfo;
+import android.media.AudioManager;
+import android.os.Build;
+import androidx.annotation.RequiresApi;
+import java.util.concurrent.ScheduledExecutorService;
+import org.webrtc.JniCommon;
+import org.webrtc.Logging;
+
+/**
+ * AudioDeviceModule implemented using android.media.AudioRecord as input and
+ * android.media.AudioTrack as output.
+ */
+public class JavaAudioDeviceModule implements AudioDeviceModule {
+  private static final String TAG = "JavaAudioDeviceModule";
+
+  public static Builder builder(Context context) {
+    return new Builder(context);
+  }
+
+  public static class Builder {
+    private final Context context;
+    private ScheduledExecutorService scheduler;
+    private final AudioManager audioManager;
+    private int inputSampleRate;
+    private int outputSampleRate;
+    private int audioSource = WebRtcAudioRecord.DEFAULT_AUDIO_SOURCE;
+    private int audioFormat = WebRtcAudioRecord.DEFAULT_AUDIO_FORMAT;
+    private AudioTrackErrorCallback audioTrackErrorCallback;
+    private AudioRecordErrorCallback audioRecordErrorCallback;
+    private SamplesReadyCallback samplesReadyCallback;
+    private AudioTrackStateCallback audioTrackStateCallback;
+    private AudioRecordStateCallback audioRecordStateCallback;
+    private boolean useHardwareAcousticEchoCanceler = isBuiltInAcousticEchoCancelerSupported();
+    private boolean useHardwareNoiseSuppressor = isBuiltInNoiseSuppressorSupported();
+    private boolean useStereoInput;
+    private boolean useStereoOutput;
+    private AudioAttributes audioAttributes;
+    private boolean useLowLatency;
+    private boolean enableVolumeLogger;
+
+    private Builder(Context context) {
+      this.context = context;
+      this.audioManager = (AudioManager) context.getSystemService(Context.AUDIO_SERVICE);
+      this.inputSampleRate = WebRtcAudioManager.getSampleRate(audioManager);
+      this.outputSampleRate = WebRtcAudioManager.getSampleRate(audioManager);
+      this.useLowLatency = false;
+      this.enableVolumeLogger = true;
+    }
+
+    public Builder setScheduler(ScheduledExecutorService scheduler) {
+      this.scheduler = scheduler;
+      return this;
+    }
+
+    /**
+     * Call this method if the default handling of querying the native sample rate shall be
+     * overridden. Can be useful on some devices where the available Android APIs are known to
+     * return invalid results.
+     */
+    public Builder setSampleRate(int sampleRate) {
+      Logging.d(TAG, "Input/Output sample rate overridden to: " + sampleRate);
+      this.inputSampleRate = sampleRate;
+      this.outputSampleRate = sampleRate;
+      return this;
+    }
+
+    /**
+     * Call this method to specifically override input sample rate.
+     */
+    public Builder setInputSampleRate(int inputSampleRate) {
+      Logging.d(TAG, "Input sample rate overridden to: " + inputSampleRate);
+      this.inputSampleRate = inputSampleRate;
+      return this;
+    }
+
+    /**
+     * Call this method to specifically override output sample rate.
+     */
+    public Builder setOutputSampleRate(int outputSampleRate) {
+      Logging.d(TAG, "Output sample rate overridden to: " + outputSampleRate);
+      this.outputSampleRate = outputSampleRate;
+      return this;
+    }
+
+    /**
+     * Call this to change the audio source. The argument should be one of the values from
+     * android.media.MediaRecorder.AudioSource. The default is AudioSource.VOICE_COMMUNICATION.
+     */
+    public Builder setAudioSource(int audioSource) {
+      this.audioSource = audioSource;
+      return this;
+    }
+
+    /**
+     * Call this to change the audio format. The argument should be one of the values from
+     * android.media.AudioFormat ENCODING_PCM_8BIT, ENCODING_PCM_16BIT or ENCODING_PCM_FLOAT.
+     * Default audio data format is PCM 16 bit per sample.
+     * Guaranteed to be supported by all devices.
+     */
+    public Builder setAudioFormat(int audioFormat) {
+      this.audioFormat = audioFormat;
+      return this;
+    }
+
+    /**
+     * Set a callback to retrieve errors from the AudioTrack.
+     */
+    public Builder setAudioTrackErrorCallback(AudioTrackErrorCallback audioTrackErrorCallback) {
+      this.audioTrackErrorCallback = audioTrackErrorCallback;
+      return this;
+    }
+
+    /**
+     * Set a callback to retrieve errors from the AudioRecord.
+     */
+    public Builder setAudioRecordErrorCallback(AudioRecordErrorCallback audioRecordErrorCallback) {
+      this.audioRecordErrorCallback = audioRecordErrorCallback;
+      return this;
+    }
+
+    /**
+     * Set a callback to listen to the raw audio input from the AudioRecord.
+     */
+    public Builder setSamplesReadyCallback(SamplesReadyCallback samplesReadyCallback) {
+      this.samplesReadyCallback = samplesReadyCallback;
+      return this;
+    }
+
+    /**
+     * Set a callback to retrieve information from the AudioTrack on when audio starts and stop.
+     */
+    public Builder setAudioTrackStateCallback(AudioTrackStateCallback audioTrackStateCallback) {
+      this.audioTrackStateCallback = audioTrackStateCallback;
+      return this;
+    }
+
+    /**
+     * Set a callback to retrieve information from the AudioRecord on when audio starts and stops.
+     */
+    public Builder setAudioRecordStateCallback(AudioRecordStateCallback audioRecordStateCallback) {
+      this.audioRecordStateCallback = audioRecordStateCallback;
+      return this;
+    }
+
+    /**
+     * Control if the built-in HW noise suppressor should be used or not. The default is on if it is
+     * supported. It is possible to query support by calling isBuiltInNoiseSuppressorSupported().
+     */
+    public Builder setUseHardwareNoiseSuppressor(boolean useHardwareNoiseSuppressor) {
+      if (useHardwareNoiseSuppressor && !isBuiltInNoiseSuppressorSupported()) {
+        Logging.e(TAG, "HW NS not supported");
+        useHardwareNoiseSuppressor = false;
+      }
+      this.useHardwareNoiseSuppressor = useHardwareNoiseSuppressor;
+      return this;
+    }
+
+    /**
+     * Control if the built-in HW acoustic echo canceler should be used or not. The default is on if
+     * it is supported. It is possible to query support by calling
+     * isBuiltInAcousticEchoCancelerSupported().
+     */
+    public Builder setUseHardwareAcousticEchoCanceler(boolean useHardwareAcousticEchoCanceler) {
+      if (useHardwareAcousticEchoCanceler && !isBuiltInAcousticEchoCancelerSupported()) {
+        Logging.e(TAG, "HW AEC not supported");
+        useHardwareAcousticEchoCanceler = false;
+      }
+      this.useHardwareAcousticEchoCanceler = useHardwareAcousticEchoCanceler;
+      return this;
+    }
+
+    /**
+     * Control if stereo input should be used or not. The default is mono.
+     */
+    public Builder setUseStereoInput(boolean useStereoInput) {
+      this.useStereoInput = useStereoInput;
+      return this;
+    }
+
+    /**
+     * Control if stereo output should be used or not. The default is mono.
+     */
+    public Builder setUseStereoOutput(boolean useStereoOutput) {
+      this.useStereoOutput = useStereoOutput;
+      return this;
+    }
+
+    /**
+     * Control if the low-latency mode should be used. The default is disabled.
+     */
+    public Builder setUseLowLatency(boolean useLowLatency) {
+      this.useLowLatency = useLowLatency;
+      return this;
+    }
+
+    /**
+     * Set custom {@link AudioAttributes} to use.
+     */
+    public Builder setAudioAttributes(AudioAttributes audioAttributes) {
+      this.audioAttributes = audioAttributes;
+      return this;
+    }
+
+    /** Disables the volume logger on the audio output track. */
+    public Builder setEnableVolumeLogger(boolean enableVolumeLogger) {
+      this.enableVolumeLogger = enableVolumeLogger;
+      return this;
+    }
+
+    /**
+     * Construct an AudioDeviceModule based on the supplied arguments. The caller takes ownership
+     * and is responsible for calling release().
+     */
+    public JavaAudioDeviceModule createAudioDeviceModule() {
+      Logging.d(TAG, "createAudioDeviceModule");
+      if (useHardwareNoiseSuppressor) {
+        Logging.d(TAG, "HW NS will be used.");
+      } else {
+        if (isBuiltInNoiseSuppressorSupported()) {
+          Logging.d(TAG, "Overriding default behavior; now using WebRTC NS!");
+        }
+        Logging.d(TAG, "HW NS will not be used.");
+      }
+      if (useHardwareAcousticEchoCanceler) {
+        Logging.d(TAG, "HW AEC will be used.");
+      } else {
+        if (isBuiltInAcousticEchoCancelerSupported()) {
+          Logging.d(TAG, "Overriding default behavior; now using WebRTC AEC!");
+        }
+        Logging.d(TAG, "HW AEC will not be used.");
+      }
+      // Low-latency mode was introduced in API version 26, see
+      // https://developer.android.com/reference/android/media/AudioTrack#PERFORMANCE_MODE_LOW_LATENCY
+      final int MIN_LOW_LATENCY_SDK_VERSION = 26;
+      if (useLowLatency && Build.VERSION.SDK_INT >= MIN_LOW_LATENCY_SDK_VERSION) {
+        Logging.d(TAG, "Low latency mode will be used.");
+      }
+      ScheduledExecutorService executor = this.scheduler;
+      if (executor == null) {
+        executor = WebRtcAudioRecord.newDefaultScheduler();
+      }
+      final WebRtcAudioRecord audioInput = new WebRtcAudioRecord(context, executor, audioManager,
+          audioSource, audioFormat, audioRecordErrorCallback, audioRecordStateCallback,
+          samplesReadyCallback, useHardwareAcousticEchoCanceler, useHardwareNoiseSuppressor);
+      final WebRtcAudioTrack audioOutput =
+          new WebRtcAudioTrack(context, audioManager, audioAttributes, audioTrackErrorCallback,
+              audioTrackStateCallback, useLowLatency, enableVolumeLogger);
+      return new JavaAudioDeviceModule(context, audioManager, audioInput, audioOutput,
+          inputSampleRate, outputSampleRate, useStereoInput, useStereoOutput);
+    }
+  }
+
+  /* AudioRecord */
+  // Audio recording error handler functions.
+  public enum AudioRecordStartErrorCode {
+    AUDIO_RECORD_START_EXCEPTION,
+    AUDIO_RECORD_START_STATE_MISMATCH,
+  }
+
+  public static interface AudioRecordErrorCallback {
+    void onWebRtcAudioRecordInitError(String errorMessage);
+    void onWebRtcAudioRecordStartError(AudioRecordStartErrorCode errorCode, String errorMessage);
+    void onWebRtcAudioRecordError(String errorMessage);
+  }
+
+  /** Called when audio recording starts and stops. */
+  public static interface AudioRecordStateCallback {
+    void onWebRtcAudioRecordStart();
+    void onWebRtcAudioRecordStop();
+  }
+
+  /**
+   * Contains audio sample information.
+   */
+  public static class AudioSamples {
+    /** See {@link AudioRecord#getAudioFormat()} */
+    private final int audioFormat;
+    /** See {@link AudioRecord#getChannelCount()} */
+    private final int channelCount;
+    /** See {@link AudioRecord#getSampleRate()} */
+    private final int sampleRate;
+
+    private final byte[] data;
+
+    public AudioSamples(int audioFormat, int channelCount, int sampleRate, byte[] data) {
+      this.audioFormat = audioFormat;
+      this.channelCount = channelCount;
+      this.sampleRate = sampleRate;
+      this.data = data;
+    }
+
+    public int getAudioFormat() {
+      return audioFormat;
+    }
+
+    public int getChannelCount() {
+      return channelCount;
+    }
+
+    public int getSampleRate() {
+      return sampleRate;
+    }
+
+    public byte[] getData() {
+      return data;
+    }
+  }
+
+  /** Called when new audio samples are ready. This should only be set for debug purposes */
+  public static interface SamplesReadyCallback {
+    void onWebRtcAudioRecordSamplesReady(AudioSamples samples);
+  }
+
+  /* AudioTrack */
+  // Audio playout/track error handler functions.
+  public enum AudioTrackStartErrorCode {
+    AUDIO_TRACK_START_EXCEPTION,
+    AUDIO_TRACK_START_STATE_MISMATCH,
+  }
+
+  public static interface AudioTrackErrorCallback {
+    void onWebRtcAudioTrackInitError(String errorMessage);
+    void onWebRtcAudioTrackStartError(AudioTrackStartErrorCode errorCode, String errorMessage);
+    void onWebRtcAudioTrackError(String errorMessage);
+  }
+
+  /** Called when audio playout starts and stops. */
+  public static interface AudioTrackStateCallback {
+    void onWebRtcAudioTrackStart();
+    void onWebRtcAudioTrackStop();
+  }
+
+  /**
+   * Returns true if the device supports built-in HW AEC, and the UUID is approved (some UUIDs can
+   * be excluded).
+   */
+  public static boolean isBuiltInAcousticEchoCancelerSupported() {
+    return WebRtcAudioEffects.isAcousticEchoCancelerSupported();
+  }
+
+  /**
+   * Returns true if the device supports built-in HW NS, and the UUID is approved (some UUIDs can be
+   * excluded).
+   */
+  public static boolean isBuiltInNoiseSuppressorSupported() {
+    return WebRtcAudioEffects.isNoiseSuppressorSupported();
+  }
+
+  private final Context context;
+  private final AudioManager audioManager;
+  private final WebRtcAudioRecord audioInput;
+  private final WebRtcAudioTrack audioOutput;
+  private final int inputSampleRate;
+  private final int outputSampleRate;
+  private final boolean useStereoInput;
+  private final boolean useStereoOutput;
+
+  private final Object nativeLock = new Object();
+  private long nativeAudioDeviceModule;
+
+  private JavaAudioDeviceModule(Context context, AudioManager audioManager,
+      WebRtcAudioRecord audioInput, WebRtcAudioTrack audioOutput, int inputSampleRate,
+      int outputSampleRate, boolean useStereoInput, boolean useStereoOutput) {
+    this.context = context;
+    this.audioManager = audioManager;
+    this.audioInput = audioInput;
+    this.audioOutput = audioOutput;
+    this.inputSampleRate = inputSampleRate;
+    this.outputSampleRate = outputSampleRate;
+    this.useStereoInput = useStereoInput;
+    this.useStereoOutput = useStereoOutput;
+  }
+
+  @Override
+  public long getNativeAudioDeviceModulePointer() {
+    synchronized (nativeLock) {
+      if (nativeAudioDeviceModule == 0) {
+        nativeAudioDeviceModule = nativeCreateAudioDeviceModule(context, audioManager, audioInput,
+            audioOutput, inputSampleRate, outputSampleRate, useStereoInput, useStereoOutput);
+      }
+      return nativeAudioDeviceModule;
+    }
+  }
+
+  @Override
+  public void release() {
+    synchronized (nativeLock) {
+      if (nativeAudioDeviceModule != 0) {
+        JniCommon.nativeReleaseRef(nativeAudioDeviceModule);
+        nativeAudioDeviceModule = 0;
+      }
+    }
+  }
+
+  @Override
+  public void setSpeakerMute(boolean mute) {
+    Logging.d(TAG, "setSpeakerMute: " + mute);
+    audioOutput.setSpeakerMute(mute);
+  }
+
+  @Override
+  public void setMicrophoneMute(boolean mute) {
+    Logging.d(TAG, "setMicrophoneMute: " + mute);
+    audioInput.setMicrophoneMute(mute);
+  }
+
+  /**
+   * Start to prefer a specific {@link AudioDeviceInfo} device for recording. Typically this should
+   * only be used if a client gives an explicit option for choosing a physical device to record
+   * from. Otherwise the best-matching device for other parameters will be used. Calling after
+   * recording is started may cause a temporary interruption if the audio routing changes.
+   */
+  @RequiresApi(Build.VERSION_CODES.M)
+  public void setPreferredInputDevice(AudioDeviceInfo preferredInputDevice) {
+    Logging.d(TAG, "setPreferredInputDevice: " + preferredInputDevice);
+    audioInput.setPreferredDevice(preferredInputDevice);
+  }
+
+  private static native long nativeCreateAudioDeviceModule(Context context,
+      AudioManager audioManager, WebRtcAudioRecord audioInput, WebRtcAudioTrack audioOutput,
+      int inputSampleRate, int outputSampleRate, boolean useStereoInput, boolean useStereoOutput);
+}
diff --git a/third_party/libwebrtc/sdk/android/api/org/webrtc/audio/LegacyAudioDeviceModule.java b/third_party/libwebrtc/sdk/android/api/org/webrtc/audio/LegacyAudioDeviceModule.java
new file mode 100644
index 0000000000..de0d0d61f9
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/api/org/webrtc/audio/LegacyAudioDeviceModule.java
@@ -0,0 +1,46 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc.audio;
+
+import org.webrtc.voiceengine.WebRtcAudioRecord;
+import org.webrtc.voiceengine.WebRtcAudioTrack;
+
+/**
+ * This class represents the legacy AudioDeviceModule that is currently hardcoded into C++ WebRTC.
+ * It will return a null native AudioDeviceModule pointer, leading to an internal object being
+ * created inside WebRTC that is controlled by static calls to the classes under the voiceengine
+ * package. Please use the new JavaAudioDeviceModule instead of this class.
+ */
+@Deprecated
+public class LegacyAudioDeviceModule implements AudioDeviceModule {
+  @Override
+  public long getNativeAudioDeviceModulePointer() {
+    // Returning a null pointer will make WebRTC construct the built-in legacy AudioDeviceModule for
+    // Android internally.
+    return 0;
+  }
+
+  @Override
+  public void release() {
+    // All control for this ADM goes through static global methods and the C++ object is owned
+    // internally by WebRTC.
+  }
+
+  @Override
+  public void setSpeakerMute(boolean mute) {
+    WebRtcAudioTrack.setSpeakerMute(mute);
+  }
+
+  @Override
+  public void setMicrophoneMute(boolean mute) {
+    WebRtcAudioRecord.setMicrophoneMute(mute);
+  }
+}
diff --git a/third_party/libwebrtc/sdk/android/src/java/org/webrtc/AndroidVideoDecoder.java b/third_party/libwebrtc/sdk/android/src/java/org/webrtc/AndroidVideoDecoder.java
new file mode 100644
index 0000000000..ad40898e4c
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/src/java/org/webrtc/AndroidVideoDecoder.java
@@ -0,0 +1,684 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+import android.media.MediaCodec;
+import android.media.MediaCodecInfo.CodecCapabilities;
+import android.media.MediaFormat;
+import android.os.SystemClock;
+import android.view.Surface;
+import androidx.annotation.Nullable;
+import java.io.IOException;
+import java.nio.ByteBuffer;
+import java.util.concurrent.BlockingDeque;
+import java.util.concurrent.LinkedBlockingDeque;
+import java.util.concurrent.TimeUnit;
+import org.webrtc.ThreadUtils.ThreadChecker;
+
+/**
+ * Android hardware video decoder.
+ */
+@SuppressWarnings("deprecation")
+// Cannot support API 16 without using deprecated methods.
+// TODO(sakal): Rename to MediaCodecVideoDecoder once the deprecated implementation is removed.
+class AndroidVideoDecoder implements VideoDecoder, VideoSink {
+  private static final String TAG = "AndroidVideoDecoder";
+
+  // TODO(magjed): Use MediaFormat.KEY_* constants when part of the public API.
+  private static final String MEDIA_FORMAT_KEY_STRIDE = "stride";
+  private static final String MEDIA_FORMAT_KEY_SLICE_HEIGHT = "slice-height";
+  private static final String MEDIA_FORMAT_KEY_CROP_LEFT = "crop-left";
+  private static final String MEDIA_FORMAT_KEY_CROP_RIGHT = "crop-right";
+  private static final String MEDIA_FORMAT_KEY_CROP_TOP = "crop-top";
+  private static final String MEDIA_FORMAT_KEY_CROP_BOTTOM = "crop-bottom";
+
+  // MediaCodec.release() occasionally hangs.  Release stops waiting and reports failure after
+  // this timeout.
+  private static final int MEDIA_CODEC_RELEASE_TIMEOUT_MS = 5000;
+
+  // WebRTC queues input frames quickly in the beginning on the call. Wait for input buffers with a
+  // long timeout (500 ms) to prevent this from causing the codec to return an error.
+  private static final int DEQUEUE_INPUT_TIMEOUT_US = 500000;
+
+  // Dequeuing an output buffer will block until a buffer is available (up to 100 milliseconds).
+  // If this timeout is exceeded, the output thread will unblock and check if the decoder is still
+  // running.  If it is, it will block on dequeue again.  Otherwise, it will stop and release the
+  // MediaCodec.
+  private static final int DEQUEUE_OUTPUT_BUFFER_TIMEOUT_US = 100000;
+
+  private final MediaCodecWrapperFactory mediaCodecWrapperFactory;
+  private final String codecName;
+  private final VideoCodecMimeType codecType;
+
+  private static class FrameInfo {
+    final long decodeStartTimeMs;
+    final int rotation;
+
+    FrameInfo(long decodeStartTimeMs, int rotation) {
+      this.decodeStartTimeMs = decodeStartTimeMs;
+      this.rotation = rotation;
+    }
+  }
+
+  private final BlockingDeque<FrameInfo> frameInfos;
+  private int colorFormat;
+
+  // Output thread runs a loop which polls MediaCodec for decoded output buffers.  It reformats
+  // those buffers into VideoFrames and delivers them to the callback.  Variable is set on decoder
+  // thread and is immutable while the codec is running.
+  @Nullable private Thread outputThread;
+
+  // Checker that ensures work is run on the output thread.
+  private ThreadChecker outputThreadChecker;
+
+  // Checker that ensures work is run on the decoder thread.  The decoder thread is owned by the
+  // caller and must be used to call initDecode, decode, and release.
+  private ThreadChecker decoderThreadChecker;
+
+  private volatile boolean running;
+  @Nullable private volatile Exception shutdownException;
+
+  // Dimensions (width, height, stride, and sliceHeight) may be accessed by either the decode thread
+  // or the output thread.  Accesses should be protected with this lock.
+  private final Object dimensionLock = new Object();
+  private int width;
+  private int height;
+  private int stride;
+  private int sliceHeight;
+
+  // Whether the decoder has finished the first frame.  The codec may not change output dimensions
+  // after delivering the first frame.  Only accessed on the output thread while the decoder is
+  // running.
+  private boolean hasDecodedFirstFrame;
+  // Whether the decoder has seen a key frame.  The first frame must be a key frame.  Only accessed
+  // on the decoder thread.
+  private boolean keyFrameRequired;
+
+  private final @Nullable EglBase.Context sharedContext;
+  // Valid and immutable while the decoder is running.
+  @Nullable private SurfaceTextureHelper surfaceTextureHelper;
+  @Nullable private Surface surface;
+
+  private static class DecodedTextureMetadata {
+    final long presentationTimestampUs;
+    final Integer decodeTimeMs;
+
+    DecodedTextureMetadata(long presentationTimestampUs, Integer decodeTimeMs) {
+      this.presentationTimestampUs = presentationTimestampUs;
+      this.decodeTimeMs = decodeTimeMs;
+    }
+  }
+
+  // Metadata for the last frame rendered to the texture.
+  private final Object renderedTextureMetadataLock = new Object();
+  @Nullable private DecodedTextureMetadata renderedTextureMetadata;
+
+  // Decoding proceeds asynchronously.  This callback returns decoded frames to the caller.  Valid
+  // and immutable while the decoder is running.
+  @Nullable private Callback callback;
+
+  // Valid and immutable while the decoder is running.
+  @Nullable private MediaCodecWrapper codec;
+
+  AndroidVideoDecoder(MediaCodecWrapperFactory mediaCodecWrapperFactory, String codecName,
+      VideoCodecMimeType codecType, int colorFormat, @Nullable EglBase.Context sharedContext) {
+    if (!isSupportedColorFormat(colorFormat)) {
+      throw new IllegalArgumentException("Unsupported color format: " + colorFormat);
+    }
+    Logging.d(TAG,
+        "ctor name: " + codecName + " type: " + codecType + " color format: " + colorFormat
+            + " context: " + sharedContext);
+    this.mediaCodecWrapperFactory = mediaCodecWrapperFactory;
+    this.codecName = codecName;
+    this.codecType = codecType;
+    this.colorFormat = colorFormat;
+    this.sharedContext = sharedContext;
+    this.frameInfos = new LinkedBlockingDeque<>();
+  }
+
+  @Override
+  public VideoCodecStatus initDecode(Settings settings, Callback callback) {
+    this.decoderThreadChecker = new ThreadChecker();
+
+    this.callback = callback;
+    if (sharedContext != null) {
+      surfaceTextureHelper = createSurfaceTextureHelper();
+      surface = new Surface(surfaceTextureHelper.getSurfaceTexture());
+      surfaceTextureHelper.startListening(this);
+    }
+    return initDecodeInternal(settings.width, settings.height);
+  }
+
+  // Internal variant is used when restarting the codec due to reconfiguration.
+  private VideoCodecStatus initDecodeInternal(int width, int height) {
+    decoderThreadChecker.checkIsOnValidThread();
+    Logging.d(TAG,
+        "initDecodeInternal name: " + codecName + " type: " + codecType + " width: " + width
+            + " height: " + height);
+    if (outputThread != null) {
+      Logging.e(TAG, "initDecodeInternal called while the codec is already running");
+      return VideoCodecStatus.FALLBACK_SOFTWARE;
+    }
+
+    // Note:  it is not necessary to initialize dimensions under the lock, since the output thread
+    // is not running.
+    this.width = width;
+    this.height = height;
+
+    stride = width;
+    sliceHeight = height;
+    hasDecodedFirstFrame = false;
+    keyFrameRequired = true;
+
+    try {
+      codec = mediaCodecWrapperFactory.createByCodecName(codecName);
+    } catch (IOException | IllegalArgumentException | IllegalStateException e) {
+      Logging.e(TAG, "Cannot create media decoder " + codecName);
+      return VideoCodecStatus.FALLBACK_SOFTWARE;
+    }
+    try {
+      MediaFormat format = MediaFormat.createVideoFormat(codecType.mimeType(), width, height);
+      if (sharedContext == null) {
+        format.setInteger(MediaFormat.KEY_COLOR_FORMAT, colorFormat);
+      }
+      codec.configure(format, surface, null, 0);
+      codec.start();
+    } catch (IllegalStateException | IllegalArgumentException e) {
+      Logging.e(TAG, "initDecode failed", e);
+      release();
+      return VideoCodecStatus.FALLBACK_SOFTWARE;
+    }
+    running = true;
+    outputThread = createOutputThread();
+    outputThread.start();
+
+    Logging.d(TAG, "initDecodeInternal done");
+    return VideoCodecStatus.OK;
+  }
+
+  @Override
+  public VideoCodecStatus decode(EncodedImage frame, DecodeInfo info) {
+    decoderThreadChecker.checkIsOnValidThread();
+    if (codec == null || callback == null) {
+      Logging.d(TAG, "decode uninitalized, codec: " + (codec != null) + ", callback: " + callback);
+      return VideoCodecStatus.UNINITIALIZED;
+    }
+
+    if (frame.buffer == null) {
+      Logging.e(TAG, "decode() - no input data");
+      return VideoCodecStatus.ERR_PARAMETER;
+    }
+
+    int size = frame.buffer.remaining();
+    if (size == 0) {
+      Logging.e(TAG, "decode() - input buffer empty");
+      return VideoCodecStatus.ERR_PARAMETER;
+    }
+
+    // Load dimensions from shared memory under the dimension lock.
+    final int width;
+    final int height;
+    synchronized (dimensionLock) {
+      width = this.width;
+      height = this.height;
+    }
+
+    // Check if the resolution changed and reset the codec if necessary.
+    if (frame.encodedWidth * frame.encodedHeight > 0
+        && (frame.encodedWidth != width || frame.encodedHeight != height)) {
+      VideoCodecStatus status = reinitDecode(frame.encodedWidth, frame.encodedHeight);
+      if (status != VideoCodecStatus.OK) {
+        return status;
+      }
+    }
+
+    if (keyFrameRequired) {
+      // Need to process a key frame first.
+      if (frame.frameType != EncodedImage.FrameType.VideoFrameKey) {
+        Logging.e(TAG, "decode() - key frame required first");
+        return VideoCodecStatus.NO_OUTPUT;
+      }
+    }
+
+    int index;
+    try {
+      index = codec.dequeueInputBuffer(DEQUEUE_INPUT_TIMEOUT_US);
+    } catch (IllegalStateException e) {
+      Logging.e(TAG, "dequeueInputBuffer failed", e);
+      return VideoCodecStatus.ERROR;
+    }
+    if (index < 0) {
+      // Decoder is falling behind.  No input buffers available.
+      // The decoder can't simply drop frames; it might lose a key frame.
+      Logging.e(TAG, "decode() - no HW buffers available; decoder falling behind");
+      return VideoCodecStatus.ERROR;
+    }
+
+    ByteBuffer buffer;
+    try {
+      buffer = codec.getInputBuffer(index);
+    } catch (IllegalStateException e) {
+      Logging.e(TAG, "getInputBuffer with index=" + index + " failed", e);
+      return VideoCodecStatus.ERROR;
+    }
+
+    if (buffer.capacity() < size) {
+      Logging.e(TAG, "decode() - HW buffer too small");
+      return VideoCodecStatus.ERROR;
+    }
+    buffer.put(frame.buffer);
+
+    frameInfos.offer(new FrameInfo(SystemClock.elapsedRealtime(), frame.rotation));
+    try {
+      codec.queueInputBuffer(index, 0 /* offset */, size,
+          TimeUnit.NANOSECONDS.toMicros(frame.captureTimeNs), 0 /* flags */);
+    } catch (IllegalStateException e) {
+      Logging.e(TAG, "queueInputBuffer failed", e);
+      frameInfos.pollLast();
+      return VideoCodecStatus.ERROR;
+    }
+    if (keyFrameRequired) {
+      keyFrameRequired = false;
+    }
+    return VideoCodecStatus.OK;
+  }
+
+  @Override
+  public String getImplementationName() {
+    return codecName;
+  }
+
+  @Override
+  public VideoCodecStatus release() {
+    // TODO(sakal): This is not called on the correct thread but is still called synchronously.
+    // Re-enable the check once this is called on the correct thread.
+    // decoderThreadChecker.checkIsOnValidThread();
+    Logging.d(TAG, "release");
+    VideoCodecStatus status = releaseInternal();
+    if (surface != null) {
+      releaseSurface();
+      surface = null;
+      surfaceTextureHelper.stopListening();
+      surfaceTextureHelper.dispose();
+      surfaceTextureHelper = null;
+    }
+    synchronized (renderedTextureMetadataLock) {
+      renderedTextureMetadata = null;
+    }
+    callback = null;
+    frameInfos.clear();
+    return status;
+  }
+
+  // Internal variant is used when restarting the codec due to reconfiguration.
+  private VideoCodecStatus releaseInternal() {
+    if (!running) {
+      Logging.d(TAG, "release: Decoder is not running.");
+      return VideoCodecStatus.OK;
+    }
+    try {
+      // The outputThread actually stops and releases the codec once running is false.
+      running = false;
+      if (!ThreadUtils.joinUninterruptibly(outputThread, MEDIA_CODEC_RELEASE_TIMEOUT_MS)) {
+        // Log an exception to capture the stack trace and turn it into a TIMEOUT error.
+        Logging.e(TAG, "Media decoder release timeout", new RuntimeException());
+        return VideoCodecStatus.TIMEOUT;
+      }
+      if (shutdownException != null) {
+        // Log the exception and turn it into an error.  Wrap the exception in a new exception to
+        // capture both the output thread's stack trace and this thread's stack trace.
+        Logging.e(TAG, "Media decoder release error", new RuntimeException(shutdownException));
+        shutdownException = null;
+        return VideoCodecStatus.ERROR;
+      }
+    } finally {
+      codec = null;
+      outputThread = null;
+    }
+    return VideoCodecStatus.OK;
+  }
+
+  private VideoCodecStatus reinitDecode(int newWidth, int newHeight) {
+    decoderThreadChecker.checkIsOnValidThread();
+    VideoCodecStatus status = releaseInternal();
+    if (status != VideoCodecStatus.OK) {
+      return status;
+    }
+    return initDecodeInternal(newWidth, newHeight);
+  }
+
+  private Thread createOutputThread() {
+    return new Thread("AndroidVideoDecoder.outputThread") {
+      @Override
+      public void run() {
+        outputThreadChecker = new ThreadChecker();
+        while (running) {
+          deliverDecodedFrame();
+        }
+        releaseCodecOnOutputThread();
+      }
+    };
+  }
+
+  // Visible for testing.
+  protected void deliverDecodedFrame() {
+    outputThreadChecker.checkIsOnValidThread();
+    try {
+      MediaCodec.BufferInfo info = new MediaCodec.BufferInfo();
+      // Block until an output buffer is available (up to 100 milliseconds).  If the timeout is
+      // exceeded, deliverDecodedFrame() will be called again on the next iteration of the output
+      // thread's loop.  Blocking here prevents the output thread from busy-waiting while the codec
+      // is idle.
+      int index = codec.dequeueOutputBuffer(info, DEQUEUE_OUTPUT_BUFFER_TIMEOUT_US);
+      if (index == MediaCodec.INFO_OUTPUT_FORMAT_CHANGED) {
+        reformat(codec.getOutputFormat());
+        return;
+      }
+
+      if (index < 0) {
+        Logging.v(TAG, "dequeueOutputBuffer returned " + index);
+        return;
+      }
+
+      FrameInfo frameInfo = frameInfos.poll();
+      Integer decodeTimeMs = null;
+      int rotation = 0;
+      if (frameInfo != null) {
+        decodeTimeMs = (int) (SystemClock.elapsedRealtime() - frameInfo.decodeStartTimeMs);
+        rotation = frameInfo.rotation;
+      }
+
+      hasDecodedFirstFrame = true;
+
+      if (surfaceTextureHelper != null) {
+        deliverTextureFrame(index, info, rotation, decodeTimeMs);
+      } else {
+        deliverByteFrame(index, info, rotation, decodeTimeMs);
+      }
+
+    } catch (IllegalStateException e) {
+      Logging.e(TAG, "deliverDecodedFrame failed", e);
+    }
+  }
+
+  private void deliverTextureFrame(final int index, final MediaCodec.BufferInfo info,
+      final int rotation, final Integer decodeTimeMs) {
+    // Load dimensions from shared memory under the dimension lock.
+    final int width;
+    final int height;
+    synchronized (dimensionLock) {
+      width = this.width;
+      height = this.height;
+    }
+
+    synchronized (renderedTextureMetadataLock) {
+      if (renderedTextureMetadata != null) {
+        codec.releaseOutputBuffer(index, false);
+        return; // We are still waiting for texture for the previous frame, drop this one.
+      }
+      surfaceTextureHelper.setTextureSize(width, height);
+      surfaceTextureHelper.setFrameRotation(rotation);
+      renderedTextureMetadata = new DecodedTextureMetadata(info.presentationTimeUs, decodeTimeMs);
+      codec.releaseOutputBuffer(index, /* render= */ true);
+    }
+  }
+
+  @Override
+  public void onFrame(VideoFrame frame) {
+    final VideoFrame newFrame;
+    final Integer decodeTimeMs;
+    final long timestampNs;
+    synchronized (renderedTextureMetadataLock) {
+      if (renderedTextureMetadata == null) {
+        throw new IllegalStateException(
+            "Rendered texture metadata was null in onTextureFrameAvailable.");
+      }
+      timestampNs = renderedTextureMetadata.presentationTimestampUs * 1000;
+      decodeTimeMs = renderedTextureMetadata.decodeTimeMs;
+      renderedTextureMetadata = null;
+    }
+    // Change timestamp of frame.
+    final VideoFrame frameWithModifiedTimeStamp =
+        new VideoFrame(frame.getBuffer(), frame.getRotation(), timestampNs);
+    callback.onDecodedFrame(frameWithModifiedTimeStamp, decodeTimeMs, null /* qp */);
+  }
+
+  private void deliverByteFrame(
+      int index, MediaCodec.BufferInfo info, int rotation, Integer decodeTimeMs) {
+    // Load dimensions from shared memory under the dimension lock.
+    int width;
+    int height;
+    int stride;
+    int sliceHeight;
+    synchronized (dimensionLock) {
+      width = this.width;
+      height = this.height;
+      stride = this.stride;
+      sliceHeight = this.sliceHeight;
+    }
+
+    // Output must be at least width * height bytes for Y channel, plus (width / 2) * (height / 2)
+    // bytes for each of the U and V channels.
+    if (info.size < width * height * 3 / 2) {
+      Logging.e(TAG, "Insufficient output buffer size: " + info.size);
+      return;
+    }
+
+    if (info.size < stride * height * 3 / 2 && sliceHeight == height && stride > width) {
+      // Some codecs (Exynos) report an incorrect stride.  Correct it here.
+      // Expected size == stride * height * 3 / 2.  A bit of algebra gives the correct stride as
+      // 2 * size / (3 * height).
+      stride = info.size * 2 / (height * 3);
+    }
+
+    ByteBuffer buffer = codec.getOutputBuffer(index);
+    buffer.position(info.offset);
+    buffer.limit(info.offset + info.size);
+    buffer = buffer.slice();
+
+    final VideoFrame.Buffer frameBuffer;
+    if (colorFormat == CodecCapabilities.COLOR_FormatYUV420Planar) {
+      frameBuffer = copyI420Buffer(buffer, stride, sliceHeight, width, height);
+    } else {
+      // All other supported color formats are NV12.
+      frameBuffer = copyNV12ToI420Buffer(buffer, stride, sliceHeight, width, height);
+    }
+    codec.releaseOutputBuffer(index, /* render= */ false);
+
+    long presentationTimeNs = info.presentationTimeUs * 1000;
+    VideoFrame frame = new VideoFrame(frameBuffer, rotation, presentationTimeNs);
+
+    // Note that qp is parsed on the C++ side.
+    callback.onDecodedFrame(frame, decodeTimeMs, null /* qp */);
+    frame.release();
+  }
+
+  private VideoFrame.Buffer copyNV12ToI420Buffer(
+      ByteBuffer buffer, int stride, int sliceHeight, int width, int height) {
+    // toI420 copies the buffer.
+    return new NV12Buffer(width, height, stride, sliceHeight, buffer, null /* releaseCallback */)
+        .toI420();
+  }
+
+  private VideoFrame.Buffer copyI420Buffer(
+      ByteBuffer buffer, int stride, int sliceHeight, int width, int height) {
+    if (stride % 2 != 0) {
+      throw new AssertionError("Stride is not divisible by two: " + stride);
+    }
+
+    // Note that the case with odd `sliceHeight` is handled in a special way.
+    // The chroma height contained in the payload is rounded down instead of
+    // up, making it one row less than what we expect in WebRTC. Therefore, we
+    // have to duplicate the last chroma rows for this case. Also, the offset
+    // between the Y plane and the U plane is unintuitive for this case. See
+    // http://bugs.webrtc.org/6651 for more info.
+    final int chromaWidth = (width + 1) / 2;
+    final int chromaHeight = (sliceHeight % 2 == 0) ? (height + 1) / 2 : height / 2;
+
+    final int uvStride = stride / 2;
+
+    final int yPos = 0;
+    final int yEnd = yPos + stride * height;
+    final int uPos = yPos + stride * sliceHeight;
+    final int uEnd = uPos + uvStride * chromaHeight;
+    final int vPos = uPos + uvStride * sliceHeight / 2;
+    final int vEnd = vPos + uvStride * chromaHeight;
+
+    VideoFrame.I420Buffer frameBuffer = allocateI420Buffer(width, height);
+
+    buffer.limit(yEnd);
+    buffer.position(yPos);
+    copyPlane(
+        buffer.slice(), stride, frameBuffer.getDataY(), frameBuffer.getStrideY(), width, height);
+
+    buffer.limit(uEnd);
+    buffer.position(uPos);
+    copyPlane(buffer.slice(), uvStride, frameBuffer.getDataU(), frameBuffer.getStrideU(),
+        chromaWidth, chromaHeight);
+    if (sliceHeight % 2 == 1) {
+      buffer.position(uPos + uvStride * (chromaHeight - 1)); // Seek to beginning of last full row.
+
+      ByteBuffer dataU = frameBuffer.getDataU();
+      dataU.position(frameBuffer.getStrideU() * chromaHeight); // Seek to beginning of last row.
+      dataU.put(buffer); // Copy the last row.
+    }
+
+    buffer.limit(vEnd);
+    buffer.position(vPos);
+    copyPlane(buffer.slice(), uvStride, frameBuffer.getDataV(), frameBuffer.getStrideV(),
+        chromaWidth, chromaHeight);
+    if (sliceHeight % 2 == 1) {
+      buffer.position(vPos + uvStride * (chromaHeight - 1)); // Seek to beginning of last full row.
+
+      ByteBuffer dataV = frameBuffer.getDataV();
+      dataV.position(frameBuffer.getStrideV() * chromaHeight); // Seek to beginning of last row.
+      dataV.put(buffer); // Copy the last row.
+    }
+
+    return frameBuffer;
+  }
+
+  private void reformat(MediaFormat format) {
+    outputThreadChecker.checkIsOnValidThread();
+    Logging.d(TAG, "Decoder format changed: " + format.toString());
+    final int newWidth;
+    final int newHeight;
+    if (format.containsKey(MEDIA_FORMAT_KEY_CROP_LEFT)
+        && format.containsKey(MEDIA_FORMAT_KEY_CROP_RIGHT)
+        && format.containsKey(MEDIA_FORMAT_KEY_CROP_BOTTOM)
+        && format.containsKey(MEDIA_FORMAT_KEY_CROP_TOP)) {
+      newWidth = 1 + format.getInteger(MEDIA_FORMAT_KEY_CROP_RIGHT)
+          - format.getInteger(MEDIA_FORMAT_KEY_CROP_LEFT);
+      newHeight = 1 + format.getInteger(MEDIA_FORMAT_KEY_CROP_BOTTOM)
+          - format.getInteger(MEDIA_FORMAT_KEY_CROP_TOP);
+    } else {
+      newWidth = format.getInteger(MediaFormat.KEY_WIDTH);
+      newHeight = format.getInteger(MediaFormat.KEY_HEIGHT);
+    }
+    // Compare to existing width, height, and save values under the dimension lock.
+    synchronized (dimensionLock) {
+      if (newWidth != width || newHeight != height) {
+        if (hasDecodedFirstFrame) {
+          stopOnOutputThread(new RuntimeException("Unexpected size change. "
+              + "Configured " + width + "*" + height + ". "
+              + "New " + newWidth + "*" + newHeight));
+          return;
+        } else if (newWidth <= 0 || newHeight <= 0) {
+          Logging.w(TAG,
+              "Unexpected format dimensions. Configured " + width + "*" + height + ". "
+                  + "New " + newWidth + "*" + newHeight + ". Skip it");
+          return;
+        }
+        width = newWidth;
+        height = newHeight;
+      }
+    }
+
+    // Note:  texture mode ignores colorFormat.  Hence, if the texture helper is non-null, skip
+    // color format updates.
+    if (surfaceTextureHelper == null && format.containsKey(MediaFormat.KEY_COLOR_FORMAT)) {
+      colorFormat = format.getInteger(MediaFormat.KEY_COLOR_FORMAT);
+      Logging.d(TAG, "Color: 0x" + Integer.toHexString(colorFormat));
+      if (!isSupportedColorFormat(colorFormat)) {
+        stopOnOutputThread(new IllegalStateException("Unsupported color format: " + colorFormat));
+        return;
+      }
+    }
+
+    // Save stride and sliceHeight under the dimension lock.
+    synchronized (dimensionLock) {
+      if (format.containsKey(MEDIA_FORMAT_KEY_STRIDE)) {
+        stride = format.getInteger(MEDIA_FORMAT_KEY_STRIDE);
+      }
+      if (format.containsKey(MEDIA_FORMAT_KEY_SLICE_HEIGHT)) {
+        sliceHeight = format.getInteger(MEDIA_FORMAT_KEY_SLICE_HEIGHT);
+      }
+      Logging.d(TAG, "Frame stride and slice height: " + stride + " x " + sliceHeight);
+      stride = Math.max(width, stride);
+      sliceHeight = Math.max(height, sliceHeight);
+    }
+  }
+
+  private void releaseCodecOnOutputThread() {
+    outputThreadChecker.checkIsOnValidThread();
+    Logging.d(TAG, "Releasing MediaCodec on output thread");
+    try {
+      codec.stop();
+    } catch (Exception e) {
+      Logging.e(TAG, "Media decoder stop failed", e);
+    }
+    try {
+      codec.release();
+    } catch (Exception e) {
+      Logging.e(TAG, "Media decoder release failed", e);
+      // Propagate exceptions caught during release back to the main thread.
+      shutdownException = e;
+    }
+    Logging.d(TAG, "Release on output thread done");
+  }
+
+  private void stopOnOutputThread(Exception e) {
+    outputThreadChecker.checkIsOnValidThread();
+    running = false;
+    shutdownException = e;
+  }
+
+  private boolean isSupportedColorFormat(int colorFormat) {
+    for (int supported : MediaCodecUtils.DECODER_COLOR_FORMATS) {
+      if (supported == colorFormat) {
+        return true;
+      }
+    }
+    return false;
+  }
+
+  // Visible for testing.
+  protected SurfaceTextureHelper createSurfaceTextureHelper() {
+    return SurfaceTextureHelper.create("decoder-texture-thread", sharedContext);
+  }
+
+  // Visible for testing.
+  // TODO(sakal): Remove once Robolectric commit fa991a0 has been rolled to WebRTC.
+  protected void releaseSurface() {
+    surface.release();
+  }
+
+  // Visible for testing.
+  protected VideoFrame.I420Buffer allocateI420Buffer(int width, int height) {
+    return JavaI420Buffer.allocate(width, height);
+  }
+
+  // Visible for testing.
+  protected void copyPlane(
+      ByteBuffer src, int srcStride, ByteBuffer dst, int dstStride, int width, int height) {
+    YuvHelper.copyPlane(src, srcStride, dst, dstStride, width, height);
+  }
+}
diff --git a/third_party/libwebrtc/sdk/android/src/java/org/webrtc/BaseBitrateAdjuster.java b/third_party/libwebrtc/sdk/android/src/java/org/webrtc/BaseBitrateAdjuster.java
new file mode 100644
index 0000000000..3b5f5d2931
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/src/java/org/webrtc/BaseBitrateAdjuster.java
@@ -0,0 +1,38 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+/** BitrateAdjuster that tracks bitrate and framerate but does not adjust them. */
+class BaseBitrateAdjuster implements BitrateAdjuster {
+  protected int targetBitrateBps;
+  protected double targetFramerateFps;
+
+  @Override
+  public void setTargets(int targetBitrateBps, double targetFramerateFps) {
+    this.targetBitrateBps = targetBitrateBps;
+    this.targetFramerateFps = targetFramerateFps;
+  }
+
+  @Override
+  public void reportEncodedFrame(int size) {
+    // No op.
+  }
+
+  @Override
+  public int getAdjustedBitrateBps() {
+    return targetBitrateBps;
+  }
+
+  @Override
+  public double getAdjustedFramerateFps() {
+    return targetFramerateFps;
+  }
+}
diff --git a/third_party/libwebrtc/sdk/android/src/java/org/webrtc/BitrateAdjuster.java b/third_party/libwebrtc/sdk/android/src/java/org/webrtc/BitrateAdjuster.java
new file mode 100644
index 0000000000..bfa08bad89
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/src/java/org/webrtc/BitrateAdjuster.java
@@ -0,0 +1,31 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+/** Object that adjusts the bitrate of a hardware codec. */
+interface BitrateAdjuster {
+  /**
+   * Sets the target bitrate in bits per second and framerate in frames per second.
+   */
+  void setTargets(int targetBitrateBps, double targetFramerateFps);
+
+  /**
+   * Should be used to report the size of an encoded frame to the bitrate adjuster. Use
+   * getAdjustedBitrateBps to get the updated bitrate after calling this method.
+   */
+  void reportEncodedFrame(int size);
+
+  /** Gets the current bitrate. */
+  int getAdjustedBitrateBps();
+
+  /** Gets the current framerate. */
+  double getAdjustedFramerateFps();
+}
diff --git a/third_party/libwebrtc/sdk/android/src/java/org/webrtc/CalledByNative.java b/third_party/libwebrtc/sdk/android/src/java/org/webrtc/CalledByNative.java
new file mode 100644
index 0000000000..9b410ceaef
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/src/java/org/webrtc/CalledByNative.java
@@ -0,0 +1,29 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+import java.lang.annotation.ElementType;
+import java.lang.annotation.Retention;
+import java.lang.annotation.RetentionPolicy;
+import java.lang.annotation.Target;
+
+/**
+ * @CalledByNative is used by the JNI generator to create the necessary JNI
+ * bindings and expose this method to native code.
+ */
+@Target({ElementType.CONSTRUCTOR, ElementType.METHOD})
+@Retention(RetentionPolicy.CLASS)
+public @interface CalledByNative {
+  /*
+   *  If present, tells which inner class the method belongs to.
+   */
+  public String value() default "";
+}
diff --git a/third_party/libwebrtc/sdk/android/src/java/org/webrtc/CalledByNativeUnchecked.java b/third_party/libwebrtc/sdk/android/src/java/org/webrtc/CalledByNativeUnchecked.java
new file mode 100644
index 0000000000..8a00a7fadb
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/src/java/org/webrtc/CalledByNativeUnchecked.java
@@ -0,0 +1,33 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+import java.lang.annotation.ElementType;
+import java.lang.annotation.Retention;
+import java.lang.annotation.RetentionPolicy;
+import java.lang.annotation.Target;
+
+/**
+ *  @CalledByNativeUnchecked is used to generate JNI bindings that do not check for exceptions.
+ *  It only makes sense to use this annotation on methods that declare a throws... spec.
+ *  However, note that the exception received native side maybe an 'unchecked' (RuntimeExpception)
+ *  such as NullPointerException, so the native code should differentiate these cases.
+ *  Usage of this should be very rare; where possible handle exceptions in the Java side and use a
+ *  return value to indicate success / failure.
+ */
+@Target(ElementType.METHOD)
+@Retention(RetentionPolicy.CLASS)
+public @interface CalledByNativeUnchecked {
+  /*
+   *  If present, tells which inner class the method belongs to.
+   */
+  public String value() default "";
+}
diff --git a/third_party/libwebrtc/sdk/android/src/java/org/webrtc/Camera1Session.java b/third_party/libwebrtc/sdk/android/src/java/org/webrtc/Camera1Session.java
new file mode 100644
index 0000000000..a54f7201b2
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/src/java/org/webrtc/Camera1Session.java
@@ -0,0 +1,340 @@
+/*
+ *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+import android.content.Context;
+import android.hardware.Camera;
+import android.os.Handler;
+import android.os.SystemClock;
+import java.io.IOException;
+import java.nio.ByteBuffer;
+import java.util.List;
+import java.util.concurrent.TimeUnit;
+import org.webrtc.CameraEnumerationAndroid.CaptureFormat;
+
+@SuppressWarnings("deprecation")
+class Camera1Session implements CameraSession {
+  private static final String TAG = "Camera1Session";
+  private static final int NUMBER_OF_CAPTURE_BUFFERS = 3;
+
+  private static final Histogram camera1StartTimeMsHistogram =
+      Histogram.createCounts("WebRTC.Android.Camera1.StartTimeMs", 1, 10000, 50);
+  private static final Histogram camera1StopTimeMsHistogram =
+      Histogram.createCounts("WebRTC.Android.Camera1.StopTimeMs", 1, 10000, 50);
+  private static final Histogram camera1ResolutionHistogram = Histogram.createEnumeration(
+      "WebRTC.Android.Camera1.Resolution", CameraEnumerationAndroid.COMMON_RESOLUTIONS.size());
+
+  private static enum SessionState { RUNNING, STOPPED }
+
+  private final Handler cameraThreadHandler;
+  private final Events events;
+  private final boolean captureToTexture;
+  private final Context applicationContext;
+  private final SurfaceTextureHelper surfaceTextureHelper;
+  private final int cameraId;
+  private final Camera camera;
+  private final Camera.CameraInfo info;
+  private final CaptureFormat captureFormat;
+  // Used only for stats. Only used on the camera thread.
+  private final long constructionTimeNs; // Construction time of this class.
+
+  private SessionState state;
+  private boolean firstFrameReported;
+
+  // TODO(titovartem) make correct fix during webrtc:9175
+  @SuppressWarnings("ByteBufferBackingArray")
+  public static void create(final CreateSessionCallback callback, final Events events,
+      final boolean captureToTexture, final Context applicationContext,
+      final SurfaceTextureHelper surfaceTextureHelper, final String cameraName,
+      final int width, final int height, final int framerate) {
+    final long constructionTimeNs = System.nanoTime();
+    Logging.d(TAG, "Open camera " + cameraName);
+    events.onCameraOpening();
+
+    final int cameraId;
+    try {
+      cameraId = Camera1Enumerator.getCameraIndex(cameraName);
+    } catch (IllegalArgumentException e) {
+      callback.onFailure(FailureType.ERROR, e.getMessage());
+      return;
+    }
+
+    final Camera camera;
+    try {
+      camera = Camera.open(cameraId);
+    } catch (RuntimeException e) {
+      callback.onFailure(FailureType.ERROR, e.getMessage());
+      return;
+    }
+
+    if (camera == null) {
+      callback.onFailure(
+          FailureType.ERROR, "Camera.open returned null for camera id = " + cameraId);
+      return;
+    }
+
+    try {
+      camera.setPreviewTexture(surfaceTextureHelper.getSurfaceTexture());
+    } catch (IOException | RuntimeException e) {
+      camera.release();
+      callback.onFailure(FailureType.ERROR, e.getMessage());
+      return;
+    }
+
+    final Camera.CameraInfo info = new Camera.CameraInfo();
+    Camera.getCameraInfo(cameraId, info);
+
+    final CaptureFormat captureFormat;
+    try {
+      final Camera.Parameters parameters = camera.getParameters();
+      captureFormat = findClosestCaptureFormat(parameters, width, height, framerate);
+      final Size pictureSize = findClosestPictureSize(parameters, width, height);
+      updateCameraParameters(camera, parameters, captureFormat, pictureSize, captureToTexture);
+    } catch (RuntimeException e) {
+      camera.release();
+      callback.onFailure(FailureType.ERROR, e.getMessage());
+      return;
+    }
+
+    if (!captureToTexture) {
+      final int frameSize = captureFormat.frameSize();
+      for (int i = 0; i < NUMBER_OF_CAPTURE_BUFFERS; ++i) {
+        final ByteBuffer buffer = ByteBuffer.allocateDirect(frameSize);
+        camera.addCallbackBuffer(buffer.array());
+      }
+    }
+
+    // Calculate orientation manually and send it as CVO instead.
+    try {
+      camera.setDisplayOrientation(0 /* degrees */);
+    } catch (RuntimeException e) {
+      camera.release();
+      callback.onFailure(FailureType.ERROR, e.getMessage());
+      return;
+    }
+
+    callback.onDone(new Camera1Session(events, captureToTexture, applicationContext,
+        surfaceTextureHelper, cameraId, camera, info, captureFormat, constructionTimeNs));
+  }
+
+  private static void updateCameraParameters(Camera camera, Camera.Parameters parameters,
+      CaptureFormat captureFormat, Size pictureSize, boolean captureToTexture) {
+    final List<String> focusModes = parameters.getSupportedFocusModes();
+
+    parameters.setPreviewFpsRange(captureFormat.framerate.min, captureFormat.framerate.max);
+    parameters.setPreviewSize(captureFormat.width, captureFormat.height);
+    parameters.setPictureSize(pictureSize.width, pictureSize.height);
+    if (!captureToTexture) {
+      parameters.setPreviewFormat(captureFormat.imageFormat);
+    }
+
+    if (parameters.isVideoStabilizationSupported()) {
+      parameters.setVideoStabilization(true);
+    }
+    if (focusModes != null && focusModes.contains(Camera.Parameters.FOCUS_MODE_CONTINUOUS_VIDEO)) {
+      parameters.setFocusMode(Camera.Parameters.FOCUS_MODE_CONTINUOUS_VIDEO);
+    }
+    camera.setParameters(parameters);
+  }
+
+  private static CaptureFormat findClosestCaptureFormat(
+      Camera.Parameters parameters, int width, int height, int framerate) {
+    // Find closest supported format for `width` x `height` @ `framerate`.
+    final List<CaptureFormat.FramerateRange> supportedFramerates =
+        Camera1Enumerator.convertFramerates(parameters.getSupportedPreviewFpsRange());
+    Logging.d(TAG, "Available fps ranges: " + supportedFramerates);
+
+    final CaptureFormat.FramerateRange fpsRange =
+        CameraEnumerationAndroid.getClosestSupportedFramerateRange(supportedFramerates, framerate);
+
+    final Size previewSize = CameraEnumerationAndroid.getClosestSupportedSize(
+        Camera1Enumerator.convertSizes(parameters.getSupportedPreviewSizes()), width, height);
+    CameraEnumerationAndroid.reportCameraResolution(camera1ResolutionHistogram, previewSize);
+
+    return new CaptureFormat(previewSize.width, previewSize.height, fpsRange);
+  }
+
+  private static Size findClosestPictureSize(Camera.Parameters parameters, int width, int height) {
+    return CameraEnumerationAndroid.getClosestSupportedSize(
+        Camera1Enumerator.convertSizes(parameters.getSupportedPictureSizes()), width, height);
+  }
+
+  private Camera1Session(Events events, boolean captureToTexture, Context applicationContext,
+      SurfaceTextureHelper surfaceTextureHelper, int cameraId, Camera camera,
+      Camera.CameraInfo info, CaptureFormat captureFormat, long constructionTimeNs) {
+    Logging.d(TAG, "Create new camera1 session on camera " + cameraId);
+
+    this.cameraThreadHandler = new Handler();
+    this.events = events;
+    this.captureToTexture = captureToTexture;
+    this.applicationContext = applicationContext;
+    this.surfaceTextureHelper = surfaceTextureHelper;
+    this.cameraId = cameraId;
+    this.camera = camera;
+    this.info = info;
+    this.captureFormat = captureFormat;
+    this.constructionTimeNs = constructionTimeNs;
+
+    surfaceTextureHelper.setTextureSize(captureFormat.width, captureFormat.height);
+
+    startCapturing();
+  }
+
+  @Override
+  public void stop() {
+    Logging.d(TAG, "Stop camera1 session on camera " + cameraId);
+    checkIsOnCameraThread();
+    if (state != SessionState.STOPPED) {
+      final long stopStartTime = System.nanoTime();
+      stopInternal();
+      final int stopTimeMs = (int) TimeUnit.NANOSECONDS.toMillis(System.nanoTime() - stopStartTime);
+      camera1StopTimeMsHistogram.addSample(stopTimeMs);
+    }
+  }
+
+  private void startCapturing() {
+    Logging.d(TAG, "Start capturing");
+    checkIsOnCameraThread();
+
+    state = SessionState.RUNNING;
+
+    camera.setErrorCallback(new Camera.ErrorCallback() {
+      @Override
+      public void onError(int error, Camera camera) {
+        String errorMessage;
+        if (error == Camera.CAMERA_ERROR_SERVER_DIED) {
+          errorMessage = "Camera server died!";
+        } else {
+          errorMessage = "Camera error: " + error;
+        }
+        Logging.e(TAG, errorMessage);
+        stopInternal();
+        if (error == Camera.CAMERA_ERROR_EVICTED) {
+          events.onCameraDisconnected(Camera1Session.this);
+        } else {
+          events.onCameraError(Camera1Session.this, errorMessage);
+        }
+      }
+    });
+
+    if (captureToTexture) {
+      listenForTextureFrames();
+    } else {
+      listenForBytebufferFrames();
+    }
+    try {
+      camera.startPreview();
+    } catch (RuntimeException e) {
+      stopInternal();
+      events.onCameraError(this, e.getMessage());
+    }
+  }
+
+  private void stopInternal() {
+    Logging.d(TAG, "Stop internal");
+    checkIsOnCameraThread();
+    if (state == SessionState.STOPPED) {
+      Logging.d(TAG, "Camera is already stopped");
+      return;
+    }
+
+    state = SessionState.STOPPED;
+    surfaceTextureHelper.stopListening();
+    // Note: stopPreview or other driver code might deadlock. Deadlock in
+    // Camera._stopPreview(Native Method) has been observed on
+    // Nexus 5 (hammerhead), OS version LMY48I.
+    camera.stopPreview();
+    camera.release();
+    events.onCameraClosed(this);
+    Logging.d(TAG, "Stop done");
+  }
+
+  private void listenForTextureFrames() {
+    surfaceTextureHelper.startListening((VideoFrame frame) -> {
+      checkIsOnCameraThread();
+
+      if (state != SessionState.RUNNING) {
+        Logging.d(TAG, "Texture frame captured but camera is no longer running.");
+        return;
+      }
+
+      if (!firstFrameReported) {
+        final int startTimeMs =
+            (int) TimeUnit.NANOSECONDS.toMillis(System.nanoTime() - constructionTimeNs);
+        camera1StartTimeMsHistogram.addSample(startTimeMs);
+        firstFrameReported = true;
+      }
+
+      // Undo the mirror that the OS "helps" us with.
+      // http://developer.android.com/reference/android/hardware/Camera.html#setDisplayOrientation(int)
+      final VideoFrame modifiedFrame =
+          new VideoFrame(CameraSession.createTextureBufferWithModifiedTransformMatrix(
+                             (TextureBufferImpl) frame.getBuffer(),
+                             /* mirror= */ info.facing == Camera.CameraInfo.CAMERA_FACING_FRONT,
+                             /* rotation= */ 0),
+              /* rotation= */ getFrameOrientation(), frame.getTimestampNs());
+      events.onFrameCaptured(Camera1Session.this, modifiedFrame);
+      modifiedFrame.release();
+    });
+  }
+
+  private void listenForBytebufferFrames() {
+    camera.setPreviewCallbackWithBuffer(new Camera.PreviewCallback() {
+      @Override
+      public void onPreviewFrame(final byte[] data, Camera callbackCamera) {
+        checkIsOnCameraThread();
+
+        if (callbackCamera != camera) {
+          Logging.e(TAG, "Callback from a different camera. This should never happen.");
+          return;
+        }
+
+        if (state != SessionState.RUNNING) {
+          Logging.d(TAG, "Bytebuffer frame captured but camera is no longer running.");
+          return;
+        }
+
+        final long captureTimeNs = TimeUnit.MILLISECONDS.toNanos(SystemClock.elapsedRealtime());
+
+        if (!firstFrameReported) {
+          final int startTimeMs =
+              (int) TimeUnit.NANOSECONDS.toMillis(System.nanoTime() - constructionTimeNs);
+          camera1StartTimeMsHistogram.addSample(startTimeMs);
+          firstFrameReported = true;
+        }
+
+        VideoFrame.Buffer frameBuffer = new NV21Buffer(
+            data, captureFormat.width, captureFormat.height, () -> cameraThreadHandler.post(() -> {
+              if (state == SessionState.RUNNING) {
+                camera.addCallbackBuffer(data);
+              }
+            }));
+        final VideoFrame frame = new VideoFrame(frameBuffer, getFrameOrientation(), captureTimeNs);
+        events.onFrameCaptured(Camera1Session.this, frame);
+        frame.release();
+      }
+    });
+  }
+
+  private int getFrameOrientation() {
+    int rotation = CameraSession.getDeviceOrientation(applicationContext);
+    if (info.facing == Camera.CameraInfo.CAMERA_FACING_BACK) {
+      rotation = 360 - rotation;
+    }
+    return (info.orientation + rotation) % 360;
+  }
+
+  private void checkIsOnCameraThread() {
+    if (Thread.currentThread() != cameraThreadHandler.getLooper().getThread()) {
+      throw new IllegalStateException("Wrong thread");
+    }
+  }
+}
diff --git a/third_party/libwebrtc/sdk/android/src/java/org/webrtc/Camera2Session.java b/third_party/libwebrtc/sdk/android/src/java/org/webrtc/Camera2Session.java
new file mode 100644
index 0000000000..d5ee80c73e
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/src/java/org/webrtc/Camera2Session.java
@@ -0,0 +1,428 @@
+/*
+ *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+import android.annotation.SuppressLint;
+import android.content.Context;
+import android.hardware.camera2.CameraAccessException;
+import android.hardware.camera2.CameraCaptureSession;
+import android.hardware.camera2.CameraCharacteristics;
+import android.hardware.camera2.CameraDevice;
+import android.hardware.camera2.CameraManager;
+import android.hardware.camera2.CameraMetadata;
+import android.hardware.camera2.CaptureFailure;
+import android.hardware.camera2.CaptureRequest;
+import android.os.Handler;
+import android.util.Range;
+import android.view.Surface;
+import androidx.annotation.Nullable;
+import java.util.Arrays;
+import java.util.List;
+import java.util.concurrent.TimeUnit;
+import org.webrtc.CameraEnumerationAndroid.CaptureFormat;
+
+class Camera2Session implements CameraSession {
+  private static final String TAG = "Camera2Session";
+
+  private static final Histogram camera2StartTimeMsHistogram =
+      Histogram.createCounts("WebRTC.Android.Camera2.StartTimeMs", 1, 10000, 50);
+  private static final Histogram camera2StopTimeMsHistogram =
+      Histogram.createCounts("WebRTC.Android.Camera2.StopTimeMs", 1, 10000, 50);
+  private static final Histogram camera2ResolutionHistogram = Histogram.createEnumeration(
+      "WebRTC.Android.Camera2.Resolution", CameraEnumerationAndroid.COMMON_RESOLUTIONS.size());
+
+  private static enum SessionState { RUNNING, STOPPED }
+
+  private final Handler cameraThreadHandler;
+  private final CreateSessionCallback callback;
+  private final Events events;
+  private final Context applicationContext;
+  private final CameraManager cameraManager;
+  private final SurfaceTextureHelper surfaceTextureHelper;
+  private final String cameraId;
+  private final int width;
+  private final int height;
+  private final int framerate;
+
+  // Initialized at start
+  private CameraCharacteristics cameraCharacteristics;
+  private int cameraOrientation;
+  private boolean isCameraFrontFacing;
+  private int fpsUnitFactor;
+  private CaptureFormat captureFormat;
+
+  // Initialized when camera opens
+  @Nullable private CameraDevice cameraDevice;
+  @Nullable private Surface surface;
+
+  // Initialized when capture session is created
+  @Nullable private CameraCaptureSession captureSession;
+
+  // State
+  private SessionState state = SessionState.RUNNING;
+  private boolean firstFrameReported;
+
+  // Used only for stats. Only used on the camera thread.
+  private final long constructionTimeNs; // Construction time of this class.
+
+  private class CameraStateCallback extends CameraDevice.StateCallback {
+    private String getErrorDescription(int errorCode) {
+      switch (errorCode) {
+        case CameraDevice.StateCallback.ERROR_CAMERA_DEVICE:
+          return "Camera device has encountered a fatal error.";
+        case CameraDevice.StateCallback.ERROR_CAMERA_DISABLED:
+          return "Camera device could not be opened due to a device policy.";
+        case CameraDevice.StateCallback.ERROR_CAMERA_IN_USE:
+          return "Camera device is in use already.";
+        case CameraDevice.StateCallback.ERROR_CAMERA_SERVICE:
+          return "Camera service has encountered a fatal error.";
+        case CameraDevice.StateCallback.ERROR_MAX_CAMERAS_IN_USE:
+          return "Camera device could not be opened because"
+              + " there are too many other open camera devices.";
+        default:
+          return "Unknown camera error: " + errorCode;
+      }
+    }
+
+    @Override
+    public void onDisconnected(CameraDevice camera) {
+      checkIsOnCameraThread();
+      final boolean startFailure = (captureSession == null) && (state != SessionState.STOPPED);
+      state = SessionState.STOPPED;
+      stopInternal();
+      if (startFailure) {
+        callback.onFailure(FailureType.DISCONNECTED, "Camera disconnected / evicted.");
+      } else {
+        events.onCameraDisconnected(Camera2Session.this);
+      }
+    }
+
+    @Override
+    public void onError(CameraDevice camera, int errorCode) {
+      checkIsOnCameraThread();
+      reportError(getErrorDescription(errorCode));
+    }
+
+    @Override
+    public void onOpened(CameraDevice camera) {
+      checkIsOnCameraThread();
+
+      Logging.d(TAG, "Camera opened.");
+      cameraDevice = camera;
+
+      surfaceTextureHelper.setTextureSize(captureFormat.width, captureFormat.height);
+      surface = new Surface(surfaceTextureHelper.getSurfaceTexture());
+      try {
+        camera.createCaptureSession(
+            Arrays.asList(surface), new CaptureSessionCallback(), cameraThreadHandler);
+      } catch (CameraAccessException e) {
+        reportError("Failed to create capture session. " + e);
+        return;
+      }
+    }
+
+    @Override
+    public void onClosed(CameraDevice camera) {
+      checkIsOnCameraThread();
+
+      Logging.d(TAG, "Camera device closed.");
+      events.onCameraClosed(Camera2Session.this);
+    }
+  }
+
+  private class CaptureSessionCallback extends CameraCaptureSession.StateCallback {
+    @Override
+    public void onConfigureFailed(CameraCaptureSession session) {
+      checkIsOnCameraThread();
+      session.close();
+      reportError("Failed to configure capture session.");
+    }
+
+    @Override
+    public void onConfigured(CameraCaptureSession session) {
+      checkIsOnCameraThread();
+      Logging.d(TAG, "Camera capture session configured.");
+      captureSession = session;
+      try {
+        /*
+         * The viable options for video capture requests are:
+         * TEMPLATE_PREVIEW: High frame rate is given priority over the highest-quality
+         *   post-processing.
+         * TEMPLATE_RECORD: Stable frame rate is used, and post-processing is set for recording
+         *   quality.
+         */
+        final CaptureRequest.Builder captureRequestBuilder =
+            cameraDevice.createCaptureRequest(CameraDevice.TEMPLATE_RECORD);
+        // Set auto exposure fps range.
+        captureRequestBuilder.set(CaptureRequest.CONTROL_AE_TARGET_FPS_RANGE,
+            new Range<Integer>(captureFormat.framerate.min / fpsUnitFactor,
+                captureFormat.framerate.max / fpsUnitFactor));
+        captureRequestBuilder.set(
+            CaptureRequest.CONTROL_AE_MODE, CaptureRequest.CONTROL_AE_MODE_ON);
+        captureRequestBuilder.set(CaptureRequest.CONTROL_AE_LOCK, false);
+        chooseStabilizationMode(captureRequestBuilder);
+        chooseFocusMode(captureRequestBuilder);
+
+        captureRequestBuilder.addTarget(surface);
+        session.setRepeatingRequest(
+            captureRequestBuilder.build(), new CameraCaptureCallback(), cameraThreadHandler);
+      } catch (CameraAccessException e) {
+        reportError("Failed to start capture request. " + e);
+        return;
+      }
+
+      surfaceTextureHelper.startListening((VideoFrame frame) -> {
+        checkIsOnCameraThread();
+
+        if (state != SessionState.RUNNING) {
+          Logging.d(TAG, "Texture frame captured but camera is no longer running.");
+          return;
+        }
+
+        if (!firstFrameReported) {
+          firstFrameReported = true;
+          final int startTimeMs =
+              (int) TimeUnit.NANOSECONDS.toMillis(System.nanoTime() - constructionTimeNs);
+          camera2StartTimeMsHistogram.addSample(startTimeMs);
+        }
+
+        // Undo the mirror that the OS "helps" us with.
+        // http://developer.android.com/reference/android/hardware/Camera.html#setDisplayOrientation(int)
+        // Also, undo camera orientation, we report it as rotation instead.
+        final VideoFrame modifiedFrame =
+            new VideoFrame(CameraSession.createTextureBufferWithModifiedTransformMatrix(
+                               (TextureBufferImpl) frame.getBuffer(),
+                               /* mirror= */ isCameraFrontFacing,
+                               /* rotation= */ -cameraOrientation),
+                /* rotation= */ getFrameOrientation(), frame.getTimestampNs());
+        events.onFrameCaptured(Camera2Session.this, modifiedFrame);
+        modifiedFrame.release();
+      });
+      Logging.d(TAG, "Camera device successfully started.");
+      callback.onDone(Camera2Session.this);
+    }
+
+    // Prefers optical stabilization over software stabilization if available. Only enables one of
+    // the stabilization modes at a time because having both enabled can cause strange results.
+    private void chooseStabilizationMode(CaptureRequest.Builder captureRequestBuilder) {
+      final int[] availableOpticalStabilization = cameraCharacteristics.get(
+          CameraCharacteristics.LENS_INFO_AVAILABLE_OPTICAL_STABILIZATION);
+      if (availableOpticalStabilization != null) {
+        for (int mode : availableOpticalStabilization) {
+          if (mode == CaptureRequest.LENS_OPTICAL_STABILIZATION_MODE_ON) {
+            captureRequestBuilder.set(CaptureRequest.LENS_OPTICAL_STABILIZATION_MODE,
+                CaptureRequest.LENS_OPTICAL_STABILIZATION_MODE_ON);
+            captureRequestBuilder.set(CaptureRequest.CONTROL_VIDEO_STABILIZATION_MODE,
+                CaptureRequest.CONTROL_VIDEO_STABILIZATION_MODE_OFF);
+            Logging.d(TAG, "Using optical stabilization.");
+            return;
+          }
+        }
+      }
+      // If no optical mode is available, try software.
+      final int[] availableVideoStabilization = cameraCharacteristics.get(
+          CameraCharacteristics.CONTROL_AVAILABLE_VIDEO_STABILIZATION_MODES);
+      if (availableVideoStabilization != null) {
+        for (int mode : availableVideoStabilization) {
+          if (mode == CaptureRequest.CONTROL_VIDEO_STABILIZATION_MODE_ON) {
+            captureRequestBuilder.set(CaptureRequest.CONTROL_VIDEO_STABILIZATION_MODE,
+                CaptureRequest.CONTROL_VIDEO_STABILIZATION_MODE_ON);
+            captureRequestBuilder.set(CaptureRequest.LENS_OPTICAL_STABILIZATION_MODE,
+                CaptureRequest.LENS_OPTICAL_STABILIZATION_MODE_OFF);
+            Logging.d(TAG, "Using video stabilization.");
+            return;
+          }
+        }
+      }
+      Logging.d(TAG, "Stabilization not available.");
+    }
+
+    private void chooseFocusMode(CaptureRequest.Builder captureRequestBuilder) {
+      final int[] availableFocusModes =
+          cameraCharacteristics.get(CameraCharacteristics.CONTROL_AF_AVAILABLE_MODES);
+      for (int mode : availableFocusModes) {
+        if (mode == CaptureRequest.CONTROL_AF_MODE_CONTINUOUS_VIDEO) {
+          captureRequestBuilder.set(
+              CaptureRequest.CONTROL_AF_MODE, CaptureRequest.CONTROL_AF_MODE_CONTINUOUS_VIDEO);
+          Logging.d(TAG, "Using continuous video auto-focus.");
+          return;
+        }
+      }
+      Logging.d(TAG, "Auto-focus is not available.");
+    }
+  }
+
+  private static class CameraCaptureCallback extends CameraCaptureSession.CaptureCallback {
+    @Override
+    public void onCaptureFailed(
+        CameraCaptureSession session, CaptureRequest request, CaptureFailure failure) {
+      Logging.d(TAG, "Capture failed: " + failure);
+    }
+  }
+
+  public static void create(CreateSessionCallback callback, Events events,
+      Context applicationContext, CameraManager cameraManager,
+      SurfaceTextureHelper surfaceTextureHelper, String cameraId, int width, int height,
+      int framerate) {
+    new Camera2Session(callback, events, applicationContext, cameraManager, surfaceTextureHelper,
+        cameraId, width, height, framerate);
+  }
+
+  private Camera2Session(CreateSessionCallback callback, Events events, Context applicationContext,
+      CameraManager cameraManager, SurfaceTextureHelper surfaceTextureHelper, String cameraId,
+      int width, int height, int framerate) {
+    Logging.d(TAG, "Create new camera2 session on camera " + cameraId);
+
+    constructionTimeNs = System.nanoTime();
+
+    this.cameraThreadHandler = new Handler();
+    this.callback = callback;
+    this.events = events;
+    this.applicationContext = applicationContext;
+    this.cameraManager = cameraManager;
+    this.surfaceTextureHelper = surfaceTextureHelper;
+    this.cameraId = cameraId;
+    this.width = width;
+    this.height = height;
+    this.framerate = framerate;
+
+    start();
+  }
+
+  private void start() {
+    checkIsOnCameraThread();
+    Logging.d(TAG, "start");
+
+    try {
+      cameraCharacteristics = cameraManager.getCameraCharacteristics(cameraId);
+    } catch (CameraAccessException | IllegalArgumentException e) {
+      reportError("getCameraCharacteristics(): " + e.getMessage());
+      return;
+    }
+    cameraOrientation = cameraCharacteristics.get(CameraCharacteristics.SENSOR_ORIENTATION);
+    isCameraFrontFacing = cameraCharacteristics.get(CameraCharacteristics.LENS_FACING)
+        == CameraMetadata.LENS_FACING_FRONT;
+
+    findCaptureFormat();
+
+    if (captureFormat == null) {
+      // findCaptureFormat reports an error already.
+      return;
+    }
+
+    openCamera();
+  }
+
+  private void findCaptureFormat() {
+    checkIsOnCameraThread();
+
+    Range<Integer>[] fpsRanges =
+        cameraCharacteristics.get(CameraCharacteristics.CONTROL_AE_AVAILABLE_TARGET_FPS_RANGES);
+    fpsUnitFactor = Camera2Enumerator.getFpsUnitFactor(fpsRanges);
+    List<CaptureFormat.FramerateRange> framerateRanges =
+        Camera2Enumerator.convertFramerates(fpsRanges, fpsUnitFactor);
+    List<Size> sizes = Camera2Enumerator.getSupportedSizes(cameraCharacteristics);
+    Logging.d(TAG, "Available preview sizes: " + sizes);
+    Logging.d(TAG, "Available fps ranges: " + framerateRanges);
+
+    if (framerateRanges.isEmpty() || sizes.isEmpty()) {
+      reportError("No supported capture formats.");
+      return;
+    }
+
+    final CaptureFormat.FramerateRange bestFpsRange =
+        CameraEnumerationAndroid.getClosestSupportedFramerateRange(framerateRanges, framerate);
+
+    final Size bestSize = CameraEnumerationAndroid.getClosestSupportedSize(sizes, width, height);
+    CameraEnumerationAndroid.reportCameraResolution(camera2ResolutionHistogram, bestSize);
+
+    captureFormat = new CaptureFormat(bestSize.width, bestSize.height, bestFpsRange);
+    Logging.d(TAG, "Using capture format: " + captureFormat);
+  }
+
+  @SuppressLint("MissingPermission")
+  private void openCamera() {
+    checkIsOnCameraThread();
+
+    Logging.d(TAG, "Opening camera " + cameraId);
+    events.onCameraOpening();
+
+    try {
+      cameraManager.openCamera(cameraId, new CameraStateCallback(), cameraThreadHandler);
+    } catch (CameraAccessException | IllegalArgumentException | SecurityException e) {
+      reportError("Failed to open camera: " + e);
+      return;
+    }
+  }
+
+  @Override
+  public void stop() {
+    Logging.d(TAG, "Stop camera2 session on camera " + cameraId);
+    checkIsOnCameraThread();
+    if (state != SessionState.STOPPED) {
+      final long stopStartTime = System.nanoTime();
+      state = SessionState.STOPPED;
+      stopInternal();
+      final int stopTimeMs = (int) TimeUnit.NANOSECONDS.toMillis(System.nanoTime() - stopStartTime);
+      camera2StopTimeMsHistogram.addSample(stopTimeMs);
+    }
+  }
+
+  private void stopInternal() {
+    Logging.d(TAG, "Stop internal");
+    checkIsOnCameraThread();
+
+    surfaceTextureHelper.stopListening();
+
+    if (captureSession != null) {
+      captureSession.close();
+      captureSession = null;
+    }
+    if (surface != null) {
+      surface.release();
+      surface = null;
+    }
+    if (cameraDevice != null) {
+      cameraDevice.close();
+      cameraDevice = null;
+    }
+
+    Logging.d(TAG, "Stop done");
+  }
+
+  private void reportError(String error) {
+    checkIsOnCameraThread();
+    Logging.e(TAG, "Error: " + error);
+
+    final boolean startFailure = (captureSession == null) && (state != SessionState.STOPPED);
+    state = SessionState.STOPPED;
+    stopInternal();
+    if (startFailure) {
+      callback.onFailure(FailureType.ERROR, error);
+    } else {
+      events.onCameraError(this, error);
+    }
+  }
+
+  private int getFrameOrientation() {
+    int rotation = CameraSession.getDeviceOrientation(applicationContext);
+    if (!isCameraFrontFacing) {
+      rotation = 360 - rotation;
+    }
+    return (cameraOrientation + rotation) % 360;
+  }
+
+  private void checkIsOnCameraThread() {
+    if (Thread.currentThread() != cameraThreadHandler.getLooper().getThread()) {
+      throw new IllegalStateException("Wrong thread");
+    }
+  }
+}
diff --git a/third_party/libwebrtc/sdk/android/src/java/org/webrtc/CameraCapturer.java b/third_party/libwebrtc/sdk/android/src/java/org/webrtc/CameraCapturer.java
new file mode 100644
index 0000000000..1922a529e2
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/src/java/org/webrtc/CameraCapturer.java
@@ -0,0 +1,458 @@
+/*
+ *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+import android.content.Context;
+import android.os.Handler;
+import android.os.Looper;
+import androidx.annotation.Nullable;
+import java.util.Arrays;
+import java.util.List;
+
+@SuppressWarnings("deprecation")
+abstract class CameraCapturer implements CameraVideoCapturer {
+  enum SwitchState {
+    IDLE, // No switch requested.
+    PENDING, // Waiting for previous capture session to open.
+    IN_PROGRESS, // Waiting for new switched capture session to start.
+  }
+
+  private static final String TAG = "CameraCapturer";
+  private final static int MAX_OPEN_CAMERA_ATTEMPTS = 3;
+  private final static int OPEN_CAMERA_DELAY_MS = 500;
+  private final static int OPEN_CAMERA_TIMEOUT = 10000;
+
+  private final CameraEnumerator cameraEnumerator;
+  private final CameraEventsHandler eventsHandler;
+  private final Handler uiThreadHandler;
+
+  @Nullable
+  private final CameraSession.CreateSessionCallback createSessionCallback =
+      new CameraSession.CreateSessionCallback() {
+        @Override
+        public void onDone(CameraSession session) {
+          checkIsOnCameraThread();
+          Logging.d(TAG, "Create session done. Switch state: " + switchState);
+          uiThreadHandler.removeCallbacks(openCameraTimeoutRunnable);
+          synchronized (stateLock) {
+            capturerObserver.onCapturerStarted(true /* success */);
+            sessionOpening = false;
+            currentSession = session;
+            cameraStatistics = new CameraStatistics(surfaceHelper, eventsHandler);
+            firstFrameObserved = false;
+            stateLock.notifyAll();
+
+            if (switchState == SwitchState.IN_PROGRESS) {
+              switchState = SwitchState.IDLE;
+              if (switchEventsHandler != null) {
+                switchEventsHandler.onCameraSwitchDone(cameraEnumerator.isFrontFacing(cameraName));
+                switchEventsHandler = null;
+              }
+            } else if (switchState == SwitchState.PENDING) {
+              String selectedCameraName = pendingCameraName;
+              pendingCameraName = null;
+              switchState = SwitchState.IDLE;
+              switchCameraInternal(switchEventsHandler, selectedCameraName);
+            }
+          }
+        }
+
+        @Override
+        public void onFailure(CameraSession.FailureType failureType, String error) {
+          checkIsOnCameraThread();
+          uiThreadHandler.removeCallbacks(openCameraTimeoutRunnable);
+          synchronized (stateLock) {
+            capturerObserver.onCapturerStarted(false /* success */);
+            openAttemptsRemaining--;
+
+            if (openAttemptsRemaining <= 0) {
+              Logging.w(TAG, "Opening camera failed, passing: " + error);
+              sessionOpening = false;
+              stateLock.notifyAll();
+
+              if (switchState != SwitchState.IDLE) {
+                if (switchEventsHandler != null) {
+                  switchEventsHandler.onCameraSwitchError(error);
+                  switchEventsHandler = null;
+                }
+                switchState = SwitchState.IDLE;
+              }
+
+              if (failureType == CameraSession.FailureType.DISCONNECTED) {
+                eventsHandler.onCameraDisconnected();
+              } else {
+                eventsHandler.onCameraError(error);
+              }
+            } else {
+              Logging.w(TAG, "Opening camera failed, retry: " + error);
+              createSessionInternal(OPEN_CAMERA_DELAY_MS);
+            }
+          }
+        }
+      };
+
+  @Nullable
+  private final CameraSession.Events cameraSessionEventsHandler = new CameraSession.Events() {
+    @Override
+    public void onCameraOpening() {
+      checkIsOnCameraThread();
+      synchronized (stateLock) {
+        if (currentSession != null) {
+          Logging.w(TAG, "onCameraOpening while session was open.");
+          return;
+        }
+        eventsHandler.onCameraOpening(cameraName);
+      }
+    }
+
+    @Override
+    public void onCameraError(CameraSession session, String error) {
+      checkIsOnCameraThread();
+      synchronized (stateLock) {
+        if (session != currentSession) {
+          Logging.w(TAG, "onCameraError from another session: " + error);
+          return;
+        }
+        eventsHandler.onCameraError(error);
+        stopCapture();
+      }
+    }
+
+    @Override
+    public void onCameraDisconnected(CameraSession session) {
+      checkIsOnCameraThread();
+      synchronized (stateLock) {
+        if (session != currentSession) {
+          Logging.w(TAG, "onCameraDisconnected from another session.");
+          return;
+        }
+        eventsHandler.onCameraDisconnected();
+        stopCapture();
+      }
+    }
+
+    @Override
+    public void onCameraClosed(CameraSession session) {
+      checkIsOnCameraThread();
+      synchronized (stateLock) {
+        if (session != currentSession && currentSession != null) {
+          Logging.d(TAG, "onCameraClosed from another session.");
+          return;
+        }
+        eventsHandler.onCameraClosed();
+      }
+    }
+
+    @Override
+    public void onFrameCaptured(CameraSession session, VideoFrame frame) {
+      checkIsOnCameraThread();
+      synchronized (stateLock) {
+        if (session != currentSession) {
+          Logging.w(TAG, "onFrameCaptured from another session.");
+          return;
+        }
+        if (!firstFrameObserved) {
+          eventsHandler.onFirstFrameAvailable();
+          firstFrameObserved = true;
+        }
+        cameraStatistics.addFrame();
+        capturerObserver.onFrameCaptured(frame);
+      }
+    }
+  };
+
+  private final Runnable openCameraTimeoutRunnable = new Runnable() {
+    @Override
+    public void run() {
+      eventsHandler.onCameraError("Camera failed to start within timeout.");
+    }
+  };
+
+  // Initialized on initialize
+  // -------------------------
+  private Handler cameraThreadHandler;
+  private Context applicationContext;
+  private org.webrtc.CapturerObserver capturerObserver;
+  private SurfaceTextureHelper surfaceHelper;
+
+  private final Object stateLock = new Object();
+  private boolean sessionOpening; /* guarded by stateLock */
+  @Nullable private CameraSession currentSession; /* guarded by stateLock */
+  private String cameraName; /* guarded by stateLock */
+  private String pendingCameraName; /* guarded by stateLock */
+  private int width; /* guarded by stateLock */
+  private int height; /* guarded by stateLock */
+  private int framerate; /* guarded by stateLock */
+  private int openAttemptsRemaining; /* guarded by stateLock */
+  private SwitchState switchState = SwitchState.IDLE; /* guarded by stateLock */
+  @Nullable private CameraSwitchHandler switchEventsHandler; /* guarded by stateLock */
+  // Valid from onDone call until stopCapture, otherwise null.
+  @Nullable private CameraStatistics cameraStatistics; /* guarded by stateLock */
+  private boolean firstFrameObserved; /* guarded by stateLock */
+
+  public CameraCapturer(String cameraName, @Nullable CameraEventsHandler eventsHandler,
+      CameraEnumerator cameraEnumerator) {
+    if (eventsHandler == null) {
+      eventsHandler = new CameraEventsHandler() {
+        @Override
+        public void onCameraError(String errorDescription) {}
+        @Override
+        public void onCameraDisconnected() {}
+        @Override
+        public void onCameraFreezed(String errorDescription) {}
+        @Override
+        public void onCameraOpening(String cameraName) {}
+        @Override
+        public void onFirstFrameAvailable() {}
+        @Override
+        public void onCameraClosed() {}
+      };
+    }
+
+    this.eventsHandler = eventsHandler;
+    this.cameraEnumerator = cameraEnumerator;
+    this.cameraName = cameraName;
+    List<String> deviceNames = Arrays.asList(cameraEnumerator.getDeviceNames());
+    uiThreadHandler = new Handler(Looper.getMainLooper());
+
+    if (deviceNames.isEmpty()) {
+      throw new RuntimeException("No cameras attached.");
+    }
+    if (!deviceNames.contains(this.cameraName)) {
+      throw new IllegalArgumentException(
+          "Camera name " + this.cameraName + " does not match any known camera device.");
+    }
+  }
+
+  @Override
+  public void initialize(SurfaceTextureHelper surfaceTextureHelper, Context applicationContext,
+      org.webrtc.CapturerObserver capturerObserver) {
+    this.applicationContext = applicationContext;
+    this.capturerObserver = capturerObserver;
+    this.surfaceHelper = surfaceTextureHelper;
+    this.cameraThreadHandler = surfaceTextureHelper.getHandler();
+  }
+
+  @Override
+  public void startCapture(int width, int height, int framerate) {
+    Logging.d(TAG, "startCapture: " + width + "x" + height + "@" + framerate);
+    if (applicationContext == null) {
+      throw new RuntimeException("CameraCapturer must be initialized before calling startCapture.");
+    }
+
+    synchronized (stateLock) {
+      if (sessionOpening || currentSession != null) {
+        Logging.w(TAG, "Session already open");
+        return;
+      }
+
+      this.width = width;
+      this.height = height;
+      this.framerate = framerate;
+
+      sessionOpening = true;
+      openAttemptsRemaining = MAX_OPEN_CAMERA_ATTEMPTS;
+      createSessionInternal(0);
+    }
+  }
+
+  private void createSessionInternal(int delayMs) {
+    uiThreadHandler.postDelayed(openCameraTimeoutRunnable, delayMs + OPEN_CAMERA_TIMEOUT);
+    cameraThreadHandler.postDelayed(new Runnable() {
+      @Override
+      public void run() {
+        createCameraSession(createSessionCallback, cameraSessionEventsHandler, applicationContext,
+            surfaceHelper, cameraName, width, height, framerate);
+      }
+    }, delayMs);
+  }
+
+  @Override
+  public void stopCapture() {
+    Logging.d(TAG, "Stop capture");
+
+    synchronized (stateLock) {
+      while (sessionOpening) {
+        Logging.d(TAG, "Stop capture: Waiting for session to open");
+        try {
+          stateLock.wait();
+        } catch (InterruptedException e) {
+          Logging.w(TAG, "Stop capture interrupted while waiting for the session to open.");
+          Thread.currentThread().interrupt();
+          return;
+        }
+      }
+
+      if (currentSession != null) {
+        Logging.d(TAG, "Stop capture: Nulling session");
+        cameraStatistics.release();
+        cameraStatistics = null;
+        final CameraSession oldSession = currentSession;
+        cameraThreadHandler.post(new Runnable() {
+          @Override
+          public void run() {
+            oldSession.stop();
+          }
+        });
+        currentSession = null;
+        capturerObserver.onCapturerStopped();
+      } else {
+        Logging.d(TAG, "Stop capture: No session open");
+      }
+    }
+
+    Logging.d(TAG, "Stop capture done");
+  }
+
+  @Override
+  public void changeCaptureFormat(int width, int height, int framerate) {
+    Logging.d(TAG, "changeCaptureFormat: " + width + "x" + height + "@" + framerate);
+    synchronized (stateLock) {
+      stopCapture();
+      startCapture(width, height, framerate);
+    }
+  }
+
+  @Override
+  public void dispose() {
+    Logging.d(TAG, "dispose");
+    stopCapture();
+  }
+
+  @Override
+  public void switchCamera(final CameraSwitchHandler switchEventsHandler) {
+    Logging.d(TAG, "switchCamera");
+    cameraThreadHandler.post(new Runnable() {
+      @Override
+      public void run() {
+        List<String> deviceNames = Arrays.asList(cameraEnumerator.getDeviceNames());
+
+        if (deviceNames.size() < 2) {
+          reportCameraSwitchError("No camera to switch to.", switchEventsHandler);
+          return;
+        }
+
+        int cameraNameIndex = deviceNames.indexOf(cameraName);
+        String cameraName = deviceNames.get((cameraNameIndex + 1) % deviceNames.size());
+        switchCameraInternal(switchEventsHandler, cameraName);
+      }
+    });
+  }
+
+  @Override
+  public void switchCamera(final CameraSwitchHandler switchEventsHandler, final String cameraName) {
+    Logging.d(TAG, "switchCamera");
+    cameraThreadHandler.post(new Runnable() {
+      @Override
+      public void run() {
+        switchCameraInternal(switchEventsHandler, cameraName);
+      }
+    });
+  }
+
+  @Override
+  public boolean isScreencast() {
+    return false;
+  }
+
+  public void printStackTrace() {
+    Thread cameraThread = null;
+    if (cameraThreadHandler != null) {
+      cameraThread = cameraThreadHandler.getLooper().getThread();
+    }
+    if (cameraThread != null) {
+      StackTraceElement[] cameraStackTrace = cameraThread.getStackTrace();
+      if (cameraStackTrace.length > 0) {
+        Logging.d(TAG, "CameraCapturer stack trace:");
+        for (StackTraceElement traceElem : cameraStackTrace) {
+          Logging.d(TAG, traceElem.toString());
+        }
+      }
+    }
+  }
+
+  private void reportCameraSwitchError(
+      String error, @Nullable CameraSwitchHandler switchEventsHandler) {
+    Logging.e(TAG, error);
+    if (switchEventsHandler != null) {
+      switchEventsHandler.onCameraSwitchError(error);
+    }
+  }
+
+  private void switchCameraInternal(
+      @Nullable final CameraSwitchHandler switchEventsHandler, final String selectedCameraName) {
+    Logging.d(TAG, "switchCamera internal");
+    List<String> deviceNames = Arrays.asList(cameraEnumerator.getDeviceNames());
+
+    if (!deviceNames.contains(selectedCameraName)) {
+      reportCameraSwitchError("Attempted to switch to unknown camera device " + selectedCameraName,
+          switchEventsHandler);
+      return;
+    }
+
+    synchronized (stateLock) {
+      if (switchState != SwitchState.IDLE) {
+        reportCameraSwitchError("Camera switch already in progress.", switchEventsHandler);
+        return;
+      }
+      if (!sessionOpening && currentSession == null) {
+        reportCameraSwitchError("switchCamera: camera is not running.", switchEventsHandler);
+        return;
+      }
+
+      this.switchEventsHandler = switchEventsHandler;
+      if (sessionOpening) {
+        switchState = SwitchState.PENDING;
+        pendingCameraName = selectedCameraName;
+        return;
+      } else {
+        switchState = SwitchState.IN_PROGRESS;
+      }
+
+      Logging.d(TAG, "switchCamera: Stopping session");
+      cameraStatistics.release();
+      cameraStatistics = null;
+      final CameraSession oldSession = currentSession;
+      cameraThreadHandler.post(new Runnable() {
+        @Override
+        public void run() {
+          oldSession.stop();
+        }
+      });
+      currentSession = null;
+
+      cameraName = selectedCameraName;
+
+      sessionOpening = true;
+      openAttemptsRemaining = 1;
+      createSessionInternal(0);
+    }
+    Logging.d(TAG, "switchCamera done");
+  }
+
+  private void checkIsOnCameraThread() {
+    if (Thread.currentThread() != cameraThreadHandler.getLooper().getThread()) {
+      Logging.e(TAG, "Check is on camera thread failed.");
+      throw new RuntimeException("Not on camera thread.");
+    }
+  }
+
+  protected String getCameraName() {
+    synchronized (stateLock) {
+      return cameraName;
+    }
+  }
+
+  abstract protected void createCameraSession(
+      CameraSession.CreateSessionCallback createSessionCallback, CameraSession.Events events,
+      Context applicationContext, SurfaceTextureHelper surfaceTextureHelper, String cameraName,
+      int width, int height, int framerate);
+}
diff --git a/third_party/libwebrtc/sdk/android/src/java/org/webrtc/CameraSession.java b/third_party/libwebrtc/sdk/android/src/java/org/webrtc/CameraSession.java
new file mode 100644
index 0000000000..8d137854d8
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/src/java/org/webrtc/CameraSession.java
@@ -0,0 +1,72 @@
+/*
+ *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+import android.content.Context;
+import android.graphics.Matrix;
+import android.view.WindowManager;
+import android.view.Surface;
+
+interface CameraSession {
+  enum FailureType { ERROR, DISCONNECTED }
+
+  // Callbacks are fired on the camera thread.
+  interface CreateSessionCallback {
+    void onDone(CameraSession session);
+    void onFailure(FailureType failureType, String error);
+  }
+
+  // Events are fired on the camera thread.
+  interface Events {
+    void onCameraOpening();
+    void onCameraError(CameraSession session, String error);
+    void onCameraDisconnected(CameraSession session);
+    void onCameraClosed(CameraSession session);
+    void onFrameCaptured(CameraSession session, VideoFrame frame);
+  }
+
+  /**
+   * Stops the capture. Waits until no more calls to capture observer will be made.
+   * If waitCameraStop is true, also waits for the camera to stop.
+   */
+  void stop();
+
+  static int getDeviceOrientation(Context context) {
+    final WindowManager wm = (WindowManager) context.getSystemService(Context.WINDOW_SERVICE);
+    switch (wm.getDefaultDisplay().getRotation()) {
+      case Surface.ROTATION_90:
+        return 90;
+      case Surface.ROTATION_180:
+        return 180;
+      case Surface.ROTATION_270:
+        return 270;
+      case Surface.ROTATION_0:
+      default:
+        return 0;
+    }
+  }
+
+  static VideoFrame.TextureBuffer createTextureBufferWithModifiedTransformMatrix(
+      TextureBufferImpl buffer, boolean mirror, int rotation) {
+    final Matrix transformMatrix = new Matrix();
+    // Perform mirror and rotation around (0.5, 0.5) since that is the center of the texture.
+    transformMatrix.preTranslate(/* dx= */ 0.5f, /* dy= */ 0.5f);
+    if (mirror) {
+      transformMatrix.preScale(/* sx= */ -1f, /* sy= */ 1f);
+    }
+    transformMatrix.preRotate(rotation);
+    transformMatrix.preTranslate(/* dx= */ -0.5f, /* dy= */ -0.5f);
+
+    // The width and height are not affected by rotation since Camera2Session has set them to the
+    // value they should be after undoing the rotation.
+    return buffer.applyTransformMatrix(transformMatrix, buffer.getWidth(), buffer.getHeight());
+  }
+}
diff --git a/third_party/libwebrtc/sdk/android/src/java/org/webrtc/DynamicBitrateAdjuster.java b/third_party/libwebrtc/sdk/android/src/java/org/webrtc/DynamicBitrateAdjuster.java
new file mode 100644
index 0000000000..96a15bbfe1
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/src/java/org/webrtc/DynamicBitrateAdjuster.java
@@ -0,0 +1,98 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+/**
+ * BitrateAdjuster that tracks the bandwidth produced by an encoder and dynamically adjusts the
+ * bitrate.  Used for hardware codecs that pay attention to framerate but still deviate from the
+ * target bitrate by unacceptable margins.
+ */
+class DynamicBitrateAdjuster extends BaseBitrateAdjuster {
+  // Change the bitrate at most once every three seconds.
+  private static final double BITRATE_ADJUSTMENT_SEC = 3.0;
+  // Maximum bitrate adjustment scale - no more than 4 times.
+  private static final double BITRATE_ADJUSTMENT_MAX_SCALE = 4;
+  // Amount of adjustment steps to reach maximum scale.
+  private static final int BITRATE_ADJUSTMENT_STEPS = 20;
+
+  private static final double BITS_PER_BYTE = 8.0;
+
+  // How far the codec has deviated above (or below) the target bitrate (tracked in bytes).
+  private double deviationBytes;
+  private double timeSinceLastAdjustmentMs;
+  private int bitrateAdjustmentScaleExp;
+
+  @Override
+  public void setTargets(int targetBitrateBps, double targetFramerateFps) {
+    if (this.targetBitrateBps > 0 && targetBitrateBps < this.targetBitrateBps) {
+      // Rescale the accumulator level if the accumulator max decreases
+      deviationBytes = deviationBytes * targetBitrateBps / this.targetBitrateBps;
+    }
+    super.setTargets(targetBitrateBps, targetFramerateFps);
+  }
+
+  @Override
+  public void reportEncodedFrame(int size) {
+    if (targetFramerateFps == 0) {
+      return;
+    }
+
+    // Accumulate the difference between actual and expected frame sizes.
+    double expectedBytesPerFrame = (targetBitrateBps / BITS_PER_BYTE) / targetFramerateFps;
+    deviationBytes += (size - expectedBytesPerFrame);
+    timeSinceLastAdjustmentMs += 1000.0 / targetFramerateFps;
+
+    // Adjust the bitrate when the encoder accumulates one second's worth of data in excess or
+    // shortfall of the target.
+    double deviationThresholdBytes = targetBitrateBps / BITS_PER_BYTE;
+
+    // Cap the deviation, i.e., don't let it grow beyond some level to avoid using too old data for
+    // bitrate adjustment.  This also prevents taking more than 3 "steps" in a given 3-second cycle.
+    double deviationCap = BITRATE_ADJUSTMENT_SEC * deviationThresholdBytes;
+    deviationBytes = Math.min(deviationBytes, deviationCap);
+    deviationBytes = Math.max(deviationBytes, -deviationCap);
+
+    // Do bitrate adjustment every 3 seconds if actual encoder bitrate deviates too much
+    // from the target value.
+    if (timeSinceLastAdjustmentMs <= 1000 * BITRATE_ADJUSTMENT_SEC) {
+      return;
+    }
+
+    if (deviationBytes > deviationThresholdBytes) {
+      // Encoder generates too high bitrate - need to reduce the scale.
+      int bitrateAdjustmentInc = (int) (deviationBytes / deviationThresholdBytes + 0.5);
+      bitrateAdjustmentScaleExp -= bitrateAdjustmentInc;
+      // Don't let the adjustment scale drop below -BITRATE_ADJUSTMENT_STEPS.
+      // This sets a minimum exponent of -1 (bitrateAdjustmentScaleExp / BITRATE_ADJUSTMENT_STEPS).
+      bitrateAdjustmentScaleExp = Math.max(bitrateAdjustmentScaleExp, -BITRATE_ADJUSTMENT_STEPS);
+      deviationBytes = deviationThresholdBytes;
+    } else if (deviationBytes < -deviationThresholdBytes) {
+      // Encoder generates too low bitrate - need to increase the scale.
+      int bitrateAdjustmentInc = (int) (-deviationBytes / deviationThresholdBytes + 0.5);
+      bitrateAdjustmentScaleExp += bitrateAdjustmentInc;
+      // Don't let the adjustment scale exceed BITRATE_ADJUSTMENT_STEPS.
+      // This sets a maximum exponent of 1 (bitrateAdjustmentScaleExp / BITRATE_ADJUSTMENT_STEPS).
+      bitrateAdjustmentScaleExp = Math.min(bitrateAdjustmentScaleExp, BITRATE_ADJUSTMENT_STEPS);
+      deviationBytes = -deviationThresholdBytes;
+    }
+    timeSinceLastAdjustmentMs = 0;
+  }
+
+  private double getBitrateAdjustmentScale() {
+    return Math.pow(BITRATE_ADJUSTMENT_MAX_SCALE,
+        (double) bitrateAdjustmentScaleExp / BITRATE_ADJUSTMENT_STEPS);
+  }
+
+  @Override
+  public int getAdjustedBitrateBps() {
+    return (int) (targetBitrateBps * getBitrateAdjustmentScale());
+  }
+}
diff --git a/third_party/libwebrtc/sdk/android/src/java/org/webrtc/EglBase10Impl.java b/third_party/libwebrtc/sdk/android/src/java/org/webrtc/EglBase10Impl.java
new file mode 100644
index 0000000000..254a17c750
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/src/java/org/webrtc/EglBase10Impl.java
@@ -0,0 +1,365 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+import android.graphics.Canvas;
+import android.graphics.Rect;
+import android.graphics.SurfaceTexture;
+import android.opengl.EGL14;
+import android.opengl.GLException;
+import android.view.Surface;
+import android.view.SurfaceHolder;
+import androidx.annotation.Nullable;
+import javax.microedition.khronos.egl.EGL10;
+import javax.microedition.khronos.egl.EGLConfig;
+import javax.microedition.khronos.egl.EGLContext;
+import javax.microedition.khronos.egl.EGLDisplay;
+import javax.microedition.khronos.egl.EGLSurface;
+
+/**
+ * Holds EGL state and utility methods for handling an egl 1.0 EGLContext, an EGLDisplay,
+ * and an EGLSurface.
+ */
+class EglBase10Impl implements EglBase10 {
+  private static final String TAG = "EglBase10Impl";
+  // This constant is taken from EGL14.EGL_CONTEXT_CLIENT_VERSION.
+  private static final int EGL_CONTEXT_CLIENT_VERSION = 0x3098;
+
+  private final EGL10 egl;
+  private EGLContext eglContext;
+  @Nullable private EGLConfig eglConfig;
+  private EGLDisplay eglDisplay;
+  private EGLSurface eglSurface = EGL10.EGL_NO_SURFACE;
+
+  // EGL wrapper for an actual EGLContext.
+  private static class Context implements EglBase10.Context {
+    private final EGL10 egl;
+    private final EGLContext eglContext;
+    private final EGLConfig eglContextConfig;
+
+    @Override
+    public EGLContext getRawContext() {
+      return eglContext;
+    }
+
+    @Override
+    public long getNativeEglContext() {
+      EGLContext previousContext = egl.eglGetCurrentContext();
+      EGLDisplay currentDisplay = egl.eglGetCurrentDisplay();
+      EGLSurface previousDrawSurface = egl.eglGetCurrentSurface(EGL10.EGL_DRAW);
+      EGLSurface previousReadSurface = egl.eglGetCurrentSurface(EGL10.EGL_READ);
+      EGLSurface tempEglSurface = null;
+
+      if (currentDisplay == EGL10.EGL_NO_DISPLAY) {
+        currentDisplay = egl.eglGetDisplay(EGL10.EGL_DEFAULT_DISPLAY);
+      }
+
+      try {
+        if (previousContext != eglContext) {
+          int[] surfaceAttribs = {EGL10.EGL_WIDTH, 1, EGL10.EGL_HEIGHT, 1, EGL10.EGL_NONE};
+          tempEglSurface =
+              egl.eglCreatePbufferSurface(currentDisplay, eglContextConfig, surfaceAttribs);
+          if (!egl.eglMakeCurrent(currentDisplay, tempEglSurface, tempEglSurface, eglContext)) {
+            throw new GLException(egl.eglGetError(),
+                "Failed to make temporary EGL surface active: " + egl.eglGetError());
+          }
+        }
+
+        return nativeGetCurrentNativeEGLContext();
+      } finally {
+        if (tempEglSurface != null) {
+          egl.eglMakeCurrent(
+              currentDisplay, previousDrawSurface, previousReadSurface, previousContext);
+          egl.eglDestroySurface(currentDisplay, tempEglSurface);
+        }
+      }
+    }
+
+    public Context(EGL10 egl, EGLContext eglContext, EGLConfig eglContextConfig) {
+      this.egl = egl;
+      this.eglContext = eglContext;
+      this.eglContextConfig = eglContextConfig;
+    }
+  }
+
+  // Create a new context with the specified config type, sharing data with sharedContext.
+  public EglBase10Impl(EGLContext sharedContext, int[] configAttributes) {
+    this.egl = (EGL10) EGLContext.getEGL();
+    eglDisplay = getEglDisplay();
+    eglConfig = getEglConfig(egl, eglDisplay, configAttributes);
+    final int openGlesVersion = EglBase.getOpenGlesVersionFromConfig(configAttributes);
+    Logging.d(TAG, "Using OpenGL ES version " + openGlesVersion);
+    eglContext = createEglContext(sharedContext, eglDisplay, eglConfig, openGlesVersion);
+  }
+
+  @Override
+  public void createSurface(Surface surface) {
+    /**
+     * We have to wrap Surface in a SurfaceHolder because for some reason eglCreateWindowSurface
+     * couldn't actually take a Surface object until API 17. Older versions fortunately just call
+     * SurfaceHolder.getSurface(), so we'll do that. No other methods are relevant.
+     */
+    class FakeSurfaceHolder implements SurfaceHolder {
+      private final Surface surface;
+
+      FakeSurfaceHolder(Surface surface) {
+        this.surface = surface;
+      }
+
+      @Override
+      public void addCallback(Callback callback) {}
+
+      @Override
+      public void removeCallback(Callback callback) {}
+
+      @Override
+      public boolean isCreating() {
+        return false;
+      }
+
+      @Deprecated
+      @Override
+      public void setType(int i) {}
+
+      @Override
+      public void setFixedSize(int i, int i2) {}
+
+      @Override
+      public void setSizeFromLayout() {}
+
+      @Override
+      public void setFormat(int i) {}
+
+      @Override
+      public void setKeepScreenOn(boolean b) {}
+
+      @Nullable
+      @Override
+      public Canvas lockCanvas() {
+        return null;
+      }
+
+      @Nullable
+      @Override
+      public Canvas lockCanvas(Rect rect) {
+        return null;
+      }
+
+      @Override
+      public void unlockCanvasAndPost(Canvas canvas) {}
+
+      @Nullable
+      @Override
+      public Rect getSurfaceFrame() {
+        return null;
+      }
+
+      @Override
+      public Surface getSurface() {
+        return surface;
+      }
+    }
+
+    createSurfaceInternal(new FakeSurfaceHolder(surface));
+  }
+
+  // Create EGLSurface from the Android SurfaceTexture.
+  @Override
+  public void createSurface(SurfaceTexture surfaceTexture) {
+    createSurfaceInternal(surfaceTexture);
+  }
+
+  // Create EGLSurface from either a SurfaceHolder or a SurfaceTexture.
+  private void createSurfaceInternal(Object nativeWindow) {
+    if (!(nativeWindow instanceof SurfaceHolder) && !(nativeWindow instanceof SurfaceTexture)) {
+      throw new IllegalStateException("Input must be either a SurfaceHolder or SurfaceTexture");
+    }
+    checkIsNotReleased();
+    if (eglSurface != EGL10.EGL_NO_SURFACE) {
+      throw new RuntimeException("Already has an EGLSurface");
+    }
+    int[] surfaceAttribs = {EGL10.EGL_NONE};
+    eglSurface = egl.eglCreateWindowSurface(eglDisplay, eglConfig, nativeWindow, surfaceAttribs);
+    if (eglSurface == EGL10.EGL_NO_SURFACE) {
+      throw new GLException(egl.eglGetError(),
+          "Failed to create window surface: 0x" + Integer.toHexString(egl.eglGetError()));
+    }
+  }
+
+  // Create dummy 1x1 pixel buffer surface so the context can be made current.
+  @Override
+  public void createDummyPbufferSurface() {
+    createPbufferSurface(1, 1);
+  }
+
+  @Override
+  public void createPbufferSurface(int width, int height) {
+    checkIsNotReleased();
+    if (eglSurface != EGL10.EGL_NO_SURFACE) {
+      throw new RuntimeException("Already has an EGLSurface");
+    }
+    int[] surfaceAttribs = {EGL10.EGL_WIDTH, width, EGL10.EGL_HEIGHT, height, EGL10.EGL_NONE};
+    eglSurface = egl.eglCreatePbufferSurface(eglDisplay, eglConfig, surfaceAttribs);
+    if (eglSurface == EGL10.EGL_NO_SURFACE) {
+      throw new GLException(egl.eglGetError(),
+          "Failed to create pixel buffer surface with size " + width + "x" + height + ": 0x"
+              + Integer.toHexString(egl.eglGetError()));
+    }
+  }
+
+  @Override
+  public org.webrtc.EglBase.Context getEglBaseContext() {
+    return new Context(egl, eglContext, eglConfig);
+  }
+
+  @Override
+  public boolean hasSurface() {
+    return eglSurface != EGL10.EGL_NO_SURFACE;
+  }
+
+  @Override
+  public int surfaceWidth() {
+    final int widthArray[] = new int[1];
+    egl.eglQuerySurface(eglDisplay, eglSurface, EGL10.EGL_WIDTH, widthArray);
+    return widthArray[0];
+  }
+
+  @Override
+  public int surfaceHeight() {
+    final int heightArray[] = new int[1];
+    egl.eglQuerySurface(eglDisplay, eglSurface, EGL10.EGL_HEIGHT, heightArray);
+    return heightArray[0];
+  }
+
+  @Override
+  public void releaseSurface() {
+    if (eglSurface != EGL10.EGL_NO_SURFACE) {
+      egl.eglDestroySurface(eglDisplay, eglSurface);
+      eglSurface = EGL10.EGL_NO_SURFACE;
+    }
+  }
+
+  private void checkIsNotReleased() {
+    if (eglDisplay == EGL10.EGL_NO_DISPLAY || eglContext == EGL10.EGL_NO_CONTEXT
+        || eglConfig == null) {
+      throw new RuntimeException("This object has been released");
+    }
+  }
+
+  @Override
+  public void release() {
+    checkIsNotReleased();
+    releaseSurface();
+    detachCurrent();
+    egl.eglDestroyContext(eglDisplay, eglContext);
+    egl.eglTerminate(eglDisplay);
+    eglContext = EGL10.EGL_NO_CONTEXT;
+    eglDisplay = EGL10.EGL_NO_DISPLAY;
+    eglConfig = null;
+  }
+
+  @Override
+  public void makeCurrent() {
+    checkIsNotReleased();
+    if (eglSurface == EGL10.EGL_NO_SURFACE) {
+      throw new RuntimeException("No EGLSurface - can't make current");
+    }
+    synchronized (EglBase.lock) {
+      if (!egl.eglMakeCurrent(eglDisplay, eglSurface, eglSurface, eglContext)) {
+        throw new GLException(egl.eglGetError(),
+            "eglMakeCurrent failed: 0x" + Integer.toHexString(egl.eglGetError()));
+      }
+    }
+  }
+
+  // Detach the current EGL context, so that it can be made current on another thread.
+  @Override
+  public void detachCurrent() {
+    synchronized (EglBase.lock) {
+      if (!egl.eglMakeCurrent(
+              eglDisplay, EGL10.EGL_NO_SURFACE, EGL10.EGL_NO_SURFACE, EGL10.EGL_NO_CONTEXT)) {
+        throw new GLException(egl.eglGetError(),
+            "eglDetachCurrent failed: 0x" + Integer.toHexString(egl.eglGetError()));
+      }
+    }
+  }
+
+  @Override
+  public void swapBuffers() {
+    checkIsNotReleased();
+    if (eglSurface == EGL10.EGL_NO_SURFACE) {
+      throw new RuntimeException("No EGLSurface - can't swap buffers");
+    }
+    synchronized (EglBase.lock) {
+      egl.eglSwapBuffers(eglDisplay, eglSurface);
+    }
+  }
+
+  @Override
+  public void swapBuffers(long timeStampNs) {
+    // Setting presentation time is not supported for EGL 1.0.
+    swapBuffers();
+  }
+
+  // Return an EGLDisplay, or die trying.
+  private EGLDisplay getEglDisplay() {
+    EGLDisplay eglDisplay = egl.eglGetDisplay(EGL10.EGL_DEFAULT_DISPLAY);
+    if (eglDisplay == EGL10.EGL_NO_DISPLAY) {
+      throw new GLException(egl.eglGetError(),
+          "Unable to get EGL10 display: 0x" + Integer.toHexString(egl.eglGetError()));
+    }
+    int[] version = new int[2];
+    if (!egl.eglInitialize(eglDisplay, version)) {
+      throw new GLException(egl.eglGetError(),
+          "Unable to initialize EGL10: 0x" + Integer.toHexString(egl.eglGetError()));
+    }
+    return eglDisplay;
+  }
+
+  // Return an EGLConfig, or die trying.
+  private static EGLConfig getEglConfig(EGL10 egl, EGLDisplay eglDisplay, int[] configAttributes) {
+    EGLConfig[] configs = new EGLConfig[1];
+    int[] numConfigs = new int[1];
+    if (!egl.eglChooseConfig(eglDisplay, configAttributes, configs, configs.length, numConfigs)) {
+      throw new GLException(
+          egl.eglGetError(), "eglChooseConfig failed: 0x" + Integer.toHexString(egl.eglGetError()));
+    }
+    if (numConfigs[0] <= 0) {
+      throw new RuntimeException("Unable to find any matching EGL config");
+    }
+    final EGLConfig eglConfig = configs[0];
+    if (eglConfig == null) {
+      throw new RuntimeException("eglChooseConfig returned null");
+    }
+    return eglConfig;
+  }
+
+  // Return an EGLConfig, or die trying.
+  private EGLContext createEglContext(@Nullable EGLContext sharedContext, EGLDisplay eglDisplay,
+      EGLConfig eglConfig, int openGlesVersion) {
+    if (sharedContext != null && sharedContext == EGL10.EGL_NO_CONTEXT) {
+      throw new RuntimeException("Invalid sharedContext");
+    }
+    int[] contextAttributes = {EGL_CONTEXT_CLIENT_VERSION, openGlesVersion, EGL10.EGL_NONE};
+    EGLContext rootContext = sharedContext == null ? EGL10.EGL_NO_CONTEXT : sharedContext;
+    final EGLContext eglContext;
+    synchronized (EglBase.lock) {
+      eglContext = egl.eglCreateContext(eglDisplay, eglConfig, rootContext, contextAttributes);
+    }
+    if (eglContext == EGL10.EGL_NO_CONTEXT) {
+      throw new GLException(egl.eglGetError(),
+          "Failed to create EGL context: 0x" + Integer.toHexString(egl.eglGetError()));
+    }
+    return eglContext;
+  }
+
+  private static native long nativeGetCurrentNativeEGLContext();
+}
diff --git a/third_party/libwebrtc/sdk/android/src/java/org/webrtc/EglBase14Impl.java b/third_party/libwebrtc/sdk/android/src/java/org/webrtc/EglBase14Impl.java
new file mode 100644
index 0000000000..caf45b091e
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/src/java/org/webrtc/EglBase14Impl.java
@@ -0,0 +1,271 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+import android.graphics.SurfaceTexture;
+import android.opengl.EGL14;
+import android.opengl.EGLConfig;
+import android.opengl.EGLContext;
+import android.opengl.EGLDisplay;
+import android.opengl.EGLExt;
+import android.opengl.EGLSurface;
+import android.opengl.GLException;
+import android.os.Build;
+import android.view.Surface;
+import androidx.annotation.Nullable;
+
+/**
+ * Holds EGL state and utility methods for handling an EGL14 EGLContext, an EGLDisplay,
+ * and an EGLSurface.
+ */
+@SuppressWarnings("ReferenceEquality") // We want to compare to EGL14 constants.
+class EglBase14Impl implements EglBase14 {
+  private static final String TAG = "EglBase14Impl";
+  private EGLContext eglContext;
+  @Nullable private EGLConfig eglConfig;
+  private EGLDisplay eglDisplay;
+  private EGLSurface eglSurface = EGL14.EGL_NO_SURFACE;
+
+  public static class Context implements EglBase14.Context {
+    private final EGLContext egl14Context;
+
+    @Override
+    public EGLContext getRawContext() {
+      return egl14Context;
+    }
+
+    @Override
+    public long getNativeEglContext() {
+      return egl14Context.getNativeHandle();
+    }
+
+    public Context(android.opengl.EGLContext eglContext) {
+      this.egl14Context = eglContext;
+    }
+  }
+
+  // Create a new context with the specified config type, sharing data with sharedContext.
+  // `sharedContext` may be null.
+  public EglBase14Impl(EGLContext sharedContext, int[] configAttributes) {
+    eglDisplay = getEglDisplay();
+    eglConfig = getEglConfig(eglDisplay, configAttributes);
+    final int openGlesVersion = EglBase.getOpenGlesVersionFromConfig(configAttributes);
+    Logging.d(TAG, "Using OpenGL ES version " + openGlesVersion);
+    eglContext = createEglContext(sharedContext, eglDisplay, eglConfig, openGlesVersion);
+  }
+
+  // Create EGLSurface from the Android Surface.
+  @Override
+  public void createSurface(Surface surface) {
+    createSurfaceInternal(surface);
+  }
+
+  // Create EGLSurface from the Android SurfaceTexture.
+  @Override
+  public void createSurface(SurfaceTexture surfaceTexture) {
+    createSurfaceInternal(surfaceTexture);
+  }
+
+  // Create EGLSurface from either Surface or SurfaceTexture.
+  private void createSurfaceInternal(Object surface) {
+    if (!(surface instanceof Surface) && !(surface instanceof SurfaceTexture)) {
+      throw new IllegalStateException("Input must be either a Surface or SurfaceTexture");
+    }
+    checkIsNotReleased();
+    if (eglSurface != EGL14.EGL_NO_SURFACE) {
+      throw new RuntimeException("Already has an EGLSurface");
+    }
+    int[] surfaceAttribs = {EGL14.EGL_NONE};
+    eglSurface = EGL14.eglCreateWindowSurface(eglDisplay, eglConfig, surface, surfaceAttribs, 0);
+    if (eglSurface == EGL14.EGL_NO_SURFACE) {
+      throw new GLException(EGL14.eglGetError(),
+          "Failed to create window surface: 0x" + Integer.toHexString(EGL14.eglGetError()));
+    }
+  }
+
+  @Override
+  public void createDummyPbufferSurface() {
+    createPbufferSurface(1, 1);
+  }
+
+  @Override
+  public void createPbufferSurface(int width, int height) {
+    checkIsNotReleased();
+    if (eglSurface != EGL14.EGL_NO_SURFACE) {
+      throw new RuntimeException("Already has an EGLSurface");
+    }
+    int[] surfaceAttribs = {EGL14.EGL_WIDTH, width, EGL14.EGL_HEIGHT, height, EGL14.EGL_NONE};
+    eglSurface = EGL14.eglCreatePbufferSurface(eglDisplay, eglConfig, surfaceAttribs, 0);
+    if (eglSurface == EGL14.EGL_NO_SURFACE) {
+      throw new GLException(EGL14.eglGetError(),
+          "Failed to create pixel buffer surface with size " + width + "x" + height + ": 0x"
+              + Integer.toHexString(EGL14.eglGetError()));
+    }
+  }
+
+  @Override
+  public Context getEglBaseContext() {
+    return new Context(eglContext);
+  }
+
+  @Override
+  public boolean hasSurface() {
+    return eglSurface != EGL14.EGL_NO_SURFACE;
+  }
+
+  @Override
+  public int surfaceWidth() {
+    final int widthArray[] = new int[1];
+    EGL14.eglQuerySurface(eglDisplay, eglSurface, EGL14.EGL_WIDTH, widthArray, 0);
+    return widthArray[0];
+  }
+
+  @Override
+  public int surfaceHeight() {
+    final int heightArray[] = new int[1];
+    EGL14.eglQuerySurface(eglDisplay, eglSurface, EGL14.EGL_HEIGHT, heightArray, 0);
+    return heightArray[0];
+  }
+
+  @Override
+  public void releaseSurface() {
+    if (eglSurface != EGL14.EGL_NO_SURFACE) {
+      EGL14.eglDestroySurface(eglDisplay, eglSurface);
+      eglSurface = EGL14.EGL_NO_SURFACE;
+    }
+  }
+
+  private void checkIsNotReleased() {
+    if (eglDisplay == EGL14.EGL_NO_DISPLAY || eglContext == EGL14.EGL_NO_CONTEXT
+        || eglConfig == null) {
+      throw new RuntimeException("This object has been released");
+    }
+  }
+
+  @Override
+  public void release() {
+    checkIsNotReleased();
+    releaseSurface();
+    detachCurrent();
+    synchronized (EglBase.lock) {
+      EGL14.eglDestroyContext(eglDisplay, eglContext);
+    }
+    EGL14.eglReleaseThread();
+    EGL14.eglTerminate(eglDisplay);
+    eglContext = EGL14.EGL_NO_CONTEXT;
+    eglDisplay = EGL14.EGL_NO_DISPLAY;
+    eglConfig = null;
+  }
+
+  @Override
+  public void makeCurrent() {
+    checkIsNotReleased();
+    if (eglSurface == EGL14.EGL_NO_SURFACE) {
+      throw new RuntimeException("No EGLSurface - can't make current");
+    }
+    synchronized (EglBase.lock) {
+      if (!EGL14.eglMakeCurrent(eglDisplay, eglSurface, eglSurface, eglContext)) {
+        throw new GLException(EGL14.eglGetError(),
+            "eglMakeCurrent failed: 0x" + Integer.toHexString(EGL14.eglGetError()));
+      }
+    }
+  }
+
+  // Detach the current EGL context, so that it can be made current on another thread.
+  @Override
+  public void detachCurrent() {
+    synchronized (EglBase.lock) {
+      if (!EGL14.eglMakeCurrent(
+              eglDisplay, EGL14.EGL_NO_SURFACE, EGL14.EGL_NO_SURFACE, EGL14.EGL_NO_CONTEXT)) {
+        throw new GLException(EGL14.eglGetError(),
+            "eglDetachCurrent failed: 0x" + Integer.toHexString(EGL14.eglGetError()));
+      }
+    }
+  }
+
+  @Override
+  public void swapBuffers() {
+    checkIsNotReleased();
+    if (eglSurface == EGL14.EGL_NO_SURFACE) {
+      throw new RuntimeException("No EGLSurface - can't swap buffers");
+    }
+    synchronized (EglBase.lock) {
+      EGL14.eglSwapBuffers(eglDisplay, eglSurface);
+    }
+  }
+
+  @Override
+  public void swapBuffers(long timeStampNs) {
+    checkIsNotReleased();
+    if (eglSurface == EGL14.EGL_NO_SURFACE) {
+      throw new RuntimeException("No EGLSurface - can't swap buffers");
+    }
+    synchronized (EglBase.lock) {
+      // See
+      // https://android.googlesource.com/platform/frameworks/native/+/tools_r22.2/opengl/specs/EGL_ANDROID_presentation_time.txt
+      EGLExt.eglPresentationTimeANDROID(eglDisplay, eglSurface, timeStampNs);
+      EGL14.eglSwapBuffers(eglDisplay, eglSurface);
+    }
+  }
+
+  // Return an EGLDisplay, or die trying.
+  private static EGLDisplay getEglDisplay() {
+    EGLDisplay eglDisplay = EGL14.eglGetDisplay(EGL14.EGL_DEFAULT_DISPLAY);
+    if (eglDisplay == EGL14.EGL_NO_DISPLAY) {
+      throw new GLException(EGL14.eglGetError(),
+          "Unable to get EGL14 display: 0x" + Integer.toHexString(EGL14.eglGetError()));
+    }
+    int[] version = new int[2];
+    if (!EGL14.eglInitialize(eglDisplay, version, 0, version, 1)) {
+      throw new GLException(EGL14.eglGetError(),
+          "Unable to initialize EGL14: 0x" + Integer.toHexString(EGL14.eglGetError()));
+    }
+    return eglDisplay;
+  }
+
+  // Return an EGLConfig, or die trying.
+  private static EGLConfig getEglConfig(EGLDisplay eglDisplay, int[] configAttributes) {
+    EGLConfig[] configs = new EGLConfig[1];
+    int[] numConfigs = new int[1];
+    if (!EGL14.eglChooseConfig(
+            eglDisplay, configAttributes, 0, configs, 0, configs.length, numConfigs, 0)) {
+      throw new GLException(EGL14.eglGetError(),
+          "eglChooseConfig failed: 0x" + Integer.toHexString(EGL14.eglGetError()));
+    }
+    if (numConfigs[0] <= 0) {
+      throw new RuntimeException("Unable to find any matching EGL config");
+    }
+    final EGLConfig eglConfig = configs[0];
+    if (eglConfig == null) {
+      throw new RuntimeException("eglChooseConfig returned null");
+    }
+    return eglConfig;
+  }
+
+  // Return an EGLConfig, or die trying.
+  private static EGLContext createEglContext(@Nullable EGLContext sharedContext,
+      EGLDisplay eglDisplay, EGLConfig eglConfig, int openGlesVersion) {
+    if (sharedContext != null && sharedContext == EGL14.EGL_NO_CONTEXT) {
+      throw new RuntimeException("Invalid sharedContext");
+    }
+    int[] contextAttributes = {EGL14.EGL_CONTEXT_CLIENT_VERSION, openGlesVersion, EGL14.EGL_NONE};
+    EGLContext rootContext = sharedContext == null ? EGL14.EGL_NO_CONTEXT : sharedContext;
+    final EGLContext eglContext;
+    synchronized (EglBase.lock) {
+      eglContext = EGL14.eglCreateContext(eglDisplay, eglConfig, rootContext, contextAttributes, 0);
+    }
+    if (eglContext == EGL14.EGL_NO_CONTEXT) {
+      throw new GLException(EGL14.eglGetError(),
+          "Failed to create EGL context: 0x" + Integer.toHexString(EGL14.eglGetError()));
+    }
+    return eglContext;
+  }
+}
diff --git a/third_party/libwebrtc/sdk/android/src/java/org/webrtc/Empty.java b/third_party/libwebrtc/sdk/android/src/java/org/webrtc/Empty.java
new file mode 100644
index 0000000000..fe9481e182
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/src/java/org/webrtc/Empty.java
@@ -0,0 +1,17 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+/**
+ * Empty class for use in libjingle_peerconnection_java because all targets require at least one
+ * Java file.
+ */
+class Empty {}
diff --git a/third_party/libwebrtc/sdk/android/src/java/org/webrtc/FramerateBitrateAdjuster.java b/third_party/libwebrtc/sdk/android/src/java/org/webrtc/FramerateBitrateAdjuster.java
new file mode 100644
index 0000000000..e28b7b5a26
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/src/java/org/webrtc/FramerateBitrateAdjuster.java
@@ -0,0 +1,26 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+/**
+ * BitrateAdjuster that adjusts the bitrate to compensate for changes in the framerate.  Used with
+ * hardware codecs that assume the framerate never changes.
+ */
+class FramerateBitrateAdjuster extends BaseBitrateAdjuster {
+  private static final int DEFAULT_FRAMERATE_FPS = 30;
+
+  @Override
+  public void setTargets(int targetBitrateBps, double targetFramerateFps) {
+    // Keep frame rate unchanged and adjust bit rate.
+    this.targetFramerateFps = DEFAULT_FRAMERATE_FPS;
+    this.targetBitrateBps = (int) (targetBitrateBps * DEFAULT_FRAMERATE_FPS / targetFramerateFps);
+  }
+}
diff --git a/third_party/libwebrtc/sdk/android/src/java/org/webrtc/GlGenericDrawer.java b/third_party/libwebrtc/sdk/android/src/java/org/webrtc/GlGenericDrawer.java
new file mode 100644
index 0000000000..34144e2f75
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/src/java/org/webrtc/GlGenericDrawer.java
@@ -0,0 +1,281 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+import android.opengl.GLES11Ext;
+import android.opengl.GLES20;
+import androidx.annotation.Nullable;
+import java.nio.FloatBuffer;
+
+/**
+ * Helper class to implement an instance of RendererCommon.GlDrawer that can accept multiple input
+ * sources (OES, RGB, or YUV) using a generic fragment shader as input. The generic fragment shader
+ * should sample pixel values from the function "sample" that will be provided by this class and
+ * provides an abstraction for the input source type (OES, RGB, or YUV). The texture coordinate
+ * variable name will be "tc" and the texture matrix in the vertex shader will be "tex_mat". The
+ * simplest possible generic shader that just draws pixel from the frame unmodified looks like:
+ * void main() {
+ *   gl_FragColor = sample(tc);
+ * }
+ * This class covers the cases for most simple shaders and generates the necessary boiler plate.
+ * Advanced shaders can always implement RendererCommon.GlDrawer directly.
+ */
+class GlGenericDrawer implements RendererCommon.GlDrawer {
+  /**
+   * The different shader types representing different input sources. YUV here represents three
+   * separate Y, U, V textures.
+   */
+  public static enum ShaderType { OES, RGB, YUV }
+
+  /**
+   * The shader callbacks is used to customize behavior for a GlDrawer. It provides a hook to set
+   * uniform variables in the shader before a frame is drawn.
+   */
+  public static interface ShaderCallbacks {
+    /**
+     * This callback is called when a new shader has been compiled and created. It will be called
+     * for the first frame as well as when the shader type is changed. This callback can be used to
+     * do custom initialization of the shader that only needs to happen once.
+     */
+    void onNewShader(GlShader shader);
+
+    /**
+     * This callback is called before rendering a frame. It can be used to do custom preparation of
+     * the shader that needs to happen every frame.
+     */
+    void onPrepareShader(GlShader shader, float[] texMatrix, int frameWidth, int frameHeight,
+        int viewportWidth, int viewportHeight);
+  }
+
+  private static final String INPUT_VERTEX_COORDINATE_NAME = "in_pos";
+  private static final String INPUT_TEXTURE_COORDINATE_NAME = "in_tc";
+  private static final String TEXTURE_MATRIX_NAME = "tex_mat";
+  private static final String DEFAULT_VERTEX_SHADER_STRING = "varying vec2 tc;\n"
+      + "attribute vec4 in_pos;\n"
+      + "attribute vec4 in_tc;\n"
+      + "uniform mat4 tex_mat;\n"
+      + "void main() {\n"
+      + "  gl_Position = in_pos;\n"
+      + "  tc = (tex_mat * in_tc).xy;\n"
+      + "}\n";
+
+  // Vertex coordinates in Normalized Device Coordinates, i.e. (-1, -1) is bottom-left and (1, 1)
+  // is top-right.
+  private static final FloatBuffer FULL_RECTANGLE_BUFFER = GlUtil.createFloatBuffer(new float[] {
+      -1.0f, -1.0f, // Bottom left.
+      1.0f, -1.0f, // Bottom right.
+      -1.0f, 1.0f, // Top left.
+      1.0f, 1.0f, // Top right.
+  });
+
+  // Texture coordinates - (0, 0) is bottom-left and (1, 1) is top-right.
+  private static final FloatBuffer FULL_RECTANGLE_TEXTURE_BUFFER =
+      GlUtil.createFloatBuffer(new float[] {
+          0.0f, 0.0f, // Bottom left.
+          1.0f, 0.0f, // Bottom right.
+          0.0f, 1.0f, // Top left.
+          1.0f, 1.0f, // Top right.
+      });
+
+  static String createFragmentShaderString(String genericFragmentSource, ShaderType shaderType) {
+    final StringBuilder stringBuilder = new StringBuilder();
+    if (shaderType == ShaderType.OES) {
+      stringBuilder.append("#extension GL_OES_EGL_image_external : require\n");
+    }
+    stringBuilder.append("precision mediump float;\n");
+    stringBuilder.append("varying vec2 tc;\n");
+
+    if (shaderType == ShaderType.YUV) {
+      stringBuilder.append("uniform sampler2D y_tex;\n");
+      stringBuilder.append("uniform sampler2D u_tex;\n");
+      stringBuilder.append("uniform sampler2D v_tex;\n");
+
+      // Add separate function for sampling texture.
+      // yuv_to_rgb_mat is inverse of the matrix defined in YuvConverter.
+      stringBuilder.append("vec4 sample(vec2 p) {\n");
+      stringBuilder.append("  float y = texture2D(y_tex, p).r * 1.16438;\n");
+      stringBuilder.append("  float u = texture2D(u_tex, p).r;\n");
+      stringBuilder.append("  float v = texture2D(v_tex, p).r;\n");
+      stringBuilder.append("  return vec4(y + 1.59603 * v - 0.874202,\n");
+      stringBuilder.append("    y - 0.391762 * u - 0.812968 * v + 0.531668,\n");
+      stringBuilder.append("    y + 2.01723 * u - 1.08563, 1);\n");
+      stringBuilder.append("}\n");
+      stringBuilder.append(genericFragmentSource);
+    } else {
+      final String samplerName = shaderType == ShaderType.OES ? "samplerExternalOES" : "sampler2D";
+      stringBuilder.append("uniform ").append(samplerName).append(" tex;\n");
+
+      // Update the sampling function in-place.
+      stringBuilder.append(genericFragmentSource.replace("sample(", "texture2D(tex, "));
+    }
+
+    return stringBuilder.toString();
+  }
+
+  private final String genericFragmentSource;
+  private final String vertexShader;
+  private final ShaderCallbacks shaderCallbacks;
+  @Nullable private ShaderType currentShaderType;
+  @Nullable private GlShader currentShader;
+  private int inPosLocation;
+  private int inTcLocation;
+  private int texMatrixLocation;
+
+  public GlGenericDrawer(String genericFragmentSource, ShaderCallbacks shaderCallbacks) {
+    this(DEFAULT_VERTEX_SHADER_STRING, genericFragmentSource, shaderCallbacks);
+  }
+
+  public GlGenericDrawer(
+      String vertexShader, String genericFragmentSource, ShaderCallbacks shaderCallbacks) {
+    this.vertexShader = vertexShader;
+    this.genericFragmentSource = genericFragmentSource;
+    this.shaderCallbacks = shaderCallbacks;
+  }
+
+  // Visible for testing.
+  GlShader createShader(ShaderType shaderType) {
+    return new GlShader(
+        vertexShader, createFragmentShaderString(genericFragmentSource, shaderType));
+  }
+
+  /**
+   * Draw an OES texture frame with specified texture transformation matrix. Required resources are
+   * allocated at the first call to this function.
+   */
+  @Override
+  public void drawOes(int oesTextureId, float[] texMatrix, int frameWidth, int frameHeight,
+      int viewportX, int viewportY, int viewportWidth, int viewportHeight) {
+    prepareShader(
+        ShaderType.OES, texMatrix, frameWidth, frameHeight, viewportWidth, viewportHeight);
+    // Bind the texture.
+    GLES20.glActiveTexture(GLES20.GL_TEXTURE0);
+    GLES20.glBindTexture(GLES11Ext.GL_TEXTURE_EXTERNAL_OES, oesTextureId);
+    // Draw the texture.
+    GLES20.glViewport(viewportX, viewportY, viewportWidth, viewportHeight);
+    GLES20.glDrawArrays(GLES20.GL_TRIANGLE_STRIP, 0, 4);
+    // Unbind the texture as a precaution.
+    GLES20.glBindTexture(GLES11Ext.GL_TEXTURE_EXTERNAL_OES, 0);
+  }
+
+  /**
+   * Draw a RGB(A) texture frame with specified texture transformation matrix. Required resources
+   * are allocated at the first call to this function.
+   */
+  @Override
+  public void drawRgb(int textureId, float[] texMatrix, int frameWidth, int frameHeight,
+      int viewportX, int viewportY, int viewportWidth, int viewportHeight) {
+    prepareShader(
+        ShaderType.RGB, texMatrix, frameWidth, frameHeight, viewportWidth, viewportHeight);
+    // Bind the texture.
+    GLES20.glActiveTexture(GLES20.GL_TEXTURE0);
+    GLES20.glBindTexture(GLES20.GL_TEXTURE_2D, textureId);
+    // Draw the texture.
+    GLES20.glViewport(viewportX, viewportY, viewportWidth, viewportHeight);
+    GLES20.glDrawArrays(GLES20.GL_TRIANGLE_STRIP, 0, 4);
+    // Unbind the texture as a precaution.
+    GLES20.glBindTexture(GLES20.GL_TEXTURE_2D, 0);
+  }
+
+  /**
+   * Draw a YUV frame with specified texture transformation matrix. Required resources are allocated
+   * at the first call to this function.
+   */
+  @Override
+  public void drawYuv(int[] yuvTextures, float[] texMatrix, int frameWidth, int frameHeight,
+      int viewportX, int viewportY, int viewportWidth, int viewportHeight) {
+    prepareShader(
+        ShaderType.YUV, texMatrix, frameWidth, frameHeight, viewportWidth, viewportHeight);
+    // Bind the textures.
+    for (int i = 0; i < 3; ++i) {
+      GLES20.glActiveTexture(GLES20.GL_TEXTURE0 + i);
+      GLES20.glBindTexture(GLES20.GL_TEXTURE_2D, yuvTextures[i]);
+    }
+    // Draw the textures.
+    GLES20.glViewport(viewportX, viewportY, viewportWidth, viewportHeight);
+    GLES20.glDrawArrays(GLES20.GL_TRIANGLE_STRIP, 0, 4);
+    // Unbind the textures as a precaution.
+    for (int i = 0; i < 3; ++i) {
+      GLES20.glActiveTexture(GLES20.GL_TEXTURE0 + i);
+      GLES20.glBindTexture(GLES20.GL_TEXTURE_2D, 0);
+    }
+  }
+
+  private void prepareShader(ShaderType shaderType, float[] texMatrix, int frameWidth,
+      int frameHeight, int viewportWidth, int viewportHeight) {
+    final GlShader shader;
+    if (shaderType.equals(currentShaderType)) {
+      // Same shader type as before, reuse exising shader.
+      shader = currentShader;
+    } else {
+      // Allocate new shader.
+      currentShaderType = null;
+      if (currentShader != null) {
+        currentShader.release();
+        currentShader = null;
+      }
+
+      shader = createShader(shaderType);
+      currentShaderType = shaderType;
+      currentShader = shader;
+
+      shader.useProgram();
+      // Set input texture units.
+      if (shaderType == ShaderType.YUV) {
+        GLES20.glUniform1i(shader.getUniformLocation("y_tex"), 0);
+        GLES20.glUniform1i(shader.getUniformLocation("u_tex"), 1);
+        GLES20.glUniform1i(shader.getUniformLocation("v_tex"), 2);
+      } else {
+        GLES20.glUniform1i(shader.getUniformLocation("tex"), 0);
+      }
+
+      GlUtil.checkNoGLES2Error("Create shader");
+      shaderCallbacks.onNewShader(shader);
+      texMatrixLocation = shader.getUniformLocation(TEXTURE_MATRIX_NAME);
+      inPosLocation = shader.getAttribLocation(INPUT_VERTEX_COORDINATE_NAME);
+      inTcLocation = shader.getAttribLocation(INPUT_TEXTURE_COORDINATE_NAME);
+    }
+
+    shader.useProgram();
+
+    // Upload the vertex coordinates.
+    GLES20.glEnableVertexAttribArray(inPosLocation);
+    GLES20.glVertexAttribPointer(inPosLocation, /* size= */ 2,
+        /* type= */ GLES20.GL_FLOAT, /* normalized= */ false, /* stride= */ 0,
+        FULL_RECTANGLE_BUFFER);
+
+    // Upload the texture coordinates.
+    GLES20.glEnableVertexAttribArray(inTcLocation);
+    GLES20.glVertexAttribPointer(inTcLocation, /* size= */ 2,
+        /* type= */ GLES20.GL_FLOAT, /* normalized= */ false, /* stride= */ 0,
+        FULL_RECTANGLE_TEXTURE_BUFFER);
+
+    // Upload the texture transformation matrix.
+    GLES20.glUniformMatrix4fv(
+        texMatrixLocation, 1 /* count= */, false /* transpose= */, texMatrix, 0 /* offset= */);
+
+    // Do custom per-frame shader preparation.
+    shaderCallbacks.onPrepareShader(
+        shader, texMatrix, frameWidth, frameHeight, viewportWidth, viewportHeight);
+    GlUtil.checkNoGLES2Error("Prepare shader");
+  }
+
+  /**
+   * Release all GLES resources. This needs to be done manually, otherwise the resources are leaked.
+   */
+  @Override
+  public void release() {
+    if (currentShader != null) {
+      currentShader.release();
+      currentShader = null;
+      currentShaderType = null;
+    }
+  }
+}
diff --git a/third_party/libwebrtc/sdk/android/src/java/org/webrtc/H264Utils.java b/third_party/libwebrtc/sdk/android/src/java/org/webrtc/H264Utils.java
new file mode 100644
index 0000000000..abb79c6582
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/src/java/org/webrtc/H264Utils.java
@@ -0,0 +1,52 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+import java.util.Map;
+import java.util.HashMap;
+
+/** Container for static helper functions related to dealing with H264 codecs. */
+class H264Utils {
+  public static final String H264_FMTP_PROFILE_LEVEL_ID = "profile-level-id";
+  public static final String H264_FMTP_LEVEL_ASYMMETRY_ALLOWED = "level-asymmetry-allowed";
+  public static final String H264_FMTP_PACKETIZATION_MODE = "packetization-mode";
+
+  public static final String H264_PROFILE_CONSTRAINED_BASELINE = "42e0";
+  public static final String H264_PROFILE_CONSTRAINED_HIGH = "640c";
+  public static final String H264_LEVEL_3_1 = "1f"; // 31 in hex.
+  public static final String H264_CONSTRAINED_HIGH_3_1 =
+      H264_PROFILE_CONSTRAINED_HIGH + H264_LEVEL_3_1;
+  public static final String H264_CONSTRAINED_BASELINE_3_1 =
+      H264_PROFILE_CONSTRAINED_BASELINE + H264_LEVEL_3_1;
+
+  public static Map<String, String> getDefaultH264Params(boolean isHighProfile) {
+    final Map<String, String> params = new HashMap<>();
+    params.put(VideoCodecInfo.H264_FMTP_LEVEL_ASYMMETRY_ALLOWED, "1");
+    params.put(VideoCodecInfo.H264_FMTP_PACKETIZATION_MODE, "1");
+    params.put(VideoCodecInfo.H264_FMTP_PROFILE_LEVEL_ID,
+        isHighProfile ? VideoCodecInfo.H264_CONSTRAINED_HIGH_3_1
+                      : VideoCodecInfo.H264_CONSTRAINED_BASELINE_3_1);
+    return params;
+  }
+
+  public static VideoCodecInfo DEFAULT_H264_BASELINE_PROFILE_CODEC =
+      new VideoCodecInfo("H264", getDefaultH264Params(/* isHighProfile= */ false));
+  public static VideoCodecInfo DEFAULT_H264_HIGH_PROFILE_CODEC =
+      new VideoCodecInfo("H264", getDefaultH264Params(/* isHighProfile= */ true));
+
+  public static boolean isSameH264Profile(
+      Map<String, String> params1, Map<String, String> params2) {
+    return nativeIsSameH264Profile(params1, params2);
+  }
+
+  private static native boolean nativeIsSameH264Profile(
+      Map<String, String> params1, Map<String, String> params2);
+}
diff --git a/third_party/libwebrtc/sdk/android/src/java/org/webrtc/HardwareVideoEncoder.java b/third_party/libwebrtc/sdk/android/src/java/org/webrtc/HardwareVideoEncoder.java
new file mode 100644
index 0000000000..42a3ccfbfd
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/src/java/org/webrtc/HardwareVideoEncoder.java
@@ -0,0 +1,763 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+import android.media.MediaCodec;
+import android.media.MediaCodecInfo;
+import android.media.MediaFormat;
+import android.opengl.GLES20;
+import android.os.Build;
+import android.os.Bundle;
+import android.view.Surface;
+import androidx.annotation.Nullable;
+import java.io.IOException;
+import java.nio.ByteBuffer;
+import java.util.Map;
+import java.util.concurrent.BlockingDeque;
+import java.util.concurrent.LinkedBlockingDeque;
+import java.util.concurrent.TimeUnit;
+import org.webrtc.ThreadUtils.ThreadChecker;
+
+/**
+ * Android hardware video encoder.
+ */
+class HardwareVideoEncoder implements VideoEncoder {
+  private static final String TAG = "HardwareVideoEncoder";
+
+  // Bitrate modes - should be in sync with OMX_VIDEO_CONTROLRATETYPE defined
+  // in OMX_Video.h
+  private static final int VIDEO_ControlRateConstant = 2;
+  // Key associated with the bitrate control mode value (above). Not present as a MediaFormat
+  // constant until API level 21.
+  private static final String KEY_BITRATE_MODE = "bitrate-mode";
+
+  private static final int VIDEO_AVC_PROFILE_HIGH = 8;
+  private static final int VIDEO_AVC_LEVEL_3 = 0x100;
+
+  private static final int MAX_VIDEO_FRAMERATE = 30;
+
+  // See MAX_ENCODER_Q_SIZE in androidmediaencoder.cc.
+  private static final int MAX_ENCODER_Q_SIZE = 2;
+
+  private static final int MEDIA_CODEC_RELEASE_TIMEOUT_MS = 5000;
+  private static final int DEQUEUE_OUTPUT_BUFFER_TIMEOUT_US = 100000;
+
+  // Size of the input frames should be multiple of 16 for the H/W encoder.
+  private static final int REQUIRED_RESOLUTION_ALIGNMENT = 16;
+
+  /**
+   * Keeps track of the number of output buffers that have been passed down the pipeline and not yet
+   * released. We need to wait for this to go down to zero before operations invalidating the output
+   * buffers, i.e., stop() and getOutputBuffer().
+   */
+  private static class BusyCount {
+    private final Object countLock = new Object();
+    private int count;
+
+    public void increment() {
+      synchronized (countLock) {
+        count++;
+      }
+    }
+
+    // This method may be called on an arbitrary thread.
+    public void decrement() {
+      synchronized (countLock) {
+        count--;
+        if (count == 0) {
+          countLock.notifyAll();
+        }
+      }
+    }
+
+    // The increment and waitForZero methods are called on the same thread (deliverEncodedImage,
+    // running on the output thread). Hence, after waitForZero returns, the count will stay zero
+    // until the same thread calls increment.
+    public void waitForZero() {
+      boolean wasInterrupted = false;
+      synchronized (countLock) {
+        while (count > 0) {
+          try {
+            countLock.wait();
+          } catch (InterruptedException e) {
+            Logging.e(TAG, "Interrupted while waiting on busy count", e);
+            wasInterrupted = true;
+          }
+        }
+      }
+
+      if (wasInterrupted) {
+        Thread.currentThread().interrupt();
+      }
+    }
+  }
+  // --- Initialized on construction.
+  private final MediaCodecWrapperFactory mediaCodecWrapperFactory;
+  private final String codecName;
+  private final VideoCodecMimeType codecType;
+  private final Integer surfaceColorFormat;
+  private final Integer yuvColorFormat;
+  private final YuvFormat yuvFormat;
+  private final Map<String, String> params;
+  private final int keyFrameIntervalSec; // Base interval for generating key frames.
+  // Interval at which to force a key frame. Used to reduce color distortions caused by some
+  // Qualcomm video encoders.
+  private final long forcedKeyFrameNs;
+  private final BitrateAdjuster bitrateAdjuster;
+  // EGL context shared with the application.  Used to access texture inputs.
+  private final EglBase14.Context sharedContext;
+
+  // Drawer used to draw input textures onto the codec's input surface.
+  private final GlRectDrawer textureDrawer = new GlRectDrawer();
+  private final VideoFrameDrawer videoFrameDrawer = new VideoFrameDrawer();
+  // A queue of EncodedImage.Builders that correspond to frames in the codec.  These builders are
+  // pre-populated with all the information that can't be sent through MediaCodec.
+  private final BlockingDeque<EncodedImage.Builder> outputBuilders = new LinkedBlockingDeque<>();
+
+  private final ThreadChecker encodeThreadChecker = new ThreadChecker();
+  private final ThreadChecker outputThreadChecker = new ThreadChecker();
+  private final BusyCount outputBuffersBusyCount = new BusyCount();
+
+  // --- Set on initialize and immutable until release.
+  private Callback callback;
+  private boolean automaticResizeOn;
+
+  // --- Valid and immutable while an encoding session is running.
+  @Nullable private MediaCodecWrapper codec;
+  // Thread that delivers encoded frames to the user callback.
+  @Nullable private Thread outputThread;
+
+  // EGL base wrapping the shared texture context.  Holds hooks to both the shared context and the
+  // input surface.  Making this base current allows textures from the context to be drawn onto the
+  // surface.
+  @Nullable private EglBase14 textureEglBase;
+  // Input surface for the codec.  The encoder will draw input textures onto this surface.
+  @Nullable private Surface textureInputSurface;
+
+  private int width;
+  private int height;
+  // Y-plane strides in the encoder's input
+  private int stride;
+  // Y-plane slice-height in the encoder's input
+  private int sliceHeight;
+  private boolean useSurfaceMode;
+
+  // --- Only accessed from the encoding thread.
+  // Presentation timestamp of next frame to encode.
+  private long nextPresentationTimestampUs;
+  // Presentation timestamp of the last requested (or forced) key frame.
+  private long lastKeyFrameNs;
+
+  // --- Only accessed on the output thread.
+  // Contents of the last observed config frame output by the MediaCodec. Used by H.264.
+  @Nullable private ByteBuffer configBuffer;
+  private int adjustedBitrate;
+
+  // Whether the encoder is running.  Volatile so that the output thread can watch this value and
+  // exit when the encoder stops.
+  private volatile boolean running;
+  // Any exception thrown during shutdown.  The output thread releases the MediaCodec and uses this
+  // value to send exceptions thrown during release back to the encoder thread.
+  @Nullable private volatile Exception shutdownException;
+
+  /**
+   * Creates a new HardwareVideoEncoder with the given codecName, codecType, colorFormat, key frame
+   * intervals, and bitrateAdjuster.
+   *
+   * @param codecName the hardware codec implementation to use
+   * @param codecType the type of the given video codec (eg. VP8, VP9, H264 or AV1)
+   * @param surfaceColorFormat color format for surface mode or null if not available
+   * @param yuvColorFormat color format for bytebuffer mode
+   * @param keyFrameIntervalSec interval in seconds between key frames; used to initialize the codec
+   * @param forceKeyFrameIntervalMs interval at which to force a key frame if one is not requested;
+   *     used to reduce distortion caused by some codec implementations
+   * @param bitrateAdjuster algorithm used to correct codec implementations that do not produce the
+   *     desired bitrates
+   * @throws IllegalArgumentException if colorFormat is unsupported
+   */
+  public HardwareVideoEncoder(MediaCodecWrapperFactory mediaCodecWrapperFactory, String codecName,
+      VideoCodecMimeType codecType, Integer surfaceColorFormat, Integer yuvColorFormat,
+      Map<String, String> params, int keyFrameIntervalSec, int forceKeyFrameIntervalMs,
+      BitrateAdjuster bitrateAdjuster, EglBase14.Context sharedContext) {
+    this.mediaCodecWrapperFactory = mediaCodecWrapperFactory;
+    this.codecName = codecName;
+    this.codecType = codecType;
+    this.surfaceColorFormat = surfaceColorFormat;
+    this.yuvColorFormat = yuvColorFormat;
+    this.yuvFormat = YuvFormat.valueOf(yuvColorFormat);
+    this.params = params;
+    this.keyFrameIntervalSec = keyFrameIntervalSec;
+    this.forcedKeyFrameNs = TimeUnit.MILLISECONDS.toNanos(forceKeyFrameIntervalMs);
+    this.bitrateAdjuster = bitrateAdjuster;
+    this.sharedContext = sharedContext;
+
+    // Allow construction on a different thread.
+    encodeThreadChecker.detachThread();
+  }
+
+  @Override
+  public VideoCodecStatus initEncode(Settings settings, Callback callback) {
+    encodeThreadChecker.checkIsOnValidThread();
+
+    this.callback = callback;
+    automaticResizeOn = settings.automaticResizeOn;
+
+    if (settings.width % REQUIRED_RESOLUTION_ALIGNMENT != 0
+        || settings.height % REQUIRED_RESOLUTION_ALIGNMENT != 0) {
+      Logging.e(TAG, "MediaCodec is only tested with resolutions that are 16x16 aligned.");
+      return VideoCodecStatus.ERR_SIZE;
+    }
+    this.width = settings.width;
+    this.height = settings.height;
+    useSurfaceMode = canUseSurface();
+
+    if (settings.startBitrate != 0 && settings.maxFramerate != 0) {
+      bitrateAdjuster.setTargets(settings.startBitrate * 1000, settings.maxFramerate);
+    }
+    adjustedBitrate = bitrateAdjuster.getAdjustedBitrateBps();
+
+    Logging.d(TAG,
+        "initEncode: " + width + " x " + height + ". @ " + settings.startBitrate
+            + "kbps. Fps: " + settings.maxFramerate + " Use surface mode: " + useSurfaceMode);
+    return initEncodeInternal();
+  }
+
+  private VideoCodecStatus initEncodeInternal() {
+    encodeThreadChecker.checkIsOnValidThread();
+
+    nextPresentationTimestampUs = 0;
+    lastKeyFrameNs = -1;
+
+    try {
+      codec = mediaCodecWrapperFactory.createByCodecName(codecName);
+    } catch (IOException | IllegalArgumentException e) {
+      Logging.e(TAG, "Cannot create media encoder " + codecName);
+      return VideoCodecStatus.FALLBACK_SOFTWARE;
+    }
+
+    final int colorFormat = useSurfaceMode ? surfaceColorFormat : yuvColorFormat;
+    try {
+      MediaFormat format = MediaFormat.createVideoFormat(codecType.mimeType(), width, height);
+      format.setInteger(MediaFormat.KEY_BIT_RATE, adjustedBitrate);
+      format.setInteger(KEY_BITRATE_MODE, VIDEO_ControlRateConstant);
+      format.setInteger(MediaFormat.KEY_COLOR_FORMAT, colorFormat);
+      format.setFloat(
+          MediaFormat.KEY_FRAME_RATE, (float) bitrateAdjuster.getAdjustedFramerateFps());
+      format.setInteger(MediaFormat.KEY_I_FRAME_INTERVAL, keyFrameIntervalSec);
+      if (codecType == VideoCodecMimeType.H264) {
+        String profileLevelId = params.get(VideoCodecInfo.H264_FMTP_PROFILE_LEVEL_ID);
+        if (profileLevelId == null) {
+          profileLevelId = VideoCodecInfo.H264_CONSTRAINED_BASELINE_3_1;
+        }
+        switch (profileLevelId) {
+          case VideoCodecInfo.H264_CONSTRAINED_HIGH_3_1:
+            format.setInteger("profile", VIDEO_AVC_PROFILE_HIGH);
+            format.setInteger("level", VIDEO_AVC_LEVEL_3);
+            break;
+          case VideoCodecInfo.H264_CONSTRAINED_BASELINE_3_1:
+            break;
+          default:
+            Logging.w(TAG, "Unknown profile level id: " + profileLevelId);
+        }
+      }
+      Logging.d(TAG, "Format: " + format);
+      codec.configure(
+          format, null /* surface */, null /* crypto */, MediaCodec.CONFIGURE_FLAG_ENCODE);
+
+      if (useSurfaceMode) {
+        textureEglBase = EglBase.createEgl14(sharedContext, EglBase.CONFIG_RECORDABLE);
+        textureInputSurface = codec.createInputSurface();
+        textureEglBase.createSurface(textureInputSurface);
+        textureEglBase.makeCurrent();
+      }
+
+      MediaFormat inputFormat = codec.getInputFormat();
+      stride = getStride(inputFormat, width);
+      sliceHeight = getSliceHeight(inputFormat, height);
+
+      codec.start();
+    } catch (IllegalStateException e) {
+      Logging.e(TAG, "initEncodeInternal failed", e);
+      release();
+      return VideoCodecStatus.FALLBACK_SOFTWARE;
+    }
+
+    running = true;
+    outputThreadChecker.detachThread();
+    outputThread = createOutputThread();
+    outputThread.start();
+
+    return VideoCodecStatus.OK;
+  }
+
+  @Override
+  public VideoCodecStatus release() {
+    encodeThreadChecker.checkIsOnValidThread();
+
+    final VideoCodecStatus returnValue;
+    if (outputThread == null) {
+      returnValue = VideoCodecStatus.OK;
+    } else {
+      // The outputThread actually stops and releases the codec once running is false.
+      running = false;
+      if (!ThreadUtils.joinUninterruptibly(outputThread, MEDIA_CODEC_RELEASE_TIMEOUT_MS)) {
+        Logging.e(TAG, "Media encoder release timeout");
+        returnValue = VideoCodecStatus.TIMEOUT;
+      } else if (shutdownException != null) {
+        // Log the exception and turn it into an error.
+        Logging.e(TAG, "Media encoder release exception", shutdownException);
+        returnValue = VideoCodecStatus.ERROR;
+      } else {
+        returnValue = VideoCodecStatus.OK;
+      }
+    }
+
+    textureDrawer.release();
+    videoFrameDrawer.release();
+    if (textureEglBase != null) {
+      textureEglBase.release();
+      textureEglBase = null;
+    }
+    if (textureInputSurface != null) {
+      textureInputSurface.release();
+      textureInputSurface = null;
+    }
+    outputBuilders.clear();
+
+    codec = null;
+    outputThread = null;
+
+    // Allow changing thread after release.
+    encodeThreadChecker.detachThread();
+
+    return returnValue;
+  }
+
+  @Override
+  public VideoCodecStatus encode(VideoFrame videoFrame, EncodeInfo encodeInfo) {
+    encodeThreadChecker.checkIsOnValidThread();
+    if (codec == null) {
+      return VideoCodecStatus.UNINITIALIZED;
+    }
+
+    final VideoFrame.Buffer videoFrameBuffer = videoFrame.getBuffer();
+    final boolean isTextureBuffer = videoFrameBuffer instanceof VideoFrame.TextureBuffer;
+
+    // If input resolution changed, restart the codec with the new resolution.
+    final int frameWidth = videoFrame.getBuffer().getWidth();
+    final int frameHeight = videoFrame.getBuffer().getHeight();
+    final boolean shouldUseSurfaceMode = canUseSurface() && isTextureBuffer;
+    if (frameWidth != width || frameHeight != height || shouldUseSurfaceMode != useSurfaceMode) {
+      VideoCodecStatus status = resetCodec(frameWidth, frameHeight, shouldUseSurfaceMode);
+      if (status != VideoCodecStatus.OK) {
+        return status;
+      }
+    }
+
+    if (outputBuilders.size() > MAX_ENCODER_Q_SIZE) {
+      // Too many frames in the encoder.  Drop this frame.
+      Logging.e(TAG, "Dropped frame, encoder queue full");
+      return VideoCodecStatus.NO_OUTPUT; // See webrtc bug 2887.
+    }
+
+    boolean requestedKeyFrame = false;
+    for (EncodedImage.FrameType frameType : encodeInfo.frameTypes) {
+      if (frameType == EncodedImage.FrameType.VideoFrameKey) {
+        requestedKeyFrame = true;
+      }
+    }
+
+    if (requestedKeyFrame || shouldForceKeyFrame(videoFrame.getTimestampNs())) {
+      requestKeyFrame(videoFrame.getTimestampNs());
+    }
+
+    // Number of bytes in the video buffer. Y channel is sampled at one byte per pixel; U and V are
+    // subsampled at one byte per four pixels.
+    int bufferSize = videoFrameBuffer.getHeight() * videoFrameBuffer.getWidth() * 3 / 2;
+    EncodedImage.Builder builder = EncodedImage.builder()
+                                       .setCaptureTimeNs(videoFrame.getTimestampNs())
+                                       .setEncodedWidth(videoFrame.getBuffer().getWidth())
+                                       .setEncodedHeight(videoFrame.getBuffer().getHeight())
+                                       .setRotation(videoFrame.getRotation());
+    outputBuilders.offer(builder);
+
+    long presentationTimestampUs = nextPresentationTimestampUs;
+    // Round frame duration down to avoid bitrate overshoot.
+    long frameDurationUs =
+        (long) (TimeUnit.SECONDS.toMicros(1) / bitrateAdjuster.getAdjustedFramerateFps());
+    nextPresentationTimestampUs += frameDurationUs;
+
+    final VideoCodecStatus returnValue;
+    if (useSurfaceMode) {
+      returnValue = encodeTextureBuffer(videoFrame, presentationTimestampUs);
+    } else {
+      returnValue =
+          encodeByteBuffer(videoFrame, presentationTimestampUs, videoFrameBuffer, bufferSize);
+    }
+
+    // Check if the queue was successful.
+    if (returnValue != VideoCodecStatus.OK) {
+      // Keep the output builders in sync with buffers in the codec.
+      outputBuilders.pollLast();
+    }
+
+    return returnValue;
+  }
+
+  private VideoCodecStatus encodeTextureBuffer(
+      VideoFrame videoFrame, long presentationTimestampUs) {
+    encodeThreadChecker.checkIsOnValidThread();
+    try {
+      // TODO(perkj): glClear() shouldn't be necessary since every pixel is covered anyway,
+      // but it's a workaround for bug webrtc:5147.
+      GLES20.glClear(GLES20.GL_COLOR_BUFFER_BIT);
+      // It is not necessary to release this frame because it doesn't own the buffer.
+      VideoFrame derotatedFrame =
+          new VideoFrame(videoFrame.getBuffer(), 0 /* rotation */, videoFrame.getTimestampNs());
+      videoFrameDrawer.drawFrame(derotatedFrame, textureDrawer, null /* additionalRenderMatrix */);
+      textureEglBase.swapBuffers(TimeUnit.MICROSECONDS.toNanos(presentationTimestampUs));
+    } catch (RuntimeException e) {
+      Logging.e(TAG, "encodeTexture failed", e);
+      return VideoCodecStatus.ERROR;
+    }
+    return VideoCodecStatus.OK;
+  }
+
+  private VideoCodecStatus encodeByteBuffer(VideoFrame videoFrame, long presentationTimestampUs,
+      VideoFrame.Buffer videoFrameBuffer, int bufferSize) {
+    encodeThreadChecker.checkIsOnValidThread();
+    // No timeout.  Don't block for an input buffer, drop frames if the encoder falls behind.
+    int index;
+    try {
+      index = codec.dequeueInputBuffer(0 /* timeout */);
+    } catch (IllegalStateException e) {
+      Logging.e(TAG, "dequeueInputBuffer failed", e);
+      return VideoCodecStatus.ERROR;
+    }
+
+    if (index == -1) {
+      // Encoder is falling behind.  No input buffers available.  Drop the frame.
+      Logging.d(TAG, "Dropped frame, no input buffers available");
+      return VideoCodecStatus.NO_OUTPUT; // See webrtc bug 2887.
+    }
+
+    ByteBuffer buffer;
+    try {
+      buffer = codec.getInputBuffer(index);
+    } catch (IllegalStateException e) {
+      Logging.e(TAG, "getInputBuffer with index=" + index + " failed", e);
+      return VideoCodecStatus.ERROR;
+    }
+    fillInputBuffer(buffer, videoFrameBuffer);
+
+    try {
+      codec.queueInputBuffer(
+          index, 0 /* offset */, bufferSize, presentationTimestampUs, 0 /* flags */);
+    } catch (IllegalStateException e) {
+      Logging.e(TAG, "queueInputBuffer failed", e);
+      // IllegalStateException thrown when the codec is in the wrong state.
+      return VideoCodecStatus.ERROR;
+    }
+    return VideoCodecStatus.OK;
+  }
+
+  @Override
+  public VideoCodecStatus setRateAllocation(BitrateAllocation bitrateAllocation, int framerate) {
+    encodeThreadChecker.checkIsOnValidThread();
+    if (framerate > MAX_VIDEO_FRAMERATE) {
+      framerate = MAX_VIDEO_FRAMERATE;
+    }
+    bitrateAdjuster.setTargets(bitrateAllocation.getSum(), framerate);
+    return VideoCodecStatus.OK;
+  }
+
+  @Override
+  public VideoCodecStatus setRates(RateControlParameters rcParameters) {
+    encodeThreadChecker.checkIsOnValidThread();
+    bitrateAdjuster.setTargets(rcParameters.bitrate.getSum(), rcParameters.framerateFps);
+    return VideoCodecStatus.OK;
+  }
+
+  @Override
+  public ScalingSettings getScalingSettings() {
+    encodeThreadChecker.checkIsOnValidThread();
+    if (automaticResizeOn) {
+      if (codecType == VideoCodecMimeType.VP8) {
+        final int kLowVp8QpThreshold = 29;
+        final int kHighVp8QpThreshold = 95;
+        return new ScalingSettings(kLowVp8QpThreshold, kHighVp8QpThreshold);
+      } else if (codecType == VideoCodecMimeType.H264) {
+        final int kLowH264QpThreshold = 24;
+        final int kHighH264QpThreshold = 37;
+        return new ScalingSettings(kLowH264QpThreshold, kHighH264QpThreshold);
+      }
+    }
+    return ScalingSettings.OFF;
+  }
+
+  @Override
+  public String getImplementationName() {
+    return codecName;
+  }
+
+  @Override
+  public EncoderInfo getEncoderInfo() {
+    // Since our MediaCodec is guaranteed to encode 16-pixel-aligned frames only, we set alignment
+    // value to be 16. Additionally, this encoder produces a single stream. So it should not require
+    // alignment for all layers.
+    return new EncoderInfo(
+        /* requestedResolutionAlignment= */ REQUIRED_RESOLUTION_ALIGNMENT,
+        /* applyAlignmentToAllSimulcastLayers= */ false);
+  }
+
+  private VideoCodecStatus resetCodec(int newWidth, int newHeight, boolean newUseSurfaceMode) {
+    encodeThreadChecker.checkIsOnValidThread();
+    VideoCodecStatus status = release();
+    if (status != VideoCodecStatus.OK) {
+      return status;
+    }
+
+    if (newWidth % REQUIRED_RESOLUTION_ALIGNMENT != 0
+        || newHeight % REQUIRED_RESOLUTION_ALIGNMENT != 0) {
+      Logging.e(TAG, "MediaCodec is only tested with resolutions that are 16x16 aligned.");
+      return VideoCodecStatus.ERR_SIZE;
+    }
+    width = newWidth;
+    height = newHeight;
+    useSurfaceMode = newUseSurfaceMode;
+    return initEncodeInternal();
+  }
+
+  private boolean shouldForceKeyFrame(long presentationTimestampNs) {
+    encodeThreadChecker.checkIsOnValidThread();
+    return forcedKeyFrameNs > 0 && presentationTimestampNs > lastKeyFrameNs + forcedKeyFrameNs;
+  }
+
+  private void requestKeyFrame(long presentationTimestampNs) {
+    encodeThreadChecker.checkIsOnValidThread();
+    // Ideally MediaCodec would honor BUFFER_FLAG_SYNC_FRAME so we could
+    // indicate this in queueInputBuffer() below and guarantee _this_ frame
+    // be encoded as a key frame, but sadly that flag is ignored.  Instead,
+    // we request a key frame "soon".
+    try {
+      Bundle b = new Bundle();
+      b.putInt(MediaCodec.PARAMETER_KEY_REQUEST_SYNC_FRAME, 0);
+      codec.setParameters(b);
+    } catch (IllegalStateException e) {
+      Logging.e(TAG, "requestKeyFrame failed", e);
+      return;
+    }
+    lastKeyFrameNs = presentationTimestampNs;
+  }
+
+  private Thread createOutputThread() {
+    return new Thread() {
+      @Override
+      public void run() {
+        while (running) {
+          deliverEncodedImage();
+        }
+        releaseCodecOnOutputThread();
+      }
+    };
+  }
+
+  // Visible for testing.
+  protected void deliverEncodedImage() {
+    outputThreadChecker.checkIsOnValidThread();
+    try {
+      MediaCodec.BufferInfo info = new MediaCodec.BufferInfo();
+      int index = codec.dequeueOutputBuffer(info, DEQUEUE_OUTPUT_BUFFER_TIMEOUT_US);
+      if (index < 0) {
+        if (index == MediaCodec.INFO_OUTPUT_BUFFERS_CHANGED) {
+          outputBuffersBusyCount.waitForZero();
+        }
+        return;
+      }
+
+      ByteBuffer codecOutputBuffer = codec.getOutputBuffer(index);
+      codecOutputBuffer.position(info.offset);
+      codecOutputBuffer.limit(info.offset + info.size);
+
+      if ((info.flags & MediaCodec.BUFFER_FLAG_CODEC_CONFIG) != 0) {
+        Logging.d(TAG, "Config frame generated. Offset: " + info.offset + ". Size: " + info.size);
+        configBuffer = ByteBuffer.allocateDirect(info.size);
+        configBuffer.put(codecOutputBuffer);
+      } else {
+        bitrateAdjuster.reportEncodedFrame(info.size);
+        if (adjustedBitrate != bitrateAdjuster.getAdjustedBitrateBps()) {
+          updateBitrate();
+        }
+
+        final boolean isKeyFrame = (info.flags & MediaCodec.BUFFER_FLAG_SYNC_FRAME) != 0;
+        if (isKeyFrame) {
+          Logging.d(TAG, "Sync frame generated");
+        }
+
+        final ByteBuffer frameBuffer;
+        if (isKeyFrame && codecType == VideoCodecMimeType.H264) {
+          Logging.d(TAG,
+              "Prepending config frame of size " + configBuffer.capacity()
+                  + " to output buffer with offset " + info.offset + ", size " + info.size);
+          // For H.264 key frame prepend SPS and PPS NALs at the start.
+          frameBuffer = ByteBuffer.allocateDirect(info.size + configBuffer.capacity());
+          configBuffer.rewind();
+          frameBuffer.put(configBuffer);
+          frameBuffer.put(codecOutputBuffer);
+          frameBuffer.rewind();
+        } else {
+          frameBuffer = codecOutputBuffer.slice();
+        }
+
+        final EncodedImage.FrameType frameType = isKeyFrame
+            ? EncodedImage.FrameType.VideoFrameKey
+            : EncodedImage.FrameType.VideoFrameDelta;
+
+        outputBuffersBusyCount.increment();
+        EncodedImage.Builder builder = outputBuilders.poll();
+        EncodedImage encodedImage = builder
+                                        .setBuffer(frameBuffer,
+                                            () -> {
+                                              // This callback should not throw any exceptions since
+                                              // it may be called on an arbitrary thread.
+                                              // Check bug webrtc:11230 for more details.
+                                              try {
+                                                codec.releaseOutputBuffer(index, false);
+                                              } catch (Exception e) {
+                                                Logging.e(TAG, "releaseOutputBuffer failed", e);
+                                              }
+                                              outputBuffersBusyCount.decrement();
+                                            })
+                                        .setFrameType(frameType)
+                                        .createEncodedImage();
+        // TODO(mellem):  Set codec-specific info.
+        callback.onEncodedFrame(encodedImage, new CodecSpecificInfo());
+        // Note that the callback may have retained the image.
+        encodedImage.release();
+      }
+    } catch (IllegalStateException e) {
+      Logging.e(TAG, "deliverOutput failed", e);
+    }
+  }
+
+  private void releaseCodecOnOutputThread() {
+    outputThreadChecker.checkIsOnValidThread();
+    Logging.d(TAG, "Releasing MediaCodec on output thread");
+    outputBuffersBusyCount.waitForZero();
+    try {
+      codec.stop();
+    } catch (Exception e) {
+      Logging.e(TAG, "Media encoder stop failed", e);
+    }
+    try {
+      codec.release();
+    } catch (Exception e) {
+      Logging.e(TAG, "Media encoder release failed", e);
+      // Propagate exceptions caught during release back to the main thread.
+      shutdownException = e;
+    }
+    configBuffer = null;
+    Logging.d(TAG, "Release on output thread done");
+  }
+
+  private VideoCodecStatus updateBitrate() {
+    outputThreadChecker.checkIsOnValidThread();
+    adjustedBitrate = bitrateAdjuster.getAdjustedBitrateBps();
+    try {
+      Bundle params = new Bundle();
+      params.putInt(MediaCodec.PARAMETER_KEY_VIDEO_BITRATE, adjustedBitrate);
+      codec.setParameters(params);
+      return VideoCodecStatus.OK;
+    } catch (IllegalStateException e) {
+      Logging.e(TAG, "updateBitrate failed", e);
+      return VideoCodecStatus.ERROR;
+    }
+  }
+
+  private boolean canUseSurface() {
+    return sharedContext != null && surfaceColorFormat != null;
+  }
+
+  private static int getStride(MediaFormat inputFormat, int width) {
+    if (Build.VERSION.SDK_INT >= Build.VERSION_CODES.M && inputFormat != null
+        && inputFormat.containsKey(MediaFormat.KEY_STRIDE)) {
+      return inputFormat.getInteger(MediaFormat.KEY_STRIDE);
+    }
+    return width;
+  }
+
+  private static int getSliceHeight(MediaFormat inputFormat, int height) {
+    if (Build.VERSION.SDK_INT >= Build.VERSION_CODES.M && inputFormat != null
+        && inputFormat.containsKey(MediaFormat.KEY_SLICE_HEIGHT)) {
+      return inputFormat.getInteger(MediaFormat.KEY_SLICE_HEIGHT);
+    }
+    return height;
+  }
+
+  // Visible for testing.
+  protected void fillInputBuffer(ByteBuffer buffer, VideoFrame.Buffer videoFrameBuffer) {
+    yuvFormat.fillBuffer(buffer, videoFrameBuffer, stride, sliceHeight);
+  }
+
+  /**
+   * Enumeration of supported YUV color formats used for MediaCodec's input.
+   */
+  private enum YuvFormat {
+    I420 {
+      @Override
+      void fillBuffer(
+          ByteBuffer dstBuffer, VideoFrame.Buffer srcBuffer, int dstStrideY, int dstSliceHeightY) {
+        /*
+         * According to the docs in Android MediaCodec, the stride of the U and V planes can be
+         * calculated based on the color format, though it is generally undefined and depends on the
+         * device and release.
+         * <p/> Assuming the width and height, dstStrideY and dstSliceHeightY are
+         * even, it works fine when we define the stride and slice-height of the dst U/V plane to be
+         * half of the dst Y plane.
+         */
+        int dstStrideU = dstStrideY / 2;
+        int dstSliceHeight = dstSliceHeightY / 2;
+        VideoFrame.I420Buffer i420 = srcBuffer.toI420();
+        YuvHelper.I420Copy(i420.getDataY(), i420.getStrideY(), i420.getDataU(), i420.getStrideU(),
+            i420.getDataV(), i420.getStrideV(), dstBuffer, i420.getWidth(), i420.getHeight(),
+            dstStrideY, dstSliceHeightY, dstStrideU, dstSliceHeight);
+        i420.release();
+      }
+    },
+    NV12 {
+      @Override
+      void fillBuffer(
+          ByteBuffer dstBuffer, VideoFrame.Buffer srcBuffer, int dstStrideY, int dstSliceHeightY) {
+        VideoFrame.I420Buffer i420 = srcBuffer.toI420();
+        YuvHelper.I420ToNV12(i420.getDataY(), i420.getStrideY(), i420.getDataU(), i420.getStrideU(),
+            i420.getDataV(), i420.getStrideV(), dstBuffer, i420.getWidth(), i420.getHeight(),
+            dstStrideY, dstSliceHeightY);
+        i420.release();
+      }
+    };
+
+    abstract void fillBuffer(
+        ByteBuffer dstBuffer, VideoFrame.Buffer srcBuffer, int dstStrideY, int dstSliceHeightY);
+
+    static YuvFormat valueOf(int colorFormat) {
+      switch (colorFormat) {
+        case MediaCodecInfo.CodecCapabilities.COLOR_FormatYUV420Planar:
+          return I420;
+        case MediaCodecInfo.CodecCapabilities.COLOR_FormatYUV420SemiPlanar:
+        case MediaCodecInfo.CodecCapabilities.COLOR_QCOM_FormatYUV420SemiPlanar:
+        case MediaCodecUtils.COLOR_QCOM_FORMATYUV420PackedSemiPlanar32m:
+          return NV12;
+        default:
+          throw new IllegalArgumentException("Unsupported colorFormat: " + colorFormat);
+      }
+    }
+  }
+}
diff --git a/third_party/libwebrtc/sdk/android/src/java/org/webrtc/Histogram.java b/third_party/libwebrtc/sdk/android/src/java/org/webrtc/Histogram.java
new file mode 100644
index 0000000000..c1d2d61a71
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/src/java/org/webrtc/Histogram.java
@@ -0,0 +1,39 @@
+/*
+ *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+/**
+ * Class for holding the native pointer of a histogram. Since there is no way to destroy a
+ * histogram, please don't create unnecessary instances of this object. This class is thread safe.
+ *
+ * Usage example:
+ * private static final Histogram someMetricHistogram =
+ *     Histogram.createCounts("WebRTC.Video.SomeMetric", 1, 10000, 50);
+ * someMetricHistogram.addSample(someVariable);
+ */
+class Histogram {
+  private final long handle;
+
+  private Histogram(long handle) {
+    this.handle = handle;
+  }
+
+  static public Histogram createCounts(String name, int min, int max, int bucketCount) {
+    return new Histogram(0);
+  }
+
+  static public Histogram createEnumeration(String name, int max) {
+    return new Histogram(0);
+  }
+
+  public void addSample(int sample) {
+  }
+}
diff --git a/third_party/libwebrtc/sdk/android/src/java/org/webrtc/JNILogging.java b/third_party/libwebrtc/sdk/android/src/java/org/webrtc/JNILogging.java
new file mode 100644
index 0000000000..f391db61a1
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/src/java/org/webrtc/JNILogging.java
@@ -0,0 +1,28 @@
+/*
+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+import org.webrtc.CalledByNative;
+import org.webrtc.Loggable;
+import org.webrtc.Logging.Severity;
+
+class JNILogging {
+  private final Loggable loggable;
+
+  public JNILogging(Loggable loggable) {
+    this.loggable = loggable;
+  }
+
+  @CalledByNative
+  public void logToInjectable(String message, Integer severity, String tag) {
+    loggable.onLogMessage(message, Severity.values()[severity], tag);
+  }
+}
diff --git a/third_party/libwebrtc/sdk/android/src/java/org/webrtc/JniCommon.java b/third_party/libwebrtc/sdk/android/src/java/org/webrtc/JniCommon.java
new file mode 100644
index 0000000000..e1b2e513d7
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/src/java/org/webrtc/JniCommon.java
@@ -0,0 +1,23 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+import java.nio.ByteBuffer;
+
+/** Class with static JNI helper functions that are used in many places. */
+public class JniCommon {
+  /** Functions to increment/decrement an rtc::RefCountInterface pointer. */
+  public static native void nativeAddRef(long refCountedPointer);
+  public static native void nativeReleaseRef(long refCountedPointer);
+
+  public static native ByteBuffer nativeAllocateByteBuffer(int size);
+  public static native void nativeFreeByteBuffer(ByteBuffer buffer);
+}
diff --git a/third_party/libwebrtc/sdk/android/src/java/org/webrtc/MediaCodecUtils.java b/third_party/libwebrtc/sdk/android/src/java/org/webrtc/MediaCodecUtils.java
new file mode 100644
index 0000000000..d5ccae9688
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/src/java/org/webrtc/MediaCodecUtils.java
@@ -0,0 +1,129 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+import android.annotation.TargetApi;
+import android.media.MediaCodecInfo;
+import android.media.MediaCodecInfo.CodecCapabilities;
+import android.os.Build;
+import androidx.annotation.Nullable;
+import java.util.HashMap;
+import java.util.Map;
+
+/** Container class for static constants and helpers used with MediaCodec. */
+// We are forced to use the old API because we want to support API level < 21.
+@SuppressWarnings("deprecation")
+class MediaCodecUtils {
+  private static final String TAG = "MediaCodecUtils";
+
+  // Prefixes for supported hardware encoder/decoder component names.
+  static final String EXYNOS_PREFIX = "OMX.Exynos.";
+  static final String INTEL_PREFIX = "OMX.Intel.";
+  static final String NVIDIA_PREFIX = "OMX.Nvidia.";
+  static final String QCOM_PREFIX = "OMX.qcom.";
+  static final String[] SOFTWARE_IMPLEMENTATION_PREFIXES = {
+      "OMX.google.", "OMX.SEC.", "c2.android"};
+
+  // NV12 color format supported by QCOM codec, but not declared in MediaCodec -
+  // see /hardware/qcom/media/mm-core/inc/OMX_QCOMExtns.h
+  static final int COLOR_QCOM_FORMATYVU420PackedSemiPlanar32m4ka = 0x7FA30C01;
+  static final int COLOR_QCOM_FORMATYVU420PackedSemiPlanar16m4ka = 0x7FA30C02;
+  static final int COLOR_QCOM_FORMATYVU420PackedSemiPlanar64x32Tile2m8ka = 0x7FA30C03;
+  static final int COLOR_QCOM_FORMATYUV420PackedSemiPlanar32m = 0x7FA30C04;
+
+  // Color formats supported by hardware decoder - in order of preference.
+  static final int[] DECODER_COLOR_FORMATS = new int[] {CodecCapabilities.COLOR_FormatYUV420Planar,
+      CodecCapabilities.COLOR_FormatYUV420SemiPlanar,
+      CodecCapabilities.COLOR_QCOM_FormatYUV420SemiPlanar,
+      MediaCodecUtils.COLOR_QCOM_FORMATYVU420PackedSemiPlanar32m4ka,
+      MediaCodecUtils.COLOR_QCOM_FORMATYVU420PackedSemiPlanar16m4ka,
+      MediaCodecUtils.COLOR_QCOM_FORMATYVU420PackedSemiPlanar64x32Tile2m8ka,
+      MediaCodecUtils.COLOR_QCOM_FORMATYUV420PackedSemiPlanar32m};
+
+  // Color formats supported by hardware encoder - in order of preference.
+  static final int[] ENCODER_COLOR_FORMATS = {
+      MediaCodecInfo.CodecCapabilities.COLOR_FormatYUV420Planar,
+      MediaCodecInfo.CodecCapabilities.COLOR_FormatYUV420SemiPlanar,
+      MediaCodecInfo.CodecCapabilities.COLOR_QCOM_FormatYUV420SemiPlanar,
+      MediaCodecUtils.COLOR_QCOM_FORMATYUV420PackedSemiPlanar32m};
+
+  // Color formats supported by texture mode encoding - in order of preference.
+  static final int[] TEXTURE_COLOR_FORMATS =
+      new int[] {MediaCodecInfo.CodecCapabilities.COLOR_FormatSurface};
+
+  static @Nullable Integer selectColorFormat(
+      int[] supportedColorFormats, CodecCapabilities capabilities) {
+    for (int supportedColorFormat : supportedColorFormats) {
+      for (int codecColorFormat : capabilities.colorFormats) {
+        if (codecColorFormat == supportedColorFormat) {
+          return codecColorFormat;
+        }
+      }
+    }
+    return null;
+  }
+
+  static boolean codecSupportsType(MediaCodecInfo info, VideoCodecMimeType type) {
+    for (String mimeType : info.getSupportedTypes()) {
+      if (type.mimeType().equals(mimeType)) {
+        return true;
+      }
+    }
+    return false;
+  }
+
+  static Map<String, String> getCodecProperties(VideoCodecMimeType type, boolean highProfile) {
+    switch (type) {
+      case VP8:
+      case VP9:
+      case AV1:
+        return new HashMap<String, String>();
+      case H264:
+        return H264Utils.getDefaultH264Params(highProfile);
+      default:
+        throw new IllegalArgumentException("Unsupported codec: " + type);
+    }
+  }
+
+  static boolean isHardwareAccelerated(MediaCodecInfo info) {
+    if (Build.VERSION.SDK_INT >= Build.VERSION_CODES.Q) {
+      return isHardwareAcceleratedQOrHigher(info);
+    }
+    return !isSoftwareOnly(info);
+  }
+
+  @TargetApi(29)
+  private static boolean isHardwareAcceleratedQOrHigher(android.media.MediaCodecInfo codecInfo) {
+    return codecInfo.isHardwareAccelerated();
+  }
+
+  static boolean isSoftwareOnly(android.media.MediaCodecInfo codecInfo) {
+    if (Build.VERSION.SDK_INT >= Build.VERSION_CODES.Q) {
+      return isSoftwareOnlyQOrHigher(codecInfo);
+    }
+    String name = codecInfo.getName();
+    for (String prefix : SOFTWARE_IMPLEMENTATION_PREFIXES) {
+      if (name.startsWith(prefix)) {
+        return true;
+      }
+    }
+    return false;
+  }
+
+  @TargetApi(29)
+  private static boolean isSoftwareOnlyQOrHigher(android.media.MediaCodecInfo codecInfo) {
+    return codecInfo.isSoftwareOnly();
+  }
+
+  private MediaCodecUtils() {
+    // This class should not be instantiated.
+  }
+}
diff --git a/third_party/libwebrtc/sdk/android/src/java/org/webrtc/MediaCodecVideoDecoderFactory.java b/third_party/libwebrtc/sdk/android/src/java/org/webrtc/MediaCodecVideoDecoderFactory.java
new file mode 100644
index 0000000000..bf591dda26
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/src/java/org/webrtc/MediaCodecVideoDecoderFactory.java
@@ -0,0 +1,139 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+import static org.webrtc.MediaCodecUtils.EXYNOS_PREFIX;
+import static org.webrtc.MediaCodecUtils.QCOM_PREFIX;
+
+import android.media.MediaCodecInfo;
+import android.media.MediaCodecInfo.CodecCapabilities;
+import android.media.MediaCodecList;
+import android.os.Build;
+import androidx.annotation.Nullable;
+import java.util.ArrayList;
+import java.util.List;
+
+/** Factory for decoders backed by Android MediaCodec API. */
+@SuppressWarnings("deprecation") // API level 16 requires use of deprecated methods.
+class MediaCodecVideoDecoderFactory implements VideoDecoderFactory {
+  private static final String TAG = "MediaCodecVideoDecoderFactory";
+
+  private final @Nullable EglBase.Context sharedContext;
+  private final @Nullable Predicate<MediaCodecInfo> codecAllowedPredicate;
+
+  /**
+   * MediaCodecVideoDecoderFactory with support of codecs filtering.
+   *
+   * @param sharedContext The textures generated will be accessible from this context. May be null,
+   *                      this disables texture support.
+   * @param codecAllowedPredicate optional predicate to test if codec allowed. All codecs are
+   *                              allowed when predicate is not provided.
+   */
+  public MediaCodecVideoDecoderFactory(@Nullable EglBase.Context sharedContext,
+      @Nullable Predicate<MediaCodecInfo> codecAllowedPredicate) {
+    this.sharedContext = sharedContext;
+    this.codecAllowedPredicate = codecAllowedPredicate;
+  }
+
+  @Nullable
+  @Override
+  public VideoDecoder createDecoder(VideoCodecInfo codecType) {
+    VideoCodecMimeType type = VideoCodecMimeType.valueOf(codecType.getName());
+    MediaCodecInfo info = findCodecForType(type);
+
+    if (info == null) {
+      return null;
+    }
+
+    CodecCapabilities capabilities = info.getCapabilitiesForType(type.mimeType());
+    return new AndroidVideoDecoder(new MediaCodecWrapperFactoryImpl(), info.getName(), type,
+        MediaCodecUtils.selectColorFormat(MediaCodecUtils.DECODER_COLOR_FORMATS, capabilities),
+        sharedContext);
+  }
+
+  @Override
+  public VideoCodecInfo[] getSupportedCodecs() {
+    List<VideoCodecInfo> supportedCodecInfos = new ArrayList<VideoCodecInfo>();
+    // Generate a list of supported codecs in order of preference:
+    // VP8, VP9, H264 (high profile), and H264 (baseline profile).
+    for (VideoCodecMimeType type : new VideoCodecMimeType[] {VideoCodecMimeType.VP8,
+             VideoCodecMimeType.VP9, VideoCodecMimeType.H264, VideoCodecMimeType.AV1}) {
+      MediaCodecInfo codec = findCodecForType(type);
+      if (codec != null) {
+        String name = type.name();
+        if (type == VideoCodecMimeType.H264 && isH264HighProfileSupported(codec)) {
+          supportedCodecInfos.add(new VideoCodecInfo(
+              name, MediaCodecUtils.getCodecProperties(type, /* highProfile= */ true)));
+        }
+
+        supportedCodecInfos.add(new VideoCodecInfo(
+            name, MediaCodecUtils.getCodecProperties(type, /* highProfile= */ false)));
+      }
+    }
+
+    return supportedCodecInfos.toArray(new VideoCodecInfo[supportedCodecInfos.size()]);
+  }
+
+  private @Nullable MediaCodecInfo findCodecForType(VideoCodecMimeType type) {
+    for (int i = 0; i < MediaCodecList.getCodecCount(); ++i) {
+      MediaCodecInfo info = null;
+      try {
+        info = MediaCodecList.getCodecInfoAt(i);
+      } catch (IllegalArgumentException e) {
+        Logging.e(TAG, "Cannot retrieve decoder codec info", e);
+      }
+
+      if (info == null || info.isEncoder()) {
+        continue;
+      }
+
+      if (isSupportedCodec(info, type)) {
+        return info;
+      }
+    }
+
+    return null; // No support for this type.
+  }
+
+  // Returns true if the given MediaCodecInfo indicates a supported encoder for the given type.
+  private boolean isSupportedCodec(MediaCodecInfo info, VideoCodecMimeType type) {
+    if (!MediaCodecUtils.codecSupportsType(info, type)) {
+      return false;
+    }
+    // Check for a supported color format.
+    if (MediaCodecUtils.selectColorFormat(
+            MediaCodecUtils.DECODER_COLOR_FORMATS, info.getCapabilitiesForType(type.mimeType()))
+        == null) {
+      return false;
+    }
+    return isCodecAllowed(info);
+  }
+
+  private boolean isCodecAllowed(MediaCodecInfo info) {
+    if (codecAllowedPredicate == null) {
+      return true;
+    }
+    return codecAllowedPredicate.test(info);
+  }
+
+  private boolean isH264HighProfileSupported(MediaCodecInfo info) {
+    String name = info.getName();
+    // Support H.264 HP decoding on QCOM chips.
+    if (name.startsWith(QCOM_PREFIX)) {
+      return true;
+    }
+    // Support H.264 HP decoding on Exynos chips for Android M and above.
+    if (Build.VERSION.SDK_INT >= Build.VERSION_CODES.M && name.startsWith(EXYNOS_PREFIX)) {
+      return true;
+    }
+    return false;
+  }
+}
diff --git a/third_party/libwebrtc/sdk/android/src/java/org/webrtc/MediaCodecWrapper.java b/third_party/libwebrtc/sdk/android/src/java/org/webrtc/MediaCodecWrapper.java
new file mode 100644
index 0000000000..60c853df35
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/src/java/org/webrtc/MediaCodecWrapper.java
@@ -0,0 +1,55 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+import android.media.MediaCodec;
+import android.media.MediaCrypto;
+import android.media.MediaFormat;
+import android.os.Bundle;
+import android.view.Surface;
+import java.nio.ByteBuffer;
+
+/**
+ * Subset of methods defined in {@link android.media.MediaCodec} needed by
+ * {@link HardwareVideoEncoder} and {@link AndroidVideoDecoder}. This interface
+ * exists to allow mocking and using a fake implementation in tests.
+ */
+interface MediaCodecWrapper {
+  void configure(MediaFormat format, Surface surface, MediaCrypto crypto, int flags);
+
+  void start();
+
+  void flush();
+
+  void stop();
+
+  void release();
+
+  int dequeueInputBuffer(long timeoutUs);
+
+  void queueInputBuffer(int index, int offset, int size, long presentationTimeUs, int flags);
+
+  int dequeueOutputBuffer(MediaCodec.BufferInfo info, long timeoutUs);
+
+  void releaseOutputBuffer(int index, boolean render);
+
+  MediaFormat getInputFormat();
+
+  MediaFormat getOutputFormat();
+
+  ByteBuffer getInputBuffer(int index);
+
+  ByteBuffer getOutputBuffer(int index);
+
+  Surface createInputSurface();
+
+  void setParameters(Bundle params);
+}
diff --git a/third_party/libwebrtc/sdk/android/src/java/org/webrtc/MediaCodecWrapperFactory.java b/third_party/libwebrtc/sdk/android/src/java/org/webrtc/MediaCodecWrapperFactory.java
new file mode 100644
index 0000000000..2962cb62a7
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/src/java/org/webrtc/MediaCodecWrapperFactory.java
@@ -0,0 +1,22 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+import java.io.IOException;
+
+interface MediaCodecWrapperFactory {
+  /**
+   * Creates a new {@link MediaCodecWrapper} by codec name.
+   *
+   * <p>For additional information see {@link android.media.MediaCodec#createByCodecName}.
+   */
+  MediaCodecWrapper createByCodecName(String name) throws IOException;
+}
diff --git a/third_party/libwebrtc/sdk/android/src/java/org/webrtc/MediaCodecWrapperFactoryImpl.java b/third_party/libwebrtc/sdk/android/src/java/org/webrtc/MediaCodecWrapperFactoryImpl.java
new file mode 100644
index 0000000000..2ba62ac7d6
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/src/java/org/webrtc/MediaCodecWrapperFactoryImpl.java
@@ -0,0 +1,115 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+import android.media.MediaCodec;
+import android.media.MediaCodec.BufferInfo;
+import android.media.MediaCrypto;
+import android.media.MediaFormat;
+import android.os.Bundle;
+import android.view.Surface;
+import java.io.IOException;
+import java.nio.ByteBuffer;
+
+/**
+ * Implementation of MediaCodecWrapperFactory that returns MediaCodecInterfaces wrapping
+ * {@link android.media.MediaCodec} objects.
+ */
+class MediaCodecWrapperFactoryImpl implements MediaCodecWrapperFactory {
+  private static class MediaCodecWrapperImpl implements MediaCodecWrapper {
+    private final MediaCodec mediaCodec;
+
+    public MediaCodecWrapperImpl(MediaCodec mediaCodec) {
+      this.mediaCodec = mediaCodec;
+    }
+
+    @Override
+    public void configure(MediaFormat format, Surface surface, MediaCrypto crypto, int flags) {
+      mediaCodec.configure(format, surface, crypto, flags);
+    }
+
+    @Override
+    public void start() {
+      mediaCodec.start();
+    }
+
+    @Override
+    public void flush() {
+      mediaCodec.flush();
+    }
+
+    @Override
+    public void stop() {
+      mediaCodec.stop();
+    }
+
+    @Override
+    public void release() {
+      mediaCodec.release();
+    }
+
+    @Override
+    public int dequeueInputBuffer(long timeoutUs) {
+      return mediaCodec.dequeueInputBuffer(timeoutUs);
+    }
+
+    @Override
+    public void queueInputBuffer(
+        int index, int offset, int size, long presentationTimeUs, int flags) {
+      mediaCodec.queueInputBuffer(index, offset, size, presentationTimeUs, flags);
+    }
+
+    @Override
+    public int dequeueOutputBuffer(BufferInfo info, long timeoutUs) {
+      return mediaCodec.dequeueOutputBuffer(info, timeoutUs);
+    }
+
+    @Override
+    public void releaseOutputBuffer(int index, boolean render) {
+      mediaCodec.releaseOutputBuffer(index, render);
+    }
+
+    @Override
+    public MediaFormat getInputFormat() {
+      return mediaCodec.getInputFormat();
+    }
+
+    @Override
+    public MediaFormat getOutputFormat() {
+      return mediaCodec.getOutputFormat();
+    }
+
+    @Override
+    public ByteBuffer getInputBuffer(int index) {
+      return mediaCodec.getInputBuffer(index);
+    }
+
+    @Override
+    public ByteBuffer getOutputBuffer(int index) {
+      return mediaCodec.getOutputBuffer(index);
+    }
+
+    @Override
+    public Surface createInputSurface() {
+      return mediaCodec.createInputSurface();
+    }
+
+    @Override
+    public void setParameters(Bundle params) {
+      mediaCodec.setParameters(params);
+    }
+  }
+
+  @Override
+  public MediaCodecWrapper createByCodecName(String name) throws IOException {
+    return new MediaCodecWrapperImpl(MediaCodec.createByCodecName(name));
+  }
+}
diff --git a/third_party/libwebrtc/sdk/android/src/java/org/webrtc/NV12Buffer.java b/third_party/libwebrtc/sdk/android/src/java/org/webrtc/NV12Buffer.java
new file mode 100644
index 0000000000..fe0221d826
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/src/java/org/webrtc/NV12Buffer.java
@@ -0,0 +1,73 @@
+/*
+ * Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS.  All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+import androidx.annotation.Nullable;
+import java.nio.ByteBuffer;
+
+public class NV12Buffer implements VideoFrame.Buffer {
+  private final int width;
+  private final int height;
+  private final int stride;
+  private final int sliceHeight;
+  private final ByteBuffer buffer;
+  private final RefCountDelegate refCountDelegate;
+
+  public NV12Buffer(int width, int height, int stride, int sliceHeight, ByteBuffer buffer,
+      @Nullable Runnable releaseCallback) {
+    this.width = width;
+    this.height = height;
+    this.stride = stride;
+    this.sliceHeight = sliceHeight;
+    this.buffer = buffer;
+    this.refCountDelegate = new RefCountDelegate(releaseCallback);
+  }
+
+  @Override
+  public int getWidth() {
+    return width;
+  }
+
+  @Override
+  public int getHeight() {
+    return height;
+  }
+
+  @Override
+  public VideoFrame.I420Buffer toI420() {
+    return (VideoFrame.I420Buffer) cropAndScale(0, 0, width, height, width, height);
+  }
+
+  @Override
+  public void retain() {
+    refCountDelegate.retain();
+  }
+
+  @Override
+  public void release() {
+    refCountDelegate.release();
+  }
+
+  @Override
+  public VideoFrame.Buffer cropAndScale(
+      int cropX, int cropY, int cropWidth, int cropHeight, int scaleWidth, int scaleHeight) {
+    JavaI420Buffer newBuffer = JavaI420Buffer.allocate(scaleWidth, scaleHeight);
+    nativeCropAndScale(cropX, cropY, cropWidth, cropHeight, scaleWidth, scaleHeight, buffer, width,
+        height, stride, sliceHeight, newBuffer.getDataY(), newBuffer.getStrideY(),
+        newBuffer.getDataU(), newBuffer.getStrideU(), newBuffer.getDataV(), newBuffer.getStrideV());
+    return newBuffer;
+  }
+
+  private static native void nativeCropAndScale(int cropX, int cropY, int cropWidth, int cropHeight,
+      int scaleWidth, int scaleHeight, ByteBuffer src, int srcWidth, int srcHeight, int srcStride,
+      int srcSliceHeight, ByteBuffer dstY, int dstStrideY, ByteBuffer dstU, int dstStrideU,
+      ByteBuffer dstV, int dstStrideV);
+}
diff --git a/third_party/libwebrtc/sdk/android/src/java/org/webrtc/NV21Buffer.java b/third_party/libwebrtc/sdk/android/src/java/org/webrtc/NV21Buffer.java
new file mode 100644
index 0000000000..0fb1afe74b
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/src/java/org/webrtc/NV21Buffer.java
@@ -0,0 +1,69 @@
+/*
+ * Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS.  All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+import androidx.annotation.Nullable;
+import java.nio.ByteBuffer;
+
+public class NV21Buffer implements VideoFrame.Buffer {
+  private final byte[] data;
+  private final int width;
+  private final int height;
+  private final RefCountDelegate refCountDelegate;
+
+  public NV21Buffer(byte[] data, int width, int height, @Nullable Runnable releaseCallback) {
+    this.data = data;
+    this.width = width;
+    this.height = height;
+    this.refCountDelegate = new RefCountDelegate(releaseCallback);
+  }
+
+  @Override
+  public int getWidth() {
+    return width;
+  }
+
+  @Override
+  public int getHeight() {
+    return height;
+  }
+
+  @Override
+  public VideoFrame.I420Buffer toI420() {
+    // Cropping converts the frame to I420. Just crop and scale to the whole image.
+    return (VideoFrame.I420Buffer) cropAndScale(0 /* cropX */, 0 /* cropY */, width /* cropWidth */,
+        height /* cropHeight */, width /* scaleWidth */, height /* scaleHeight */);
+  }
+
+  @Override
+  public void retain() {
+    refCountDelegate.retain();
+  }
+
+  @Override
+  public void release() {
+    refCountDelegate.release();
+  }
+
+  @Override
+  public VideoFrame.Buffer cropAndScale(
+      int cropX, int cropY, int cropWidth, int cropHeight, int scaleWidth, int scaleHeight) {
+    JavaI420Buffer newBuffer = JavaI420Buffer.allocate(scaleWidth, scaleHeight);
+    nativeCropAndScale(cropX, cropY, cropWidth, cropHeight, scaleWidth, scaleHeight, data, width,
+        height, newBuffer.getDataY(), newBuffer.getStrideY(), newBuffer.getDataU(),
+        newBuffer.getStrideU(), newBuffer.getDataV(), newBuffer.getStrideV());
+    return newBuffer;
+  }
+
+  private static native void nativeCropAndScale(int cropX, int cropY, int cropWidth, int cropHeight,
+      int scaleWidth, int scaleHeight, byte[] src, int srcWidth, int srcHeight, ByteBuffer dstY,
+      int dstStrideY, ByteBuffer dstU, int dstStrideU, ByteBuffer dstV, int dstStrideV);
+}
diff --git a/third_party/libwebrtc/sdk/android/src/java/org/webrtc/NativeAndroidVideoTrackSource.java b/third_party/libwebrtc/sdk/android/src/java/org/webrtc/NativeAndroidVideoTrackSource.java
new file mode 100644
index 0000000000..d4fba481e8
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/src/java/org/webrtc/NativeAndroidVideoTrackSource.java
@@ -0,0 +1,99 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+import androidx.annotation.Nullable;
+import org.webrtc.VideoFrame;
+import org.webrtc.VideoProcessor;
+
+/**
+ * This class is meant to be a simple layer that only handles the JNI wrapping of a C++
+ * AndroidVideoTrackSource, that can easily be mocked out in Java unit tests. Refrain from adding
+ * any unnecessary logic to this class.
+ * This class is thred safe and methods can be called from any thread, but if frames A, B, ..., are
+ * sent to adaptFrame(), the adapted frames adaptedA, adaptedB, ..., needs to be passed in the same
+ * order to onFrameCaptured().
+ */
+class NativeAndroidVideoTrackSource {
+  // Pointer to webrtc::jni::AndroidVideoTrackSource.
+  private final long nativeAndroidVideoTrackSource;
+
+  public NativeAndroidVideoTrackSource(long nativeAndroidVideoTrackSource) {
+    this.nativeAndroidVideoTrackSource = nativeAndroidVideoTrackSource;
+  }
+
+  /**
+   * Set the state for the native MediaSourceInterface. Maps boolean to either
+   * SourceState::kLive or SourceState::kEnded.
+   */
+  public void setState(boolean isLive) {
+    nativeSetState(nativeAndroidVideoTrackSource, isLive);
+  }
+
+  /**
+   * This function should be called before delivering any frame to determine if the frame should be
+   * dropped or what the cropping and scaling parameters should be. If the return value is null, the
+   * frame should be dropped, otherwise the frame should be adapted in accordance to the frame
+   * adaptation parameters before calling onFrameCaptured().
+   */
+  @Nullable
+  public VideoProcessor.FrameAdaptationParameters adaptFrame(VideoFrame frame) {
+    return nativeAdaptFrame(nativeAndroidVideoTrackSource, frame.getBuffer().getWidth(),
+        frame.getBuffer().getHeight(), frame.getRotation(), frame.getTimestampNs());
+  }
+
+  /**
+   * Pass an adapted frame to the native AndroidVideoTrackSource. Note that adaptFrame() is
+   * expected to be called first and that the passed frame conforms to those parameters.
+   */
+  public void onFrameCaptured(VideoFrame frame) {
+    nativeOnFrameCaptured(nativeAndroidVideoTrackSource, frame.getRotation(),
+        frame.getTimestampNs(), frame.getBuffer());
+  }
+
+  /**
+   * Calling this function will cause frames to be scaled down to the requested resolution. Also,
+   * frames will be cropped to match the requested aspect ratio, and frames will be dropped to match
+   * the requested fps.
+   */
+  public void adaptOutputFormat(VideoSource.AspectRatio targetLandscapeAspectRatio,
+      @Nullable Integer maxLandscapePixelCount, VideoSource.AspectRatio targetPortraitAspectRatio,
+      @Nullable Integer maxPortraitPixelCount, @Nullable Integer maxFps) {
+    nativeAdaptOutputFormat(nativeAndroidVideoTrackSource, targetLandscapeAspectRatio.width,
+        targetLandscapeAspectRatio.height, maxLandscapePixelCount, targetPortraitAspectRatio.width,
+        targetPortraitAspectRatio.height, maxPortraitPixelCount, maxFps);
+  }
+
+  public void setIsScreencast(boolean isScreencast) {
+    nativeSetIsScreencast(nativeAndroidVideoTrackSource, isScreencast);
+  }
+
+  @CalledByNative
+  static VideoProcessor.FrameAdaptationParameters createFrameAdaptationParameters(int cropX,
+      int cropY, int cropWidth, int cropHeight, int scaleWidth, int scaleHeight, long timestampNs,
+      boolean drop) {
+    return new VideoProcessor.FrameAdaptationParameters(
+        cropX, cropY, cropWidth, cropHeight, scaleWidth, scaleHeight, timestampNs, drop);
+  }
+
+  private static native void nativeSetIsScreencast(
+      long nativeAndroidVideoTrackSource, boolean isScreencast);
+  private static native void nativeSetState(long nativeAndroidVideoTrackSource, boolean isLive);
+  private static native void nativeAdaptOutputFormat(long nativeAndroidVideoTrackSource,
+      int landscapeWidth, int landscapeHeight, @Nullable Integer maxLandscapePixelCount,
+      int portraitWidth, int portraitHeight, @Nullable Integer maxPortraitPixelCount,
+      @Nullable Integer maxFps);
+  @Nullable
+  private static native VideoProcessor.FrameAdaptationParameters nativeAdaptFrame(
+      long nativeAndroidVideoTrackSource, int width, int height, int rotation, long timestampNs);
+  private static native void nativeOnFrameCaptured(
+      long nativeAndroidVideoTrackSource, int rotation, long timestampNs, VideoFrame.Buffer buffer);
+}
diff --git a/third_party/libwebrtc/sdk/android/src/java/org/webrtc/NativeCapturerObserver.java b/third_party/libwebrtc/sdk/android/src/java/org/webrtc/NativeCapturerObserver.java
new file mode 100644
index 0000000000..c195fb3a4c
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/src/java/org/webrtc/NativeCapturerObserver.java
@@ -0,0 +1,53 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+import org.webrtc.VideoFrame;
+
+/**
+ * Used from native api and implements a simple VideoCapturer.CapturerObserver that feeds frames to
+ * a webrtc::jni::AndroidVideoTrackSource.
+ */
+class NativeCapturerObserver implements CapturerObserver {
+  private final NativeAndroidVideoTrackSource nativeAndroidVideoTrackSource;
+
+  @CalledByNative
+  public NativeCapturerObserver(long nativeSource) {
+    this.nativeAndroidVideoTrackSource = new NativeAndroidVideoTrackSource(nativeSource);
+  }
+
+  @Override
+  public void onCapturerStarted(boolean success) {
+    nativeAndroidVideoTrackSource.setState(success);
+  }
+
+  @Override
+  public void onCapturerStopped() {
+    nativeAndroidVideoTrackSource.setState(/* isLive= */ false);
+  }
+
+  @Override
+  public void onFrameCaptured(VideoFrame frame) {
+    final VideoProcessor.FrameAdaptationParameters parameters =
+        nativeAndroidVideoTrackSource.adaptFrame(frame);
+    if (parameters == null) {
+      // Drop frame.
+      return;
+    }
+
+    final VideoFrame.Buffer adaptedBuffer =
+        frame.getBuffer().cropAndScale(parameters.cropX, parameters.cropY, parameters.cropWidth,
+            parameters.cropHeight, parameters.scaleWidth, parameters.scaleHeight);
+    nativeAndroidVideoTrackSource.onFrameCaptured(
+        new VideoFrame(adaptedBuffer, frame.getRotation(), parameters.timestampNs));
+    adaptedBuffer.release();
+  }
+}
diff --git a/third_party/libwebrtc/sdk/android/src/java/org/webrtc/NativeLibrary.java b/third_party/libwebrtc/sdk/android/src/java/org/webrtc/NativeLibrary.java
new file mode 100644
index 0000000000..531c216302
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/src/java/org/webrtc/NativeLibrary.java
@@ -0,0 +1,51 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+class NativeLibrary {
+  private static String TAG = "NativeLibrary";
+
+  static class DefaultLoader implements NativeLibraryLoader {
+    @Override
+    public boolean load(String name) {
+      Logging.d(TAG, "Loading library: " + name);
+      System.loadLibrary(name);
+
+      // Not relevant, but kept for API compatibility.
+      return true;
+    }
+  }
+
+  private static Object lock = new Object();
+  private static boolean libraryLoaded;
+
+  /**
+   * Loads the native library. Clients should call PeerConnectionFactory.initialize. It will call
+   * this method for them.
+   */
+  static void initialize(NativeLibraryLoader loader, String libraryName) {
+    synchronized (lock) {
+      if (libraryLoaded) {
+        Logging.d(TAG, "Native library has already been loaded.");
+        return;
+      }
+      Logging.d(TAG, "Loading native library: " + libraryName);
+      libraryLoaded = loader.load(libraryName);
+    }
+  }
+
+  /** Returns true if the library has been loaded successfully. */
+  static boolean isLoaded() {
+    synchronized (lock) {
+      return libraryLoaded;
+    }
+  }
+}
diff --git a/third_party/libwebrtc/sdk/android/src/java/org/webrtc/RefCountDelegate.java b/third_party/libwebrtc/sdk/android/src/java/org/webrtc/RefCountDelegate.java
new file mode 100644
index 0000000000..b9210d26a4
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/src/java/org/webrtc/RefCountDelegate.java
@@ -0,0 +1,63 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+import androidx.annotation.Nullable;
+import java.util.concurrent.atomic.AtomicInteger;
+
+/**
+ * Implementation of RefCounted that executes a Runnable once the ref count reaches zero.
+ */
+class RefCountDelegate implements RefCounted {
+  private final AtomicInteger refCount = new AtomicInteger(1);
+  private final @Nullable Runnable releaseCallback;
+
+  /**
+   * @param releaseCallback Callback that will be executed once the ref count reaches zero.
+   */
+  public RefCountDelegate(@Nullable Runnable releaseCallback) {
+    this.releaseCallback = releaseCallback;
+  }
+
+  @Override
+  public void retain() {
+    int updated_count = refCount.incrementAndGet();
+    if (updated_count < 2) {
+      throw new IllegalStateException("retain() called on an object with refcount < 1");
+    }
+  }
+
+  @Override
+  public void release() {
+    int updated_count = refCount.decrementAndGet();
+    if (updated_count < 0) {
+      throw new IllegalStateException("release() called on an object with refcount < 1");
+    }
+    if (updated_count == 0 && releaseCallback != null) {
+      releaseCallback.run();
+    }
+  }
+
+  /**
+   * Tries to retain the object. Can be used in scenarios where it is unknown if the object has
+   * already been released. Returns true if successful or false if the object was already released.
+   */
+  boolean safeRetain() {
+    int currentRefCount = refCount.get();
+    while (currentRefCount != 0) {
+      if (refCount.weakCompareAndSet(currentRefCount, currentRefCount + 1)) {
+        return true;
+      }
+      currentRefCount = refCount.get();
+    }
+    return false;
+  }
+}
diff --git a/third_party/libwebrtc/sdk/android/src/java/org/webrtc/VideoCodecMimeType.java b/third_party/libwebrtc/sdk/android/src/java/org/webrtc/VideoCodecMimeType.java
new file mode 100644
index 0000000000..26a030919d
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/src/java/org/webrtc/VideoCodecMimeType.java
@@ -0,0 +1,29 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+/** Enumeration of supported video codec types. */
+enum VideoCodecMimeType {
+  VP8("video/x-vnd.on2.vp8"),
+  VP9("video/x-vnd.on2.vp9"),
+  H264("video/avc"),
+  AV1("video/av01");
+
+  private final String mimeType;
+
+  private VideoCodecMimeType(String mimeType) {
+    this.mimeType = mimeType;
+  }
+
+  String mimeType() {
+    return mimeType;
+  }
+}
diff --git a/third_party/libwebrtc/sdk/android/src/java/org/webrtc/VideoDecoderWrapper.java b/third_party/libwebrtc/sdk/android/src/java/org/webrtc/VideoDecoderWrapper.java
new file mode 100644
index 0000000000..2aae041640
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/src/java/org/webrtc/VideoDecoderWrapper.java
@@ -0,0 +1,27 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+import org.webrtc.VideoDecoder;
+
+/**
+ * This class contains the Java glue code for JNI generation of VideoDecoder.
+ */
+class VideoDecoderWrapper {
+  @CalledByNative
+  static VideoDecoder.Callback createDecoderCallback(final long nativeDecoder) {
+    return (VideoFrame frame, Integer decodeTimeMs,
+               Integer qp) -> nativeOnDecodedFrame(nativeDecoder, frame, decodeTimeMs, qp);
+  }
+
+  private static native void nativeOnDecodedFrame(
+      long nativeVideoDecoderWrapper, VideoFrame frame, Integer decodeTimeMs, Integer qp);
+}
diff --git a/third_party/libwebrtc/sdk/android/src/java/org/webrtc/VideoEncoderWrapper.java b/third_party/libwebrtc/sdk/android/src/java/org/webrtc/VideoEncoderWrapper.java
new file mode 100644
index 0000000000..b5485d4edb
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/src/java/org/webrtc/VideoEncoderWrapper.java
@@ -0,0 +1,46 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+// Explicit imports necessary for JNI generation.
+import androidx.annotation.Nullable;
+import org.webrtc.VideoEncoder;
+
+/**
+ * This class contains the Java glue code for JNI generation of VideoEncoder.
+ */
+class VideoEncoderWrapper {
+  @CalledByNative
+  static boolean getScalingSettingsOn(VideoEncoder.ScalingSettings scalingSettings) {
+    return scalingSettings.on;
+  }
+
+  @Nullable
+  @CalledByNative
+  static Integer getScalingSettingsLow(VideoEncoder.ScalingSettings scalingSettings) {
+    return scalingSettings.low;
+  }
+
+  @Nullable
+  @CalledByNative
+  static Integer getScalingSettingsHigh(VideoEncoder.ScalingSettings scalingSettings) {
+    return scalingSettings.high;
+  }
+
+  @CalledByNative
+  static VideoEncoder.Callback createEncoderCallback(final long nativeEncoder) {
+    return (EncodedImage frame,
+               VideoEncoder.CodecSpecificInfo info) -> nativeOnEncodedFrame(nativeEncoder, frame);
+  }
+
+  private static native void nativeOnEncodedFrame(
+      long nativeVideoEncoderWrapper, EncodedImage frame);
+}
diff --git a/third_party/libwebrtc/sdk/android/src/java/org/webrtc/WebRtcClassLoader.java b/third_party/libwebrtc/sdk/android/src/java/org/webrtc/WebRtcClassLoader.java
new file mode 100644
index 0000000000..023e92cfb1
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/src/java/org/webrtc/WebRtcClassLoader.java
@@ -0,0 +1,27 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+/**
+ * This class provides a ClassLoader that is capable of loading WebRTC Java classes regardless of
+ * what thread it's called from. Such a ClassLoader is needed for the few cases where the JNI
+ * mechanism is unable to automatically determine the appropriate ClassLoader instance.
+ */
+class WebRtcClassLoader {
+  @CalledByNative
+  static Object getClassLoader() {
+    Object loader = WebRtcClassLoader.class.getClassLoader();
+    if (loader == null) {
+      throw new RuntimeException("Failed to get WebRTC class loader.");
+    }
+    return loader;
+  }
+}
diff --git a/third_party/libwebrtc/sdk/android/src/java/org/webrtc/WrappedNativeI420Buffer.java b/third_party/libwebrtc/sdk/android/src/java/org/webrtc/WrappedNativeI420Buffer.java
new file mode 100644
index 0000000000..0461660fcf
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/src/java/org/webrtc/WrappedNativeI420Buffer.java
@@ -0,0 +1,110 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc;
+
+import java.nio.ByteBuffer;
+
+/**
+ * This class wraps a webrtc::I420BufferInterface into a VideoFrame.I420Buffer.
+ */
+class WrappedNativeI420Buffer implements VideoFrame.I420Buffer {
+  private final int width;
+  private final int height;
+  private final ByteBuffer dataY;
+  private final int strideY;
+  private final ByteBuffer dataU;
+  private final int strideU;
+  private final ByteBuffer dataV;
+  private final int strideV;
+  private final long nativeBuffer;
+
+  @CalledByNative
+  WrappedNativeI420Buffer(int width, int height, ByteBuffer dataY, int strideY, ByteBuffer dataU,
+      int strideU, ByteBuffer dataV, int strideV, long nativeBuffer) {
+    this.width = width;
+    this.height = height;
+    this.dataY = dataY;
+    this.strideY = strideY;
+    this.dataU = dataU;
+    this.strideU = strideU;
+    this.dataV = dataV;
+    this.strideV = strideV;
+    this.nativeBuffer = nativeBuffer;
+
+    retain();
+  }
+
+  @Override
+  public int getWidth() {
+    return width;
+  }
+
+  @Override
+  public int getHeight() {
+    return height;
+  }
+
+  @Override
+  public ByteBuffer getDataY() {
+    // Return a slice to prevent relative reads from changing the position.
+    return dataY.slice();
+  }
+
+  @Override
+  public ByteBuffer getDataU() {
+    // Return a slice to prevent relative reads from changing the position.
+    return dataU.slice();
+  }
+
+  @Override
+  public ByteBuffer getDataV() {
+    // Return a slice to prevent relative reads from changing the position.
+    return dataV.slice();
+  }
+
+  @Override
+  public int getStrideY() {
+    return strideY;
+  }
+
+  @Override
+  public int getStrideU() {
+    return strideU;
+  }
+
+  @Override
+  public int getStrideV() {
+    return strideV;
+  }
+
+  @Override
+  public VideoFrame.I420Buffer toI420() {
+    retain();
+    return this;
+  }
+
+  @Override
+  public void retain() {
+    JniCommon.nativeAddRef(nativeBuffer);
+  }
+
+  @Override
+  public void release() {
+    JniCommon.nativeReleaseRef(nativeBuffer);
+  }
+
+  @Override
+  public VideoFrame.Buffer cropAndScale(
+      int cropX, int cropY, int cropWidth, int cropHeight, int scaleWidth, int scaleHeight) {
+    return JavaI420Buffer.cropAndScaleI420(
+        this, cropX, cropY, cropWidth, cropHeight, scaleWidth, scaleHeight);
+  }
+}
diff --git a/third_party/libwebrtc/sdk/android/src/java/org/webrtc/audio/LowLatencyAudioBufferManager.java b/third_party/libwebrtc/sdk/android/src/java/org/webrtc/audio/LowLatencyAudioBufferManager.java
new file mode 100644
index 0000000000..70c625ab4f
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/src/java/org/webrtc/audio/LowLatencyAudioBufferManager.java
@@ -0,0 +1,81 @@
+/*
+ *  Copyright (c) 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc.audio;
+
+import android.media.AudioTrack;
+import android.os.Build;
+import org.webrtc.Logging;
+
+// Lowers the buffer size if no underruns are detected for 100 ms. Once an
+// underrun is detected, the buffer size is increased by 10 ms and it will not
+// be lowered further. The buffer size will never be increased more than
+// 5 times, to avoid the possibility of the buffer size increasing without
+// bounds.
+class LowLatencyAudioBufferManager {
+  private static final String TAG = "LowLatencyAudioBufferManager";
+  // The underrun count that was valid during the previous call to maybeAdjustBufferSize(). Used to
+  // detect increases in the value.
+  private int prevUnderrunCount;
+  // The number of ticks to wait without an underrun before decreasing the buffer size.
+  private int ticksUntilNextDecrease;
+  // Indicate if we should continue to decrease the buffer size.
+  private boolean keepLoweringBufferSize;
+  // How often the buffer size was increased.
+  private int bufferIncreaseCounter;
+
+  public LowLatencyAudioBufferManager() {
+    this.prevUnderrunCount = 0;
+    this.ticksUntilNextDecrease = 10;
+    this.keepLoweringBufferSize = true;
+    this.bufferIncreaseCounter = 0;
+  }
+
+  public void maybeAdjustBufferSize(AudioTrack audioTrack) {
+    if (Build.VERSION.SDK_INT >= 26) {
+      final int underrunCount = audioTrack.getUnderrunCount();
+      if (underrunCount > prevUnderrunCount) {
+        // Don't increase buffer more than 5 times. Continuing to increase the buffer size
+        // could be harmful on low-power devices that regularly experience underruns under
+        // normal conditions.
+        if (bufferIncreaseCounter < 5) {
+          // Underrun detected, increase buffer size by 10ms.
+          final int currentBufferSize = audioTrack.getBufferSizeInFrames();
+          final int newBufferSize = currentBufferSize + audioTrack.getPlaybackRate() / 100;
+          Logging.d(TAG,
+              "Underrun detected! Increasing AudioTrack buffer size from " + currentBufferSize
+                  + " to " + newBufferSize);
+          audioTrack.setBufferSizeInFrames(newBufferSize);
+          bufferIncreaseCounter++;
+        }
+        // Stop trying to lower the buffer size.
+        keepLoweringBufferSize = false;
+        prevUnderrunCount = underrunCount;
+        ticksUntilNextDecrease = 10;
+      } else if (keepLoweringBufferSize) {
+        ticksUntilNextDecrease--;
+        if (ticksUntilNextDecrease <= 0) {
+          // No underrun seen for 100 ms, try to lower the buffer size by 10ms.
+          final int bufferSize10ms = audioTrack.getPlaybackRate() / 100;
+          // Never go below a buffer size of 10ms.
+          final int currentBufferSize = audioTrack.getBufferSizeInFrames();
+          final int newBufferSize = Math.max(bufferSize10ms, currentBufferSize - bufferSize10ms);
+          if (newBufferSize != currentBufferSize) {
+            Logging.d(TAG,
+                "Lowering AudioTrack buffer size from " + currentBufferSize + " to "
+                    + newBufferSize);
+            audioTrack.setBufferSizeInFrames(newBufferSize);
+          }
+          ticksUntilNextDecrease = 10;
+        }
+      }
+    }
+  }
+}
diff --git a/third_party/libwebrtc/sdk/android/src/java/org/webrtc/audio/VolumeLogger.java b/third_party/libwebrtc/sdk/android/src/java/org/webrtc/audio/VolumeLogger.java
new file mode 100644
index 0000000000..06d5cd3a8e
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/src/java/org/webrtc/audio/VolumeLogger.java
@@ -0,0 +1,83 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc.audio;
+
+import android.media.AudioManager;
+import androidx.annotation.Nullable;
+import java.util.Timer;
+import java.util.TimerTask;
+import org.webrtc.Logging;
+
+// TODO(magjed): Do we really need to spawn a new thread just to log volume? Can we re-use the
+// AudioTrackThread instead?
+/**
+ * Private utility class that periodically checks and logs the volume level of the audio stream that
+ * is currently controlled by the volume control. A timer triggers logs once every 30 seconds and
+ * the timer's associated thread is named "WebRtcVolumeLevelLoggerThread".
+ */
+class VolumeLogger {
+  private static final String TAG = "VolumeLogger";
+  private static final String THREAD_NAME = "WebRtcVolumeLevelLoggerThread";
+  private static final int TIMER_PERIOD_IN_SECONDS = 30;
+
+  private final AudioManager audioManager;
+  private @Nullable Timer timer;
+
+  public VolumeLogger(AudioManager audioManager) {
+    this.audioManager = audioManager;
+  }
+
+  public void start() {
+    Logging.d(TAG, "start" + WebRtcAudioUtils.getThreadInfo());
+    if (timer != null) {
+      return;
+    }
+    Logging.d(TAG, "audio mode is: " + WebRtcAudioUtils.modeToString(audioManager.getMode()));
+
+    timer = new Timer(THREAD_NAME);
+    timer.schedule(new LogVolumeTask(audioManager.getStreamMaxVolume(AudioManager.STREAM_RING),
+                       audioManager.getStreamMaxVolume(AudioManager.STREAM_VOICE_CALL)),
+        0, TIMER_PERIOD_IN_SECONDS * 1000);
+  }
+
+  private class LogVolumeTask extends TimerTask {
+    private final int maxRingVolume;
+    private final int maxVoiceCallVolume;
+
+    LogVolumeTask(int maxRingVolume, int maxVoiceCallVolume) {
+      this.maxRingVolume = maxRingVolume;
+      this.maxVoiceCallVolume = maxVoiceCallVolume;
+    }
+
+    @Override
+    public void run() {
+      final int mode = audioManager.getMode();
+      if (mode == AudioManager.MODE_RINGTONE) {
+        Logging.d(TAG,
+            "STREAM_RING stream volume: " + audioManager.getStreamVolume(AudioManager.STREAM_RING)
+                + " (max=" + maxRingVolume + ")");
+      } else if (mode == AudioManager.MODE_IN_COMMUNICATION) {
+        Logging.d(TAG,
+            "VOICE_CALL stream volume: "
+                + audioManager.getStreamVolume(AudioManager.STREAM_VOICE_CALL)
+                + " (max=" + maxVoiceCallVolume + ")");
+      }
+    }
+  }
+
+  public void stop() {
+    Logging.d(TAG, "stop" + WebRtcAudioUtils.getThreadInfo());
+    if (timer != null) {
+      timer.cancel();
+      timer = null;
+    }
+  }
+}
diff --git a/third_party/libwebrtc/sdk/android/src/java/org/webrtc/audio/WebRtcAudioEffects.java b/third_party/libwebrtc/sdk/android/src/java/org/webrtc/audio/WebRtcAudioEffects.java
new file mode 100644
index 0000000000..a9ff1011b6
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/src/java/org/webrtc/audio/WebRtcAudioEffects.java
@@ -0,0 +1,227 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc.audio;
+
+import android.media.audiofx.AcousticEchoCanceler;
+import android.media.audiofx.AudioEffect;
+import android.media.audiofx.AudioEffect.Descriptor;
+import android.media.audiofx.NoiseSuppressor;
+import android.os.Build;
+import androidx.annotation.Nullable;
+import java.util.UUID;
+import org.webrtc.Logging;
+
+// This class wraps control of three different platform effects. Supported
+// effects are: AcousticEchoCanceler (AEC) and NoiseSuppressor (NS).
+// Calling enable() will active all effects that are
+// supported by the device if the corresponding `shouldEnableXXX` member is set.
+class WebRtcAudioEffects {
+  private static final boolean DEBUG = false;
+
+  private static final String TAG = "WebRtcAudioEffectsExternal";
+
+  // UUIDs for Software Audio Effects that we want to avoid using.
+  // The implementor field will be set to "The Android Open Source Project".
+  private static final UUID AOSP_ACOUSTIC_ECHO_CANCELER =
+      UUID.fromString("bb392ec0-8d4d-11e0-a896-0002a5d5c51b");
+  private static final UUID AOSP_NOISE_SUPPRESSOR =
+      UUID.fromString("c06c8400-8e06-11e0-9cb6-0002a5d5c51b");
+
+  // Contains the available effect descriptors returned from the
+  // AudioEffect.getEffects() call. This result is cached to avoid doing the
+  // slow OS call multiple times.
+  private static @Nullable Descriptor[] cachedEffects;
+
+  // Contains the audio effect objects. Created in enable() and destroyed
+  // in release().
+  private @Nullable AcousticEchoCanceler aec;
+  private @Nullable NoiseSuppressor ns;
+
+  // Affects the final state given to the setEnabled() method on each effect.
+  // The default state is set to "disabled" but each effect can also be enabled
+  // by calling setAEC() and setNS().
+  private boolean shouldEnableAec;
+  private boolean shouldEnableNs;
+
+  // Returns true if all conditions for supporting HW Acoustic Echo Cancellation (AEC) are
+  // fulfilled.
+  public static boolean isAcousticEchoCancelerSupported() {
+    return isEffectTypeAvailable(AudioEffect.EFFECT_TYPE_AEC, AOSP_ACOUSTIC_ECHO_CANCELER);
+  }
+
+  // Returns true if all conditions for supporting HW Noise Suppression (NS) are fulfilled.
+  public static boolean isNoiseSuppressorSupported() {
+    return isEffectTypeAvailable(AudioEffect.EFFECT_TYPE_NS, AOSP_NOISE_SUPPRESSOR);
+  }
+
+  public WebRtcAudioEffects() {
+    Logging.d(TAG, "ctor" + WebRtcAudioUtils.getThreadInfo());
+  }
+
+  // Call this method to enable or disable the platform AEC. It modifies
+  // `shouldEnableAec` which is used in enable() where the actual state
+  // of the AEC effect is modified. Returns true if HW AEC is supported and
+  // false otherwise.
+  public boolean setAEC(boolean enable) {
+    Logging.d(TAG, "setAEC(" + enable + ")");
+    if (!isAcousticEchoCancelerSupported()) {
+      Logging.w(TAG, "Platform AEC is not supported");
+      shouldEnableAec = false;
+      return false;
+    }
+    if (aec != null && (enable != shouldEnableAec)) {
+      Logging.e(TAG, "Platform AEC state can't be modified while recording");
+      return false;
+    }
+    shouldEnableAec = enable;
+    return true;
+  }
+
+  // Call this method to enable or disable the platform NS. It modifies
+  // `shouldEnableNs` which is used in enable() where the actual state
+  // of the NS effect is modified. Returns true if HW NS is supported and
+  // false otherwise.
+  public boolean setNS(boolean enable) {
+    Logging.d(TAG, "setNS(" + enable + ")");
+    if (!isNoiseSuppressorSupported()) {
+      Logging.w(TAG, "Platform NS is not supported");
+      shouldEnableNs = false;
+      return false;
+    }
+    if (ns != null && (enable != shouldEnableNs)) {
+      Logging.e(TAG, "Platform NS state can't be modified while recording");
+      return false;
+    }
+    shouldEnableNs = enable;
+    return true;
+  }
+
+  public void enable(int audioSession) {
+    Logging.d(TAG, "enable(audioSession=" + audioSession + ")");
+    assertTrue(aec == null);
+    assertTrue(ns == null);
+
+    if (DEBUG) {
+      // Add logging of supported effects but filter out "VoIP effects", i.e.,
+      // AEC, AEC and NS. Avoid calling AudioEffect.queryEffects() unless the
+      // DEBUG flag is set since we have seen crashes in this API.
+      for (Descriptor d : AudioEffect.queryEffects()) {
+        if (effectTypeIsVoIP(d.type)) {
+          Logging.d(TAG,
+              "name: " + d.name + ", "
+                  + "mode: " + d.connectMode + ", "
+                  + "implementor: " + d.implementor + ", "
+                  + "UUID: " + d.uuid);
+        }
+      }
+    }
+
+    if (isAcousticEchoCancelerSupported()) {
+      // Create an AcousticEchoCanceler and attach it to the AudioRecord on
+      // the specified audio session.
+      aec = AcousticEchoCanceler.create(audioSession);
+      if (aec != null) {
+        boolean enabled = aec.getEnabled();
+        boolean enable = shouldEnableAec && isAcousticEchoCancelerSupported();
+        if (aec.setEnabled(enable) != AudioEffect.SUCCESS) {
+          Logging.e(TAG, "Failed to set the AcousticEchoCanceler state");
+        }
+        Logging.d(TAG,
+            "AcousticEchoCanceler: was " + (enabled ? "enabled" : "disabled") + ", enable: "
+                + enable + ", is now: " + (aec.getEnabled() ? "enabled" : "disabled"));
+      } else {
+        Logging.e(TAG, "Failed to create the AcousticEchoCanceler instance");
+      }
+    }
+
+    if (isNoiseSuppressorSupported()) {
+      // Create an NoiseSuppressor and attach it to the AudioRecord on the
+      // specified audio session.
+      ns = NoiseSuppressor.create(audioSession);
+      if (ns != null) {
+        boolean enabled = ns.getEnabled();
+        boolean enable = shouldEnableNs && isNoiseSuppressorSupported();
+        if (ns.setEnabled(enable) != AudioEffect.SUCCESS) {
+          Logging.e(TAG, "Failed to set the NoiseSuppressor state");
+        }
+        Logging.d(TAG,
+            "NoiseSuppressor: was " + (enabled ? "enabled" : "disabled") + ", enable: " + enable
+                + ", is now: " + (ns.getEnabled() ? "enabled" : "disabled"));
+      } else {
+        Logging.e(TAG, "Failed to create the NoiseSuppressor instance");
+      }
+    }
+  }
+
+  // Releases all native audio effect resources. It is a good practice to
+  // release the effect engine when not in use as control can be returned
+  // to other applications or the native resources released.
+  public void release() {
+    Logging.d(TAG, "release");
+    if (aec != null) {
+      aec.release();
+      aec = null;
+    }
+    if (ns != null) {
+      ns.release();
+      ns = null;
+    }
+  }
+
+  // Returns true for effect types in `type` that are of "VoIP" types:
+  // Acoustic Echo Canceler (AEC) or Automatic Gain Control (AGC) or
+  // Noise Suppressor (NS). Note that, an extra check for support is needed
+  // in each comparison since some devices includes effects in the
+  // AudioEffect.Descriptor array that are actually not available on the device.
+  // As an example: Samsung Galaxy S6 includes an AGC in the descriptor but
+  // AutomaticGainControl.isAvailable() returns false.
+  private boolean effectTypeIsVoIP(UUID type) {
+    return (AudioEffect.EFFECT_TYPE_AEC.equals(type) && isAcousticEchoCancelerSupported())
+        || (AudioEffect.EFFECT_TYPE_NS.equals(type) && isNoiseSuppressorSupported());
+  }
+
+  // Helper method which throws an exception when an assertion has failed.
+  private static void assertTrue(boolean condition) {
+    if (!condition) {
+      throw new AssertionError("Expected condition to be true");
+    }
+  }
+
+  // Returns the cached copy of the audio effects array, if available, or
+  // queries the operating system for the list of effects.
+  private static @Nullable Descriptor[] getAvailableEffects() {
+    if (cachedEffects != null) {
+      return cachedEffects;
+    }
+    // The caching is best effort only - if this method is called from several
+    // threads in parallel, they may end up doing the underlying OS call
+    // multiple times. It's normally only called on one thread so there's no
+    // real need to optimize for the multiple threads case.
+    cachedEffects = AudioEffect.queryEffects();
+    return cachedEffects;
+  }
+
+  // Returns true if an effect of the specified type is available. Functionally
+  // equivalent to (NoiseSuppressor`AutomaticGainControl`...).isAvailable(), but
+  // faster as it avoids the expensive OS call to enumerate effects.
+  private static boolean isEffectTypeAvailable(UUID effectType, UUID blockListedUuid) {
+    Descriptor[] effects = getAvailableEffects();
+    if (effects == null) {
+      return false;
+    }
+    for (Descriptor d : effects) {
+      if (d.type.equals(effectType)) {
+        return !d.uuid.equals(blockListedUuid);
+      }
+    }
+    return false;
+  }
+}
diff --git a/third_party/libwebrtc/sdk/android/src/java/org/webrtc/audio/WebRtcAudioManager.java b/third_party/libwebrtc/sdk/android/src/java/org/webrtc/audio/WebRtcAudioManager.java
new file mode 100644
index 0000000000..f398602a28
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/src/java/org/webrtc/audio/WebRtcAudioManager.java
@@ -0,0 +1,122 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc.audio;
+
+import android.content.Context;
+import android.content.pm.PackageManager;
+import android.media.AudioFormat;
+import android.media.AudioManager;
+import android.media.AudioRecord;
+import android.media.AudioTrack;
+import android.os.Build;
+import org.webrtc.Logging;
+import org.webrtc.CalledByNative;
+
+/**
+ * This class contains static functions to query sample rate and input/output audio buffer sizes.
+ */
+class WebRtcAudioManager {
+  private static final String TAG = "WebRtcAudioManagerExternal";
+
+  private static final int DEFAULT_SAMPLE_RATE_HZ = 16000;
+
+  // Default audio data format is PCM 16 bit per sample.
+  // Guaranteed to be supported by all devices.
+  private static final int BITS_PER_SAMPLE = 16;
+
+  private static final int DEFAULT_FRAME_PER_BUFFER = 256;
+
+  @CalledByNative
+  static AudioManager getAudioManager(Context context) {
+    return (AudioManager) context.getSystemService(Context.AUDIO_SERVICE);
+  }
+
+  @CalledByNative
+  static int getOutputBufferSize(
+      Context context, AudioManager audioManager, int sampleRate, int numberOfOutputChannels) {
+    return isLowLatencyOutputSupported(context)
+        ? getLowLatencyFramesPerBuffer(audioManager)
+        : getMinOutputFrameSize(sampleRate, numberOfOutputChannels);
+  }
+
+  @CalledByNative
+  static int getInputBufferSize(
+      Context context, AudioManager audioManager, int sampleRate, int numberOfInputChannels) {
+    return isLowLatencyInputSupported(context)
+        ? getLowLatencyFramesPerBuffer(audioManager)
+        : getMinInputFrameSize(sampleRate, numberOfInputChannels);
+  }
+
+  private static boolean isLowLatencyOutputSupported(Context context) {
+    return context.getPackageManager().hasSystemFeature(PackageManager.FEATURE_AUDIO_LOW_LATENCY);
+  }
+
+  private static boolean isLowLatencyInputSupported(Context context) {
+    // TODO(henrika): investigate if some sort of device list is needed here
+    // as well. The NDK doc states that: "As of API level 21, lower latency
+    // audio input is supported on select devices. To take advantage of this
+    // feature, first confirm that lower latency output is available".
+    return isLowLatencyOutputSupported(context);
+  }
+
+  /**
+   * Returns the native input/output sample rate for this device's output stream.
+   */
+  @CalledByNative
+  static int getSampleRate(AudioManager audioManager) {
+    // Override this if we're running on an old emulator image which only
+    // supports 8 kHz and doesn't support PROPERTY_OUTPUT_SAMPLE_RATE.
+    if (WebRtcAudioUtils.runningOnEmulator()) {
+      Logging.d(TAG, "Running emulator, overriding sample rate to 8 kHz.");
+      return 8000;
+    }
+    // Deliver best possible estimate based on default Android AudioManager APIs.
+    final int sampleRateHz = getSampleRateForApiLevel(audioManager);
+    Logging.d(TAG, "Sample rate is set to " + sampleRateHz + " Hz");
+    return sampleRateHz;
+  }
+
+  private static int getSampleRateForApiLevel(AudioManager audioManager) {
+    String sampleRateString = audioManager.getProperty(AudioManager.PROPERTY_OUTPUT_SAMPLE_RATE);
+    return (sampleRateString == null) ? DEFAULT_SAMPLE_RATE_HZ : Integer.parseInt(sampleRateString);
+  }
+
+  // Returns the native output buffer size for low-latency output streams.
+  private static int getLowLatencyFramesPerBuffer(AudioManager audioManager) {
+    String framesPerBuffer =
+        audioManager.getProperty(AudioManager.PROPERTY_OUTPUT_FRAMES_PER_BUFFER);
+    return framesPerBuffer == null ? DEFAULT_FRAME_PER_BUFFER : Integer.parseInt(framesPerBuffer);
+  }
+
+  // Returns the minimum output buffer size for Java based audio (AudioTrack).
+  // This size can also be used for OpenSL ES implementations on devices that
+  // lacks support of low-latency output.
+  private static int getMinOutputFrameSize(int sampleRateInHz, int numChannels) {
+    final int bytesPerFrame = numChannels * (BITS_PER_SAMPLE / 8);
+    final int channelConfig =
+        (numChannels == 1 ? AudioFormat.CHANNEL_OUT_MONO : AudioFormat.CHANNEL_OUT_STEREO);
+    return AudioTrack.getMinBufferSize(
+               sampleRateInHz, channelConfig, AudioFormat.ENCODING_PCM_16BIT)
+        / bytesPerFrame;
+  }
+
+  // Returns the minimum input buffer size for Java based audio (AudioRecord).
+  // This size can calso be used for OpenSL ES implementations on devices that
+  // lacks support of low-latency input.
+  private static int getMinInputFrameSize(int sampleRateInHz, int numChannels) {
+    final int bytesPerFrame = numChannels * (BITS_PER_SAMPLE / 8);
+    final int channelConfig =
+        (numChannels == 1 ? AudioFormat.CHANNEL_IN_MONO : AudioFormat.CHANNEL_IN_STEREO);
+    return AudioRecord.getMinBufferSize(
+               sampleRateInHz, channelConfig, AudioFormat.ENCODING_PCM_16BIT)
+        / bytesPerFrame;
+  }
+}
diff --git a/third_party/libwebrtc/sdk/android/src/java/org/webrtc/audio/WebRtcAudioRecord.java b/third_party/libwebrtc/sdk/android/src/java/org/webrtc/audio/WebRtcAudioRecord.java
new file mode 100644
index 0000000000..6647e5fcbb
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/src/java/org/webrtc/audio/WebRtcAudioRecord.java
@@ -0,0 +1,743 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc.audio;
+
+import android.annotation.TargetApi;
+import android.content.Context;
+import android.media.AudioDeviceInfo;
+import android.media.AudioFormat;
+import android.media.AudioManager;
+import android.media.AudioRecord;
+import android.media.AudioRecordingConfiguration;
+import android.media.AudioTimestamp;
+import android.media.MediaRecorder.AudioSource;
+import android.os.Build;
+import android.os.Process;
+import androidx.annotation.Nullable;
+import androidx.annotation.RequiresApi;
+import java.lang.System;
+import java.nio.ByteBuffer;
+import java.util.Arrays;
+import java.util.Iterator;
+import java.util.List;
+import java.util.concurrent.Callable;
+import java.util.concurrent.Executors;
+import java.util.concurrent.ScheduledExecutorService;
+import java.util.concurrent.ScheduledFuture;
+import java.util.concurrent.ThreadFactory;
+import java.util.concurrent.TimeUnit;
+import java.util.concurrent.atomic.AtomicInteger;
+import java.util.concurrent.atomic.AtomicReference;
+import org.webrtc.CalledByNative;
+import org.webrtc.Logging;
+import org.webrtc.ThreadUtils;
+import org.webrtc.audio.JavaAudioDeviceModule.AudioRecordErrorCallback;
+import org.webrtc.audio.JavaAudioDeviceModule.AudioRecordStartErrorCode;
+import org.webrtc.audio.JavaAudioDeviceModule.AudioRecordStateCallback;
+import org.webrtc.audio.JavaAudioDeviceModule.SamplesReadyCallback;
+
+class WebRtcAudioRecord {
+  private static final String TAG = "WebRtcAudioRecordExternal";
+
+  // Requested size of each recorded buffer provided to the client.
+  private static final int CALLBACK_BUFFER_SIZE_MS = 10;
+
+  // Average number of callbacks per second.
+  private static final int BUFFERS_PER_SECOND = 1000 / CALLBACK_BUFFER_SIZE_MS;
+
+  // We ask for a native buffer size of BUFFER_SIZE_FACTOR * (minimum required
+  // buffer size). The extra space is allocated to guard against glitches under
+  // high load.
+  private static final int BUFFER_SIZE_FACTOR = 2;
+
+  // The AudioRecordJavaThread is allowed to wait for successful call to join()
+  // but the wait times out afther this amount of time.
+  private static final long AUDIO_RECORD_THREAD_JOIN_TIMEOUT_MS = 2000;
+
+  public static final int DEFAULT_AUDIO_SOURCE = AudioSource.VOICE_COMMUNICATION;
+
+  // Default audio data format is PCM 16 bit per sample.
+  // Guaranteed to be supported by all devices.
+  public static final int DEFAULT_AUDIO_FORMAT = AudioFormat.ENCODING_PCM_16BIT;
+
+  // Indicates AudioRecord has started recording audio.
+  private static final int AUDIO_RECORD_START = 0;
+
+  // Indicates AudioRecord has stopped recording audio.
+  private static final int AUDIO_RECORD_STOP = 1;
+
+  // Time to wait before checking recording status after start has been called. Tests have
+  // shown that the result can sometimes be invalid (our own status might be missing) if we check
+  // directly after start.
+  private static final int CHECK_REC_STATUS_DELAY_MS = 100;
+
+  private final Context context;
+  private final AudioManager audioManager;
+  private final int audioSource;
+  private final int audioFormat;
+
+  private long nativeAudioRecord;
+
+  private final WebRtcAudioEffects effects = new WebRtcAudioEffects();
+
+  private @Nullable ByteBuffer byteBuffer;
+
+  private @Nullable AudioRecord audioRecord;
+  private @Nullable AudioRecordThread audioThread;
+  private @Nullable AudioDeviceInfo preferredDevice;
+
+  private final ScheduledExecutorService executor;
+  private @Nullable ScheduledFuture<String> future;
+
+  private volatile boolean microphoneMute;
+  private final AtomicReference<Boolean> audioSourceMatchesRecordingSessionRef =
+      new AtomicReference<>();
+  private byte[] emptyBytes;
+
+  private final @Nullable AudioRecordErrorCallback errorCallback;
+  private final @Nullable AudioRecordStateCallback stateCallback;
+  private final @Nullable SamplesReadyCallback audioSamplesReadyCallback;
+  private final boolean isAcousticEchoCancelerSupported;
+  private final boolean isNoiseSuppressorSupported;
+
+  /**
+   * Audio thread which keeps calling ByteBuffer.read() waiting for audio
+   * to be recorded. Feeds recorded data to the native counterpart as a
+   * periodic sequence of callbacks using DataIsRecorded().
+   * This thread uses a Process.THREAD_PRIORITY_URGENT_AUDIO priority.
+   */
+  private class AudioRecordThread extends Thread {
+    private volatile boolean keepAlive = true;
+
+    public AudioRecordThread(String name) {
+      super(name);
+    }
+
+    @Override
+    public void run() {
+      Process.setThreadPriority(Process.THREAD_PRIORITY_URGENT_AUDIO);
+      Logging.d(TAG, "AudioRecordThread" + WebRtcAudioUtils.getThreadInfo());
+      assertTrue(audioRecord.getRecordingState() == AudioRecord.RECORDSTATE_RECORDING);
+
+      // Audio recording has started and the client is informed about it.
+      doAudioRecordStateCallback(AUDIO_RECORD_START);
+
+      long lastTime = System.nanoTime();
+      AudioTimestamp audioTimestamp = null;
+      if (Build.VERSION.SDK_INT >= 24) {
+        audioTimestamp = new AudioTimestamp();
+      }
+      while (keepAlive) {
+        int bytesRead = audioRecord.read(byteBuffer, byteBuffer.capacity());
+        if (bytesRead == byteBuffer.capacity()) {
+          if (microphoneMute) {
+            byteBuffer.clear();
+            byteBuffer.put(emptyBytes);
+          }
+          // It's possible we've been shut down during the read, and stopRecording() tried and
+          // failed to join this thread. To be a bit safer, try to avoid calling any native methods
+          // in case they've been unregistered after stopRecording() returned.
+          if (keepAlive) {
+            long captureTimeNs = 0;
+            if (Build.VERSION.SDK_INT >= 24) {
+              if (audioRecord.getTimestamp(audioTimestamp, AudioTimestamp.TIMEBASE_MONOTONIC)
+                  == AudioRecord.SUCCESS) {
+                captureTimeNs = audioTimestamp.nanoTime;
+              }
+            }
+            nativeDataIsRecorded(nativeAudioRecord, bytesRead, captureTimeNs);
+          }
+          if (audioSamplesReadyCallback != null) {
+            // Copy the entire byte buffer array. The start of the byteBuffer is not necessarily
+            // at index 0.
+            byte[] data = Arrays.copyOfRange(byteBuffer.array(), byteBuffer.arrayOffset(),
+                byteBuffer.capacity() + byteBuffer.arrayOffset());
+            audioSamplesReadyCallback.onWebRtcAudioRecordSamplesReady(
+                new JavaAudioDeviceModule.AudioSamples(audioRecord.getAudioFormat(),
+                    audioRecord.getChannelCount(), audioRecord.getSampleRate(), data));
+          }
+        } else {
+          String errorMessage = "AudioRecord.read failed: " + bytesRead;
+          Logging.e(TAG, errorMessage);
+          if (bytesRead == AudioRecord.ERROR_INVALID_OPERATION) {
+            keepAlive = false;
+            reportWebRtcAudioRecordError(errorMessage);
+          }
+        }
+      }
+
+      try {
+        if (audioRecord != null) {
+          audioRecord.stop();
+          doAudioRecordStateCallback(AUDIO_RECORD_STOP);
+        }
+      } catch (IllegalStateException e) {
+        Logging.e(TAG, "AudioRecord.stop failed: " + e.getMessage());
+      }
+    }
+
+    // Stops the inner thread loop and also calls AudioRecord.stop().
+    // Does not block the calling thread.
+    public void stopThread() {
+      Logging.d(TAG, "stopThread");
+      keepAlive = false;
+    }
+  }
+
+  @CalledByNative
+  WebRtcAudioRecord(Context context, AudioManager audioManager) {
+    this(context, newDefaultScheduler() /* scheduler */, audioManager, DEFAULT_AUDIO_SOURCE,
+        DEFAULT_AUDIO_FORMAT, null /* errorCallback */, null /* stateCallback */,
+        null /* audioSamplesReadyCallback */, WebRtcAudioEffects.isAcousticEchoCancelerSupported(),
+        WebRtcAudioEffects.isNoiseSuppressorSupported());
+  }
+
+  public WebRtcAudioRecord(Context context, ScheduledExecutorService scheduler,
+      AudioManager audioManager, int audioSource, int audioFormat,
+      @Nullable AudioRecordErrorCallback errorCallback,
+      @Nullable AudioRecordStateCallback stateCallback,
+      @Nullable SamplesReadyCallback audioSamplesReadyCallback,
+      boolean isAcousticEchoCancelerSupported, boolean isNoiseSuppressorSupported) {
+    if (isAcousticEchoCancelerSupported && !WebRtcAudioEffects.isAcousticEchoCancelerSupported()) {
+      throw new IllegalArgumentException("HW AEC not supported");
+    }
+    if (isNoiseSuppressorSupported && !WebRtcAudioEffects.isNoiseSuppressorSupported()) {
+      throw new IllegalArgumentException("HW NS not supported");
+    }
+    this.context = context;
+    this.executor = scheduler;
+    this.audioManager = audioManager;
+    this.audioSource = audioSource;
+    this.audioFormat = audioFormat;
+    this.errorCallback = errorCallback;
+    this.stateCallback = stateCallback;
+    this.audioSamplesReadyCallback = audioSamplesReadyCallback;
+    this.isAcousticEchoCancelerSupported = isAcousticEchoCancelerSupported;
+    this.isNoiseSuppressorSupported = isNoiseSuppressorSupported;
+    Logging.d(TAG, "ctor" + WebRtcAudioUtils.getThreadInfo());
+  }
+
+  @CalledByNative
+  public void setNativeAudioRecord(long nativeAudioRecord) {
+    this.nativeAudioRecord = nativeAudioRecord;
+  }
+
+  @CalledByNative
+  boolean isAcousticEchoCancelerSupported() {
+    return isAcousticEchoCancelerSupported;
+  }
+
+  @CalledByNative
+  boolean isNoiseSuppressorSupported() {
+    return isNoiseSuppressorSupported;
+  }
+
+  // Returns true if a valid call to verifyAudioConfig() has been done. Should always be
+  // checked before using the returned value of isAudioSourceMatchingRecordingSession().
+  @CalledByNative
+  boolean isAudioConfigVerified() {
+    return audioSourceMatchesRecordingSessionRef.get() != null;
+  }
+
+  // Returns true if verifyAudioConfig() succeeds. This value is set after a specific delay when
+  // startRecording() has been called. Hence, should preferably be called in combination with
+  // stopRecording() to ensure that it has been set properly. `isAudioConfigVerified` is
+  // enabled in WebRtcAudioRecord to ensure that the returned value is valid.
+  @CalledByNative
+  boolean isAudioSourceMatchingRecordingSession() {
+    Boolean audioSourceMatchesRecordingSession = audioSourceMatchesRecordingSessionRef.get();
+    if (audioSourceMatchesRecordingSession == null) {
+      Logging.w(TAG, "Audio configuration has not yet been verified");
+      return false;
+    }
+    return audioSourceMatchesRecordingSession;
+  }
+
+  @CalledByNative
+  private boolean enableBuiltInAEC(boolean enable) {
+    Logging.d(TAG, "enableBuiltInAEC(" + enable + ")");
+    return effects.setAEC(enable);
+  }
+
+  @CalledByNative
+  private boolean enableBuiltInNS(boolean enable) {
+    Logging.d(TAG, "enableBuiltInNS(" + enable + ")");
+    return effects.setNS(enable);
+  }
+
+  @CalledByNative
+  private int initRecording(int sampleRate, int channels) {
+    Logging.d(TAG, "initRecording(sampleRate=" + sampleRate + ", channels=" + channels + ")");
+    if (audioRecord != null) {
+      reportWebRtcAudioRecordInitError("InitRecording called twice without StopRecording.");
+      return -1;
+    }
+    final int bytesPerFrame = channels * getBytesPerSample(audioFormat);
+    final int framesPerBuffer = sampleRate / BUFFERS_PER_SECOND;
+    byteBuffer = ByteBuffer.allocateDirect(bytesPerFrame * framesPerBuffer);
+    if (!(byteBuffer.hasArray())) {
+      reportWebRtcAudioRecordInitError("ByteBuffer does not have backing array.");
+      return -1;
+    }
+    Logging.d(TAG, "byteBuffer.capacity: " + byteBuffer.capacity());
+    emptyBytes = new byte[byteBuffer.capacity()];
+    // Rather than passing the ByteBuffer with every callback (requiring
+    // the potentially expensive GetDirectBufferAddress) we simply have the
+    // the native class cache the address to the memory once.
+    nativeCacheDirectBufferAddress(nativeAudioRecord, byteBuffer);
+
+    // Get the minimum buffer size required for the successful creation of
+    // an AudioRecord object, in byte units.
+    // Note that this size doesn't guarantee a smooth recording under load.
+    final int channelConfig = channelCountToConfiguration(channels);
+    int minBufferSize = AudioRecord.getMinBufferSize(sampleRate, channelConfig, audioFormat);
+    if (minBufferSize == AudioRecord.ERROR || minBufferSize == AudioRecord.ERROR_BAD_VALUE) {
+      reportWebRtcAudioRecordInitError("AudioRecord.getMinBufferSize failed: " + minBufferSize);
+      return -1;
+    }
+    Logging.d(TAG, "AudioRecord.getMinBufferSize: " + minBufferSize);
+
+    // Use a larger buffer size than the minimum required when creating the
+    // AudioRecord instance to ensure smooth recording under load. It has been
+    // verified that it does not increase the actual recording latency.
+    int bufferSizeInBytes = Math.max(BUFFER_SIZE_FACTOR * minBufferSize, byteBuffer.capacity());
+    Logging.d(TAG, "bufferSizeInBytes: " + bufferSizeInBytes);
+    try {
+      if (Build.VERSION.SDK_INT >= Build.VERSION_CODES.M) {
+        // Use the AudioRecord.Builder class on Android M (23) and above.
+        // Throws IllegalArgumentException.
+        audioRecord = createAudioRecordOnMOrHigher(
+            audioSource, sampleRate, channelConfig, audioFormat, bufferSizeInBytes);
+        audioSourceMatchesRecordingSessionRef.set(null);
+        if (preferredDevice != null) {
+          setPreferredDevice(preferredDevice);
+        }
+      } else {
+        // Use the old AudioRecord constructor for API levels below 23.
+        // Throws UnsupportedOperationException.
+        audioRecord = createAudioRecordOnLowerThanM(
+            audioSource, sampleRate, channelConfig, audioFormat, bufferSizeInBytes);
+        audioSourceMatchesRecordingSessionRef.set(null);
+      }
+    } catch (IllegalArgumentException | UnsupportedOperationException e) {
+      // Report of exception message is sufficient. Example: "Cannot create AudioRecord".
+      reportWebRtcAudioRecordInitError(e.getMessage());
+      releaseAudioResources();
+      return -1;
+    }
+    if (audioRecord == null || audioRecord.getState() != AudioRecord.STATE_INITIALIZED) {
+      reportWebRtcAudioRecordInitError("Creation or initialization of audio recorder failed.");
+      releaseAudioResources();
+      return -1;
+    }
+    effects.enable(audioRecord.getAudioSessionId());
+    logMainParameters();
+    logMainParametersExtended();
+    // Check number of active recording sessions. Should be zero but we have seen conflict cases
+    // and adding a log for it can help us figure out details about conflicting sessions.
+    final int numActiveRecordingSessions =
+        logRecordingConfigurations(audioRecord, false /* verifyAudioConfig */);
+    if (numActiveRecordingSessions != 0) {
+      // Log the conflict as a warning since initialization did in fact succeed. Most likely, the
+      // upcoming call to startRecording() will fail under these conditions.
+      Logging.w(
+          TAG, "Potential microphone conflict. Active sessions: " + numActiveRecordingSessions);
+    }
+    return framesPerBuffer;
+  }
+
+  /**
+   * Prefer a specific {@link AudioDeviceInfo} device for recording. Calling after recording starts
+   * is valid but may cause a temporary interruption if the audio routing changes.
+   */
+  @RequiresApi(Build.VERSION_CODES.M)
+  @TargetApi(Build.VERSION_CODES.M)
+  void setPreferredDevice(@Nullable AudioDeviceInfo preferredDevice) {
+    Logging.d(
+        TAG, "setPreferredDevice " + (preferredDevice != null ? preferredDevice.getId() : null));
+    this.preferredDevice = preferredDevice;
+    if (audioRecord != null) {
+      if (!audioRecord.setPreferredDevice(preferredDevice)) {
+        Logging.e(TAG, "setPreferredDevice failed");
+      }
+    }
+  }
+
+  @CalledByNative
+  private boolean startRecording() {
+    Logging.d(TAG, "startRecording");
+    assertTrue(audioRecord != null);
+    assertTrue(audioThread == null);
+    try {
+      audioRecord.startRecording();
+    } catch (IllegalStateException e) {
+      reportWebRtcAudioRecordStartError(AudioRecordStartErrorCode.AUDIO_RECORD_START_EXCEPTION,
+          "AudioRecord.startRecording failed: " + e.getMessage());
+      return false;
+    }
+    if (audioRecord.getRecordingState() != AudioRecord.RECORDSTATE_RECORDING) {
+      reportWebRtcAudioRecordStartError(AudioRecordStartErrorCode.AUDIO_RECORD_START_STATE_MISMATCH,
+          "AudioRecord.startRecording failed - incorrect state: "
+              + audioRecord.getRecordingState());
+      return false;
+    }
+    audioThread = new AudioRecordThread("AudioRecordJavaThread");
+    audioThread.start();
+    scheduleLogRecordingConfigurationsTask(audioRecord);
+    return true;
+  }
+
+  @CalledByNative
+  private boolean stopRecording() {
+    Logging.d(TAG, "stopRecording");
+    assertTrue(audioThread != null);
+    if (future != null) {
+      if (!future.isDone()) {
+        // Might be needed if the client calls startRecording(), stopRecording() back-to-back.
+        future.cancel(true /* mayInterruptIfRunning */);
+      }
+      future = null;
+    }
+    audioThread.stopThread();
+    if (!ThreadUtils.joinUninterruptibly(audioThread, AUDIO_RECORD_THREAD_JOIN_TIMEOUT_MS)) {
+      Logging.e(TAG, "Join of AudioRecordJavaThread timed out");
+      WebRtcAudioUtils.logAudioState(TAG, context, audioManager);
+    }
+    audioThread = null;
+    effects.release();
+    releaseAudioResources();
+    return true;
+  }
+
+  @TargetApi(Build.VERSION_CODES.M)
+  private static AudioRecord createAudioRecordOnMOrHigher(
+      int audioSource, int sampleRate, int channelConfig, int audioFormat, int bufferSizeInBytes) {
+    Logging.d(TAG, "createAudioRecordOnMOrHigher");
+    return new AudioRecord.Builder()
+        .setAudioSource(audioSource)
+        .setAudioFormat(new AudioFormat.Builder()
+                            .setEncoding(audioFormat)
+                            .setSampleRate(sampleRate)
+                            .setChannelMask(channelConfig)
+                            .build())
+        .setBufferSizeInBytes(bufferSizeInBytes)
+        .build();
+  }
+
+  private static AudioRecord createAudioRecordOnLowerThanM(
+      int audioSource, int sampleRate, int channelConfig, int audioFormat, int bufferSizeInBytes) {
+    Logging.d(TAG, "createAudioRecordOnLowerThanM");
+    return new AudioRecord(audioSource, sampleRate, channelConfig, audioFormat, bufferSizeInBytes);
+  }
+
+  private void logMainParameters() {
+    Logging.d(TAG,
+        "AudioRecord: "
+            + "session ID: " + audioRecord.getAudioSessionId() + ", "
+            + "channels: " + audioRecord.getChannelCount() + ", "
+            + "sample rate: " + audioRecord.getSampleRate());
+  }
+
+  @TargetApi(Build.VERSION_CODES.M)
+  private void logMainParametersExtended() {
+    if (Build.VERSION.SDK_INT >= Build.VERSION_CODES.M) {
+      Logging.d(TAG,
+          "AudioRecord: "
+              // The frame count of the native AudioRecord buffer.
+              + "buffer size in frames: " + audioRecord.getBufferSizeInFrames());
+    }
+  }
+
+  @TargetApi(Build.VERSION_CODES.N)
+  // Checks the number of active recording sessions and logs the states of all active sessions.
+  // Returns number of active sessions. Note that this could occur on arbituary thread.
+  private int logRecordingConfigurations(AudioRecord audioRecord, boolean verifyAudioConfig) {
+    if (Build.VERSION.SDK_INT < Build.VERSION_CODES.N) {
+      Logging.w(TAG, "AudioManager#getActiveRecordingConfigurations() requires N or higher");
+      return 0;
+    }
+    if (audioRecord == null) {
+      return 0;
+    }
+
+    // Get a list of the currently active audio recording configurations of the device (can be more
+    // than one). An empty list indicates there is no recording active when queried.
+    List<AudioRecordingConfiguration> configs = audioManager.getActiveRecordingConfigurations();
+    final int numActiveRecordingSessions = configs.size();
+    Logging.d(TAG, "Number of active recording sessions: " + numActiveRecordingSessions);
+    if (numActiveRecordingSessions > 0) {
+      logActiveRecordingConfigs(audioRecord.getAudioSessionId(), configs);
+      if (verifyAudioConfig) {
+        // Run an extra check to verify that the existing audio source doing the recording (tied
+        // to the AudioRecord instance) is matching what the audio recording configuration lists
+        // as its client parameters. If these do not match, recording might work but under invalid
+        // conditions.
+        audioSourceMatchesRecordingSessionRef.set(
+            verifyAudioConfig(audioRecord.getAudioSource(), audioRecord.getAudioSessionId(),
+                audioRecord.getFormat(), audioRecord.getRoutedDevice(), configs));
+      }
+    }
+    return numActiveRecordingSessions;
+  }
+
+  // Helper method which throws an exception  when an assertion has failed.
+  private static void assertTrue(boolean condition) {
+    if (!condition) {
+      throw new AssertionError("Expected condition to be true");
+    }
+  }
+
+  private int channelCountToConfiguration(int channels) {
+    return (channels == 1 ? AudioFormat.CHANNEL_IN_MONO : AudioFormat.CHANNEL_IN_STEREO);
+  }
+
+  private native void nativeCacheDirectBufferAddress(
+      long nativeAudioRecordJni, ByteBuffer byteBuffer);
+  private native void nativeDataIsRecorded(
+      long nativeAudioRecordJni, int bytes, long captureTimestampNs);
+
+  // Sets all recorded samples to zero if `mute` is true, i.e., ensures that
+  // the microphone is muted.
+  public void setMicrophoneMute(boolean mute) {
+    Logging.w(TAG, "setMicrophoneMute(" + mute + ")");
+    microphoneMute = mute;
+  }
+
+  // Releases the native AudioRecord resources.
+  private void releaseAudioResources() {
+    Logging.d(TAG, "releaseAudioResources");
+    if (audioRecord != null) {
+      audioRecord.release();
+      audioRecord = null;
+    }
+    audioSourceMatchesRecordingSessionRef.set(null);
+  }
+
+  private void reportWebRtcAudioRecordInitError(String errorMessage) {
+    Logging.e(TAG, "Init recording error: " + errorMessage);
+    WebRtcAudioUtils.logAudioState(TAG, context, audioManager);
+    logRecordingConfigurations(audioRecord, false /* verifyAudioConfig */);
+    if (errorCallback != null) {
+      errorCallback.onWebRtcAudioRecordInitError(errorMessage);
+    }
+  }
+
+  private void reportWebRtcAudioRecordStartError(
+      AudioRecordStartErrorCode errorCode, String errorMessage) {
+    Logging.e(TAG, "Start recording error: " + errorCode + ". " + errorMessage);
+    WebRtcAudioUtils.logAudioState(TAG, context, audioManager);
+    logRecordingConfigurations(audioRecord, false /* verifyAudioConfig */);
+    if (errorCallback != null) {
+      errorCallback.onWebRtcAudioRecordStartError(errorCode, errorMessage);
+    }
+  }
+
+  private void reportWebRtcAudioRecordError(String errorMessage) {
+    Logging.e(TAG, "Run-time recording error: " + errorMessage);
+    WebRtcAudioUtils.logAudioState(TAG, context, audioManager);
+    if (errorCallback != null) {
+      errorCallback.onWebRtcAudioRecordError(errorMessage);
+    }
+  }
+
+  private void doAudioRecordStateCallback(int audioState) {
+    Logging.d(TAG, "doAudioRecordStateCallback: " + audioStateToString(audioState));
+    if (stateCallback != null) {
+      if (audioState == WebRtcAudioRecord.AUDIO_RECORD_START) {
+        stateCallback.onWebRtcAudioRecordStart();
+      } else if (audioState == WebRtcAudioRecord.AUDIO_RECORD_STOP) {
+        stateCallback.onWebRtcAudioRecordStop();
+      } else {
+        Logging.e(TAG, "Invalid audio state");
+      }
+    }
+  }
+
+  // Reference from Android code, AudioFormat.getBytesPerSample. BitPerSample / 8
+  // Default audio data format is PCM 16 bits per sample.
+  // Guaranteed to be supported by all devices
+  private static int getBytesPerSample(int audioFormat) {
+    switch (audioFormat) {
+      case AudioFormat.ENCODING_PCM_8BIT:
+        return 1;
+      case AudioFormat.ENCODING_PCM_16BIT:
+      case AudioFormat.ENCODING_IEC61937:
+      case AudioFormat.ENCODING_DEFAULT:
+        return 2;
+      case AudioFormat.ENCODING_PCM_FLOAT:
+        return 4;
+      case AudioFormat.ENCODING_INVALID:
+      default:
+        throw new IllegalArgumentException("Bad audio format " + audioFormat);
+    }
+  }
+
+  // Use an ExecutorService to schedule a task after a given delay where the task consists of
+  // checking (by logging) the current status of active recording sessions.
+  private void scheduleLogRecordingConfigurationsTask(AudioRecord audioRecord) {
+    Logging.d(TAG, "scheduleLogRecordingConfigurationsTask");
+    if (Build.VERSION.SDK_INT < Build.VERSION_CODES.N) {
+      return;
+    }
+
+    Callable<String> callable = () -> {
+      if (this.audioRecord == audioRecord) {
+        logRecordingConfigurations(audioRecord, true /* verifyAudioConfig */);
+      } else {
+        Logging.d(TAG, "audio record has changed");
+      }
+      return "Scheduled task is done";
+    };
+
+    if (future != null && !future.isDone()) {
+      future.cancel(true /* mayInterruptIfRunning */);
+    }
+    // Schedule call to logRecordingConfigurations() from executor thread after fixed delay.
+    future = executor.schedule(callable, CHECK_REC_STATUS_DELAY_MS, TimeUnit.MILLISECONDS);
+  };
+
+  @TargetApi(Build.VERSION_CODES.N)
+  private static boolean logActiveRecordingConfigs(
+      int session, List<AudioRecordingConfiguration> configs) {
+    assertTrue(!configs.isEmpty());
+    final Iterator<AudioRecordingConfiguration> it = configs.iterator();
+    Logging.d(TAG, "AudioRecordingConfigurations: ");
+    while (it.hasNext()) {
+      final AudioRecordingConfiguration config = it.next();
+      StringBuilder conf = new StringBuilder();
+      // The audio source selected by the client.
+      final int audioSource = config.getClientAudioSource();
+      conf.append("  client audio source=")
+          .append(WebRtcAudioUtils.audioSourceToString(audioSource))
+          .append(", client session id=")
+          .append(config.getClientAudioSessionId())
+          // Compare with our own id (based on AudioRecord#getAudioSessionId()).
+          .append(" (")
+          .append(session)
+          .append(")")
+          .append("\n");
+      // Audio format at which audio is recorded on this Android device. Note that it may differ
+      // from the client application recording format (see getClientFormat()).
+      AudioFormat format = config.getFormat();
+      conf.append("  Device AudioFormat: ")
+          .append("channel count=")
+          .append(format.getChannelCount())
+          .append(", channel index mask=")
+          .append(format.getChannelIndexMask())
+          // Only AudioFormat#CHANNEL_IN_MONO is guaranteed to work on all devices.
+          .append(", channel mask=")
+          .append(WebRtcAudioUtils.channelMaskToString(format.getChannelMask()))
+          .append(", encoding=")
+          .append(WebRtcAudioUtils.audioEncodingToString(format.getEncoding()))
+          .append(", sample rate=")
+          .append(format.getSampleRate())
+          .append("\n");
+      // Audio format at which the client application is recording audio.
+      format = config.getClientFormat();
+      conf.append("  Client AudioFormat: ")
+          .append("channel count=")
+          .append(format.getChannelCount())
+          .append(", channel index mask=")
+          .append(format.getChannelIndexMask())
+          // Only AudioFormat#CHANNEL_IN_MONO is guaranteed to work on all devices.
+          .append(", channel mask=")
+          .append(WebRtcAudioUtils.channelMaskToString(format.getChannelMask()))
+          .append(", encoding=")
+          .append(WebRtcAudioUtils.audioEncodingToString(format.getEncoding()))
+          .append(", sample rate=")
+          .append(format.getSampleRate())
+          .append("\n");
+      // Audio input device used for this recording session.
+      final AudioDeviceInfo device = config.getAudioDevice();
+      if (device != null) {
+        assertTrue(device.isSource());
+        conf.append("  AudioDevice: ")
+            .append("type=")
+            .append(WebRtcAudioUtils.deviceTypeToString(device.getType()))
+            .append(", id=")
+            .append(device.getId());
+      }
+      Logging.d(TAG, conf.toString());
+    }
+    return true;
+  }
+
+  // Verify that the client audio configuration (device and format) matches the requested
+  // configuration (same as AudioRecord's).
+  @TargetApi(Build.VERSION_CODES.N)
+  private static boolean verifyAudioConfig(int source, int session, AudioFormat format,
+      AudioDeviceInfo device, List<AudioRecordingConfiguration> configs) {
+    assertTrue(!configs.isEmpty());
+    final Iterator<AudioRecordingConfiguration> it = configs.iterator();
+    while (it.hasNext()) {
+      final AudioRecordingConfiguration config = it.next();
+      final AudioDeviceInfo configDevice = config.getAudioDevice();
+      if (configDevice == null) {
+        continue;
+      }
+      if ((config.getClientAudioSource() == source)
+          && (config.getClientAudioSessionId() == session)
+          // Check the client format (should match the format of the AudioRecord instance).
+          && (config.getClientFormat().getEncoding() == format.getEncoding())
+          && (config.getClientFormat().getSampleRate() == format.getSampleRate())
+          && (config.getClientFormat().getChannelMask() == format.getChannelMask())
+          && (config.getClientFormat().getChannelIndexMask() == format.getChannelIndexMask())
+          // Ensure that the device format is properly configured.
+          && (config.getFormat().getEncoding() != AudioFormat.ENCODING_INVALID)
+          && (config.getFormat().getSampleRate() > 0)
+          //  For the channel mask, either the position or index-based value must be valid.
+          && ((config.getFormat().getChannelMask() != AudioFormat.CHANNEL_INVALID)
+              || (config.getFormat().getChannelIndexMask() != AudioFormat.CHANNEL_INVALID))
+          && checkDeviceMatch(configDevice, device)) {
+        Logging.d(TAG, "verifyAudioConfig: PASS");
+        return true;
+      }
+    }
+    Logging.e(TAG, "verifyAudioConfig: FAILED");
+    return false;
+  }
+
+  @TargetApi(Build.VERSION_CODES.N)
+  // Returns true if device A parameters matches those of device B.
+  // TODO(henrika): can be improved by adding AudioDeviceInfo#getAddress() but it requires API 29.
+  private static boolean checkDeviceMatch(AudioDeviceInfo devA, AudioDeviceInfo devB) {
+    return ((devA.getId() == devB.getId() && (devA.getType() == devB.getType())));
+  }
+
+  private static String audioStateToString(int state) {
+    switch (state) {
+      case WebRtcAudioRecord.AUDIO_RECORD_START:
+        return "START";
+      case WebRtcAudioRecord.AUDIO_RECORD_STOP:
+        return "STOP";
+      default:
+        return "INVALID";
+    }
+  }
+
+  private static final AtomicInteger nextSchedulerId = new AtomicInteger(0);
+
+  static ScheduledExecutorService newDefaultScheduler() {
+    AtomicInteger nextThreadId = new AtomicInteger(0);
+    return Executors.newScheduledThreadPool(0, new ThreadFactory() {
+      /**
+       * Constructs a new {@code Thread}
+       */
+      @Override
+      public Thread newThread(Runnable r) {
+        Thread thread = Executors.defaultThreadFactory().newThread(r);
+        thread.setName(String.format("WebRtcAudioRecordScheduler-%s-%s",
+            nextSchedulerId.getAndIncrement(), nextThreadId.getAndIncrement()));
+        return thread;
+      }
+    });
+  }
+}
diff --git a/third_party/libwebrtc/sdk/android/src/java/org/webrtc/audio/WebRtcAudioTrack.java b/third_party/libwebrtc/sdk/android/src/java/org/webrtc/audio/WebRtcAudioTrack.java
new file mode 100644
index 0000000000..2b34e34013
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/src/java/org/webrtc/audio/WebRtcAudioTrack.java
@@ -0,0 +1,585 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc.audio;
+
+import android.annotation.TargetApi;
+import android.content.Context;
+import android.media.AudioAttributes;
+import android.media.AudioFormat;
+import android.media.AudioManager;
+import android.media.AudioTrack;
+import android.os.Build;
+import android.os.Process;
+import androidx.annotation.Nullable;
+import java.nio.ByteBuffer;
+import org.webrtc.CalledByNative;
+import org.webrtc.Logging;
+import org.webrtc.ThreadUtils;
+import org.webrtc.audio.JavaAudioDeviceModule.AudioTrackErrorCallback;
+import org.webrtc.audio.JavaAudioDeviceModule.AudioTrackStartErrorCode;
+import org.webrtc.audio.JavaAudioDeviceModule.AudioTrackStateCallback;
+import org.webrtc.audio.LowLatencyAudioBufferManager;
+
+class WebRtcAudioTrack {
+  private static final String TAG = "WebRtcAudioTrackExternal";
+
+  // Default audio data format is PCM 16 bit per sample.
+  // Guaranteed to be supported by all devices.
+  private static final int BITS_PER_SAMPLE = 16;
+
+  // Requested size of each recorded buffer provided to the client.
+  private static final int CALLBACK_BUFFER_SIZE_MS = 10;
+
+  // Average number of callbacks per second.
+  private static final int BUFFERS_PER_SECOND = 1000 / CALLBACK_BUFFER_SIZE_MS;
+
+  // The AudioTrackThread is allowed to wait for successful call to join()
+  // but the wait times out afther this amount of time.
+  private static final long AUDIO_TRACK_THREAD_JOIN_TIMEOUT_MS = 2000;
+
+  // By default, WebRTC creates audio tracks with a usage attribute
+  // corresponding to voice communications, such as telephony or VoIP.
+  private static final int DEFAULT_USAGE = AudioAttributes.USAGE_VOICE_COMMUNICATION;
+
+  // Indicates the AudioTrack has started playing audio.
+  private static final int AUDIO_TRACK_START = 0;
+
+  // Indicates the AudioTrack has stopped playing audio.
+  private static final int AUDIO_TRACK_STOP = 1;
+
+  private long nativeAudioTrack;
+  private final Context context;
+  private final AudioManager audioManager;
+  private final ThreadUtils.ThreadChecker threadChecker = new ThreadUtils.ThreadChecker();
+
+  private ByteBuffer byteBuffer;
+
+  private @Nullable final AudioAttributes audioAttributes;
+  private @Nullable AudioTrack audioTrack;
+  private @Nullable AudioTrackThread audioThread;
+  private final VolumeLogger volumeLogger;
+
+  // Samples to be played are replaced by zeros if `speakerMute` is set to true.
+  // Can be used to ensure that the speaker is fully muted.
+  private volatile boolean speakerMute;
+  private byte[] emptyBytes;
+  private boolean useLowLatency;
+  private int initialBufferSizeInFrames;
+
+  private final @Nullable AudioTrackErrorCallback errorCallback;
+  private final @Nullable AudioTrackStateCallback stateCallback;
+
+  /**
+   * Audio thread which keeps calling AudioTrack.write() to stream audio.
+   * Data is periodically acquired from the native WebRTC layer using the
+   * nativeGetPlayoutData callback function.
+   * This thread uses a Process.THREAD_PRIORITY_URGENT_AUDIO priority.
+   */
+  private class AudioTrackThread extends Thread {
+    private volatile boolean keepAlive = true;
+    private LowLatencyAudioBufferManager bufferManager;
+
+    public AudioTrackThread(String name) {
+      super(name);
+      bufferManager = new LowLatencyAudioBufferManager();
+    }
+
+    @Override
+    public void run() {
+      Process.setThreadPriority(Process.THREAD_PRIORITY_URGENT_AUDIO);
+      Logging.d(TAG, "AudioTrackThread" + WebRtcAudioUtils.getThreadInfo());
+      assertTrue(audioTrack.getPlayState() == AudioTrack.PLAYSTATE_PLAYING);
+
+      // Audio playout has started and the client is informed about it.
+      doAudioTrackStateCallback(AUDIO_TRACK_START);
+
+      // Fixed size in bytes of each 10ms block of audio data that we ask for
+      // using callbacks to the native WebRTC client.
+      final int sizeInBytes = byteBuffer.capacity();
+
+      while (keepAlive) {
+        // Get 10ms of PCM data from the native WebRTC client. Audio data is
+        // written into the common ByteBuffer using the address that was
+        // cached at construction.
+        nativeGetPlayoutData(nativeAudioTrack, sizeInBytes);
+        // Write data until all data has been written to the audio sink.
+        // Upon return, the buffer position will have been advanced to reflect
+        // the amount of data that was successfully written to the AudioTrack.
+        assertTrue(sizeInBytes <= byteBuffer.remaining());
+        if (speakerMute) {
+          byteBuffer.clear();
+          byteBuffer.put(emptyBytes);
+          byteBuffer.position(0);
+        }
+        int bytesWritten = audioTrack.write(byteBuffer, sizeInBytes, AudioTrack.WRITE_BLOCKING);
+        if (bytesWritten != sizeInBytes) {
+          Logging.e(TAG, "AudioTrack.write played invalid number of bytes: " + bytesWritten);
+          // If a write() returns a negative value, an error has occurred.
+          // Stop playing and report an error in this case.
+          if (bytesWritten < 0) {
+            keepAlive = false;
+            reportWebRtcAudioTrackError("AudioTrack.write failed: " + bytesWritten);
+          }
+        }
+        if (useLowLatency) {
+          bufferManager.maybeAdjustBufferSize(audioTrack);
+        }
+        // The byte buffer must be rewinded since byteBuffer.position() is
+        // increased at each call to AudioTrack.write(). If we don't do this,
+        // next call to AudioTrack.write() will fail.
+        byteBuffer.rewind();
+
+        // TODO(henrika): it is possible to create a delay estimate here by
+        // counting number of written frames and subtracting the result from
+        // audioTrack.getPlaybackHeadPosition().
+      }
+    }
+
+    // Stops the inner thread loop which results in calling AudioTrack.stop().
+    // Does not block the calling thread.
+    public void stopThread() {
+      Logging.d(TAG, "stopThread");
+      keepAlive = false;
+    }
+  }
+
+  @CalledByNative
+  WebRtcAudioTrack(Context context, AudioManager audioManager) {
+    this(context, audioManager, null /* audioAttributes */, null /* errorCallback */,
+        null /* stateCallback */, false /* useLowLatency */, true /* enableVolumeLogger */);
+  }
+
+  WebRtcAudioTrack(Context context, AudioManager audioManager,
+      @Nullable AudioAttributes audioAttributes, @Nullable AudioTrackErrorCallback errorCallback,
+      @Nullable AudioTrackStateCallback stateCallback, boolean useLowLatency,
+      boolean enableVolumeLogger) {
+    threadChecker.detachThread();
+    this.context = context;
+    this.audioManager = audioManager;
+    this.audioAttributes = audioAttributes;
+    this.errorCallback = errorCallback;
+    this.stateCallback = stateCallback;
+    this.volumeLogger = enableVolumeLogger ? new VolumeLogger(audioManager) : null;
+    this.useLowLatency = useLowLatency;
+    Logging.d(TAG, "ctor" + WebRtcAudioUtils.getThreadInfo());
+  }
+
+  @CalledByNative
+  public void setNativeAudioTrack(long nativeAudioTrack) {
+    this.nativeAudioTrack = nativeAudioTrack;
+  }
+
+  @CalledByNative
+  private int initPlayout(int sampleRate, int channels, double bufferSizeFactor) {
+    threadChecker.checkIsOnValidThread();
+    Logging.d(TAG,
+        "initPlayout(sampleRate=" + sampleRate + ", channels=" + channels
+            + ", bufferSizeFactor=" + bufferSizeFactor + ")");
+    final int bytesPerFrame = channels * (BITS_PER_SAMPLE / 8);
+    byteBuffer = ByteBuffer.allocateDirect(bytesPerFrame * (sampleRate / BUFFERS_PER_SECOND));
+    Logging.d(TAG, "byteBuffer.capacity: " + byteBuffer.capacity());
+    emptyBytes = new byte[byteBuffer.capacity()];
+    // Rather than passing the ByteBuffer with every callback (requiring
+    // the potentially expensive GetDirectBufferAddress) we simply have the
+    // the native class cache the address to the memory once.
+    nativeCacheDirectBufferAddress(nativeAudioTrack, byteBuffer);
+
+    // Get the minimum buffer size required for the successful creation of an
+    // AudioTrack object to be created in the MODE_STREAM mode.
+    // Note that this size doesn't guarantee a smooth playback under load.
+    final int channelConfig = channelCountToConfiguration(channels);
+    final int minBufferSizeInBytes = (int) (AudioTrack.getMinBufferSize(sampleRate, channelConfig,
+                                                AudioFormat.ENCODING_PCM_16BIT)
+        * bufferSizeFactor);
+    Logging.d(TAG, "minBufferSizeInBytes: " + minBufferSizeInBytes);
+    // For the streaming mode, data must be written to the audio sink in
+    // chunks of size (given by byteBuffer.capacity()) less than or equal
+    // to the total buffer size `minBufferSizeInBytes`. But, we have seen
+    // reports of "getMinBufferSize(): error querying hardware". Hence, it
+    // can happen that `minBufferSizeInBytes` contains an invalid value.
+    if (minBufferSizeInBytes < byteBuffer.capacity()) {
+      reportWebRtcAudioTrackInitError("AudioTrack.getMinBufferSize returns an invalid value.");
+      return -1;
+    }
+
+    // Don't use low-latency mode when a bufferSizeFactor > 1 is used. When bufferSizeFactor > 1
+    // we want to use a larger buffer to prevent underruns. However, low-latency mode would
+    // decrease the buffer size, which makes the bufferSizeFactor have no effect.
+    if (bufferSizeFactor > 1.0) {
+      useLowLatency = false;
+    }
+
+    // Ensure that prevision audio session was stopped correctly before trying
+    // to create a new AudioTrack.
+    if (audioTrack != null) {
+      reportWebRtcAudioTrackInitError("Conflict with existing AudioTrack.");
+      return -1;
+    }
+    try {
+      // Create an AudioTrack object and initialize its associated audio buffer.
+      // The size of this buffer determines how long an AudioTrack can play
+      // before running out of data.
+      if (useLowLatency && Build.VERSION.SDK_INT >= Build.VERSION_CODES.O) {
+        // On API level 26 or higher, we can use a low latency mode.
+        audioTrack = createAudioTrackOnOreoOrHigher(
+            sampleRate, channelConfig, minBufferSizeInBytes, audioAttributes);
+      } else {
+        // As we are on API level 21 or higher, it is possible to use a special AudioTrack
+        // constructor that uses AudioAttributes and AudioFormat as input. It allows us to
+        // supersede the notion of stream types for defining the behavior of audio playback,
+        // and to allow certain platforms or routing policies to use this information for more
+        // refined volume or routing decisions.
+        audioTrack = createAudioTrackBeforeOreo(
+            sampleRate, channelConfig, minBufferSizeInBytes, audioAttributes);
+      }
+    } catch (IllegalArgumentException e) {
+      reportWebRtcAudioTrackInitError(e.getMessage());
+      releaseAudioResources();
+      return -1;
+    }
+
+    // It can happen that an AudioTrack is created but it was not successfully
+    // initialized upon creation. Seems to be the case e.g. when the maximum
+    // number of globally available audio tracks is exceeded.
+    if (audioTrack == null || audioTrack.getState() != AudioTrack.STATE_INITIALIZED) {
+      reportWebRtcAudioTrackInitError("Initialization of audio track failed.");
+      releaseAudioResources();
+      return -1;
+    }
+    if (Build.VERSION.SDK_INT >= Build.VERSION_CODES.M) {
+      initialBufferSizeInFrames = audioTrack.getBufferSizeInFrames();
+    } else {
+      initialBufferSizeInFrames = -1;
+    }
+    logMainParameters();
+    logMainParametersExtended();
+    return minBufferSizeInBytes;
+  }
+
+  @CalledByNative
+  private boolean startPlayout() {
+    threadChecker.checkIsOnValidThread();
+    if (volumeLogger != null) {
+      volumeLogger.start();
+    }
+    Logging.d(TAG, "startPlayout");
+    assertTrue(audioTrack != null);
+    assertTrue(audioThread == null);
+
+    // Starts playing an audio track.
+    try {
+      audioTrack.play();
+    } catch (IllegalStateException e) {
+      reportWebRtcAudioTrackStartError(AudioTrackStartErrorCode.AUDIO_TRACK_START_EXCEPTION,
+          "AudioTrack.play failed: " + e.getMessage());
+      releaseAudioResources();
+      return false;
+    }
+    if (audioTrack.getPlayState() != AudioTrack.PLAYSTATE_PLAYING) {
+      reportWebRtcAudioTrackStartError(AudioTrackStartErrorCode.AUDIO_TRACK_START_STATE_MISMATCH,
+          "AudioTrack.play failed - incorrect state :" + audioTrack.getPlayState());
+      releaseAudioResources();
+      return false;
+    }
+
+    // Create and start new high-priority thread which calls AudioTrack.write()
+    // and where we also call the native nativeGetPlayoutData() callback to
+    // request decoded audio from WebRTC.
+    audioThread = new AudioTrackThread("AudioTrackJavaThread");
+    audioThread.start();
+    return true;
+  }
+
+  @CalledByNative
+  private boolean stopPlayout() {
+    threadChecker.checkIsOnValidThread();
+    if (volumeLogger != null) {
+      volumeLogger.stop();
+    }
+    Logging.d(TAG, "stopPlayout");
+    assertTrue(audioThread != null);
+    logUnderrunCount();
+    audioThread.stopThread();
+
+    Logging.d(TAG, "Stopping the AudioTrackThread...");
+    audioThread.interrupt();
+    if (!ThreadUtils.joinUninterruptibly(audioThread, AUDIO_TRACK_THREAD_JOIN_TIMEOUT_MS)) {
+      Logging.e(TAG, "Join of AudioTrackThread timed out.");
+      WebRtcAudioUtils.logAudioState(TAG, context, audioManager);
+    }
+    Logging.d(TAG, "AudioTrackThread has now been stopped.");
+    audioThread = null;
+    if (audioTrack != null) {
+      Logging.d(TAG, "Calling AudioTrack.stop...");
+      try {
+        audioTrack.stop();
+        Logging.d(TAG, "AudioTrack.stop is done.");
+        doAudioTrackStateCallback(AUDIO_TRACK_STOP);
+      } catch (IllegalStateException e) {
+        Logging.e(TAG, "AudioTrack.stop failed: " + e.getMessage());
+      }
+    }
+    releaseAudioResources();
+    return true;
+  }
+
+  // Get max possible volume index for a phone call audio stream.
+  @CalledByNative
+  private int getStreamMaxVolume() {
+    threadChecker.checkIsOnValidThread();
+    Logging.d(TAG, "getStreamMaxVolume");
+    return audioManager.getStreamMaxVolume(AudioManager.STREAM_VOICE_CALL);
+  }
+
+  // Set current volume level for a phone call audio stream.
+  @CalledByNative
+  private boolean setStreamVolume(int volume) {
+    threadChecker.checkIsOnValidThread();
+    Logging.d(TAG, "setStreamVolume(" + volume + ")");
+    if (audioManager.isVolumeFixed()) {
+      Logging.e(TAG, "The device implements a fixed volume policy.");
+      return false;
+    }
+    audioManager.setStreamVolume(AudioManager.STREAM_VOICE_CALL, volume, 0);
+    return true;
+  }
+
+  /** Get current volume level for a phone call audio stream. */
+  @CalledByNative
+  private int getStreamVolume() {
+    threadChecker.checkIsOnValidThread();
+    Logging.d(TAG, "getStreamVolume");
+    return audioManager.getStreamVolume(AudioManager.STREAM_VOICE_CALL);
+  }
+
+  @CalledByNative
+  private int GetPlayoutUnderrunCount() {
+    if (Build.VERSION.SDK_INT >= 24) {
+      if (audioTrack != null) {
+        return audioTrack.getUnderrunCount();
+      } else {
+        return -1;
+      }
+    } else {
+      return -2;
+    }
+  }
+
+  private void logMainParameters() {
+    Logging.d(TAG,
+        "AudioTrack: "
+            + "session ID: " + audioTrack.getAudioSessionId() + ", "
+            + "channels: " + audioTrack.getChannelCount() + ", "
+            + "sample rate: " + audioTrack.getSampleRate()
+            + ", "
+            // Gain (>=1.0) expressed as linear multiplier on sample values.
+            + "max gain: " + AudioTrack.getMaxVolume());
+  }
+
+  private static void logNativeOutputSampleRate(int requestedSampleRateInHz) {
+    final int nativeOutputSampleRate =
+        AudioTrack.getNativeOutputSampleRate(AudioManager.STREAM_VOICE_CALL);
+    Logging.d(TAG, "nativeOutputSampleRate: " + nativeOutputSampleRate);
+    if (requestedSampleRateInHz != nativeOutputSampleRate) {
+      Logging.w(TAG, "Unable to use fast mode since requested sample rate is not native");
+    }
+  }
+
+  private static AudioAttributes getAudioAttributes(@Nullable AudioAttributes overrideAttributes) {
+    AudioAttributes.Builder attributesBuilder =
+        new AudioAttributes.Builder()
+            .setUsage(DEFAULT_USAGE)
+            .setContentType(AudioAttributes.CONTENT_TYPE_SPEECH);
+
+    if (overrideAttributes != null) {
+      if (overrideAttributes.getUsage() != AudioAttributes.USAGE_UNKNOWN) {
+        attributesBuilder.setUsage(overrideAttributes.getUsage());
+      }
+      if (overrideAttributes.getContentType() != AudioAttributes.CONTENT_TYPE_UNKNOWN) {
+        attributesBuilder.setContentType(overrideAttributes.getContentType());
+      }
+
+      attributesBuilder.setFlags(overrideAttributes.getFlags());
+
+      if (Build.VERSION.SDK_INT >= Build.VERSION_CODES.Q) {
+        attributesBuilder = applyAttributesOnQOrHigher(attributesBuilder, overrideAttributes);
+      }
+    }
+    return attributesBuilder.build();
+  }
+
+  // Creates and AudioTrack instance using AudioAttributes and AudioFormat as input.
+  // It allows certain platforms or routing policies to use this information for more
+  // refined volume or routing decisions.
+  private static AudioTrack createAudioTrackBeforeOreo(int sampleRateInHz, int channelConfig,
+      int bufferSizeInBytes, @Nullable AudioAttributes overrideAttributes) {
+    Logging.d(TAG, "createAudioTrackBeforeOreo");
+    logNativeOutputSampleRate(sampleRateInHz);
+
+    // Create an audio track where the audio usage is for VoIP and the content type is speech.
+    return new AudioTrack(getAudioAttributes(overrideAttributes),
+        new AudioFormat.Builder()
+            .setEncoding(AudioFormat.ENCODING_PCM_16BIT)
+            .setSampleRate(sampleRateInHz)
+            .setChannelMask(channelConfig)
+            .build(),
+        bufferSizeInBytes, AudioTrack.MODE_STREAM, AudioManager.AUDIO_SESSION_ID_GENERATE);
+  }
+
+  // Creates and AudioTrack instance using AudioAttributes and AudioFormat as input.
+  // Use the low-latency mode to improve audio latency. Note that the low-latency mode may
+  // prevent effects (such as AEC) from working. Assuming AEC is working, the delay changes
+  // that happen in low-latency mode during the call will cause the AEC to perform worse.
+  // The behavior of the low-latency mode may be device dependent, use at your own risk.
+  @TargetApi(Build.VERSION_CODES.O)
+  private static AudioTrack createAudioTrackOnOreoOrHigher(int sampleRateInHz, int channelConfig,
+      int bufferSizeInBytes, @Nullable AudioAttributes overrideAttributes) {
+    Logging.d(TAG, "createAudioTrackOnOreoOrHigher");
+    logNativeOutputSampleRate(sampleRateInHz);
+
+    // Create an audio track where the audio usage is for VoIP and the content type is speech.
+    return new AudioTrack.Builder()
+        .setAudioAttributes(getAudioAttributes(overrideAttributes))
+        .setAudioFormat(new AudioFormat.Builder()
+                            .setEncoding(AudioFormat.ENCODING_PCM_16BIT)
+                            .setSampleRate(sampleRateInHz)
+                            .setChannelMask(channelConfig)
+                            .build())
+        .setBufferSizeInBytes(bufferSizeInBytes)
+        .setPerformanceMode(AudioTrack.PERFORMANCE_MODE_LOW_LATENCY)
+        .setTransferMode(AudioTrack.MODE_STREAM)
+        .setSessionId(AudioManager.AUDIO_SESSION_ID_GENERATE)
+        .build();
+  }
+
+  @TargetApi(Build.VERSION_CODES.Q)
+  private static AudioAttributes.Builder applyAttributesOnQOrHigher(
+      AudioAttributes.Builder builder, AudioAttributes overrideAttributes) {
+    return builder.setAllowedCapturePolicy(overrideAttributes.getAllowedCapturePolicy());
+  }
+
+  private void logBufferSizeInFrames() {
+    if (Build.VERSION.SDK_INT >= Build.VERSION_CODES.M) {
+      Logging.d(TAG,
+          "AudioTrack: "
+              // The effective size of the AudioTrack buffer that the app writes to.
+              + "buffer size in frames: " + audioTrack.getBufferSizeInFrames());
+    }
+  }
+
+  @CalledByNative
+  private int getBufferSizeInFrames() {
+    if (Build.VERSION.SDK_INT >= Build.VERSION_CODES.M) {
+      return audioTrack.getBufferSizeInFrames();
+    }
+    return -1;
+  }
+
+  @CalledByNative
+  private int getInitialBufferSizeInFrames() {
+    return initialBufferSizeInFrames;
+  }
+
+  private void logBufferCapacityInFrames() {
+    if (Build.VERSION.SDK_INT >= Build.VERSION_CODES.N) {
+      Logging.d(TAG,
+          "AudioTrack: "
+              // Maximum size of the AudioTrack buffer in frames.
+              + "buffer capacity in frames: " + audioTrack.getBufferCapacityInFrames());
+    }
+  }
+
+  private void logMainParametersExtended() {
+    logBufferSizeInFrames();
+    logBufferCapacityInFrames();
+  }
+
+  // Prints the number of underrun occurrences in the application-level write
+  // buffer since the AudioTrack was created. An underrun occurs if the app does
+  // not write audio data quickly enough, causing the buffer to underflow and a
+  // potential audio glitch.
+  // TODO(henrika): keep track of this value in the field and possibly add new
+  // UMA stat if needed.
+  private void logUnderrunCount() {
+    if (Build.VERSION.SDK_INT >= Build.VERSION_CODES.N) {
+      Logging.d(TAG, "underrun count: " + audioTrack.getUnderrunCount());
+    }
+  }
+
+  // Helper method which throws an exception  when an assertion has failed.
+  private static void assertTrue(boolean condition) {
+    if (!condition) {
+      throw new AssertionError("Expected condition to be true");
+    }
+  }
+
+  private int channelCountToConfiguration(int channels) {
+    return (channels == 1 ? AudioFormat.CHANNEL_OUT_MONO : AudioFormat.CHANNEL_OUT_STEREO);
+  }
+
+  private static native void nativeCacheDirectBufferAddress(
+      long nativeAudioTrackJni, ByteBuffer byteBuffer);
+  private static native void nativeGetPlayoutData(long nativeAudioTrackJni, int bytes);
+
+  // Sets all samples to be played out to zero if `mute` is true, i.e.,
+  // ensures that the speaker is muted.
+  public void setSpeakerMute(boolean mute) {
+    Logging.w(TAG, "setSpeakerMute(" + mute + ")");
+    speakerMute = mute;
+  }
+
+  // Releases the native AudioTrack resources.
+  private void releaseAudioResources() {
+    Logging.d(TAG, "releaseAudioResources");
+    if (audioTrack != null) {
+      audioTrack.release();
+      audioTrack = null;
+    }
+  }
+
+  private void reportWebRtcAudioTrackInitError(String errorMessage) {
+    Logging.e(TAG, "Init playout error: " + errorMessage);
+    WebRtcAudioUtils.logAudioState(TAG, context, audioManager);
+    if (errorCallback != null) {
+      errorCallback.onWebRtcAudioTrackInitError(errorMessage);
+    }
+  }
+
+  private void reportWebRtcAudioTrackStartError(
+      AudioTrackStartErrorCode errorCode, String errorMessage) {
+    Logging.e(TAG, "Start playout error: " + errorCode + ". " + errorMessage);
+    WebRtcAudioUtils.logAudioState(TAG, context, audioManager);
+    if (errorCallback != null) {
+      errorCallback.onWebRtcAudioTrackStartError(errorCode, errorMessage);
+    }
+  }
+
+  private void reportWebRtcAudioTrackError(String errorMessage) {
+    Logging.e(TAG, "Run-time playback error: " + errorMessage);
+    WebRtcAudioUtils.logAudioState(TAG, context, audioManager);
+    if (errorCallback != null) {
+      errorCallback.onWebRtcAudioTrackError(errorMessage);
+    }
+  }
+
+  private void doAudioTrackStateCallback(int audioState) {
+    Logging.d(TAG, "doAudioTrackStateCallback: " + audioState);
+    if (stateCallback != null) {
+      if (audioState == WebRtcAudioTrack.AUDIO_TRACK_START) {
+        stateCallback.onWebRtcAudioTrackStart();
+      } else if (audioState == WebRtcAudioTrack.AUDIO_TRACK_STOP) {
+        stateCallback.onWebRtcAudioTrackStop();
+      } else {
+        Logging.e(TAG, "Invalid audio state");
+      }
+    }
+  }
+}
diff --git a/third_party/libwebrtc/sdk/android/src/java/org/webrtc/audio/WebRtcAudioUtils.java b/third_party/libwebrtc/sdk/android/src/java/org/webrtc/audio/WebRtcAudioUtils.java
new file mode 100644
index 0000000000..7b4b809ab1
--- /dev/null
+++ b/third_party/libwebrtc/sdk/android/src/java/org/webrtc/audio/WebRtcAudioUtils.java
@@ -0,0 +1,308 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+package org.webrtc.audio;
+
+import static android.media.AudioManager.MODE_IN_CALL;
+import static android.media.AudioManager.MODE_IN_COMMUNICATION;
+import static android.media.AudioManager.MODE_NORMAL;
+import static android.media.AudioManager.MODE_RINGTONE;
+
+import android.annotation.SuppressLint;
+import android.annotation.TargetApi;
+import android.content.Context;
+import android.content.pm.PackageManager;
+import android.media.AudioDeviceInfo;
+import android.media.AudioFormat;
+import android.media.AudioManager;
+import android.media.MediaRecorder.AudioSource;
+import android.os.Build;
+import java.lang.Thread;
+import java.util.Arrays;
+import org.webrtc.Logging;
+
+final class WebRtcAudioUtils {
+  private static final String TAG = "WebRtcAudioUtilsExternal";
+
+  // Helper method for building a string of thread information.
+  public static String getThreadInfo() {
+    return "@[name=" + Thread.currentThread().getName() + ", id=" + Thread.currentThread().getId()
+        + "]";
+  }
+
+  // Returns true if we're running on emulator.
+  public static boolean runningOnEmulator() {
+    return Build.HARDWARE.equals("goldfish") && Build.BRAND.startsWith("generic_");
+  }
+
+  // Information about the current build, taken from system properties.
+  static void logDeviceInfo(String tag) {
+    Logging.d(tag,
+        "Android SDK: " + Build.VERSION.SDK_INT + ", "
+            + "Release: " + Build.VERSION.RELEASE + ", "
+            + "Brand: " + Build.BRAND + ", "
+            + "Device: " + Build.DEVICE + ", "
+            + "Id: " + Build.ID + ", "
+            + "Hardware: " + Build.HARDWARE + ", "
+            + "Manufacturer: " + Build.MANUFACTURER + ", "
+            + "Model: " + Build.MODEL + ", "
+            + "Product: " + Build.PRODUCT);
+  }
+
+  // Logs information about the current audio state. The idea is to call this
+  // method when errors are detected to log under what conditions the error
+  // occurred. Hopefully it will provide clues to what might be the root cause.
+  static void logAudioState(String tag, Context context, AudioManager audioManager) {
+    logDeviceInfo(tag);
+    logAudioStateBasic(tag, context, audioManager);
+    logAudioStateVolume(tag, audioManager);
+    logAudioDeviceInfo(tag, audioManager);
+  }
+
+  // Converts AudioDeviceInfo types to local string representation.
+  static String deviceTypeToString(int type) {
+    switch (type) {
+      case AudioDeviceInfo.TYPE_UNKNOWN:
+        return "TYPE_UNKNOWN";
+      case AudioDeviceInfo.TYPE_BUILTIN_EARPIECE:
+        return "TYPE_BUILTIN_EARPIECE";
+      case AudioDeviceInfo.TYPE_BUILTIN_SPEAKER:
+        return "TYPE_BUILTIN_SPEAKER";
+      case AudioDeviceInfo.TYPE_WIRED_HEADSET:
+        return "TYPE_WIRED_HEADSET";
+      case AudioDeviceInfo.TYPE_WIRED_HEADPHONES:
+        return "TYPE_WIRED_HEADPHONES";
+      case AudioDeviceInfo.TYPE_LINE_ANALOG:
+        return "TYPE_LINE_ANALOG";
+      case AudioDeviceInfo.TYPE_LINE_DIGITAL:
+        return "TYPE_LINE_DIGITAL";
+      case AudioDeviceInfo.TYPE_BLUETOOTH_SCO:
+        return "TYPE_BLUETOOTH_SCO";
+      case AudioDeviceInfo.TYPE_BLUETOOTH_A2DP:
+        return "TYPE_BLUETOOTH_A2DP";
+      case AudioDeviceInfo.TYPE_HDMI:
+        return "TYPE_HDMI";
+      case AudioDeviceInfo.TYPE_HDMI_ARC:
+        return "TYPE_HDMI_ARC";
+      case AudioDeviceInfo.TYPE_USB_DEVICE:
+        return "TYPE_USB_DEVICE";
+      case AudioDeviceInfo.TYPE_USB_ACCESSORY:
+        return "TYPE_USB_ACCESSORY";
+      case AudioDeviceInfo.TYPE_DOCK:
+        return "TYPE_DOCK";
+      case AudioDeviceInfo.TYPE_FM:
+        return "TYPE_FM";
+      case AudioDeviceInfo.TYPE_BUILTIN_MIC:
+        return "TYPE_BUILTIN_MIC";
+      case AudioDeviceInfo.TYPE_FM_TUNER:
+        return "TYPE_FM_TUNER";
+      case AudioDeviceInfo.TYPE_TV_TUNER:
+        return "TYPE_TV_TUNER";
+      case AudioDeviceInfo.TYPE_TELEPHONY:
+        return "TYPE_TELEPHONY";
+      case AudioDeviceInfo.TYPE_AUX_LINE:
+        return "TYPE_AUX_LINE";
+      case AudioDeviceInfo.TYPE_IP:
+        return "TYPE_IP";
+      case AudioDeviceInfo.TYPE_BUS:
+        return "TYPE_BUS";
+      case AudioDeviceInfo.TYPE_USB_HEADSET:
+        return "TYPE_USB_HEADSET";
+      default:
+        return "TYPE_UNKNOWN";
+    }
+  }
+
+  @TargetApi(Build.VERSION_CODES.N)
+  public static String audioSourceToString(int source) {
+    // AudioSource.UNPROCESSED requires API level 29. Use local define instead.
+    final int VOICE_PERFORMANCE = 10;
+    switch (source) {
+      case AudioSource.DEFAULT:
+        return "DEFAULT";
+      case AudioSource.MIC:
+        return "MIC";
+      case AudioSource.VOICE_UPLINK:
+        return "VOICE_UPLINK";
+      case AudioSource.VOICE_DOWNLINK:
+        return "VOICE_DOWNLINK";
+      case AudioSource.VOICE_CALL:
+        return "VOICE_CALL";
+      case AudioSource.CAMCORDER:
+        return "CAMCORDER";
+      case AudioSource.VOICE_RECOGNITION:
+        return "VOICE_RECOGNITION";
+      case AudioSource.VOICE_COMMUNICATION:
+        return "VOICE_COMMUNICATION";
+      case AudioSource.UNPROCESSED:
+        return "UNPROCESSED";
+      case VOICE_PERFORMANCE:
+        return "VOICE_PERFORMANCE";
+      default:
+        return "INVALID";
+    }
+  }
+
+  public static String channelMaskToString(int mask) {
+    // For input or AudioRecord, the mask should be AudioFormat#CHANNEL_IN_MONO or
+    // AudioFormat#CHANNEL_IN_STEREO. AudioFormat#CHANNEL_IN_MONO is guaranteed to work on all
+    // devices.
+    switch (mask) {
+      case AudioFormat.CHANNEL_IN_STEREO:
+        return "IN_STEREO";
+      case AudioFormat.CHANNEL_IN_MONO:
+        return "IN_MONO";
+      default:
+        return "INVALID";
+    }
+  }
+
+  @TargetApi(Build.VERSION_CODES.N)
+  public static String audioEncodingToString(int enc) {
+    switch (enc) {
+      case AudioFormat.ENCODING_INVALID:
+        return "INVALID";
+      case AudioFormat.ENCODING_PCM_16BIT:
+        return "PCM_16BIT";
+      case AudioFormat.ENCODING_PCM_8BIT:
+        return "PCM_8BIT";
+      case AudioFormat.ENCODING_PCM_FLOAT:
+        return "PCM_FLOAT";
+      case AudioFormat.ENCODING_AC3:
+        return "AC3";
+      case AudioFormat.ENCODING_E_AC3:
+        return "AC3";
+      case AudioFormat.ENCODING_DTS:
+        return "DTS";
+      case AudioFormat.ENCODING_DTS_HD:
+        return "DTS_HD";
+      case AudioFormat.ENCODING_MP3:
+        return "MP3";
+      default:
+        return "Invalid encoding: " + enc;
+    }
+  }
+
+  // Reports basic audio statistics.
+  private static void logAudioStateBasic(String tag, Context context, AudioManager audioManager) {
+    Logging.d(tag,
+        "Audio State: "
+            + "audio mode: " + modeToString(audioManager.getMode()) + ", "
+            + "has mic: " + hasMicrophone(context) + ", "
+            + "mic muted: " + audioManager.isMicrophoneMute() + ", "
+            + "music active: " + audioManager.isMusicActive() + ", "
+            + "speakerphone: " + audioManager.isSpeakerphoneOn() + ", "
+            + "BT SCO: " + audioManager.isBluetoothScoOn());
+  }
+
+  // Adds volume information for all possible stream types.
+  private static void logAudioStateVolume(String tag, AudioManager audioManager) {
+    final int[] streams = {AudioManager.STREAM_VOICE_CALL, AudioManager.STREAM_MUSIC,
+        AudioManager.STREAM_RING, AudioManager.STREAM_ALARM, AudioManager.STREAM_NOTIFICATION,
+        AudioManager.STREAM_SYSTEM};
+    Logging.d(tag, "Audio State: ");
+    // Some devices may not have volume controls and might use a fixed volume.
+    boolean fixedVolume = audioManager.isVolumeFixed();
+    Logging.d(tag, "  fixed volume=" + fixedVolume);
+    if (!fixedVolume) {
+      for (int stream : streams) {
+        StringBuilder info = new StringBuilder();
+        info.append("  " + streamTypeToString(stream) + ": ");
+        info.append("volume=").append(audioManager.getStreamVolume(stream));
+        info.append(", max=").append(audioManager.getStreamMaxVolume(stream));
+        logIsStreamMute(tag, audioManager, stream, info);
+        Logging.d(tag, info.toString());
+      }
+    }
+  }
+
+  private static void logIsStreamMute(
+      String tag, AudioManager audioManager, int stream, StringBuilder info) {
+    if (Build.VERSION.SDK_INT >= 23) {
+      info.append(", muted=").append(audioManager.isStreamMute(stream));
+    }
+  }
+
+  // Moz linting complains even though AudioManager.GET_DEVICES_ALL is
+  // listed in the docs here:
+  // https://developer.android.com/reference/android/media/AudioManager#GET_DEVICES_ALL
+  @SuppressLint("WrongConstant")
+  private static void logAudioDeviceInfo(String tag, AudioManager audioManager) {
+    if (Build.VERSION.SDK_INT < 23) {
+      return;
+    }
+    final AudioDeviceInfo[] devices = audioManager.getDevices(AudioManager.GET_DEVICES_ALL);
+    if (devices.length == 0) {
+      return;
+    }
+    Logging.d(tag, "Audio Devices: ");
+    for (AudioDeviceInfo device : devices) {
+      StringBuilder info = new StringBuilder();
+      info.append("  ").append(deviceTypeToString(device.getType()));
+      info.append(device.isSource() ? "(in): " : "(out): ");
+      // An empty array indicates that the device supports arbitrary channel counts.
+      if (device.getChannelCounts().length > 0) {
+        info.append("channels=").append(Arrays.toString(device.getChannelCounts()));
+        info.append(", ");
+      }
+      if (device.getEncodings().length > 0) {
+        // Examples: ENCODING_PCM_16BIT = 2, ENCODING_PCM_FLOAT = 4.
+        info.append("encodings=").append(Arrays.toString(device.getEncodings()));
+        info.append(", ");
+      }
+      if (device.getSampleRates().length > 0) {
+        info.append("sample rates=").append(Arrays.toString(device.getSampleRates()));
+        info.append(", ");
+      }
+      info.append("id=").append(device.getId());
+      Logging.d(tag, info.toString());
+    }
+  }
+
+  // Converts media.AudioManager modes into local string representation.
+  static String modeToString(int mode) {
+    switch (mode) {
+      case MODE_IN_CALL:
+        return "MODE_IN_CALL";
+      case MODE_IN_COMMUNICATION:
+        return "MODE_IN_COMMUNICATION";
+      case MODE_NORMAL:
+        return "MODE_NORMAL";
+      case MODE_RINGTONE:
+        return "MODE_RINGTONE";
+      default:
+        return "MODE_INVALID";
+    }
+  }
+
+  private static String streamTypeToString(int stream) {
+    switch (stream) {
+      case AudioManager.STREAM_VOICE_CALL:
+        return "STREAM_VOICE_CALL";
+      case AudioManager.STREAM_MUSIC:
+        return "STREAM_MUSIC";
+      case AudioManager.STREAM_RING:
+        return "STREAM_RING";
+      case AudioManager.STREAM_ALARM:
+        return "STREAM_ALARM";
+      case AudioManager.STREAM_NOTIFICATION:
+        return "STREAM_NOTIFICATION";
+      case AudioManager.STREAM_SYSTEM:
+        return "STREAM_SYSTEM";
+      default:
+        return "STREAM_INVALID";
+    }
+  }
+
+  // Returns true if the device can record audio via a microphone.
+  private static boolean hasMicrophone(Context context) {
+    return context.getPackageManager().hasSystemFeature(PackageManager.FEATURE_MICROPHONE);
+  }
+}
diff --git a/third_party/libwebrtc/sdk/base_objc_gn/moz.build b/third_party/libwebrtc/sdk/base_objc_gn/moz.build
new file mode 100644
index 0000000000..976c7fceb7
--- /dev/null
+++ b/third_party/libwebrtc/sdk/base_objc_gn/moz.build
@@ -0,0 +1,79 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+CMFLAGS += [
+    "-fobjc-arc"
+]
+
+CMMFLAGS += [
+    "-fobjc-arc"
+]
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MAC"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_POSIX"] = True
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+DEFINES["__STDC_CONSTANT_MACROS"] = True
+DEFINES["__STDC_FORMAT_MACROS"] = True
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/sdk/objc/",
+    "/third_party/libwebrtc/sdk/objc/base/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/sdk/objc/base/RTCEncodedImage.m",
+    "/third_party/libwebrtc/sdk/objc/base/RTCLogging.mm",
+    "/third_party/libwebrtc/sdk/objc/base/RTCVideoCapturer.m",
+    "/third_party/libwebrtc/sdk/objc/base/RTCVideoCodecInfo.m",
+    "/third_party/libwebrtc/sdk/objc/base/RTCVideoEncoderQpThresholds.m",
+    "/third_party/libwebrtc/sdk/objc/base/RTCVideoEncoderSettings.m",
+    "/third_party/libwebrtc/sdk/objc/base/RTCVideoFrame.mm"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+Library("base_objc_gn")
diff --git a/third_party/libwebrtc/sdk/helpers_objc_gn/moz.build b/third_party/libwebrtc/sdk/helpers_objc_gn/moz.build
new file mode 100644
index 0000000000..7634e1aa7b
--- /dev/null
+++ b/third_party/libwebrtc/sdk/helpers_objc_gn/moz.build
@@ -0,0 +1,75 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+CMFLAGS += [
+    "-fobjc-arc"
+]
+
+CMMFLAGS += [
+    "-fobjc-arc"
+]
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MAC"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_POSIX"] = True
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+DEFINES["__STDC_CONSTANT_MACROS"] = True
+DEFINES["__STDC_FORMAT_MACROS"] = True
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/sdk/objc/",
+    "/third_party/libwebrtc/sdk/objc/base/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/sdk/objc/helpers/AVCaptureSession+DevicePosition.mm",
+    "/third_party/libwebrtc/sdk/objc/helpers/NSString+StdString.mm",
+    "/third_party/libwebrtc/sdk/objc/helpers/RTCDispatcher.m"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+Library("helpers_objc_gn")
diff --git a/third_party/libwebrtc/sdk/media_constraints.cc b/third_party/libwebrtc/sdk/media_constraints.cc
new file mode 100644
index 0000000000..bbb46edaae
--- /dev/null
+++ b/third_party/libwebrtc/sdk/media_constraints.cc
@@ -0,0 +1,253 @@
+/*
+ *  Copyright 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "sdk/media_constraints.h"
+
+#include "absl/types/optional.h"
+#include "api/peer_connection_interface.h"
+
+namespace webrtc {
+namespace {
+
+// Find the highest-priority instance of the T-valued constraint named by
+// `key` and return its value as `value`. `constraints` can be null.
+// If `mandatory_constraints` is non-null, it is incremented if the key appears
+// among the mandatory constraints.
+// Returns true if the key was found and has a valid value for type T.
+// If the key appears multiple times as an optional constraint, appearances
+// after the first are ignored.
+// Note: Because this uses FindFirst, repeated optional constraints whose
+// first instance has an unrecognized value are not handled precisely in
+// accordance with the specification.
+template <typename T>
+bool FindConstraint(const MediaConstraints* constraints,
+                    const std::string& key,
+                    T* value,
+                    size_t* mandatory_constraints) {
+  std::string string_value;
+  if (!FindConstraint(constraints, key, &string_value, mandatory_constraints)) {
+    return false;
+  }
+  return rtc::FromString(string_value, value);
+}
+
+// Specialization for std::string, since a string doesn't need conversion.
+template <>
+bool FindConstraint(const MediaConstraints* constraints,
+                    const std::string& key,
+                    std::string* value,
+                    size_t* mandatory_constraints) {
+  if (!constraints) {
+    return false;
+  }
+  if (constraints->GetMandatory().FindFirst(key, value)) {
+    if (mandatory_constraints) {
+      ++*mandatory_constraints;
+    }
+    return true;
+  }
+  if (constraints->GetOptional().FindFirst(key, value)) {
+    return true;
+  }
+  return false;
+}
+
+bool FindConstraint(const MediaConstraints* constraints,
+                    const std::string& key,
+                    bool* value,
+                    size_t* mandatory_constraints) {
+  return FindConstraint<bool>(constraints, key, value, mandatory_constraints);
+}
+
+bool FindConstraint(const MediaConstraints* constraints,
+                    const std::string& key,
+                    int* value,
+                    size_t* mandatory_constraints) {
+  return FindConstraint<int>(constraints, key, value, mandatory_constraints);
+}
+
+// Converts a constraint (mandatory takes precedence over optional) to an
+// absl::optional.
+template <typename T>
+void ConstraintToOptional(const MediaConstraints* constraints,
+                          const std::string& key,
+                          absl::optional<T>* value_out) {
+  T value;
+  bool present = FindConstraint<T>(constraints, key, &value, nullptr);
+  if (present) {
+    *value_out = value;
+  }
+}
+}  // namespace
+
+const char MediaConstraints::kValueTrue[] = "true";
+const char MediaConstraints::kValueFalse[] = "false";
+
+// Constraints declared as static members in mediastreaminterface.h
+
+// Audio constraints.
+const char MediaConstraints::kGoogEchoCancellation[] = "googEchoCancellation";
+const char MediaConstraints::kAutoGainControl[] = "googAutoGainControl";
+const char MediaConstraints::kNoiseSuppression[] = "googNoiseSuppression";
+const char MediaConstraints::kHighpassFilter[] = "googHighpassFilter";
+const char MediaConstraints::kAudioMirroring[] = "googAudioMirroring";
+const char MediaConstraints::kAudioNetworkAdaptorConfig[] =
+    "googAudioNetworkAdaptorConfig";
+const char MediaConstraints::kInitAudioRecordingOnSend[] =
+    "InitAudioRecordingOnSend";
+
+// Constraint keys for CreateOffer / CreateAnswer defined in W3C specification.
+const char MediaConstraints::kOfferToReceiveAudio[] = "OfferToReceiveAudio";
+const char MediaConstraints::kOfferToReceiveVideo[] = "OfferToReceiveVideo";
+const char MediaConstraints::kVoiceActivityDetection[] =
+    "VoiceActivityDetection";
+const char MediaConstraints::kIceRestart[] = "IceRestart";
+// Google specific constraint for BUNDLE enable/disable.
+const char MediaConstraints::kUseRtpMux[] = "googUseRtpMUX";
+
+// Below constraints should be used during PeerConnection construction.
+// Google-specific constraint keys.
+const char MediaConstraints::kEnableDscp[] = "googDscp";
+const char MediaConstraints::kEnableVideoSuspendBelowMinBitrate[] =
+    "googSuspendBelowMinBitrate";
+const char MediaConstraints::kCombinedAudioVideoBwe[] =
+    "googCombinedAudioVideoBwe";
+const char MediaConstraints::kScreencastMinBitrate[] =
+    "googScreencastMinBitrate";
+// TODO(ronghuawu): Remove once cpu overuse detection is stable.
+const char MediaConstraints::kCpuOveruseDetection[] = "googCpuOveruseDetection";
+
+const char MediaConstraints::kRawPacketizationForVideoEnabled[] =
+    "googRawPacketizationForVideoEnabled";
+
+const char MediaConstraints::kNumSimulcastLayers[] = "googNumSimulcastLayers";
+
+// Set `value` to the value associated with the first appearance of `key`, or
+// return false if `key` is not found.
+bool MediaConstraints::Constraints::FindFirst(const std::string& key,
+                                              std::string* value) const {
+  for (Constraints::const_iterator iter = begin(); iter != end(); ++iter) {
+    if (iter->key == key) {
+      *value = iter->value;
+      return true;
+    }
+  }
+  return false;
+}
+
+void CopyConstraintsIntoRtcConfiguration(
+    const MediaConstraints* constraints,
+    PeerConnectionInterface::RTCConfiguration* configuration) {
+  // Copy info from constraints into configuration, if present.
+  if (!constraints) {
+    return;
+  }
+
+  FindConstraint(constraints, MediaConstraints::kEnableDscp,
+                 &configuration->media_config.enable_dscp, nullptr);
+  FindConstraint(constraints, MediaConstraints::kCpuOveruseDetection,
+                 &configuration->media_config.video.enable_cpu_adaptation,
+                 nullptr);
+  // Find Suspend Below Min Bitrate constraint.
+  FindConstraint(
+      constraints, MediaConstraints::kEnableVideoSuspendBelowMinBitrate,
+      &configuration->media_config.video.suspend_below_min_bitrate, nullptr);
+  ConstraintToOptional<int>(constraints,
+                            MediaConstraints::kScreencastMinBitrate,
+                            &configuration->screencast_min_bitrate);
+  ConstraintToOptional<bool>(constraints,
+                             MediaConstraints::kCombinedAudioVideoBwe,
+                             &configuration->combined_audio_video_bwe);
+}
+
+void CopyConstraintsIntoAudioOptions(const MediaConstraints* constraints,
+                                     cricket::AudioOptions* options) {
+  if (!constraints) {
+    return;
+  }
+
+  ConstraintToOptional<bool>(constraints,
+                             MediaConstraints::kGoogEchoCancellation,
+                             &options->echo_cancellation);
+  ConstraintToOptional<bool>(constraints, MediaConstraints::kAutoGainControl,
+                             &options->auto_gain_control);
+  ConstraintToOptional<bool>(constraints, MediaConstraints::kNoiseSuppression,
+                             &options->noise_suppression);
+  ConstraintToOptional<bool>(constraints, MediaConstraints::kHighpassFilter,
+                             &options->highpass_filter);
+  ConstraintToOptional<bool>(constraints, MediaConstraints::kAudioMirroring,
+                             &options->stereo_swapping);
+  ConstraintToOptional<std::string>(
+      constraints, MediaConstraints::kAudioNetworkAdaptorConfig,
+      &options->audio_network_adaptor_config);
+  // When `kAudioNetworkAdaptorConfig` is defined, it both means that audio
+  // network adaptor is desired, and provides the config string.
+  if (options->audio_network_adaptor_config) {
+    options->audio_network_adaptor = true;
+  }
+  ConstraintToOptional<bool>(constraints,
+                             MediaConstraints::kInitAudioRecordingOnSend,
+                             &options->init_recording_on_send);
+}
+
+bool CopyConstraintsIntoOfferAnswerOptions(
+    const MediaConstraints* constraints,
+    PeerConnectionInterface::RTCOfferAnswerOptions* offer_answer_options) {
+  if (!constraints) {
+    return true;
+  }
+
+  bool value = false;
+  size_t mandatory_constraints_satisfied = 0;
+
+  if (FindConstraint(constraints, MediaConstraints::kOfferToReceiveAudio,
+                     &value, &mandatory_constraints_satisfied)) {
+    offer_answer_options->offer_to_receive_audio =
+        value ? PeerConnectionInterface::RTCOfferAnswerOptions::
+                    kOfferToReceiveMediaTrue
+              : 0;
+  }
+
+  if (FindConstraint(constraints, MediaConstraints::kOfferToReceiveVideo,
+                     &value, &mandatory_constraints_satisfied)) {
+    offer_answer_options->offer_to_receive_video =
+        value ? PeerConnectionInterface::RTCOfferAnswerOptions::
+                    kOfferToReceiveMediaTrue
+              : 0;
+  }
+  if (FindConstraint(constraints, MediaConstraints::kVoiceActivityDetection,
+                     &value, &mandatory_constraints_satisfied)) {
+    offer_answer_options->voice_activity_detection = value;
+  }
+  if (FindConstraint(constraints, MediaConstraints::kUseRtpMux, &value,
+                     &mandatory_constraints_satisfied)) {
+    offer_answer_options->use_rtp_mux = value;
+  }
+  if (FindConstraint(constraints, MediaConstraints::kIceRestart, &value,
+                     &mandatory_constraints_satisfied)) {
+    offer_answer_options->ice_restart = value;
+  }
+
+  if (FindConstraint(constraints,
+                     MediaConstraints::kRawPacketizationForVideoEnabled, &value,
+                     &mandatory_constraints_satisfied)) {
+    offer_answer_options->raw_packetization_for_video = value;
+  }
+
+  int layers;
+  if (FindConstraint(constraints, MediaConstraints::kNumSimulcastLayers,
+                     &layers, &mandatory_constraints_satisfied)) {
+    offer_answer_options->num_simulcast_layers = layers;
+  }
+
+  return mandatory_constraints_satisfied == constraints->GetMandatory().size();
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/sdk/media_constraints.h b/third_party/libwebrtc/sdk/media_constraints.h
new file mode 100644
index 0000000000..c946e4fab1
--- /dev/null
+++ b/third_party/libwebrtc/sdk/media_constraints.h
@@ -0,0 +1,131 @@
+/*
+ *  Copyright 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+// Implementation of the w3c constraints spec is the responsibility of the
+// browser. Chrome no longer uses the constraints api declared here, and it will
+// be removed from WebRTC.
+// https://bugs.chromium.org/p/webrtc/issues/detail?id=9239
+
+#ifndef SDK_MEDIA_CONSTRAINTS_H_
+#define SDK_MEDIA_CONSTRAINTS_H_
+
+#include <stddef.h>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "api/audio_options.h"
+#include "api/peer_connection_interface.h"
+
+namespace webrtc {
+
+// Class representing constraints, as used by the android and objc apis.
+//
+// Constraints may be either "mandatory", which means that unless satisfied,
+// the method taking the constraints should fail, or "optional", which means
+// they may not be satisfied..
+class MediaConstraints {
+ public:
+  struct Constraint {
+    Constraint() {}
+    Constraint(const std::string& key, const std::string value)
+        : key(key), value(value) {}
+    std::string key;
+    std::string value;
+  };
+
+  class Constraints : public std::vector<Constraint> {
+   public:
+    Constraints() = default;
+    Constraints(std::initializer_list<Constraint> l)
+        : std::vector<Constraint>(l) {}
+
+    bool FindFirst(const std::string& key, std::string* value) const;
+  };
+
+  MediaConstraints() = default;
+  MediaConstraints(Constraints mandatory, Constraints optional)
+      : mandatory_(std::move(mandatory)), optional_(std::move(optional)) {}
+
+  // Constraint keys used by a local audio source.
+
+  // These keys are google specific.
+  static const char kGoogEchoCancellation[];  // googEchoCancellation
+
+  static const char kAutoGainControl[];       // googAutoGainControl
+  static const char kNoiseSuppression[];      // googNoiseSuppression
+  static const char kHighpassFilter[];        // googHighpassFilter
+  static const char kAudioMirroring[];        // googAudioMirroring
+  static const char
+      kAudioNetworkAdaptorConfig[];  // googAudioNetworkAdaptorConfig
+  static const char kInitAudioRecordingOnSend[];  // InitAudioRecordingOnSend;
+
+  // Constraint keys for CreateOffer / CreateAnswer
+  // Specified by the W3C PeerConnection spec
+  static const char kOfferToReceiveVideo[];     // OfferToReceiveVideo
+  static const char kOfferToReceiveAudio[];     // OfferToReceiveAudio
+  static const char kVoiceActivityDetection[];  // VoiceActivityDetection
+  static const char kIceRestart[];              // IceRestart
+  // These keys are google specific.
+  static const char kUseRtpMux[];  // googUseRtpMUX
+
+  // Constraints values.
+  static const char kValueTrue[];   // true
+  static const char kValueFalse[];  // false
+
+  // PeerConnection constraint keys.
+  // Google-specific constraint keys.
+  // Temporary pseudo-constraint for enabling DSCP through JS.
+  static const char kEnableDscp[];  // googDscp
+  // Constraint to enable IPv6 through JS.
+  static const char kEnableIPv6[];  // googIPv6
+  // Temporary constraint to enable suspend below min bitrate feature.
+  static const char kEnableVideoSuspendBelowMinBitrate[];
+  // googSuspendBelowMinBitrate
+  // Constraint to enable combined audio+video bandwidth estimation.
+  static const char kCombinedAudioVideoBwe[];  // googCombinedAudioVideoBwe
+  static const char kScreencastMinBitrate[];   // googScreencastMinBitrate
+  static const char kCpuOveruseDetection[];    // googCpuOveruseDetection
+
+  // Constraint to enable negotiating raw RTP packetization using attribute
+  // "a=packetization:<payload_type> raw" in the SDP for all video payload.
+  static const char kRawPacketizationForVideoEnabled[];
+
+  // Specifies number of simulcast layers for all video tracks
+  // with a Plan B offer/answer
+  // (see RTCOfferAnswerOptions::num_simulcast_layers).
+  static const char kNumSimulcastLayers[];
+
+  ~MediaConstraints() = default;
+
+  const Constraints& GetMandatory() const { return mandatory_; }
+  const Constraints& GetOptional() const { return optional_; }
+
+ private:
+  const Constraints mandatory_ = {};
+  const Constraints optional_ = {};
+};
+
+// Copy all relevant constraints into an RTCConfiguration object.
+void CopyConstraintsIntoRtcConfiguration(
+    const MediaConstraints* constraints,
+    PeerConnectionInterface::RTCConfiguration* configuration);
+
+// Copy all relevant constraints into an AudioOptions object.
+void CopyConstraintsIntoAudioOptions(const MediaConstraints* constraints,
+                                     cricket::AudioOptions* options);
+
+bool CopyConstraintsIntoOfferAnswerOptions(
+    const MediaConstraints* constraints,
+    PeerConnectionInterface::RTCOfferAnswerOptions* offer_answer_options);
+
+}  // namespace webrtc
+
+#endif  // SDK_MEDIA_CONSTRAINTS_H_
diff --git a/third_party/libwebrtc/sdk/media_constraints_unittest.cc b/third_party/libwebrtc/sdk/media_constraints_unittest.cc
new file mode 100644
index 0000000000..2d25da03e7
--- /dev/null
+++ b/third_party/libwebrtc/sdk/media_constraints_unittest.cc
@@ -0,0 +1,56 @@
+/*
+ *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "sdk/media_constraints.h"
+
+#include "test/gtest.h"
+
+namespace webrtc {
+
+namespace {
+
+// Checks all settings touched by CopyConstraintsIntoRtcConfiguration,
+// plus audio_jitter_buffer_max_packets.
+bool Matches(const PeerConnectionInterface::RTCConfiguration& a,
+             const PeerConnectionInterface::RTCConfiguration& b) {
+  return a.audio_jitter_buffer_max_packets ==
+             b.audio_jitter_buffer_max_packets &&
+         a.screencast_min_bitrate == b.screencast_min_bitrate &&
+         a.combined_audio_video_bwe == b.combined_audio_video_bwe &&
+         a.media_config == b.media_config;
+}
+
+TEST(MediaConstraints, CopyConstraintsIntoRtcConfiguration) {
+  const MediaConstraints constraints_empty;
+  PeerConnectionInterface::RTCConfiguration old_configuration;
+  PeerConnectionInterface::RTCConfiguration configuration;
+
+  CopyConstraintsIntoRtcConfiguration(&constraints_empty, &configuration);
+  EXPECT_TRUE(Matches(old_configuration, configuration));
+
+  const MediaConstraints constraints_screencast(
+      {MediaConstraints::Constraint(MediaConstraints::kScreencastMinBitrate,
+                                    "27")},
+      {});
+  CopyConstraintsIntoRtcConfiguration(&constraints_screencast, &configuration);
+  EXPECT_TRUE(configuration.screencast_min_bitrate);
+  EXPECT_EQ(27, *(configuration.screencast_min_bitrate));
+
+  // An empty set of constraints will not overwrite
+  // values that are already present.
+  configuration = old_configuration;
+  configuration.audio_jitter_buffer_max_packets = 34;
+  CopyConstraintsIntoRtcConfiguration(&constraints_empty, &configuration);
+  EXPECT_EQ(34, configuration.audio_jitter_buffer_max_packets);
+}
+
+}  // namespace
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/sdk/objc/DEPS b/third_party/libwebrtc/sdk/objc/DEPS
new file mode 100644
index 0000000000..4cff92caf1
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/DEPS
@@ -0,0 +1,18 @@
+include_rules = [
+  "+base",
+  "+components",
+  "+helpers",
+  "+sdk",
+  "+common_video/h264",
+  "+common_video/include",
+  "+common_video/libyuv/include",
+  "+logging/rtc_event_log/rtc_event_log_factory.h",
+  "+media",
+  "+modules/video_coding",
+  "+pc",
+  "+system_wrappers",
+  "+modules/audio_device",
+  "+modules/audio_processing",
+  "+native",
+  "+third_party/libyuv",
+]
diff --git a/third_party/libwebrtc/sdk/objc/Framework/Classes/Common/NSString+StdString.h b/third_party/libwebrtc/sdk/objc/Framework/Classes/Common/NSString+StdString.h
new file mode 100644
index 0000000000..3ec1b613ef
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/Framework/Classes/Common/NSString+StdString.h
@@ -0,0 +1,11 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "helpers/NSString+StdString.h"
diff --git a/third_party/libwebrtc/sdk/objc/Framework/Classes/Common/scoped_cftyperef.h b/third_party/libwebrtc/sdk/objc/Framework/Classes/Common/scoped_cftyperef.h
new file mode 100644
index 0000000000..e5e376b0bc
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/Framework/Classes/Common/scoped_cftyperef.h
@@ -0,0 +1,12 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ *
+ */
+
+#import "helpers/scoped_cftyperef.h"
diff --git a/third_party/libwebrtc/sdk/objc/Framework/Classes/PeerConnection/RTCConfiguration+Native.h b/third_party/libwebrtc/sdk/objc/Framework/Classes/PeerConnection/RTCConfiguration+Native.h
new file mode 100644
index 0000000000..529aa8dcf5
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/Framework/Classes/PeerConnection/RTCConfiguration+Native.h
@@ -0,0 +1,11 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "api/peerconnection/RTCConfiguration+Native.h"
diff --git a/third_party/libwebrtc/sdk/objc/Framework/Classes/PeerConnection/RTCPeerConnectionFactory+Native.h b/third_party/libwebrtc/sdk/objc/Framework/Classes/PeerConnection/RTCPeerConnectionFactory+Native.h
new file mode 100644
index 0000000000..222e06ef33
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/Framework/Classes/PeerConnection/RTCPeerConnectionFactory+Native.h
@@ -0,0 +1,11 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "api/peerconnection/RTCPeerConnectionFactory+Native.h"
diff --git a/third_party/libwebrtc/sdk/objc/Framework/Classes/Video/RTCDefaultShader.h b/third_party/libwebrtc/sdk/objc/Framework/Classes/Video/RTCDefaultShader.h
new file mode 100644
index 0000000000..136d7003c6
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/Framework/Classes/Video/RTCDefaultShader.h
@@ -0,0 +1,11 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "components/renderer/opengl/RTCDefaultShader.h"
diff --git a/third_party/libwebrtc/sdk/objc/Framework/Classes/Video/RTCNV12TextureCache.h b/third_party/libwebrtc/sdk/objc/Framework/Classes/Video/RTCNV12TextureCache.h
new file mode 100644
index 0000000000..4ba1caa41d
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/Framework/Classes/Video/RTCNV12TextureCache.h
@@ -0,0 +1,11 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "components/renderer/opengl/RTCNV12TextureCache.h"
diff --git a/third_party/libwebrtc/sdk/objc/Framework/Classes/VideoToolbox/nalu_rewriter.h b/third_party/libwebrtc/sdk/objc/Framework/Classes/VideoToolbox/nalu_rewriter.h
new file mode 100644
index 0000000000..21281f36ac
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/Framework/Classes/VideoToolbox/nalu_rewriter.h
@@ -0,0 +1,11 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "components/video_codec/nalu_rewriter.h"
diff --git a/third_party/libwebrtc/sdk/objc/Framework/Native/api/audio_device_module.h b/third_party/libwebrtc/sdk/objc/Framework/Native/api/audio_device_module.h
new file mode 100644
index 0000000000..7b448024de
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/Framework/Native/api/audio_device_module.h
@@ -0,0 +1,11 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "native/api/audio_device_module.h"
diff --git a/third_party/libwebrtc/sdk/objc/Framework/Native/api/video_decoder_factory.h b/third_party/libwebrtc/sdk/objc/Framework/Native/api/video_decoder_factory.h
new file mode 100644
index 0000000000..ca9371c54d
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/Framework/Native/api/video_decoder_factory.h
@@ -0,0 +1,11 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "native/api/video_decoder_factory.h"
diff --git a/third_party/libwebrtc/sdk/objc/Framework/Native/api/video_encoder_factory.h b/third_party/libwebrtc/sdk/objc/Framework/Native/api/video_encoder_factory.h
new file mode 100644
index 0000000000..35e1e6c99f
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/Framework/Native/api/video_encoder_factory.h
@@ -0,0 +1,11 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "native/api/video_encoder_factory.h"
diff --git a/third_party/libwebrtc/sdk/objc/Framework/Native/api/video_frame_buffer.h b/third_party/libwebrtc/sdk/objc/Framework/Native/api/video_frame_buffer.h
new file mode 100644
index 0000000000..0e862cfa07
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/Framework/Native/api/video_frame_buffer.h
@@ -0,0 +1,11 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "native/api/video_frame_buffer.h"
diff --git a/third_party/libwebrtc/sdk/objc/Framework/Native/src/objc_video_decoder_factory.h b/third_party/libwebrtc/sdk/objc/Framework/Native/src/objc_video_decoder_factory.h
new file mode 100644
index 0000000000..bd8513c342
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/Framework/Native/src/objc_video_decoder_factory.h
@@ -0,0 +1,11 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "native/src/objc_video_decoder_factory.h"
diff --git a/third_party/libwebrtc/sdk/objc/Framework/Native/src/objc_video_encoder_factory.h b/third_party/libwebrtc/sdk/objc/Framework/Native/src/objc_video_encoder_factory.h
new file mode 100644
index 0000000000..b6bd650a4f
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/Framework/Native/src/objc_video_encoder_factory.h
@@ -0,0 +1,11 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "native/src/objc_video_encoder_factory.h"
diff --git a/third_party/libwebrtc/sdk/objc/Info.plist b/third_party/libwebrtc/sdk/objc/Info.plist
new file mode 100644
index 0000000000..38c437e7fe
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/Info.plist
@@ -0,0 +1,26 @@
+<?xml version="1.0" encoding="UTF-8"?>
+<!DOCTYPE plist PUBLIC "-//Apple//DTD PLIST 1.0//EN" "http://www.apple.com/DTDs/PropertyList-1.0.dtd">
+<plist version="1.0">
+<dict>
+	<key>CFBundleDevelopmentRegion</key>
+	<string>en</string>
+	<key>CFBundleExecutable</key>
+	<string>WebRTC</string>
+	<key>CFBundleIdentifier</key>
+	<string>org.webrtc.WebRTC</string>
+	<key>CFBundleInfoDictionaryVersion</key>
+	<string>6.0</string>
+	<key>CFBundleName</key>
+	<string>WebRTC</string>
+	<key>CFBundlePackageType</key>
+	<string>FMWK</string>
+	<key>CFBundleShortVersionString</key>
+	<string>1.0</string>
+	<key>CFBundleSignature</key>
+	<string>????</string>
+	<key>CFBundleVersion</key>
+	<string>1.0</string>
+	<key>NSPrincipalClass</key>
+	<string></string>
+</dict>
+</plist>
diff --git a/third_party/libwebrtc/sdk/objc/OWNERS b/third_party/libwebrtc/sdk/objc/OWNERS
new file mode 100644
index 0000000000..6af9062b2d
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/OWNERS
@@ -0,0 +1,9 @@
+# Normal code changes.
+kthelgason@webrtc.org
+andersc@webrtc.org
+peterhanspers@webrtc.org
+denicija@webrtc.org
+
+# Rubberstamps of e.g. reverts and critical bug fixes.
+magjed@webrtc.org
+tkchin@webrtc.org
diff --git a/third_party/libwebrtc/sdk/objc/README.md b/third_party/libwebrtc/sdk/objc/README.md
new file mode 100644
index 0000000000..ff294a266f
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/README.md
@@ -0,0 +1,37 @@
+# WebRTC Obj-C SDK
+
+This directory contains the Obj-C SDK for WebRTC. This includes wrappers for the
+C++ PeerConnection API and some platform specific components for iOS and macOS.
+
+## Organization
+
+- api/
+
+  Wrappers around classes and functions in the C++ API for creating and
+  configuring peer connections, etc.
+
+- base/
+
+  This directory contains some base protocols and classes that are used by both
+  the platform specific components and the SDK wrappers.
+
+- components/
+
+  These are the platform specific components. Contains components for handling
+  audio, capturing and rendering video, encoding and decoding using the
+  platform's hardware codec implementation and for representing video frames
+  in the platform's native format.
+
+- helpers/
+
+  These files are not WebRTC specific, but are general helper classes and
+  utilities for the Cocoa platforms.
+
+- native/
+
+  APIs for wrapping the platform specific components and using them with the
+  C++ API.
+
+- unittests/
+
+  This directory contains the tests.
diff --git a/third_party/libwebrtc/sdk/objc/api/RTCVideoRendererAdapter+Private.h b/third_party/libwebrtc/sdk/objc/api/RTCVideoRendererAdapter+Private.h
new file mode 100644
index 0000000000..9b123d2d05
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/RTCVideoRendererAdapter+Private.h
@@ -0,0 +1,41 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCVideoRendererAdapter.h"
+
+#import "base/RTCVideoRenderer.h"
+
+#include "api/media_stream_interface.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+@interface RTCVideoRendererAdapter ()
+
+/**
+ * The Objective-C video renderer passed to this adapter during construction.
+ * Calls made to the webrtc::VideoRenderInterface will be adapted and passed to
+ * this video renderer.
+ */
+@property(nonatomic, readonly) id<RTC_OBJC_TYPE(RTCVideoRenderer)> videoRenderer;
+
+/**
+ * The native VideoSinkInterface surface exposed by this adapter. Calls made
+ * to this interface will be adapted and passed to the RTCVideoRenderer supplied
+ * during construction. This pointer is unsafe and owned by this class.
+ */
+@property(nonatomic, readonly) rtc::VideoSinkInterface<webrtc::VideoFrame> *nativeVideoRenderer;
+
+/** Initialize an RTCVideoRendererAdapter with an RTCVideoRenderer. */
+- (instancetype)initWithNativeRenderer:(id<RTC_OBJC_TYPE(RTCVideoRenderer)>)videoRenderer
+    NS_DESIGNATED_INITIALIZER;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/api/RTCVideoRendererAdapter.h b/third_party/libwebrtc/sdk/objc/api/RTCVideoRendererAdapter.h
new file mode 100644
index 0000000000..b0b6f04488
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/RTCVideoRendererAdapter.h
@@ -0,0 +1,27 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+NS_ASSUME_NONNULL_BEGIN
+
+/*
+ * Creates a rtc::VideoSinkInterface surface for an RTCVideoRenderer. The
+ * rtc::VideoSinkInterface is used by WebRTC rendering code - this
+ * adapter adapts calls made to that interface to the RTCVideoRenderer supplied
+ * during construction.
+ */
+@interface RTCVideoRendererAdapter : NSObject
+
+- (instancetype)init NS_UNAVAILABLE;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/api/RTCVideoRendererAdapter.mm b/third_party/libwebrtc/sdk/objc/api/RTCVideoRendererAdapter.mm
new file mode 100644
index 0000000000..ef02f72f60
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/RTCVideoRendererAdapter.mm
@@ -0,0 +1,67 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCVideoRendererAdapter+Private.h"
+#import "base/RTCVideoFrame.h"
+
+#include <memory>
+
+#include "sdk/objc/native/api/video_frame.h"
+
+namespace webrtc {
+
+class VideoRendererAdapter
+    : public rtc::VideoSinkInterface<webrtc::VideoFrame> {
+ public:
+  VideoRendererAdapter(RTCVideoRendererAdapter* adapter) {
+    adapter_ = adapter;
+    size_ = CGSizeZero;
+  }
+
+  void OnFrame(const webrtc::VideoFrame& nativeVideoFrame) override {
+    RTC_OBJC_TYPE(RTCVideoFrame)* videoFrame = NativeToObjCVideoFrame(nativeVideoFrame);
+
+    CGSize current_size = (videoFrame.rotation % 180 == 0)
+                              ? CGSizeMake(videoFrame.width, videoFrame.height)
+                              : CGSizeMake(videoFrame.height, videoFrame.width);
+
+    if (!CGSizeEqualToSize(size_, current_size)) {
+      size_ = current_size;
+      [adapter_.videoRenderer setSize:size_];
+    }
+    [adapter_.videoRenderer renderFrame:videoFrame];
+  }
+
+ private:
+  __weak RTCVideoRendererAdapter *adapter_;
+  CGSize size_;
+};
+}
+
+@implementation RTCVideoRendererAdapter {
+  std::unique_ptr<webrtc::VideoRendererAdapter> _adapter;
+}
+
+@synthesize videoRenderer = _videoRenderer;
+
+- (instancetype)initWithNativeRenderer:(id<RTC_OBJC_TYPE(RTCVideoRenderer)>)videoRenderer {
+  NSParameterAssert(videoRenderer);
+  if (self = [super init]) {
+    _videoRenderer = videoRenderer;
+    _adapter.reset(new webrtc::VideoRendererAdapter(self));
+  }
+  return self;
+}
+
+- (rtc::VideoSinkInterface<webrtc::VideoFrame> *)nativeVideoRenderer {
+  return _adapter.get();
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/api/logging/RTCCallbackLogger.h b/third_party/libwebrtc/sdk/objc/api/logging/RTCCallbackLogger.h
new file mode 100644
index 0000000000..c1aeb825cb
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/logging/RTCCallbackLogger.h
@@ -0,0 +1,41 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCLogging.h"
+#import "RTCMacros.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+typedef void (^RTCCallbackLoggerMessageHandler)(NSString *message);
+typedef void (^RTCCallbackLoggerMessageAndSeverityHandler)(NSString *message,
+                                                           RTCLoggingSeverity severity);
+
+// This class intercepts WebRTC logs and forwards them to a registered block.
+// This class is not threadsafe.
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCCallbackLogger) : NSObject
+
+// The severity level to capture. The default is kRTCLoggingSeverityInfo.
+@property(nonatomic, assign) RTCLoggingSeverity severity;
+
+// The callback handler will be called on the same thread that does the
+// logging, so if the logging callback can be slow it may be a good idea
+// to implement dispatching to some other queue.
+- (void)start:(nullable RTCCallbackLoggerMessageHandler)handler;
+- (void)startWithMessageAndSeverityHandler:
+        (nullable RTCCallbackLoggerMessageAndSeverityHandler)handler;
+
+- (void)stop;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/api/logging/RTCCallbackLogger.mm b/third_party/libwebrtc/sdk/objc/api/logging/RTCCallbackLogger.mm
new file mode 100644
index 0000000000..ba6fe1b1cc
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/logging/RTCCallbackLogger.mm
@@ -0,0 +1,151 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCCallbackLogger.h"
+
+#import "helpers/NSString+StdString.h"
+
+#include <memory>
+
+#include "absl/strings/string_view.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/log_sinks.h"
+#include "rtc_base/logging.h"
+
+namespace {
+
+class CallbackLogSink final : public rtc::LogSink {
+ public:
+  CallbackLogSink(RTCCallbackLoggerMessageHandler callbackHandler)
+      : callback_handler_(callbackHandler) {}
+
+  void OnLogMessage(const std::string& message) override {
+    OnLogMessage(absl::string_view(message));
+  }
+
+  void OnLogMessage(absl::string_view message) override {
+    if (callback_handler_) {
+      callback_handler_([NSString stringForAbslStringView:message]);
+    }
+  }
+
+ private:
+  RTCCallbackLoggerMessageHandler callback_handler_;
+};
+
+class CallbackWithSeverityLogSink final : public rtc::LogSink {
+ public:
+  CallbackWithSeverityLogSink(RTCCallbackLoggerMessageAndSeverityHandler callbackHandler)
+      : callback_handler_(callbackHandler) {}
+
+  void OnLogMessage(const std::string& message) override { RTC_DCHECK_NOTREACHED(); }
+
+  void OnLogMessage(const std::string& message, rtc::LoggingSeverity severity) override {
+    OnLogMessage(absl::string_view(message), severity);
+  }
+
+  void OnLogMessage(absl::string_view message, rtc::LoggingSeverity severity) override {
+    if (callback_handler_) {
+      RTCLoggingSeverity loggingSeverity = NativeSeverityToObjcSeverity(severity);
+      callback_handler_([NSString stringForAbslStringView:message], loggingSeverity);
+    }
+  }
+
+ private:
+  static RTCLoggingSeverity NativeSeverityToObjcSeverity(rtc::LoggingSeverity severity) {
+    switch (severity) {
+      case rtc::LS_VERBOSE:
+        return RTCLoggingSeverityVerbose;
+      case rtc::LS_INFO:
+        return RTCLoggingSeverityInfo;
+      case rtc::LS_WARNING:
+        return RTCLoggingSeverityWarning;
+      case rtc::LS_ERROR:
+        return RTCLoggingSeverityError;
+      case rtc::LS_NONE:
+        return RTCLoggingSeverityNone;
+    }
+  }
+
+  RTCCallbackLoggerMessageAndSeverityHandler callback_handler_;
+};
+
+}
+
+@implementation RTC_OBJC_TYPE (RTCCallbackLogger) {
+  BOOL _hasStarted;
+  std::unique_ptr<rtc::LogSink> _logSink;
+}
+
+@synthesize severity = _severity;
+
+- (instancetype)init {
+  self = [super init];
+  if (self != nil) {
+    _severity = RTCLoggingSeverityInfo;
+  }
+  return self;
+}
+
+- (void)dealloc {
+  [self stop];
+}
+
+- (void)start:(nullable RTCCallbackLoggerMessageHandler)handler {
+  if (_hasStarted) {
+    return;
+  }
+
+  _logSink.reset(new CallbackLogSink(handler));
+
+  rtc::LogMessage::AddLogToStream(_logSink.get(), [self rtcSeverity]);
+  _hasStarted = YES;
+}
+
+- (void)startWithMessageAndSeverityHandler:
+        (nullable RTCCallbackLoggerMessageAndSeverityHandler)handler {
+  if (_hasStarted) {
+    return;
+  }
+
+  _logSink.reset(new CallbackWithSeverityLogSink(handler));
+
+  rtc::LogMessage::AddLogToStream(_logSink.get(), [self rtcSeverity]);
+  _hasStarted = YES;
+}
+
+- (void)stop {
+  if (!_hasStarted) {
+    return;
+  }
+  RTC_DCHECK(_logSink);
+  rtc::LogMessage::RemoveLogToStream(_logSink.get());
+  _hasStarted = NO;
+  _logSink.reset();
+}
+
+#pragma mark - Private
+
+- (rtc::LoggingSeverity)rtcSeverity {
+  switch (_severity) {
+    case RTCLoggingSeverityVerbose:
+      return rtc::LS_VERBOSE;
+    case RTCLoggingSeverityInfo:
+      return rtc::LS_INFO;
+    case RTCLoggingSeverityWarning:
+      return rtc::LS_WARNING;
+    case RTCLoggingSeverityError:
+      return rtc::LS_ERROR;
+    case RTCLoggingSeverityNone:
+      return rtc::LS_NONE;
+  }
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCAudioSource+Private.h b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCAudioSource+Private.h
new file mode 100644
index 0000000000..2c333f9d73
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCAudioSource+Private.h
@@ -0,0 +1,34 @@
+/*
+ *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCAudioSource.h"
+
+#import "RTCMediaSource+Private.h"
+
+@interface RTC_OBJC_TYPE (RTCAudioSource)
+()
+
+    /**
+     * The AudioSourceInterface object passed to this RTCAudioSource during
+     * construction.
+     */
+    @property(nonatomic,
+              readonly) rtc::scoped_refptr<webrtc::AudioSourceInterface> nativeAudioSource;
+
+/** Initialize an RTCAudioSource from a native AudioSourceInterface. */
+- (instancetype)initWithFactory:(RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory
+              nativeAudioSource:(rtc::scoped_refptr<webrtc::AudioSourceInterface>)nativeAudioSource
+    NS_DESIGNATED_INITIALIZER;
+
+- (instancetype)initWithFactory:(RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory
+              nativeMediaSource:(rtc::scoped_refptr<webrtc::MediaSourceInterface>)nativeMediaSource
+                           type:(RTCMediaSourceType)type NS_UNAVAILABLE;
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCAudioSource.h b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCAudioSource.h
new file mode 100644
index 0000000000..9272fdf2d8
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCAudioSource.h
@@ -0,0 +1,32 @@
+/*
+ *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+#import "RTCMediaSource.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCAudioSource) : RTC_OBJC_TYPE(RTCMediaSource)
+
+- (instancetype)init NS_UNAVAILABLE;
+
+// Sets the volume for the RTCMediaSource. `volume` is a gain value in the range
+// [0, 10].
+// Temporary fix to be able to modify volume of remote audio tracks.
+// TODO(kthelgason): Property stays here temporarily until a proper volume-api
+// is available on the surface exposed by webrtc.
+@property(nonatomic, assign) double volume;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCAudioSource.mm b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCAudioSource.mm
new file mode 100644
index 0000000000..1541045099
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCAudioSource.mm
@@ -0,0 +1,52 @@
+/*
+ *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCAudioSource+Private.h"
+
+#include "rtc_base/checks.h"
+
+@implementation RTC_OBJC_TYPE (RTCAudioSource) {
+}
+
+@synthesize volume = _volume;
+@synthesize nativeAudioSource = _nativeAudioSource;
+
+- (instancetype)initWithFactory:(RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory
+              nativeAudioSource:
+                  (rtc::scoped_refptr<webrtc::AudioSourceInterface>)nativeAudioSource {
+  RTC_DCHECK(factory);
+  RTC_DCHECK(nativeAudioSource);
+
+  if (self = [super initWithFactory:factory
+                  nativeMediaSource:nativeAudioSource
+                               type:RTCMediaSourceTypeAudio]) {
+    _nativeAudioSource = nativeAudioSource;
+  }
+  return self;
+}
+
+- (instancetype)initWithFactory:(RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory
+              nativeMediaSource:(rtc::scoped_refptr<webrtc::MediaSourceInterface>)nativeMediaSource
+                           type:(RTCMediaSourceType)type {
+  RTC_DCHECK_NOTREACHED();
+  return nil;
+}
+
+- (NSString *)description {
+  NSString *stateString = [[self class] stringForState:self.state];
+  return [NSString stringWithFormat:@"RTC_OBJC_TYPE(RTCAudioSource)( %p ): %@", self, stateString];
+}
+
+- (void)setVolume:(double)volume {
+  _volume = volume;
+  _nativeAudioSource->SetVolume(volume);
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCAudioTrack+Private.h b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCAudioTrack+Private.h
new file mode 100644
index 0000000000..6495500484
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCAudioTrack+Private.h
@@ -0,0 +1,31 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCAudioTrack.h"
+
+#include "api/media_stream_interface.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+@class RTC_OBJC_TYPE(RTCPeerConnectionFactory);
+@interface RTC_OBJC_TYPE (RTCAudioTrack)
+()
+
+    /** AudioTrackInterface created or passed in at construction. */
+    @property(nonatomic, readonly) rtc::scoped_refptr<webrtc::AudioTrackInterface> nativeAudioTrack;
+
+/** Initialize an RTCAudioTrack with an id. */
+- (instancetype)initWithFactory:(RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory
+                         source:(RTC_OBJC_TYPE(RTCAudioSource) *)source
+                        trackId:(NSString *)trackId;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCAudioTrack.h b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCAudioTrack.h
new file mode 100644
index 0000000000..95eb5d3d48
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCAudioTrack.h
@@ -0,0 +1,28 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCMacros.h"
+#import "RTCMediaStreamTrack.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+@class RTC_OBJC_TYPE(RTCAudioSource);
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCAudioTrack) : RTC_OBJC_TYPE(RTCMediaStreamTrack)
+
+- (instancetype)init NS_UNAVAILABLE;
+
+/** The audio source for this audio track. */
+@property(nonatomic, readonly) RTC_OBJC_TYPE(RTCAudioSource) * source;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCAudioTrack.mm b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCAudioTrack.mm
new file mode 100644
index 0000000000..5c1736f436
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCAudioTrack.mm
@@ -0,0 +1,67 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCAudioTrack+Private.h"
+
+#import "RTCAudioSource+Private.h"
+#import "RTCMediaStreamTrack+Private.h"
+#import "RTCPeerConnectionFactory+Private.h"
+#import "helpers/NSString+StdString.h"
+
+#include "rtc_base/checks.h"
+
+@implementation RTC_OBJC_TYPE (RTCAudioTrack)
+
+@synthesize source = _source;
+
+- (instancetype)initWithFactory:(RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory
+                         source:(RTC_OBJC_TYPE(RTCAudioSource) *)source
+                        trackId:(NSString *)trackId {
+  RTC_DCHECK(factory);
+  RTC_DCHECK(source);
+  RTC_DCHECK(trackId.length);
+
+  std::string nativeId = [NSString stdStringForString:trackId];
+  rtc::scoped_refptr<webrtc::AudioTrackInterface> track =
+      factory.nativeFactory->CreateAudioTrack(nativeId, source.nativeAudioSource.get());
+  if (self = [self initWithFactory:factory nativeTrack:track type:RTCMediaStreamTrackTypeAudio]) {
+    _source = source;
+  }
+  return self;
+}
+
+- (instancetype)initWithFactory:(RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory
+                    nativeTrack:(rtc::scoped_refptr<webrtc::MediaStreamTrackInterface>)nativeTrack
+                           type:(RTCMediaStreamTrackType)type {
+  NSParameterAssert(factory);
+  NSParameterAssert(nativeTrack);
+  NSParameterAssert(type == RTCMediaStreamTrackTypeAudio);
+  return [super initWithFactory:factory nativeTrack:nativeTrack type:type];
+}
+
+- (RTC_OBJC_TYPE(RTCAudioSource) *)source {
+  if (!_source) {
+    rtc::scoped_refptr<webrtc::AudioSourceInterface> source(self.nativeAudioTrack->GetSource());
+    if (source) {
+      _source = [[RTC_OBJC_TYPE(RTCAudioSource) alloc] initWithFactory:self.factory
+                                                     nativeAudioSource:source];
+    }
+  }
+  return _source;
+}
+
+#pragma mark - Private
+
+- (rtc::scoped_refptr<webrtc::AudioTrackInterface>)nativeAudioTrack {
+  return rtc::scoped_refptr<webrtc::AudioTrackInterface>(
+      static_cast<webrtc::AudioTrackInterface *>(self.nativeTrack.get()));
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCCertificate.h b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCCertificate.h
new file mode 100644
index 0000000000..5ac8984d4a
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCCertificate.h
@@ -0,0 +1,44 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCCertificate) : NSObject <NSCopying>
+
+/** Private key in PEM. */
+@property(nonatomic, readonly, copy) NSString *private_key;
+
+/** Public key in an x509 cert encoded in PEM. */
+@property(nonatomic, readonly, copy) NSString *certificate;
+
+/**
+ * Initialize an RTCCertificate with PEM strings for private_key and certificate.
+ */
+- (instancetype)initWithPrivateKey:(NSString *)private_key
+                       certificate:(NSString *)certificate NS_DESIGNATED_INITIALIZER;
+
+- (instancetype)init NS_UNAVAILABLE;
+
+/** Generate a new certificate for 're' use.
+ *
+ *  Optional dictionary of parameters. Defaults to KeyType ECDSA if none are
+ *  provided.
+ *  - name: "ECDSA" or "RSASSA-PKCS1-v1_5"
+ */
++ (nullable RTC_OBJC_TYPE(RTCCertificate) *)generateCertificateWithParams:(NSDictionary *)params;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCCertificate.mm b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCCertificate.mm
new file mode 100644
index 0000000000..e5c33e407c
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCCertificate.mm
@@ -0,0 +1,72 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCCertificate.h"
+
+#import "base/RTCLogging.h"
+
+#include "rtc_base/logging.h"
+#include "rtc_base/rtc_certificate_generator.h"
+#include "rtc_base/ssl_identity.h"
+
+@implementation RTC_OBJC_TYPE (RTCCertificate)
+
+@synthesize private_key = _private_key;
+@synthesize certificate = _certificate;
+
+- (id)copyWithZone:(NSZone *)zone {
+  id copy = [[[self class] alloc] initWithPrivateKey:[self.private_key copyWithZone:zone]
+                                         certificate:[self.certificate copyWithZone:zone]];
+  return copy;
+}
+
+- (instancetype)initWithPrivateKey:(NSString *)private_key certificate:(NSString *)certificate {
+  if (self = [super init]) {
+    _private_key = [private_key copy];
+    _certificate = [certificate copy];
+  }
+  return self;
+}
+
++ (nullable RTC_OBJC_TYPE(RTCCertificate) *)generateCertificateWithParams:(NSDictionary *)params {
+  rtc::KeyType keyType = rtc::KT_ECDSA;
+  NSString *keyTypeString = [params valueForKey:@"name"];
+  if (keyTypeString && [keyTypeString isEqualToString:@"RSASSA-PKCS1-v1_5"]) {
+    keyType = rtc::KT_RSA;
+  }
+
+  NSNumber *expires = [params valueForKey:@"expires"];
+  rtc::scoped_refptr<rtc::RTCCertificate> cc_certificate = nullptr;
+  if (expires != nil) {
+    uint64_t expirationTimestamp = [expires unsignedLongLongValue];
+    cc_certificate = rtc::RTCCertificateGenerator::GenerateCertificate(rtc::KeyParams(keyType),
+                                                                       expirationTimestamp);
+  } else {
+    cc_certificate =
+        rtc::RTCCertificateGenerator::GenerateCertificate(rtc::KeyParams(keyType), absl::nullopt);
+  }
+  if (!cc_certificate) {
+    RTCLogError(@"Failed to generate certificate.");
+    return nullptr;
+  }
+  // grab PEMs and create an NS RTCCerticicate
+  rtc::RTCCertificatePEM pem = cc_certificate->ToPEM();
+  std::string pem_private_key = pem.private_key();
+  std::string pem_certificate = pem.certificate();
+  RTC_LOG(LS_INFO) << "CERT PEM ";
+  RTC_LOG(LS_INFO) << pem_certificate;
+
+  RTC_OBJC_TYPE(RTCCertificate) *cert =
+      [[RTC_OBJC_TYPE(RTCCertificate) alloc] initWithPrivateKey:@(pem_private_key.c_str())
+                                                    certificate:@(pem_certificate.c_str())];
+  return cert;
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCConfiguration+Native.h b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCConfiguration+Native.h
new file mode 100644
index 0000000000..07c0da6041
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCConfiguration+Native.h
@@ -0,0 +1,29 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCConfiguration.h"
+
+#include "api/peer_connection_interface.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+@interface RTC_OBJC_TYPE (RTCConfiguration)
+()
+
+    /** Optional TurnCustomizer.
+     *  With this class one can modify outgoing TURN messages.
+     *  The object passed in must remain valid until PeerConnection::Close() is
+     * called.
+     */
+    @property(nonatomic, nullable) webrtc::TurnCustomizer* turnCustomizer;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCConfiguration+Private.h b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCConfiguration+Private.h
new file mode 100644
index 0000000000..70a6532dbc
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCConfiguration+Private.h
@@ -0,0 +1,79 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCConfiguration.h"
+
+#include "api/peer_connection_interface.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+@interface RTC_OBJC_TYPE (RTCConfiguration)
+()
+
+    + (webrtc::PeerConnectionInterface::IceTransportsType)nativeTransportsTypeForTransportPolicy
+    : (RTCIceTransportPolicy)policy;
+
++ (RTCIceTransportPolicy)transportPolicyForTransportsType:
+        (webrtc::PeerConnectionInterface::IceTransportsType)nativeType;
+
++ (NSString *)stringForTransportPolicy:(RTCIceTransportPolicy)policy;
+
++ (webrtc::PeerConnectionInterface::BundlePolicy)nativeBundlePolicyForPolicy:
+        (RTCBundlePolicy)policy;
+
++ (RTCBundlePolicy)bundlePolicyForNativePolicy:
+        (webrtc::PeerConnectionInterface::BundlePolicy)nativePolicy;
+
++ (NSString *)stringForBundlePolicy:(RTCBundlePolicy)policy;
+
++ (webrtc::PeerConnectionInterface::RtcpMuxPolicy)nativeRtcpMuxPolicyForPolicy:
+        (RTCRtcpMuxPolicy)policy;
+
++ (RTCRtcpMuxPolicy)rtcpMuxPolicyForNativePolicy:
+        (webrtc::PeerConnectionInterface::RtcpMuxPolicy)nativePolicy;
+
++ (NSString *)stringForRtcpMuxPolicy:(RTCRtcpMuxPolicy)policy;
+
++ (webrtc::PeerConnectionInterface::TcpCandidatePolicy)nativeTcpCandidatePolicyForPolicy:
+        (RTCTcpCandidatePolicy)policy;
+
++ (RTCTcpCandidatePolicy)tcpCandidatePolicyForNativePolicy:
+        (webrtc::PeerConnectionInterface::TcpCandidatePolicy)nativePolicy;
+
++ (NSString *)stringForTcpCandidatePolicy:(RTCTcpCandidatePolicy)policy;
+
++ (webrtc::PeerConnectionInterface::CandidateNetworkPolicy)nativeCandidateNetworkPolicyForPolicy:
+        (RTCCandidateNetworkPolicy)policy;
+
++ (RTCCandidateNetworkPolicy)candidateNetworkPolicyForNativePolicy:
+        (webrtc::PeerConnectionInterface::CandidateNetworkPolicy)nativePolicy;
+
++ (NSString *)stringForCandidateNetworkPolicy:(RTCCandidateNetworkPolicy)policy;
+
++ (rtc::KeyType)nativeEncryptionKeyTypeForKeyType:(RTCEncryptionKeyType)keyType;
+
++ (webrtc::SdpSemantics)nativeSdpSemanticsForSdpSemantics:(RTCSdpSemantics)sdpSemantics;
+
++ (RTCSdpSemantics)sdpSemanticsForNativeSdpSemantics:(webrtc::SdpSemantics)sdpSemantics;
+
++ (NSString *)stringForSdpSemantics:(RTCSdpSemantics)sdpSemantics;
+
+/**
+ * RTCConfiguration struct representation of this RTCConfiguration.
+ * This is needed to pass to the underlying C++ APIs.
+ */
+- (nullable webrtc::PeerConnectionInterface::RTCConfiguration *)createNativeConfiguration;
+
+- (instancetype)initWithNativeConfiguration:
+        (const webrtc::PeerConnectionInterface::RTCConfiguration &)config NS_DESIGNATED_INITIALIZER;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCConfiguration.h b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCConfiguration.h
new file mode 100644
index 0000000000..1b0d14baf1
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCConfiguration.h
@@ -0,0 +1,268 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCCertificate.h"
+#import "RTCCryptoOptions.h"
+#import "RTCMacros.h"
+
+@class RTC_OBJC_TYPE(RTCIceServer);
+
+/**
+ * Represents the ice transport policy. This exposes the same states in C++,
+ * which include one more state than what exists in the W3C spec.
+ */
+typedef NS_ENUM(NSInteger, RTCIceTransportPolicy) {
+  RTCIceTransportPolicyNone,
+  RTCIceTransportPolicyRelay,
+  RTCIceTransportPolicyNoHost,
+  RTCIceTransportPolicyAll
+};
+
+/** Represents the bundle policy. */
+typedef NS_ENUM(NSInteger, RTCBundlePolicy) {
+  RTCBundlePolicyBalanced,
+  RTCBundlePolicyMaxCompat,
+  RTCBundlePolicyMaxBundle
+};
+
+/** Represents the rtcp mux policy. */
+typedef NS_ENUM(NSInteger, RTCRtcpMuxPolicy) { RTCRtcpMuxPolicyNegotiate, RTCRtcpMuxPolicyRequire };
+
+/** Represents the tcp candidate policy. */
+typedef NS_ENUM(NSInteger, RTCTcpCandidatePolicy) {
+  RTCTcpCandidatePolicyEnabled,
+  RTCTcpCandidatePolicyDisabled
+};
+
+/** Represents the candidate network policy. */
+typedef NS_ENUM(NSInteger, RTCCandidateNetworkPolicy) {
+  RTCCandidateNetworkPolicyAll,
+  RTCCandidateNetworkPolicyLowCost
+};
+
+/** Represents the continual gathering policy. */
+typedef NS_ENUM(NSInteger, RTCContinualGatheringPolicy) {
+  RTCContinualGatheringPolicyGatherOnce,
+  RTCContinualGatheringPolicyGatherContinually
+};
+
+/** Represents the encryption key type. */
+typedef NS_ENUM(NSInteger, RTCEncryptionKeyType) {
+  RTCEncryptionKeyTypeRSA,
+  RTCEncryptionKeyTypeECDSA,
+};
+
+/** Represents the chosen SDP semantics for the RTCPeerConnection. */
+typedef NS_ENUM(NSInteger, RTCSdpSemantics) {
+  // TODO(https://crbug.com/webrtc/13528): Remove support for Plan B.
+  RTCSdpSemanticsPlanB,
+  RTCSdpSemanticsUnifiedPlan,
+};
+
+NS_ASSUME_NONNULL_BEGIN
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCConfiguration) : NSObject
+
+/** If true, allows DSCP codes to be set on outgoing packets, configured using
+ *  networkPriority field of RTCRtpEncodingParameters. Defaults to false.
+ */
+@property(nonatomic, assign) BOOL enableDscp;
+
+/** An array of Ice Servers available to be used by ICE. */
+@property(nonatomic, copy) NSArray<RTC_OBJC_TYPE(RTCIceServer) *> *iceServers;
+
+/** An RTCCertificate for 're' use. */
+@property(nonatomic, nullable) RTC_OBJC_TYPE(RTCCertificate) * certificate;
+
+/** Which candidates the ICE agent is allowed to use. The W3C calls it
+ * `iceTransportPolicy`, while in C++ it is called `type`. */
+@property(nonatomic, assign) RTCIceTransportPolicy iceTransportPolicy;
+
+/** The media-bundling policy to use when gathering ICE candidates. */
+@property(nonatomic, assign) RTCBundlePolicy bundlePolicy;
+
+/** The rtcp-mux policy to use when gathering ICE candidates. */
+@property(nonatomic, assign) RTCRtcpMuxPolicy rtcpMuxPolicy;
+@property(nonatomic, assign) RTCTcpCandidatePolicy tcpCandidatePolicy;
+@property(nonatomic, assign) RTCCandidateNetworkPolicy candidateNetworkPolicy;
+@property(nonatomic, assign) RTCContinualGatheringPolicy continualGatheringPolicy;
+
+/** If set to YES, don't gather IPv6 ICE candidates on Wi-Fi.
+ *  Only intended to be used on specific devices. Certain phones disable IPv6
+ *  when the screen is turned off and it would be better to just disable the
+ *  IPv6 ICE candidates on Wi-Fi in those cases.
+ *  Default is NO.
+ */
+@property(nonatomic, assign) BOOL disableIPV6OnWiFi;
+
+/** By default, the PeerConnection will use a limited number of IPv6 network
+ *  interfaces, in order to avoid too many ICE candidate pairs being created
+ *  and delaying ICE completion.
+ *
+ *  Can be set to INT_MAX to effectively disable the limit.
+ */
+@property(nonatomic, assign) int maxIPv6Networks;
+
+/** Exclude link-local network interfaces
+ *  from considertaion for gathering ICE candidates.
+ *  Defaults to NO.
+ */
+@property(nonatomic, assign) BOOL disableLinkLocalNetworks;
+
+@property(nonatomic, assign) int audioJitterBufferMaxPackets;
+@property(nonatomic, assign) BOOL audioJitterBufferFastAccelerate;
+@property(nonatomic, assign) int iceConnectionReceivingTimeout;
+@property(nonatomic, assign) int iceBackupCandidatePairPingInterval;
+
+/** Key type used to generate SSL identity. Default is ECDSA. */
+@property(nonatomic, assign) RTCEncryptionKeyType keyType;
+
+/** ICE candidate pool size as defined in JSEP. Default is 0. */
+@property(nonatomic, assign) int iceCandidatePoolSize;
+
+/** Prune turn ports on the same network to the same turn server.
+ *  Default is NO.
+ */
+@property(nonatomic, assign) BOOL shouldPruneTurnPorts;
+
+/** If set to YES, this means the ICE transport should presume TURN-to-TURN
+ *  candidate pairs will succeed, even before a binding response is received.
+ */
+@property(nonatomic, assign) BOOL shouldPresumeWritableWhenFullyRelayed;
+
+/* This flag is only effective when `continualGatheringPolicy` is
+ * RTCContinualGatheringPolicyGatherContinually.
+ *
+ * If YES, after the ICE transport type is changed such that new types of
+ * ICE candidates are allowed by the new transport type, e.g. from
+ * RTCIceTransportPolicyRelay to RTCIceTransportPolicyAll, candidates that
+ * have been gathered by the ICE transport but not matching the previous
+ * transport type and as a result not observed by PeerConnectionDelegateAdapter,
+ * will be surfaced to the delegate.
+ */
+@property(nonatomic, assign) BOOL shouldSurfaceIceCandidatesOnIceTransportTypeChanged;
+
+/** If set to non-nil, controls the minimal interval between consecutive ICE
+ *  check packets.
+ */
+@property(nonatomic, copy, nullable) NSNumber *iceCheckMinInterval;
+
+/**
+ * Configure the SDP semantics used by this PeerConnection. By default, this
+ * is RTCSdpSemanticsUnifiedPlan which is compliant to the WebRTC 1.0
+ * specification. It is possible to overrwite this to the deprecated
+ * RTCSdpSemanticsPlanB SDP format, but note that RTCSdpSemanticsPlanB will be
+ * deleted at some future date, see https://crbug.com/webrtc/13528.
+ *
+ * RTCSdpSemanticsUnifiedPlan will cause RTCPeerConnection to create offers and
+ * answers with multiple m= sections where each m= section maps to one
+ * RTCRtpSender and one RTCRtpReceiver (an RTCRtpTransceiver), either both audio
+ * or both video. This will also cause RTCPeerConnection to ignore all but the
+ * first a=ssrc lines that form a Plan B stream.
+ *
+ * RTCSdpSemanticsPlanB will cause RTCPeerConnection to create offers and
+ * answers with at most one audio and one video m= section with multiple
+ * RTCRtpSenders and RTCRtpReceivers specified as multiple a=ssrc lines within
+ * the section. This will also cause RTCPeerConnection to ignore all but the
+ * first m= section of the same media type.
+ */
+@property(nonatomic, assign) RTCSdpSemantics sdpSemantics;
+
+/** Actively reset the SRTP parameters when the DTLS transports underneath are
+ *  changed after offer/answer negotiation. This is only intended to be a
+ *  workaround for crbug.com/835958
+ */
+@property(nonatomic, assign) BOOL activeResetSrtpParams;
+
+/** If the remote side support mid-stream codec switches then allow encoder
+ *  switching to be performed.
+ */
+
+@property(nonatomic, assign) BOOL allowCodecSwitching;
+
+/**
+ * Defines advanced optional cryptographic settings related to SRTP and
+ * frame encryption for native WebRTC. Setting this will overwrite any
+ * options set through the PeerConnectionFactory (which is deprecated).
+ */
+@property(nonatomic, nullable) RTC_OBJC_TYPE(RTCCryptoOptions) * cryptoOptions;
+
+/**
+ * An optional string that will be attached to the TURN_ALLOCATE_REQUEST which
+ * which can be used to correlate client logs with backend logs.
+ */
+@property(nonatomic, nullable, copy) NSString *turnLoggingId;
+
+/**
+ * Time interval between audio RTCP reports.
+ */
+@property(nonatomic, assign) int rtcpAudioReportIntervalMs;
+
+/**
+ * Time interval between video RTCP reports.
+ */
+@property(nonatomic, assign) int rtcpVideoReportIntervalMs;
+
+/**
+ * Allow implicit rollback of local description when remote description
+ * conflicts with local description.
+ * See: https://w3c.github.io/webrtc-pc/#dom-peerconnection-setremotedescription
+ */
+@property(nonatomic, assign) BOOL enableImplicitRollback;
+
+/**
+ * Control if "a=extmap-allow-mixed" is included in the offer.
+ * See: https://www.chromestatus.com/feature/6269234631933952
+ */
+@property(nonatomic, assign) BOOL offerExtmapAllowMixed;
+
+/**
+ * Defines the interval applied to ALL candidate pairs
+ * when ICE is strongly connected, and it overrides the
+ * default value of this interval in the ICE implementation;
+ */
+@property(nonatomic, copy, nullable) NSNumber *iceCheckIntervalStrongConnectivity;
+
+/**
+ * Defines the counterpart for ALL pairs when ICE is
+ * weakly connected, and it overrides the default value of
+ * this interval in the ICE implementation
+ */
+@property(nonatomic, copy, nullable) NSNumber *iceCheckIntervalWeakConnectivity;
+
+/**
+ * The min time period for which a candidate pair must wait for response to
+ * connectivity checks before it becomes unwritable. This parameter
+ * overrides the default value in the ICE implementation if set.
+ */
+@property(nonatomic, copy, nullable) NSNumber *iceUnwritableTimeout;
+
+/**
+ * The min number of connectivity checks that a candidate pair must sent
+ * without receiving response before it becomes unwritable. This parameter
+ * overrides the default value in the ICE implementation if set.
+ */
+@property(nonatomic, copy, nullable) NSNumber *iceUnwritableMinChecks;
+
+/**
+ * The min time period for which a candidate pair must wait for response to
+ * connectivity checks it becomes inactive. This parameter overrides the
+ * default value in the ICE implementation if set.
+ */
+@property(nonatomic, copy, nullable) NSNumber *iceInactiveTimeout;
+
+- (instancetype)init;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCConfiguration.mm b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCConfiguration.mm
new file mode 100644
index 0000000000..8e42cb2a82
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCConfiguration.mm
@@ -0,0 +1,544 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCConfiguration+Private.h"
+
+#include <memory>
+
+#import "RTCCertificate.h"
+#import "RTCConfiguration+Native.h"
+#import "RTCIceServer+Private.h"
+#import "base/RTCLogging.h"
+
+#include "rtc_base/checks.h"
+#include "rtc_base/rtc_certificate_generator.h"
+#include "rtc_base/ssl_identity.h"
+
+@implementation RTC_OBJC_TYPE (RTCConfiguration)
+
+@synthesize enableDscp = _enableDscp;
+@synthesize iceServers = _iceServers;
+@synthesize certificate = _certificate;
+@synthesize iceTransportPolicy = _iceTransportPolicy;
+@synthesize bundlePolicy = _bundlePolicy;
+@synthesize rtcpMuxPolicy = _rtcpMuxPolicy;
+@synthesize tcpCandidatePolicy = _tcpCandidatePolicy;
+@synthesize candidateNetworkPolicy = _candidateNetworkPolicy;
+@synthesize continualGatheringPolicy = _continualGatheringPolicy;
+@synthesize disableIPV6OnWiFi = _disableIPV6OnWiFi;
+@synthesize maxIPv6Networks = _maxIPv6Networks;
+@synthesize disableLinkLocalNetworks = _disableLinkLocalNetworks;
+@synthesize audioJitterBufferMaxPackets = _audioJitterBufferMaxPackets;
+@synthesize audioJitterBufferFastAccelerate = _audioJitterBufferFastAccelerate;
+@synthesize iceConnectionReceivingTimeout = _iceConnectionReceivingTimeout;
+@synthesize iceBackupCandidatePairPingInterval =
+    _iceBackupCandidatePairPingInterval;
+@synthesize keyType = _keyType;
+@synthesize iceCandidatePoolSize = _iceCandidatePoolSize;
+@synthesize shouldPruneTurnPorts = _shouldPruneTurnPorts;
+@synthesize shouldPresumeWritableWhenFullyRelayed =
+    _shouldPresumeWritableWhenFullyRelayed;
+@synthesize shouldSurfaceIceCandidatesOnIceTransportTypeChanged =
+    _shouldSurfaceIceCandidatesOnIceTransportTypeChanged;
+@synthesize iceCheckMinInterval = _iceCheckMinInterval;
+@synthesize sdpSemantics = _sdpSemantics;
+@synthesize turnCustomizer = _turnCustomizer;
+@synthesize activeResetSrtpParams = _activeResetSrtpParams;
+@synthesize allowCodecSwitching = _allowCodecSwitching;
+@synthesize cryptoOptions = _cryptoOptions;
+@synthesize turnLoggingId = _turnLoggingId;
+@synthesize rtcpAudioReportIntervalMs = _rtcpAudioReportIntervalMs;
+@synthesize rtcpVideoReportIntervalMs = _rtcpVideoReportIntervalMs;
+@synthesize enableImplicitRollback = _enableImplicitRollback;
+@synthesize offerExtmapAllowMixed = _offerExtmapAllowMixed;
+@synthesize iceCheckIntervalStrongConnectivity = _iceCheckIntervalStrongConnectivity;
+@synthesize iceCheckIntervalWeakConnectivity = _iceCheckIntervalWeakConnectivity;
+@synthesize iceUnwritableTimeout = _iceUnwritableTimeout;
+@synthesize iceUnwritableMinChecks = _iceUnwritableMinChecks;
+@synthesize iceInactiveTimeout = _iceInactiveTimeout;
+
+- (instancetype)init {
+  // Copy defaults.
+  webrtc::PeerConnectionInterface::RTCConfiguration config;
+  config.sdp_semantics = webrtc::SdpSemantics::kUnifiedPlan;
+  return [self initWithNativeConfiguration:config];
+}
+
+- (instancetype)initWithNativeConfiguration:
+    (const webrtc::PeerConnectionInterface::RTCConfiguration &)config {
+  if (self = [super init]) {
+    _enableDscp = config.dscp();
+    NSMutableArray *iceServers = [NSMutableArray array];
+    for (const webrtc::PeerConnectionInterface::IceServer& server : config.servers) {
+      RTC_OBJC_TYPE(RTCIceServer) *iceServer =
+          [[RTC_OBJC_TYPE(RTCIceServer) alloc] initWithNativeServer:server];
+      [iceServers addObject:iceServer];
+    }
+    _iceServers = iceServers;
+    if (!config.certificates.empty()) {
+      rtc::scoped_refptr<rtc::RTCCertificate> native_cert;
+      native_cert = config.certificates[0];
+      rtc::RTCCertificatePEM native_pem = native_cert->ToPEM();
+      _certificate = [[RTC_OBJC_TYPE(RTCCertificate) alloc]
+          initWithPrivateKey:@(native_pem.private_key().c_str())
+                 certificate:@(native_pem.certificate().c_str())];
+    }
+    _iceTransportPolicy =
+        [[self class] transportPolicyForTransportsType:config.type];
+    _bundlePolicy =
+        [[self class] bundlePolicyForNativePolicy:config.bundle_policy];
+    _rtcpMuxPolicy =
+        [[self class] rtcpMuxPolicyForNativePolicy:config.rtcp_mux_policy];
+    _tcpCandidatePolicy = [[self class] tcpCandidatePolicyForNativePolicy:
+        config.tcp_candidate_policy];
+    _candidateNetworkPolicy = [[self class]
+        candidateNetworkPolicyForNativePolicy:config.candidate_network_policy];
+    webrtc::PeerConnectionInterface::ContinualGatheringPolicy nativePolicy =
+    config.continual_gathering_policy;
+    _continualGatheringPolicy = [[self class] continualGatheringPolicyForNativePolicy:nativePolicy];
+    _disableIPV6OnWiFi = config.disable_ipv6_on_wifi;
+    _maxIPv6Networks = config.max_ipv6_networks;
+    _disableLinkLocalNetworks = config.disable_link_local_networks;
+    _audioJitterBufferMaxPackets = config.audio_jitter_buffer_max_packets;
+    _audioJitterBufferFastAccelerate = config.audio_jitter_buffer_fast_accelerate;
+    _iceConnectionReceivingTimeout = config.ice_connection_receiving_timeout;
+    _iceBackupCandidatePairPingInterval =
+        config.ice_backup_candidate_pair_ping_interval;
+    _keyType = RTCEncryptionKeyTypeECDSA;
+    _iceCandidatePoolSize = config.ice_candidate_pool_size;
+    _shouldPruneTurnPorts = config.prune_turn_ports;
+    _shouldPresumeWritableWhenFullyRelayed =
+        config.presume_writable_when_fully_relayed;
+    _shouldSurfaceIceCandidatesOnIceTransportTypeChanged =
+        config.surface_ice_candidates_on_ice_transport_type_changed;
+    if (config.ice_check_min_interval) {
+      _iceCheckMinInterval =
+          [NSNumber numberWithInt:*config.ice_check_min_interval];
+    }
+    _sdpSemantics = [[self class] sdpSemanticsForNativeSdpSemantics:config.sdp_semantics];
+    _turnCustomizer = config.turn_customizer;
+    _activeResetSrtpParams = config.active_reset_srtp_params;
+    if (config.crypto_options) {
+      _cryptoOptions = [[RTC_OBJC_TYPE(RTCCryptoOptions) alloc]
+               initWithSrtpEnableGcmCryptoSuites:config.crypto_options->srtp
+                                                     .enable_gcm_crypto_suites
+             srtpEnableAes128Sha1_32CryptoCipher:config.crypto_options->srtp
+                                                     .enable_aes128_sha1_32_crypto_cipher
+          srtpEnableEncryptedRtpHeaderExtensions:config.crypto_options->srtp
+                                                     .enable_encrypted_rtp_header_extensions
+                    sframeRequireFrameEncryption:config.crypto_options->sframe
+                                                     .require_frame_encryption];
+    }
+    _turnLoggingId = [NSString stringWithUTF8String:config.turn_logging_id.c_str()];
+    _rtcpAudioReportIntervalMs = config.audio_rtcp_report_interval_ms();
+    _rtcpVideoReportIntervalMs = config.video_rtcp_report_interval_ms();
+    _allowCodecSwitching = config.allow_codec_switching.value_or(false);
+    _enableImplicitRollback = config.enable_implicit_rollback;
+    _offerExtmapAllowMixed = config.offer_extmap_allow_mixed;
+    _iceCheckIntervalStrongConnectivity =
+        config.ice_check_interval_strong_connectivity.has_value() ?
+        [NSNumber numberWithInt:*config.ice_check_interval_strong_connectivity] :
+        nil;
+    _iceCheckIntervalWeakConnectivity = config.ice_check_interval_weak_connectivity.has_value() ?
+        [NSNumber numberWithInt:*config.ice_check_interval_weak_connectivity] :
+        nil;
+    _iceUnwritableTimeout = config.ice_unwritable_timeout.has_value() ?
+        [NSNumber numberWithInt:*config.ice_unwritable_timeout] :
+        nil;
+    _iceUnwritableMinChecks = config.ice_unwritable_min_checks.has_value() ?
+        [NSNumber numberWithInt:*config.ice_unwritable_min_checks] :
+        nil;
+    _iceInactiveTimeout = config.ice_inactive_timeout.has_value() ?
+        [NSNumber numberWithInt:*config.ice_inactive_timeout] :
+        nil;
+  }
+  return self;
+}
+
+- (NSString *)description {
+  static NSString *formatString = @"RTC_OBJC_TYPE(RTCConfiguration): "
+                                  @"{\n%@\n%@\n%@\n%@\n%@\n%@\n%@\n%@\n%d\n%d\n%d\n%d\n%d\n%d\n"
+                                  @"%d\n%@\n%d\n%d\n%d\n%d\n%d\n%d\n%d\n}\n";
+
+  return [NSString
+      stringWithFormat:formatString,
+                       _iceServers,
+                       [[self class] stringForTransportPolicy:_iceTransportPolicy],
+                       [[self class] stringForBundlePolicy:_bundlePolicy],
+                       [[self class] stringForRtcpMuxPolicy:_rtcpMuxPolicy],
+                       [[self class] stringForTcpCandidatePolicy:_tcpCandidatePolicy],
+                       [[self class] stringForCandidateNetworkPolicy:_candidateNetworkPolicy],
+                       [[self class] stringForContinualGatheringPolicy:_continualGatheringPolicy],
+                       [[self class] stringForSdpSemantics:_sdpSemantics],
+                       _audioJitterBufferMaxPackets,
+                       _audioJitterBufferFastAccelerate,
+                       _iceConnectionReceivingTimeout,
+                       _iceBackupCandidatePairPingInterval,
+                       _iceCandidatePoolSize,
+                       _shouldPruneTurnPorts,
+                       _shouldPresumeWritableWhenFullyRelayed,
+                       _shouldSurfaceIceCandidatesOnIceTransportTypeChanged,
+                       _iceCheckMinInterval,
+                       _disableLinkLocalNetworks,
+                       _disableIPV6OnWiFi,
+                       _maxIPv6Networks,
+                       _activeResetSrtpParams,
+                       _enableDscp,
+                       _enableImplicitRollback];
+}
+
+#pragma mark - Private
+
+- (webrtc::PeerConnectionInterface::RTCConfiguration *)
+    createNativeConfiguration {
+  std::unique_ptr<webrtc::PeerConnectionInterface::RTCConfiguration>
+      nativeConfig(new webrtc::PeerConnectionInterface::RTCConfiguration(
+          webrtc::PeerConnectionInterface::RTCConfigurationType::kAggressive));
+
+  nativeConfig->set_dscp(_enableDscp);
+  for (RTC_OBJC_TYPE(RTCIceServer) * iceServer in _iceServers) {
+    nativeConfig->servers.push_back(iceServer.nativeServer);
+  }
+  nativeConfig->type =
+      [[self class] nativeTransportsTypeForTransportPolicy:_iceTransportPolicy];
+  nativeConfig->bundle_policy =
+      [[self class] nativeBundlePolicyForPolicy:_bundlePolicy];
+  nativeConfig->rtcp_mux_policy =
+      [[self class] nativeRtcpMuxPolicyForPolicy:_rtcpMuxPolicy];
+  nativeConfig->tcp_candidate_policy =
+      [[self class] nativeTcpCandidatePolicyForPolicy:_tcpCandidatePolicy];
+  nativeConfig->candidate_network_policy = [[self class]
+      nativeCandidateNetworkPolicyForPolicy:_candidateNetworkPolicy];
+  nativeConfig->continual_gathering_policy =
+      [[self class] nativeContinualGatheringPolicyForPolicy:_continualGatheringPolicy];
+  nativeConfig->disable_ipv6_on_wifi = _disableIPV6OnWiFi;
+  nativeConfig->max_ipv6_networks = _maxIPv6Networks;
+  nativeConfig->disable_link_local_networks = _disableLinkLocalNetworks;
+  nativeConfig->audio_jitter_buffer_max_packets = _audioJitterBufferMaxPackets;
+  nativeConfig->audio_jitter_buffer_fast_accelerate =
+      _audioJitterBufferFastAccelerate  ? true : false;
+  nativeConfig->ice_connection_receiving_timeout =
+      _iceConnectionReceivingTimeout;
+  nativeConfig->ice_backup_candidate_pair_ping_interval =
+      _iceBackupCandidatePairPingInterval;
+  rtc::KeyType keyType =
+      [[self class] nativeEncryptionKeyTypeForKeyType:_keyType];
+  if (_certificate != nullptr) {
+    // if offered a pemcert use it...
+    RTC_LOG(LS_INFO) << "Have configured cert - using it.";
+    std::string pem_private_key = [[_certificate private_key] UTF8String];
+    std::string pem_certificate = [[_certificate certificate] UTF8String];
+    rtc::RTCCertificatePEM pem = rtc::RTCCertificatePEM(pem_private_key, pem_certificate);
+    rtc::scoped_refptr<rtc::RTCCertificate> certificate = rtc::RTCCertificate::FromPEM(pem);
+    RTC_LOG(LS_INFO) << "Created cert from PEM strings.";
+    if (!certificate) {
+      RTC_LOG(LS_ERROR) << "Failed to generate certificate from PEM.";
+      return nullptr;
+    }
+    nativeConfig->certificates.push_back(certificate);
+  } else {
+    RTC_LOG(LS_INFO) << "Don't have configured cert.";
+    // Generate non-default certificate.
+    if (keyType != rtc::KT_DEFAULT) {
+      rtc::scoped_refptr<rtc::RTCCertificate> certificate =
+          rtc::RTCCertificateGenerator::GenerateCertificate(rtc::KeyParams(keyType),
+                                                            absl::optional<uint64_t>());
+      if (!certificate) {
+        RTCLogError(@"Failed to generate certificate.");
+        return nullptr;
+      }
+      nativeConfig->certificates.push_back(certificate);
+    }
+  }
+  nativeConfig->ice_candidate_pool_size = _iceCandidatePoolSize;
+  nativeConfig->prune_turn_ports = _shouldPruneTurnPorts ? true : false;
+  nativeConfig->presume_writable_when_fully_relayed =
+      _shouldPresumeWritableWhenFullyRelayed ? true : false;
+  nativeConfig->surface_ice_candidates_on_ice_transport_type_changed =
+      _shouldSurfaceIceCandidatesOnIceTransportTypeChanged ? true : false;
+  if (_iceCheckMinInterval != nil) {
+    nativeConfig->ice_check_min_interval = absl::optional<int>(_iceCheckMinInterval.intValue);
+  }
+  nativeConfig->sdp_semantics = [[self class] nativeSdpSemanticsForSdpSemantics:_sdpSemantics];
+  if (_turnCustomizer) {
+    nativeConfig->turn_customizer = _turnCustomizer;
+  }
+  nativeConfig->active_reset_srtp_params = _activeResetSrtpParams ? true : false;
+  if (_cryptoOptions) {
+    webrtc::CryptoOptions nativeCryptoOptions;
+    nativeCryptoOptions.srtp.enable_gcm_crypto_suites =
+        _cryptoOptions.srtpEnableGcmCryptoSuites ? true : false;
+    nativeCryptoOptions.srtp.enable_aes128_sha1_32_crypto_cipher =
+        _cryptoOptions.srtpEnableAes128Sha1_32CryptoCipher ? true : false;
+    nativeCryptoOptions.srtp.enable_encrypted_rtp_header_extensions =
+        _cryptoOptions.srtpEnableEncryptedRtpHeaderExtensions ? true : false;
+    nativeCryptoOptions.sframe.require_frame_encryption =
+        _cryptoOptions.sframeRequireFrameEncryption ? true : false;
+    nativeConfig->crypto_options = absl::optional<webrtc::CryptoOptions>(nativeCryptoOptions);
+  }
+  nativeConfig->turn_logging_id = [_turnLoggingId UTF8String];
+  nativeConfig->set_audio_rtcp_report_interval_ms(_rtcpAudioReportIntervalMs);
+  nativeConfig->set_video_rtcp_report_interval_ms(_rtcpVideoReportIntervalMs);
+  nativeConfig->allow_codec_switching = _allowCodecSwitching;
+  nativeConfig->enable_implicit_rollback = _enableImplicitRollback;
+  nativeConfig->offer_extmap_allow_mixed = _offerExtmapAllowMixed;
+  if (_iceCheckIntervalStrongConnectivity != nil) {
+    nativeConfig->ice_check_interval_strong_connectivity =
+        absl::optional<int>(_iceCheckIntervalStrongConnectivity.intValue);
+  }
+  if (_iceCheckIntervalWeakConnectivity != nil) {
+    nativeConfig->ice_check_interval_weak_connectivity =
+        absl::optional<int>(_iceCheckIntervalWeakConnectivity.intValue);
+  }
+  if (_iceUnwritableTimeout != nil) {
+    nativeConfig->ice_unwritable_timeout = absl::optional<int>(_iceUnwritableTimeout.intValue);
+  }
+  if (_iceUnwritableMinChecks != nil) {
+    nativeConfig->ice_unwritable_min_checks = absl::optional<int>(_iceUnwritableMinChecks.intValue);
+  }
+  if (_iceInactiveTimeout != nil) {
+    nativeConfig->ice_inactive_timeout = absl::optional<int>(_iceInactiveTimeout.intValue);
+  }
+  return nativeConfig.release();
+}
+
++ (webrtc::PeerConnectionInterface::IceTransportsType)
+    nativeTransportsTypeForTransportPolicy:(RTCIceTransportPolicy)policy {
+  switch (policy) {
+    case RTCIceTransportPolicyNone:
+      return webrtc::PeerConnectionInterface::kNone;
+    case RTCIceTransportPolicyRelay:
+      return webrtc::PeerConnectionInterface::kRelay;
+    case RTCIceTransportPolicyNoHost:
+      return webrtc::PeerConnectionInterface::kNoHost;
+    case RTCIceTransportPolicyAll:
+      return webrtc::PeerConnectionInterface::kAll;
+  }
+}
+
++ (RTCIceTransportPolicy)transportPolicyForTransportsType:
+    (webrtc::PeerConnectionInterface::IceTransportsType)nativeType {
+  switch (nativeType) {
+    case webrtc::PeerConnectionInterface::kNone:
+      return RTCIceTransportPolicyNone;
+    case webrtc::PeerConnectionInterface::kRelay:
+      return RTCIceTransportPolicyRelay;
+    case webrtc::PeerConnectionInterface::kNoHost:
+      return RTCIceTransportPolicyNoHost;
+    case webrtc::PeerConnectionInterface::kAll:
+      return RTCIceTransportPolicyAll;
+  }
+}
+
++ (NSString *)stringForTransportPolicy:(RTCIceTransportPolicy)policy {
+  switch (policy) {
+    case RTCIceTransportPolicyNone:
+      return @"NONE";
+    case RTCIceTransportPolicyRelay:
+      return @"RELAY";
+    case RTCIceTransportPolicyNoHost:
+      return @"NO_HOST";
+    case RTCIceTransportPolicyAll:
+      return @"ALL";
+  }
+}
+
++ (webrtc::PeerConnectionInterface::BundlePolicy)nativeBundlePolicyForPolicy:
+    (RTCBundlePolicy)policy {
+  switch (policy) {
+    case RTCBundlePolicyBalanced:
+      return webrtc::PeerConnectionInterface::kBundlePolicyBalanced;
+    case RTCBundlePolicyMaxCompat:
+      return webrtc::PeerConnectionInterface::kBundlePolicyMaxCompat;
+    case RTCBundlePolicyMaxBundle:
+      return webrtc::PeerConnectionInterface::kBundlePolicyMaxBundle;
+  }
+}
+
++ (RTCBundlePolicy)bundlePolicyForNativePolicy:
+    (webrtc::PeerConnectionInterface::BundlePolicy)nativePolicy {
+  switch (nativePolicy) {
+    case webrtc::PeerConnectionInterface::kBundlePolicyBalanced:
+      return RTCBundlePolicyBalanced;
+    case webrtc::PeerConnectionInterface::kBundlePolicyMaxCompat:
+      return RTCBundlePolicyMaxCompat;
+    case webrtc::PeerConnectionInterface::kBundlePolicyMaxBundle:
+      return RTCBundlePolicyMaxBundle;
+  }
+}
+
++ (NSString *)stringForBundlePolicy:(RTCBundlePolicy)policy {
+  switch (policy) {
+    case RTCBundlePolicyBalanced:
+      return @"BALANCED";
+    case RTCBundlePolicyMaxCompat:
+      return @"MAX_COMPAT";
+    case RTCBundlePolicyMaxBundle:
+      return @"MAX_BUNDLE";
+  }
+}
+
++ (webrtc::PeerConnectionInterface::RtcpMuxPolicy)nativeRtcpMuxPolicyForPolicy:
+    (RTCRtcpMuxPolicy)policy {
+  switch (policy) {
+    case RTCRtcpMuxPolicyNegotiate:
+      return webrtc::PeerConnectionInterface::kRtcpMuxPolicyNegotiate;
+    case RTCRtcpMuxPolicyRequire:
+      return webrtc::PeerConnectionInterface::kRtcpMuxPolicyRequire;
+  }
+}
+
++ (RTCRtcpMuxPolicy)rtcpMuxPolicyForNativePolicy:
+    (webrtc::PeerConnectionInterface::RtcpMuxPolicy)nativePolicy {
+  switch (nativePolicy) {
+    case webrtc::PeerConnectionInterface::kRtcpMuxPolicyNegotiate:
+      return RTCRtcpMuxPolicyNegotiate;
+    case webrtc::PeerConnectionInterface::kRtcpMuxPolicyRequire:
+      return RTCRtcpMuxPolicyRequire;
+  }
+}
+
++ (NSString *)stringForRtcpMuxPolicy:(RTCRtcpMuxPolicy)policy {
+  switch (policy) {
+    case RTCRtcpMuxPolicyNegotiate:
+      return @"NEGOTIATE";
+    case RTCRtcpMuxPolicyRequire:
+      return @"REQUIRE";
+  }
+}
+
++ (webrtc::PeerConnectionInterface::TcpCandidatePolicy)
+    nativeTcpCandidatePolicyForPolicy:(RTCTcpCandidatePolicy)policy {
+  switch (policy) {
+    case RTCTcpCandidatePolicyEnabled:
+      return webrtc::PeerConnectionInterface::kTcpCandidatePolicyEnabled;
+    case RTCTcpCandidatePolicyDisabled:
+      return webrtc::PeerConnectionInterface::kTcpCandidatePolicyDisabled;
+  }
+}
+
++ (webrtc::PeerConnectionInterface::CandidateNetworkPolicy)
+    nativeCandidateNetworkPolicyForPolicy:(RTCCandidateNetworkPolicy)policy {
+  switch (policy) {
+    case RTCCandidateNetworkPolicyAll:
+      return webrtc::PeerConnectionInterface::kCandidateNetworkPolicyAll;
+    case RTCCandidateNetworkPolicyLowCost:
+      return webrtc::PeerConnectionInterface::kCandidateNetworkPolicyLowCost;
+  }
+}
+
++ (RTCTcpCandidatePolicy)tcpCandidatePolicyForNativePolicy:
+    (webrtc::PeerConnectionInterface::TcpCandidatePolicy)nativePolicy {
+  switch (nativePolicy) {
+    case webrtc::PeerConnectionInterface::kTcpCandidatePolicyEnabled:
+      return RTCTcpCandidatePolicyEnabled;
+    case webrtc::PeerConnectionInterface::kTcpCandidatePolicyDisabled:
+      return RTCTcpCandidatePolicyDisabled;
+  }
+}
+
++ (NSString *)stringForTcpCandidatePolicy:(RTCTcpCandidatePolicy)policy {
+  switch (policy) {
+    case RTCTcpCandidatePolicyEnabled:
+      return @"TCP_ENABLED";
+    case RTCTcpCandidatePolicyDisabled:
+      return @"TCP_DISABLED";
+  }
+}
+
++ (RTCCandidateNetworkPolicy)candidateNetworkPolicyForNativePolicy:
+    (webrtc::PeerConnectionInterface::CandidateNetworkPolicy)nativePolicy {
+  switch (nativePolicy) {
+    case webrtc::PeerConnectionInterface::kCandidateNetworkPolicyAll:
+      return RTCCandidateNetworkPolicyAll;
+    case webrtc::PeerConnectionInterface::kCandidateNetworkPolicyLowCost:
+      return RTCCandidateNetworkPolicyLowCost;
+  }
+}
+
++ (NSString *)stringForCandidateNetworkPolicy:
+    (RTCCandidateNetworkPolicy)policy {
+  switch (policy) {
+    case RTCCandidateNetworkPolicyAll:
+      return @"CANDIDATE_ALL_NETWORKS";
+    case RTCCandidateNetworkPolicyLowCost:
+      return @"CANDIDATE_LOW_COST_NETWORKS";
+  }
+}
+
++ (webrtc::PeerConnectionInterface::ContinualGatheringPolicy)
+    nativeContinualGatheringPolicyForPolicy:
+        (RTCContinualGatheringPolicy)policy {
+  switch (policy) {
+    case RTCContinualGatheringPolicyGatherOnce:
+      return webrtc::PeerConnectionInterface::GATHER_ONCE;
+    case RTCContinualGatheringPolicyGatherContinually:
+      return webrtc::PeerConnectionInterface::GATHER_CONTINUALLY;
+  }
+}
+
++ (RTCContinualGatheringPolicy)continualGatheringPolicyForNativePolicy:
+    (webrtc::PeerConnectionInterface::ContinualGatheringPolicy)nativePolicy {
+  switch (nativePolicy) {
+    case webrtc::PeerConnectionInterface::GATHER_ONCE:
+      return RTCContinualGatheringPolicyGatherOnce;
+    case webrtc::PeerConnectionInterface::GATHER_CONTINUALLY:
+      return RTCContinualGatheringPolicyGatherContinually;
+  }
+}
+
++ (NSString *)stringForContinualGatheringPolicy:
+    (RTCContinualGatheringPolicy)policy {
+  switch (policy) {
+    case RTCContinualGatheringPolicyGatherOnce:
+      return @"GATHER_ONCE";
+    case RTCContinualGatheringPolicyGatherContinually:
+      return @"GATHER_CONTINUALLY";
+  }
+}
+
++ (rtc::KeyType)nativeEncryptionKeyTypeForKeyType:
+    (RTCEncryptionKeyType)keyType {
+  switch (keyType) {
+    case RTCEncryptionKeyTypeRSA:
+      return rtc::KT_RSA;
+    case RTCEncryptionKeyTypeECDSA:
+      return rtc::KT_ECDSA;
+  }
+}
+
++ (webrtc::SdpSemantics)nativeSdpSemanticsForSdpSemantics:(RTCSdpSemantics)sdpSemantics {
+  switch (sdpSemantics) {
+    case RTCSdpSemanticsPlanB:
+      return webrtc::SdpSemantics::kPlanB_DEPRECATED;
+    case RTCSdpSemanticsUnifiedPlan:
+      return webrtc::SdpSemantics::kUnifiedPlan;
+  }
+}
+
++ (RTCSdpSemantics)sdpSemanticsForNativeSdpSemantics:(webrtc::SdpSemantics)sdpSemantics {
+  switch (sdpSemantics) {
+    case webrtc::SdpSemantics::kPlanB_DEPRECATED:
+      return RTCSdpSemanticsPlanB;
+    case webrtc::SdpSemantics::kUnifiedPlan:
+      return RTCSdpSemanticsUnifiedPlan;
+  }
+}
+
++ (NSString *)stringForSdpSemantics:(RTCSdpSemantics)sdpSemantics {
+  switch (sdpSemantics) {
+    case RTCSdpSemanticsPlanB:
+      return @"PLAN_B";
+    case RTCSdpSemanticsUnifiedPlan:
+      return @"UNIFIED_PLAN";
+  }
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCCryptoOptions.h b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCCryptoOptions.h
new file mode 100644
index 0000000000..7894c8d50c
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCCryptoOptions.h
@@ -0,0 +1,63 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+/**
+ * Objective-C bindings for webrtc::CryptoOptions. This API had to be flattened
+ * as Objective-C doesn't support nested structures.
+ */
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCCryptoOptions) : NSObject
+
+/**
+ * Enable GCM crypto suites from RFC 7714 for SRTP. GCM will only be used
+ * if both sides enable it
+ */
+@property(nonatomic, assign) BOOL srtpEnableGcmCryptoSuites;
+/**
+ * If set to true, the (potentially insecure) crypto cipher
+ * kSrtpAes128CmSha1_32 will be included in the list of supported ciphers
+ * during negotiation. It will only be used if both peers support it and no
+ * other ciphers get preferred.
+ */
+@property(nonatomic, assign) BOOL srtpEnableAes128Sha1_32CryptoCipher;
+/**
+ * If set to true, encrypted RTP header extensions as defined in RFC 6904
+ * will be negotiated. They will only be used if both peers support them.
+ */
+@property(nonatomic, assign) BOOL srtpEnableEncryptedRtpHeaderExtensions;
+
+/**
+ * If set all RtpSenders must have an FrameEncryptor attached to them before
+ * they are allowed to send packets. All RtpReceivers must have a
+ * FrameDecryptor attached to them before they are able to receive packets.
+ */
+@property(nonatomic, assign) BOOL sframeRequireFrameEncryption;
+
+/**
+ * Initializes CryptoOptions with all possible options set explicitly. This
+ * is done when converting from a native RTCConfiguration.crypto_options.
+ */
+- (instancetype)initWithSrtpEnableGcmCryptoSuites:(BOOL)srtpEnableGcmCryptoSuites
+              srtpEnableAes128Sha1_32CryptoCipher:(BOOL)srtpEnableAes128Sha1_32CryptoCipher
+           srtpEnableEncryptedRtpHeaderExtensions:(BOOL)srtpEnableEncryptedRtpHeaderExtensions
+                     sframeRequireFrameEncryption:(BOOL)sframeRequireFrameEncryption
+    NS_DESIGNATED_INITIALIZER;
+
+- (instancetype)init NS_UNAVAILABLE;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCCryptoOptions.mm b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCCryptoOptions.mm
new file mode 100644
index 0000000000..fbaa1de58d
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCCryptoOptions.mm
@@ -0,0 +1,33 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCCryptoOptions.h"
+
+@implementation RTC_OBJC_TYPE (RTCCryptoOptions)
+
+@synthesize srtpEnableGcmCryptoSuites = _srtpEnableGcmCryptoSuites;
+@synthesize srtpEnableAes128Sha1_32CryptoCipher = _srtpEnableAes128Sha1_32CryptoCipher;
+@synthesize srtpEnableEncryptedRtpHeaderExtensions = _srtpEnableEncryptedRtpHeaderExtensions;
+@synthesize sframeRequireFrameEncryption = _sframeRequireFrameEncryption;
+
+- (instancetype)initWithSrtpEnableGcmCryptoSuites:(BOOL)srtpEnableGcmCryptoSuites
+              srtpEnableAes128Sha1_32CryptoCipher:(BOOL)srtpEnableAes128Sha1_32CryptoCipher
+           srtpEnableEncryptedRtpHeaderExtensions:(BOOL)srtpEnableEncryptedRtpHeaderExtensions
+                     sframeRequireFrameEncryption:(BOOL)sframeRequireFrameEncryption {
+  if (self = [super init]) {
+    _srtpEnableGcmCryptoSuites = srtpEnableGcmCryptoSuites;
+    _srtpEnableAes128Sha1_32CryptoCipher = srtpEnableAes128Sha1_32CryptoCipher;
+    _srtpEnableEncryptedRtpHeaderExtensions = srtpEnableEncryptedRtpHeaderExtensions;
+    _sframeRequireFrameEncryption = sframeRequireFrameEncryption;
+  }
+  return self;
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCDataChannel+Private.h b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCDataChannel+Private.h
new file mode 100644
index 0000000000..2cdbdabec6
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCDataChannel+Private.h
@@ -0,0 +1,52 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCDataChannel.h"
+
+#include "api/data_channel_interface.h"
+#include "api/scoped_refptr.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+@class RTC_OBJC_TYPE(RTCPeerConnectionFactory);
+
+@interface RTC_OBJC_TYPE (RTCDataBuffer)
+()
+
+    /**
+     * The native DataBuffer representation of this RTCDatabuffer object. This is
+     * needed to pass to the underlying C++ APIs.
+     */
+    @property(nonatomic, readonly) const webrtc::DataBuffer *nativeDataBuffer;
+
+/** Initialize an RTCDataBuffer from a native DataBuffer. */
+- (instancetype)initWithNativeBuffer:(const webrtc::DataBuffer &)nativeBuffer;
+
+@end
+
+@interface RTC_OBJC_TYPE (RTCDataChannel)
+()
+
+    /** Initialize an RTCDataChannel from a native DataChannelInterface. */
+    - (instancetype)initWithFactory
+    : (RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory nativeDataChannel
+    : (rtc::scoped_refptr<webrtc::DataChannelInterface>)nativeDataChannel NS_DESIGNATED_INITIALIZER;
+
++ (webrtc::DataChannelInterface::DataState)nativeDataChannelStateForState:
+        (RTCDataChannelState)state;
+
++ (RTCDataChannelState)dataChannelStateForNativeState:
+        (webrtc::DataChannelInterface::DataState)nativeState;
+
++ (NSString *)stringForState:(RTCDataChannelState)state;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCDataChannel.h b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCDataChannel.h
new file mode 100644
index 0000000000..89eb58bc3f
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCDataChannel.h
@@ -0,0 +1,132 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <AvailabilityMacros.h>
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCDataBuffer) : NSObject
+
+/** NSData representation of the underlying buffer. */
+@property(nonatomic, readonly) NSData *data;
+
+/** Indicates whether `data` contains UTF-8 or binary data. */
+@property(nonatomic, readonly) BOOL isBinary;
+
+- (instancetype)init NS_UNAVAILABLE;
+
+/**
+ * Initialize an RTCDataBuffer from NSData. `isBinary` indicates whether `data`
+ * contains UTF-8 or binary data.
+ */
+- (instancetype)initWithData:(NSData *)data isBinary:(BOOL)isBinary;
+
+@end
+
+@class RTC_OBJC_TYPE(RTCDataChannel);
+RTC_OBJC_EXPORT
+@protocol RTC_OBJC_TYPE
+(RTCDataChannelDelegate)<NSObject>
+
+    /** The data channel state changed. */
+    - (void)dataChannelDidChangeState : (RTC_OBJC_TYPE(RTCDataChannel) *)dataChannel;
+
+/** The data channel successfully received a data buffer. */
+- (void)dataChannel:(RTC_OBJC_TYPE(RTCDataChannel) *)dataChannel
+    didReceiveMessageWithBuffer:(RTC_OBJC_TYPE(RTCDataBuffer) *)buffer;
+
+@optional
+/** The data channel's `bufferedAmount` changed. */
+- (void)dataChannel:(RTC_OBJC_TYPE(RTCDataChannel) *)dataChannel
+    didChangeBufferedAmount:(uint64_t)amount;
+
+@end
+
+/** Represents the state of the data channel. */
+typedef NS_ENUM(NSInteger, RTCDataChannelState) {
+  RTCDataChannelStateConnecting,
+  RTCDataChannelStateOpen,
+  RTCDataChannelStateClosing,
+  RTCDataChannelStateClosed,
+};
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCDataChannel) : NSObject
+
+/**
+ * A label that can be used to distinguish this data channel from other data
+ * channel objects.
+ */
+@property(nonatomic, readonly) NSString *label;
+
+/** Whether the data channel can send messages in unreliable mode. */
+@property(nonatomic, readonly) BOOL isReliable DEPRECATED_ATTRIBUTE;
+
+/** Returns whether this data channel is ordered or not. */
+@property(nonatomic, readonly) BOOL isOrdered;
+
+/** Deprecated. Use maxPacketLifeTime. */
+@property(nonatomic, readonly) NSUInteger maxRetransmitTime DEPRECATED_ATTRIBUTE;
+
+/**
+ * The length of the time window (in milliseconds) during which transmissions
+ * and retransmissions may occur in unreliable mode.
+ */
+@property(nonatomic, readonly) uint16_t maxPacketLifeTime;
+
+/**
+ * The maximum number of retransmissions that are attempted in unreliable mode.
+ */
+@property(nonatomic, readonly) uint16_t maxRetransmits;
+
+/**
+ * The name of the sub-protocol used with this data channel, if any. Otherwise
+ * this returns an empty string.
+ */
+@property(nonatomic, readonly) NSString *protocol;
+
+/**
+ * Returns whether this data channel was negotiated by the application or not.
+ */
+@property(nonatomic, readonly) BOOL isNegotiated;
+
+/** Deprecated. Use channelId. */
+@property(nonatomic, readonly) NSInteger streamId DEPRECATED_ATTRIBUTE;
+
+/** The identifier for this data channel. */
+@property(nonatomic, readonly) int channelId;
+
+/** The state of the data channel. */
+@property(nonatomic, readonly) RTCDataChannelState readyState;
+
+/**
+ * The number of bytes of application data that have been queued using
+ * `sendData:` but that have not yet been transmitted to the network.
+ */
+@property(nonatomic, readonly) uint64_t bufferedAmount;
+
+/** The delegate for this data channel. */
+@property(nonatomic, weak) id<RTC_OBJC_TYPE(RTCDataChannelDelegate)> delegate;
+
+- (instancetype)init NS_UNAVAILABLE;
+
+/** Closes the data channel. */
+- (void)close;
+
+/** Attempt to send `data` on this data channel's underlying data transport. */
+- (BOOL)sendData:(RTC_OBJC_TYPE(RTCDataBuffer) *)data;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCDataChannel.mm b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCDataChannel.mm
new file mode 100644
index 0000000000..4a79cefdb4
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCDataChannel.mm
@@ -0,0 +1,220 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCDataChannel+Private.h"
+
+#import "helpers/NSString+StdString.h"
+
+#include <memory>
+
+namespace webrtc {
+
+class DataChannelDelegateAdapter : public DataChannelObserver {
+ public:
+  DataChannelDelegateAdapter(RTC_OBJC_TYPE(RTCDataChannel) * channel) { channel_ = channel; }
+
+  void OnStateChange() override {
+    [channel_.delegate dataChannelDidChangeState:channel_];
+  }
+
+  void OnMessage(const DataBuffer& buffer) override {
+    RTC_OBJC_TYPE(RTCDataBuffer) *data_buffer =
+        [[RTC_OBJC_TYPE(RTCDataBuffer) alloc] initWithNativeBuffer:buffer];
+    [channel_.delegate dataChannel:channel_
+       didReceiveMessageWithBuffer:data_buffer];
+  }
+
+  void OnBufferedAmountChange(uint64_t previousAmount) override {
+    id<RTC_OBJC_TYPE(RTCDataChannelDelegate)> delegate = channel_.delegate;
+    SEL sel = @selector(dataChannel:didChangeBufferedAmount:);
+    if ([delegate respondsToSelector:sel]) {
+      [delegate dataChannel:channel_ didChangeBufferedAmount:previousAmount];
+    }
+  }
+
+ private:
+  __weak RTC_OBJC_TYPE(RTCDataChannel) * channel_;
+};
+}
+
+@implementation RTC_OBJC_TYPE (RTCDataBuffer) {
+  std::unique_ptr<webrtc::DataBuffer> _dataBuffer;
+}
+
+- (instancetype)initWithData:(NSData *)data isBinary:(BOOL)isBinary {
+  NSParameterAssert(data);
+  if (self = [super init]) {
+    rtc::CopyOnWriteBuffer buffer(
+        reinterpret_cast<const uint8_t*>(data.bytes), data.length);
+    _dataBuffer.reset(new webrtc::DataBuffer(buffer, isBinary));
+  }
+  return self;
+}
+
+- (NSData *)data {
+  return [NSData dataWithBytes:_dataBuffer->data.data()
+                        length:_dataBuffer->data.size()];
+}
+
+- (BOOL)isBinary {
+  return _dataBuffer->binary;
+}
+
+#pragma mark - Private
+
+- (instancetype)initWithNativeBuffer:(const webrtc::DataBuffer&)nativeBuffer {
+  if (self = [super init]) {
+    _dataBuffer.reset(new webrtc::DataBuffer(nativeBuffer));
+  }
+  return self;
+}
+
+- (const webrtc::DataBuffer *)nativeDataBuffer {
+  return _dataBuffer.get();
+}
+
+@end
+
+@implementation RTC_OBJC_TYPE (RTCDataChannel) {
+  RTC_OBJC_TYPE(RTCPeerConnectionFactory) * _factory;
+  rtc::scoped_refptr<webrtc::DataChannelInterface> _nativeDataChannel;
+  std::unique_ptr<webrtc::DataChannelDelegateAdapter> _observer;
+  BOOL _isObserverRegistered;
+}
+
+@synthesize delegate = _delegate;
+
+- (void)dealloc {
+  // Handles unregistering the observer properly. We need to do this because
+  // there may still be other references to the underlying data channel.
+  _nativeDataChannel->UnregisterObserver();
+}
+
+- (NSString *)label {
+  return [NSString stringForStdString:_nativeDataChannel->label()];
+}
+
+- (BOOL)isReliable {
+  return _nativeDataChannel->reliable();
+}
+
+- (BOOL)isOrdered {
+  return _nativeDataChannel->ordered();
+}
+
+- (NSUInteger)maxRetransmitTime {
+  return self.maxPacketLifeTime;
+}
+
+- (uint16_t)maxPacketLifeTime {
+  return _nativeDataChannel->maxRetransmitTime();
+}
+
+- (uint16_t)maxRetransmits {
+  return _nativeDataChannel->maxRetransmits();
+}
+
+- (NSString *)protocol {
+  return [NSString stringForStdString:_nativeDataChannel->protocol()];
+}
+
+- (BOOL)isNegotiated {
+  return _nativeDataChannel->negotiated();
+}
+
+- (NSInteger)streamId {
+  return self.channelId;
+}
+
+- (int)channelId {
+  return _nativeDataChannel->id();
+}
+
+- (RTCDataChannelState)readyState {
+  return [[self class] dataChannelStateForNativeState:
+      _nativeDataChannel->state()];
+}
+
+- (uint64_t)bufferedAmount {
+  return _nativeDataChannel->buffered_amount();
+}
+
+- (void)close {
+  _nativeDataChannel->Close();
+}
+
+- (BOOL)sendData:(RTC_OBJC_TYPE(RTCDataBuffer) *)data {
+  return _nativeDataChannel->Send(*data.nativeDataBuffer);
+}
+
+- (NSString *)description {
+  return [NSString stringWithFormat:@"RTC_OBJC_TYPE(RTCDataChannel):\n%ld\n%@\n%@",
+                                    (long)self.channelId,
+                                    self.label,
+                                    [[self class] stringForState:self.readyState]];
+}
+
+#pragma mark - Private
+
+- (instancetype)initWithFactory:(RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory
+              nativeDataChannel:
+                  (rtc::scoped_refptr<webrtc::DataChannelInterface>)nativeDataChannel {
+  NSParameterAssert(nativeDataChannel);
+  if (self = [super init]) {
+    _factory = factory;
+    _nativeDataChannel = nativeDataChannel;
+    _observer.reset(new webrtc::DataChannelDelegateAdapter(self));
+    _nativeDataChannel->RegisterObserver(_observer.get());
+  }
+  return self;
+}
+
++ (webrtc::DataChannelInterface::DataState)
+    nativeDataChannelStateForState:(RTCDataChannelState)state {
+  switch (state) {
+    case RTCDataChannelStateConnecting:
+      return webrtc::DataChannelInterface::DataState::kConnecting;
+    case RTCDataChannelStateOpen:
+      return webrtc::DataChannelInterface::DataState::kOpen;
+    case RTCDataChannelStateClosing:
+      return webrtc::DataChannelInterface::DataState::kClosing;
+    case RTCDataChannelStateClosed:
+      return webrtc::DataChannelInterface::DataState::kClosed;
+  }
+}
+
++ (RTCDataChannelState)dataChannelStateForNativeState:
+    (webrtc::DataChannelInterface::DataState)nativeState {
+  switch (nativeState) {
+    case webrtc::DataChannelInterface::DataState::kConnecting:
+      return RTCDataChannelStateConnecting;
+    case webrtc::DataChannelInterface::DataState::kOpen:
+      return RTCDataChannelStateOpen;
+    case webrtc::DataChannelInterface::DataState::kClosing:
+      return RTCDataChannelStateClosing;
+    case webrtc::DataChannelInterface::DataState::kClosed:
+      return RTCDataChannelStateClosed;
+  }
+}
+
++ (NSString *)stringForState:(RTCDataChannelState)state {
+  switch (state) {
+    case RTCDataChannelStateConnecting:
+      return @"Connecting";
+    case RTCDataChannelStateOpen:
+      return @"Open";
+    case RTCDataChannelStateClosing:
+      return @"Closing";
+    case RTCDataChannelStateClosed:
+      return @"Closed";
+  }
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCDataChannelConfiguration+Private.h b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCDataChannelConfiguration+Private.h
new file mode 100644
index 0000000000..5aef10fcef
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCDataChannelConfiguration+Private.h
@@ -0,0 +1,24 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCDataChannelConfiguration.h"
+
+#include "api/data_channel_interface.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+@interface RTC_OBJC_TYPE (RTCDataChannelConfiguration)
+()
+
+    @property(nonatomic, readonly) webrtc::DataChannelInit nativeDataChannelInit;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCDataChannelConfiguration.h b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCDataChannelConfiguration.h
new file mode 100644
index 0000000000..9459ae0a13
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCDataChannelConfiguration.h
@@ -0,0 +1,52 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <AvailabilityMacros.h>
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCDataChannelConfiguration) : NSObject
+
+/** Set to YES if ordered delivery is required. */
+@property(nonatomic, assign) BOOL isOrdered;
+
+/** Deprecated. Use maxPacketLifeTime. */
+@property(nonatomic, assign) NSInteger maxRetransmitTimeMs DEPRECATED_ATTRIBUTE;
+
+/**
+ * Max period in milliseconds in which retransmissions will be sent. After this
+ * time, no more retransmissions will be sent. -1 if unset.
+ */
+@property(nonatomic, assign) int maxPacketLifeTime;
+
+/** The max number of retransmissions. -1 if unset. */
+@property(nonatomic, assign) int maxRetransmits;
+
+/** Set to YES if the channel has been externally negotiated and we do not send
+ * an in-band signalling in the form of an "open" message.
+ */
+@property(nonatomic, assign) BOOL isNegotiated;
+
+/** Deprecated. Use channelId. */
+@property(nonatomic, assign) int streamId DEPRECATED_ATTRIBUTE;
+
+/** The id of the data channel. */
+@property(nonatomic, assign) int channelId;
+
+/** Set by the application and opaque to the WebRTC implementation. */
+@property(nonatomic) NSString* protocol;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCDataChannelConfiguration.mm b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCDataChannelConfiguration.mm
new file mode 100644
index 0000000000..bf775b1afd
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCDataChannelConfiguration.mm
@@ -0,0 +1,87 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCDataChannelConfiguration+Private.h"
+
+#import "helpers/NSString+StdString.h"
+
+@implementation RTC_OBJC_TYPE (RTCDataChannelConfiguration)
+
+@synthesize nativeDataChannelInit = _nativeDataChannelInit;
+
+- (BOOL)isOrdered {
+  return _nativeDataChannelInit.ordered;
+}
+
+- (void)setIsOrdered:(BOOL)isOrdered {
+  _nativeDataChannelInit.ordered = isOrdered;
+}
+
+- (NSInteger)maxRetransmitTimeMs {
+  return self.maxPacketLifeTime;
+}
+
+- (void)setMaxRetransmitTimeMs:(NSInteger)maxRetransmitTimeMs {
+  self.maxPacketLifeTime = maxRetransmitTimeMs;
+}
+
+- (int)maxPacketLifeTime {
+  return *_nativeDataChannelInit.maxRetransmitTime;
+}
+
+- (void)setMaxPacketLifeTime:(int)maxPacketLifeTime {
+  _nativeDataChannelInit.maxRetransmitTime = maxPacketLifeTime;
+}
+
+- (int)maxRetransmits {
+  if (_nativeDataChannelInit.maxRetransmits) {
+    return *_nativeDataChannelInit.maxRetransmits;
+  } else {
+    return -1;
+  }
+}
+
+- (void)setMaxRetransmits:(int)maxRetransmits {
+  _nativeDataChannelInit.maxRetransmits = maxRetransmits;
+}
+
+- (NSString *)protocol {
+  return [NSString stringForStdString:_nativeDataChannelInit.protocol];
+}
+
+- (void)setProtocol:(NSString *)protocol {
+  _nativeDataChannelInit.protocol = [NSString stdStringForString:protocol];
+}
+
+- (BOOL)isNegotiated {
+  return _nativeDataChannelInit.negotiated;
+}
+
+- (void)setIsNegotiated:(BOOL)isNegotiated {
+  _nativeDataChannelInit.negotiated = isNegotiated;
+}
+
+- (int)streamId {
+  return self.channelId;
+}
+
+- (void)setStreamId:(int)streamId {
+  self.channelId = streamId;
+}
+
+- (int)channelId {
+  return _nativeDataChannelInit.id;
+}
+
+- (void)setChannelId:(int)channelId {
+  _nativeDataChannelInit.id = channelId;
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCDtmfSender+Private.h b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCDtmfSender+Private.h
new file mode 100644
index 0000000000..49a62164cd
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCDtmfSender+Private.h
@@ -0,0 +1,29 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCDtmfSender.h"
+
+#include "api/dtmf_sender_interface.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+@interface RTC_OBJC_TYPE (RTCDtmfSender) : NSObject <RTC_OBJC_TYPE(RTCDtmfSender)>
+
+@property(nonatomic, readonly) rtc::scoped_refptr<webrtc::DtmfSenderInterface> nativeDtmfSender;
+
+- (instancetype)init NS_UNAVAILABLE;
+
+/** Initialize an RTCDtmfSender with a native DtmfSenderInterface. */
+- (instancetype)initWithNativeDtmfSender:
+        (rtc::scoped_refptr<webrtc::DtmfSenderInterface>)nativeDtmfSender NS_DESIGNATED_INITIALIZER;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCDtmfSender.h b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCDtmfSender.h
new file mode 100644
index 0000000000..0f1b6ba4da
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCDtmfSender.h
@@ -0,0 +1,71 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+RTC_OBJC_EXPORT
+@protocol RTC_OBJC_TYPE
+(RTCDtmfSender)<NSObject>
+
+    /**
+     * Returns true if this RTCDtmfSender is capable of sending DTMF. Otherwise
+     * returns false. To be able to send DTMF, the associated RTCRtpSender must be
+     * able to send packets, and a "telephone-event" codec must be negotiated.
+     */
+    @property(nonatomic, readonly) BOOL canInsertDtmf;
+
+/**
+ * Queues a task that sends the DTMF tones. The tones parameter is treated
+ * as a series of characters. The characters 0 through 9, A through D, #, and *
+ * generate the associated DTMF tones. The characters a to d are equivalent
+ * to A to D. The character ',' indicates a delay of 2 seconds before
+ * processing the next character in the tones parameter.
+ *
+ * Unrecognized characters are ignored.
+ *
+ * @param duration The parameter indicates the duration to use for each
+ * character passed in the tones parameter. The duration cannot be more
+ * than 6000 or less than 70 ms.
+ *
+ * @param interToneGap The parameter indicates the gap between tones.
+ * This parameter must be at least 50 ms but should be as short as
+ * possible.
+ *
+ * If InsertDtmf is called on the same object while an existing task for this
+ * object to generate DTMF is still running, the previous task is canceled.
+ * Returns true on success and false on failure.
+ */
+- (BOOL)insertDtmf:(nonnull NSString *)tones
+          duration:(NSTimeInterval)duration
+      interToneGap:(NSTimeInterval)interToneGap;
+
+/** The tones remaining to be played out */
+- (nonnull NSString *)remainingTones;
+
+/**
+ * The current tone duration value. This value will be the value last set via the
+ * insertDtmf method, or the default value of 100 ms if insertDtmf was never called.
+ */
+- (NSTimeInterval)duration;
+
+/**
+ * The current value of the between-tone gap. This value will be the value last set
+ * via the insertDtmf() method, or the default value of 50 ms if insertDtmf() was never
+ * called.
+ */
+- (NSTimeInterval)interToneGap;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCDtmfSender.mm b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCDtmfSender.mm
new file mode 100644
index 0000000000..ee3b79cd37
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCDtmfSender.mm
@@ -0,0 +1,74 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCDtmfSender+Private.h"
+
+#import "base/RTCLogging.h"
+#import "helpers/NSString+StdString.h"
+
+#include "rtc_base/time_utils.h"
+
+@implementation RTC_OBJC_TYPE (RTCDtmfSender) {
+  rtc::scoped_refptr<webrtc::DtmfSenderInterface> _nativeDtmfSender;
+}
+
+- (BOOL)canInsertDtmf {
+  return _nativeDtmfSender->CanInsertDtmf();
+}
+
+- (BOOL)insertDtmf:(nonnull NSString *)tones
+          duration:(NSTimeInterval)duration
+      interToneGap:(NSTimeInterval)interToneGap {
+  RTC_DCHECK(tones != nil);
+
+  int durationMs = static_cast<int>(duration * rtc::kNumMillisecsPerSec);
+  int interToneGapMs = static_cast<int>(interToneGap * rtc::kNumMillisecsPerSec);
+  return _nativeDtmfSender->InsertDtmf(
+      [NSString stdStringForString:tones], durationMs, interToneGapMs);
+}
+
+- (nonnull NSString *)remainingTones {
+  return [NSString stringForStdString:_nativeDtmfSender->tones()];
+}
+
+- (NSTimeInterval)duration {
+  return static_cast<NSTimeInterval>(_nativeDtmfSender->duration()) / rtc::kNumMillisecsPerSec;
+}
+
+- (NSTimeInterval)interToneGap {
+  return static_cast<NSTimeInterval>(_nativeDtmfSender->inter_tone_gap()) /
+      rtc::kNumMillisecsPerSec;
+}
+
+- (NSString *)description {
+  return [NSString stringWithFormat:@"RTC_OBJC_TYPE(RTCDtmfSender) {\n  remainingTones: %@\n  "
+                                    @"duration: %f sec\n  interToneGap: %f sec\n}",
+                                    [self remainingTones],
+                                    [self duration],
+                                    [self interToneGap]];
+}
+
+#pragma mark - Private
+
+- (rtc::scoped_refptr<webrtc::DtmfSenderInterface>)nativeDtmfSender {
+  return _nativeDtmfSender;
+}
+
+- (instancetype)initWithNativeDtmfSender:
+        (rtc::scoped_refptr<webrtc::DtmfSenderInterface>)nativeDtmfSender {
+  NSParameterAssert(nativeDtmfSender);
+  if (self = [super init]) {
+    _nativeDtmfSender = nativeDtmfSender;
+    RTCLogInfo(
+        @"RTC_OBJC_TYPE(RTCDtmfSender)(%p): created DTMF sender: %@", self, self.description);
+  }
+  return self;
+}
+@end
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCEncodedImage+Private.h b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCEncodedImage+Private.h
new file mode 100644
index 0000000000..a078b0aded
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCEncodedImage+Private.h
@@ -0,0 +1,26 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "base/RTCEncodedImage.h"
+
+#include "api/video/encoded_image.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+/* Interfaces for converting to/from internal C++ formats. */
+@interface RTC_OBJC_TYPE (RTCEncodedImage)
+(Private)
+
+    - (instancetype)initWithNativeEncodedImage : (const webrtc::EncodedImage &)encodedImage;
+- (webrtc::EncodedImage)nativeEncodedImage;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCEncodedImage+Private.mm b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCEncodedImage+Private.mm
new file mode 100644
index 0000000000..7f8ae739e0
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCEncodedImage+Private.mm
@@ -0,0 +1,130 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCEncodedImage+Private.h"
+
+#import <objc/runtime.h>
+
+#include "rtc_base/numerics/safe_conversions.h"
+
+namespace {
+// An implementation of EncodedImageBufferInterface that doesn't perform any copies.
+class ObjCEncodedImageBuffer : public webrtc::EncodedImageBufferInterface {
+ public:
+  static rtc::scoped_refptr<ObjCEncodedImageBuffer> Create(NSData *data) {
+    return rtc::make_ref_counted<ObjCEncodedImageBuffer>(data);
+  }
+  const uint8_t *data() const override { return static_cast<const uint8_t *>(data_.bytes); }
+  // TODO(bugs.webrtc.org/9378): delete this non-const data method.
+  uint8_t *data() override {
+    return const_cast<uint8_t *>(static_cast<const uint8_t *>(data_.bytes));
+  }
+  size_t size() const override { return data_.length; }
+
+ protected:
+  explicit ObjCEncodedImageBuffer(NSData *data) : data_(data) {}
+  ~ObjCEncodedImageBuffer() {}
+
+  NSData *data_;
+};
+}
+
+// A simple wrapper around webrtc::EncodedImageBufferInterface to make it usable with associated
+// objects.
+@interface RTCWrappedEncodedImageBuffer : NSObject
+@property(nonatomic) rtc::scoped_refptr<webrtc::EncodedImageBufferInterface> buffer;
+- (instancetype)initWithEncodedImageBuffer:
+    (rtc::scoped_refptr<webrtc::EncodedImageBufferInterface>)buffer;
+@end
+@implementation RTCWrappedEncodedImageBuffer
+@synthesize buffer = _buffer;
+- (instancetype)initWithEncodedImageBuffer:
+    (rtc::scoped_refptr<webrtc::EncodedImageBufferInterface>)buffer {
+  self = [super init];
+  if (self) {
+    _buffer = buffer;
+  }
+  return self;
+}
+@end
+
+@implementation RTC_OBJC_TYPE (RTCEncodedImage)
+(Private)
+
+    - (rtc::scoped_refptr<webrtc::EncodedImageBufferInterface>)encodedData {
+  RTCWrappedEncodedImageBuffer *wrappedBuffer =
+      objc_getAssociatedObject(self, @selector(encodedData));
+  return wrappedBuffer.buffer;
+}
+
+- (void)setEncodedData:(rtc::scoped_refptr<webrtc::EncodedImageBufferInterface>)buffer {
+  return objc_setAssociatedObject(
+      self,
+      @selector(encodedData),
+      [[RTCWrappedEncodedImageBuffer alloc] initWithEncodedImageBuffer:buffer],
+      OBJC_ASSOCIATION_RETAIN_NONATOMIC);
+}
+
+- (instancetype)initWithNativeEncodedImage:(const webrtc::EncodedImage &)encodedImage {
+  if (self = [super init]) {
+    // A reference to the encodedData must be stored so that it's kept alive as long
+    // self.buffer references its underlying data.
+    self.encodedData = encodedImage.GetEncodedData();
+    // Wrap the buffer in NSData without copying, do not take ownership.
+    self.buffer = [NSData dataWithBytesNoCopy:self.encodedData->data()
+                                       length:encodedImage.size()
+                                 freeWhenDone:NO];
+    self.encodedWidth = rtc::dchecked_cast<int32_t>(encodedImage._encodedWidth);
+    self.encodedHeight = rtc::dchecked_cast<int32_t>(encodedImage._encodedHeight);
+    self.timeStamp = encodedImage.Timestamp();
+    self.captureTimeMs = encodedImage.capture_time_ms_;
+    self.ntpTimeMs = encodedImage.ntp_time_ms_;
+    self.flags = encodedImage.timing_.flags;
+    self.encodeStartMs = encodedImage.timing_.encode_start_ms;
+    self.encodeFinishMs = encodedImage.timing_.encode_finish_ms;
+    self.frameType = static_cast<RTCFrameType>(encodedImage._frameType);
+    self.rotation = static_cast<RTCVideoRotation>(encodedImage.rotation_);
+    self.qp = @(encodedImage.qp_);
+    self.contentType = (encodedImage.content_type_ == webrtc::VideoContentType::SCREENSHARE) ?
+        RTCVideoContentTypeScreenshare :
+        RTCVideoContentTypeUnspecified;
+  }
+
+  return self;
+}
+
+- (webrtc::EncodedImage)nativeEncodedImage {
+  // Return the pointer without copying.
+  webrtc::EncodedImage encodedImage;
+  if (self.encodedData) {
+    encodedImage.SetEncodedData(self.encodedData);
+  } else if (self.buffer) {
+    encodedImage.SetEncodedData(ObjCEncodedImageBuffer::Create(self.buffer));
+  }
+  encodedImage.set_size(self.buffer.length);
+  encodedImage._encodedWidth = rtc::dchecked_cast<uint32_t>(self.encodedWidth);
+  encodedImage._encodedHeight = rtc::dchecked_cast<uint32_t>(self.encodedHeight);
+  encodedImage.SetTimestamp(self.timeStamp);
+  encodedImage.capture_time_ms_ = self.captureTimeMs;
+  encodedImage.ntp_time_ms_ = self.ntpTimeMs;
+  encodedImage.timing_.flags = self.flags;
+  encodedImage.timing_.encode_start_ms = self.encodeStartMs;
+  encodedImage.timing_.encode_finish_ms = self.encodeFinishMs;
+  encodedImage._frameType = webrtc::VideoFrameType(self.frameType);
+  encodedImage.rotation_ = webrtc::VideoRotation(self.rotation);
+  encodedImage.qp_ = self.qp ? self.qp.intValue : -1;
+  encodedImage.content_type_ = (self.contentType == RTCVideoContentTypeScreenshare) ?
+      webrtc::VideoContentType::SCREENSHARE :
+      webrtc::VideoContentType::UNSPECIFIED;
+
+  return encodedImage;
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCFieldTrials.h b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCFieldTrials.h
new file mode 100644
index 0000000000..3e8fcc8075
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCFieldTrials.h
@@ -0,0 +1,30 @@
+/*
+ *  Copyright 2016 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+/** The only valid value for the following if set is kRTCFieldTrialEnabledValue. */
+RTC_EXTERN NSString *const kRTCFieldTrialAudioForceABWENoTWCCKey;
+RTC_EXTERN NSString * const kRTCFieldTrialFlexFec03AdvertisedKey;
+RTC_EXTERN NSString * const kRTCFieldTrialFlexFec03Key;
+RTC_EXTERN NSString * const kRTCFieldTrialH264HighProfileKey;
+RTC_EXTERN NSString * const kRTCFieldTrialMinimizeResamplingOnMobileKey;
+RTC_EXTERN NSString *const kRTCFieldTrialUseNWPathMonitor;
+
+/** The valid value for field trials above. */
+RTC_EXTERN NSString * const kRTCFieldTrialEnabledValue;
+
+/** Initialize field trials using a dictionary mapping field trial keys to their
+ * values. See above for valid keys and values. Must be called before any other
+ * call into WebRTC. See: webrtc/system_wrappers/include/field_trial.h
+ */
+RTC_EXTERN void RTCInitFieldTrialDictionary(NSDictionary<NSString *, NSString *> *fieldTrials);
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCFieldTrials.mm b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCFieldTrials.mm
new file mode 100644
index 0000000000..193da9e4f7
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCFieldTrials.mm
@@ -0,0 +1,56 @@
+/*
+ *  Copyright 2016 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCFieldTrials.h"
+
+#include <memory>
+
+#import "base/RTCLogging.h"
+
+#include "system_wrappers/include/field_trial.h"
+
+NSString *const kRTCFieldTrialAudioForceABWENoTWCCKey = @"WebRTC-Audio-ABWENoTWCC";
+NSString * const kRTCFieldTrialFlexFec03AdvertisedKey = @"WebRTC-FlexFEC-03-Advertised";
+NSString * const kRTCFieldTrialFlexFec03Key = @"WebRTC-FlexFEC-03";
+NSString * const kRTCFieldTrialH264HighProfileKey = @"WebRTC-H264HighProfile";
+NSString * const kRTCFieldTrialMinimizeResamplingOnMobileKey =
+    @"WebRTC-Audio-MinimizeResamplingOnMobile";
+NSString *const kRTCFieldTrialUseNWPathMonitor = @"WebRTC-Network-UseNWPathMonitor";
+NSString * const kRTCFieldTrialEnabledValue = @"Enabled";
+
+// InitFieldTrialsFromString stores the char*, so the char array must outlive
+// the application.
+static char *gFieldTrialInitString = nullptr;
+
+void RTCInitFieldTrialDictionary(NSDictionary<NSString *, NSString *> *fieldTrials) {
+  if (!fieldTrials) {
+    RTCLogWarning(@"No fieldTrials provided.");
+    return;
+  }
+  // Assemble the keys and values into the field trial string.
+  // We don't perform any extra format checking. That should be done by the underlying WebRTC calls.
+  NSMutableString *fieldTrialInitString = [NSMutableString string];
+  for (NSString *key in fieldTrials) {
+    NSString *fieldTrialEntry = [NSString stringWithFormat:@"%@/%@/", key, fieldTrials[key]];
+    [fieldTrialInitString appendString:fieldTrialEntry];
+  }
+  size_t len = fieldTrialInitString.length + 1;
+  if (gFieldTrialInitString != nullptr) {
+    delete[] gFieldTrialInitString;
+  }
+  gFieldTrialInitString = new char[len];
+  if (![fieldTrialInitString getCString:gFieldTrialInitString
+                              maxLength:len
+                               encoding:NSUTF8StringEncoding]) {
+    RTCLogError(@"Failed to convert field trial string.");
+    return;
+  }
+  webrtc::field_trial::InitFieldTrialsFromString(gFieldTrialInitString);
+}
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCFileLogger.h b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCFileLogger.h
new file mode 100644
index 0000000000..cb397c9633
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCFileLogger.h
@@ -0,0 +1,74 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+typedef NS_ENUM(NSUInteger, RTCFileLoggerSeverity) {
+  RTCFileLoggerSeverityVerbose,
+  RTCFileLoggerSeverityInfo,
+  RTCFileLoggerSeverityWarning,
+  RTCFileLoggerSeverityError
+};
+
+typedef NS_ENUM(NSUInteger, RTCFileLoggerRotationType) {
+  RTCFileLoggerTypeCall,
+  RTCFileLoggerTypeApp,
+};
+
+NS_ASSUME_NONNULL_BEGIN
+
+// This class intercepts WebRTC logs and saves them to a file. The file size
+// will not exceed the given maximum bytesize. When the maximum bytesize is
+// reached, logs are rotated according to the rotationType specified.
+// For kRTCFileLoggerTypeCall, logs from the beginning and the end
+// are preserved while the middle section is overwritten instead.
+// For kRTCFileLoggerTypeApp, the oldest log is overwritten.
+// This class is not threadsafe.
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCFileLogger) : NSObject
+
+// The severity level to capture. The default is kRTCFileLoggerSeverityInfo.
+@property(nonatomic, assign) RTCFileLoggerSeverity severity;
+
+// The rotation type for this file logger. The default is
+// kRTCFileLoggerTypeCall.
+@property(nonatomic, readonly) RTCFileLoggerRotationType rotationType;
+
+// Disables buffering disk writes. Should be set before `start`. Buffering
+// is enabled by default for performance.
+@property(nonatomic, assign) BOOL shouldDisableBuffering;
+
+// Default constructor provides default settings for dir path, file size and
+// rotation type.
+- (instancetype)init;
+
+// Create file logger with default rotation type.
+- (instancetype)initWithDirPath:(NSString *)dirPath maxFileSize:(NSUInteger)maxFileSize;
+
+- (instancetype)initWithDirPath:(NSString *)dirPath
+                    maxFileSize:(NSUInteger)maxFileSize
+                   rotationType:(RTCFileLoggerRotationType)rotationType NS_DESIGNATED_INITIALIZER;
+
+// Starts writing WebRTC logs to disk if not already started. Overwrites any
+// existing file(s).
+- (void)start;
+
+// Stops writing WebRTC logs to disk. This method is also called on dealloc.
+- (void)stop;
+
+// Returns the current contents of the logs, or nil if start has been called
+// without a stop.
+- (nullable NSData *)logData;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCFileLogger.mm b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCFileLogger.mm
new file mode 100644
index 0000000000..9562245611
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCFileLogger.mm
@@ -0,0 +1,170 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCFileLogger.h"
+
+#include <memory>
+
+#include "rtc_base/checks.h"
+#include "rtc_base/file_rotating_stream.h"
+#include "rtc_base/log_sinks.h"
+#include "rtc_base/logging.h"
+
+NSString *const kDefaultLogDirName = @"webrtc_logs";
+NSUInteger const kDefaultMaxFileSize = 10 * 1024 * 1024; // 10MB.
+const char *kRTCFileLoggerRotatingLogPrefix = "rotating_log";
+
+@implementation RTC_OBJC_TYPE (RTCFileLogger) {
+  BOOL _hasStarted;
+  NSString *_dirPath;
+  NSUInteger _maxFileSize;
+  std::unique_ptr<rtc::FileRotatingLogSink> _logSink;
+}
+
+@synthesize severity = _severity;
+@synthesize rotationType = _rotationType;
+@synthesize shouldDisableBuffering = _shouldDisableBuffering;
+
+- (instancetype)init {
+  NSArray *paths = NSSearchPathForDirectoriesInDomains(
+      NSDocumentDirectory, NSUserDomainMask, YES);
+  NSString *documentsDirPath = [paths firstObject];
+  NSString *defaultDirPath =
+      [documentsDirPath stringByAppendingPathComponent:kDefaultLogDirName];
+  return [self initWithDirPath:defaultDirPath
+                   maxFileSize:kDefaultMaxFileSize];
+}
+
+- (instancetype)initWithDirPath:(NSString *)dirPath
+                    maxFileSize:(NSUInteger)maxFileSize {
+  return [self initWithDirPath:dirPath
+                   maxFileSize:maxFileSize
+                  rotationType:RTCFileLoggerTypeCall];
+}
+
+- (instancetype)initWithDirPath:(NSString *)dirPath
+                    maxFileSize:(NSUInteger)maxFileSize
+                   rotationType:(RTCFileLoggerRotationType)rotationType {
+  NSParameterAssert(dirPath.length);
+  NSParameterAssert(maxFileSize);
+  if (self = [super init]) {
+    BOOL isDir = NO;
+    NSFileManager *fileManager = [NSFileManager defaultManager];
+    if ([fileManager fileExistsAtPath:dirPath isDirectory:&isDir]) {
+      if (!isDir) {
+        // Bail if something already exists there.
+        return nil;
+      }
+    } else {
+      if (![fileManager createDirectoryAtPath:dirPath
+                  withIntermediateDirectories:NO
+                                   attributes:nil
+                                        error:nil]) {
+        // Bail if we failed to create a directory.
+        return nil;
+      }
+    }
+    _dirPath = dirPath;
+    _maxFileSize = maxFileSize;
+    _severity = RTCFileLoggerSeverityInfo;
+  }
+  return self;
+}
+
+- (void)dealloc {
+  [self stop];
+}
+
+- (void)start {
+  if (_hasStarted) {
+    return;
+  }
+  switch (_rotationType) {
+    case RTCFileLoggerTypeApp:
+      _logSink.reset(
+          new rtc::FileRotatingLogSink(_dirPath.UTF8String,
+                                       kRTCFileLoggerRotatingLogPrefix,
+                                       _maxFileSize,
+                                       _maxFileSize / 10));
+      break;
+    case RTCFileLoggerTypeCall:
+      _logSink.reset(
+          new rtc::CallSessionFileRotatingLogSink(_dirPath.UTF8String,
+                                                  _maxFileSize));
+      break;
+  }
+  if (!_logSink->Init()) {
+    RTC_LOG(LS_ERROR) << "Failed to open log files at path: " << _dirPath.UTF8String;
+    _logSink.reset();
+    return;
+  }
+  if (_shouldDisableBuffering) {
+    _logSink->DisableBuffering();
+  }
+  rtc::LogMessage::LogThreads(true);
+  rtc::LogMessage::LogTimestamps(true);
+  rtc::LogMessage::AddLogToStream(_logSink.get(), [self rtcSeverity]);
+  _hasStarted = YES;
+}
+
+- (void)stop {
+  if (!_hasStarted) {
+    return;
+  }
+  RTC_DCHECK(_logSink);
+  rtc::LogMessage::RemoveLogToStream(_logSink.get());
+  _hasStarted = NO;
+  _logSink.reset();
+}
+
+- (nullable NSData *)logData {
+  if (_hasStarted) {
+    return nil;
+  }
+  NSMutableData* logData = [NSMutableData data];
+  std::unique_ptr<rtc::FileRotatingStreamReader> stream;
+  switch(_rotationType) {
+    case RTCFileLoggerTypeApp:
+      stream = std::make_unique<rtc::FileRotatingStreamReader>(_dirPath.UTF8String,
+                                                               kRTCFileLoggerRotatingLogPrefix);
+      break;
+    case RTCFileLoggerTypeCall:
+      stream = std::make_unique<rtc::CallSessionFileRotatingStreamReader>(_dirPath.UTF8String);
+      break;
+  }
+  size_t bufferSize = stream->GetSize();
+  if (bufferSize == 0) {
+    return logData;
+  }
+  // Allocate memory using malloc so we can pass it direcly to NSData without
+  // copying.
+  std::unique_ptr<uint8_t[]> buffer(static_cast<uint8_t*>(malloc(bufferSize)));
+  size_t read = stream->ReadAll(buffer.get(), bufferSize);
+  logData = [[NSMutableData alloc] initWithBytesNoCopy:buffer.release()
+                                                length:read];
+  return logData;
+}
+
+#pragma mark - Private
+
+- (rtc::LoggingSeverity)rtcSeverity {
+  switch (_severity) {
+    case RTCFileLoggerSeverityVerbose:
+      return rtc::LS_VERBOSE;
+    case RTCFileLoggerSeverityInfo:
+      return rtc::LS_INFO;
+    case RTCFileLoggerSeverityWarning:
+      return rtc::LS_WARNING;
+    case RTCFileLoggerSeverityError:
+      return rtc::LS_ERROR;
+  }
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCIceCandidate+Private.h b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCIceCandidate+Private.h
new file mode 100644
index 0000000000..409e16b608
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCIceCandidate+Private.h
@@ -0,0 +1,36 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCIceCandidate.h"
+
+#include <memory>
+
+#include "api/jsep.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+@interface RTC_OBJC_TYPE (RTCIceCandidate)
+()
+
+    /**
+     * The native IceCandidateInterface representation of this RTCIceCandidate
+     * object. This is needed to pass to the underlying C++ APIs.
+     */
+    @property(nonatomic, readonly) std::unique_ptr<webrtc::IceCandidateInterface> nativeCandidate;
+
+/**
+ * Initialize an RTCIceCandidate from a native IceCandidateInterface. No
+ * ownership is taken of the native candidate.
+ */
+- (instancetype)initWithNativeCandidate:(const webrtc::IceCandidateInterface *)candidate;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCIceCandidate.h b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCIceCandidate.h
new file mode 100644
index 0000000000..f84843af6c
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCIceCandidate.h
@@ -0,0 +1,49 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCIceCandidate) : NSObject
+
+/**
+ * If present, the identifier of the "media stream identification" for the media
+ * component this candidate is associated with.
+ */
+@property(nonatomic, readonly, nullable) NSString *sdpMid;
+
+/**
+ * The index (starting at zero) of the media description this candidate is
+ * associated with in the SDP.
+ */
+@property(nonatomic, readonly) int sdpMLineIndex;
+
+/** The SDP string for this candidate. */
+@property(nonatomic, readonly) NSString *sdp;
+
+/** The URL of the ICE server which this candidate is gathered from. */
+@property(nonatomic, readonly, nullable) NSString *serverUrl;
+
+- (instancetype)init NS_UNAVAILABLE;
+
+/**
+ * Initialize an RTCIceCandidate from SDP.
+ */
+- (instancetype)initWithSdp:(NSString *)sdp
+              sdpMLineIndex:(int)sdpMLineIndex
+                     sdpMid:(nullable NSString *)sdpMid NS_DESIGNATED_INITIALIZER;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCIceCandidate.mm b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCIceCandidate.mm
new file mode 100644
index 0000000000..48385ef5b4
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCIceCandidate.mm
@@ -0,0 +1,76 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCIceCandidate+Private.h"
+
+#include <memory>
+
+#import "base/RTCLogging.h"
+#import "helpers/NSString+StdString.h"
+
+@implementation RTC_OBJC_TYPE (RTCIceCandidate)
+
+@synthesize sdpMid = _sdpMid;
+@synthesize sdpMLineIndex = _sdpMLineIndex;
+@synthesize sdp = _sdp;
+@synthesize serverUrl = _serverUrl;
+
+- (instancetype)initWithSdp:(NSString *)sdp
+              sdpMLineIndex:(int)sdpMLineIndex
+                     sdpMid:(NSString *)sdpMid {
+  NSParameterAssert(sdp.length);
+  if (self = [super init]) {
+    _sdpMid = [sdpMid copy];
+    _sdpMLineIndex = sdpMLineIndex;
+    _sdp = [sdp copy];
+  }
+  return self;
+}
+
+- (NSString *)description {
+  return [NSString stringWithFormat:@"RTC_OBJC_TYPE(RTCIceCandidate):\n%@\n%d\n%@\n%@",
+                                    _sdpMid,
+                                    _sdpMLineIndex,
+                                    _sdp,
+                                    _serverUrl];
+}
+
+#pragma mark - Private
+
+- (instancetype)initWithNativeCandidate:
+    (const webrtc::IceCandidateInterface *)candidate {
+  NSParameterAssert(candidate);
+  std::string sdp;
+  candidate->ToString(&sdp);
+
+  RTC_OBJC_TYPE(RTCIceCandidate) *rtcCandidate =
+      [self initWithSdp:[NSString stringForStdString:sdp]
+          sdpMLineIndex:candidate->sdp_mline_index()
+                 sdpMid:[NSString stringForStdString:candidate->sdp_mid()]];
+  rtcCandidate->_serverUrl = [NSString stringForStdString:candidate->server_url()];
+  return rtcCandidate;
+}
+
+- (std::unique_ptr<webrtc::IceCandidateInterface>)nativeCandidate {
+  webrtc::SdpParseError error;
+
+  webrtc::IceCandidateInterface *candidate = webrtc::CreateIceCandidate(
+      _sdpMid.stdString, _sdpMLineIndex, _sdp.stdString, &error);
+
+  if (!candidate) {
+    RTCLog(@"Failed to create ICE candidate: %s\nline: %s",
+           error.description.c_str(),
+           error.line.c_str());
+  }
+
+  return std::unique_ptr<webrtc::IceCandidateInterface>(candidate);
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCIceCandidateErrorEvent+Private.h b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCIceCandidateErrorEvent+Private.h
new file mode 100644
index 0000000000..8502da08a8
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCIceCandidateErrorEvent+Private.h
@@ -0,0 +1,26 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCIceCandidateErrorEvent.h"
+
+#include <string>
+
+NS_ASSUME_NONNULL_BEGIN
+
+@interface RTC_OBJC_TYPE (RTCIceCandidateErrorEvent)
+()
+
+    - (instancetype)initWithAddress : (const std::string&)address port : (const int)port url
+    : (const std::string&)url errorCode : (const int)errorCode errorText
+    : (const std::string&)errorText;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCIceCandidateErrorEvent.h b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCIceCandidateErrorEvent.h
new file mode 100644
index 0000000000..e0906fdbdd
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCIceCandidateErrorEvent.h
@@ -0,0 +1,42 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCIceCandidateErrorEvent) : NSObject
+
+/** The local IP address used to communicate with the STUN or TURN server. */
+@property(nonatomic, readonly) NSString *address;
+
+/** The port used to communicate with the STUN or TURN server. */
+@property(nonatomic, readonly) int port;
+
+/** The STUN or TURN URL that identifies the STUN or TURN server for which the failure occurred. */
+@property(nonatomic, readonly) NSString *url;
+
+/** The numeric STUN error code returned by the STUN or TURN server. If no host candidate can reach
+ * the server, errorCode will be set to the value 701 which is outside the STUN error code range.
+ * This error is only fired once per server URL while in the RTCIceGatheringState of "gathering". */
+@property(nonatomic, readonly) int errorCode;
+
+/** The STUN reason text returned by the STUN or TURN server. If the server could not be reached,
+ * errorText will be set to an implementation-specific value providing details about the error. */
+@property(nonatomic, readonly) NSString *errorText;
+
+- (instancetype)init NS_DESIGNATED_INITIALIZER;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCIceCandidateErrorEvent.mm b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCIceCandidateErrorEvent.mm
new file mode 100644
index 0000000000..573e30642b
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCIceCandidateErrorEvent.mm
@@ -0,0 +1,42 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCIceCandidateErrorEvent+Private.h"
+
+#import "helpers/NSString+StdString.h"
+
+@implementation RTC_OBJC_TYPE (RTCIceCandidateErrorEvent)
+
+@synthesize address = _address;
+@synthesize port = _port;
+@synthesize url = _url;
+@synthesize errorCode = _errorCode;
+@synthesize errorText = _errorText;
+
+- (instancetype)init {
+  return [super init];
+}
+
+- (instancetype)initWithAddress:(const std::string&)address
+                           port:(const int)port
+                            url:(const std::string&)url
+                      errorCode:(const int)errorCode
+                      errorText:(const std::string&)errorText {
+  if (self = [self init]) {
+    _address = [NSString stringForStdString:address];
+    _port = port;
+    _url = [NSString stringForStdString:url];
+    _errorCode = errorCode;
+    _errorText = [NSString stringForStdString:errorText];
+  }
+  return self;
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCIceServer+Private.h b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCIceServer+Private.h
new file mode 100644
index 0000000000..3eee819965
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCIceServer+Private.h
@@ -0,0 +1,31 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCIceServer.h"
+
+#include "api/peer_connection_interface.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+@interface RTC_OBJC_TYPE (RTCIceServer)
+()
+
+    /**
+     * IceServer struct representation of this RTCIceServer object's data.
+     * This is needed to pass to the underlying C++ APIs.
+     */
+    @property(nonatomic, readonly) webrtc::PeerConnectionInterface::IceServer nativeServer;
+
+/** Initialize an RTCIceServer from a native IceServer. */
+- (instancetype)initWithNativeServer:(webrtc::PeerConnectionInterface::IceServer)nativeServer;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCIceServer.h b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCIceServer.h
new file mode 100644
index 0000000000..7ddcbc1a1f
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCIceServer.h
@@ -0,0 +1,114 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+typedef NS_ENUM(NSUInteger, RTCTlsCertPolicy) {
+  RTCTlsCertPolicySecure,
+  RTCTlsCertPolicyInsecureNoCheck
+};
+
+NS_ASSUME_NONNULL_BEGIN
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCIceServer) : NSObject
+
+/** URI(s) for this server represented as NSStrings. */
+@property(nonatomic, readonly) NSArray<NSString *> *urlStrings;
+
+/** Username to use if this RTCIceServer object is a TURN server. */
+@property(nonatomic, readonly, nullable) NSString *username;
+
+/** Credential to use if this RTCIceServer object is a TURN server. */
+@property(nonatomic, readonly, nullable) NSString *credential;
+
+/**
+ * TLS certificate policy to use if this RTCIceServer object is a TURN server.
+ */
+@property(nonatomic, readonly) RTCTlsCertPolicy tlsCertPolicy;
+
+/**
+  If the URIs in `urls` only contain IP addresses, this field can be used
+  to indicate the hostname, which may be necessary for TLS (using the SNI
+  extension). If `urls` itself contains the hostname, this isn't necessary.
+ */
+@property(nonatomic, readonly, nullable) NSString *hostname;
+
+/** List of protocols to be used in the TLS ALPN extension. */
+@property(nonatomic, readonly) NSArray<NSString *> *tlsAlpnProtocols;
+
+/**
+  List elliptic curves to be used in the TLS elliptic curves extension.
+  Only curve names supported by OpenSSL should be used (eg. "P-256","X25519").
+  */
+@property(nonatomic, readonly) NSArray<NSString *> *tlsEllipticCurves;
+
+- (nonnull instancetype)init NS_UNAVAILABLE;
+
+/** Convenience initializer for a server with no authentication (e.g. STUN). */
+- (instancetype)initWithURLStrings:(NSArray<NSString *> *)urlStrings;
+
+/**
+ * Initialize an RTCIceServer with its associated URLs, optional username,
+ * optional credential, and credentialType.
+ */
+- (instancetype)initWithURLStrings:(NSArray<NSString *> *)urlStrings
+                          username:(nullable NSString *)username
+                        credential:(nullable NSString *)credential;
+
+/**
+ * Initialize an RTCIceServer with its associated URLs, optional username,
+ * optional credential, and TLS cert policy.
+ */
+- (instancetype)initWithURLStrings:(NSArray<NSString *> *)urlStrings
+                          username:(nullable NSString *)username
+                        credential:(nullable NSString *)credential
+                     tlsCertPolicy:(RTCTlsCertPolicy)tlsCertPolicy;
+
+/**
+ * Initialize an RTCIceServer with its associated URLs, optional username,
+ * optional credential, TLS cert policy and hostname.
+ */
+- (instancetype)initWithURLStrings:(NSArray<NSString *> *)urlStrings
+                          username:(nullable NSString *)username
+                        credential:(nullable NSString *)credential
+                     tlsCertPolicy:(RTCTlsCertPolicy)tlsCertPolicy
+                          hostname:(nullable NSString *)hostname;
+
+/**
+ * Initialize an RTCIceServer with its associated URLs, optional username,
+ * optional credential, TLS cert policy, hostname and ALPN protocols.
+ */
+- (instancetype)initWithURLStrings:(NSArray<NSString *> *)urlStrings
+                          username:(nullable NSString *)username
+                        credential:(nullable NSString *)credential
+                     tlsCertPolicy:(RTCTlsCertPolicy)tlsCertPolicy
+                          hostname:(nullable NSString *)hostname
+                  tlsAlpnProtocols:(NSArray<NSString *> *)tlsAlpnProtocols;
+
+/**
+ * Initialize an RTCIceServer with its associated URLs, optional username,
+ * optional credential, TLS cert policy, hostname, ALPN protocols and
+ * elliptic curves.
+ */
+- (instancetype)initWithURLStrings:(NSArray<NSString *> *)urlStrings
+                          username:(nullable NSString *)username
+                        credential:(nullable NSString *)credential
+                     tlsCertPolicy:(RTCTlsCertPolicy)tlsCertPolicy
+                          hostname:(nullable NSString *)hostname
+                  tlsAlpnProtocols:(nullable NSArray<NSString *> *)tlsAlpnProtocols
+                 tlsEllipticCurves:(nullable NSArray<NSString *> *)tlsEllipticCurves
+    NS_DESIGNATED_INITIALIZER;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCIceServer.mm b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCIceServer.mm
new file mode 100644
index 0000000000..19a0a7e9e8
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCIceServer.mm
@@ -0,0 +1,196 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCIceServer+Private.h"
+
+#import "helpers/NSString+StdString.h"
+
+@implementation RTC_OBJC_TYPE (RTCIceServer)
+
+@synthesize urlStrings = _urlStrings;
+@synthesize username = _username;
+@synthesize credential = _credential;
+@synthesize tlsCertPolicy = _tlsCertPolicy;
+@synthesize hostname = _hostname;
+@synthesize tlsAlpnProtocols = _tlsAlpnProtocols;
+@synthesize tlsEllipticCurves = _tlsEllipticCurves;
+
+- (instancetype)initWithURLStrings:(NSArray<NSString *> *)urlStrings {
+  return [self initWithURLStrings:urlStrings
+                         username:nil
+                       credential:nil];
+}
+
+- (instancetype)initWithURLStrings:(NSArray<NSString *> *)urlStrings
+                          username:(NSString *)username
+                        credential:(NSString *)credential {
+  return [self initWithURLStrings:urlStrings
+                         username:username
+                       credential:credential
+                    tlsCertPolicy:RTCTlsCertPolicySecure];
+}
+
+- (instancetype)initWithURLStrings:(NSArray<NSString *> *)urlStrings
+                          username:(NSString *)username
+                        credential:(NSString *)credential
+                     tlsCertPolicy:(RTCTlsCertPolicy)tlsCertPolicy {
+  return [self initWithURLStrings:urlStrings
+                         username:username
+                       credential:credential
+                    tlsCertPolicy:tlsCertPolicy
+                         hostname:nil];
+}
+
+- (instancetype)initWithURLStrings:(NSArray<NSString *> *)urlStrings
+                          username:(NSString *)username
+                        credential:(NSString *)credential
+                     tlsCertPolicy:(RTCTlsCertPolicy)tlsCertPolicy
+                          hostname:(NSString *)hostname {
+  return [self initWithURLStrings:urlStrings
+                         username:username
+                       credential:credential
+                    tlsCertPolicy:tlsCertPolicy
+                         hostname:hostname
+                 tlsAlpnProtocols:[NSArray array]];
+}
+
+- (instancetype)initWithURLStrings:(NSArray<NSString *> *)urlStrings
+                          username:(NSString *)username
+                        credential:(NSString *)credential
+                     tlsCertPolicy:(RTCTlsCertPolicy)tlsCertPolicy
+                          hostname:(NSString *)hostname
+                  tlsAlpnProtocols:(NSArray<NSString *> *)tlsAlpnProtocols {
+  return [self initWithURLStrings:urlStrings
+                         username:username
+                       credential:credential
+                    tlsCertPolicy:tlsCertPolicy
+                         hostname:hostname
+                 tlsAlpnProtocols:tlsAlpnProtocols
+                tlsEllipticCurves:[NSArray array]];
+}
+
+- (instancetype)initWithURLStrings:(NSArray<NSString *> *)urlStrings
+                          username:(NSString *)username
+                        credential:(NSString *)credential
+                     tlsCertPolicy:(RTCTlsCertPolicy)tlsCertPolicy
+                          hostname:(NSString *)hostname
+                  tlsAlpnProtocols:(NSArray<NSString *> *)tlsAlpnProtocols
+                 tlsEllipticCurves:(NSArray<NSString *> *)tlsEllipticCurves {
+  NSParameterAssert(urlStrings.count);
+  if (self = [super init]) {
+    _urlStrings = [[NSArray alloc] initWithArray:urlStrings copyItems:YES];
+    _username = [username copy];
+    _credential = [credential copy];
+    _tlsCertPolicy = tlsCertPolicy;
+    _hostname = [hostname copy];
+    _tlsAlpnProtocols = [[NSArray alloc] initWithArray:tlsAlpnProtocols copyItems:YES];
+    _tlsEllipticCurves = [[NSArray alloc] initWithArray:tlsEllipticCurves copyItems:YES];
+  }
+  return self;
+}
+
+- (NSString *)description {
+  return [NSString stringWithFormat:@"RTC_OBJC_TYPE(RTCIceServer):\n%@\n%@\n%@\n%@\n%@\n%@\n%@",
+                                    _urlStrings,
+                                    _username,
+                                    _credential,
+                                    [self stringForTlsCertPolicy:_tlsCertPolicy],
+                                    _hostname,
+                                    _tlsAlpnProtocols,
+                                    _tlsEllipticCurves];
+}
+
+#pragma mark - Private
+
+- (NSString *)stringForTlsCertPolicy:(RTCTlsCertPolicy)tlsCertPolicy {
+  switch (tlsCertPolicy) {
+    case RTCTlsCertPolicySecure:
+      return @"RTCTlsCertPolicySecure";
+    case RTCTlsCertPolicyInsecureNoCheck:
+      return @"RTCTlsCertPolicyInsecureNoCheck";
+  }
+}
+
+- (webrtc::PeerConnectionInterface::IceServer)nativeServer {
+  __block webrtc::PeerConnectionInterface::IceServer iceServer;
+
+  iceServer.username = [NSString stdStringForString:_username];
+  iceServer.password = [NSString stdStringForString:_credential];
+  iceServer.hostname = [NSString stdStringForString:_hostname];
+
+  [_tlsAlpnProtocols enumerateObjectsUsingBlock:^(NSString *proto, NSUInteger idx, BOOL *stop) {
+    iceServer.tls_alpn_protocols.push_back(proto.stdString);
+  }];
+
+  [_tlsEllipticCurves enumerateObjectsUsingBlock:^(NSString *curve, NSUInteger idx, BOOL *stop) {
+    iceServer.tls_elliptic_curves.push_back(curve.stdString);
+  }];
+
+  [_urlStrings enumerateObjectsUsingBlock:^(NSString *url,
+                                            NSUInteger idx,
+                                            BOOL *stop) {
+    iceServer.urls.push_back(url.stdString);
+  }];
+
+  switch (_tlsCertPolicy) {
+    case RTCTlsCertPolicySecure:
+      iceServer.tls_cert_policy =
+          webrtc::PeerConnectionInterface::kTlsCertPolicySecure;
+      break;
+    case RTCTlsCertPolicyInsecureNoCheck:
+      iceServer.tls_cert_policy =
+          webrtc::PeerConnectionInterface::kTlsCertPolicyInsecureNoCheck;
+      break;
+  }
+  return iceServer;
+}
+
+- (instancetype)initWithNativeServer:
+    (webrtc::PeerConnectionInterface::IceServer)nativeServer {
+  NSMutableArray *urls =
+      [NSMutableArray arrayWithCapacity:nativeServer.urls.size()];
+  for (auto const &url : nativeServer.urls) {
+    [urls addObject:[NSString stringForStdString:url]];
+  }
+  NSString *username = [NSString stringForStdString:nativeServer.username];
+  NSString *credential = [NSString stringForStdString:nativeServer.password];
+  NSString *hostname = [NSString stringForStdString:nativeServer.hostname];
+  NSMutableArray *tlsAlpnProtocols =
+      [NSMutableArray arrayWithCapacity:nativeServer.tls_alpn_protocols.size()];
+  for (auto const &proto : nativeServer.tls_alpn_protocols) {
+    [tlsAlpnProtocols addObject:[NSString stringForStdString:proto]];
+  }
+  NSMutableArray *tlsEllipticCurves =
+      [NSMutableArray arrayWithCapacity:nativeServer.tls_elliptic_curves.size()];
+  for (auto const &curve : nativeServer.tls_elliptic_curves) {
+    [tlsEllipticCurves addObject:[NSString stringForStdString:curve]];
+  }
+  RTCTlsCertPolicy tlsCertPolicy;
+
+  switch (nativeServer.tls_cert_policy) {
+    case webrtc::PeerConnectionInterface::kTlsCertPolicySecure:
+      tlsCertPolicy = RTCTlsCertPolicySecure;
+      break;
+    case webrtc::PeerConnectionInterface::kTlsCertPolicyInsecureNoCheck:
+      tlsCertPolicy = RTCTlsCertPolicyInsecureNoCheck;
+      break;
+  }
+
+  self = [self initWithURLStrings:urls
+                         username:username
+                       credential:credential
+                    tlsCertPolicy:tlsCertPolicy
+                         hostname:hostname
+                 tlsAlpnProtocols:tlsAlpnProtocols
+                tlsEllipticCurves:tlsEllipticCurves];
+  return self;
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCLegacyStatsReport+Private.h b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCLegacyStatsReport+Private.h
new file mode 100644
index 0000000000..7374b2b72f
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCLegacyStatsReport+Private.h
@@ -0,0 +1,25 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCLegacyStatsReport.h"
+
+#include "api/legacy_stats_types.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+@interface RTC_OBJC_TYPE (RTCLegacyStatsReport)
+()
+
+    /** Initialize an RTCLegacyStatsReport object from a native StatsReport. */
+    - (instancetype)initWithNativeReport : (const webrtc::StatsReport &)nativeReport;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCLegacyStatsReport.h b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCLegacyStatsReport.h
new file mode 100644
index 0000000000..b3bd12c5d7
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCLegacyStatsReport.h
@@ -0,0 +1,37 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+/** This does not currently conform to the spec. */
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCLegacyStatsReport) : NSObject
+
+/** Time since 1970-01-01T00:00:00Z in milliseconds. */
+@property(nonatomic, readonly) CFTimeInterval timestamp;
+
+/** The type of stats held by this object. */
+@property(nonatomic, readonly) NSString *type;
+
+/** The identifier for this object. */
+@property(nonatomic, readonly) NSString *reportId;
+
+/** A dictionary holding the actual stats. */
+@property(nonatomic, readonly) NSDictionary<NSString *, NSString *> *values;
+
+- (instancetype)init NS_UNAVAILABLE;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCLegacyStatsReport.mm b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCLegacyStatsReport.mm
new file mode 100644
index 0000000000..bd7a1ad9c9
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCLegacyStatsReport.mm
@@ -0,0 +1,60 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCLegacyStatsReport+Private.h"
+
+#import "base/RTCLogging.h"
+#import "helpers/NSString+StdString.h"
+
+#include "rtc_base/checks.h"
+
+@implementation RTC_OBJC_TYPE (RTCLegacyStatsReport)
+
+@synthesize timestamp = _timestamp;
+@synthesize type = _type;
+@synthesize reportId = _reportId;
+@synthesize values = _values;
+
+- (NSString *)description {
+  return [NSString stringWithFormat:@"RTC_OBJC_TYPE(RTCLegacyStatsReport):\n%@\n%@\n%f\n%@",
+                                    _reportId,
+                                    _type,
+                                    _timestamp,
+                                    _values];
+}
+
+#pragma mark - Private
+
+- (instancetype)initWithNativeReport:(const webrtc::StatsReport &)nativeReport {
+  if (self = [super init]) {
+    _timestamp = nativeReport.timestamp();
+    _type = [NSString stringForStdString:nativeReport.TypeToString()];
+    _reportId = [NSString stringForStdString:
+        nativeReport.id()->ToString()];
+
+    NSUInteger capacity = nativeReport.values().size();
+    NSMutableDictionary *values =
+        [NSMutableDictionary dictionaryWithCapacity:capacity];
+    for (auto const &valuePair : nativeReport.values()) {
+      NSString *key = [NSString stringForStdString:
+          valuePair.second->display_name()];
+      NSString *value = [NSString stringForStdString:
+          valuePair.second->ToString()];
+
+      // Not expecting duplicate keys.
+      RTC_DCHECK(![values objectForKey:key]);
+      [values setObject:value forKey:key];
+    }
+    _values = values;
+  }
+  return self;
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCMediaConstraints+Private.h b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCMediaConstraints+Private.h
new file mode 100644
index 0000000000..97eee8307d
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCMediaConstraints+Private.h
@@ -0,0 +1,34 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCMediaConstraints.h"
+
+#include <memory>
+
+#include "sdk/media_constraints.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+@interface RTC_OBJC_TYPE (RTCMediaConstraints)
+()
+
+    /**
+     * A MediaConstraints representation of this RTCMediaConstraints object. This is
+     * needed to pass to the underlying C++ APIs.
+     */
+    - (std::unique_ptr<webrtc::MediaConstraints>)nativeConstraints;
+
+/** Return a native Constraints object representing these constraints */
++ (webrtc::MediaConstraints::Constraints)nativeConstraintsForConstraints:
+    (NSDictionary<NSString*, NSString*>*)constraints;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCMediaConstraints.h b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCMediaConstraints.h
new file mode 100644
index 0000000000..c5baf20c1d
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCMediaConstraints.h
@@ -0,0 +1,46 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+/** Constraint keys for media sources. */
+/** The value for this key should be a base64 encoded string containing
+ *  the data from the serialized configuration proto.
+ */
+RTC_EXTERN NSString *const kRTCMediaConstraintsAudioNetworkAdaptorConfig;
+
+/** Constraint keys for generating offers and answers. */
+RTC_EXTERN NSString *const kRTCMediaConstraintsIceRestart;
+RTC_EXTERN NSString *const kRTCMediaConstraintsOfferToReceiveAudio;
+RTC_EXTERN NSString *const kRTCMediaConstraintsOfferToReceiveVideo;
+RTC_EXTERN NSString *const kRTCMediaConstraintsVoiceActivityDetection;
+
+/** Constraint values for Boolean parameters. */
+RTC_EXTERN NSString *const kRTCMediaConstraintsValueTrue;
+RTC_EXTERN NSString *const kRTCMediaConstraintsValueFalse;
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCMediaConstraints) : NSObject
+
+- (instancetype)init NS_UNAVAILABLE;
+
+/** Initialize with mandatory and/or optional constraints. */
+- (instancetype)
+    initWithMandatoryConstraints:(nullable NSDictionary<NSString *, NSString *> *)mandatory
+             optionalConstraints:(nullable NSDictionary<NSString *, NSString *> *)optional
+    NS_DESIGNATED_INITIALIZER;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCMediaConstraints.mm b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCMediaConstraints.mm
new file mode 100644
index 0000000000..0f46e4b8fe
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCMediaConstraints.mm
@@ -0,0 +1,90 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCMediaConstraints+Private.h"
+
+#import "helpers/NSString+StdString.h"
+
+#include <memory>
+
+NSString *const kRTCMediaConstraintsAudioNetworkAdaptorConfig =
+    @(webrtc::MediaConstraints::kAudioNetworkAdaptorConfig);
+
+NSString *const kRTCMediaConstraintsIceRestart = @(webrtc::MediaConstraints::kIceRestart);
+NSString *const kRTCMediaConstraintsOfferToReceiveAudio =
+    @(webrtc::MediaConstraints::kOfferToReceiveAudio);
+NSString *const kRTCMediaConstraintsOfferToReceiveVideo =
+    @(webrtc::MediaConstraints::kOfferToReceiveVideo);
+NSString *const kRTCMediaConstraintsVoiceActivityDetection =
+    @(webrtc::MediaConstraints::kVoiceActivityDetection);
+
+NSString *const kRTCMediaConstraintsValueTrue = @(webrtc::MediaConstraints::kValueTrue);
+NSString *const kRTCMediaConstraintsValueFalse = @(webrtc::MediaConstraints::kValueFalse);
+
+@implementation RTC_OBJC_TYPE (RTCMediaConstraints) {
+  NSDictionary<NSString *, NSString *> *_mandatory;
+  NSDictionary<NSString *, NSString *> *_optional;
+}
+
+- (instancetype)initWithMandatoryConstraints:
+    (NSDictionary<NSString *, NSString *> *)mandatory
+                         optionalConstraints:
+    (NSDictionary<NSString *, NSString *> *)optional {
+  if (self = [super init]) {
+    _mandatory = [[NSDictionary alloc] initWithDictionary:mandatory
+                                                copyItems:YES];
+    _optional = [[NSDictionary alloc] initWithDictionary:optional
+                                               copyItems:YES];
+  }
+  return self;
+}
+
+- (NSString *)description {
+  return [NSString
+      stringWithFormat:@"RTC_OBJC_TYPE(RTCMediaConstraints):\n%@\n%@", _mandatory, _optional];
+}
+
+#pragma mark - Private
+
+- (std::unique_ptr<webrtc::MediaConstraints>)nativeConstraints {
+  webrtc::MediaConstraints::Constraints mandatory =
+      [[self class] nativeConstraintsForConstraints:_mandatory];
+  webrtc::MediaConstraints::Constraints optional =
+      [[self class] nativeConstraintsForConstraints:_optional];
+
+  webrtc::MediaConstraints *nativeConstraints =
+      new webrtc::MediaConstraints(mandatory, optional);
+  return std::unique_ptr<webrtc::MediaConstraints>(nativeConstraints);
+}
+
++ (webrtc::MediaConstraints::Constraints)nativeConstraintsForConstraints:
+    (NSDictionary<NSString *, NSString *> *)constraints {
+  webrtc::MediaConstraints::Constraints nativeConstraints;
+  for (NSString *key in constraints) {
+    NSAssert([key isKindOfClass:[NSString class]],
+             @"%@ is not an NSString.", key);
+    NSString *value = [constraints objectForKey:key];
+    NSAssert([value isKindOfClass:[NSString class]],
+             @"%@ is not an NSString.", value);
+    if ([kRTCMediaConstraintsAudioNetworkAdaptorConfig isEqualToString:key]) {
+      // This value is base64 encoded.
+      NSData *charData = [[NSData alloc] initWithBase64EncodedString:value options:0];
+      std::string configValue =
+          std::string(reinterpret_cast<const char *>(charData.bytes), charData.length);
+      nativeConstraints.push_back(webrtc::MediaConstraints::Constraint(key.stdString, configValue));
+    } else {
+      nativeConstraints.push_back(
+          webrtc::MediaConstraints::Constraint(key.stdString, value.stdString));
+    }
+  }
+  return nativeConstraints;
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCMediaSource+Private.h b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCMediaSource+Private.h
new file mode 100644
index 0000000000..edda892e50
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCMediaSource+Private.h
@@ -0,0 +1,42 @@
+/*
+ *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCMediaSource.h"
+
+#include "api/media_stream_interface.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+@class RTC_OBJC_TYPE(RTCPeerConnectionFactory);
+
+typedef NS_ENUM(NSInteger, RTCMediaSourceType) {
+  RTCMediaSourceTypeAudio,
+  RTCMediaSourceTypeVideo,
+};
+
+@interface RTC_OBJC_TYPE (RTCMediaSource)
+()
+
+    @property(nonatomic,
+              readonly) rtc::scoped_refptr<webrtc::MediaSourceInterface> nativeMediaSource;
+
+- (instancetype)initWithFactory:(RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory
+              nativeMediaSource:(rtc::scoped_refptr<webrtc::MediaSourceInterface>)nativeMediaSource
+                           type:(RTCMediaSourceType)type NS_DESIGNATED_INITIALIZER;
+
++ (webrtc::MediaSourceInterface::SourceState)nativeSourceStateForState:(RTCSourceState)state;
+
++ (RTCSourceState)sourceStateForNativeState:(webrtc::MediaSourceInterface::SourceState)nativeState;
+
++ (NSString *)stringForState:(RTCSourceState)state;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCMediaSource.h b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCMediaSource.h
new file mode 100644
index 0000000000..ba19c2a352
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCMediaSource.h
@@ -0,0 +1,34 @@
+/*
+ *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+typedef NS_ENUM(NSInteger, RTCSourceState) {
+  RTCSourceStateInitializing,
+  RTCSourceStateLive,
+  RTCSourceStateEnded,
+  RTCSourceStateMuted,
+};
+
+NS_ASSUME_NONNULL_BEGIN
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCMediaSource) : NSObject
+
+/** The current state of the RTCMediaSource. */
+@property(nonatomic, readonly) RTCSourceState state;
+
+- (instancetype)init NS_UNAVAILABLE;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCMediaSource.mm b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCMediaSource.mm
new file mode 100644
index 0000000000..61472a782a
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCMediaSource.mm
@@ -0,0 +1,82 @@
+/*
+ *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCMediaSource+Private.h"
+
+#include "rtc_base/checks.h"
+
+@implementation RTC_OBJC_TYPE (RTCMediaSource) {
+  RTC_OBJC_TYPE(RTCPeerConnectionFactory) * _factory;
+  RTCMediaSourceType _type;
+}
+
+@synthesize nativeMediaSource = _nativeMediaSource;
+
+- (instancetype)initWithFactory:(RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory
+              nativeMediaSource:(rtc::scoped_refptr<webrtc::MediaSourceInterface>)nativeMediaSource
+                           type:(RTCMediaSourceType)type {
+  RTC_DCHECK(factory);
+  RTC_DCHECK(nativeMediaSource);
+  if (self = [super init]) {
+    _factory = factory;
+    _nativeMediaSource = nativeMediaSource;
+    _type = type;
+  }
+  return self;
+}
+
+- (RTCSourceState)state {
+  return [[self class] sourceStateForNativeState:_nativeMediaSource->state()];
+}
+
+#pragma mark - Private
+
++ (webrtc::MediaSourceInterface::SourceState)nativeSourceStateForState:
+    (RTCSourceState)state {
+  switch (state) {
+    case RTCSourceStateInitializing:
+      return webrtc::MediaSourceInterface::kInitializing;
+    case RTCSourceStateLive:
+      return webrtc::MediaSourceInterface::kLive;
+    case RTCSourceStateEnded:
+      return webrtc::MediaSourceInterface::kEnded;
+    case RTCSourceStateMuted:
+      return webrtc::MediaSourceInterface::kMuted;
+  }
+}
+
++ (RTCSourceState)sourceStateForNativeState:
+    (webrtc::MediaSourceInterface::SourceState)nativeState {
+  switch (nativeState) {
+    case webrtc::MediaSourceInterface::kInitializing:
+      return RTCSourceStateInitializing;
+    case webrtc::MediaSourceInterface::kLive:
+      return RTCSourceStateLive;
+    case webrtc::MediaSourceInterface::kEnded:
+      return RTCSourceStateEnded;
+    case webrtc::MediaSourceInterface::kMuted:
+      return RTCSourceStateMuted;
+  }
+}
+
++ (NSString *)stringForState:(RTCSourceState)state {
+  switch (state) {
+    case RTCSourceStateInitializing:
+      return @"Initializing";
+    case RTCSourceStateLive:
+      return @"Live";
+    case RTCSourceStateEnded:
+      return @"Ended";
+    case RTCSourceStateMuted:
+      return @"Muted";
+  }
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCMediaStream+Private.h b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCMediaStream+Private.h
new file mode 100644
index 0000000000..6c8a602766
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCMediaStream+Private.h
@@ -0,0 +1,37 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCMediaStream.h"
+
+#include "api/media_stream_interface.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+@interface RTC_OBJC_TYPE (RTCMediaStream)
+()
+
+    /**
+     * MediaStreamInterface representation of this RTCMediaStream object. This is
+     * needed to pass to the underlying C++ APIs.
+     */
+    @property(nonatomic,
+              readonly) rtc::scoped_refptr<webrtc::MediaStreamInterface> nativeMediaStream;
+
+/** Initialize an RTCMediaStream with an id. */
+- (instancetype)initWithFactory:(RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory
+                       streamId:(NSString *)streamId;
+
+/** Initialize an RTCMediaStream from a native MediaStreamInterface. */
+- (instancetype)initWithFactory:(RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory
+              nativeMediaStream:(rtc::scoped_refptr<webrtc::MediaStreamInterface>)nativeMediaStream;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCMediaStream.h b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCMediaStream.h
new file mode 100644
index 0000000000..2d56f15c7d
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCMediaStream.h
@@ -0,0 +1,49 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+@class RTC_OBJC_TYPE(RTCAudioTrack);
+@class RTC_OBJC_TYPE(RTCPeerConnectionFactory);
+@class RTC_OBJC_TYPE(RTCVideoTrack);
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCMediaStream) : NSObject
+
+/** The audio tracks in this stream. */
+@property(nonatomic, strong, readonly) NSArray<RTC_OBJC_TYPE(RTCAudioTrack) *> *audioTracks;
+
+/** The video tracks in this stream. */
+@property(nonatomic, strong, readonly) NSArray<RTC_OBJC_TYPE(RTCVideoTrack) *> *videoTracks;
+
+/** An identifier for this media stream. */
+@property(nonatomic, readonly) NSString *streamId;
+
+- (instancetype)init NS_UNAVAILABLE;
+
+/** Adds the given audio track to this media stream. */
+- (void)addAudioTrack:(RTC_OBJC_TYPE(RTCAudioTrack) *)audioTrack;
+
+/** Adds the given video track to this media stream. */
+- (void)addVideoTrack:(RTC_OBJC_TYPE(RTCVideoTrack) *)videoTrack;
+
+/** Removes the given audio track to this media stream. */
+- (void)removeAudioTrack:(RTC_OBJC_TYPE(RTCAudioTrack) *)audioTrack;
+
+/** Removes the given video track to this media stream. */
+- (void)removeVideoTrack:(RTC_OBJC_TYPE(RTCVideoTrack) *)videoTrack;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCMediaStream.mm b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCMediaStream.mm
new file mode 100644
index 0000000000..0018dd6945
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCMediaStream.mm
@@ -0,0 +1,155 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCMediaStream+Private.h"
+
+#import "RTCAudioTrack+Private.h"
+#import "RTCMediaStreamTrack+Private.h"
+#import "RTCPeerConnectionFactory+Private.h"
+#import "RTCVideoTrack+Private.h"
+#import "helpers/NSString+StdString.h"
+
+@implementation RTC_OBJC_TYPE (RTCMediaStream) {
+  RTC_OBJC_TYPE(RTCPeerConnectionFactory) * _factory;
+  rtc::Thread *_signalingThread;
+  NSMutableArray *_audioTracks /* accessed on _signalingThread */;
+  NSMutableArray *_videoTracks /* accessed on _signalingThread */;
+  rtc::scoped_refptr<webrtc::MediaStreamInterface> _nativeMediaStream;
+}
+
+- (instancetype)initWithFactory:(RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory
+                       streamId:(NSString *)streamId {
+  NSParameterAssert(factory);
+  NSParameterAssert(streamId.length);
+  std::string nativeId = [NSString stdStringForString:streamId];
+  rtc::scoped_refptr<webrtc::MediaStreamInterface> stream =
+      factory.nativeFactory->CreateLocalMediaStream(nativeId);
+  return [self initWithFactory:factory nativeMediaStream:stream];
+}
+
+- (NSArray<RTC_OBJC_TYPE(RTCAudioTrack) *> *)audioTracks {
+  if (!_signalingThread->IsCurrent()) {
+    return _signalingThread->BlockingCall([self]() { return self.audioTracks; });
+  }
+  return [_audioTracks copy];
+}
+
+- (NSArray<RTC_OBJC_TYPE(RTCVideoTrack) *> *)videoTracks {
+  if (!_signalingThread->IsCurrent()) {
+    return _signalingThread->BlockingCall([self]() { return self.videoTracks; });
+  }
+  return [_videoTracks copy];
+}
+
+- (NSString *)streamId {
+  return [NSString stringForStdString:_nativeMediaStream->id()];
+}
+
+- (void)addAudioTrack:(RTC_OBJC_TYPE(RTCAudioTrack) *)audioTrack {
+  if (!_signalingThread->IsCurrent()) {
+    return _signalingThread->BlockingCall(
+        [audioTrack, self]() { return [self addAudioTrack:audioTrack]; });
+  }
+  if (_nativeMediaStream->AddTrack(audioTrack.nativeAudioTrack)) {
+    [_audioTracks addObject:audioTrack];
+  }
+}
+
+- (void)addVideoTrack:(RTC_OBJC_TYPE(RTCVideoTrack) *)videoTrack {
+  if (!_signalingThread->IsCurrent()) {
+    return _signalingThread->BlockingCall(
+        [videoTrack, self]() { return [self addVideoTrack:videoTrack]; });
+  }
+  if (_nativeMediaStream->AddTrack(videoTrack.nativeVideoTrack)) {
+    [_videoTracks addObject:videoTrack];
+  }
+}
+
+- (void)removeAudioTrack:(RTC_OBJC_TYPE(RTCAudioTrack) *)audioTrack {
+  if (!_signalingThread->IsCurrent()) {
+    return _signalingThread->BlockingCall(
+        [audioTrack, self]() { return [self removeAudioTrack:audioTrack]; });
+  }
+  NSUInteger index = [_audioTracks indexOfObjectIdenticalTo:audioTrack];
+  if (index == NSNotFound) {
+    RTC_LOG(LS_INFO) << "|removeAudioTrack| called on unexpected RTC_OBJC_TYPE(RTCAudioTrack)";
+    return;
+  }
+  if (_nativeMediaStream->RemoveTrack(audioTrack.nativeAudioTrack)) {
+    [_audioTracks removeObjectAtIndex:index];
+  }
+}
+
+- (void)removeVideoTrack:(RTC_OBJC_TYPE(RTCVideoTrack) *)videoTrack {
+  if (!_signalingThread->IsCurrent()) {
+    return _signalingThread->BlockingCall(
+        [videoTrack, self]() { return [self removeVideoTrack:videoTrack]; });
+  }
+  NSUInteger index = [_videoTracks indexOfObjectIdenticalTo:videoTrack];
+  if (index == NSNotFound) {
+    RTC_LOG(LS_INFO) << "|removeVideoTrack| called on unexpected RTC_OBJC_TYPE(RTCVideoTrack)";
+    return;
+  }
+
+  if (_nativeMediaStream->RemoveTrack(videoTrack.nativeVideoTrack)) {
+    [_videoTracks removeObjectAtIndex:index];
+  }
+}
+
+- (NSString *)description {
+  return [NSString stringWithFormat:@"RTC_OBJC_TYPE(RTCMediaStream):\n%@\nA=%lu\nV=%lu",
+                                    self.streamId,
+                                    (unsigned long)self.audioTracks.count,
+                                    (unsigned long)self.videoTracks.count];
+}
+
+#pragma mark - Private
+
+- (rtc::scoped_refptr<webrtc::MediaStreamInterface>)nativeMediaStream {
+  return _nativeMediaStream;
+}
+
+- (instancetype)initWithFactory:(RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory
+              nativeMediaStream:
+                  (rtc::scoped_refptr<webrtc::MediaStreamInterface>)nativeMediaStream {
+  NSParameterAssert(nativeMediaStream);
+  if (self = [super init]) {
+    _factory = factory;
+    _signalingThread = factory.signalingThread;
+
+    webrtc::AudioTrackVector audioTracks = nativeMediaStream->GetAudioTracks();
+    webrtc::VideoTrackVector videoTracks = nativeMediaStream->GetVideoTracks();
+
+    _audioTracks = [NSMutableArray arrayWithCapacity:audioTracks.size()];
+    _videoTracks = [NSMutableArray arrayWithCapacity:videoTracks.size()];
+    _nativeMediaStream = nativeMediaStream;
+
+    for (auto &track : audioTracks) {
+      RTCMediaStreamTrackType type = RTCMediaStreamTrackTypeAudio;
+      RTC_OBJC_TYPE(RTCAudioTrack) *audioTrack =
+          [[RTC_OBJC_TYPE(RTCAudioTrack) alloc] initWithFactory:_factory
+                                                    nativeTrack:track
+                                                           type:type];
+      [_audioTracks addObject:audioTrack];
+    }
+
+    for (auto &track : videoTracks) {
+      RTCMediaStreamTrackType type = RTCMediaStreamTrackTypeVideo;
+      RTC_OBJC_TYPE(RTCVideoTrack) *videoTrack =
+          [[RTC_OBJC_TYPE(RTCVideoTrack) alloc] initWithFactory:_factory
+                                                    nativeTrack:track
+                                                           type:type];
+      [_videoTracks addObject:videoTrack];
+    }
+  }
+  return self;
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCMediaStreamTrack+Private.h b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCMediaStreamTrack+Private.h
new file mode 100644
index 0000000000..ee51e27b2d
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCMediaStreamTrack+Private.h
@@ -0,0 +1,62 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCMediaStreamTrack.h"
+
+#include "api/media_stream_interface.h"
+
+typedef NS_ENUM(NSInteger, RTCMediaStreamTrackType) {
+  RTCMediaStreamTrackTypeAudio,
+  RTCMediaStreamTrackTypeVideo,
+};
+
+NS_ASSUME_NONNULL_BEGIN
+
+@class RTC_OBJC_TYPE(RTCPeerConnectionFactory);
+
+@interface RTC_OBJC_TYPE (RTCMediaStreamTrack)
+()
+
+        @property(nonatomic, readonly) RTC_OBJC_TYPE(RTCPeerConnectionFactory) *
+    factory;
+
+/**
+ * The native MediaStreamTrackInterface passed in or created during
+ * construction.
+ */
+@property(nonatomic, readonly) rtc::scoped_refptr<webrtc::MediaStreamTrackInterface> nativeTrack;
+
+/**
+ * Initialize an RTCMediaStreamTrack from a native MediaStreamTrackInterface.
+ */
+- (instancetype)initWithFactory:(RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory
+                    nativeTrack:(rtc::scoped_refptr<webrtc::MediaStreamTrackInterface>)nativeTrack
+                           type:(RTCMediaStreamTrackType)type NS_DESIGNATED_INITIALIZER;
+
+- (instancetype)initWithFactory:(RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory
+                    nativeTrack:(rtc::scoped_refptr<webrtc::MediaStreamTrackInterface>)nativeTrack;
+
+- (BOOL)isEqualToTrack:(RTC_OBJC_TYPE(RTCMediaStreamTrack) *)track;
+
++ (webrtc::MediaStreamTrackInterface::TrackState)nativeTrackStateForState:
+        (RTCMediaStreamTrackState)state;
+
++ (RTCMediaStreamTrackState)trackStateForNativeState:
+        (webrtc::MediaStreamTrackInterface::TrackState)nativeState;
+
++ (NSString *)stringForState:(RTCMediaStreamTrackState)state;
+
++ (RTC_OBJC_TYPE(RTCMediaStreamTrack) *)
+    mediaTrackForNativeTrack:(rtc::scoped_refptr<webrtc::MediaStreamTrackInterface>)nativeTrack
+                     factory:(RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCMediaStreamTrack.h b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCMediaStreamTrack.h
new file mode 100644
index 0000000000..2200122ccd
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCMediaStreamTrack.h
@@ -0,0 +1,50 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+/**
+ * Represents the state of the track. This exposes the same states in C++.
+ */
+typedef NS_ENUM(NSInteger, RTCMediaStreamTrackState) {
+  RTCMediaStreamTrackStateLive,
+  RTCMediaStreamTrackStateEnded
+};
+
+NS_ASSUME_NONNULL_BEGIN
+
+RTC_EXTERN NSString *const kRTCMediaStreamTrackKindAudio;
+RTC_EXTERN NSString *const kRTCMediaStreamTrackKindVideo;
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCMediaStreamTrack) : NSObject
+
+/**
+ * The kind of track. For example, "audio" if this track represents an audio
+ * track and "video" if this track represents a video track.
+ */
+@property(nonatomic, readonly) NSString *kind;
+
+/** An identifier string. */
+@property(nonatomic, readonly) NSString *trackId;
+
+/** The enabled state of the track. */
+@property(nonatomic, assign) BOOL isEnabled;
+
+/** The state of the track. */
+@property(nonatomic, readonly) RTCMediaStreamTrackState readyState;
+
+- (instancetype)init NS_UNAVAILABLE;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCMediaStreamTrack.mm b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCMediaStreamTrack.mm
new file mode 100644
index 0000000000..f1e128ca60
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCMediaStreamTrack.mm
@@ -0,0 +1,161 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCAudioTrack+Private.h"
+#import "RTCMediaStreamTrack+Private.h"
+#import "RTCVideoTrack+Private.h"
+
+#import "helpers/NSString+StdString.h"
+
+NSString * const kRTCMediaStreamTrackKindAudio =
+    @(webrtc::MediaStreamTrackInterface::kAudioKind);
+NSString * const kRTCMediaStreamTrackKindVideo =
+    @(webrtc::MediaStreamTrackInterface::kVideoKind);
+
+@implementation RTC_OBJC_TYPE (RTCMediaStreamTrack) {
+  RTC_OBJC_TYPE(RTCPeerConnectionFactory) * _factory;
+  rtc::scoped_refptr<webrtc::MediaStreamTrackInterface> _nativeTrack;
+  RTCMediaStreamTrackType _type;
+}
+
+- (NSString *)kind {
+  return [NSString stringForStdString:_nativeTrack->kind()];
+}
+
+- (NSString *)trackId {
+  return [NSString stringForStdString:_nativeTrack->id()];
+}
+
+- (BOOL)isEnabled {
+  return _nativeTrack->enabled();
+}
+
+- (void)setIsEnabled:(BOOL)isEnabled {
+  _nativeTrack->set_enabled(isEnabled);
+}
+
+- (RTCMediaStreamTrackState)readyState {
+  return [[self class] trackStateForNativeState:_nativeTrack->state()];
+}
+
+- (NSString *)description {
+  NSString *readyState = [[self class] stringForState:self.readyState];
+  return [NSString stringWithFormat:@"RTC_OBJC_TYPE(RTCMediaStreamTrack):\n%@\n%@\n%@\n%@",
+                                    self.kind,
+                                    self.trackId,
+                                    self.isEnabled ? @"enabled" : @"disabled",
+                                    readyState];
+}
+
+- (BOOL)isEqual:(id)object {
+  if (self == object) {
+    return YES;
+  }
+  if (![object isMemberOfClass:[self class]]) {
+    return NO;
+  }
+  return [self isEqualToTrack:(RTC_OBJC_TYPE(RTCMediaStreamTrack) *)object];
+}
+
+- (NSUInteger)hash {
+  return (NSUInteger)_nativeTrack.get();
+}
+
+#pragma mark - Private
+
+- (rtc::scoped_refptr<webrtc::MediaStreamTrackInterface>)nativeTrack {
+  return _nativeTrack;
+}
+
+@synthesize factory = _factory;
+
+- (instancetype)initWithFactory:(RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory
+                    nativeTrack:(rtc::scoped_refptr<webrtc::MediaStreamTrackInterface>)nativeTrack
+                           type:(RTCMediaStreamTrackType)type {
+  NSParameterAssert(nativeTrack);
+  NSParameterAssert(factory);
+  if (self = [super init]) {
+    _factory = factory;
+    _nativeTrack = nativeTrack;
+    _type = type;
+  }
+  return self;
+}
+
+- (instancetype)initWithFactory:(RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory
+                    nativeTrack:(rtc::scoped_refptr<webrtc::MediaStreamTrackInterface>)nativeTrack {
+  NSParameterAssert(nativeTrack);
+  if (nativeTrack->kind() ==
+      std::string(webrtc::MediaStreamTrackInterface::kAudioKind)) {
+    return [self initWithFactory:factory nativeTrack:nativeTrack type:RTCMediaStreamTrackTypeAudio];
+  }
+  if (nativeTrack->kind() ==
+      std::string(webrtc::MediaStreamTrackInterface::kVideoKind)) {
+    return [self initWithFactory:factory nativeTrack:nativeTrack type:RTCMediaStreamTrackTypeVideo];
+  }
+  return nil;
+}
+
+- (BOOL)isEqualToTrack:(RTC_OBJC_TYPE(RTCMediaStreamTrack) *)track {
+  if (!track) {
+    return NO;
+  }
+  return _nativeTrack == track.nativeTrack;
+}
+
++ (webrtc::MediaStreamTrackInterface::TrackState)nativeTrackStateForState:
+    (RTCMediaStreamTrackState)state {
+  switch (state) {
+    case RTCMediaStreamTrackStateLive:
+      return webrtc::MediaStreamTrackInterface::kLive;
+    case RTCMediaStreamTrackStateEnded:
+      return webrtc::MediaStreamTrackInterface::kEnded;
+  }
+}
+
++ (RTCMediaStreamTrackState)trackStateForNativeState:
+    (webrtc::MediaStreamTrackInterface::TrackState)nativeState {
+  switch (nativeState) {
+    case webrtc::MediaStreamTrackInterface::kLive:
+      return RTCMediaStreamTrackStateLive;
+    case webrtc::MediaStreamTrackInterface::kEnded:
+      return RTCMediaStreamTrackStateEnded;
+  }
+}
+
++ (NSString *)stringForState:(RTCMediaStreamTrackState)state {
+  switch (state) {
+    case RTCMediaStreamTrackStateLive:
+      return @"Live";
+    case RTCMediaStreamTrackStateEnded:
+      return @"Ended";
+  }
+}
+
++ (RTC_OBJC_TYPE(RTCMediaStreamTrack) *)
+    mediaTrackForNativeTrack:(rtc::scoped_refptr<webrtc::MediaStreamTrackInterface>)nativeTrack
+                     factory:(RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory {
+  NSParameterAssert(nativeTrack);
+  NSParameterAssert(factory);
+  if (nativeTrack->kind() == webrtc::MediaStreamTrackInterface::kAudioKind) {
+    return [[RTC_OBJC_TYPE(RTCAudioTrack) alloc] initWithFactory:factory
+                                                     nativeTrack:nativeTrack
+                                                            type:RTCMediaStreamTrackTypeAudio];
+  } else if (nativeTrack->kind() == webrtc::MediaStreamTrackInterface::kVideoKind) {
+    return [[RTC_OBJC_TYPE(RTCVideoTrack) alloc] initWithFactory:factory
+                                                     nativeTrack:nativeTrack
+                                                            type:RTCMediaStreamTrackTypeVideo];
+  } else {
+    return [[RTC_OBJC_TYPE(RTCMediaStreamTrack) alloc] initWithFactory:factory
+                                                           nativeTrack:nativeTrack];
+  }
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCMetrics.h b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCMetrics.h
new file mode 100644
index 0000000000..fddbb27c90
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCMetrics.h
@@ -0,0 +1,23 @@
+/*
+ *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+#import "RTCMetricsSampleInfo.h"
+
+/**
+ * Enables gathering of metrics (which can be fetched with
+ * RTCGetAndResetMetrics). Must be called before any other call into WebRTC.
+ */
+RTC_EXTERN void RTCEnableMetrics(void);
+
+/** Gets and clears native histograms. */
+RTC_EXTERN NSArray<RTC_OBJC_TYPE(RTCMetricsSampleInfo) *>* RTCGetAndResetMetrics(void);
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCMetrics.mm b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCMetrics.mm
new file mode 100644
index 0000000000..87eb8c0210
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCMetrics.mm
@@ -0,0 +1,34 @@
+/*
+ *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCMetrics.h"
+
+#import "RTCMetricsSampleInfo+Private.h"
+
+#include "rtc_base/string_utils.h"
+
+void RTCEnableMetrics(void) {
+  webrtc::metrics::Enable();
+}
+
+NSArray<RTC_OBJC_TYPE(RTCMetricsSampleInfo) *> *RTCGetAndResetMetrics(void) {
+  std::map<std::string, std::unique_ptr<webrtc::metrics::SampleInfo>, rtc::AbslStringViewCmp>
+      histograms;
+  webrtc::metrics::GetAndReset(&histograms);
+
+  NSMutableArray *metrics =
+      [NSMutableArray arrayWithCapacity:histograms.size()];
+  for (auto const &histogram : histograms) {
+    RTC_OBJC_TYPE(RTCMetricsSampleInfo) *metric =
+        [[RTC_OBJC_TYPE(RTCMetricsSampleInfo) alloc] initWithNativeSampleInfo:*histogram.second];
+    [metrics addObject:metric];
+  }
+  return metrics;
+}
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCMetricsSampleInfo+Private.h b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCMetricsSampleInfo+Private.h
new file mode 100644
index 0000000000..e4aa41f6c7
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCMetricsSampleInfo+Private.h
@@ -0,0 +1,25 @@
+/*
+ *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCMetricsSampleInfo.h"
+
+#include "system_wrappers/include/metrics.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+@interface RTC_OBJC_TYPE (RTCMetricsSampleInfo)
+()
+
+    /** Initialize an RTCMetricsSampleInfo object from native SampleInfo. */
+    - (instancetype)initWithNativeSampleInfo : (const webrtc::metrics::SampleInfo &)info;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCMetricsSampleInfo.h b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCMetricsSampleInfo.h
new file mode 100644
index 0000000000..47a877b6fb
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCMetricsSampleInfo.h
@@ -0,0 +1,48 @@
+/*
+ *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCMetricsSampleInfo) : NSObject
+
+/**
+ * Example of RTCMetricsSampleInfo:
+ * name: "WebRTC.Video.InputFramesPerSecond"
+ * min: 1
+ * max: 100
+ * bucketCount: 50
+ * samples: [29]:2 [30]:1
+ */
+
+/** The name of the histogram. */
+@property(nonatomic, readonly) NSString *name;
+
+/** The minimum bucket value. */
+@property(nonatomic, readonly) int min;
+
+/** The maximum bucket value. */
+@property(nonatomic, readonly) int max;
+
+/** The number of buckets. */
+@property(nonatomic, readonly) int bucketCount;
+
+/** A dictionary holding the samples <value, # of events>. */
+@property(nonatomic, readonly) NSDictionary<NSNumber *, NSNumber *> *samples;
+
+- (instancetype)init NS_UNAVAILABLE;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCMetricsSampleInfo.mm b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCMetricsSampleInfo.mm
new file mode 100644
index 0000000000..e4be94e90a
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCMetricsSampleInfo.mm
@@ -0,0 +1,43 @@
+/*
+ *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCMetricsSampleInfo+Private.h"
+
+#import "helpers/NSString+StdString.h"
+
+@implementation RTC_OBJC_TYPE (RTCMetricsSampleInfo)
+
+@synthesize name = _name;
+@synthesize min = _min;
+@synthesize max = _max;
+@synthesize bucketCount = _bucketCount;
+@synthesize samples = _samples;
+
+#pragma mark - Private
+
+- (instancetype)initWithNativeSampleInfo:
+    (const webrtc::metrics::SampleInfo &)info {
+  if (self = [super init]) {
+    _name = [NSString stringForStdString:info.name];
+    _min = info.min;
+    _max = info.max;
+    _bucketCount = info.bucket_count;
+
+    NSMutableDictionary *samples =
+        [NSMutableDictionary dictionaryWithCapacity:info.samples.size()];
+    for (auto const &sample : info.samples) {
+      [samples setObject:@(sample.second) forKey:@(sample.first)];
+    }
+    _samples = samples;
+  }
+  return self;
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCPeerConnection+DataChannel.mm b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCPeerConnection+DataChannel.mm
new file mode 100644
index 0000000000..cb75f061d8
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCPeerConnection+DataChannel.mm
@@ -0,0 +1,34 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCPeerConnection+Private.h"
+
+#import "RTCDataChannel+Private.h"
+#import "RTCDataChannelConfiguration+Private.h"
+#import "helpers/NSString+StdString.h"
+
+@implementation RTC_OBJC_TYPE (RTCPeerConnection)
+(DataChannel)
+
+    - (nullable RTC_OBJC_TYPE(RTCDataChannel) *)dataChannelForLabel
+    : (NSString *)label configuration
+    : (RTC_OBJC_TYPE(RTCDataChannelConfiguration) *)configuration {
+  std::string labelString = [NSString stdStringForString:label];
+  const webrtc::DataChannelInit nativeInit =
+      configuration.nativeDataChannelInit;
+  auto result = self.nativePeerConnection->CreateDataChannelOrError(labelString, &nativeInit);
+  if (!result.ok()) {
+    return nil;
+  }
+  return [[RTC_OBJC_TYPE(RTCDataChannel) alloc] initWithFactory:self.factory
+                                              nativeDataChannel:result.MoveValue()];
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCPeerConnection+Private.h b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCPeerConnection+Private.h
new file mode 100644
index 0000000000..00f2ef7834
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCPeerConnection+Private.h
@@ -0,0 +1,143 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCPeerConnection.h"
+
+#include "api/peer_connection_interface.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+namespace webrtc {
+
+/**
+ * These objects are created by RTCPeerConnectionFactory to wrap an
+ * id<RTCPeerConnectionDelegate> and call methods on that interface.
+ */
+class PeerConnectionDelegateAdapter : public PeerConnectionObserver {
+ public:
+  PeerConnectionDelegateAdapter(RTC_OBJC_TYPE(RTCPeerConnection) * peerConnection);
+  ~PeerConnectionDelegateAdapter() override;
+
+  void OnSignalingChange(PeerConnectionInterface::SignalingState new_state) override;
+
+  void OnAddStream(rtc::scoped_refptr<MediaStreamInterface> stream) override;
+
+  void OnRemoveStream(rtc::scoped_refptr<MediaStreamInterface> stream) override;
+
+  void OnTrack(rtc::scoped_refptr<RtpTransceiverInterface> transceiver) override;
+
+  void OnDataChannel(rtc::scoped_refptr<DataChannelInterface> data_channel) override;
+
+  void OnRenegotiationNeeded() override;
+
+  void OnIceConnectionChange(PeerConnectionInterface::IceConnectionState new_state) override;
+
+  void OnStandardizedIceConnectionChange(
+      PeerConnectionInterface::IceConnectionState new_state) override;
+
+  void OnConnectionChange(PeerConnectionInterface::PeerConnectionState new_state) override;
+
+  void OnIceGatheringChange(PeerConnectionInterface::IceGatheringState new_state) override;
+
+  void OnIceCandidate(const IceCandidateInterface *candidate) override;
+
+  void OnIceCandidateError(const std::string &address,
+                           int port,
+                           const std::string &url,
+                           int error_code,
+                           const std::string &error_text) override;
+
+  void OnIceCandidatesRemoved(const std::vector<cricket::Candidate> &candidates) override;
+
+  void OnIceSelectedCandidatePairChanged(const cricket::CandidatePairChangeEvent &event) override;
+
+  void OnAddTrack(rtc::scoped_refptr<RtpReceiverInterface> receiver,
+                  const std::vector<rtc::scoped_refptr<MediaStreamInterface>> &streams) override;
+
+  void OnRemoveTrack(rtc::scoped_refptr<RtpReceiverInterface> receiver) override;
+
+ private:
+  __weak RTC_OBJC_TYPE(RTCPeerConnection) * peer_connection_;
+};
+
+}  // namespace webrtc
+@protocol RTC_OBJC_TYPE
+(RTCSSLCertificateVerifier);
+
+@interface RTC_OBJC_TYPE (RTCPeerConnection)
+()
+
+    /** The factory used to create this RTCPeerConnection */
+    @property(nonatomic, readonly) RTC_OBJC_TYPE(RTCPeerConnectionFactory) *
+    factory;
+
+/** The native PeerConnectionInterface created during construction. */
+@property(nonatomic, readonly) rtc::scoped_refptr<webrtc::PeerConnectionInterface>
+    nativePeerConnection;
+
+/** Initialize an RTCPeerConnection with a configuration, constraints, and
+ *  delegate.
+ */
+- (nullable instancetype)
+        initWithFactory:(RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory
+          configuration:(RTC_OBJC_TYPE(RTCConfiguration) *)configuration
+            constraints:(RTC_OBJC_TYPE(RTCMediaConstraints) *)constraints
+    certificateVerifier:(nullable id<RTC_OBJC_TYPE(RTCSSLCertificateVerifier)>)certificateVerifier
+               delegate:(nullable id<RTC_OBJC_TYPE(RTCPeerConnectionDelegate)>)delegate;
+
+/** Initialize an RTCPeerConnection with a configuration, constraints,
+ *  delegate and PeerConnectionDependencies.
+ */
+- (nullable instancetype)
+    initWithDependencies:(RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory
+           configuration:(RTC_OBJC_TYPE(RTCConfiguration) *)configuration
+             constraints:(RTC_OBJC_TYPE(RTCMediaConstraints) *)constraints
+            dependencies:(std::unique_ptr<webrtc::PeerConnectionDependencies>)dependencies
+                delegate:(nullable id<RTC_OBJC_TYPE(RTCPeerConnectionDelegate)>)delegate
+    NS_DESIGNATED_INITIALIZER;
+
++ (webrtc::PeerConnectionInterface::SignalingState)nativeSignalingStateForState:
+        (RTCSignalingState)state;
+
++ (RTCSignalingState)signalingStateForNativeState:
+        (webrtc::PeerConnectionInterface::SignalingState)nativeState;
+
++ (NSString *)stringForSignalingState:(RTCSignalingState)state;
+
++ (webrtc::PeerConnectionInterface::IceConnectionState)nativeIceConnectionStateForState:
+        (RTCIceConnectionState)state;
+
++ (webrtc::PeerConnectionInterface::PeerConnectionState)nativeConnectionStateForState:
+        (RTCPeerConnectionState)state;
+
++ (RTCIceConnectionState)iceConnectionStateForNativeState:
+        (webrtc::PeerConnectionInterface::IceConnectionState)nativeState;
+
++ (RTCPeerConnectionState)connectionStateForNativeState:
+        (webrtc::PeerConnectionInterface::PeerConnectionState)nativeState;
+
++ (NSString *)stringForIceConnectionState:(RTCIceConnectionState)state;
+
++ (NSString *)stringForConnectionState:(RTCPeerConnectionState)state;
+
++ (webrtc::PeerConnectionInterface::IceGatheringState)nativeIceGatheringStateForState:
+        (RTCIceGatheringState)state;
+
++ (RTCIceGatheringState)iceGatheringStateForNativeState:
+        (webrtc::PeerConnectionInterface::IceGatheringState)nativeState;
+
++ (NSString *)stringForIceGatheringState:(RTCIceGatheringState)state;
+
++ (webrtc::PeerConnectionInterface::StatsOutputLevel)nativeStatsOutputLevelForLevel:
+        (RTCStatsOutputLevel)level;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCPeerConnection+Stats.mm b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCPeerConnection+Stats.mm
new file mode 100644
index 0000000000..f8d38143f3
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCPeerConnection+Stats.mm
@@ -0,0 +1,102 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCPeerConnection+Private.h"
+
+#import "RTCLegacyStatsReport+Private.h"
+#import "RTCMediaStreamTrack+Private.h"
+#import "RTCRtpReceiver+Private.h"
+#import "RTCRtpSender+Private.h"
+#import "RTCStatisticsReport+Private.h"
+#import "helpers/NSString+StdString.h"
+
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+class StatsCollectorCallbackAdapter : public RTCStatsCollectorCallback {
+ public:
+  StatsCollectorCallbackAdapter(RTCStatisticsCompletionHandler completion_handler)
+      : completion_handler_(completion_handler) {}
+
+  void OnStatsDelivered(const rtc::scoped_refptr<const RTCStatsReport> &report) override {
+    RTC_DCHECK(completion_handler_);
+    RTC_OBJC_TYPE(RTCStatisticsReport) *statisticsReport =
+        [[RTC_OBJC_TYPE(RTCStatisticsReport) alloc] initWithReport:*report];
+    completion_handler_(statisticsReport);
+    completion_handler_ = nil;
+  }
+
+ private:
+  RTCStatisticsCompletionHandler completion_handler_;
+};
+
+class StatsObserverAdapter : public StatsObserver {
+ public:
+  StatsObserverAdapter(
+      void (^completionHandler)(NSArray<RTC_OBJC_TYPE(RTCLegacyStatsReport) *> *stats)) {
+    completion_handler_ = completionHandler;
+  }
+
+  ~StatsObserverAdapter() override { completion_handler_ = nil; }
+
+  void OnComplete(const StatsReports& reports) override {
+    RTC_DCHECK(completion_handler_);
+    NSMutableArray *stats = [NSMutableArray arrayWithCapacity:reports.size()];
+    for (const auto* report : reports) {
+      RTC_OBJC_TYPE(RTCLegacyStatsReport) *statsReport =
+          [[RTC_OBJC_TYPE(RTCLegacyStatsReport) alloc] initWithNativeReport:*report];
+      [stats addObject:statsReport];
+    }
+    completion_handler_(stats);
+    completion_handler_ = nil;
+  }
+
+ private:
+  void (^completion_handler_)(NSArray<RTC_OBJC_TYPE(RTCLegacyStatsReport) *> *stats);
+};
+}  // namespace webrtc
+
+@implementation RTC_OBJC_TYPE (RTCPeerConnection)
+(Stats)
+
+    - (void)statisticsForSender : (RTC_OBJC_TYPE(RTCRtpSender) *)sender completionHandler
+    : (RTCStatisticsCompletionHandler)completionHandler {
+  rtc::scoped_refptr<webrtc::StatsCollectorCallbackAdapter> collector =
+      rtc::make_ref_counted<webrtc::StatsCollectorCallbackAdapter>(completionHandler);
+  self.nativePeerConnection->GetStats(sender.nativeRtpSender, collector);
+}
+
+- (void)statisticsForReceiver:(RTC_OBJC_TYPE(RTCRtpReceiver) *)receiver
+            completionHandler:(RTCStatisticsCompletionHandler)completionHandler {
+  rtc::scoped_refptr<webrtc::StatsCollectorCallbackAdapter> collector =
+      rtc::make_ref_counted<webrtc::StatsCollectorCallbackAdapter>(completionHandler);
+  self.nativePeerConnection->GetStats(receiver.nativeRtpReceiver, collector);
+}
+
+- (void)statisticsWithCompletionHandler:(RTCStatisticsCompletionHandler)completionHandler {
+  rtc::scoped_refptr<webrtc::StatsCollectorCallbackAdapter> collector =
+      rtc::make_ref_counted<webrtc::StatsCollectorCallbackAdapter>(completionHandler);
+  self.nativePeerConnection->GetStats(collector.get());
+}
+
+- (void)statsForTrack:(RTC_OBJC_TYPE(RTCMediaStreamTrack) *)mediaStreamTrack
+     statsOutputLevel:(RTCStatsOutputLevel)statsOutputLevel
+    completionHandler:
+        (void (^)(NSArray<RTC_OBJC_TYPE(RTCLegacyStatsReport) *> *stats))completionHandler {
+  rtc::scoped_refptr<webrtc::StatsObserverAdapter> observer =
+      rtc::make_ref_counted<webrtc::StatsObserverAdapter>(completionHandler);
+  webrtc::PeerConnectionInterface::StatsOutputLevel nativeOutputLevel =
+      [[self class] nativeStatsOutputLevelForLevel:statsOutputLevel];
+  self.nativePeerConnection->GetStats(
+      observer.get(), mediaStreamTrack.nativeTrack.get(), nativeOutputLevel);
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCPeerConnection.h b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCPeerConnection.h
new file mode 100644
index 0000000000..55af6868fd
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCPeerConnection.h
@@ -0,0 +1,398 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+@class RTC_OBJC_TYPE(RTCConfiguration);
+@class RTC_OBJC_TYPE(RTCDataChannel);
+@class RTC_OBJC_TYPE(RTCDataChannelConfiguration);
+@class RTC_OBJC_TYPE(RTCIceCandidate);
+@class RTC_OBJC_TYPE(RTCIceCandidateErrorEvent);
+@class RTC_OBJC_TYPE(RTCMediaConstraints);
+@class RTC_OBJC_TYPE(RTCMediaStream);
+@class RTC_OBJC_TYPE(RTCMediaStreamTrack);
+@class RTC_OBJC_TYPE(RTCPeerConnectionFactory);
+@class RTC_OBJC_TYPE(RTCRtpReceiver);
+@class RTC_OBJC_TYPE(RTCRtpSender);
+@class RTC_OBJC_TYPE(RTCRtpTransceiver);
+@class RTC_OBJC_TYPE(RTCRtpTransceiverInit);
+@class RTC_OBJC_TYPE(RTCSessionDescription);
+@class RTC_OBJC_TYPE(RTCStatisticsReport);
+@class RTC_OBJC_TYPE(RTCLegacyStatsReport);
+
+typedef NS_ENUM(NSInteger, RTCRtpMediaType);
+
+NS_ASSUME_NONNULL_BEGIN
+
+extern NSString *const kRTCPeerConnectionErrorDomain;
+extern int const kRTCSessionDescriptionErrorCode;
+
+/** Represents the signaling state of the peer connection. */
+typedef NS_ENUM(NSInteger, RTCSignalingState) {
+  RTCSignalingStateStable,
+  RTCSignalingStateHaveLocalOffer,
+  RTCSignalingStateHaveLocalPrAnswer,
+  RTCSignalingStateHaveRemoteOffer,
+  RTCSignalingStateHaveRemotePrAnswer,
+  // Not an actual state, represents the total number of states.
+  RTCSignalingStateClosed,
+};
+
+/** Represents the ice connection state of the peer connection. */
+typedef NS_ENUM(NSInteger, RTCIceConnectionState) {
+  RTCIceConnectionStateNew,
+  RTCIceConnectionStateChecking,
+  RTCIceConnectionStateConnected,
+  RTCIceConnectionStateCompleted,
+  RTCIceConnectionStateFailed,
+  RTCIceConnectionStateDisconnected,
+  RTCIceConnectionStateClosed,
+  RTCIceConnectionStateCount,
+};
+
+/** Represents the combined ice+dtls connection state of the peer connection. */
+typedef NS_ENUM(NSInteger, RTCPeerConnectionState) {
+  RTCPeerConnectionStateNew,
+  RTCPeerConnectionStateConnecting,
+  RTCPeerConnectionStateConnected,
+  RTCPeerConnectionStateDisconnected,
+  RTCPeerConnectionStateFailed,
+  RTCPeerConnectionStateClosed,
+};
+
+/** Represents the ice gathering state of the peer connection. */
+typedef NS_ENUM(NSInteger, RTCIceGatheringState) {
+  RTCIceGatheringStateNew,
+  RTCIceGatheringStateGathering,
+  RTCIceGatheringStateComplete,
+};
+
+/** Represents the stats output level. */
+typedef NS_ENUM(NSInteger, RTCStatsOutputLevel) {
+  RTCStatsOutputLevelStandard,
+  RTCStatsOutputLevelDebug,
+};
+
+typedef void (^RTCCreateSessionDescriptionCompletionHandler)(RTC_OBJC_TYPE(RTCSessionDescription) *
+                                                                 _Nullable sdp,
+                                                             NSError *_Nullable error);
+
+typedef void (^RTCSetSessionDescriptionCompletionHandler)(NSError *_Nullable error);
+
+@class RTC_OBJC_TYPE(RTCPeerConnection);
+
+RTC_OBJC_EXPORT
+@protocol RTC_OBJC_TYPE
+(RTCPeerConnectionDelegate)<NSObject>
+
+    /** Called when the SignalingState changed. */
+    - (void)peerConnection
+    : (RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection didChangeSignalingState
+    : (RTCSignalingState)stateChanged;
+
+/** Called when media is received on a new stream from remote peer. */
+- (void)peerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection
+          didAddStream:(RTC_OBJC_TYPE(RTCMediaStream) *)stream;
+
+/** Called when a remote peer closes a stream.
+ *  This is not called when RTCSdpSemanticsUnifiedPlan is specified.
+ */
+- (void)peerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection
+       didRemoveStream:(RTC_OBJC_TYPE(RTCMediaStream) *)stream;
+
+/** Called when negotiation is needed, for example ICE has restarted. */
+- (void)peerConnectionShouldNegotiate:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection;
+
+/** Called any time the IceConnectionState changes. */
+- (void)peerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection
+    didChangeIceConnectionState:(RTCIceConnectionState)newState;
+
+/** Called any time the IceGatheringState changes. */
+- (void)peerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection
+    didChangeIceGatheringState:(RTCIceGatheringState)newState;
+
+/** New ice candidate has been found. */
+- (void)peerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection
+    didGenerateIceCandidate:(RTC_OBJC_TYPE(RTCIceCandidate) *)candidate;
+
+/** Called when a group of local Ice candidates have been removed. */
+- (void)peerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection
+    didRemoveIceCandidates:(NSArray<RTC_OBJC_TYPE(RTCIceCandidate) *> *)candidates;
+
+/** New data channel has been opened. */
+- (void)peerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection
+    didOpenDataChannel:(RTC_OBJC_TYPE(RTCDataChannel) *)dataChannel;
+
+/** Called when signaling indicates a transceiver will be receiving media from
+ *  the remote endpoint.
+ *  This is only called with RTCSdpSemanticsUnifiedPlan specified.
+ */
+@optional
+/** Called any time the IceConnectionState changes following standardized
+ * transition. */
+- (void)peerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection
+    didChangeStandardizedIceConnectionState:(RTCIceConnectionState)newState;
+
+/** Called any time the PeerConnectionState changes. */
+- (void)peerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection
+    didChangeConnectionState:(RTCPeerConnectionState)newState;
+
+- (void)peerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection
+    didStartReceivingOnTransceiver:(RTC_OBJC_TYPE(RTCRtpTransceiver) *)transceiver;
+
+/** Called when a receiver and its track are created. */
+- (void)peerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection
+        didAddReceiver:(RTC_OBJC_TYPE(RTCRtpReceiver) *)rtpReceiver
+               streams:(NSArray<RTC_OBJC_TYPE(RTCMediaStream) *> *)mediaStreams;
+
+/** Called when the receiver and its track are removed. */
+- (void)peerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection
+     didRemoveReceiver:(RTC_OBJC_TYPE(RTCRtpReceiver) *)rtpReceiver;
+
+/** Called when the selected ICE candidate pair is changed. */
+- (void)peerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection
+    didChangeLocalCandidate:(RTC_OBJC_TYPE(RTCIceCandidate) *)local
+            remoteCandidate:(RTC_OBJC_TYPE(RTCIceCandidate) *)remote
+             lastReceivedMs:(int)lastDataReceivedMs
+               changeReason:(NSString *)reason;
+
+/** Called when gathering of an ICE candidate failed. */
+- (void)peerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)peerConnection
+    didFailToGatherIceCandidate:(RTC_OBJC_TYPE(RTCIceCandidateErrorEvent) *)event;
+
+@end
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCPeerConnection) : NSObject
+
+/** The object that will be notifed about events such as state changes and
+ *  streams being added or removed.
+ */
+@property(nonatomic, weak, nullable) id<RTC_OBJC_TYPE(RTCPeerConnectionDelegate)> delegate;
+/** This property is not available with RTCSdpSemanticsUnifiedPlan. Please use
+ *  `senders` instead.
+ */
+@property(nonatomic, readonly) NSArray<RTC_OBJC_TYPE(RTCMediaStream) *> *localStreams;
+@property(nonatomic, readonly, nullable) RTC_OBJC_TYPE(RTCSessionDescription) * localDescription;
+@property(nonatomic, readonly, nullable) RTC_OBJC_TYPE(RTCSessionDescription) * remoteDescription;
+@property(nonatomic, readonly) RTCSignalingState signalingState;
+@property(nonatomic, readonly) RTCIceConnectionState iceConnectionState;
+@property(nonatomic, readonly) RTCPeerConnectionState connectionState;
+@property(nonatomic, readonly) RTCIceGatheringState iceGatheringState;
+@property(nonatomic, readonly, copy) RTC_OBJC_TYPE(RTCConfiguration) * configuration;
+
+/** Gets all RTCRtpSenders associated with this peer connection.
+ *  Note: reading this property returns different instances of RTCRtpSender.
+ *  Use isEqual: instead of == to compare RTCRtpSender instances.
+ */
+@property(nonatomic, readonly) NSArray<RTC_OBJC_TYPE(RTCRtpSender) *> *senders;
+
+/** Gets all RTCRtpReceivers associated with this peer connection.
+ *  Note: reading this property returns different instances of RTCRtpReceiver.
+ *  Use isEqual: instead of == to compare RTCRtpReceiver instances.
+ */
+@property(nonatomic, readonly) NSArray<RTC_OBJC_TYPE(RTCRtpReceiver) *> *receivers;
+
+/** Gets all RTCRtpTransceivers associated with this peer connection.
+ *  Note: reading this property returns different instances of
+ *  RTCRtpTransceiver. Use isEqual: instead of == to compare
+ *  RTCRtpTransceiver instances. This is only available with
+ * RTCSdpSemanticsUnifiedPlan specified.
+ */
+@property(nonatomic, readonly) NSArray<RTC_OBJC_TYPE(RTCRtpTransceiver) *> *transceivers;
+
+- (instancetype)init NS_UNAVAILABLE;
+
+/** Sets the PeerConnection's global configuration to `configuration`.
+ *  Any changes to STUN/TURN servers or ICE candidate policy will affect the
+ *  next gathering phase, and cause the next call to createOffer to generate
+ *  new ICE credentials. Note that the BUNDLE and RTCP-multiplexing policies
+ *  cannot be changed with this method.
+ */
+- (BOOL)setConfiguration:(RTC_OBJC_TYPE(RTCConfiguration) *)configuration;
+
+/** Terminate all media and close the transport. */
+- (void)close;
+
+/** Provide a remote candidate to the ICE Agent. */
+- (void)addIceCandidate:(RTC_OBJC_TYPE(RTCIceCandidate) *)candidate
+    DEPRECATED_MSG_ATTRIBUTE("Please use addIceCandidate:completionHandler: instead");
+
+/** Provide a remote candidate to the ICE Agent. */
+- (void)addIceCandidate:(RTC_OBJC_TYPE(RTCIceCandidate) *)candidate
+      completionHandler:(void (^)(NSError *_Nullable error))completionHandler;
+
+/** Remove a group of remote candidates from the ICE Agent. */
+- (void)removeIceCandidates:(NSArray<RTC_OBJC_TYPE(RTCIceCandidate) *> *)candidates;
+
+/** Add a new media stream to be sent on this peer connection.
+ *  This method is not supported with RTCSdpSemanticsUnifiedPlan. Please use
+ *  addTrack instead.
+ */
+- (void)addStream:(RTC_OBJC_TYPE(RTCMediaStream) *)stream;
+
+/** Remove the given media stream from this peer connection.
+ *  This method is not supported with RTCSdpSemanticsUnifiedPlan. Please use
+ *  removeTrack instead.
+ */
+- (void)removeStream:(RTC_OBJC_TYPE(RTCMediaStream) *)stream;
+
+/** Add a new media stream track to be sent on this peer connection, and return
+ *  the newly created RTCRtpSender. The RTCRtpSender will be
+ * associated with the streams specified in the `streamIds` list.
+ *
+ *  Errors: If an error occurs, returns nil. An error can occur if:
+ *  - A sender already exists for the track.
+ *  - The peer connection is closed.
+ */
+- (nullable RTC_OBJC_TYPE(RTCRtpSender) *)addTrack:(RTC_OBJC_TYPE(RTCMediaStreamTrack) *)track
+                                         streamIds:(NSArray<NSString *> *)streamIds;
+
+/** With PlanB semantics, removes an RTCRtpSender from this peer connection.
+ *
+ *  With UnifiedPlan semantics, sets sender's track to null and removes the
+ *  send component from the associated RTCRtpTransceiver's direction.
+ *
+ *  Returns YES on success.
+ */
+- (BOOL)removeTrack:(RTC_OBJC_TYPE(RTCRtpSender) *)sender;
+
+/** addTransceiver creates a new RTCRtpTransceiver and adds it to the set of
+ *  transceivers. Adding a transceiver will cause future calls to CreateOffer
+ *  to add a media description for the corresponding transceiver.
+ *
+ *  The initial value of `mid` in the returned transceiver is nil. Setting a
+ *  new session description may change it to a non-nil value.
+ *
+ *  https://w3c.github.io/webrtc-pc/#dom-rtcpeerconnection-addtransceiver
+ *
+ *  Optionally, an RtpTransceiverInit structure can be specified to configure
+ *  the transceiver from construction. If not specified, the transceiver will
+ *  default to having a direction of kSendRecv and not be part of any streams.
+ *
+ *  These methods are only available when Unified Plan is enabled (see
+ *  RTCConfiguration).
+ */
+
+/** Adds a transceiver with a sender set to transmit the given track. The kind
+ *  of the transceiver (and sender/receiver) will be derived from the kind of
+ *  the track.
+ */
+- (nullable RTC_OBJC_TYPE(RTCRtpTransceiver) *)addTransceiverWithTrack:
+    (RTC_OBJC_TYPE(RTCMediaStreamTrack) *)track;
+- (nullable RTC_OBJC_TYPE(RTCRtpTransceiver) *)
+    addTransceiverWithTrack:(RTC_OBJC_TYPE(RTCMediaStreamTrack) *)track
+                       init:(RTC_OBJC_TYPE(RTCRtpTransceiverInit) *)init;
+
+/** Adds a transceiver with the given kind. Can either be RTCRtpMediaTypeAudio
+ *  or RTCRtpMediaTypeVideo.
+ */
+- (nullable RTC_OBJC_TYPE(RTCRtpTransceiver) *)addTransceiverOfType:(RTCRtpMediaType)mediaType;
+- (nullable RTC_OBJC_TYPE(RTCRtpTransceiver) *)
+    addTransceiverOfType:(RTCRtpMediaType)mediaType
+                    init:(RTC_OBJC_TYPE(RTCRtpTransceiverInit) *)init;
+
+/** Tells the PeerConnection that ICE should be restarted. This triggers a need
+ * for negotiation and subsequent offerForConstraints:completionHandler call will act as if
+ * RTCOfferAnswerOptions::ice_restart is true.
+ */
+- (void)restartIce;
+
+/** Generate an SDP offer. */
+- (void)offerForConstraints:(RTC_OBJC_TYPE(RTCMediaConstraints) *)constraints
+          completionHandler:(RTCCreateSessionDescriptionCompletionHandler)completionHandler;
+
+/** Generate an SDP answer. */
+- (void)answerForConstraints:(RTC_OBJC_TYPE(RTCMediaConstraints) *)constraints
+           completionHandler:(RTCCreateSessionDescriptionCompletionHandler)completionHandler;
+
+/** Apply the supplied RTCSessionDescription as the local description. */
+- (void)setLocalDescription:(RTC_OBJC_TYPE(RTCSessionDescription) *)sdp
+          completionHandler:(RTCSetSessionDescriptionCompletionHandler)completionHandler;
+
+/** Creates an offer or answer (depending on current signaling state) and sets
+ * it as the local session description. */
+- (void)setLocalDescriptionWithCompletionHandler:
+    (RTCSetSessionDescriptionCompletionHandler)completionHandler;
+
+/** Apply the supplied RTCSessionDescription as the remote description. */
+- (void)setRemoteDescription:(RTC_OBJC_TYPE(RTCSessionDescription) *)sdp
+           completionHandler:(RTCSetSessionDescriptionCompletionHandler)completionHandler;
+
+/** Limits the bandwidth allocated for all RTP streams sent by this
+ *  PeerConnection. Nil parameters will be unchanged. Setting
+ * `currentBitrateBps` will force the available bitrate estimate to the given
+ *  value. Returns YES if the parameters were successfully updated.
+ */
+- (BOOL)setBweMinBitrateBps:(nullable NSNumber *)minBitrateBps
+          currentBitrateBps:(nullable NSNumber *)currentBitrateBps
+              maxBitrateBps:(nullable NSNumber *)maxBitrateBps;
+
+/** Start or stop recording an Rtc EventLog. */
+- (BOOL)startRtcEventLogWithFilePath:(NSString *)filePath maxSizeInBytes:(int64_t)maxSizeInBytes;
+- (void)stopRtcEventLog;
+
+@end
+
+@interface RTC_OBJC_TYPE (RTCPeerConnection)
+(Media)
+
+    /** Create an RTCRtpSender with the specified kind and media stream ID.
+     *  See RTCMediaStreamTrack.h for available kinds.
+     *  This method is not supported with RTCSdpSemanticsUnifiedPlan. Please use
+     *  addTransceiver instead.
+     */
+    - (RTC_OBJC_TYPE(RTCRtpSender) *)senderWithKind : (NSString *)kind streamId
+    : (NSString *)streamId;
+
+@end
+
+@interface RTC_OBJC_TYPE (RTCPeerConnection)
+(DataChannel)
+
+    /** Create a new data channel with the given label and configuration. */
+    - (nullable RTC_OBJC_TYPE(RTCDataChannel) *)dataChannelForLabel
+    : (NSString *)label configuration : (RTC_OBJC_TYPE(RTCDataChannelConfiguration) *)configuration;
+
+@end
+
+typedef void (^RTCStatisticsCompletionHandler)(RTC_OBJC_TYPE(RTCStatisticsReport) *);
+
+@interface RTC_OBJC_TYPE (RTCPeerConnection)
+(Stats)
+
+    /** Gather stats for the given RTCMediaStreamTrack. If `mediaStreamTrack` is nil
+     *  statistics are gathered for all tracks.
+     */
+    - (void)statsForTrack
+    : (nullable RTC_OBJC_TYPE(RTCMediaStreamTrack) *)mediaStreamTrack statsOutputLevel
+    : (RTCStatsOutputLevel)statsOutputLevel completionHandler
+    : (nullable void (^)(NSArray<RTC_OBJC_TYPE(RTCLegacyStatsReport) *> *stats))completionHandler;
+
+/** Gather statistic through the v2 statistics API. */
+- (void)statisticsWithCompletionHandler:(RTCStatisticsCompletionHandler)completionHandler;
+
+/** Spec-compliant getStats() performing the stats selection algorithm with the
+ *  sender.
+ */
+- (void)statisticsForSender:(RTC_OBJC_TYPE(RTCRtpSender) *)sender
+          completionHandler:(RTCStatisticsCompletionHandler)completionHandler;
+
+/** Spec-compliant getStats() performing the stats selection algorithm with the
+ *  receiver.
+ */
+- (void)statisticsForReceiver:(RTC_OBJC_TYPE(RTCRtpReceiver) *)receiver
+            completionHandler:(RTCStatisticsCompletionHandler)completionHandler;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCPeerConnection.mm b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCPeerConnection.mm
new file mode 100644
index 0000000000..df99030111
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCPeerConnection.mm
@@ -0,0 +1,935 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCPeerConnection+Private.h"
+
+#import "RTCConfiguration+Private.h"
+#import "RTCDataChannel+Private.h"
+#import "RTCIceCandidate+Private.h"
+#import "RTCIceCandidateErrorEvent+Private.h"
+#import "RTCLegacyStatsReport+Private.h"
+#import "RTCMediaConstraints+Private.h"
+#import "RTCMediaStream+Private.h"
+#import "RTCMediaStreamTrack+Private.h"
+#import "RTCPeerConnectionFactory+Private.h"
+#import "RTCRtpReceiver+Private.h"
+#import "RTCRtpSender+Private.h"
+#import "RTCRtpTransceiver+Private.h"
+#import "RTCSessionDescription+Private.h"
+#import "base/RTCLogging.h"
+#import "helpers/NSString+StdString.h"
+
+#include <memory>
+
+#include "api/jsep_ice_candidate.h"
+#include "api/rtc_event_log_output_file.h"
+#include "api/set_local_description_observer_interface.h"
+#include "api/set_remote_description_observer_interface.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "sdk/objc/native/api/ssl_certificate_verifier.h"
+
+NSString *const kRTCPeerConnectionErrorDomain = @"org.webrtc.RTC_OBJC_TYPE(RTCPeerConnection)";
+int const kRTCPeerConnnectionSessionDescriptionError = -1;
+
+namespace {
+
+class SetSessionDescriptionObserver : public webrtc::SetLocalDescriptionObserverInterface,
+                                      public webrtc::SetRemoteDescriptionObserverInterface {
+ public:
+  SetSessionDescriptionObserver(RTCSetSessionDescriptionCompletionHandler completionHandler) {
+    completion_handler_ = completionHandler;
+  }
+
+  virtual void OnSetLocalDescriptionComplete(webrtc::RTCError error) override {
+    OnCompelete(error);
+  }
+
+  virtual void OnSetRemoteDescriptionComplete(webrtc::RTCError error) override {
+    OnCompelete(error);
+  }
+
+ private:
+  void OnCompelete(webrtc::RTCError error) {
+    RTC_DCHECK(completion_handler_ != nil);
+    if (error.ok()) {
+      completion_handler_(nil);
+    } else {
+      // TODO(hta): Add handling of error.type()
+      NSString *str = [NSString stringForStdString:error.message()];
+      NSError *err = [NSError errorWithDomain:kRTCPeerConnectionErrorDomain
+                                         code:kRTCPeerConnnectionSessionDescriptionError
+                                     userInfo:@{NSLocalizedDescriptionKey : str}];
+      completion_handler_(err);
+    }
+    completion_handler_ = nil;
+  }
+  RTCSetSessionDescriptionCompletionHandler completion_handler_;
+};
+
+}  // anonymous namespace
+
+namespace webrtc {
+
+class CreateSessionDescriptionObserverAdapter
+    : public CreateSessionDescriptionObserver {
+ public:
+  CreateSessionDescriptionObserverAdapter(void (^completionHandler)(
+      RTC_OBJC_TYPE(RTCSessionDescription) * sessionDescription, NSError *error)) {
+    completion_handler_ = completionHandler;
+  }
+
+  ~CreateSessionDescriptionObserverAdapter() override { completion_handler_ = nil; }
+
+  void OnSuccess(SessionDescriptionInterface *desc) override {
+    RTC_DCHECK(completion_handler_);
+    std::unique_ptr<webrtc::SessionDescriptionInterface> description =
+        std::unique_ptr<webrtc::SessionDescriptionInterface>(desc);
+    RTC_OBJC_TYPE(RTCSessionDescription) *session =
+        [[RTC_OBJC_TYPE(RTCSessionDescription) alloc] initWithNativeDescription:description.get()];
+    completion_handler_(session, nil);
+    completion_handler_ = nil;
+  }
+
+  void OnFailure(RTCError error) override {
+    RTC_DCHECK(completion_handler_);
+    // TODO(hta): Add handling of error.type()
+    NSString *str = [NSString stringForStdString:error.message()];
+    NSError* err =
+        [NSError errorWithDomain:kRTCPeerConnectionErrorDomain
+                            code:kRTCPeerConnnectionSessionDescriptionError
+                        userInfo:@{ NSLocalizedDescriptionKey : str }];
+    completion_handler_(nil, err);
+    completion_handler_ = nil;
+  }
+
+ private:
+  void (^completion_handler_)(RTC_OBJC_TYPE(RTCSessionDescription) * sessionDescription,
+                              NSError *error);
+};
+
+PeerConnectionDelegateAdapter::PeerConnectionDelegateAdapter(RTC_OBJC_TYPE(RTCPeerConnection) *
+                                                             peerConnection) {
+  peer_connection_ = peerConnection;
+}
+
+PeerConnectionDelegateAdapter::~PeerConnectionDelegateAdapter() {
+  peer_connection_ = nil;
+}
+
+void PeerConnectionDelegateAdapter::OnSignalingChange(
+    PeerConnectionInterface::SignalingState new_state) {
+  RTCSignalingState state =
+      [[RTC_OBJC_TYPE(RTCPeerConnection) class] signalingStateForNativeState:new_state];
+  RTC_OBJC_TYPE(RTCPeerConnection) *peer_connection = peer_connection_;
+  [peer_connection.delegate peerConnection:peer_connection
+                   didChangeSignalingState:state];
+}
+
+void PeerConnectionDelegateAdapter::OnAddStream(
+    rtc::scoped_refptr<MediaStreamInterface> stream) {
+  RTC_OBJC_TYPE(RTCPeerConnection) *peer_connection = peer_connection_;
+  RTC_OBJC_TYPE(RTCMediaStream) *mediaStream =
+      [[RTC_OBJC_TYPE(RTCMediaStream) alloc] initWithFactory:peer_connection.factory
+                                           nativeMediaStream:stream];
+  [peer_connection.delegate peerConnection:peer_connection
+                              didAddStream:mediaStream];
+}
+
+void PeerConnectionDelegateAdapter::OnRemoveStream(
+    rtc::scoped_refptr<MediaStreamInterface> stream) {
+  RTC_OBJC_TYPE(RTCPeerConnection) *peer_connection = peer_connection_;
+  RTC_OBJC_TYPE(RTCMediaStream) *mediaStream =
+      [[RTC_OBJC_TYPE(RTCMediaStream) alloc] initWithFactory:peer_connection.factory
+                                           nativeMediaStream:stream];
+
+  [peer_connection.delegate peerConnection:peer_connection
+                           didRemoveStream:mediaStream];
+}
+
+void PeerConnectionDelegateAdapter::OnTrack(
+    rtc::scoped_refptr<RtpTransceiverInterface> nativeTransceiver) {
+  RTC_OBJC_TYPE(RTCPeerConnection) *peer_connection = peer_connection_;
+  RTC_OBJC_TYPE(RTCRtpTransceiver) *transceiver =
+      [[RTC_OBJC_TYPE(RTCRtpTransceiver) alloc] initWithFactory:peer_connection.factory
+                                           nativeRtpTransceiver:nativeTransceiver];
+  if ([peer_connection.delegate
+          respondsToSelector:@selector(peerConnection:didStartReceivingOnTransceiver:)]) {
+    [peer_connection.delegate peerConnection:peer_connection
+              didStartReceivingOnTransceiver:transceiver];
+  }
+}
+
+void PeerConnectionDelegateAdapter::OnDataChannel(
+    rtc::scoped_refptr<DataChannelInterface> data_channel) {
+  RTC_OBJC_TYPE(RTCPeerConnection) *peer_connection = peer_connection_;
+  RTC_OBJC_TYPE(RTCDataChannel) *dataChannel =
+      [[RTC_OBJC_TYPE(RTCDataChannel) alloc] initWithFactory:peer_connection.factory
+                                           nativeDataChannel:data_channel];
+  [peer_connection.delegate peerConnection:peer_connection
+                        didOpenDataChannel:dataChannel];
+}
+
+void PeerConnectionDelegateAdapter::OnRenegotiationNeeded() {
+  RTC_OBJC_TYPE(RTCPeerConnection) *peer_connection = peer_connection_;
+  [peer_connection.delegate peerConnectionShouldNegotiate:peer_connection];
+}
+
+void PeerConnectionDelegateAdapter::OnIceConnectionChange(
+    PeerConnectionInterface::IceConnectionState new_state) {
+  RTCIceConnectionState state =
+      [RTC_OBJC_TYPE(RTCPeerConnection) iceConnectionStateForNativeState:new_state];
+  [peer_connection_.delegate peerConnection:peer_connection_ didChangeIceConnectionState:state];
+}
+
+void PeerConnectionDelegateAdapter::OnStandardizedIceConnectionChange(
+    PeerConnectionInterface::IceConnectionState new_state) {
+  if ([peer_connection_.delegate
+          respondsToSelector:@selector(peerConnection:didChangeStandardizedIceConnectionState:)]) {
+    RTCIceConnectionState state =
+        [RTC_OBJC_TYPE(RTCPeerConnection) iceConnectionStateForNativeState:new_state];
+    [peer_connection_.delegate peerConnection:peer_connection_
+        didChangeStandardizedIceConnectionState:state];
+  }
+}
+
+void PeerConnectionDelegateAdapter::OnConnectionChange(
+    PeerConnectionInterface::PeerConnectionState new_state) {
+  if ([peer_connection_.delegate
+          respondsToSelector:@selector(peerConnection:didChangeConnectionState:)]) {
+    RTCPeerConnectionState state =
+        [RTC_OBJC_TYPE(RTCPeerConnection) connectionStateForNativeState:new_state];
+    [peer_connection_.delegate peerConnection:peer_connection_ didChangeConnectionState:state];
+  }
+}
+
+void PeerConnectionDelegateAdapter::OnIceGatheringChange(
+    PeerConnectionInterface::IceGatheringState new_state) {
+  RTCIceGatheringState state =
+      [[RTC_OBJC_TYPE(RTCPeerConnection) class] iceGatheringStateForNativeState:new_state];
+  RTC_OBJC_TYPE(RTCPeerConnection) *peer_connection = peer_connection_;
+  [peer_connection.delegate peerConnection:peer_connection
+                didChangeIceGatheringState:state];
+}
+
+void PeerConnectionDelegateAdapter::OnIceCandidate(
+    const IceCandidateInterface *candidate) {
+  RTC_OBJC_TYPE(RTCIceCandidate) *iceCandidate =
+      [[RTC_OBJC_TYPE(RTCIceCandidate) alloc] initWithNativeCandidate:candidate];
+  RTC_OBJC_TYPE(RTCPeerConnection) *peer_connection = peer_connection_;
+  [peer_connection.delegate peerConnection:peer_connection
+                   didGenerateIceCandidate:iceCandidate];
+}
+
+void PeerConnectionDelegateAdapter::OnIceCandidateError(const std::string &address,
+                                                        int port,
+                                                        const std::string &url,
+                                                        int error_code,
+                                                        const std::string &error_text) {
+  RTC_OBJC_TYPE(RTCPeerConnection) *peer_connection = peer_connection_;
+  RTC_OBJC_TYPE(RTCIceCandidateErrorEvent) *event =
+      [[RTC_OBJC_TYPE(RTCIceCandidateErrorEvent) alloc] initWithAddress:address
+                                                                   port:port
+                                                                    url:url
+                                                              errorCode:error_code
+                                                              errorText:error_text];
+  if ([peer_connection.delegate respondsToSelector:@selector(peerConnection:
+                                                             didFailToGatherIceCandidate:)]) {
+    [peer_connection.delegate peerConnection:peer_connection didFailToGatherIceCandidate:event];
+  }
+}
+
+void PeerConnectionDelegateAdapter::OnIceCandidatesRemoved(
+    const std::vector<cricket::Candidate>& candidates) {
+  NSMutableArray* ice_candidates =
+      [NSMutableArray arrayWithCapacity:candidates.size()];
+  for (const auto& candidate : candidates) {
+    std::unique_ptr<JsepIceCandidate> candidate_wrapper(
+        new JsepIceCandidate(candidate.transport_name(), -1, candidate));
+    RTC_OBJC_TYPE(RTCIceCandidate) *ice_candidate =
+        [[RTC_OBJC_TYPE(RTCIceCandidate) alloc] initWithNativeCandidate:candidate_wrapper.get()];
+    [ice_candidates addObject:ice_candidate];
+  }
+  RTC_OBJC_TYPE(RTCPeerConnection) *peer_connection = peer_connection_;
+  [peer_connection.delegate peerConnection:peer_connection
+                    didRemoveIceCandidates:ice_candidates];
+}
+
+void PeerConnectionDelegateAdapter::OnIceSelectedCandidatePairChanged(
+    const cricket::CandidatePairChangeEvent &event) {
+  const auto &selected_pair = event.selected_candidate_pair;
+  auto local_candidate_wrapper = std::make_unique<JsepIceCandidate>(
+      selected_pair.local_candidate().transport_name(), -1, selected_pair.local_candidate());
+  RTC_OBJC_TYPE(RTCIceCandidate) *local_candidate = [[RTC_OBJC_TYPE(RTCIceCandidate) alloc]
+      initWithNativeCandidate:local_candidate_wrapper.release()];
+  auto remote_candidate_wrapper = std::make_unique<JsepIceCandidate>(
+      selected_pair.remote_candidate().transport_name(), -1, selected_pair.remote_candidate());
+  RTC_OBJC_TYPE(RTCIceCandidate) *remote_candidate = [[RTC_OBJC_TYPE(RTCIceCandidate) alloc]
+      initWithNativeCandidate:remote_candidate_wrapper.release()];
+  RTC_OBJC_TYPE(RTCPeerConnection) *peer_connection = peer_connection_;
+  NSString *nsstr_reason = [NSString stringForStdString:event.reason];
+  if ([peer_connection.delegate
+          respondsToSelector:@selector
+          (peerConnection:didChangeLocalCandidate:remoteCandidate:lastReceivedMs:changeReason:)]) {
+    [peer_connection.delegate peerConnection:peer_connection
+                     didChangeLocalCandidate:local_candidate
+                             remoteCandidate:remote_candidate
+                              lastReceivedMs:event.last_data_received_ms
+                                changeReason:nsstr_reason];
+  }
+}
+
+void PeerConnectionDelegateAdapter::OnAddTrack(
+    rtc::scoped_refptr<RtpReceiverInterface> receiver,
+    const std::vector<rtc::scoped_refptr<MediaStreamInterface>> &streams) {
+  RTC_OBJC_TYPE(RTCPeerConnection) *peer_connection = peer_connection_;
+  if ([peer_connection.delegate respondsToSelector:@selector(peerConnection:
+                                                             didAddReceiver:streams:)]) {
+    NSMutableArray *mediaStreams = [NSMutableArray arrayWithCapacity:streams.size()];
+    for (const auto &nativeStream : streams) {
+      RTC_OBJC_TYPE(RTCMediaStream) *mediaStream =
+          [[RTC_OBJC_TYPE(RTCMediaStream) alloc] initWithFactory:peer_connection.factory
+                                               nativeMediaStream:nativeStream];
+      [mediaStreams addObject:mediaStream];
+    }
+    RTC_OBJC_TYPE(RTCRtpReceiver) *rtpReceiver =
+        [[RTC_OBJC_TYPE(RTCRtpReceiver) alloc] initWithFactory:peer_connection.factory
+                                             nativeRtpReceiver:receiver];
+
+    [peer_connection.delegate peerConnection:peer_connection
+                              didAddReceiver:rtpReceiver
+                                     streams:mediaStreams];
+  }
+}
+
+void PeerConnectionDelegateAdapter::OnRemoveTrack(
+    rtc::scoped_refptr<RtpReceiverInterface> receiver) {
+  RTC_OBJC_TYPE(RTCPeerConnection) *peer_connection = peer_connection_;
+  if ([peer_connection.delegate respondsToSelector:@selector(peerConnection:didRemoveReceiver:)]) {
+    RTC_OBJC_TYPE(RTCRtpReceiver) *rtpReceiver =
+        [[RTC_OBJC_TYPE(RTCRtpReceiver) alloc] initWithFactory:peer_connection.factory
+                                             nativeRtpReceiver:receiver];
+    [peer_connection.delegate peerConnection:peer_connection didRemoveReceiver:rtpReceiver];
+  }
+}
+
+}  // namespace webrtc
+
+@implementation RTC_OBJC_TYPE (RTCPeerConnection) {
+  RTC_OBJC_TYPE(RTCPeerConnectionFactory) * _factory;
+  NSMutableArray<RTC_OBJC_TYPE(RTCMediaStream) *> *_localStreams;
+  std::unique_ptr<webrtc::PeerConnectionDelegateAdapter> _observer;
+  rtc::scoped_refptr<webrtc::PeerConnectionInterface> _peerConnection;
+  std::unique_ptr<webrtc::MediaConstraints> _nativeConstraints;
+  BOOL _hasStartedRtcEventLog;
+}
+
+@synthesize delegate = _delegate;
+@synthesize factory = _factory;
+
+- (nullable instancetype)initWithFactory:(RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory
+                           configuration:(RTC_OBJC_TYPE(RTCConfiguration) *)configuration
+                             constraints:(RTC_OBJC_TYPE(RTCMediaConstraints) *)constraints
+                     certificateVerifier:
+                         (nullable id<RTC_OBJC_TYPE(RTCSSLCertificateVerifier)>)certificateVerifier
+                                delegate:(id<RTC_OBJC_TYPE(RTCPeerConnectionDelegate)>)delegate {
+  NSParameterAssert(factory);
+  std::unique_ptr<webrtc::PeerConnectionDependencies> dependencies =
+      std::make_unique<webrtc::PeerConnectionDependencies>(nullptr);
+  if (certificateVerifier != nil) {
+    dependencies->tls_cert_verifier = webrtc::ObjCToNativeCertificateVerifier(certificateVerifier);
+  }
+  return [self initWithDependencies:factory
+                      configuration:configuration
+                        constraints:constraints
+                       dependencies:std::move(dependencies)
+                           delegate:delegate];
+}
+
+- (nullable instancetype)
+    initWithDependencies:(RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory
+           configuration:(RTC_OBJC_TYPE(RTCConfiguration) *)configuration
+             constraints:(RTC_OBJC_TYPE(RTCMediaConstraints) *)constraints
+            dependencies:(std::unique_ptr<webrtc::PeerConnectionDependencies>)dependencies
+                delegate:(id<RTC_OBJC_TYPE(RTCPeerConnectionDelegate)>)delegate {
+  NSParameterAssert(factory);
+  NSParameterAssert(dependencies.get());
+  std::unique_ptr<webrtc::PeerConnectionInterface::RTCConfiguration> config(
+      [configuration createNativeConfiguration]);
+  if (!config) {
+    return nil;
+  }
+  if (self = [super init]) {
+    _observer.reset(new webrtc::PeerConnectionDelegateAdapter(self));
+    _nativeConstraints = constraints.nativeConstraints;
+    CopyConstraintsIntoRtcConfiguration(_nativeConstraints.get(), config.get());
+
+    webrtc::PeerConnectionDependencies deps = std::move(*dependencies.release());
+    deps.observer = _observer.get();
+    auto result = factory.nativeFactory->CreatePeerConnectionOrError(*config, std::move(deps));
+
+    if (!result.ok()) {
+      return nil;
+    }
+    _peerConnection = result.MoveValue();
+    _factory = factory;
+    _localStreams = [[NSMutableArray alloc] init];
+    _delegate = delegate;
+  }
+  return self;
+}
+
+- (NSArray<RTC_OBJC_TYPE(RTCMediaStream) *> *)localStreams {
+  return [_localStreams copy];
+}
+
+- (RTC_OBJC_TYPE(RTCSessionDescription) *)localDescription {
+  // It's only safe to operate on SessionDescriptionInterface on the signaling thread.
+  return _peerConnection->signaling_thread()->BlockingCall([self] {
+    const webrtc::SessionDescriptionInterface *description = _peerConnection->local_description();
+    return description ?
+        [[RTC_OBJC_TYPE(RTCSessionDescription) alloc] initWithNativeDescription:description] :
+        nil;
+  });
+}
+
+- (RTC_OBJC_TYPE(RTCSessionDescription) *)remoteDescription {
+  // It's only safe to operate on SessionDescriptionInterface on the signaling thread.
+  return _peerConnection->signaling_thread()->BlockingCall([self] {
+    const webrtc::SessionDescriptionInterface *description = _peerConnection->remote_description();
+    return description ?
+        [[RTC_OBJC_TYPE(RTCSessionDescription) alloc] initWithNativeDescription:description] :
+        nil;
+  });
+}
+
+- (RTCSignalingState)signalingState {
+  return [[self class]
+      signalingStateForNativeState:_peerConnection->signaling_state()];
+}
+
+- (RTCIceConnectionState)iceConnectionState {
+  return [[self class] iceConnectionStateForNativeState:
+      _peerConnection->ice_connection_state()];
+}
+
+- (RTCPeerConnectionState)connectionState {
+  return [[self class] connectionStateForNativeState:_peerConnection->peer_connection_state()];
+}
+
+- (RTCIceGatheringState)iceGatheringState {
+  return [[self class] iceGatheringStateForNativeState:
+      _peerConnection->ice_gathering_state()];
+}
+
+- (BOOL)setConfiguration:(RTC_OBJC_TYPE(RTCConfiguration) *)configuration {
+  std::unique_ptr<webrtc::PeerConnectionInterface::RTCConfiguration> config(
+      [configuration createNativeConfiguration]);
+  if (!config) {
+    return NO;
+  }
+  CopyConstraintsIntoRtcConfiguration(_nativeConstraints.get(),
+                                      config.get());
+  return _peerConnection->SetConfiguration(*config).ok();
+}
+
+- (RTC_OBJC_TYPE(RTCConfiguration) *)configuration {
+  webrtc::PeerConnectionInterface::RTCConfiguration config =
+    _peerConnection->GetConfiguration();
+  return [[RTC_OBJC_TYPE(RTCConfiguration) alloc] initWithNativeConfiguration:config];
+}
+
+- (void)close {
+  _peerConnection->Close();
+}
+
+- (void)addIceCandidate:(RTC_OBJC_TYPE(RTCIceCandidate) *)candidate {
+  std::unique_ptr<const webrtc::IceCandidateInterface> iceCandidate(
+      candidate.nativeCandidate);
+  _peerConnection->AddIceCandidate(iceCandidate.get());
+}
+- (void)addIceCandidate:(RTC_OBJC_TYPE(RTCIceCandidate) *)candidate
+      completionHandler:(void (^)(NSError *_Nullable error))completionHandler {
+  RTC_DCHECK(completionHandler != nil);
+  _peerConnection->AddIceCandidate(
+      candidate.nativeCandidate, [completionHandler](const auto &error) {
+        if (error.ok()) {
+          completionHandler(nil);
+        } else {
+          NSString *str = [NSString stringForStdString:error.message()];
+          NSError *err = [NSError errorWithDomain:kRTCPeerConnectionErrorDomain
+                                             code:static_cast<NSInteger>(error.type())
+                                         userInfo:@{NSLocalizedDescriptionKey : str}];
+          completionHandler(err);
+        }
+      });
+}
+- (void)removeIceCandidates:(NSArray<RTC_OBJC_TYPE(RTCIceCandidate) *> *)iceCandidates {
+  std::vector<cricket::Candidate> candidates;
+  for (RTC_OBJC_TYPE(RTCIceCandidate) * iceCandidate in iceCandidates) {
+    std::unique_ptr<const webrtc::IceCandidateInterface> candidate(
+        iceCandidate.nativeCandidate);
+    if (candidate) {
+      candidates.push_back(candidate->candidate());
+      // Need to fill the transport name from the sdp_mid.
+      candidates.back().set_transport_name(candidate->sdp_mid());
+    }
+  }
+  if (!candidates.empty()) {
+    _peerConnection->RemoveIceCandidates(candidates);
+  }
+}
+
+- (void)addStream:(RTC_OBJC_TYPE(RTCMediaStream) *)stream {
+  if (!_peerConnection->AddStream(stream.nativeMediaStream.get())) {
+    RTCLogError(@"Failed to add stream: %@", stream);
+    return;
+  }
+  [_localStreams addObject:stream];
+}
+
+- (void)removeStream:(RTC_OBJC_TYPE(RTCMediaStream) *)stream {
+  _peerConnection->RemoveStream(stream.nativeMediaStream.get());
+  [_localStreams removeObject:stream];
+}
+
+- (nullable RTC_OBJC_TYPE(RTCRtpSender) *)addTrack:(RTC_OBJC_TYPE(RTCMediaStreamTrack) *)track
+                                         streamIds:(NSArray<NSString *> *)streamIds {
+  std::vector<std::string> nativeStreamIds;
+  for (NSString *streamId in streamIds) {
+    nativeStreamIds.push_back([streamId UTF8String]);
+  }
+  webrtc::RTCErrorOr<rtc::scoped_refptr<webrtc::RtpSenderInterface>> nativeSenderOrError =
+      _peerConnection->AddTrack(track.nativeTrack, nativeStreamIds);
+  if (!nativeSenderOrError.ok()) {
+    RTCLogError(@"Failed to add track %@: %s", track, nativeSenderOrError.error().message());
+    return nil;
+  }
+  return [[RTC_OBJC_TYPE(RTCRtpSender) alloc] initWithFactory:self.factory
+                                              nativeRtpSender:nativeSenderOrError.MoveValue()];
+}
+
+- (BOOL)removeTrack:(RTC_OBJC_TYPE(RTCRtpSender) *)sender {
+  bool result = _peerConnection->RemoveTrackOrError(sender.nativeRtpSender).ok();
+  if (!result) {
+    RTCLogError(@"Failed to remote track %@", sender);
+  }
+  return result;
+}
+
+- (nullable RTC_OBJC_TYPE(RTCRtpTransceiver) *)addTransceiverWithTrack:
+    (RTC_OBJC_TYPE(RTCMediaStreamTrack) *)track {
+  return [self addTransceiverWithTrack:track
+                                  init:[[RTC_OBJC_TYPE(RTCRtpTransceiverInit) alloc] init]];
+}
+
+- (nullable RTC_OBJC_TYPE(RTCRtpTransceiver) *)
+    addTransceiverWithTrack:(RTC_OBJC_TYPE(RTCMediaStreamTrack) *)track
+                       init:(RTC_OBJC_TYPE(RTCRtpTransceiverInit) *)init {
+  webrtc::RTCErrorOr<rtc::scoped_refptr<webrtc::RtpTransceiverInterface>> nativeTransceiverOrError =
+      _peerConnection->AddTransceiver(track.nativeTrack, init.nativeInit);
+  if (!nativeTransceiverOrError.ok()) {
+    RTCLogError(
+        @"Failed to add transceiver %@: %s", track, nativeTransceiverOrError.error().message());
+    return nil;
+  }
+  return [[RTC_OBJC_TYPE(RTCRtpTransceiver) alloc]
+           initWithFactory:self.factory
+      nativeRtpTransceiver:nativeTransceiverOrError.MoveValue()];
+}
+
+- (nullable RTC_OBJC_TYPE(RTCRtpTransceiver) *)addTransceiverOfType:(RTCRtpMediaType)mediaType {
+  return [self addTransceiverOfType:mediaType
+                               init:[[RTC_OBJC_TYPE(RTCRtpTransceiverInit) alloc] init]];
+}
+
+- (nullable RTC_OBJC_TYPE(RTCRtpTransceiver) *)
+    addTransceiverOfType:(RTCRtpMediaType)mediaType
+                    init:(RTC_OBJC_TYPE(RTCRtpTransceiverInit) *)init {
+  webrtc::RTCErrorOr<rtc::scoped_refptr<webrtc::RtpTransceiverInterface>> nativeTransceiverOrError =
+      _peerConnection->AddTransceiver(
+          [RTC_OBJC_TYPE(RTCRtpReceiver) nativeMediaTypeForMediaType:mediaType], init.nativeInit);
+  if (!nativeTransceiverOrError.ok()) {
+    RTCLogError(@"Failed to add transceiver %@: %s",
+                [RTC_OBJC_TYPE(RTCRtpReceiver) stringForMediaType:mediaType],
+                nativeTransceiverOrError.error().message());
+    return nil;
+  }
+  return [[RTC_OBJC_TYPE(RTCRtpTransceiver) alloc]
+           initWithFactory:self.factory
+      nativeRtpTransceiver:nativeTransceiverOrError.MoveValue()];
+}
+
+- (void)restartIce {
+  _peerConnection->RestartIce();
+}
+
+- (void)offerForConstraints:(RTC_OBJC_TYPE(RTCMediaConstraints) *)constraints
+          completionHandler:(RTCCreateSessionDescriptionCompletionHandler)completionHandler {
+  RTC_DCHECK(completionHandler != nil);
+  rtc::scoped_refptr<webrtc::CreateSessionDescriptionObserverAdapter> observer =
+      rtc::make_ref_counted<webrtc::CreateSessionDescriptionObserverAdapter>(completionHandler);
+  webrtc::PeerConnectionInterface::RTCOfferAnswerOptions options;
+  CopyConstraintsIntoOfferAnswerOptions(constraints.nativeConstraints.get(), &options);
+
+  _peerConnection->CreateOffer(observer.get(), options);
+}
+
+- (void)answerForConstraints:(RTC_OBJC_TYPE(RTCMediaConstraints) *)constraints
+           completionHandler:(RTCCreateSessionDescriptionCompletionHandler)completionHandler {
+  RTC_DCHECK(completionHandler != nil);
+  rtc::scoped_refptr<webrtc::CreateSessionDescriptionObserverAdapter> observer =
+      rtc::make_ref_counted<webrtc::CreateSessionDescriptionObserverAdapter>(completionHandler);
+  webrtc::PeerConnectionInterface::RTCOfferAnswerOptions options;
+  CopyConstraintsIntoOfferAnswerOptions(constraints.nativeConstraints.get(), &options);
+
+  _peerConnection->CreateAnswer(observer.get(), options);
+}
+
+- (void)setLocalDescription:(RTC_OBJC_TYPE(RTCSessionDescription) *)sdp
+          completionHandler:(RTCSetSessionDescriptionCompletionHandler)completionHandler {
+  RTC_DCHECK(completionHandler != nil);
+  rtc::scoped_refptr<webrtc::SetLocalDescriptionObserverInterface> observer =
+      rtc::make_ref_counted<::SetSessionDescriptionObserver>(completionHandler);
+  _peerConnection->SetLocalDescription(sdp.nativeDescription, observer);
+}
+
+- (void)setLocalDescriptionWithCompletionHandler:
+    (RTCSetSessionDescriptionCompletionHandler)completionHandler {
+  RTC_DCHECK(completionHandler != nil);
+  rtc::scoped_refptr<webrtc::SetLocalDescriptionObserverInterface> observer =
+      rtc::make_ref_counted<::SetSessionDescriptionObserver>(completionHandler);
+  _peerConnection->SetLocalDescription(observer);
+}
+
+- (void)setRemoteDescription:(RTC_OBJC_TYPE(RTCSessionDescription) *)sdp
+           completionHandler:(RTCSetSessionDescriptionCompletionHandler)completionHandler {
+  RTC_DCHECK(completionHandler != nil);
+  rtc::scoped_refptr<webrtc::SetRemoteDescriptionObserverInterface> observer =
+      rtc::make_ref_counted<::SetSessionDescriptionObserver>(completionHandler);
+  _peerConnection->SetRemoteDescription(sdp.nativeDescription, observer);
+}
+
+- (BOOL)setBweMinBitrateBps:(nullable NSNumber *)minBitrateBps
+          currentBitrateBps:(nullable NSNumber *)currentBitrateBps
+              maxBitrateBps:(nullable NSNumber *)maxBitrateBps {
+  webrtc::BitrateSettings params;
+  if (minBitrateBps != nil) {
+    params.min_bitrate_bps = absl::optional<int>(minBitrateBps.intValue);
+  }
+  if (currentBitrateBps != nil) {
+    params.start_bitrate_bps = absl::optional<int>(currentBitrateBps.intValue);
+  }
+  if (maxBitrateBps != nil) {
+    params.max_bitrate_bps = absl::optional<int>(maxBitrateBps.intValue);
+  }
+  return _peerConnection->SetBitrate(params).ok();
+}
+
+- (BOOL)startRtcEventLogWithFilePath:(NSString *)filePath
+                      maxSizeInBytes:(int64_t)maxSizeInBytes {
+  RTC_DCHECK(filePath.length);
+  RTC_DCHECK_GT(maxSizeInBytes, 0);
+  RTC_DCHECK(!_hasStartedRtcEventLog);
+  if (_hasStartedRtcEventLog) {
+    RTCLogError(@"Event logging already started.");
+    return NO;
+  }
+  FILE *f = fopen(filePath.UTF8String, "wb");
+  if (!f) {
+    RTCLogError(@"Error opening file: %@. Error: %d", filePath, errno);
+    return NO;
+  }
+  // TODO(eladalon): It would be better to not allow negative values into PC.
+  const size_t max_size = (maxSizeInBytes < 0) ? webrtc::RtcEventLog::kUnlimitedOutput :
+                                                 rtc::saturated_cast<size_t>(maxSizeInBytes);
+
+  _hasStartedRtcEventLog = _peerConnection->StartRtcEventLog(
+      std::make_unique<webrtc::RtcEventLogOutputFile>(f, max_size));
+  return _hasStartedRtcEventLog;
+}
+
+- (void)stopRtcEventLog {
+  _peerConnection->StopRtcEventLog();
+  _hasStartedRtcEventLog = NO;
+}
+
+- (RTC_OBJC_TYPE(RTCRtpSender) *)senderWithKind:(NSString *)kind streamId:(NSString *)streamId {
+  std::string nativeKind = [NSString stdStringForString:kind];
+  std::string nativeStreamId = [NSString stdStringForString:streamId];
+  rtc::scoped_refptr<webrtc::RtpSenderInterface> nativeSender(
+      _peerConnection->CreateSender(nativeKind, nativeStreamId));
+  return nativeSender ? [[RTC_OBJC_TYPE(RTCRtpSender) alloc] initWithFactory:self.factory
+                                                             nativeRtpSender:nativeSender] :
+                        nil;
+}
+
+- (NSArray<RTC_OBJC_TYPE(RTCRtpSender) *> *)senders {
+  std::vector<rtc::scoped_refptr<webrtc::RtpSenderInterface>> nativeSenders(
+      _peerConnection->GetSenders());
+  NSMutableArray *senders = [[NSMutableArray alloc] init];
+  for (const auto &nativeSender : nativeSenders) {
+    RTC_OBJC_TYPE(RTCRtpSender) *sender =
+        [[RTC_OBJC_TYPE(RTCRtpSender) alloc] initWithFactory:self.factory
+                                             nativeRtpSender:nativeSender];
+    [senders addObject:sender];
+  }
+  return senders;
+}
+
+- (NSArray<RTC_OBJC_TYPE(RTCRtpReceiver) *> *)receivers {
+  std::vector<rtc::scoped_refptr<webrtc::RtpReceiverInterface>> nativeReceivers(
+      _peerConnection->GetReceivers());
+  NSMutableArray *receivers = [[NSMutableArray alloc] init];
+  for (const auto &nativeReceiver : nativeReceivers) {
+    RTC_OBJC_TYPE(RTCRtpReceiver) *receiver =
+        [[RTC_OBJC_TYPE(RTCRtpReceiver) alloc] initWithFactory:self.factory
+                                             nativeRtpReceiver:nativeReceiver];
+    [receivers addObject:receiver];
+  }
+  return receivers;
+}
+
+- (NSArray<RTC_OBJC_TYPE(RTCRtpTransceiver) *> *)transceivers {
+  std::vector<rtc::scoped_refptr<webrtc::RtpTransceiverInterface>> nativeTransceivers(
+      _peerConnection->GetTransceivers());
+  NSMutableArray *transceivers = [[NSMutableArray alloc] init];
+  for (const auto &nativeTransceiver : nativeTransceivers) {
+    RTC_OBJC_TYPE(RTCRtpTransceiver) *transceiver =
+        [[RTC_OBJC_TYPE(RTCRtpTransceiver) alloc] initWithFactory:self.factory
+                                             nativeRtpTransceiver:nativeTransceiver];
+    [transceivers addObject:transceiver];
+  }
+  return transceivers;
+}
+
+#pragma mark - Private
+
++ (webrtc::PeerConnectionInterface::SignalingState)nativeSignalingStateForState:
+    (RTCSignalingState)state {
+  switch (state) {
+    case RTCSignalingStateStable:
+      return webrtc::PeerConnectionInterface::kStable;
+    case RTCSignalingStateHaveLocalOffer:
+      return webrtc::PeerConnectionInterface::kHaveLocalOffer;
+    case RTCSignalingStateHaveLocalPrAnswer:
+      return webrtc::PeerConnectionInterface::kHaveLocalPrAnswer;
+    case RTCSignalingStateHaveRemoteOffer:
+      return webrtc::PeerConnectionInterface::kHaveRemoteOffer;
+    case RTCSignalingStateHaveRemotePrAnswer:
+      return webrtc::PeerConnectionInterface::kHaveRemotePrAnswer;
+    case RTCSignalingStateClosed:
+      return webrtc::PeerConnectionInterface::kClosed;
+  }
+}
+
++ (RTCSignalingState)signalingStateForNativeState:
+    (webrtc::PeerConnectionInterface::SignalingState)nativeState {
+  switch (nativeState) {
+    case webrtc::PeerConnectionInterface::kStable:
+      return RTCSignalingStateStable;
+    case webrtc::PeerConnectionInterface::kHaveLocalOffer:
+      return RTCSignalingStateHaveLocalOffer;
+    case webrtc::PeerConnectionInterface::kHaveLocalPrAnswer:
+      return RTCSignalingStateHaveLocalPrAnswer;
+    case webrtc::PeerConnectionInterface::kHaveRemoteOffer:
+      return RTCSignalingStateHaveRemoteOffer;
+    case webrtc::PeerConnectionInterface::kHaveRemotePrAnswer:
+      return RTCSignalingStateHaveRemotePrAnswer;
+    case webrtc::PeerConnectionInterface::kClosed:
+      return RTCSignalingStateClosed;
+  }
+}
+
++ (NSString *)stringForSignalingState:(RTCSignalingState)state {
+  switch (state) {
+    case RTCSignalingStateStable:
+      return @"STABLE";
+    case RTCSignalingStateHaveLocalOffer:
+      return @"HAVE_LOCAL_OFFER";
+    case RTCSignalingStateHaveLocalPrAnswer:
+      return @"HAVE_LOCAL_PRANSWER";
+    case RTCSignalingStateHaveRemoteOffer:
+      return @"HAVE_REMOTE_OFFER";
+    case RTCSignalingStateHaveRemotePrAnswer:
+      return @"HAVE_REMOTE_PRANSWER";
+    case RTCSignalingStateClosed:
+      return @"CLOSED";
+  }
+}
+
++ (webrtc::PeerConnectionInterface::PeerConnectionState)nativeConnectionStateForState:
+        (RTCPeerConnectionState)state {
+  switch (state) {
+    case RTCPeerConnectionStateNew:
+      return webrtc::PeerConnectionInterface::PeerConnectionState::kNew;
+    case RTCPeerConnectionStateConnecting:
+      return webrtc::PeerConnectionInterface::PeerConnectionState::kConnecting;
+    case RTCPeerConnectionStateConnected:
+      return webrtc::PeerConnectionInterface::PeerConnectionState::kConnected;
+    case RTCPeerConnectionStateFailed:
+      return webrtc::PeerConnectionInterface::PeerConnectionState::kFailed;
+    case RTCPeerConnectionStateDisconnected:
+      return webrtc::PeerConnectionInterface::PeerConnectionState::kDisconnected;
+    case RTCPeerConnectionStateClosed:
+      return webrtc::PeerConnectionInterface::PeerConnectionState::kClosed;
+  }
+}
+
++ (RTCPeerConnectionState)connectionStateForNativeState:
+        (webrtc::PeerConnectionInterface::PeerConnectionState)nativeState {
+  switch (nativeState) {
+    case webrtc::PeerConnectionInterface::PeerConnectionState::kNew:
+      return RTCPeerConnectionStateNew;
+    case webrtc::PeerConnectionInterface::PeerConnectionState::kConnecting:
+      return RTCPeerConnectionStateConnecting;
+    case webrtc::PeerConnectionInterface::PeerConnectionState::kConnected:
+      return RTCPeerConnectionStateConnected;
+    case webrtc::PeerConnectionInterface::PeerConnectionState::kFailed:
+      return RTCPeerConnectionStateFailed;
+    case webrtc::PeerConnectionInterface::PeerConnectionState::kDisconnected:
+      return RTCPeerConnectionStateDisconnected;
+    case webrtc::PeerConnectionInterface::PeerConnectionState::kClosed:
+      return RTCPeerConnectionStateClosed;
+  }
+}
+
++ (NSString *)stringForConnectionState:(RTCPeerConnectionState)state {
+  switch (state) {
+    case RTCPeerConnectionStateNew:
+      return @"NEW";
+    case RTCPeerConnectionStateConnecting:
+      return @"CONNECTING";
+    case RTCPeerConnectionStateConnected:
+      return @"CONNECTED";
+    case RTCPeerConnectionStateFailed:
+      return @"FAILED";
+    case RTCPeerConnectionStateDisconnected:
+      return @"DISCONNECTED";
+    case RTCPeerConnectionStateClosed:
+      return @"CLOSED";
+  }
+}
+
++ (webrtc::PeerConnectionInterface::IceConnectionState)
+    nativeIceConnectionStateForState:(RTCIceConnectionState)state {
+  switch (state) {
+    case RTCIceConnectionStateNew:
+      return webrtc::PeerConnectionInterface::kIceConnectionNew;
+    case RTCIceConnectionStateChecking:
+      return webrtc::PeerConnectionInterface::kIceConnectionChecking;
+    case RTCIceConnectionStateConnected:
+      return webrtc::PeerConnectionInterface::kIceConnectionConnected;
+    case RTCIceConnectionStateCompleted:
+      return webrtc::PeerConnectionInterface::kIceConnectionCompleted;
+    case RTCIceConnectionStateFailed:
+      return webrtc::PeerConnectionInterface::kIceConnectionFailed;
+    case RTCIceConnectionStateDisconnected:
+      return webrtc::PeerConnectionInterface::kIceConnectionDisconnected;
+    case RTCIceConnectionStateClosed:
+      return webrtc::PeerConnectionInterface::kIceConnectionClosed;
+    case RTCIceConnectionStateCount:
+      return webrtc::PeerConnectionInterface::kIceConnectionMax;
+  }
+}
+
++ (RTCIceConnectionState)iceConnectionStateForNativeState:
+    (webrtc::PeerConnectionInterface::IceConnectionState)nativeState {
+  switch (nativeState) {
+    case webrtc::PeerConnectionInterface::kIceConnectionNew:
+      return RTCIceConnectionStateNew;
+    case webrtc::PeerConnectionInterface::kIceConnectionChecking:
+      return RTCIceConnectionStateChecking;
+    case webrtc::PeerConnectionInterface::kIceConnectionConnected:
+      return RTCIceConnectionStateConnected;
+    case webrtc::PeerConnectionInterface::kIceConnectionCompleted:
+      return RTCIceConnectionStateCompleted;
+    case webrtc::PeerConnectionInterface::kIceConnectionFailed:
+      return RTCIceConnectionStateFailed;
+    case webrtc::PeerConnectionInterface::kIceConnectionDisconnected:
+      return RTCIceConnectionStateDisconnected;
+    case webrtc::PeerConnectionInterface::kIceConnectionClosed:
+      return RTCIceConnectionStateClosed;
+    case webrtc::PeerConnectionInterface::kIceConnectionMax:
+      return RTCIceConnectionStateCount;
+  }
+}
+
++ (NSString *)stringForIceConnectionState:(RTCIceConnectionState)state {
+  switch (state) {
+    case RTCIceConnectionStateNew:
+      return @"NEW";
+    case RTCIceConnectionStateChecking:
+      return @"CHECKING";
+    case RTCIceConnectionStateConnected:
+      return @"CONNECTED";
+    case RTCIceConnectionStateCompleted:
+      return @"COMPLETED";
+    case RTCIceConnectionStateFailed:
+      return @"FAILED";
+    case RTCIceConnectionStateDisconnected:
+      return @"DISCONNECTED";
+    case RTCIceConnectionStateClosed:
+      return @"CLOSED";
+    case RTCIceConnectionStateCount:
+      return @"COUNT";
+  }
+}
+
++ (webrtc::PeerConnectionInterface::IceGatheringState)
+    nativeIceGatheringStateForState:(RTCIceGatheringState)state {
+  switch (state) {
+    case RTCIceGatheringStateNew:
+      return webrtc::PeerConnectionInterface::kIceGatheringNew;
+    case RTCIceGatheringStateGathering:
+      return webrtc::PeerConnectionInterface::kIceGatheringGathering;
+    case RTCIceGatheringStateComplete:
+      return webrtc::PeerConnectionInterface::kIceGatheringComplete;
+  }
+}
+
++ (RTCIceGatheringState)iceGatheringStateForNativeState:
+    (webrtc::PeerConnectionInterface::IceGatheringState)nativeState {
+  switch (nativeState) {
+    case webrtc::PeerConnectionInterface::kIceGatheringNew:
+      return RTCIceGatheringStateNew;
+    case webrtc::PeerConnectionInterface::kIceGatheringGathering:
+      return RTCIceGatheringStateGathering;
+    case webrtc::PeerConnectionInterface::kIceGatheringComplete:
+      return RTCIceGatheringStateComplete;
+  }
+}
+
++ (NSString *)stringForIceGatheringState:(RTCIceGatheringState)state {
+  switch (state) {
+    case RTCIceGatheringStateNew:
+      return @"NEW";
+    case RTCIceGatheringStateGathering:
+      return @"GATHERING";
+    case RTCIceGatheringStateComplete:
+      return @"COMPLETE";
+  }
+}
+
++ (webrtc::PeerConnectionInterface::StatsOutputLevel)
+    nativeStatsOutputLevelForLevel:(RTCStatsOutputLevel)level {
+  switch (level) {
+    case RTCStatsOutputLevelStandard:
+      return webrtc::PeerConnectionInterface::kStatsOutputLevelStandard;
+    case RTCStatsOutputLevelDebug:
+      return webrtc::PeerConnectionInterface::kStatsOutputLevelDebug;
+  }
+}
+
+- (rtc::scoped_refptr<webrtc::PeerConnectionInterface>)nativePeerConnection {
+  return _peerConnection;
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCPeerConnectionFactory+Native.h b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCPeerConnectionFactory+Native.h
new file mode 100644
index 0000000000..f361b9f0ea
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCPeerConnectionFactory+Native.h
@@ -0,0 +1,85 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCPeerConnectionFactory.h"
+
+#include "api/scoped_refptr.h"
+
+namespace webrtc {
+
+class AudioDeviceModule;
+class AudioEncoderFactory;
+class AudioDecoderFactory;
+class NetworkControllerFactoryInterface;
+class VideoEncoderFactory;
+class VideoDecoderFactory;
+class AudioProcessing;
+struct PeerConnectionDependencies;
+
+}  // namespace webrtc
+
+NS_ASSUME_NONNULL_BEGIN
+
+/**
+ * This class extension exposes methods that work directly with injectable C++ components.
+ */
+@interface RTC_OBJC_TYPE (RTCPeerConnectionFactory)
+()
+
+    - (instancetype)initNative NS_DESIGNATED_INITIALIZER;
+
+/* Initializer used when WebRTC is compiled with no media support */
+- (instancetype)initWithNoMedia;
+
+/* Initialize object with injectable native audio/video encoder/decoder factories */
+- (instancetype)initWithNativeAudioEncoderFactory:
+                    (rtc::scoped_refptr<webrtc::AudioEncoderFactory>)audioEncoderFactory
+                        nativeAudioDecoderFactory:
+                            (rtc::scoped_refptr<webrtc::AudioDecoderFactory>)audioDecoderFactory
+                        nativeVideoEncoderFactory:
+                            (std::unique_ptr<webrtc::VideoEncoderFactory>)videoEncoderFactory
+                        nativeVideoDecoderFactory:
+                            (std::unique_ptr<webrtc::VideoDecoderFactory>)videoDecoderFactory
+                                audioDeviceModule:
+                                    (nullable webrtc::AudioDeviceModule *)audioDeviceModule
+                            audioProcessingModule:
+                                (rtc::scoped_refptr<webrtc::AudioProcessing>)audioProcessingModule;
+
+- (instancetype)
+    initWithNativeAudioEncoderFactory:
+        (rtc::scoped_refptr<webrtc::AudioEncoderFactory>)audioEncoderFactory
+            nativeAudioDecoderFactory:
+                (rtc::scoped_refptr<webrtc::AudioDecoderFactory>)audioDecoderFactory
+            nativeVideoEncoderFactory:
+                (std::unique_ptr<webrtc::VideoEncoderFactory>)videoEncoderFactory
+            nativeVideoDecoderFactory:
+                (std::unique_ptr<webrtc::VideoDecoderFactory>)videoDecoderFactory
+                    audioDeviceModule:(nullable webrtc::AudioDeviceModule *)audioDeviceModule
+                audioProcessingModule:
+                    (rtc::scoped_refptr<webrtc::AudioProcessing>)audioProcessingModule
+             networkControllerFactory:(std::unique_ptr<webrtc::NetworkControllerFactoryInterface>)
+                                          networkControllerFactory;
+
+- (instancetype)
+    initWithEncoderFactory:(nullable id<RTC_OBJC_TYPE(RTCVideoEncoderFactory)>)encoderFactory
+            decoderFactory:(nullable id<RTC_OBJC_TYPE(RTCVideoDecoderFactory)>)decoderFactory;
+
+/** Initialize an RTCPeerConnection with a configuration, constraints, and
+ *  dependencies.
+ */
+- (nullable RTC_OBJC_TYPE(RTCPeerConnection) *)
+    peerConnectionWithDependencies:(RTC_OBJC_TYPE(RTCConfiguration) *)configuration
+                       constraints:(RTC_OBJC_TYPE(RTCMediaConstraints) *)constraints
+                      dependencies:(std::unique_ptr<webrtc::PeerConnectionDependencies>)dependencies
+                          delegate:(nullable id<RTC_OBJC_TYPE(RTCPeerConnectionDelegate)>)delegate;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCPeerConnectionFactory+Private.h b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCPeerConnectionFactory+Private.h
new file mode 100644
index 0000000000..9613646270
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCPeerConnectionFactory+Private.h
@@ -0,0 +1,35 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCPeerConnectionFactory.h"
+
+#include "api/peer_connection_interface.h"
+#include "api/scoped_refptr.h"
+#include "rtc_base/thread.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+@interface RTC_OBJC_TYPE (RTCPeerConnectionFactory)
+()
+
+    /**
+     * PeerConnectionFactoryInterface created and held by this
+     * RTCPeerConnectionFactory object. This is needed to pass to the underlying
+     * C++ APIs.
+     */
+    @property(nonatomic,
+              readonly) rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> nativeFactory;
+
+@property(nonatomic, readonly) rtc::Thread* signalingThread;
+@property(nonatomic, readonly) rtc::Thread* workerThread;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCPeerConnectionFactory.h b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCPeerConnectionFactory.h
new file mode 100644
index 0000000000..5575af98c9
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCPeerConnectionFactory.h
@@ -0,0 +1,113 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+@class RTC_OBJC_TYPE(RTCAudioSource);
+@class RTC_OBJC_TYPE(RTCAudioTrack);
+@class RTC_OBJC_TYPE(RTCConfiguration);
+@class RTC_OBJC_TYPE(RTCMediaConstraints);
+@class RTC_OBJC_TYPE(RTCMediaStream);
+@class RTC_OBJC_TYPE(RTCPeerConnection);
+@class RTC_OBJC_TYPE(RTCVideoSource);
+@class RTC_OBJC_TYPE(RTCVideoTrack);
+@class RTC_OBJC_TYPE(RTCPeerConnectionFactoryOptions);
+@protocol RTC_OBJC_TYPE
+(RTCPeerConnectionDelegate);
+@protocol RTC_OBJC_TYPE
+(RTCVideoDecoderFactory);
+@protocol RTC_OBJC_TYPE
+(RTCVideoEncoderFactory);
+@protocol RTC_OBJC_TYPE
+(RTCSSLCertificateVerifier);
+@protocol RTC_OBJC_TYPE
+(RTCAudioDevice);
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCPeerConnectionFactory) : NSObject
+
+/* Initialize object with default H264 video encoder/decoder factories and default ADM */
+- (instancetype)init;
+
+/* Initialize object with injectable video encoder/decoder factories and default ADM */
+- (instancetype)
+    initWithEncoderFactory:(nullable id<RTC_OBJC_TYPE(RTCVideoEncoderFactory)>)encoderFactory
+            decoderFactory:(nullable id<RTC_OBJC_TYPE(RTCVideoDecoderFactory)>)decoderFactory;
+
+/* Initialize object with injectable video encoder/decoder factories and injectable ADM */
+- (instancetype)
+    initWithEncoderFactory:(nullable id<RTC_OBJC_TYPE(RTCVideoEncoderFactory)>)encoderFactory
+            decoderFactory:(nullable id<RTC_OBJC_TYPE(RTCVideoDecoderFactory)>)decoderFactory
+               audioDevice:(nullable id<RTC_OBJC_TYPE(RTCAudioDevice)>)audioDevice;
+
+/** Initialize an RTCAudioSource with constraints. */
+- (RTC_OBJC_TYPE(RTCAudioSource) *)audioSourceWithConstraints:
+    (nullable RTC_OBJC_TYPE(RTCMediaConstraints) *)constraints;
+
+/** Initialize an RTCAudioTrack with an id. Convenience ctor to use an audio source
+ * with no constraints.
+ */
+- (RTC_OBJC_TYPE(RTCAudioTrack) *)audioTrackWithTrackId:(NSString *)trackId;
+
+/** Initialize an RTCAudioTrack with a source and an id. */
+- (RTC_OBJC_TYPE(RTCAudioTrack) *)audioTrackWithSource:(RTC_OBJC_TYPE(RTCAudioSource) *)source
+                                               trackId:(NSString *)trackId;
+
+/** Initialize a generic RTCVideoSource. The RTCVideoSource should be
+ * passed to a RTCVideoCapturer implementation, e.g.
+ * RTCCameraVideoCapturer, in order to produce frames.
+ */
+- (RTC_OBJC_TYPE(RTCVideoSource) *)videoSource;
+
+/** Initialize a generic RTCVideoSource with he posibility of marking
+ * it as usable for screen sharing. The RTCVideoSource should be
+ * passed to a RTCVideoCapturer implementation, e.g.
+ * RTCCameraVideoCapturer, in order to produce frames.
+ */
+- (RTC_OBJC_TYPE(RTCVideoSource) *)videoSourceForScreenCast:(BOOL)forScreenCast;
+
+/** Initialize an RTCVideoTrack with a source and an id. */
+- (RTC_OBJC_TYPE(RTCVideoTrack) *)videoTrackWithSource:(RTC_OBJC_TYPE(RTCVideoSource) *)source
+                                               trackId:(NSString *)trackId;
+
+/** Initialize an RTCMediaStream with an id. */
+- (RTC_OBJC_TYPE(RTCMediaStream) *)mediaStreamWithStreamId:(NSString *)streamId;
+
+/** Initialize an RTCPeerConnection with a configuration, constraints, and
+ *  delegate.
+ */
+- (nullable RTC_OBJC_TYPE(RTCPeerConnection) *)
+    peerConnectionWithConfiguration:(RTC_OBJC_TYPE(RTCConfiguration) *)configuration
+                        constraints:(RTC_OBJC_TYPE(RTCMediaConstraints) *)constraints
+                           delegate:(nullable id<RTC_OBJC_TYPE(RTCPeerConnectionDelegate)>)delegate;
+
+- (nullable RTC_OBJC_TYPE(RTCPeerConnection) *)
+    peerConnectionWithConfiguration:(RTC_OBJC_TYPE(RTCConfiguration) *)configuration
+                        constraints:(RTC_OBJC_TYPE(RTCMediaConstraints) *)constraints
+                certificateVerifier:
+                    (id<RTC_OBJC_TYPE(RTCSSLCertificateVerifier)>)certificateVerifier
+                           delegate:(nullable id<RTC_OBJC_TYPE(RTCPeerConnectionDelegate)>)delegate;
+
+/** Set the options to be used for subsequently created RTCPeerConnections */
+- (void)setOptions:(nonnull RTC_OBJC_TYPE(RTCPeerConnectionFactoryOptions) *)options;
+
+/** Start an AecDump recording. This API call will likely change in the future. */
+- (BOOL)startAecDumpWithFilePath:(NSString *)filePath maxSizeInBytes:(int64_t)maxSizeInBytes;
+
+/* Stop an active AecDump recording */
+- (void)stopAecDump;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCPeerConnectionFactory.mm b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCPeerConnectionFactory.mm
new file mode 100644
index 0000000000..c4d89e911d
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCPeerConnectionFactory.mm
@@ -0,0 +1,342 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <memory>
+
+#import "RTCPeerConnectionFactory+Native.h"
+#import "RTCPeerConnectionFactory+Private.h"
+#import "RTCPeerConnectionFactoryOptions+Private.h"
+
+#import "RTCAudioSource+Private.h"
+#import "RTCAudioTrack+Private.h"
+#import "RTCMediaConstraints+Private.h"
+#import "RTCMediaStream+Private.h"
+#import "RTCPeerConnection+Private.h"
+#import "RTCVideoSource+Private.h"
+#import "RTCVideoTrack+Private.h"
+#import "base/RTCLogging.h"
+#import "base/RTCVideoDecoderFactory.h"
+#import "base/RTCVideoEncoderFactory.h"
+#import "helpers/NSString+StdString.h"
+#include "rtc_base/checks.h"
+#include "sdk/objc/native/api/network_monitor_factory.h"
+#include "sdk/objc/native/api/ssl_certificate_verifier.h"
+#include "system_wrappers/include/field_trial.h"
+
+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
+#include "api/audio_codecs/builtin_audio_encoder_factory.h"
+#include "api/rtc_event_log/rtc_event_log_factory.h"
+#include "api/task_queue/default_task_queue_factory.h"
+#include "api/transport/field_trial_based_config.h"
+#import "components/video_codec/RTCVideoDecoderFactoryH264.h"
+#import "components/video_codec/RTCVideoEncoderFactoryH264.h"
+#include "media/engine/webrtc_media_engine.h"
+#include "modules/audio_device/include/audio_device.h"
+#include "modules/audio_processing/include/audio_processing.h"
+
+#include "sdk/objc/native/api/objc_audio_device_module.h"
+#include "sdk/objc/native/api/video_decoder_factory.h"
+#include "sdk/objc/native/api/video_encoder_factory.h"
+#include "sdk/objc/native/src/objc_video_decoder_factory.h"
+#include "sdk/objc/native/src/objc_video_encoder_factory.h"
+
+#if defined(WEBRTC_IOS)
+#import "sdk/objc/native/api/audio_device_module.h"
+#endif
+
+@implementation RTC_OBJC_TYPE (RTCPeerConnectionFactory) {
+  std::unique_ptr<rtc::Thread> _networkThread;
+  std::unique_ptr<rtc::Thread> _workerThread;
+  std::unique_ptr<rtc::Thread> _signalingThread;
+  BOOL _hasStartedAecDump;
+}
+
+@synthesize nativeFactory = _nativeFactory;
+
+- (rtc::scoped_refptr<webrtc::AudioDeviceModule>)audioDeviceModule {
+#if defined(WEBRTC_IOS)
+  return webrtc::CreateAudioDeviceModule();
+#else
+  return nullptr;
+#endif
+}
+
+- (instancetype)init {
+  return [self
+      initWithNativeAudioEncoderFactory:webrtc::CreateBuiltinAudioEncoderFactory()
+              nativeAudioDecoderFactory:webrtc::CreateBuiltinAudioDecoderFactory()
+              nativeVideoEncoderFactory:webrtc::ObjCToNativeVideoEncoderFactory([[RTC_OBJC_TYPE(
+                                            RTCVideoEncoderFactoryH264) alloc] init])
+              nativeVideoDecoderFactory:webrtc::ObjCToNativeVideoDecoderFactory([[RTC_OBJC_TYPE(
+                                            RTCVideoDecoderFactoryH264) alloc] init])
+                      audioDeviceModule:[self audioDeviceModule].get()
+                  audioProcessingModule:nullptr];
+}
+
+- (instancetype)
+    initWithEncoderFactory:(nullable id<RTC_OBJC_TYPE(RTCVideoEncoderFactory)>)encoderFactory
+            decoderFactory:(nullable id<RTC_OBJC_TYPE(RTCVideoDecoderFactory)>)decoderFactory {
+  return [self initWithEncoderFactory:encoderFactory decoderFactory:decoderFactory audioDevice:nil];
+}
+
+- (instancetype)
+    initWithEncoderFactory:(nullable id<RTC_OBJC_TYPE(RTCVideoEncoderFactory)>)encoderFactory
+            decoderFactory:(nullable id<RTC_OBJC_TYPE(RTCVideoDecoderFactory)>)decoderFactory
+               audioDevice:(nullable id<RTC_OBJC_TYPE(RTCAudioDevice)>)audioDevice {
+#ifdef HAVE_NO_MEDIA
+  return [self initWithNoMedia];
+#else
+  std::unique_ptr<webrtc::VideoEncoderFactory> native_encoder_factory;
+  std::unique_ptr<webrtc::VideoDecoderFactory> native_decoder_factory;
+  if (encoderFactory) {
+    native_encoder_factory = webrtc::ObjCToNativeVideoEncoderFactory(encoderFactory);
+  }
+  if (decoderFactory) {
+    native_decoder_factory = webrtc::ObjCToNativeVideoDecoderFactory(decoderFactory);
+  }
+  rtc::scoped_refptr<webrtc::AudioDeviceModule> audio_device_module;
+  if (audioDevice) {
+    audio_device_module = webrtc::CreateAudioDeviceModule(audioDevice);
+  } else {
+    audio_device_module = [self audioDeviceModule];
+  }
+  return [self initWithNativeAudioEncoderFactory:webrtc::CreateBuiltinAudioEncoderFactory()
+                       nativeAudioDecoderFactory:webrtc::CreateBuiltinAudioDecoderFactory()
+                       nativeVideoEncoderFactory:std::move(native_encoder_factory)
+                       nativeVideoDecoderFactory:std::move(native_decoder_factory)
+                               audioDeviceModule:audio_device_module.get()
+                           audioProcessingModule:nullptr];
+#endif
+}
+
+- (instancetype)initNative {
+  if (self = [super init]) {
+    _networkThread = rtc::Thread::CreateWithSocketServer();
+    _networkThread->SetName("network_thread", _networkThread.get());
+    BOOL result = _networkThread->Start();
+    RTC_DCHECK(result) << "Failed to start network thread.";
+
+    _workerThread = rtc::Thread::Create();
+    _workerThread->SetName("worker_thread", _workerThread.get());
+    result = _workerThread->Start();
+    RTC_DCHECK(result) << "Failed to start worker thread.";
+
+    _signalingThread = rtc::Thread::Create();
+    _signalingThread->SetName("signaling_thread", _signalingThread.get());
+    result = _signalingThread->Start();
+    RTC_DCHECK(result) << "Failed to start signaling thread.";
+  }
+  return self;
+}
+
+- (instancetype)initWithNoMedia {
+  if (self = [self initNative]) {
+    webrtc::PeerConnectionFactoryDependencies dependencies;
+    dependencies.network_thread = _networkThread.get();
+    dependencies.worker_thread = _workerThread.get();
+    dependencies.signaling_thread = _signalingThread.get();
+    if (webrtc::field_trial::IsEnabled("WebRTC-Network-UseNWPathMonitor")) {
+      dependencies.network_monitor_factory = webrtc::CreateNetworkMonitorFactory();
+    }
+    _nativeFactory = webrtc::CreateModularPeerConnectionFactory(std::move(dependencies));
+    NSAssert(_nativeFactory, @"Failed to initialize PeerConnectionFactory!");
+  }
+  return self;
+}
+
+- (instancetype)initWithNativeAudioEncoderFactory:
+                    (rtc::scoped_refptr<webrtc::AudioEncoderFactory>)audioEncoderFactory
+                        nativeAudioDecoderFactory:
+                            (rtc::scoped_refptr<webrtc::AudioDecoderFactory>)audioDecoderFactory
+                        nativeVideoEncoderFactory:
+                            (std::unique_ptr<webrtc::VideoEncoderFactory>)videoEncoderFactory
+                        nativeVideoDecoderFactory:
+                            (std::unique_ptr<webrtc::VideoDecoderFactory>)videoDecoderFactory
+                                audioDeviceModule:(webrtc::AudioDeviceModule *)audioDeviceModule
+                            audioProcessingModule:
+                                (rtc::scoped_refptr<webrtc::AudioProcessing>)audioProcessingModule {
+  return [self initWithNativeAudioEncoderFactory:audioEncoderFactory
+                       nativeAudioDecoderFactory:audioDecoderFactory
+                       nativeVideoEncoderFactory:std::move(videoEncoderFactory)
+                       nativeVideoDecoderFactory:std::move(videoDecoderFactory)
+                               audioDeviceModule:audioDeviceModule
+                           audioProcessingModule:audioProcessingModule
+                        networkControllerFactory:nullptr];
+}
+- (instancetype)initWithNativeAudioEncoderFactory:
+                    (rtc::scoped_refptr<webrtc::AudioEncoderFactory>)audioEncoderFactory
+                        nativeAudioDecoderFactory:
+                            (rtc::scoped_refptr<webrtc::AudioDecoderFactory>)audioDecoderFactory
+                        nativeVideoEncoderFactory:
+                            (std::unique_ptr<webrtc::VideoEncoderFactory>)videoEncoderFactory
+                        nativeVideoDecoderFactory:
+                            (std::unique_ptr<webrtc::VideoDecoderFactory>)videoDecoderFactory
+                                audioDeviceModule:(webrtc::AudioDeviceModule *)audioDeviceModule
+                            audioProcessingModule:
+                                (rtc::scoped_refptr<webrtc::AudioProcessing>)audioProcessingModule
+                         networkControllerFactory:
+                             (std::unique_ptr<webrtc::NetworkControllerFactoryInterface>)
+                                 networkControllerFactory {
+  if (self = [self initNative]) {
+    webrtc::PeerConnectionFactoryDependencies dependencies;
+    dependencies.network_thread = _networkThread.get();
+    dependencies.worker_thread = _workerThread.get();
+    dependencies.signaling_thread = _signalingThread.get();
+    if (webrtc::field_trial::IsEnabled("WebRTC-Network-UseNWPathMonitor")) {
+      dependencies.network_monitor_factory = webrtc::CreateNetworkMonitorFactory();
+    }
+    dependencies.trials = std::make_unique<webrtc::FieldTrialBasedConfig>();
+    dependencies.task_queue_factory =
+        webrtc::CreateDefaultTaskQueueFactory(dependencies.trials.get());
+    cricket::MediaEngineDependencies media_deps;
+    media_deps.adm = std::move(audioDeviceModule);
+    media_deps.task_queue_factory = dependencies.task_queue_factory.get();
+    media_deps.audio_encoder_factory = std::move(audioEncoderFactory);
+    media_deps.audio_decoder_factory = std::move(audioDecoderFactory);
+    media_deps.video_encoder_factory = std::move(videoEncoderFactory);
+    media_deps.video_decoder_factory = std::move(videoDecoderFactory);
+    if (audioProcessingModule) {
+      media_deps.audio_processing = std::move(audioProcessingModule);
+    } else {
+      media_deps.audio_processing = webrtc::AudioProcessingBuilder().Create();
+    }
+    media_deps.trials = dependencies.trials.get();
+    dependencies.media_engine = cricket::CreateMediaEngine(std::move(media_deps));
+    dependencies.call_factory = webrtc::CreateCallFactory();
+    dependencies.event_log_factory =
+        std::make_unique<webrtc::RtcEventLogFactory>(dependencies.task_queue_factory.get());
+    dependencies.network_controller_factory = std::move(networkControllerFactory);
+    _nativeFactory = webrtc::CreateModularPeerConnectionFactory(std::move(dependencies));
+    NSAssert(_nativeFactory, @"Failed to initialize PeerConnectionFactory!");
+  }
+  return self;
+}
+
+- (RTC_OBJC_TYPE(RTCAudioSource) *)audioSourceWithConstraints:
+    (nullable RTC_OBJC_TYPE(RTCMediaConstraints) *)constraints {
+  std::unique_ptr<webrtc::MediaConstraints> nativeConstraints;
+  if (constraints) {
+    nativeConstraints = constraints.nativeConstraints;
+  }
+  cricket::AudioOptions options;
+  CopyConstraintsIntoAudioOptions(nativeConstraints.get(), &options);
+
+  rtc::scoped_refptr<webrtc::AudioSourceInterface> source =
+      _nativeFactory->CreateAudioSource(options);
+  return [[RTC_OBJC_TYPE(RTCAudioSource) alloc] initWithFactory:self nativeAudioSource:source];
+}
+
+- (RTC_OBJC_TYPE(RTCAudioTrack) *)audioTrackWithTrackId:(NSString *)trackId {
+  RTC_OBJC_TYPE(RTCAudioSource) *audioSource = [self audioSourceWithConstraints:nil];
+  return [self audioTrackWithSource:audioSource trackId:trackId];
+}
+
+- (RTC_OBJC_TYPE(RTCAudioTrack) *)audioTrackWithSource:(RTC_OBJC_TYPE(RTCAudioSource) *)source
+                                               trackId:(NSString *)trackId {
+  return [[RTC_OBJC_TYPE(RTCAudioTrack) alloc] initWithFactory:self source:source trackId:trackId];
+}
+
+- (RTC_OBJC_TYPE(RTCVideoSource) *)videoSource {
+  return [[RTC_OBJC_TYPE(RTCVideoSource) alloc] initWithFactory:self
+                                                signalingThread:_signalingThread.get()
+                                                   workerThread:_workerThread.get()];
+}
+
+- (RTC_OBJC_TYPE(RTCVideoSource) *)videoSourceForScreenCast:(BOOL)forScreenCast {
+  return [[RTC_OBJC_TYPE(RTCVideoSource) alloc] initWithFactory:self
+                                                signalingThread:_signalingThread.get()
+                                                   workerThread:_workerThread.get()
+                                                   isScreenCast:forScreenCast];
+}
+
+- (RTC_OBJC_TYPE(RTCVideoTrack) *)videoTrackWithSource:(RTC_OBJC_TYPE(RTCVideoSource) *)source
+                                               trackId:(NSString *)trackId {
+  return [[RTC_OBJC_TYPE(RTCVideoTrack) alloc] initWithFactory:self source:source trackId:trackId];
+}
+
+- (RTC_OBJC_TYPE(RTCMediaStream) *)mediaStreamWithStreamId:(NSString *)streamId {
+  return [[RTC_OBJC_TYPE(RTCMediaStream) alloc] initWithFactory:self streamId:streamId];
+}
+
+- (nullable RTC_OBJC_TYPE(RTCPeerConnection) *)
+    peerConnectionWithConfiguration:(RTC_OBJC_TYPE(RTCConfiguration) *)configuration
+                        constraints:(RTC_OBJC_TYPE(RTCMediaConstraints) *)constraints
+                           delegate:
+                               (nullable id<RTC_OBJC_TYPE(RTCPeerConnectionDelegate)>)delegate {
+  return [[RTC_OBJC_TYPE(RTCPeerConnection) alloc] initWithFactory:self
+                                                     configuration:configuration
+                                                       constraints:constraints
+                                               certificateVerifier:nil
+                                                          delegate:delegate];
+}
+
+- (nullable RTC_OBJC_TYPE(RTCPeerConnection) *)
+    peerConnectionWithConfiguration:(RTC_OBJC_TYPE(RTCConfiguration) *)configuration
+                        constraints:(RTC_OBJC_TYPE(RTCMediaConstraints) *)constraints
+                certificateVerifier:
+                    (id<RTC_OBJC_TYPE(RTCSSLCertificateVerifier)>)certificateVerifier
+                           delegate:
+                               (nullable id<RTC_OBJC_TYPE(RTCPeerConnectionDelegate)>)delegate {
+  return [[RTC_OBJC_TYPE(RTCPeerConnection) alloc] initWithFactory:self
+                                                     configuration:configuration
+                                                       constraints:constraints
+                                               certificateVerifier:certificateVerifier
+                                                          delegate:delegate];
+}
+
+- (nullable RTC_OBJC_TYPE(RTCPeerConnection) *)
+    peerConnectionWithDependencies:(RTC_OBJC_TYPE(RTCConfiguration) *)configuration
+                       constraints:(RTC_OBJC_TYPE(RTCMediaConstraints) *)constraints
+                      dependencies:(std::unique_ptr<webrtc::PeerConnectionDependencies>)dependencies
+                          delegate:(id<RTC_OBJC_TYPE(RTCPeerConnectionDelegate)>)delegate {
+  return [[RTC_OBJC_TYPE(RTCPeerConnection) alloc] initWithDependencies:self
+                                                          configuration:configuration
+                                                            constraints:constraints
+                                                           dependencies:std::move(dependencies)
+                                                               delegate:delegate];
+}
+
+- (void)setOptions:(nonnull RTC_OBJC_TYPE(RTCPeerConnectionFactoryOptions) *)options {
+  RTC_DCHECK(options != nil);
+  _nativeFactory->SetOptions(options.nativeOptions);
+}
+
+- (BOOL)startAecDumpWithFilePath:(NSString *)filePath
+                  maxSizeInBytes:(int64_t)maxSizeInBytes {
+  RTC_DCHECK(filePath.length);
+  RTC_DCHECK_GT(maxSizeInBytes, 0);
+
+  if (_hasStartedAecDump) {
+    RTCLogError(@"Aec dump already started.");
+    return NO;
+  }
+  FILE *f = fopen(filePath.UTF8String, "wb");
+  if (!f) {
+    RTCLogError(@"Error opening file: %@. Error: %s", filePath, strerror(errno));
+    return NO;
+  }
+  _hasStartedAecDump = _nativeFactory->StartAecDump(f, maxSizeInBytes);
+  return _hasStartedAecDump;
+}
+
+- (void)stopAecDump {
+  _nativeFactory->StopAecDump();
+  _hasStartedAecDump = NO;
+}
+
+- (rtc::Thread *)signalingThread {
+  return _signalingThread.get();
+}
+
+- (rtc::Thread *)workerThread {
+  return _workerThread.get();
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCPeerConnectionFactoryBuilder+DefaultComponents.h b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCPeerConnectionFactoryBuilder+DefaultComponents.h
new file mode 100644
index 0000000000..070a0e74a5
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCPeerConnectionFactoryBuilder+DefaultComponents.h
@@ -0,0 +1,21 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCPeerConnectionFactoryBuilder.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+@interface RTCPeerConnectionFactoryBuilder (DefaultComponents)
+
++ (RTCPeerConnectionFactoryBuilder *)defaultBuilder;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCPeerConnectionFactoryBuilder+DefaultComponents.mm b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCPeerConnectionFactoryBuilder+DefaultComponents.mm
new file mode 100644
index 0000000000..522e520e12
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCPeerConnectionFactoryBuilder+DefaultComponents.mm
@@ -0,0 +1,49 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCPeerConnectionFactory+Native.h"
+#import "RTCPeerConnectionFactoryBuilder+DefaultComponents.h"
+
+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
+#include "api/audio_codecs/builtin_audio_encoder_factory.h"
+#import "components/video_codec/RTCVideoDecoderFactoryH264.h"
+#import "components/video_codec/RTCVideoEncoderFactoryH264.h"
+#include "sdk/objc/native/api/video_decoder_factory.h"
+#include "sdk/objc/native/api/video_encoder_factory.h"
+
+#if defined(WEBRTC_IOS)
+#import "sdk/objc/native/api/audio_device_module.h"
+#endif
+
+@implementation RTCPeerConnectionFactoryBuilder (DefaultComponents)
+
++ (RTCPeerConnectionFactoryBuilder *)defaultBuilder {
+  RTCPeerConnectionFactoryBuilder *builder = [[RTCPeerConnectionFactoryBuilder alloc] init];
+  auto audioEncoderFactory = webrtc::CreateBuiltinAudioEncoderFactory();
+  [builder setAudioEncoderFactory:audioEncoderFactory];
+
+  auto audioDecoderFactory = webrtc::CreateBuiltinAudioDecoderFactory();
+  [builder setAudioDecoderFactory:audioDecoderFactory];
+
+  auto videoEncoderFactory = webrtc::ObjCToNativeVideoEncoderFactory(
+      [[RTC_OBJC_TYPE(RTCVideoEncoderFactoryH264) alloc] init]);
+  [builder setVideoEncoderFactory:std::move(videoEncoderFactory)];
+
+  auto videoDecoderFactory = webrtc::ObjCToNativeVideoDecoderFactory(
+      [[RTC_OBJC_TYPE(RTCVideoDecoderFactoryH264) alloc] init]);
+  [builder setVideoDecoderFactory:std::move(videoDecoderFactory)];
+
+#if defined(WEBRTC_IOS)
+  [builder setAudioDeviceModule:webrtc::CreateAudioDeviceModule()];
+#endif
+  return builder;
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCPeerConnectionFactoryBuilder.h b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCPeerConnectionFactoryBuilder.h
new file mode 100644
index 0000000000..f0b0de156a
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCPeerConnectionFactoryBuilder.h
@@ -0,0 +1,48 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCPeerConnectionFactory.h"
+
+#include "api/scoped_refptr.h"
+
+namespace webrtc {
+
+class AudioDeviceModule;
+class AudioEncoderFactory;
+class AudioDecoderFactory;
+class VideoEncoderFactory;
+class VideoDecoderFactory;
+class AudioProcessing;
+
+}  // namespace webrtc
+
+NS_ASSUME_NONNULL_BEGIN
+
+@interface RTCPeerConnectionFactoryBuilder : NSObject
+
++ (RTCPeerConnectionFactoryBuilder *)builder;
+
+- (RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)createPeerConnectionFactory;
+
+- (void)setVideoEncoderFactory:(std::unique_ptr<webrtc::VideoEncoderFactory>)videoEncoderFactory;
+
+- (void)setVideoDecoderFactory:(std::unique_ptr<webrtc::VideoDecoderFactory>)videoDecoderFactory;
+
+- (void)setAudioEncoderFactory:(rtc::scoped_refptr<webrtc::AudioEncoderFactory>)audioEncoderFactory;
+
+- (void)setAudioDecoderFactory:(rtc::scoped_refptr<webrtc::AudioDecoderFactory>)audioDecoderFactory;
+
+- (void)setAudioDeviceModule:(rtc::scoped_refptr<webrtc::AudioDeviceModule>)audioDeviceModule;
+
+- (void)setAudioProcessingModule:(rtc::scoped_refptr<webrtc::AudioProcessing>)audioProcessingModule;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCPeerConnectionFactoryBuilder.mm b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCPeerConnectionFactoryBuilder.mm
new file mode 100644
index 0000000000..627909a0e3
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCPeerConnectionFactoryBuilder.mm
@@ -0,0 +1,72 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCPeerConnectionFactoryBuilder.h"
+#import "RTCPeerConnectionFactory+Native.h"
+
+#include "api/audio_codecs/audio_decoder_factory.h"
+#include "api/audio_codecs/audio_encoder_factory.h"
+#include "api/video_codecs/video_decoder_factory.h"
+#include "api/video_codecs/video_encoder_factory.h"
+#include "modules/audio_device/include/audio_device.h"
+#include "modules/audio_processing/include/audio_processing.h"
+
+@implementation RTCPeerConnectionFactoryBuilder {
+  std::unique_ptr<webrtc::VideoEncoderFactory> _videoEncoderFactory;
+  std::unique_ptr<webrtc::VideoDecoderFactory> _videoDecoderFactory;
+  rtc::scoped_refptr<webrtc::AudioEncoderFactory> _audioEncoderFactory;
+  rtc::scoped_refptr<webrtc::AudioDecoderFactory> _audioDecoderFactory;
+  rtc::scoped_refptr<webrtc::AudioDeviceModule> _audioDeviceModule;
+  rtc::scoped_refptr<webrtc::AudioProcessing> _audioProcessingModule;
+}
+
++ (RTCPeerConnectionFactoryBuilder *)builder {
+  return [[RTCPeerConnectionFactoryBuilder alloc] init];
+}
+
+- (RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)createPeerConnectionFactory {
+  RTC_OBJC_TYPE(RTCPeerConnectionFactory) *factory =
+      [RTC_OBJC_TYPE(RTCPeerConnectionFactory) alloc];
+  return [factory initWithNativeAudioEncoderFactory:_audioEncoderFactory
+                          nativeAudioDecoderFactory:_audioDecoderFactory
+                          nativeVideoEncoderFactory:std::move(_videoEncoderFactory)
+                          nativeVideoDecoderFactory:std::move(_videoDecoderFactory)
+                                  audioDeviceModule:_audioDeviceModule.get()
+                              audioProcessingModule:_audioProcessingModule];
+}
+
+- (void)setVideoEncoderFactory:(std::unique_ptr<webrtc::VideoEncoderFactory>)videoEncoderFactory {
+  _videoEncoderFactory = std::move(videoEncoderFactory);
+}
+
+- (void)setVideoDecoderFactory:(std::unique_ptr<webrtc::VideoDecoderFactory>)videoDecoderFactory {
+  _videoDecoderFactory = std::move(videoDecoderFactory);
+}
+
+- (void)setAudioEncoderFactory:
+        (rtc::scoped_refptr<webrtc::AudioEncoderFactory>)audioEncoderFactory {
+  _audioEncoderFactory = audioEncoderFactory;
+}
+
+- (void)setAudioDecoderFactory:
+        (rtc::scoped_refptr<webrtc::AudioDecoderFactory>)audioDecoderFactory {
+  _audioDecoderFactory = audioDecoderFactory;
+}
+
+- (void)setAudioDeviceModule:(rtc::scoped_refptr<webrtc::AudioDeviceModule>)audioDeviceModule {
+  _audioDeviceModule = audioDeviceModule;
+}
+
+- (void)setAudioProcessingModule:
+        (rtc::scoped_refptr<webrtc::AudioProcessing>)audioProcessingModule {
+  _audioProcessingModule = audioProcessingModule;
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCPeerConnectionFactoryOptions+Private.h b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCPeerConnectionFactoryOptions+Private.h
new file mode 100644
index 0000000000..8832b23695
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCPeerConnectionFactoryOptions+Private.h
@@ -0,0 +1,26 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCPeerConnectionFactoryOptions.h"
+
+#include "api/peer_connection_interface.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+@interface RTC_OBJC_TYPE (RTCPeerConnectionFactoryOptions)
+()
+
+    /** Returns the equivalent native PeerConnectionFactoryInterface::Options
+     * structure. */
+    @property(nonatomic, readonly) webrtc::PeerConnectionFactoryInterface::Options nativeOptions;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCPeerConnectionFactoryOptions.h b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCPeerConnectionFactoryOptions.h
new file mode 100644
index 0000000000..bfc54a5d7b
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCPeerConnectionFactoryOptions.h
@@ -0,0 +1,38 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCPeerConnectionFactoryOptions) : NSObject
+
+@property(nonatomic, assign) BOOL disableEncryption;
+
+@property(nonatomic, assign) BOOL disableNetworkMonitor;
+
+@property(nonatomic, assign) BOOL ignoreLoopbackNetworkAdapter;
+
+@property(nonatomic, assign) BOOL ignoreVPNNetworkAdapter;
+
+@property(nonatomic, assign) BOOL ignoreCellularNetworkAdapter;
+
+@property(nonatomic, assign) BOOL ignoreWiFiNetworkAdapter;
+
+@property(nonatomic, assign) BOOL ignoreEthernetNetworkAdapter;
+
+- (instancetype)init NS_DESIGNATED_INITIALIZER;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCPeerConnectionFactoryOptions.mm b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCPeerConnectionFactoryOptions.mm
new file mode 100644
index 0000000000..5467bd5fc9
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCPeerConnectionFactoryOptions.mm
@@ -0,0 +1,56 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCPeerConnectionFactoryOptions+Private.h"
+
+#include "rtc_base/network_constants.h"
+
+namespace {
+
+void setNetworkBit(webrtc::PeerConnectionFactoryInterface::Options* options,
+                   rtc::AdapterType type,
+                   bool ignore) {
+  if (ignore) {
+    options->network_ignore_mask |= type;
+  } else {
+    options->network_ignore_mask &= ~type;
+  }
+}
+}  // namespace
+
+@implementation RTC_OBJC_TYPE (RTCPeerConnectionFactoryOptions)
+
+@synthesize disableEncryption = _disableEncryption;
+@synthesize disableNetworkMonitor = _disableNetworkMonitor;
+@synthesize ignoreLoopbackNetworkAdapter = _ignoreLoopbackNetworkAdapter;
+@synthesize ignoreVPNNetworkAdapter = _ignoreVPNNetworkAdapter;
+@synthesize ignoreCellularNetworkAdapter = _ignoreCellularNetworkAdapter;
+@synthesize ignoreWiFiNetworkAdapter = _ignoreWiFiNetworkAdapter;
+@synthesize ignoreEthernetNetworkAdapter = _ignoreEthernetNetworkAdapter;
+
+- (instancetype)init {
+  return [super init];
+}
+
+- (webrtc::PeerConnectionFactoryInterface::Options)nativeOptions {
+  webrtc::PeerConnectionFactoryInterface::Options options;
+  options.disable_encryption = self.disableEncryption;
+  options.disable_network_monitor = self.disableNetworkMonitor;
+
+  setNetworkBit(&options, rtc::ADAPTER_TYPE_LOOPBACK, self.ignoreLoopbackNetworkAdapter);
+  setNetworkBit(&options, rtc::ADAPTER_TYPE_VPN, self.ignoreVPNNetworkAdapter);
+  setNetworkBit(&options, rtc::ADAPTER_TYPE_CELLULAR, self.ignoreCellularNetworkAdapter);
+  setNetworkBit(&options, rtc::ADAPTER_TYPE_WIFI, self.ignoreWiFiNetworkAdapter);
+  setNetworkBit(&options, rtc::ADAPTER_TYPE_ETHERNET, self.ignoreEthernetNetworkAdapter);
+
+  return options;
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCRtcpParameters+Private.h b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCRtcpParameters+Private.h
new file mode 100644
index 0000000000..c4d196cf79
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCRtcpParameters+Private.h
@@ -0,0 +1,29 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCRtcpParameters.h"
+
+#include "api/rtp_parameters.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+@interface RTC_OBJC_TYPE (RTCRtcpParameters)
+()
+
+    /** Returns the equivalent native RtcpParameters structure. */
+    @property(nonatomic, readonly) webrtc::RtcpParameters nativeParameters;
+
+/** Initialize the object with a native RtcpParameters structure. */
+- (instancetype)initWithNativeParameters:(const webrtc::RtcpParameters &)nativeParameters
+    NS_DESIGNATED_INITIALIZER;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCRtcpParameters.h b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCRtcpParameters.h
new file mode 100644
index 0000000000..2f7aad3aef
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCRtcpParameters.h
@@ -0,0 +1,30 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCRtcpParameters) : NSObject
+
+/** The Canonical Name used by RTCP. */
+@property(nonatomic, readonly, copy) NSString *cname;
+
+/** Whether reduced size RTCP is configured or compound RTCP. */
+@property(nonatomic, assign) BOOL isReducedSize;
+
+- (instancetype)init;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCRtcpParameters.mm b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCRtcpParameters.mm
new file mode 100644
index 0000000000..e92ee4b3e7
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCRtcpParameters.mm
@@ -0,0 +1,40 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCRtcpParameters+Private.h"
+
+#import "helpers/NSString+StdString.h"
+
+@implementation RTC_OBJC_TYPE (RTCRtcpParameters)
+
+@synthesize cname = _cname;
+@synthesize isReducedSize = _isReducedSize;
+
+- (instancetype)init {
+  webrtc::RtcpParameters nativeParameters;
+  return [self initWithNativeParameters:nativeParameters];
+}
+
+- (instancetype)initWithNativeParameters:(const webrtc::RtcpParameters &)nativeParameters {
+  if (self = [super init]) {
+    _cname = [NSString stringForStdString:nativeParameters.cname];
+    _isReducedSize = nativeParameters.reduced_size;
+  }
+  return self;
+}
+
+- (webrtc::RtcpParameters)nativeParameters {
+  webrtc::RtcpParameters parameters;
+  parameters.cname = [NSString stdStringForString:_cname];
+  parameters.reduced_size = _isReducedSize;
+  return parameters;
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCRtpCodecParameters+Private.h b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCRtpCodecParameters+Private.h
new file mode 100644
index 0000000000..ff23cfd642
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCRtpCodecParameters+Private.h
@@ -0,0 +1,29 @@
+/*
+ *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCRtpCodecParameters.h"
+
+#include "api/rtp_parameters.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+@interface RTC_OBJC_TYPE (RTCRtpCodecParameters)
+()
+
+    /** Returns the equivalent native RtpCodecParameters structure. */
+    @property(nonatomic, readonly) webrtc::RtpCodecParameters nativeParameters;
+
+/** Initialize the object with a native RtpCodecParameters structure. */
+- (instancetype)initWithNativeParameters:(const webrtc::RtpCodecParameters &)nativeParameters
+    NS_DESIGNATED_INITIALIZER;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCRtpCodecParameters.h b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCRtpCodecParameters.h
new file mode 100644
index 0000000000..6135223720
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCRtpCodecParameters.h
@@ -0,0 +1,73 @@
+/*
+ *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+RTC_EXTERN const NSString *const kRTCRtxCodecName;
+RTC_EXTERN const NSString *const kRTCRedCodecName;
+RTC_EXTERN const NSString *const kRTCUlpfecCodecName;
+RTC_EXTERN const NSString *const kRTCFlexfecCodecName;
+RTC_EXTERN const NSString *const kRTCOpusCodecName;
+RTC_EXTERN const NSString *const kRTCIsacCodecName;
+RTC_EXTERN const NSString *const kRTCL16CodecName;
+RTC_EXTERN const NSString *const kRTCG722CodecName;
+RTC_EXTERN const NSString *const kRTCIlbcCodecName;
+RTC_EXTERN const NSString *const kRTCPcmuCodecName;
+RTC_EXTERN const NSString *const kRTCPcmaCodecName;
+RTC_EXTERN const NSString *const kRTCDtmfCodecName;
+RTC_EXTERN const NSString *const kRTCComfortNoiseCodecName;
+RTC_EXTERN const NSString *const kRTCVp8CodecName;
+RTC_EXTERN const NSString *const kRTCVp9CodecName;
+RTC_EXTERN const NSString *const kRTCH264CodecName;
+
+/** Defined in https://www.w3.org/TR/webrtc/#idl-def-rtcrtpcodecparameters */
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCRtpCodecParameters) : NSObject
+
+/** The RTP payload type. */
+@property(nonatomic, assign) int payloadType;
+
+/**
+ * The codec MIME subtype. Valid types are listed in:
+ * http://www.iana.org/assignments/rtp-parameters/rtp-parameters.xhtml#rtp-parameters-2
+ *
+ * Several supported types are represented by the constants above.
+ */
+@property(nonatomic, readonly, nonnull) NSString *name;
+
+/**
+ * The media type of this codec. Equivalent to MIME top-level type.
+ *
+ * Valid values are kRTCMediaStreamTrackKindAudio and
+ * kRTCMediaStreamTrackKindVideo.
+ */
+@property(nonatomic, readonly, nonnull) NSString *kind;
+
+/** The codec clock rate expressed in Hertz. */
+@property(nonatomic, readonly, nullable) NSNumber *clockRate;
+
+/**
+ * The number of channels (mono=1, stereo=2).
+ * Set to null for video codecs.
+ **/
+@property(nonatomic, readonly, nullable) NSNumber *numChannels;
+
+/** The "format specific parameters" field from the "a=fmtp" line in the SDP */
+@property(nonatomic, readonly, nonnull) NSDictionary *parameters;
+
+- (instancetype)init;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCRtpCodecParameters.mm b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCRtpCodecParameters.mm
new file mode 100644
index 0000000000..753667b635
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCRtpCodecParameters.mm
@@ -0,0 +1,113 @@
+/*
+ *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCRtpCodecParameters+Private.h"
+
+#import "RTCMediaStreamTrack.h"
+#import "helpers/NSString+StdString.h"
+
+#include "media/base/media_constants.h"
+#include "rtc_base/checks.h"
+
+const NSString * const kRTCRtxCodecName = @(cricket::kRtxCodecName);
+const NSString * const kRTCRedCodecName = @(cricket::kRedCodecName);
+const NSString * const kRTCUlpfecCodecName = @(cricket::kUlpfecCodecName);
+const NSString * const kRTCFlexfecCodecName = @(cricket::kFlexfecCodecName);
+const NSString * const kRTCOpusCodecName = @(cricket::kOpusCodecName);
+const NSString * const kRTCIsacCodecName = @(cricket::kIsacCodecName);
+const NSString * const kRTCL16CodecName  = @(cricket::kL16CodecName);
+const NSString * const kRTCG722CodecName = @(cricket::kG722CodecName);
+const NSString * const kRTCIlbcCodecName = @(cricket::kIlbcCodecName);
+const NSString * const kRTCPcmuCodecName = @(cricket::kPcmuCodecName);
+const NSString * const kRTCPcmaCodecName = @(cricket::kPcmaCodecName);
+const NSString * const kRTCDtmfCodecName = @(cricket::kDtmfCodecName);
+const NSString * const kRTCComfortNoiseCodecName =
+    @(cricket::kComfortNoiseCodecName);
+const NSString * const kRTCVp8CodecName = @(cricket::kVp8CodecName);
+const NSString * const kRTCVp9CodecName = @(cricket::kVp9CodecName);
+const NSString * const kRTCH264CodecName = @(cricket::kH264CodecName);
+
+@implementation RTC_OBJC_TYPE (RTCRtpCodecParameters)
+
+@synthesize payloadType = _payloadType;
+@synthesize name = _name;
+@synthesize kind = _kind;
+@synthesize clockRate = _clockRate;
+@synthesize numChannels = _numChannels;
+@synthesize parameters = _parameters;
+
+- (instancetype)init {
+  webrtc::RtpCodecParameters nativeParameters;
+  return [self initWithNativeParameters:nativeParameters];
+}
+
+- (instancetype)initWithNativeParameters:
+    (const webrtc::RtpCodecParameters &)nativeParameters {
+  if (self = [super init]) {
+    _payloadType = nativeParameters.payload_type;
+    _name = [NSString stringForStdString:nativeParameters.name];
+    switch (nativeParameters.kind) {
+      case cricket::MEDIA_TYPE_AUDIO:
+        _kind = kRTCMediaStreamTrackKindAudio;
+        break;
+      case cricket::MEDIA_TYPE_VIDEO:
+        _kind = kRTCMediaStreamTrackKindVideo;
+        break;
+      case cricket::MEDIA_TYPE_DATA:
+        RTC_DCHECK_NOTREACHED();
+        break;
+      case cricket::MEDIA_TYPE_UNSUPPORTED:
+        RTC_DCHECK_NOTREACHED();
+        break;
+    }
+    if (nativeParameters.clock_rate) {
+      _clockRate = [NSNumber numberWithInt:*nativeParameters.clock_rate];
+    }
+    if (nativeParameters.num_channels) {
+      _numChannels = [NSNumber numberWithInt:*nativeParameters.num_channels];
+    }
+    NSMutableDictionary *parameters = [NSMutableDictionary dictionary];
+    for (const auto &parameter : nativeParameters.parameters) {
+      [parameters setObject:[NSString stringForStdString:parameter.second]
+                     forKey:[NSString stringForStdString:parameter.first]];
+    }
+    _parameters = parameters;
+  }
+  return self;
+}
+
+- (webrtc::RtpCodecParameters)nativeParameters {
+  webrtc::RtpCodecParameters parameters;
+  parameters.payload_type = _payloadType;
+  parameters.name = [NSString stdStringForString:_name];
+  // NSString pointer comparison is safe here since "kind" is readonly and only
+  // populated above.
+  if (_kind == kRTCMediaStreamTrackKindAudio) {
+    parameters.kind = cricket::MEDIA_TYPE_AUDIO;
+  } else if (_kind == kRTCMediaStreamTrackKindVideo) {
+    parameters.kind = cricket::MEDIA_TYPE_VIDEO;
+  } else {
+    RTC_DCHECK_NOTREACHED();
+  }
+  if (_clockRate != nil) {
+    parameters.clock_rate = absl::optional<int>(_clockRate.intValue);
+  }
+  if (_numChannels != nil) {
+    parameters.num_channels = absl::optional<int>(_numChannels.intValue);
+  }
+  for (NSString *paramKey in _parameters.allKeys) {
+    std::string key = [NSString stdStringForString:paramKey];
+    std::string value = [NSString stdStringForString:_parameters[paramKey]];
+    parameters.parameters[key] = value;
+  }
+  return parameters;
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCRtpEncodingParameters+Private.h b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCRtpEncodingParameters+Private.h
new file mode 100644
index 0000000000..d12ca624e3
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCRtpEncodingParameters+Private.h
@@ -0,0 +1,29 @@
+/*
+ *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCRtpEncodingParameters.h"
+
+#include "api/rtp_parameters.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+@interface RTC_OBJC_TYPE (RTCRtpEncodingParameters)
+()
+
+    /** Returns the equivalent native RtpEncodingParameters structure. */
+    @property(nonatomic, readonly) webrtc::RtpEncodingParameters nativeParameters;
+
+/** Initialize the object with a native RtpEncodingParameters structure. */
+- (instancetype)initWithNativeParameters:(const webrtc::RtpEncodingParameters &)nativeParameters
+    NS_DESIGNATED_INITIALIZER;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCRtpEncodingParameters.h b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCRtpEncodingParameters.h
new file mode 100644
index 0000000000..07f6b7a39c
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCRtpEncodingParameters.h
@@ -0,0 +1,76 @@
+/*
+ *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+/** Corresponds to webrtc::Priority. */
+typedef NS_ENUM(NSInteger, RTCPriority) {
+  RTCPriorityVeryLow,
+  RTCPriorityLow,
+  RTCPriorityMedium,
+  RTCPriorityHigh
+};
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCRtpEncodingParameters) : NSObject
+
+/** The idenfifier for the encoding layer. This is used in simulcast. */
+@property(nonatomic, copy, nullable) NSString *rid;
+
+/** Controls whether the encoding is currently transmitted. */
+@property(nonatomic, assign) BOOL isActive;
+
+/** The maximum bitrate to use for the encoding, or nil if there is no
+ *  limit.
+ */
+@property(nonatomic, copy, nullable) NSNumber *maxBitrateBps;
+
+/** The minimum bitrate to use for the encoding, or nil if there is no
+ *  limit.
+ */
+@property(nonatomic, copy, nullable) NSNumber *minBitrateBps;
+
+/** The maximum framerate to use for the encoding, or nil if there is no
+ *  limit.
+ */
+@property(nonatomic, copy, nullable) NSNumber *maxFramerate;
+
+/** The requested number of temporal layers to use for the encoding, or nil
+ * if the default should be used.
+ */
+@property(nonatomic, copy, nullable) NSNumber *numTemporalLayers;
+
+/** Scale the width and height down by this factor for video. If nil,
+ * implementation default scaling factor will be used.
+ */
+@property(nonatomic, copy, nullable) NSNumber *scaleResolutionDownBy;
+
+/** The SSRC being used by this encoding. */
+@property(nonatomic, readonly, nullable) NSNumber *ssrc;
+
+/** The relative bitrate priority. */
+@property(nonatomic, assign) double bitratePriority;
+
+/** The relative DiffServ Code Point priority. */
+@property(nonatomic, assign) RTCPriority networkPriority;
+
+/** Allow dynamic frame length changes for audio:
+ https://w3c.github.io/webrtc-extensions/#dom-rtcrtpencodingparameters-adaptiveptime */
+@property(nonatomic, assign) BOOL adaptiveAudioPacketTime;
+
+- (instancetype)init;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCRtpEncodingParameters.mm b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCRtpEncodingParameters.mm
new file mode 100644
index 0000000000..d6087dafb0
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCRtpEncodingParameters.mm
@@ -0,0 +1,128 @@
+/*
+ *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCRtpEncodingParameters+Private.h"
+
+#import "helpers/NSString+StdString.h"
+
+@implementation RTC_OBJC_TYPE (RTCRtpEncodingParameters)
+
+@synthesize rid = _rid;
+@synthesize isActive = _isActive;
+@synthesize maxBitrateBps = _maxBitrateBps;
+@synthesize minBitrateBps = _minBitrateBps;
+@synthesize maxFramerate = _maxFramerate;
+@synthesize numTemporalLayers = _numTemporalLayers;
+@synthesize scaleResolutionDownBy = _scaleResolutionDownBy;
+@synthesize ssrc = _ssrc;
+@synthesize bitratePriority = _bitratePriority;
+@synthesize networkPriority = _networkPriority;
+@synthesize adaptiveAudioPacketTime = _adaptiveAudioPacketTime;
+
+- (instancetype)init {
+  webrtc::RtpEncodingParameters nativeParameters;
+  return [self initWithNativeParameters:nativeParameters];
+}
+
+- (instancetype)initWithNativeParameters:
+    (const webrtc::RtpEncodingParameters &)nativeParameters {
+  if (self = [super init]) {
+    if (!nativeParameters.rid.empty()) {
+      _rid = [NSString stringForStdString:nativeParameters.rid];
+    }
+    _isActive = nativeParameters.active;
+    if (nativeParameters.max_bitrate_bps) {
+      _maxBitrateBps =
+          [NSNumber numberWithInt:*nativeParameters.max_bitrate_bps];
+    }
+    if (nativeParameters.min_bitrate_bps) {
+      _minBitrateBps =
+          [NSNumber numberWithInt:*nativeParameters.min_bitrate_bps];
+    }
+    if (nativeParameters.max_framerate) {
+      _maxFramerate = [NSNumber numberWithInt:*nativeParameters.max_framerate];
+    }
+    if (nativeParameters.num_temporal_layers) {
+      _numTemporalLayers = [NSNumber numberWithInt:*nativeParameters.num_temporal_layers];
+    }
+    if (nativeParameters.scale_resolution_down_by) {
+      _scaleResolutionDownBy =
+          [NSNumber numberWithDouble:*nativeParameters.scale_resolution_down_by];
+    }
+    if (nativeParameters.ssrc) {
+      _ssrc = [NSNumber numberWithUnsignedLong:*nativeParameters.ssrc];
+    }
+    _bitratePriority = nativeParameters.bitrate_priority;
+    _networkPriority = [RTC_OBJC_TYPE(RTCRtpEncodingParameters)
+        priorityFromNativePriority:nativeParameters.network_priority];
+    _adaptiveAudioPacketTime = nativeParameters.adaptive_ptime;
+  }
+  return self;
+}
+
+- (webrtc::RtpEncodingParameters)nativeParameters {
+  webrtc::RtpEncodingParameters parameters;
+  if (_rid != nil) {
+    parameters.rid = [NSString stdStringForString:_rid];
+  }
+  parameters.active = _isActive;
+  if (_maxBitrateBps != nil) {
+    parameters.max_bitrate_bps = absl::optional<int>(_maxBitrateBps.intValue);
+  }
+  if (_minBitrateBps != nil) {
+    parameters.min_bitrate_bps = absl::optional<int>(_minBitrateBps.intValue);
+  }
+  if (_maxFramerate != nil) {
+    parameters.max_framerate = absl::optional<int>(_maxFramerate.intValue);
+  }
+  if (_numTemporalLayers != nil) {
+    parameters.num_temporal_layers = absl::optional<int>(_numTemporalLayers.intValue);
+  }
+  if (_scaleResolutionDownBy != nil) {
+    parameters.scale_resolution_down_by =
+        absl::optional<double>(_scaleResolutionDownBy.doubleValue);
+  }
+  if (_ssrc != nil) {
+    parameters.ssrc = absl::optional<uint32_t>(_ssrc.unsignedLongValue);
+  }
+  parameters.bitrate_priority = _bitratePriority;
+  parameters.network_priority =
+      [RTC_OBJC_TYPE(RTCRtpEncodingParameters) nativePriorityFromPriority:_networkPriority];
+  parameters.adaptive_ptime = _adaptiveAudioPacketTime;
+  return parameters;
+}
+
++ (webrtc::Priority)nativePriorityFromPriority:(RTCPriority)networkPriority {
+  switch (networkPriority) {
+    case RTCPriorityVeryLow:
+      return webrtc::Priority::kVeryLow;
+    case RTCPriorityLow:
+      return webrtc::Priority::kLow;
+    case RTCPriorityMedium:
+      return webrtc::Priority::kMedium;
+    case RTCPriorityHigh:
+      return webrtc::Priority::kHigh;
+  }
+}
+
++ (RTCPriority)priorityFromNativePriority:(webrtc::Priority)nativePriority {
+  switch (nativePriority) {
+    case webrtc::Priority::kVeryLow:
+      return RTCPriorityVeryLow;
+    case webrtc::Priority::kLow:
+      return RTCPriorityLow;
+    case webrtc::Priority::kMedium:
+      return RTCPriorityMedium;
+    case webrtc::Priority::kHigh:
+      return RTCPriorityHigh;
+  }
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCRtpHeaderExtension+Private.h b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCRtpHeaderExtension+Private.h
new file mode 100644
index 0000000000..0e0fbba5ac
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCRtpHeaderExtension+Private.h
@@ -0,0 +1,29 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCRtpHeaderExtension.h"
+
+#include "api/rtp_parameters.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+@interface RTC_OBJC_TYPE (RTCRtpHeaderExtension)
+()
+
+    /** Returns the equivalent native RtpExtension structure. */
+    @property(nonatomic, readonly) webrtc::RtpExtension nativeParameters;
+
+/** Initialize the object with a native RtpExtension structure. */
+- (instancetype)initWithNativeParameters:(const webrtc::RtpExtension &)nativeParameters
+    NS_DESIGNATED_INITIALIZER;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCRtpHeaderExtension.h b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCRtpHeaderExtension.h
new file mode 100644
index 0000000000..4000bf5372
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCRtpHeaderExtension.h
@@ -0,0 +1,33 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCRtpHeaderExtension) : NSObject
+
+/** The URI of the RTP header extension, as defined in RFC5285. */
+@property(nonatomic, readonly, copy) NSString *uri;
+
+/** The value put in the RTP packet to identify the header extension. */
+@property(nonatomic, readonly) int id;
+
+/** Whether the header extension is encrypted or not. */
+@property(nonatomic, readonly, getter=isEncrypted) BOOL encrypted;
+
+- (instancetype)init;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCRtpHeaderExtension.mm b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCRtpHeaderExtension.mm
new file mode 100644
index 0000000000..68093e92ea
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCRtpHeaderExtension.mm
@@ -0,0 +1,43 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCRtpHeaderExtension+Private.h"
+
+#import "helpers/NSString+StdString.h"
+
+@implementation RTC_OBJC_TYPE (RTCRtpHeaderExtension)
+
+@synthesize uri = _uri;
+@synthesize id = _id;
+@synthesize encrypted = _encrypted;
+
+- (instancetype)init {
+  webrtc::RtpExtension nativeExtension;
+  return [self initWithNativeParameters:nativeExtension];
+}
+
+- (instancetype)initWithNativeParameters:(const webrtc::RtpExtension &)nativeParameters {
+  if (self = [super init]) {
+    _uri = [NSString stringForStdString:nativeParameters.uri];
+    _id = nativeParameters.id;
+    _encrypted = nativeParameters.encrypt;
+  }
+  return self;
+}
+
+- (webrtc::RtpExtension)nativeParameters {
+  webrtc::RtpExtension extension;
+  extension.uri = [NSString stdStringForString:_uri];
+  extension.id = _id;
+  extension.encrypt = _encrypted;
+  return extension;
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCRtpParameters+Private.h b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCRtpParameters+Private.h
new file mode 100644
index 0000000000..139617f727
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCRtpParameters+Private.h
@@ -0,0 +1,29 @@
+/*
+ *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCRtpParameters.h"
+
+#include "api/rtp_parameters.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+@interface RTC_OBJC_TYPE (RTCRtpParameters)
+()
+
+    /** Returns the equivalent native RtpParameters structure. */
+    @property(nonatomic, readonly) webrtc::RtpParameters nativeParameters;
+
+/** Initialize the object with a native RtpParameters structure. */
+- (instancetype)initWithNativeParameters:(const webrtc::RtpParameters &)nativeParameters
+    NS_DESIGNATED_INITIALIZER;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCRtpParameters.h b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCRtpParameters.h
new file mode 100644
index 0000000000..3d71c55ab9
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCRtpParameters.h
@@ -0,0 +1,58 @@
+/*
+ *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+#import "RTCRtcpParameters.h"
+#import "RTCRtpCodecParameters.h"
+#import "RTCRtpEncodingParameters.h"
+#import "RTCRtpHeaderExtension.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+/** Corresponds to webrtc::DegradationPreference. */
+typedef NS_ENUM(NSInteger, RTCDegradationPreference) {
+  RTCDegradationPreferenceDisabled,
+  RTCDegradationPreferenceMaintainFramerate,
+  RTCDegradationPreferenceMaintainResolution,
+  RTCDegradationPreferenceBalanced
+};
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCRtpParameters) : NSObject
+
+/** A unique identifier for the last set of parameters applied. */
+@property(nonatomic, copy) NSString *transactionId;
+
+/** Parameters used for RTCP. */
+@property(nonatomic, readonly, copy) RTC_OBJC_TYPE(RTCRtcpParameters) * rtcp;
+
+/** An array containing parameters for RTP header extensions. */
+@property(nonatomic, readonly, copy)
+    NSArray<RTC_OBJC_TYPE(RTCRtpHeaderExtension) *> *headerExtensions;
+
+/** The currently active encodings in the order of preference. */
+@property(nonatomic, copy) NSArray<RTC_OBJC_TYPE(RTCRtpEncodingParameters) *> *encodings;
+
+/** The negotiated set of send codecs in order of preference. */
+@property(nonatomic, copy) NSArray<RTC_OBJC_TYPE(RTCRtpCodecParameters) *> *codecs;
+
+/**
+ * Degradation preference in case of CPU adaptation or constrained bandwidth.
+ * If nil, implementation default degradation preference will be used.
+ */
+@property(nonatomic, copy, nullable) NSNumber *degradationPreference;
+
+- (instancetype)init;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCRtpParameters.mm b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCRtpParameters.mm
new file mode 100644
index 0000000000..2baf0ecd80
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCRtpParameters.mm
@@ -0,0 +1,121 @@
+/*
+ *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCRtpParameters+Private.h"
+
+#import "RTCRtcpParameters+Private.h"
+#import "RTCRtpCodecParameters+Private.h"
+#import "RTCRtpEncodingParameters+Private.h"
+#import "RTCRtpHeaderExtension+Private.h"
+#import "helpers/NSString+StdString.h"
+
+@implementation RTC_OBJC_TYPE (RTCRtpParameters)
+
+@synthesize transactionId = _transactionId;
+@synthesize rtcp = _rtcp;
+@synthesize headerExtensions = _headerExtensions;
+@synthesize encodings = _encodings;
+@synthesize codecs = _codecs;
+@synthesize degradationPreference = _degradationPreference;
+
+- (instancetype)init {
+  webrtc::RtpParameters nativeParameters;
+  return [self initWithNativeParameters:nativeParameters];
+}
+
+- (instancetype)initWithNativeParameters:
+    (const webrtc::RtpParameters &)nativeParameters {
+  if (self = [super init]) {
+    _transactionId = [NSString stringForStdString:nativeParameters.transaction_id];
+    _rtcp =
+        [[RTC_OBJC_TYPE(RTCRtcpParameters) alloc] initWithNativeParameters:nativeParameters.rtcp];
+
+    NSMutableArray *headerExtensions = [[NSMutableArray alloc] init];
+    for (const auto &headerExtension : nativeParameters.header_extensions) {
+      [headerExtensions addObject:[[RTC_OBJC_TYPE(RTCRtpHeaderExtension) alloc]
+                                      initWithNativeParameters:headerExtension]];
+    }
+    _headerExtensions = headerExtensions;
+
+    NSMutableArray *encodings = [[NSMutableArray alloc] init];
+    for (const auto &encoding : nativeParameters.encodings) {
+      [encodings addObject:[[RTC_OBJC_TYPE(RTCRtpEncodingParameters) alloc]
+                               initWithNativeParameters:encoding]];
+    }
+    _encodings = encodings;
+
+    NSMutableArray *codecs = [[NSMutableArray alloc] init];
+    for (const auto &codec : nativeParameters.codecs) {
+      [codecs
+          addObject:[[RTC_OBJC_TYPE(RTCRtpCodecParameters) alloc] initWithNativeParameters:codec]];
+    }
+    _codecs = codecs;
+
+    _degradationPreference = [RTC_OBJC_TYPE(RTCRtpParameters)
+        degradationPreferenceFromNativeDegradationPreference:nativeParameters
+                                                                 .degradation_preference];
+  }
+  return self;
+}
+
+- (webrtc::RtpParameters)nativeParameters {
+  webrtc::RtpParameters parameters;
+  parameters.transaction_id = [NSString stdStringForString:_transactionId];
+  parameters.rtcp = [_rtcp nativeParameters];
+  for (RTC_OBJC_TYPE(RTCRtpHeaderExtension) * headerExtension in _headerExtensions) {
+    parameters.header_extensions.push_back(headerExtension.nativeParameters);
+  }
+  for (RTC_OBJC_TYPE(RTCRtpEncodingParameters) * encoding in _encodings) {
+    parameters.encodings.push_back(encoding.nativeParameters);
+  }
+  for (RTC_OBJC_TYPE(RTCRtpCodecParameters) * codec in _codecs) {
+    parameters.codecs.push_back(codec.nativeParameters);
+  }
+  if (_degradationPreference) {
+    parameters.degradation_preference = [RTC_OBJC_TYPE(RTCRtpParameters)
+        nativeDegradationPreferenceFromDegradationPreference:(RTCDegradationPreference)
+                                                                 _degradationPreference.intValue];
+  }
+  return parameters;
+}
+
++ (webrtc::DegradationPreference)nativeDegradationPreferenceFromDegradationPreference:
+    (RTCDegradationPreference)degradationPreference {
+  switch (degradationPreference) {
+    case RTCDegradationPreferenceDisabled:
+      return webrtc::DegradationPreference::DISABLED;
+    case RTCDegradationPreferenceMaintainFramerate:
+      return webrtc::DegradationPreference::MAINTAIN_FRAMERATE;
+    case RTCDegradationPreferenceMaintainResolution:
+      return webrtc::DegradationPreference::MAINTAIN_RESOLUTION;
+    case RTCDegradationPreferenceBalanced:
+      return webrtc::DegradationPreference::BALANCED;
+  }
+}
+
++ (NSNumber *)degradationPreferenceFromNativeDegradationPreference:
+    (absl::optional<webrtc::DegradationPreference>)nativeDegradationPreference {
+  if (!nativeDegradationPreference.has_value()) {
+    return nil;
+  }
+
+  switch (*nativeDegradationPreference) {
+    case webrtc::DegradationPreference::DISABLED:
+      return @(RTCDegradationPreferenceDisabled);
+    case webrtc::DegradationPreference::MAINTAIN_FRAMERATE:
+      return @(RTCDegradationPreferenceMaintainFramerate);
+    case webrtc::DegradationPreference::MAINTAIN_RESOLUTION:
+      return @(RTCDegradationPreferenceMaintainResolution);
+    case webrtc::DegradationPreference::BALANCED:
+      return @(RTCDegradationPreferenceBalanced);
+  }
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCRtpReceiver+Native.h b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCRtpReceiver+Native.h
new file mode 100644
index 0000000000..c15ce70079
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCRtpReceiver+Native.h
@@ -0,0 +1,32 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCRtpReceiver.h"
+
+#include "api/crypto/frame_decryptor_interface.h"
+#include "api/scoped_refptr.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+/**
+ * This class extension exposes methods that work directly with injectable C++ components.
+ */
+@interface RTC_OBJC_TYPE (RTCRtpReceiver)
+()
+
+    /** Sets a user defined frame decryptor that will decrypt the entire frame.
+     * This will decrypt the entire frame using the user provided decryption
+     * mechanism regardless of whether SRTP is enabled or not.
+     */
+    - (void)setFrameDecryptor : (rtc::scoped_refptr<webrtc::FrameDecryptorInterface>)frameDecryptor;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCRtpReceiver+Private.h b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCRtpReceiver+Private.h
new file mode 100644
index 0000000000..6aed0b4bc5
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCRtpReceiver+Private.h
@@ -0,0 +1,52 @@
+/*
+ *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCRtpReceiver.h"
+
+#include "api/rtp_receiver_interface.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+@class RTC_OBJC_TYPE(RTCPeerConnectionFactory);
+
+namespace webrtc {
+
+class RtpReceiverDelegateAdapter : public RtpReceiverObserverInterface {
+ public:
+  RtpReceiverDelegateAdapter(RTC_OBJC_TYPE(RTCRtpReceiver) * receiver);
+
+  void OnFirstPacketReceived(cricket::MediaType media_type) override;
+
+ private:
+  __weak RTC_OBJC_TYPE(RTCRtpReceiver) * receiver_;
+};
+
+}  // namespace webrtc
+
+@interface RTC_OBJC_TYPE (RTCRtpReceiver)
+()
+
+    @property(nonatomic,
+              readonly) rtc::scoped_refptr<webrtc::RtpReceiverInterface> nativeRtpReceiver;
+
+/** Initialize an RTCRtpReceiver with a native RtpReceiverInterface. */
+- (instancetype)initWithFactory:(RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory
+              nativeRtpReceiver:(rtc::scoped_refptr<webrtc::RtpReceiverInterface>)nativeRtpReceiver
+    NS_DESIGNATED_INITIALIZER;
+
++ (RTCRtpMediaType)mediaTypeForNativeMediaType:(cricket::MediaType)nativeMediaType;
+
++ (cricket::MediaType)nativeMediaTypeForMediaType:(RTCRtpMediaType)mediaType;
+
++ (NSString*)stringForMediaType:(RTCRtpMediaType)mediaType;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCRtpReceiver.h b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCRtpReceiver.h
new file mode 100644
index 0000000000..1e407fd71b
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCRtpReceiver.h
@@ -0,0 +1,86 @@
+/*
+ *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+#import "RTCMediaStreamTrack.h"
+#import "RTCRtpParameters.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+/** Represents the media type of the RtpReceiver. */
+typedef NS_ENUM(NSInteger, RTCRtpMediaType) {
+  RTCRtpMediaTypeAudio,
+  RTCRtpMediaTypeVideo,
+  RTCRtpMediaTypeData,
+  RTCRtpMediaTypeUnsupported,
+};
+
+@class RTC_OBJC_TYPE(RTCRtpReceiver);
+
+RTC_OBJC_EXPORT
+@protocol RTC_OBJC_TYPE
+(RTCRtpReceiverDelegate)<NSObject>
+
+    /** Called when the first RTP packet is received.
+     *
+     *  Note: Currently if there are multiple RtpReceivers of the same media type,
+     *  they will all call OnFirstPacketReceived at once.
+     *
+     *  For example, if we create three audio receivers, A/B/C, they will listen to
+     *  the same signal from the underneath network layer. Whenever the first audio packet
+     *  is received, the underneath signal will be fired. All the receivers A/B/C will be
+     *  notified and the callback of the receiver's delegate will be called.
+     *
+     *  The process is the same for video receivers.
+     */
+    - (void)rtpReceiver
+    : (RTC_OBJC_TYPE(RTCRtpReceiver) *)rtpReceiver didReceiveFirstPacketForMediaType
+    : (RTCRtpMediaType)mediaType;
+
+@end
+
+RTC_OBJC_EXPORT
+@protocol RTC_OBJC_TYPE
+(RTCRtpReceiver)<NSObject>
+
+    /** A unique identifier for this receiver. */
+    @property(nonatomic, readonly) NSString *receiverId;
+
+/** The currently active RTCRtpParameters, as defined in
+ *  https://www.w3.org/TR/webrtc/#idl-def-RTCRtpParameters.
+ *
+ *  The WebRTC specification only defines RTCRtpParameters in terms of senders,
+ *  but this API also applies them to receivers, similar to ORTC:
+ *  http://ortc.org/wp-content/uploads/2016/03/ortc.html#rtcrtpparameters*.
+ */
+@property(nonatomic, readonly) RTC_OBJC_TYPE(RTCRtpParameters) * parameters;
+
+/** The RTCMediaStreamTrack associated with the receiver.
+ *  Note: reading this property returns a new instance of
+ *  RTCMediaStreamTrack. Use isEqual: instead of == to compare
+ *  RTCMediaStreamTrack instances.
+ */
+@property(nonatomic, readonly, nullable) RTC_OBJC_TYPE(RTCMediaStreamTrack) * track;
+
+/** The delegate for this RtpReceiver. */
+@property(nonatomic, weak) id<RTC_OBJC_TYPE(RTCRtpReceiverDelegate)> delegate;
+
+@end
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCRtpReceiver) : NSObject <RTC_OBJC_TYPE(RTCRtpReceiver)>
+
+- (instancetype)init NS_UNAVAILABLE;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCRtpReceiver.mm b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCRtpReceiver.mm
new file mode 100644
index 0000000000..60af86ac1b
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCRtpReceiver.mm
@@ -0,0 +1,159 @@
+/*
+ *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCRtpReceiver+Private.h"
+
+#import "RTCMediaStreamTrack+Private.h"
+#import "RTCRtpParameters+Private.h"
+#import "RTCRtpReceiver+Native.h"
+#import "base/RTCLogging.h"
+#import "helpers/NSString+StdString.h"
+
+#include "api/media_stream_interface.h"
+
+namespace webrtc {
+
+RtpReceiverDelegateAdapter::RtpReceiverDelegateAdapter(RTC_OBJC_TYPE(RTCRtpReceiver) * receiver) {
+  RTC_CHECK(receiver);
+  receiver_ = receiver;
+}
+
+void RtpReceiverDelegateAdapter::OnFirstPacketReceived(
+    cricket::MediaType media_type) {
+  RTCRtpMediaType packet_media_type =
+      [RTC_OBJC_TYPE(RTCRtpReceiver) mediaTypeForNativeMediaType:media_type];
+  RTC_OBJC_TYPE(RTCRtpReceiver) *receiver = receiver_;
+  [receiver.delegate rtpReceiver:receiver didReceiveFirstPacketForMediaType:packet_media_type];
+}
+
+}  // namespace webrtc
+
+@implementation RTC_OBJC_TYPE (RTCRtpReceiver) {
+  RTC_OBJC_TYPE(RTCPeerConnectionFactory) * _factory;
+  rtc::scoped_refptr<webrtc::RtpReceiverInterface> _nativeRtpReceiver;
+  std::unique_ptr<webrtc::RtpReceiverDelegateAdapter> _observer;
+}
+
+@synthesize delegate = _delegate;
+
+- (NSString *)receiverId {
+  return [NSString stringForStdString:_nativeRtpReceiver->id()];
+}
+
+- (RTC_OBJC_TYPE(RTCRtpParameters) *)parameters {
+  return [[RTC_OBJC_TYPE(RTCRtpParameters) alloc]
+      initWithNativeParameters:_nativeRtpReceiver->GetParameters()];
+}
+
+- (nullable RTC_OBJC_TYPE(RTCMediaStreamTrack) *)track {
+  rtc::scoped_refptr<webrtc::MediaStreamTrackInterface> nativeTrack(
+    _nativeRtpReceiver->track());
+  if (nativeTrack) {
+    return [RTC_OBJC_TYPE(RTCMediaStreamTrack) mediaTrackForNativeTrack:nativeTrack
+                                                                factory:_factory];
+  }
+  return nil;
+}
+
+- (NSString *)description {
+  return [NSString
+      stringWithFormat:@"RTC_OBJC_TYPE(RTCRtpReceiver) {\n  receiverId: %@\n}", self.receiverId];
+}
+
+- (void)dealloc {
+  if (_nativeRtpReceiver) {
+    _nativeRtpReceiver->SetObserver(nullptr);
+  }
+}
+
+- (BOOL)isEqual:(id)object {
+  if (self == object) {
+    return YES;
+  }
+  if (object == nil) {
+    return NO;
+  }
+  if (![object isMemberOfClass:[self class]]) {
+    return NO;
+  }
+  RTC_OBJC_TYPE(RTCRtpReceiver) *receiver = (RTC_OBJC_TYPE(RTCRtpReceiver) *)object;
+  return _nativeRtpReceiver == receiver.nativeRtpReceiver;
+}
+
+- (NSUInteger)hash {
+  return (NSUInteger)_nativeRtpReceiver.get();
+}
+
+#pragma mark - Native
+
+- (void)setFrameDecryptor:(rtc::scoped_refptr<webrtc::FrameDecryptorInterface>)frameDecryptor {
+  _nativeRtpReceiver->SetFrameDecryptor(frameDecryptor);
+}
+
+#pragma mark - Private
+
+- (rtc::scoped_refptr<webrtc::RtpReceiverInterface>)nativeRtpReceiver {
+  return _nativeRtpReceiver;
+}
+
+- (instancetype)initWithFactory:(RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory
+              nativeRtpReceiver:
+                  (rtc::scoped_refptr<webrtc::RtpReceiverInterface>)nativeRtpReceiver {
+  if (self = [super init]) {
+    _factory = factory;
+    _nativeRtpReceiver = nativeRtpReceiver;
+    RTCLogInfo(@"RTC_OBJC_TYPE(RTCRtpReceiver)(%p): created receiver: %@", self, self.description);
+    _observer.reset(new webrtc::RtpReceiverDelegateAdapter(self));
+    _nativeRtpReceiver->SetObserver(_observer.get());
+  }
+  return self;
+}
+
++ (RTCRtpMediaType)mediaTypeForNativeMediaType:
+    (cricket::MediaType)nativeMediaType {
+  switch (nativeMediaType) {
+    case cricket::MEDIA_TYPE_AUDIO:
+      return RTCRtpMediaTypeAudio;
+    case cricket::MEDIA_TYPE_VIDEO:
+      return RTCRtpMediaTypeVideo;
+    case cricket::MEDIA_TYPE_DATA:
+      return RTCRtpMediaTypeData;
+    case cricket::MEDIA_TYPE_UNSUPPORTED:
+      return RTCRtpMediaTypeUnsupported;
+  }
+}
+
++ (cricket::MediaType)nativeMediaTypeForMediaType:(RTCRtpMediaType)mediaType {
+  switch (mediaType) {
+    case RTCRtpMediaTypeAudio:
+      return cricket::MEDIA_TYPE_AUDIO;
+    case RTCRtpMediaTypeVideo:
+      return cricket::MEDIA_TYPE_VIDEO;
+    case RTCRtpMediaTypeData:
+      return cricket::MEDIA_TYPE_DATA;
+    case RTCRtpMediaTypeUnsupported:
+      return cricket::MEDIA_TYPE_UNSUPPORTED;
+  }
+}
+
++ (NSString *)stringForMediaType:(RTCRtpMediaType)mediaType {
+  switch (mediaType) {
+    case RTCRtpMediaTypeAudio:
+      return @"AUDIO";
+    case RTCRtpMediaTypeVideo:
+      return @"VIDEO";
+    case RTCRtpMediaTypeData:
+      return @"DATA";
+    case RTCRtpMediaTypeUnsupported:
+      return @"UNSUPPORTED";
+  }
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCRtpSender+Native.h b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCRtpSender+Native.h
new file mode 100644
index 0000000000..249d5c5e09
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCRtpSender+Native.h
@@ -0,0 +1,33 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCRtpSender.h"
+
+#include "api/crypto/frame_encryptor_interface.h"
+#include "api/scoped_refptr.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+/**
+ * This class extension exposes methods that work directly with injectable C++ components.
+ */
+@interface RTC_OBJC_TYPE (RTCRtpSender)
+()
+
+    /** Sets a defined frame encryptor that will encrypt the entire frame
+     * before it is sent across the network. This will encrypt the entire frame
+     * using the user provided encryption mechanism regardless of whether SRTP is
+     * enabled or not.
+     */
+    - (void)setFrameEncryptor : (rtc::scoped_refptr<webrtc::FrameEncryptorInterface>)frameEncryptor;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCRtpSender+Private.h b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCRtpSender+Private.h
new file mode 100644
index 0000000000..6fdb42bb22
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCRtpSender+Private.h
@@ -0,0 +1,31 @@
+/*
+ *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCRtpSender.h"
+
+#include "api/rtp_sender_interface.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+@class RTC_OBJC_TYPE(RTCPeerConnectionFactory);
+
+@interface RTC_OBJC_TYPE (RTCRtpSender)
+()
+
+    @property(nonatomic, readonly) rtc::scoped_refptr<webrtc::RtpSenderInterface> nativeRtpSender;
+
+/** Initialize an RTCRtpSender with a native RtpSenderInterface. */
+- (instancetype)initWithFactory:(RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory
+                nativeRtpSender:(rtc::scoped_refptr<webrtc::RtpSenderInterface>)nativeRtpSender
+    NS_DESIGNATED_INITIALIZER;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCRtpSender.h b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCRtpSender.h
new file mode 100644
index 0000000000..fcdf199869
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCRtpSender.h
@@ -0,0 +1,54 @@
+/*
+ *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCDtmfSender.h"
+#import "RTCMacros.h"
+#import "RTCMediaStreamTrack.h"
+#import "RTCRtpParameters.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+RTC_OBJC_EXPORT
+@protocol RTC_OBJC_TYPE
+(RTCRtpSender)<NSObject>
+
+/** A unique identifier for this sender. */
+@property(nonatomic, readonly) NSString *senderId;
+
+/** The currently active RTCRtpParameters, as defined in
+ *  https://www.w3.org/TR/webrtc/#idl-def-RTCRtpParameters.
+ */
+@property(nonatomic, copy) RTC_OBJC_TYPE(RTCRtpParameters) * parameters;
+
+/** The RTCMediaStreamTrack associated with the sender.
+ *  Note: reading this property returns a new instance of
+ *  RTCMediaStreamTrack. Use isEqual: instead of == to compare
+ *  RTCMediaStreamTrack instances.
+ */
+@property(nonatomic, copy, nullable) RTC_OBJC_TYPE(RTCMediaStreamTrack) * track;
+
+/** IDs of streams associated with the RTP sender */
+@property(nonatomic, copy) NSArray<NSString *> *streamIds;
+
+/** The RTCDtmfSender accociated with the RTP sender. */
+@property(nonatomic, readonly, nullable) id<RTC_OBJC_TYPE(RTCDtmfSender)> dtmfSender;
+
+@end
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCRtpSender) : NSObject <RTC_OBJC_TYPE(RTCRtpSender)>
+
+- (instancetype)init NS_UNAVAILABLE;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCRtpSender.mm b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCRtpSender.mm
new file mode 100644
index 0000000000..4fadb30f49
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCRtpSender.mm
@@ -0,0 +1,132 @@
+/*
+ *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCRtpSender+Private.h"
+
+#import "RTCDtmfSender+Private.h"
+#import "RTCMediaStreamTrack+Private.h"
+#import "RTCRtpParameters+Private.h"
+#import "RTCRtpSender+Native.h"
+#import "base/RTCLogging.h"
+#import "helpers/NSString+StdString.h"
+
+#include "api/media_stream_interface.h"
+
+@implementation RTC_OBJC_TYPE (RTCRtpSender) {
+  RTC_OBJC_TYPE(RTCPeerConnectionFactory) * _factory;
+  rtc::scoped_refptr<webrtc::RtpSenderInterface> _nativeRtpSender;
+}
+
+@synthesize dtmfSender = _dtmfSender;
+
+- (NSString *)senderId {
+  return [NSString stringForStdString:_nativeRtpSender->id()];
+}
+
+- (RTC_OBJC_TYPE(RTCRtpParameters) *)parameters {
+  return [[RTC_OBJC_TYPE(RTCRtpParameters) alloc]
+      initWithNativeParameters:_nativeRtpSender->GetParameters()];
+}
+
+- (void)setParameters:(RTC_OBJC_TYPE(RTCRtpParameters) *)parameters {
+  if (!_nativeRtpSender->SetParameters(parameters.nativeParameters).ok()) {
+    RTCLogError(@"RTC_OBJC_TYPE(RTCRtpSender)(%p): Failed to set parameters: %@", self, parameters);
+  }
+}
+
+- (RTC_OBJC_TYPE(RTCMediaStreamTrack) *)track {
+  rtc::scoped_refptr<webrtc::MediaStreamTrackInterface> nativeTrack(
+    _nativeRtpSender->track());
+  if (nativeTrack) {
+    return [RTC_OBJC_TYPE(RTCMediaStreamTrack) mediaTrackForNativeTrack:nativeTrack
+                                                                factory:_factory];
+  }
+  return nil;
+}
+
+- (void)setTrack:(RTC_OBJC_TYPE(RTCMediaStreamTrack) *)track {
+  if (!_nativeRtpSender->SetTrack(track.nativeTrack.get())) {
+    RTCLogError(@"RTC_OBJC_TYPE(RTCRtpSender)(%p): Failed to set track %@", self, track);
+  }
+}
+
+- (NSArray<NSString *> *)streamIds {
+  std::vector<std::string> nativeStreamIds = _nativeRtpSender->stream_ids();
+  NSMutableArray *streamIds = [NSMutableArray arrayWithCapacity:nativeStreamIds.size()];
+  for (const auto &s : nativeStreamIds) {
+    [streamIds addObject:[NSString stringForStdString:s]];
+  }
+  return streamIds;
+}
+
+- (void)setStreamIds:(NSArray<NSString *> *)streamIds {
+  std::vector<std::string> nativeStreamIds;
+  for (NSString *streamId in streamIds) {
+    nativeStreamIds.push_back([streamId UTF8String]);
+  }
+  _nativeRtpSender->SetStreams(nativeStreamIds);
+}
+
+- (NSString *)description {
+  return [NSString
+      stringWithFormat:@"RTC_OBJC_TYPE(RTCRtpSender) {\n  senderId: %@\n}", self.senderId];
+}
+
+- (BOOL)isEqual:(id)object {
+  if (self == object) {
+    return YES;
+  }
+  if (object == nil) {
+    return NO;
+  }
+  if (![object isMemberOfClass:[self class]]) {
+    return NO;
+  }
+  RTC_OBJC_TYPE(RTCRtpSender) *sender = (RTC_OBJC_TYPE(RTCRtpSender) *)object;
+  return _nativeRtpSender == sender.nativeRtpSender;
+}
+
+- (NSUInteger)hash {
+  return (NSUInteger)_nativeRtpSender.get();
+}
+
+#pragma mark - Native
+
+- (void)setFrameEncryptor:(rtc::scoped_refptr<webrtc::FrameEncryptorInterface>)frameEncryptor {
+  _nativeRtpSender->SetFrameEncryptor(frameEncryptor);
+}
+
+#pragma mark - Private
+
+- (rtc::scoped_refptr<webrtc::RtpSenderInterface>)nativeRtpSender {
+  return _nativeRtpSender;
+}
+
+- (instancetype)initWithFactory:(RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory
+                nativeRtpSender:(rtc::scoped_refptr<webrtc::RtpSenderInterface>)nativeRtpSender {
+  NSParameterAssert(factory);
+  NSParameterAssert(nativeRtpSender);
+  if (self = [super init]) {
+    _factory = factory;
+    _nativeRtpSender = nativeRtpSender;
+    if (_nativeRtpSender->media_type() == cricket::MEDIA_TYPE_AUDIO) {
+      rtc::scoped_refptr<webrtc::DtmfSenderInterface> nativeDtmfSender(
+          _nativeRtpSender->GetDtmfSender());
+      if (nativeDtmfSender) {
+        _dtmfSender =
+            [[RTC_OBJC_TYPE(RTCDtmfSender) alloc] initWithNativeDtmfSender:nativeDtmfSender];
+      }
+    }
+    RTCLogInfo(@"RTC_OBJC_TYPE(RTCRtpSender)(%p): created sender: %@", self, self.description);
+  }
+  return self;
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCRtpTransceiver+Private.h b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCRtpTransceiver+Private.h
new file mode 100644
index 0000000000..65d45fb88e
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCRtpTransceiver+Private.h
@@ -0,0 +1,46 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCRtpTransceiver.h"
+
+#include "api/rtp_transceiver_interface.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+@class RTC_OBJC_TYPE(RTCPeerConnectionFactory);
+
+@interface RTC_OBJC_TYPE (RTCRtpTransceiverInit)
+()
+
+    @property(nonatomic, readonly) webrtc::RtpTransceiverInit nativeInit;
+
+@end
+
+@interface RTC_OBJC_TYPE (RTCRtpTransceiver)
+()
+
+    @property(nonatomic,
+              readonly) rtc::scoped_refptr<webrtc::RtpTransceiverInterface> nativeRtpTransceiver;
+
+/** Initialize an RTCRtpTransceiver with a native RtpTransceiverInterface. */
+- (instancetype)initWithFactory:(RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory
+           nativeRtpTransceiver:
+               (rtc::scoped_refptr<webrtc::RtpTransceiverInterface>)nativeRtpTransceiver
+    NS_DESIGNATED_INITIALIZER;
+
++ (webrtc::RtpTransceiverDirection)nativeRtpTransceiverDirectionFromDirection:
+        (RTCRtpTransceiverDirection)direction;
+
++ (RTCRtpTransceiverDirection)rtpTransceiverDirectionFromNativeDirection:
+        (webrtc::RtpTransceiverDirection)nativeDirection;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCRtpTransceiver.h b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCRtpTransceiver.h
new file mode 100644
index 0000000000..fd59013639
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCRtpTransceiver.h
@@ -0,0 +1,137 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+#import "RTCRtpReceiver.h"
+#import "RTCRtpSender.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+extern NSString *const kRTCRtpTransceiverErrorDomain;
+
+/** https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiverdirection */
+typedef NS_ENUM(NSInteger, RTCRtpTransceiverDirection) {
+  RTCRtpTransceiverDirectionSendRecv,
+  RTCRtpTransceiverDirectionSendOnly,
+  RTCRtpTransceiverDirectionRecvOnly,
+  RTCRtpTransceiverDirectionInactive,
+  RTCRtpTransceiverDirectionStopped
+};
+
+/** Structure for initializing an RTCRtpTransceiver in a call to
+ *  RTCPeerConnection.addTransceiver.
+ *  https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiverinit
+ */
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCRtpTransceiverInit) : NSObject
+
+/** Direction of the RTCRtpTransceiver. See RTCRtpTransceiver.direction. */
+@property(nonatomic) RTCRtpTransceiverDirection direction;
+
+/** The added RTCRtpTransceiver will be added to these streams. */
+@property(nonatomic) NSArray<NSString *> *streamIds;
+
+/** TODO(bugs.webrtc.org/7600): Not implemented. */
+@property(nonatomic) NSArray<RTC_OBJC_TYPE(RTCRtpEncodingParameters) *> *sendEncodings;
+
+@end
+
+@class RTC_OBJC_TYPE(RTCRtpTransceiver);
+
+/** The RTCRtpTransceiver maps to the RTCRtpTransceiver defined by the
+ *  WebRTC specification. A transceiver represents a combination of an RTCRtpSender
+ *  and an RTCRtpReceiver that share a common mid. As defined in JSEP, an
+ *  RTCRtpTransceiver is said to be associated with a media description if its
+ *  mid property is non-nil; otherwise, it is said to be disassociated.
+ *  JSEP: https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-24
+ *
+ *  Note that RTCRtpTransceivers are only supported when using
+ *  RTCPeerConnection with Unified Plan SDP.
+ *
+ *  WebRTC specification for RTCRtpTransceiver, the JavaScript analog:
+ *  https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver
+ */
+RTC_OBJC_EXPORT
+@protocol RTC_OBJC_TYPE
+(RTCRtpTransceiver)<NSObject>
+
+    /** Media type of the transceiver. The sender and receiver will also have this
+     *  type.
+     */
+    @property(nonatomic, readonly) RTCRtpMediaType mediaType;
+
+/** The mid attribute is the mid negotiated and present in the local and
+ *  remote descriptions. Before negotiation is complete, the mid value may be
+ *  nil. After rollbacks, the value may change from a non-nil value to nil.
+ *  https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-mid
+ */
+@property(nonatomic, readonly) NSString *mid;
+
+/** The sender attribute exposes the RTCRtpSender corresponding to the RTP
+ *  media that may be sent with the transceiver's mid. The sender is always
+ *  present, regardless of the direction of media.
+ *  https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-sender
+ */
+@property(nonatomic, readonly) RTC_OBJC_TYPE(RTCRtpSender) * sender;
+
+/** The receiver attribute exposes the RTCRtpReceiver corresponding to the RTP
+ *  media that may be received with the transceiver's mid. The receiver is
+ *  always present, regardless of the direction of media.
+ *  https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-receiver
+ */
+@property(nonatomic, readonly) RTC_OBJC_TYPE(RTCRtpReceiver) * receiver;
+
+/** The isStopped attribute indicates that the sender of this transceiver will
+ *  no longer send, and that the receiver will no longer receive. It is true if
+ *  either stop has been called or if setting the local or remote description
+ *  has caused the RTCRtpTransceiver to be stopped.
+ *  https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-stopped
+ */
+@property(nonatomic, readonly) BOOL isStopped;
+
+/** The direction attribute indicates the preferred direction of this
+ *  transceiver, which will be used in calls to createOffer and createAnswer.
+ *  https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-direction
+ */
+@property(nonatomic, readonly) RTCRtpTransceiverDirection direction;
+
+/** The currentDirection attribute indicates the current direction negotiated
+ *  for this transceiver. If this transceiver has never been represented in an
+ *  offer/answer exchange, or if the transceiver is stopped, the value is not
+ *  present and this method returns NO.
+ *  https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-currentdirection
+ */
+- (BOOL)currentDirection:(RTCRtpTransceiverDirection *)currentDirectionOut;
+
+/** The stop method irreversibly stops the RTCRtpTransceiver. The sender of
+ *  this transceiver will no longer send, the receiver will no longer receive.
+ *  https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-stop
+ */
+- (void)stopInternal;
+
+/** An update of directionality does not take effect immediately. Instead,
+ *  future calls to createOffer and createAnswer mark the corresponding media
+ *  descriptions as sendrecv, sendonly, recvonly, or inactive.
+ *  https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-direction
+ */
+- (void)setDirection:(RTCRtpTransceiverDirection)direction error:(NSError **)error;
+
+@end
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCRtpTransceiver) : NSObject <RTC_OBJC_TYPE(RTCRtpTransceiver)>
+
+- (instancetype)init NS_UNAVAILABLE;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCRtpTransceiver.mm b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCRtpTransceiver.mm
new file mode 100644
index 0000000000..ae1cf79864
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCRtpTransceiver.mm
@@ -0,0 +1,190 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCRtpTransceiver+Private.h"
+
+#import "RTCRtpEncodingParameters+Private.h"
+#import "RTCRtpParameters+Private.h"
+#import "RTCRtpReceiver+Private.h"
+#import "RTCRtpSender+Private.h"
+#import "base/RTCLogging.h"
+#import "helpers/NSString+StdString.h"
+
+NSString *const kRTCRtpTransceiverErrorDomain = @"org.webrtc.RTCRtpTranceiver";
+
+@implementation RTC_OBJC_TYPE (RTCRtpTransceiverInit)
+
+@synthesize direction = _direction;
+@synthesize streamIds = _streamIds;
+@synthesize sendEncodings = _sendEncodings;
+
+- (instancetype)init {
+  if (self = [super init]) {
+    _direction = RTCRtpTransceiverDirectionSendRecv;
+  }
+  return self;
+}
+
+- (webrtc::RtpTransceiverInit)nativeInit {
+  webrtc::RtpTransceiverInit init;
+  init.direction =
+      [RTC_OBJC_TYPE(RTCRtpTransceiver) nativeRtpTransceiverDirectionFromDirection:_direction];
+  for (NSString *streamId in _streamIds) {
+    init.stream_ids.push_back([streamId UTF8String]);
+  }
+  for (RTC_OBJC_TYPE(RTCRtpEncodingParameters) * sendEncoding in _sendEncodings) {
+    init.send_encodings.push_back(sendEncoding.nativeParameters);
+  }
+  return init;
+}
+
+@end
+
+@implementation RTC_OBJC_TYPE (RTCRtpTransceiver) {
+  RTC_OBJC_TYPE(RTCPeerConnectionFactory) * _factory;
+  rtc::scoped_refptr<webrtc::RtpTransceiverInterface> _nativeRtpTransceiver;
+}
+
+- (RTCRtpMediaType)mediaType {
+  return [RTC_OBJC_TYPE(RTCRtpReceiver)
+      mediaTypeForNativeMediaType:_nativeRtpTransceiver->media_type()];
+}
+
+- (NSString *)mid {
+  if (_nativeRtpTransceiver->mid()) {
+    return [NSString stringForStdString:*_nativeRtpTransceiver->mid()];
+  } else {
+    return nil;
+  }
+}
+
+@synthesize sender = _sender;
+@synthesize receiver = _receiver;
+
+- (BOOL)isStopped {
+  return _nativeRtpTransceiver->stopped();
+}
+
+- (RTCRtpTransceiverDirection)direction {
+  return [RTC_OBJC_TYPE(RTCRtpTransceiver)
+      rtpTransceiverDirectionFromNativeDirection:_nativeRtpTransceiver->direction()];
+}
+
+- (void)setDirection:(RTCRtpTransceiverDirection)direction error:(NSError **)error {
+  webrtc::RTCError nativeError = _nativeRtpTransceiver->SetDirectionWithError(
+      [RTC_OBJC_TYPE(RTCRtpTransceiver) nativeRtpTransceiverDirectionFromDirection:direction]);
+
+  if (!nativeError.ok() && error) {
+    *error = [NSError errorWithDomain:kRTCRtpTransceiverErrorDomain
+                                 code:static_cast<int>(nativeError.type())
+                             userInfo:@{
+                               @"message" : [NSString stringWithCString:nativeError.message()
+                                                               encoding:NSUTF8StringEncoding]
+                             }];
+  }
+}
+
+- (BOOL)currentDirection:(RTCRtpTransceiverDirection *)currentDirectionOut {
+  if (_nativeRtpTransceiver->current_direction()) {
+    *currentDirectionOut = [RTC_OBJC_TYPE(RTCRtpTransceiver)
+        rtpTransceiverDirectionFromNativeDirection:*_nativeRtpTransceiver->current_direction()];
+    return YES;
+  } else {
+    return NO;
+  }
+}
+
+- (void)stopInternal {
+  _nativeRtpTransceiver->StopInternal();
+}
+
+- (NSString *)description {
+  return [NSString
+      stringWithFormat:@"RTC_OBJC_TYPE(RTCRtpTransceiver) {\n  sender: %@\n  receiver: %@\n}",
+                       _sender,
+                       _receiver];
+}
+
+- (BOOL)isEqual:(id)object {
+  if (self == object) {
+    return YES;
+  }
+  if (object == nil) {
+    return NO;
+  }
+  if (![object isMemberOfClass:[self class]]) {
+    return NO;
+  }
+  RTC_OBJC_TYPE(RTCRtpTransceiver) *transceiver = (RTC_OBJC_TYPE(RTCRtpTransceiver) *)object;
+  return _nativeRtpTransceiver == transceiver.nativeRtpTransceiver;
+}
+
+- (NSUInteger)hash {
+  return (NSUInteger)_nativeRtpTransceiver.get();
+}
+
+#pragma mark - Private
+
+- (rtc::scoped_refptr<webrtc::RtpTransceiverInterface>)nativeRtpTransceiver {
+  return _nativeRtpTransceiver;
+}
+
+- (instancetype)initWithFactory:(RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory
+           nativeRtpTransceiver:
+               (rtc::scoped_refptr<webrtc::RtpTransceiverInterface>)nativeRtpTransceiver {
+  NSParameterAssert(factory);
+  NSParameterAssert(nativeRtpTransceiver);
+  if (self = [super init]) {
+    _factory = factory;
+    _nativeRtpTransceiver = nativeRtpTransceiver;
+    _sender = [[RTC_OBJC_TYPE(RTCRtpSender) alloc] initWithFactory:_factory
+                                                   nativeRtpSender:nativeRtpTransceiver->sender()];
+    _receiver =
+        [[RTC_OBJC_TYPE(RTCRtpReceiver) alloc] initWithFactory:_factory
+                                             nativeRtpReceiver:nativeRtpTransceiver->receiver()];
+    RTCLogInfo(
+        @"RTC_OBJC_TYPE(RTCRtpTransceiver)(%p): created transceiver: %@", self, self.description);
+  }
+  return self;
+}
+
++ (webrtc::RtpTransceiverDirection)nativeRtpTransceiverDirectionFromDirection:
+        (RTCRtpTransceiverDirection)direction {
+  switch (direction) {
+    case RTCRtpTransceiverDirectionSendRecv:
+      return webrtc::RtpTransceiverDirection::kSendRecv;
+    case RTCRtpTransceiverDirectionSendOnly:
+      return webrtc::RtpTransceiverDirection::kSendOnly;
+    case RTCRtpTransceiverDirectionRecvOnly:
+      return webrtc::RtpTransceiverDirection::kRecvOnly;
+    case RTCRtpTransceiverDirectionInactive:
+      return webrtc::RtpTransceiverDirection::kInactive;
+    case RTCRtpTransceiverDirectionStopped:
+      return webrtc::RtpTransceiverDirection::kStopped;
+  }
+}
+
++ (RTCRtpTransceiverDirection)rtpTransceiverDirectionFromNativeDirection:
+        (webrtc::RtpTransceiverDirection)nativeDirection {
+  switch (nativeDirection) {
+    case webrtc::RtpTransceiverDirection::kSendRecv:
+      return RTCRtpTransceiverDirectionSendRecv;
+    case webrtc::RtpTransceiverDirection::kSendOnly:
+      return RTCRtpTransceiverDirectionSendOnly;
+    case webrtc::RtpTransceiverDirection::kRecvOnly:
+      return RTCRtpTransceiverDirectionRecvOnly;
+    case webrtc::RtpTransceiverDirection::kInactive:
+      return RTCRtpTransceiverDirectionInactive;
+    case webrtc::RtpTransceiverDirection::kStopped:
+      return RTCRtpTransceiverDirectionStopped;
+  }
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCSSLAdapter.h b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCSSLAdapter.h
new file mode 100644
index 0000000000..f68bc5e9e3
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCSSLAdapter.h
@@ -0,0 +1,20 @@
+/*
+ *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+/**
+ * Initialize and clean up the SSL library. Failure is fatal. These call the
+ * corresponding functions in webrtc/rtc_base/ssladapter.h.
+ */
+RTC_EXTERN BOOL RTCInitializeSSL(void);
+RTC_EXTERN BOOL RTCCleanupSSL(void);
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCSSLAdapter.mm b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCSSLAdapter.mm
new file mode 100644
index 0000000000..430249577b
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCSSLAdapter.mm
@@ -0,0 +1,26 @@
+/*
+ *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCSSLAdapter.h"
+
+#include "rtc_base/checks.h"
+#include "rtc_base/ssl_adapter.h"
+
+BOOL RTCInitializeSSL(void) {
+  BOOL initialized = rtc::InitializeSSL();
+  RTC_DCHECK(initialized);
+  return initialized;
+}
+
+BOOL RTCCleanupSSL(void) {
+  BOOL cleanedUp = rtc::CleanupSSL();
+  RTC_DCHECK(cleanedUp);
+  return cleanedUp;
+}
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCSessionDescription+Private.h b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCSessionDescription+Private.h
new file mode 100644
index 0000000000..aa087e557f
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCSessionDescription+Private.h
@@ -0,0 +1,42 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCSessionDescription.h"
+
+#include "api/jsep.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+@interface RTC_OBJC_TYPE (RTCSessionDescription)
+()
+
+    /**
+     * The native SessionDescriptionInterface representation of this
+     * RTCSessionDescription object. This is needed to pass to the underlying C++
+     * APIs.
+     */
+    @property(nonatomic,
+              readonly) std::unique_ptr<webrtc::SessionDescriptionInterface> nativeDescription;
+
+/**
+ * Initialize an RTCSessionDescription from a native
+ * SessionDescriptionInterface. No ownership is taken of the native session
+ * description.
+ */
+- (instancetype)initWithNativeDescription:
+        (const webrtc::SessionDescriptionInterface *)nativeDescription;
+
++ (std::string)stdStringForType:(RTCSdpType)type;
+
++ (RTCSdpType)typeForStdString:(const std::string &)string;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCSessionDescription.h b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCSessionDescription.h
new file mode 100644
index 0000000000..8a9479d5cf
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCSessionDescription.h
@@ -0,0 +1,48 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+/**
+ * Represents the session description type. This exposes the same types that are
+ * in C++, which doesn't include the rollback type that is in the W3C spec.
+ */
+typedef NS_ENUM(NSInteger, RTCSdpType) {
+  RTCSdpTypeOffer,
+  RTCSdpTypePrAnswer,
+  RTCSdpTypeAnswer,
+  RTCSdpTypeRollback,
+};
+
+NS_ASSUME_NONNULL_BEGIN
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCSessionDescription) : NSObject
+
+/** The type of session description. */
+@property(nonatomic, readonly) RTCSdpType type;
+
+/** The SDP string representation of this session description. */
+@property(nonatomic, readonly) NSString *sdp;
+
+- (instancetype)init NS_UNAVAILABLE;
+
+/** Initialize a session description with a type and SDP string. */
+- (instancetype)initWithType:(RTCSdpType)type sdp:(NSString *)sdp NS_DESIGNATED_INITIALIZER;
+
++ (NSString *)stringForType:(RTCSdpType)type;
+
++ (RTCSdpType)typeForString:(NSString *)string;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCSessionDescription.mm b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCSessionDescription.mm
new file mode 100644
index 0000000000..539c90b14c
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCSessionDescription.mm
@@ -0,0 +1,103 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCSessionDescription+Private.h"
+
+#import "base/RTCLogging.h"
+#import "helpers/NSString+StdString.h"
+
+#include "rtc_base/checks.h"
+
+@implementation RTC_OBJC_TYPE (RTCSessionDescription)
+
+@synthesize type = _type;
+@synthesize sdp = _sdp;
+
++ (NSString *)stringForType:(RTCSdpType)type {
+  std::string string = [[self class] stdStringForType:type];
+  return [NSString stringForStdString:string];
+}
+
++ (RTCSdpType)typeForString:(NSString *)string {
+  std::string typeString = string.stdString;
+  return [[self class] typeForStdString:typeString];
+}
+
+- (instancetype)initWithType:(RTCSdpType)type sdp:(NSString *)sdp {
+  if (self = [super init]) {
+    _type = type;
+    _sdp = [sdp copy];
+  }
+  return self;
+}
+
+- (NSString *)description {
+  return [NSString stringWithFormat:@"RTC_OBJC_TYPE(RTCSessionDescription):\n%@\n%@",
+                                    [[self class] stringForType:_type],
+                                    _sdp];
+}
+
+#pragma mark - Private
+
+- (std::unique_ptr<webrtc::SessionDescriptionInterface>)nativeDescription {
+  webrtc::SdpParseError error;
+
+  std::unique_ptr<webrtc::SessionDescriptionInterface> description(webrtc::CreateSessionDescription(
+      [[self class] stdStringForType:_type], _sdp.stdString, &error));
+
+  if (!description) {
+    RTCLogError(@"Failed to create session description: %s\nline: %s",
+                error.description.c_str(),
+                error.line.c_str());
+  }
+
+  return description;
+}
+
+- (instancetype)initWithNativeDescription:
+    (const webrtc::SessionDescriptionInterface *)nativeDescription {
+  NSParameterAssert(nativeDescription);
+  std::string sdp;
+  nativeDescription->ToString(&sdp);
+  RTCSdpType type = [[self class] typeForStdString:nativeDescription->type()];
+
+  return [self initWithType:type
+                        sdp:[NSString stringForStdString:sdp]];
+}
+
++ (std::string)stdStringForType:(RTCSdpType)type {
+  switch (type) {
+    case RTCSdpTypeOffer:
+      return webrtc::SessionDescriptionInterface::kOffer;
+    case RTCSdpTypePrAnswer:
+      return webrtc::SessionDescriptionInterface::kPrAnswer;
+    case RTCSdpTypeAnswer:
+      return webrtc::SessionDescriptionInterface::kAnswer;
+    case RTCSdpTypeRollback:
+      return webrtc::SessionDescriptionInterface::kRollback;
+  }
+}
+
++ (RTCSdpType)typeForStdString:(const std::string &)string {
+  if (string == webrtc::SessionDescriptionInterface::kOffer) {
+    return RTCSdpTypeOffer;
+  } else if (string == webrtc::SessionDescriptionInterface::kPrAnswer) {
+    return RTCSdpTypePrAnswer;
+  } else if (string == webrtc::SessionDescriptionInterface::kAnswer) {
+    return RTCSdpTypeAnswer;
+  } else if (string == webrtc::SessionDescriptionInterface::kRollback) {
+    return RTCSdpTypeRollback;
+  } else {
+    RTC_DCHECK_NOTREACHED();
+    return RTCSdpTypeOffer;
+  }
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCStatisticsReport+Private.h b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCStatisticsReport+Private.h
new file mode 100644
index 0000000000..47c5241d51
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCStatisticsReport+Private.h
@@ -0,0 +1,19 @@
+/*
+ *  Copyright 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCStatisticsReport.h"
+
+#include "api/stats/rtc_stats_report.h"
+
+@interface RTC_OBJC_TYPE (RTCStatisticsReport) (Private)
+
+- (instancetype)initWithReport : (const webrtc::RTCStatsReport &)report;
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCStatisticsReport.h b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCStatisticsReport.h
new file mode 100644
index 0000000000..06dbf48d88
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCStatisticsReport.h
@@ -0,0 +1,55 @@
+/*
+ *  Copyright 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+@class RTC_OBJC_TYPE(RTCStatistics);
+
+NS_ASSUME_NONNULL_BEGIN
+
+/** A statistics report. Encapsulates a number of RTCStatistics objects. */
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCStatisticsReport) : NSObject
+
+/** The timestamp of the report in microseconds since 1970-01-01T00:00:00Z. */
+@property(nonatomic, readonly) CFTimeInterval timestamp_us;
+
+/** RTCStatistics objects by id. */
+@property(nonatomic, readonly) NSDictionary<NSString *, RTC_OBJC_TYPE(RTCStatistics) *> *statistics;
+
+- (instancetype)init NS_UNAVAILABLE;
+
+@end
+
+/** A part of a report (a subreport) covering a certain area. */
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCStatistics) : NSObject
+
+/** The id of this subreport, e.g. "RTCMediaStreamTrack_receiver_2". */
+@property(nonatomic, readonly) NSString *id;
+
+/** The timestamp of the subreport in microseconds since 1970-01-01T00:00:00Z. */
+@property(nonatomic, readonly) CFTimeInterval timestamp_us;
+
+/** The type of the subreport, e.g. "track", "codec". */
+@property(nonatomic, readonly) NSString *type;
+
+/** The keys and values of the subreport, e.g. "totalFramesDuration = 5.551".
+    The values are either NSNumbers or NSStrings or NSArrays encapsulating NSNumbers
+    or NSStrings, or NSDictionary of NSString keys to NSNumber values. */
+@property(nonatomic, readonly) NSDictionary<NSString *, NSObject *> *values;
+
+- (instancetype)init NS_UNAVAILABLE;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCStatisticsReport.mm b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCStatisticsReport.mm
new file mode 100644
index 0000000000..bfe2424553
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCStatisticsReport.mm
@@ -0,0 +1,193 @@
+/*
+ *  Copyright 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCStatisticsReport+Private.h"
+
+#include "helpers/NSString+StdString.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+/** Converts a single value to a suitable NSNumber, NSString or NSArray containing NSNumbers
+    or NSStrings, or NSDictionary of NSString keys to NSNumber values.*/
+NSObject *ValueFromStatsMember(const RTCStatsMemberInterface *member) {
+  if (member->is_defined()) {
+    switch (member->type()) {
+      case RTCStatsMemberInterface::kBool:
+        return [NSNumber numberWithBool:*member->cast_to<RTCStatsMember<bool>>()];
+      case RTCStatsMemberInterface::kInt32:
+        return [NSNumber numberWithInt:*member->cast_to<RTCStatsMember<int32_t>>()];
+      case RTCStatsMemberInterface::kUint32:
+        return [NSNumber numberWithUnsignedInt:*member->cast_to<RTCStatsMember<uint32_t>>()];
+      case RTCStatsMemberInterface::kInt64:
+        return [NSNumber numberWithLong:*member->cast_to<RTCStatsMember<int64_t>>()];
+      case RTCStatsMemberInterface::kUint64:
+        return [NSNumber numberWithUnsignedLong:*member->cast_to<RTCStatsMember<uint64_t>>()];
+      case RTCStatsMemberInterface::kDouble:
+        return [NSNumber numberWithDouble:*member->cast_to<RTCStatsMember<double>>()];
+      case RTCStatsMemberInterface::kString:
+        return [NSString stringForStdString:*member->cast_to<RTCStatsMember<std::string>>()];
+      case RTCStatsMemberInterface::kSequenceBool: {
+        std::vector<bool> sequence = *member->cast_to<RTCStatsMember<std::vector<bool>>>();
+        NSMutableArray *array = [NSMutableArray arrayWithCapacity:sequence.size()];
+        for (auto item : sequence) {
+          [array addObject:[NSNumber numberWithBool:item]];
+        }
+        return [array copy];
+      }
+      case RTCStatsMemberInterface::kSequenceInt32: {
+        std::vector<int32_t> sequence = *member->cast_to<RTCStatsMember<std::vector<int32_t>>>();
+        NSMutableArray<NSNumber *> *array = [NSMutableArray arrayWithCapacity:sequence.size()];
+        for (const auto &item : sequence) {
+          [array addObject:[NSNumber numberWithInt:item]];
+        }
+        return [array copy];
+      }
+      case RTCStatsMemberInterface::kSequenceUint32: {
+        std::vector<uint32_t> sequence = *member->cast_to<RTCStatsMember<std::vector<uint32_t>>>();
+        NSMutableArray<NSNumber *> *array = [NSMutableArray arrayWithCapacity:sequence.size()];
+        for (const auto &item : sequence) {
+          [array addObject:[NSNumber numberWithUnsignedInt:item]];
+        }
+        return [array copy];
+      }
+      case RTCStatsMemberInterface::kSequenceInt64: {
+        std::vector<int64_t> sequence = *member->cast_to<RTCStatsMember<std::vector<int64_t>>>();
+        NSMutableArray<NSNumber *> *array = [NSMutableArray arrayWithCapacity:sequence.size()];
+        for (const auto &item : sequence) {
+          [array addObject:[NSNumber numberWithLong:item]];
+        }
+        return [array copy];
+      }
+      case RTCStatsMemberInterface::kSequenceUint64: {
+        std::vector<uint64_t> sequence = *member->cast_to<RTCStatsMember<std::vector<uint64_t>>>();
+        NSMutableArray<NSNumber *> *array = [NSMutableArray arrayWithCapacity:sequence.size()];
+        for (const auto &item : sequence) {
+          [array addObject:[NSNumber numberWithUnsignedLong:item]];
+        }
+        return [array copy];
+      }
+      case RTCStatsMemberInterface::kSequenceDouble: {
+        std::vector<double> sequence = *member->cast_to<RTCStatsMember<std::vector<double>>>();
+        NSMutableArray<NSNumber *> *array = [NSMutableArray arrayWithCapacity:sequence.size()];
+        for (const auto &item : sequence) {
+          [array addObject:[NSNumber numberWithDouble:item]];
+        }
+        return [array copy];
+      }
+      case RTCStatsMemberInterface::kSequenceString: {
+        std::vector<std::string> sequence =
+            *member->cast_to<RTCStatsMember<std::vector<std::string>>>();
+        NSMutableArray<NSString *> *array = [NSMutableArray arrayWithCapacity:sequence.size()];
+        for (const auto &item : sequence) {
+          [array addObject:[NSString stringForStdString:item]];
+        }
+        return [array copy];
+      }
+      case RTCStatsMemberInterface::kMapStringUint64: {
+        std::map<std::string, uint64_t> map =
+            *member->cast_to<RTCStatsMember<std::map<std::string, uint64_t>>>();
+        NSMutableDictionary<NSString *, NSNumber *> *dictionary =
+            [NSMutableDictionary dictionaryWithCapacity:map.size()];
+        for (const auto &item : map) {
+          dictionary[[NSString stringForStdString:item.first]] = @(item.second);
+        }
+        return [dictionary copy];
+      }
+      case RTCStatsMemberInterface::kMapStringDouble: {
+        std::map<std::string, double> map =
+            *member->cast_to<RTCStatsMember<std::map<std::string, double>>>();
+        NSMutableDictionary<NSString *, NSNumber *> *dictionary =
+            [NSMutableDictionary dictionaryWithCapacity:map.size()];
+        for (const auto &item : map) {
+          dictionary[[NSString stringForStdString:item.first]] = @(item.second);
+        }
+        return [dictionary copy];
+      }
+      default:
+        RTC_DCHECK_NOTREACHED();
+    }
+  }
+
+  return nil;
+}
+}  // namespace webrtc
+
+@implementation RTC_OBJC_TYPE (RTCStatistics)
+
+@synthesize id = _id;
+@synthesize timestamp_us = _timestamp_us;
+@synthesize type = _type;
+@synthesize values = _values;
+
+- (instancetype)initWithStatistics:(const webrtc::RTCStats &)statistics {
+  if (self = [super init]) {
+    _id = [NSString stringForStdString:statistics.id()];
+    _timestamp_us = statistics.timestamp().us();
+    _type = [NSString stringWithCString:statistics.type() encoding:NSUTF8StringEncoding];
+
+    NSMutableDictionary<NSString *, NSObject *> *values = [NSMutableDictionary dictionary];
+    for (const webrtc::RTCStatsMemberInterface *member : statistics.Members()) {
+      NSObject *value = ValueFromStatsMember(member);
+      if (value) {
+        NSString *name = [NSString stringWithCString:member->name() encoding:NSUTF8StringEncoding];
+        RTC_DCHECK(name.length > 0);
+        RTC_DCHECK(!values[name]);
+        values[name] = value;
+      }
+    }
+    _values = [values copy];
+  }
+
+  return self;
+}
+
+- (NSString *)description {
+  return [NSString stringWithFormat:@"id = %@, type = %@, timestamp = %.0f, values = %@",
+                                    self.id,
+                                    self.type,
+                                    self.timestamp_us,
+                                    self.values];
+}
+
+@end
+
+@implementation RTC_OBJC_TYPE (RTCStatisticsReport)
+
+@synthesize timestamp_us = _timestamp_us;
+@synthesize statistics = _statistics;
+
+- (NSString *)description {
+  return [NSString
+      stringWithFormat:@"timestamp = %.0f, statistics = %@", self.timestamp_us, self.statistics];
+}
+
+@end
+
+@implementation RTC_OBJC_TYPE (RTCStatisticsReport) (Private)
+
+- (instancetype)initWithReport : (const webrtc::RTCStatsReport &)report {
+  if (self = [super init]) {
+    _timestamp_us = report.timestamp().us();
+
+    NSMutableDictionary *statisticsById =
+        [NSMutableDictionary dictionaryWithCapacity:report.size()];
+    for (const auto &stat : report) {
+      RTC_OBJC_TYPE(RTCStatistics) *statistics =
+          [[RTC_OBJC_TYPE(RTCStatistics) alloc] initWithStatistics:stat];
+      statisticsById[statistics.id] = statistics;
+    }
+    _statistics = [statisticsById copy];
+  }
+
+  return self;
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCTracing.h b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCTracing.h
new file mode 100644
index 0000000000..5c66e5a63a
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCTracing.h
@@ -0,0 +1,21 @@
+/*
+ *  Copyright 2016 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+RTC_EXTERN void RTCSetupInternalTracer(void);
+/** Starts capture to specified file. Must be a valid writable path.
+ *  Returns YES if capture starts.
+ */
+RTC_EXTERN BOOL RTCStartInternalCapture(NSString* filePath);
+RTC_EXTERN void RTCStopInternalCapture(void);
+RTC_EXTERN void RTCShutdownInternalTracer(void);
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCTracing.mm b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCTracing.mm
new file mode 100644
index 0000000000..72f9f4da13
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCTracing.mm
@@ -0,0 +1,29 @@
+/*
+ *  Copyright 2016 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCTracing.h"
+
+#include "rtc_base/event_tracer.h"
+
+void RTCSetupInternalTracer(void) {
+  rtc::tracing::SetupInternalTracer();
+}
+
+BOOL RTCStartInternalCapture(NSString *filePath) {
+  return rtc::tracing::StartInternalCapture(filePath.UTF8String);
+}
+
+void RTCStopInternalCapture(void) {
+  rtc::tracing::StopInternalCapture();
+}
+
+void RTCShutdownInternalTracer(void) {
+  rtc::tracing::ShutdownInternalTracer();
+}
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCVideoCodecInfo+Private.h b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCVideoCodecInfo+Private.h
new file mode 100644
index 0000000000..5eff996c4f
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCVideoCodecInfo+Private.h
@@ -0,0 +1,26 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "base/RTCVideoCodecInfo.h"
+
+#include "api/video_codecs/sdp_video_format.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+/* Interface for converting to/from internal C++ formats. */
+@interface RTC_OBJC_TYPE (RTCVideoCodecInfo)
+(Private)
+
+    - (instancetype)initWithNativeSdpVideoFormat : (webrtc::SdpVideoFormat)format;
+- (webrtc::SdpVideoFormat)nativeSdpVideoFormat;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCVideoCodecInfo+Private.mm b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCVideoCodecInfo+Private.mm
new file mode 100644
index 0000000000..2eb8d366d2
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCVideoCodecInfo+Private.mm
@@ -0,0 +1,38 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCVideoCodecInfo+Private.h"
+
+#import "helpers/NSString+StdString.h"
+
+@implementation RTC_OBJC_TYPE (RTCVideoCodecInfo)
+(Private)
+
+    - (instancetype)initWithNativeSdpVideoFormat : (webrtc::SdpVideoFormat)format {
+  NSMutableDictionary *params = [NSMutableDictionary dictionary];
+  for (auto it = format.parameters.begin(); it != format.parameters.end(); ++it) {
+    [params setObject:[NSString stringForStdString:it->second]
+               forKey:[NSString stringForStdString:it->first]];
+  }
+  return [self initWithName:[NSString stringForStdString:format.name] parameters:params];
+}
+
+- (webrtc::SdpVideoFormat)nativeSdpVideoFormat {
+  std::map<std::string, std::string> parameters;
+  for (NSString *paramKey in self.parameters.allKeys) {
+    std::string key = [NSString stdStringForString:paramKey];
+    std::string value = [NSString stdStringForString:self.parameters[paramKey]];
+    parameters[key] = value;
+  }
+
+  return webrtc::SdpVideoFormat([NSString stdStringForString:self.name], parameters);
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCVideoEncoderSettings+Private.h b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCVideoEncoderSettings+Private.h
new file mode 100644
index 0000000000..8323b18dc1
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCVideoEncoderSettings+Private.h
@@ -0,0 +1,26 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "base/RTCVideoEncoderSettings.h"
+
+#include "modules/video_coding/include/video_codec_interface.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+/* Interfaces for converting to/from internal C++ formats. */
+@interface RTC_OBJC_TYPE (RTCVideoEncoderSettings)
+(Private)
+
+    - (instancetype)initWithNativeVideoCodec : (const webrtc::VideoCodec *__nullable)videoCodec;
+- (webrtc::VideoCodec)nativeVideoCodec;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCVideoEncoderSettings+Private.mm b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCVideoEncoderSettings+Private.mm
new file mode 100644
index 0000000000..dec3a61090
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCVideoEncoderSettings+Private.mm
@@ -0,0 +1,52 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCVideoEncoderSettings+Private.h"
+
+#import "helpers/NSString+StdString.h"
+
+@implementation RTC_OBJC_TYPE (RTCVideoEncoderSettings)
+(Private)
+
+    - (instancetype)initWithNativeVideoCodec : (const webrtc::VideoCodec *)videoCodec {
+  if (self = [super init]) {
+    if (videoCodec) {
+      const char *codecName = CodecTypeToPayloadString(videoCodec->codecType);
+      self.name = [NSString stringWithUTF8String:codecName];
+
+      self.width = videoCodec->width;
+      self.height = videoCodec->height;
+      self.startBitrate = videoCodec->startBitrate;
+      self.maxBitrate = videoCodec->maxBitrate;
+      self.minBitrate = videoCodec->minBitrate;
+      self.maxFramerate = videoCodec->maxFramerate;
+      self.qpMax = videoCodec->qpMax;
+      self.mode = (RTCVideoCodecMode)videoCodec->mode;
+    }
+  }
+
+  return self;
+}
+
+- (webrtc::VideoCodec)nativeVideoCodec {
+  webrtc::VideoCodec videoCodec;
+  videoCodec.width = self.width;
+  videoCodec.height = self.height;
+  videoCodec.startBitrate = self.startBitrate;
+  videoCodec.maxBitrate = self.maxBitrate;
+  videoCodec.minBitrate = self.minBitrate;
+  videoCodec.maxBitrate = self.maxBitrate;
+  videoCodec.qpMax = self.qpMax;
+  videoCodec.mode = (webrtc::VideoCodecMode)self.mode;
+
+  return videoCodec;
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCVideoSource+Private.h b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCVideoSource+Private.h
new file mode 100644
index 0000000000..8e475dd21e
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCVideoSource+Private.h
@@ -0,0 +1,51 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCVideoSource.h"
+
+#import "RTCMediaSource+Private.h"
+
+#include "api/media_stream_interface.h"
+#include "rtc_base/thread.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+@interface RTC_OBJC_TYPE (RTCVideoSource)
+()
+
+    /**
+     * The VideoTrackSourceInterface object passed to this RTCVideoSource during
+     * construction.
+     */
+    @property(nonatomic,
+              readonly) rtc::scoped_refptr<webrtc::VideoTrackSourceInterface> nativeVideoSource;
+
+/** Initialize an RTCVideoSource from a native VideoTrackSourceInterface. */
+- (instancetype)initWithFactory:(RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory
+              nativeVideoSource:
+                  (rtc::scoped_refptr<webrtc::VideoTrackSourceInterface>)nativeVideoSource
+    NS_DESIGNATED_INITIALIZER;
+
+- (instancetype)initWithFactory:(RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory
+              nativeMediaSource:(rtc::scoped_refptr<webrtc::MediaSourceInterface>)nativeMediaSource
+                           type:(RTCMediaSourceType)type NS_UNAVAILABLE;
+
+- (instancetype)initWithFactory:(RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory
+                signalingThread:(rtc::Thread *)signalingThread
+                   workerThread:(rtc::Thread *)workerThread;
+
+- (instancetype)initWithFactory:(RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory
+                signalingThread:(rtc::Thread *)signalingThread
+                   workerThread:(rtc::Thread *)workerThread
+                   isScreenCast:(BOOL)isScreenCast;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCVideoSource.h b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCVideoSource.h
new file mode 100644
index 0000000000..cdef8b89a1
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCVideoSource.h
@@ -0,0 +1,37 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+#import "RTCMediaSource.h"
+#import "RTCVideoCapturer.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+RTC_OBJC_EXPORT
+
+@interface RTC_OBJC_TYPE (RTCVideoSource) : RTC_OBJC_TYPE(RTCMediaSource) <RTC_OBJC_TYPE(RTCVideoCapturerDelegate)>
+
+- (instancetype)init NS_UNAVAILABLE;
+
+/**
+ * Calling this function will cause frames to be scaled down to the
+ * requested resolution. Also, frames will be cropped to match the
+ * requested aspect ratio, and frames will be dropped to match the
+ * requested fps. The requested aspect ratio is orientation agnostic and
+ * will be adjusted to maintain the input orientation, so it doesn't
+ * matter if e.g. 1280x720 or 720x1280 is requested.
+ */
+- (void)adaptOutputFormatToWidth:(int)width height:(int)height fps:(int)fps;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCVideoSource.mm b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCVideoSource.mm
new file mode 100644
index 0000000000..486ca93771
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCVideoSource.mm
@@ -0,0 +1,92 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCVideoSource+Private.h"
+
+#include "pc/video_track_source_proxy.h"
+#include "rtc_base/checks.h"
+#include "sdk/objc/native/src/objc_video_track_source.h"
+
+static webrtc::ObjCVideoTrackSource *getObjCVideoSource(
+    const rtc::scoped_refptr<webrtc::VideoTrackSourceInterface> nativeSource) {
+  webrtc::VideoTrackSourceProxy *proxy_source =
+      static_cast<webrtc::VideoTrackSourceProxy *>(nativeSource.get());
+  return static_cast<webrtc::ObjCVideoTrackSource *>(proxy_source->internal());
+}
+
+// TODO(magjed): Refactor this class and target ObjCVideoTrackSource only once
+// RTCAVFoundationVideoSource is gone. See http://crbug/webrtc/7177 for more
+// info.
+@implementation RTC_OBJC_TYPE (RTCVideoSource) {
+  rtc::scoped_refptr<webrtc::VideoTrackSourceInterface> _nativeVideoSource;
+}
+
+- (instancetype)initWithFactory:(RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory
+              nativeVideoSource:
+                  (rtc::scoped_refptr<webrtc::VideoTrackSourceInterface>)nativeVideoSource {
+  RTC_DCHECK(factory);
+  RTC_DCHECK(nativeVideoSource);
+  if (self = [super initWithFactory:factory
+                  nativeMediaSource:nativeVideoSource
+                               type:RTCMediaSourceTypeVideo]) {
+    _nativeVideoSource = nativeVideoSource;
+  }
+  return self;
+}
+
+- (instancetype)initWithFactory:(RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory
+              nativeMediaSource:(rtc::scoped_refptr<webrtc::MediaSourceInterface>)nativeMediaSource
+                           type:(RTCMediaSourceType)type {
+  RTC_DCHECK_NOTREACHED();
+  return nil;
+}
+
+- (instancetype)initWithFactory:(RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory
+                signalingThread:(rtc::Thread *)signalingThread
+                   workerThread:(rtc::Thread *)workerThread {
+  return [self initWithFactory:factory
+               signalingThread:signalingThread
+                  workerThread:workerThread
+                  isScreenCast:NO];
+}
+
+- (instancetype)initWithFactory:(RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory
+                signalingThread:(rtc::Thread *)signalingThread
+                   workerThread:(rtc::Thread *)workerThread
+                   isScreenCast:(BOOL)isScreenCast {
+  rtc::scoped_refptr<webrtc::ObjCVideoTrackSource> objCVideoTrackSource =
+      rtc::make_ref_counted<webrtc::ObjCVideoTrackSource>(isScreenCast);
+
+  return [self initWithFactory:factory
+             nativeVideoSource:webrtc::VideoTrackSourceProxy::Create(
+                                   signalingThread, workerThread, objCVideoTrackSource)];
+}
+
+- (NSString *)description {
+  NSString *stateString = [[self class] stringForState:self.state];
+  return [NSString stringWithFormat:@"RTC_OBJC_TYPE(RTCVideoSource)( %p ): %@", self, stateString];
+}
+
+- (void)capturer:(RTC_OBJC_TYPE(RTCVideoCapturer) *)capturer
+    didCaptureVideoFrame:(RTC_OBJC_TYPE(RTCVideoFrame) *)frame {
+  getObjCVideoSource(_nativeVideoSource)->OnCapturedFrame(frame);
+}
+
+- (void)adaptOutputFormatToWidth:(int)width height:(int)height fps:(int)fps {
+  getObjCVideoSource(_nativeVideoSource)->OnOutputFormatRequest(width, height, fps);
+}
+
+#pragma mark - Private
+
+- (rtc::scoped_refptr<webrtc::VideoTrackSourceInterface>)nativeVideoSource {
+  return _nativeVideoSource;
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCVideoTrack+Private.h b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCVideoTrack+Private.h
new file mode 100644
index 0000000000..f1a8d7e4ed
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCVideoTrack+Private.h
@@ -0,0 +1,30 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCVideoTrack.h"
+
+#include "api/media_stream_interface.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+@interface RTC_OBJC_TYPE (RTCVideoTrack)
+()
+
+    /** VideoTrackInterface created or passed in at construction. */
+    @property(nonatomic, readonly) rtc::scoped_refptr<webrtc::VideoTrackInterface> nativeVideoTrack;
+
+/** Initialize an RTCVideoTrack with its source and an id. */
+- (instancetype)initWithFactory:(RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory
+                         source:(RTC_OBJC_TYPE(RTCVideoSource) *)source
+                        trackId:(NSString *)trackId;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCVideoTrack.h b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCVideoTrack.h
new file mode 100644
index 0000000000..5382b7169f
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCVideoTrack.h
@@ -0,0 +1,38 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCMediaStreamTrack.h"
+
+#import "RTCMacros.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+@protocol RTC_OBJC_TYPE
+(RTCVideoRenderer);
+@class RTC_OBJC_TYPE(RTCPeerConnectionFactory);
+@class RTC_OBJC_TYPE(RTCVideoSource);
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCVideoTrack) : RTC_OBJC_TYPE(RTCMediaStreamTrack)
+
+/** The video source for this video track. */
+@property(nonatomic, readonly) RTC_OBJC_TYPE(RTCVideoSource) *source;
+
+- (instancetype)init NS_UNAVAILABLE;
+
+/** Register a renderer that will render all frames received on this track. */
+- (void)addRenderer:(id<RTC_OBJC_TYPE(RTCVideoRenderer)>)renderer;
+
+/** Deregister a renderer. */
+- (void)removeRenderer:(id<RTC_OBJC_TYPE(RTCVideoRenderer)>)renderer;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCVideoTrack.mm b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCVideoTrack.mm
new file mode 100644
index 0000000000..d4862e3748
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/peerconnection/RTCVideoTrack.mm
@@ -0,0 +1,125 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCVideoTrack+Private.h"
+
+#import "RTCMediaStreamTrack+Private.h"
+#import "RTCPeerConnectionFactory+Private.h"
+#import "RTCVideoSource+Private.h"
+#import "api/RTCVideoRendererAdapter+Private.h"
+#import "helpers/NSString+StdString.h"
+
+@implementation RTC_OBJC_TYPE (RTCVideoTrack) {
+  rtc::Thread *_workerThread;
+  NSMutableArray *_adapters /* accessed on _workerThread */;
+}
+
+@synthesize source = _source;
+
+- (instancetype)initWithFactory:(RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory
+                         source:(RTC_OBJC_TYPE(RTCVideoSource) *)source
+                        trackId:(NSString *)trackId {
+  NSParameterAssert(factory);
+  NSParameterAssert(source);
+  NSParameterAssert(trackId.length);
+  std::string nativeId = [NSString stdStringForString:trackId];
+  rtc::scoped_refptr<webrtc::VideoTrackInterface> track =
+      factory.nativeFactory->CreateVideoTrack(nativeId, source.nativeVideoSource.get());
+  if (self = [self initWithFactory:factory nativeTrack:track type:RTCMediaStreamTrackTypeVideo]) {
+    _source = source;
+  }
+  return self;
+}
+
+- (instancetype)initWithFactory:(RTC_OBJC_TYPE(RTCPeerConnectionFactory) *)factory
+                    nativeTrack:
+                        (rtc::scoped_refptr<webrtc::MediaStreamTrackInterface>)nativeMediaTrack
+                           type:(RTCMediaStreamTrackType)type {
+  NSParameterAssert(factory);
+  NSParameterAssert(nativeMediaTrack);
+  NSParameterAssert(type == RTCMediaStreamTrackTypeVideo);
+  if (self = [super initWithFactory:factory nativeTrack:nativeMediaTrack type:type]) {
+    _adapters = [NSMutableArray array];
+    _workerThread = factory.workerThread;
+  }
+  return self;
+}
+
+- (void)dealloc {
+  for (RTCVideoRendererAdapter *adapter in _adapters) {
+    self.nativeVideoTrack->RemoveSink(adapter.nativeVideoRenderer);
+  }
+}
+
+- (RTC_OBJC_TYPE(RTCVideoSource) *)source {
+  if (!_source) {
+    rtc::scoped_refptr<webrtc::VideoTrackSourceInterface> source(
+        self.nativeVideoTrack->GetSource());
+    if (source) {
+      _source = [[RTC_OBJC_TYPE(RTCVideoSource) alloc] initWithFactory:self.factory
+                                                     nativeVideoSource:source];
+    }
+  }
+  return _source;
+}
+
+- (void)addRenderer:(id<RTC_OBJC_TYPE(RTCVideoRenderer)>)renderer {
+  if (!_workerThread->IsCurrent()) {
+    _workerThread->BlockingCall([renderer, self] { [self addRenderer:renderer]; });
+    return;
+  }
+
+  // Make sure we don't have this renderer yet.
+  for (RTCVideoRendererAdapter *adapter in _adapters) {
+    if (adapter.videoRenderer == renderer) {
+      RTC_LOG(LS_INFO) << "|renderer| is already attached to this track";
+      return;
+    }
+  }
+  // Create a wrapper that provides a native pointer for us.
+  RTCVideoRendererAdapter* adapter =
+      [[RTCVideoRendererAdapter alloc] initWithNativeRenderer:renderer];
+  [_adapters addObject:adapter];
+  self.nativeVideoTrack->AddOrUpdateSink(adapter.nativeVideoRenderer,
+                                         rtc::VideoSinkWants());
+}
+
+- (void)removeRenderer:(id<RTC_OBJC_TYPE(RTCVideoRenderer)>)renderer {
+  if (!_workerThread->IsCurrent()) {
+    _workerThread->BlockingCall([renderer, self] { [self removeRenderer:renderer]; });
+    return;
+  }
+  __block NSUInteger indexToRemove = NSNotFound;
+  [_adapters enumerateObjectsUsingBlock:^(RTCVideoRendererAdapter *adapter,
+                                          NSUInteger idx,
+                                          BOOL *stop) {
+    if (adapter.videoRenderer == renderer) {
+      indexToRemove = idx;
+      *stop = YES;
+    }
+  }];
+  if (indexToRemove == NSNotFound) {
+    RTC_LOG(LS_INFO) << "removeRenderer called with a renderer that has not been previously added";
+    return;
+  }
+  RTCVideoRendererAdapter *adapterToRemove =
+      [_adapters objectAtIndex:indexToRemove];
+  self.nativeVideoTrack->RemoveSink(adapterToRemove.nativeVideoRenderer);
+  [_adapters removeObjectAtIndex:indexToRemove];
+}
+
+#pragma mark - Private
+
+- (rtc::scoped_refptr<webrtc::VideoTrackInterface>)nativeVideoTrack {
+  return rtc::scoped_refptr<webrtc::VideoTrackInterface>(
+      static_cast<webrtc::VideoTrackInterface *>(self.nativeTrack.get()));
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/api/video_codec/RTCVideoCodecConstants.h b/third_party/libwebrtc/sdk/objc/api/video_codec/RTCVideoCodecConstants.h
new file mode 100644
index 0000000000..8b17a75aef
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/video_codec/RTCVideoCodecConstants.h
@@ -0,0 +1,17 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+RTC_EXTERN NSString* const kRTCVideoCodecVp8Name;
+RTC_EXTERN NSString* const kRTCVideoCodecVp9Name;
+RTC_EXTERN NSString* const kRTCVideoCodecAv1Name;
diff --git a/third_party/libwebrtc/sdk/objc/api/video_codec/RTCVideoCodecConstants.mm b/third_party/libwebrtc/sdk/objc/api/video_codec/RTCVideoCodecConstants.mm
new file mode 100644
index 0000000000..1ab236a2c2
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/video_codec/RTCVideoCodecConstants.mm
@@ -0,0 +1,18 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ *
+ */
+
+#import "RTCVideoCodecConstants.h"
+
+#include "media/base/media_constants.h"
+
+NSString *const kRTCVideoCodecVp8Name = @(cricket::kVp8CodecName);
+NSString *const kRTCVideoCodecVp9Name = @(cricket::kVp9CodecName);
+NSString *const kRTCVideoCodecAv1Name = @(cricket::kAv1CodecName);
diff --git a/third_party/libwebrtc/sdk/objc/api/video_codec/RTCVideoDecoderAV1.h b/third_party/libwebrtc/sdk/objc/api/video_codec/RTCVideoDecoderAV1.h
new file mode 100644
index 0000000000..3f6a689564
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/video_codec/RTCVideoDecoderAV1.h
@@ -0,0 +1,25 @@
+/*
+ *  Copyright 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+#import "RTCVideoDecoder.h"
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCVideoDecoderAV1) : NSObject
+
+/* This returns a AV1 decoder that can be returned from a RTCVideoDecoderFactory injected into
+ * RTCPeerConnectionFactory. Even though it implements the RTCVideoDecoder protocol, it can not be
+ * used independently from the RTCPeerConnectionFactory.
+ */
++ (id<RTC_OBJC_TYPE(RTCVideoDecoder)>)av1Decoder;
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/api/video_codec/RTCVideoDecoderAV1.mm b/third_party/libwebrtc/sdk/objc/api/video_codec/RTCVideoDecoderAV1.mm
new file mode 100644
index 0000000000..81f5f93eec
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/video_codec/RTCVideoDecoderAV1.mm
@@ -0,0 +1,27 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ *
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+#import "RTCVideoDecoderAV1.h"
+#import "RTCWrappedNativeVideoDecoder.h"
+
+#include "modules/video_coding/codecs/av1/dav1d_decoder.h"
+
+@implementation RTC_OBJC_TYPE (RTCVideoDecoderAV1)
+
++ (id<RTC_OBJC_TYPE(RTCVideoDecoder)>)av1Decoder {
+  return [[RTC_OBJC_TYPE(RTCWrappedNativeVideoDecoder) alloc]
+      initWithNativeDecoder:std::unique_ptr<webrtc::VideoDecoder>(webrtc::CreateDav1dDecoder())];
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/api/video_codec/RTCVideoDecoderVP8.h b/third_party/libwebrtc/sdk/objc/api/video_codec/RTCVideoDecoderVP8.h
new file mode 100644
index 0000000000..a118b25ed7
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/video_codec/RTCVideoDecoderVP8.h
@@ -0,0 +1,25 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+#import "RTCVideoDecoder.h"
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCVideoDecoderVP8) : NSObject
+
+/* This returns a VP8 decoder that can be returned from a RTCVideoDecoderFactory injected into
+ * RTCPeerConnectionFactory. Even though it implements the RTCVideoDecoder protocol, it can not be
+ * used independently from the RTCPeerConnectionFactory.
+ */
++ (id<RTC_OBJC_TYPE(RTCVideoDecoder)>)vp8Decoder;
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/api/video_codec/RTCVideoDecoderVP8.mm b/third_party/libwebrtc/sdk/objc/api/video_codec/RTCVideoDecoderVP8.mm
new file mode 100644
index 0000000000..c150cf6d3a
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/video_codec/RTCVideoDecoderVP8.mm
@@ -0,0 +1,27 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ *
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+#import "RTCVideoDecoderVP8.h"
+#import "RTCWrappedNativeVideoDecoder.h"
+
+#include "modules/video_coding/codecs/vp8/include/vp8.h"
+
+@implementation RTC_OBJC_TYPE (RTCVideoDecoderVP8)
+
++ (id<RTC_OBJC_TYPE(RTCVideoDecoder)>)vp8Decoder {
+  return [[RTC_OBJC_TYPE(RTCWrappedNativeVideoDecoder) alloc]
+      initWithNativeDecoder:std::unique_ptr<webrtc::VideoDecoder>(webrtc::VP8Decoder::Create())];
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/api/video_codec/RTCVideoDecoderVP9.h b/third_party/libwebrtc/sdk/objc/api/video_codec/RTCVideoDecoderVP9.h
new file mode 100644
index 0000000000..de7e62012b
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/video_codec/RTCVideoDecoderVP9.h
@@ -0,0 +1,27 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+#import "RTCVideoDecoder.h"
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCVideoDecoderVP9) : NSObject
+
+/* This returns a VP9 decoder that can be returned from a RTCVideoDecoderFactory injected into
+ * RTCPeerConnectionFactory. Even though it implements the RTCVideoDecoder protocol, it can not be
+ * used independently from the RTCPeerConnectionFactory.
+ */
++ (id<RTC_OBJC_TYPE(RTCVideoDecoder)>)vp9Decoder;
+
++ (bool)isSupported;
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/api/video_codec/RTCVideoDecoderVP9.mm b/third_party/libwebrtc/sdk/objc/api/video_codec/RTCVideoDecoderVP9.mm
new file mode 100644
index 0000000000..05446d436d
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/video_codec/RTCVideoDecoderVP9.mm
@@ -0,0 +1,39 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ *
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+#import "RTCVideoDecoderVP9.h"
+#import "RTCWrappedNativeVideoDecoder.h"
+
+#include "modules/video_coding/codecs/vp9/include/vp9.h"
+
+@implementation RTC_OBJC_TYPE (RTCVideoDecoderVP9)
+
++ (id<RTC_OBJC_TYPE(RTCVideoDecoder)>)vp9Decoder {
+  std::unique_ptr<webrtc::VideoDecoder> nativeDecoder(webrtc::VP9Decoder::Create());
+  if (nativeDecoder == nullptr) {
+    return nil;
+  }
+  return [[RTC_OBJC_TYPE(RTCWrappedNativeVideoDecoder) alloc]
+      initWithNativeDecoder:std::move(nativeDecoder)];
+}
+
++ (bool)isSupported {
+#if defined(RTC_ENABLE_VP9)
+  return true;
+#else
+  return false;
+#endif
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/api/video_codec/RTCVideoEncoderAV1.h b/third_party/libwebrtc/sdk/objc/api/video_codec/RTCVideoEncoderAV1.h
new file mode 100644
index 0000000000..8aa55e4bfa
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/video_codec/RTCVideoEncoderAV1.h
@@ -0,0 +1,27 @@
+/*
+ *  Copyright 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+#import "RTCVideoEncoder.h"
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCVideoEncoderAV1) : NSObject
+
+/* This returns a AV1 encoder that can be returned from a RTCVideoEncoderFactory injected into
+ * RTCPeerConnectionFactory. Even though it implements the RTCVideoEncoder protocol, it can not be
+ * used independently from the RTCPeerConnectionFactory.
+ */
++ (id<RTC_OBJC_TYPE(RTCVideoEncoder)>)av1Encoder;
+
++ (bool)isSupported;
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/api/video_codec/RTCVideoEncoderAV1.mm b/third_party/libwebrtc/sdk/objc/api/video_codec/RTCVideoEncoderAV1.mm
new file mode 100644
index 0000000000..d2fe65293b
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/video_codec/RTCVideoEncoderAV1.mm
@@ -0,0 +1,31 @@
+/*
+ *  Copyright (c) 2021 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ *
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+#import "RTCVideoEncoderAV1.h"
+#import "RTCWrappedNativeVideoEncoder.h"
+#include "modules/video_coding/codecs/av1/libaom_av1_encoder.h"
+
+@implementation RTC_OBJC_TYPE (RTCVideoEncoderAV1)
+
++ (id<RTC_OBJC_TYPE(RTCVideoEncoder)>)av1Encoder {
+  std::unique_ptr<webrtc::VideoEncoder> nativeEncoder(webrtc::CreateLibaomAv1Encoder());
+  return [[RTC_OBJC_TYPE(RTCWrappedNativeVideoEncoder) alloc]
+      initWithNativeEncoder:std::move(nativeEncoder)];
+}
+
++ (bool)isSupported {
+  return true;
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/api/video_codec/RTCVideoEncoderVP8.h b/third_party/libwebrtc/sdk/objc/api/video_codec/RTCVideoEncoderVP8.h
new file mode 100644
index 0000000000..e136a5bda8
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/video_codec/RTCVideoEncoderVP8.h
@@ -0,0 +1,25 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+#import "RTCVideoEncoder.h"
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCVideoEncoderVP8) : NSObject
+
+/* This returns a VP8 encoder that can be returned from a RTCVideoEncoderFactory injected into
+ * RTCPeerConnectionFactory. Even though it implements the RTCVideoEncoder protocol, it can not be
+ * used independently from the RTCPeerConnectionFactory.
+ */
++ (id<RTC_OBJC_TYPE(RTCVideoEncoder)>)vp8Encoder;
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/api/video_codec/RTCVideoEncoderVP8.mm b/third_party/libwebrtc/sdk/objc/api/video_codec/RTCVideoEncoderVP8.mm
new file mode 100644
index 0000000000..d72f705813
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/video_codec/RTCVideoEncoderVP8.mm
@@ -0,0 +1,27 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ *
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+#import "RTCVideoEncoderVP8.h"
+#import "RTCWrappedNativeVideoEncoder.h"
+
+#include "modules/video_coding/codecs/vp8/include/vp8.h"
+
+@implementation RTC_OBJC_TYPE (RTCVideoEncoderVP8)
+
++ (id<RTC_OBJC_TYPE(RTCVideoEncoder)>)vp8Encoder {
+  return [[RTC_OBJC_TYPE(RTCWrappedNativeVideoEncoder) alloc]
+      initWithNativeEncoder:std::unique_ptr<webrtc::VideoEncoder>(webrtc::VP8Encoder::Create())];
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/api/video_codec/RTCVideoEncoderVP9.h b/third_party/libwebrtc/sdk/objc/api/video_codec/RTCVideoEncoderVP9.h
new file mode 100644
index 0000000000..f7dac6117d
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/video_codec/RTCVideoEncoderVP9.h
@@ -0,0 +1,27 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+#import "RTCVideoEncoder.h"
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCVideoEncoderVP9) : NSObject
+
+/* This returns a VP9 encoder that can be returned from a RTCVideoEncoderFactory injected into
+ * RTCPeerConnectionFactory. Even though it implements the RTCVideoEncoder protocol, it can not be
+ * used independently from the RTCPeerConnectionFactory.
+ */
++ (id<RTC_OBJC_TYPE(RTCVideoEncoder)>)vp9Encoder;
+
++ (bool)isSupported;
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/api/video_codec/RTCVideoEncoderVP9.mm b/third_party/libwebrtc/sdk/objc/api/video_codec/RTCVideoEncoderVP9.mm
new file mode 100644
index 0000000000..18a9353f7e
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/video_codec/RTCVideoEncoderVP9.mm
@@ -0,0 +1,39 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ *
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+#import "RTCVideoEncoderVP9.h"
+#import "RTCWrappedNativeVideoEncoder.h"
+
+#include "modules/video_coding/codecs/vp9/include/vp9.h"
+
+@implementation RTC_OBJC_TYPE (RTCVideoEncoderVP9)
+
++ (id<RTC_OBJC_TYPE(RTCVideoEncoder)>)vp9Encoder {
+  std::unique_ptr<webrtc::VideoEncoder> nativeEncoder(webrtc::VP9Encoder::Create());
+  if (nativeEncoder == nullptr) {
+    return nil;
+  }
+  return [[RTC_OBJC_TYPE(RTCWrappedNativeVideoEncoder) alloc]
+      initWithNativeEncoder:std::move(nativeEncoder)];
+}
+
++ (bool)isSupported {
+#if defined(RTC_ENABLE_VP9)
+  return true;
+#else
+  return false;
+#endif
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/api/video_codec/RTCWrappedNativeVideoDecoder.h b/third_party/libwebrtc/sdk/objc/api/video_codec/RTCWrappedNativeVideoDecoder.h
new file mode 100644
index 0000000000..3a9b39e959
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/video_codec/RTCWrappedNativeVideoDecoder.h
@@ -0,0 +1,26 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "base/RTCMacros.h"
+#import "base/RTCVideoDecoder.h"
+
+#include "api/video_codecs/video_decoder.h"
+#include "media/base/codec.h"
+
+@interface RTC_OBJC_TYPE (RTCWrappedNativeVideoDecoder) : NSObject <RTC_OBJC_TYPE (RTCVideoDecoder)>
+
+- (instancetype)initWithNativeDecoder:(std::unique_ptr<webrtc::VideoDecoder>)decoder;
+
+/* This moves the ownership of the wrapped decoder to the caller. */
+- (std::unique_ptr<webrtc::VideoDecoder>)releaseWrappedDecoder;
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/api/video_codec/RTCWrappedNativeVideoDecoder.mm b/third_party/libwebrtc/sdk/objc/api/video_codec/RTCWrappedNativeVideoDecoder.mm
new file mode 100644
index 0000000000..261874d20b
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/video_codec/RTCWrappedNativeVideoDecoder.mm
@@ -0,0 +1,62 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCWrappedNativeVideoDecoder.h"
+#import "base/RTCMacros.h"
+#import "helpers/NSString+StdString.h"
+
+@implementation RTC_OBJC_TYPE (RTCWrappedNativeVideoDecoder) {
+  std::unique_ptr<webrtc::VideoDecoder> _wrappedDecoder;
+}
+
+- (instancetype)initWithNativeDecoder:(std::unique_ptr<webrtc::VideoDecoder>)decoder {
+  if (self = [super init]) {
+    _wrappedDecoder = std::move(decoder);
+  }
+
+  return self;
+}
+
+- (std::unique_ptr<webrtc::VideoDecoder>)releaseWrappedDecoder {
+  return std::move(_wrappedDecoder);
+}
+
+#pragma mark - RTC_OBJC_TYPE(RTCVideoDecoder)
+
+- (void)setCallback:(RTCVideoDecoderCallback)callback {
+  RTC_DCHECK_NOTREACHED();
+}
+
+- (NSInteger)startDecodeWithNumberOfCores:(int)numberOfCores {
+  RTC_DCHECK_NOTREACHED();
+  return 0;
+}
+
+- (NSInteger)releaseDecoder {
+  RTC_DCHECK_NOTREACHED();
+  return 0;
+}
+
+- (NSInteger)decode:(RTC_OBJC_TYPE(RTCEncodedImage) *)encodedImage
+        missingFrames:(BOOL)missingFrames
+    codecSpecificInfo:(nullable id<RTC_OBJC_TYPE(RTCCodecSpecificInfo)>)info
+         renderTimeMs:(int64_t)renderTimeMs {
+  RTC_DCHECK_NOTREACHED();
+  return 0;
+}
+
+- (NSString *)implementationName {
+  RTC_DCHECK_NOTREACHED();
+  return nil;
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/api/video_codec/RTCWrappedNativeVideoEncoder.h b/third_party/libwebrtc/sdk/objc/api/video_codec/RTCWrappedNativeVideoEncoder.h
new file mode 100644
index 0000000000..8df9ceec35
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/video_codec/RTCWrappedNativeVideoEncoder.h
@@ -0,0 +1,27 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "base/RTCMacros.h"
+#import "base/RTCVideoEncoder.h"
+
+#include "api/video_codecs/sdp_video_format.h"
+#include "api/video_codecs/video_encoder.h"
+#include "media/base/codec.h"
+
+@interface RTC_OBJC_TYPE (RTCWrappedNativeVideoEncoder) : NSObject <RTC_OBJC_TYPE (RTCVideoEncoder)>
+
+- (instancetype)initWithNativeEncoder:(std::unique_ptr<webrtc::VideoEncoder>)encoder;
+
+/* This moves the ownership of the wrapped encoder to the caller. */
+- (std::unique_ptr<webrtc::VideoEncoder>)releaseWrappedEncoder;
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/api/video_codec/RTCWrappedNativeVideoEncoder.mm b/third_party/libwebrtc/sdk/objc/api/video_codec/RTCWrappedNativeVideoEncoder.mm
new file mode 100644
index 0000000000..4160572814
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/video_codec/RTCWrappedNativeVideoEncoder.mm
@@ -0,0 +1,86 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCWrappedNativeVideoEncoder.h"
+#import "base/RTCMacros.h"
+#import "helpers/NSString+StdString.h"
+
+@implementation RTC_OBJC_TYPE (RTCWrappedNativeVideoEncoder) {
+  std::unique_ptr<webrtc::VideoEncoder> _wrappedEncoder;
+}
+
+- (instancetype)initWithNativeEncoder:(std::unique_ptr<webrtc::VideoEncoder>)encoder {
+  if (self = [super init]) {
+    _wrappedEncoder = std::move(encoder);
+  }
+
+  return self;
+}
+
+- (std::unique_ptr<webrtc::VideoEncoder>)releaseWrappedEncoder {
+  return std::move(_wrappedEncoder);
+}
+
+#pragma mark - RTC_OBJC_TYPE(RTCVideoEncoder)
+
+- (void)setCallback:(RTCVideoEncoderCallback)callback {
+  RTC_DCHECK_NOTREACHED();
+}
+
+- (NSInteger)startEncodeWithSettings:(RTC_OBJC_TYPE(RTCVideoEncoderSettings) *)settings
+                       numberOfCores:(int)numberOfCores {
+  RTC_DCHECK_NOTREACHED();
+  return 0;
+}
+
+- (NSInteger)releaseEncoder {
+  RTC_DCHECK_NOTREACHED();
+  return 0;
+}
+
+- (NSInteger)encode:(RTC_OBJC_TYPE(RTCVideoFrame) *)frame
+    codecSpecificInfo:(nullable id<RTC_OBJC_TYPE(RTCCodecSpecificInfo)>)info
+           frameTypes:(NSArray<NSNumber *> *)frameTypes {
+  RTC_DCHECK_NOTREACHED();
+  return 0;
+}
+
+- (int)setBitrate:(uint32_t)bitrateKbit framerate:(uint32_t)framerate {
+  RTC_DCHECK_NOTREACHED();
+  return 0;
+}
+
+- (NSString *)implementationName {
+  RTC_DCHECK_NOTREACHED();
+  return nil;
+}
+
+- (nullable RTC_OBJC_TYPE(RTCVideoEncoderQpThresholds) *)scalingSettings {
+  RTC_DCHECK_NOTREACHED();
+  return nil;
+}
+
+- (NSInteger)resolutionAlignment {
+  RTC_DCHECK_NOTREACHED();
+  return 1;
+}
+
+- (BOOL)applyAlignmentToAllSimulcastLayers {
+  RTC_DCHECK_NOTREACHED();
+  return NO;
+}
+
+- (BOOL)supportsNativeHandle {
+  RTC_DCHECK_NOTREACHED();
+  return NO;
+}
+@end
diff --git a/third_party/libwebrtc/sdk/objc/api/video_frame_buffer/RTCNativeI420Buffer+Private.h b/third_party/libwebrtc/sdk/objc/api/video_frame_buffer/RTCNativeI420Buffer+Private.h
new file mode 100644
index 0000000000..20dc807991
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/video_frame_buffer/RTCNativeI420Buffer+Private.h
@@ -0,0 +1,29 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCNativeI420Buffer.h"
+
+#include "api/video/i420_buffer.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+@interface RTC_OBJC_TYPE (RTCI420Buffer)
+() {
+ @protected
+  rtc::scoped_refptr<webrtc::I420BufferInterface> _i420Buffer;
+}
+
+/** Initialize an RTCI420Buffer with its backing I420BufferInterface. */
+- (instancetype)initWithFrameBuffer:(rtc::scoped_refptr<webrtc::I420BufferInterface>)i420Buffer;
+- (rtc::scoped_refptr<webrtc::I420BufferInterface>)nativeI420Buffer;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/api/video_frame_buffer/RTCNativeI420Buffer.h b/third_party/libwebrtc/sdk/objc/api/video_frame_buffer/RTCNativeI420Buffer.h
new file mode 100644
index 0000000000..3afe2090a2
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/video_frame_buffer/RTCNativeI420Buffer.h
@@ -0,0 +1,23 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <AVFoundation/AVFoundation.h>
+
+#import "RTCI420Buffer.h"
+#import "RTCMacros.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+/** RTCI420Buffer implements the RTCI420Buffer protocol */
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCI420Buffer) : NSObject<RTC_OBJC_TYPE(RTCI420Buffer)>
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/api/video_frame_buffer/RTCNativeI420Buffer.mm b/third_party/libwebrtc/sdk/objc/api/video_frame_buffer/RTCNativeI420Buffer.mm
new file mode 100644
index 0000000000..7aafd98f43
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/video_frame_buffer/RTCNativeI420Buffer.mm
@@ -0,0 +1,154 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCNativeI420Buffer+Private.h"
+
+#include "api/video/i420_buffer.h"
+
+#if !defined(NDEBUG) && defined(WEBRTC_IOS)
+#import <UIKit/UIKit.h>
+#include "third_party/libyuv/include/libyuv.h"
+#endif
+
+@implementation RTC_OBJC_TYPE (RTCI420Buffer)
+
+- (instancetype)initWithWidth:(int)width height:(int)height {
+  if (self = [super init]) {
+    _i420Buffer = webrtc::I420Buffer::Create(width, height);
+  }
+
+  return self;
+}
+
+- (instancetype)initWithWidth:(int)width
+                       height:(int)height
+                        dataY:(const uint8_t *)dataY
+                        dataU:(const uint8_t *)dataU
+                        dataV:(const uint8_t *)dataV {
+  if (self = [super init]) {
+    _i420Buffer = webrtc::I420Buffer::Copy(
+        width, height, dataY, width, dataU, (width + 1) / 2, dataV, (width + 1) / 2);
+  }
+  return self;
+}
+
+- (instancetype)initWithWidth:(int)width
+                       height:(int)height
+                      strideY:(int)strideY
+                      strideU:(int)strideU
+                      strideV:(int)strideV {
+  if (self = [super init]) {
+    _i420Buffer = webrtc::I420Buffer::Create(width, height, strideY, strideU, strideV);
+  }
+
+  return self;
+}
+
+- (instancetype)initWithFrameBuffer:(rtc::scoped_refptr<webrtc::I420BufferInterface>)i420Buffer {
+  if (self = [super init]) {
+    _i420Buffer = i420Buffer;
+  }
+
+  return self;
+}
+
+- (int)width {
+  return _i420Buffer->width();
+}
+
+- (int)height {
+  return _i420Buffer->height();
+}
+
+- (int)strideY {
+  return _i420Buffer->StrideY();
+}
+
+- (int)strideU {
+  return _i420Buffer->StrideU();
+}
+
+- (int)strideV {
+  return _i420Buffer->StrideV();
+}
+
+- (int)chromaWidth {
+  return _i420Buffer->ChromaWidth();
+}
+
+- (int)chromaHeight {
+  return _i420Buffer->ChromaHeight();
+}
+
+- (const uint8_t *)dataY {
+  return _i420Buffer->DataY();
+}
+
+- (const uint8_t *)dataU {
+  return _i420Buffer->DataU();
+}
+
+- (const uint8_t *)dataV {
+  return _i420Buffer->DataV();
+}
+
+- (id<RTC_OBJC_TYPE(RTCVideoFrameBuffer)>)cropAndScaleWith:(int)offsetX
+                                                   offsetY:(int)offsetY
+                                                 cropWidth:(int)cropWidth
+                                                cropHeight:(int)cropHeight
+                                                scaleWidth:(int)scaleWidth
+                                               scaleHeight:(int)scaleHeight {
+  rtc::scoped_refptr<webrtc::VideoFrameBuffer> scaled_buffer =
+      _i420Buffer->CropAndScale(offsetX, offsetY, cropWidth, cropHeight, scaleWidth, scaleHeight);
+  RTC_DCHECK_EQ(scaled_buffer->type(), webrtc::VideoFrameBuffer::Type::kI420);
+  // Calling ToI420() doesn't do any conversions.
+  rtc::scoped_refptr<webrtc::I420BufferInterface> buffer = scaled_buffer->ToI420();
+  RTC_OBJC_TYPE(RTCI420Buffer) *result =
+      [[RTC_OBJC_TYPE(RTCI420Buffer) alloc] initWithFrameBuffer:buffer];
+  return result;
+}
+
+- (id<RTC_OBJC_TYPE(RTCI420Buffer)>)toI420 {
+  return self;
+}
+
+#pragma mark - Private
+
+- (rtc::scoped_refptr<webrtc::I420BufferInterface>)nativeI420Buffer {
+  return _i420Buffer;
+}
+
+#pragma mark - Debugging
+
+#if !defined(NDEBUG) && defined(WEBRTC_IOS)
+- (id)debugQuickLookObject {
+  UIGraphicsBeginImageContext(CGSizeMake(_i420Buffer->width(), _i420Buffer->height()));
+  CGContextRef c = UIGraphicsGetCurrentContext();
+  uint8_t *ctxData = (uint8_t *)CGBitmapContextGetData(c);
+
+  libyuv::I420ToARGB(_i420Buffer->DataY(),
+                     _i420Buffer->StrideY(),
+                     _i420Buffer->DataU(),
+                     _i420Buffer->StrideU(),
+                     _i420Buffer->DataV(),
+                     _i420Buffer->StrideV(),
+                     ctxData,
+                     CGBitmapContextGetBytesPerRow(c),
+                     CGBitmapContextGetWidth(c),
+                     CGBitmapContextGetHeight(c));
+
+  UIImage *image = UIGraphicsGetImageFromCurrentImageContext();
+  UIGraphicsEndImageContext();
+
+  return image;
+}
+#endif
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/api/video_frame_buffer/RTCNativeMutableI420Buffer.h b/third_party/libwebrtc/sdk/objc/api/video_frame_buffer/RTCNativeMutableI420Buffer.h
new file mode 100644
index 0000000000..053a10a304
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/video_frame_buffer/RTCNativeMutableI420Buffer.h
@@ -0,0 +1,24 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <AVFoundation/AVFoundation.h>
+
+#import "RTCMacros.h"
+#import "RTCMutableI420Buffer.h"
+#import "RTCNativeI420Buffer.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+/** Mutable version of RTCI420Buffer */
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCMutableI420Buffer) : RTC_OBJC_TYPE(RTCI420Buffer)<RTC_OBJC_TYPE(RTCMutableI420Buffer)>
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/api/video_frame_buffer/RTCNativeMutableI420Buffer.mm b/third_party/libwebrtc/sdk/objc/api/video_frame_buffer/RTCNativeMutableI420Buffer.mm
new file mode 100644
index 0000000000..1e669bcb9c
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/api/video_frame_buffer/RTCNativeMutableI420Buffer.mm
@@ -0,0 +1,31 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCNativeMutableI420Buffer.h"
+
+#import "RTCNativeI420Buffer+Private.h"
+
+#include "api/video/i420_buffer.h"
+
+@implementation RTC_OBJC_TYPE (RTCMutableI420Buffer)
+
+- (uint8_t *)mutableDataY {
+  return static_cast<webrtc::I420Buffer *>(_i420Buffer.get())->MutableDataY();
+}
+
+- (uint8_t *)mutableDataU {
+  return static_cast<webrtc::I420Buffer *>(_i420Buffer.get())->MutableDataU();
+}
+
+- (uint8_t *)mutableDataV {
+  return static_cast<webrtc::I420Buffer *>(_i420Buffer.get())->MutableDataV();
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/base/RTCCodecSpecificInfo.h b/third_party/libwebrtc/sdk/objc/base/RTCCodecSpecificInfo.h
new file mode 100644
index 0000000000..5e7800e524
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/base/RTCCodecSpecificInfo.h
@@ -0,0 +1,24 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+/** Implement this protocol to pass codec specific info from the encoder.
+ *  Corresponds to webrtc::CodecSpecificInfo.
+ */
+RTC_OBJC_EXPORT
+@protocol RTC_OBJC_TYPE
+(RTCCodecSpecificInfo)<NSObject> @end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/base/RTCEncodedImage.h b/third_party/libwebrtc/sdk/objc/base/RTCEncodedImage.h
new file mode 100644
index 0000000000..28529e5906
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/base/RTCEncodedImage.h
@@ -0,0 +1,52 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+#import "RTCVideoFrame.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+/** Represents an encoded frame's type. */
+typedef NS_ENUM(NSUInteger, RTCFrameType) {
+  RTCFrameTypeEmptyFrame = 0,
+  RTCFrameTypeAudioFrameSpeech = 1,
+  RTCFrameTypeAudioFrameCN = 2,
+  RTCFrameTypeVideoFrameKey = 3,
+  RTCFrameTypeVideoFrameDelta = 4,
+};
+
+typedef NS_ENUM(NSUInteger, RTCVideoContentType) {
+  RTCVideoContentTypeUnspecified,
+  RTCVideoContentTypeScreenshare,
+};
+
+/** Represents an encoded frame. Corresponds to webrtc::EncodedImage. */
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCEncodedImage) : NSObject
+
+@property(nonatomic, strong) NSData *buffer;
+@property(nonatomic, assign) int32_t encodedWidth;
+@property(nonatomic, assign) int32_t encodedHeight;
+@property(nonatomic, assign) uint32_t timeStamp;
+@property(nonatomic, assign) int64_t captureTimeMs;
+@property(nonatomic, assign) int64_t ntpTimeMs;
+@property(nonatomic, assign) uint8_t flags;
+@property(nonatomic, assign) int64_t encodeStartMs;
+@property(nonatomic, assign) int64_t encodeFinishMs;
+@property(nonatomic, assign) RTCFrameType frameType;
+@property(nonatomic, assign) RTCVideoRotation rotation;
+@property(nonatomic, strong) NSNumber *qp;
+@property(nonatomic, assign) RTCVideoContentType contentType;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/base/RTCEncodedImage.m b/third_party/libwebrtc/sdk/objc/base/RTCEncodedImage.m
new file mode 100644
index 0000000000..ad8441aabd
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/base/RTCEncodedImage.m
@@ -0,0 +1,29 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCEncodedImage.h"
+
+@implementation RTC_OBJC_TYPE (RTCEncodedImage)
+
+@synthesize buffer = _buffer;
+@synthesize encodedWidth = _encodedWidth;
+@synthesize encodedHeight = _encodedHeight;
+@synthesize timeStamp = _timeStamp;
+@synthesize captureTimeMs = _captureTimeMs;
+@synthesize ntpTimeMs = _ntpTimeMs;
+@synthesize flags = _flags;
+@synthesize encodeStartMs = _encodeStartMs;
+@synthesize encodeFinishMs = _encodeFinishMs;
+@synthesize frameType = _frameType;
+@synthesize rotation = _rotation;
+@synthesize qp = _qp;
+@synthesize contentType = _contentType;
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/base/RTCI420Buffer.h b/third_party/libwebrtc/sdk/objc/base/RTCI420Buffer.h
new file mode 100644
index 0000000000..b97f05a5ba
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/base/RTCI420Buffer.h
@@ -0,0 +1,22 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <AVFoundation/AVFoundation.h>
+
+#import "RTCYUVPlanarBuffer.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+/** Protocol for RTCYUVPlanarBuffers containing I420 data */
+RTC_OBJC_EXPORT
+@protocol RTC_OBJC_TYPE
+(RTCI420Buffer)<RTC_OBJC_TYPE(RTCYUVPlanarBuffer)> @end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/base/RTCLogging.h b/third_party/libwebrtc/sdk/objc/base/RTCLogging.h
new file mode 100644
index 0000000000..754945c8f2
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/base/RTCLogging.h
@@ -0,0 +1,67 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+// Subset of rtc::LoggingSeverity.
+typedef NS_ENUM(NSInteger, RTCLoggingSeverity) {
+  RTCLoggingSeverityVerbose,
+  RTCLoggingSeverityInfo,
+  RTCLoggingSeverityWarning,
+  RTCLoggingSeverityError,
+  RTCLoggingSeverityNone,
+};
+
+// Wrapper for C++ RTC_LOG(sev) macros.
+// Logs the log string to the webrtc logstream for the given severity.
+RTC_EXTERN void RTCLogEx(RTCLoggingSeverity severity, NSString* log_string);
+
+// Wrapper for rtc::LogMessage::LogToDebug.
+// Sets the minimum severity to be logged to console.
+RTC_EXTERN void RTCSetMinDebugLogLevel(RTCLoggingSeverity severity);
+
+// Returns the filename with the path prefix removed.
+RTC_EXTERN NSString* RTCFileName(const char* filePath);
+
+// Some convenience macros.
+
+#define RTCLogString(format, ...)                                           \
+  [NSString stringWithFormat:@"(%@:%d %s): " format, RTCFileName(__FILE__), \
+                             __LINE__, __FUNCTION__, ##__VA_ARGS__]
+
+#define RTCLogFormat(severity, format, ...)                     \
+  do {                                                          \
+    NSString* log_string = RTCLogString(format, ##__VA_ARGS__); \
+    RTCLogEx(severity, log_string);                             \
+  } while (false)
+
+#define RTCLogVerbose(format, ...) \
+  RTCLogFormat(RTCLoggingSeverityVerbose, format, ##__VA_ARGS__)
+
+#define RTCLogInfo(format, ...) \
+  RTCLogFormat(RTCLoggingSeverityInfo, format, ##__VA_ARGS__)
+
+#define RTCLogWarning(format, ...) \
+  RTCLogFormat(RTCLoggingSeverityWarning, format, ##__VA_ARGS__)
+
+#define RTCLogError(format, ...) \
+  RTCLogFormat(RTCLoggingSeverityError, format, ##__VA_ARGS__)
+
+#if !defined(NDEBUG)
+#define RTCLogDebug(format, ...) RTCLogInfo(format, ##__VA_ARGS__)
+#else
+#define RTCLogDebug(format, ...) \
+  do {                           \
+  } while (false)
+#endif
+
+#define RTCLog(format, ...) RTCLogInfo(format, ##__VA_ARGS__)
diff --git a/third_party/libwebrtc/sdk/objc/base/RTCLogging.mm b/third_party/libwebrtc/sdk/objc/base/RTCLogging.mm
new file mode 100644
index 0000000000..e8dae02efb
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/base/RTCLogging.mm
@@ -0,0 +1,48 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCLogging.h"
+
+#include "rtc_base/logging.h"
+
+rtc::LoggingSeverity RTCGetNativeLoggingSeverity(RTCLoggingSeverity severity) {
+  switch (severity) {
+    case RTCLoggingSeverityVerbose:
+      return rtc::LS_VERBOSE;
+    case RTCLoggingSeverityInfo:
+      return rtc::LS_INFO;
+    case RTCLoggingSeverityWarning:
+      return rtc::LS_WARNING;
+    case RTCLoggingSeverityError:
+      return rtc::LS_ERROR;
+    case RTCLoggingSeverityNone:
+      return rtc::LS_NONE;
+  }
+}
+
+void RTCLogEx(RTCLoggingSeverity severity, NSString* log_string) {
+  if (log_string.length) {
+    const char* utf8_string = log_string.UTF8String;
+    RTC_LOG_V(RTCGetNativeLoggingSeverity(severity)) << utf8_string;
+  }
+}
+
+void RTCSetMinDebugLogLevel(RTCLoggingSeverity severity) {
+  rtc::LogMessage::LogToDebug(RTCGetNativeLoggingSeverity(severity));
+}
+
+NSString* RTCFileName(const char* file_path) {
+  NSString* ns_file_path =
+      [[NSString alloc] initWithBytesNoCopy:const_cast<char*>(file_path)
+                                     length:strlen(file_path)
+                                   encoding:NSUTF8StringEncoding
+                               freeWhenDone:NO];
+  return ns_file_path.lastPathComponent;
+}
diff --git a/third_party/libwebrtc/sdk/objc/base/RTCMacros.h b/third_party/libwebrtc/sdk/objc/base/RTCMacros.h
new file mode 100644
index 0000000000..469e3c93bd
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/base/RTCMacros.h
@@ -0,0 +1,62 @@
+/*
+ *  Copyright 2016 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef SDK_OBJC_BASE_RTCMACROS_H_
+#define SDK_OBJC_BASE_RTCMACROS_H_
+
+#ifdef WEBRTC_ENABLE_OBJC_SYMBOL_EXPORT
+
+#if defined(WEBRTC_LIBRARY_IMPL)
+#define RTC_OBJC_EXPORT __attribute__((visibility("default")))
+#endif
+
+#endif  // WEBRTC_ENABLE_OBJC_SYMBOL_EXPORT
+
+#ifndef RTC_OBJC_EXPORT
+#define RTC_OBJC_EXPORT
+#endif
+
+// Internal macros used to correctly concatenate symbols.
+#define RTC_SYMBOL_CONCAT_HELPER(a, b) a##b
+#define RTC_SYMBOL_CONCAT(a, b) RTC_SYMBOL_CONCAT_HELPER(a, b)
+
+// RTC_OBJC_TYPE_PREFIX
+//
+// Macro used to prepend a prefix to the API types that are exported with
+// RTC_OBJC_EXPORT.
+//
+// Clients can patch the definition of this macro locally and build
+// WebRTC.framework with their own prefix in case symbol clashing is a
+// problem.
+//
+// This macro must only be defined here and not on via compiler flag to
+// ensure it has a unique value.
+#define RTC_OBJC_TYPE_PREFIX
+
+// RCT_OBJC_TYPE
+//
+// Macro used internally to declare API types. Declaring an API type without
+// using this macro will not include the declared type in the set of types
+// that will be affected by the configurable RTC_OBJC_TYPE_PREFIX.
+#define RTC_OBJC_TYPE(type_name) RTC_SYMBOL_CONCAT(RTC_OBJC_TYPE_PREFIX, type_name)
+
+#if defined(__cplusplus)
+#define RTC_EXTERN extern "C" RTC_OBJC_EXPORT
+#else
+#define RTC_EXTERN extern RTC_OBJC_EXPORT
+#endif
+
+#ifdef __OBJC__
+#define RTC_FWD_DECL_OBJC_CLASS(classname) @class classname
+#else
+#define RTC_FWD_DECL_OBJC_CLASS(classname) typedef struct objc_object classname
+#endif
+
+#endif  // SDK_OBJC_BASE_RTCMACROS_H_
diff --git a/third_party/libwebrtc/sdk/objc/base/RTCMutableI420Buffer.h b/third_party/libwebrtc/sdk/objc/base/RTCMutableI420Buffer.h
new file mode 100644
index 0000000000..cde721980b
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/base/RTCMutableI420Buffer.h
@@ -0,0 +1,23 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <AVFoundation/AVFoundation.h>
+
+#import "RTCI420Buffer.h"
+#import "RTCMutableYUVPlanarBuffer.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+/** Extension of the I420 buffer with mutable data access */
+RTC_OBJC_EXPORT
+@protocol RTC_OBJC_TYPE
+(RTCMutableI420Buffer)<RTC_OBJC_TYPE(RTCI420Buffer), RTC_OBJC_TYPE(RTCMutableYUVPlanarBuffer)> @end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/base/RTCMutableYUVPlanarBuffer.h b/third_party/libwebrtc/sdk/objc/base/RTCMutableYUVPlanarBuffer.h
new file mode 100644
index 0000000000..bd14e3bca3
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/base/RTCMutableYUVPlanarBuffer.h
@@ -0,0 +1,28 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <AVFoundation/AVFoundation.h>
+
+#import "RTCYUVPlanarBuffer.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+/** Extension of the YUV planar data buffer with mutable data access */
+RTC_OBJC_EXPORT
+@protocol RTC_OBJC_TYPE
+(RTCMutableYUVPlanarBuffer)<RTC_OBJC_TYPE(RTCYUVPlanarBuffer)>
+
+    @property(nonatomic, readonly) uint8_t *mutableDataY;
+@property(nonatomic, readonly) uint8_t *mutableDataU;
+@property(nonatomic, readonly) uint8_t *mutableDataV;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/base/RTCSSLCertificateVerifier.h b/third_party/libwebrtc/sdk/objc/base/RTCSSLCertificateVerifier.h
new file mode 100644
index 0000000000..53da0cceff
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/base/RTCSSLCertificateVerifier.h
@@ -0,0 +1,25 @@
+/*
+ *  Copyright 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+RTC_OBJC_EXPORT @protocol RTC_OBJC_TYPE
+(RTCSSLCertificateVerifier)<NSObject>
+
+    /** The certificate to verify */
+    - (BOOL)verify : (NSData *)derCertificate;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/base/RTCVideoCapturer.h b/third_party/libwebrtc/sdk/objc/base/RTCVideoCapturer.h
new file mode 100644
index 0000000000..a1ffdcf38e
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/base/RTCVideoCapturer.h
@@ -0,0 +1,35 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCVideoFrame.h"
+
+#import "RTCMacros.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+@class RTC_OBJC_TYPE(RTCVideoCapturer);
+
+RTC_OBJC_EXPORT
+@protocol RTC_OBJC_TYPE
+(RTCVideoCapturerDelegate)<NSObject> -
+    (void)capturer : (RTC_OBJC_TYPE(RTCVideoCapturer) *)capturer didCaptureVideoFrame
+    : (RTC_OBJC_TYPE(RTCVideoFrame) *)frame;
+@end
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCVideoCapturer) : NSObject
+
+@property(nonatomic, weak) id<RTC_OBJC_TYPE(RTCVideoCapturerDelegate)> delegate;
+
+- (instancetype)initWithDelegate:(id<RTC_OBJC_TYPE(RTCVideoCapturerDelegate)>)delegate;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/base/RTCVideoCapturer.m b/third_party/libwebrtc/sdk/objc/base/RTCVideoCapturer.m
new file mode 100644
index 0000000000..ca31a731f0
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/base/RTCVideoCapturer.m
@@ -0,0 +1,24 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCVideoCapturer.h"
+
+@implementation RTC_OBJC_TYPE (RTCVideoCapturer)
+
+@synthesize delegate = _delegate;
+
+- (instancetype)initWithDelegate:(id<RTC_OBJC_TYPE(RTCVideoCapturerDelegate)>)delegate {
+  if (self = [super init]) {
+    _delegate = delegate;
+  }
+  return self;
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/base/RTCVideoCodecInfo.h b/third_party/libwebrtc/sdk/objc/base/RTCVideoCodecInfo.h
new file mode 100644
index 0000000000..fa28958f25
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/base/RTCVideoCodecInfo.h
@@ -0,0 +1,36 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+/** Holds information to identify a codec. Corresponds to webrtc::SdpVideoFormat. */
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCVideoCodecInfo) : NSObject <NSCoding>
+
+- (instancetype)init NS_UNAVAILABLE;
+
+- (instancetype)initWithName:(NSString *)name;
+
+- (instancetype)initWithName:(NSString *)name
+                  parameters:(nullable NSDictionary<NSString *, NSString *> *)parameters
+    NS_DESIGNATED_INITIALIZER;
+
+- (BOOL)isEqualToCodecInfo:(RTC_OBJC_TYPE(RTCVideoCodecInfo) *)info;
+
+@property(nonatomic, readonly) NSString *name;
+@property(nonatomic, readonly) NSDictionary<NSString *, NSString *> *parameters;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/base/RTCVideoCodecInfo.m b/third_party/libwebrtc/sdk/objc/base/RTCVideoCodecInfo.m
new file mode 100644
index 0000000000..ce26ae1de3
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/base/RTCVideoCodecInfo.m
@@ -0,0 +1,65 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCVideoCodecInfo.h"
+
+@implementation RTC_OBJC_TYPE (RTCVideoCodecInfo)
+
+@synthesize name = _name;
+@synthesize parameters = _parameters;
+
+- (instancetype)initWithName:(NSString *)name {
+  return [self initWithName:name parameters:nil];
+}
+
+- (instancetype)initWithName:(NSString *)name
+                  parameters:(nullable NSDictionary<NSString *, NSString *> *)parameters {
+  if (self = [super init]) {
+    _name = name;
+    _parameters = (parameters ? parameters : @{});
+  }
+
+  return self;
+}
+
+- (BOOL)isEqualToCodecInfo:(RTC_OBJC_TYPE(RTCVideoCodecInfo) *)info {
+  if (!info ||
+      ![self.name isEqualToString:info.name] ||
+      ![self.parameters isEqualToDictionary:info.parameters]) {
+    return NO;
+  }
+  return YES;
+}
+
+- (BOOL)isEqual:(id)object {
+  if (self == object)
+    return YES;
+  if (![object isKindOfClass:[self class]])
+    return NO;
+  return [self isEqualToCodecInfo:object];
+}
+
+- (NSUInteger)hash {
+  return [self.name hash] ^ [self.parameters hash];
+}
+
+#pragma mark - NSCoding
+
+- (instancetype)initWithCoder:(NSCoder *)decoder {
+  return [self initWithName:[decoder decodeObjectForKey:@"name"]
+                 parameters:[decoder decodeObjectForKey:@"parameters"]];
+}
+
+- (void)encodeWithCoder:(NSCoder *)encoder {
+  [encoder encodeObject:_name forKey:@"name"];
+  [encoder encodeObject:_parameters forKey:@"parameters"];
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/base/RTCVideoDecoder.h b/third_party/libwebrtc/sdk/objc/base/RTCVideoDecoder.h
new file mode 100644
index 0000000000..ccddd42d42
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/base/RTCVideoDecoder.h
@@ -0,0 +1,40 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCCodecSpecificInfo.h"
+#import "RTCEncodedImage.h"
+#import "RTCMacros.h"
+#import "RTCVideoEncoderSettings.h"
+#import "RTCVideoFrame.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+/** Callback block for decoder. */
+typedef void (^RTCVideoDecoderCallback)(RTC_OBJC_TYPE(RTCVideoFrame) * frame);
+
+/** Protocol for decoder implementations. */
+RTC_OBJC_EXPORT
+@protocol RTC_OBJC_TYPE
+(RTCVideoDecoder)<NSObject>
+
+    - (void)setCallback : (RTCVideoDecoderCallback)callback;
+- (NSInteger)startDecodeWithNumberOfCores:(int)numberOfCores;
+- (NSInteger)releaseDecoder;
+- (NSInteger)decode:(RTC_OBJC_TYPE(RTCEncodedImage) *)encodedImage
+        missingFrames:(BOOL)missingFrames
+    codecSpecificInfo:(nullable id<RTC_OBJC_TYPE(RTCCodecSpecificInfo)>)info
+         renderTimeMs:(int64_t)renderTimeMs;
+- (NSString *)implementationName;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/base/RTCVideoDecoderFactory.h b/third_party/libwebrtc/sdk/objc/base/RTCVideoDecoderFactory.h
new file mode 100644
index 0000000000..8d90138521
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/base/RTCVideoDecoderFactory.h
@@ -0,0 +1,32 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+#import "RTCVideoCodecInfo.h"
+#import "RTCVideoDecoder.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+/** RTCVideoDecoderFactory is an Objective-C version of webrtc::VideoDecoderFactory.
+ */
+RTC_OBJC_EXPORT
+@protocol RTC_OBJC_TYPE
+(RTCVideoDecoderFactory)<NSObject>
+
+    - (nullable id<RTC_OBJC_TYPE(RTCVideoDecoder)>)createDecoder
+    : (RTC_OBJC_TYPE(RTCVideoCodecInfo) *)info;
+- (NSArray<RTC_OBJC_TYPE(RTCVideoCodecInfo) *> *)
+    supportedCodecs;  // TODO(andersc): "supportedFormats" instead?
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/base/RTCVideoEncoder.h b/third_party/libwebrtc/sdk/objc/base/RTCVideoEncoder.h
new file mode 100644
index 0000000000..2445d432d6
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/base/RTCVideoEncoder.h
@@ -0,0 +1,59 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCCodecSpecificInfo.h"
+#import "RTCEncodedImage.h"
+#import "RTCMacros.h"
+#import "RTCVideoEncoderQpThresholds.h"
+#import "RTCVideoEncoderSettings.h"
+#import "RTCVideoFrame.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+/** Callback block for encoder. */
+typedef BOOL (^RTCVideoEncoderCallback)(RTC_OBJC_TYPE(RTCEncodedImage) * frame,
+                                        id<RTC_OBJC_TYPE(RTCCodecSpecificInfo)> info);
+
+/** Protocol for encoder implementations. */
+RTC_OBJC_EXPORT
+@protocol RTC_OBJC_TYPE
+(RTCVideoEncoder)<NSObject>
+
+- (void)setCallback:(nullable RTCVideoEncoderCallback)callback;
+- (NSInteger)startEncodeWithSettings:(RTC_OBJC_TYPE(RTCVideoEncoderSettings) *)settings
+                       numberOfCores:(int)numberOfCores;
+- (NSInteger)releaseEncoder;
+- (NSInteger)encode:(RTC_OBJC_TYPE(RTCVideoFrame) *)frame
+    codecSpecificInfo:(nullable id<RTC_OBJC_TYPE(RTCCodecSpecificInfo)>)info
+           frameTypes:(NSArray<NSNumber *> *)frameTypes;
+- (int)setBitrate:(uint32_t)bitrateKbit framerate:(uint32_t)framerate;
+- (NSString *)implementationName;
+
+/** Returns QP scaling settings for encoder. The quality scaler adjusts the resolution in order to
+ *  keep the QP from the encoded images within the given range. Returning nil from this function
+ *  disables quality scaling. */
+- (nullable RTC_OBJC_TYPE(RTCVideoEncoderQpThresholds) *)scalingSettings;
+
+/** Resolutions should be aligned to this value. */
+@property(nonatomic, readonly) NSInteger resolutionAlignment;
+
+/** If enabled, resolution alignment is applied to all simulcast layers simultaneously so that when
+    scaled, all resolutions comply with 'resolutionAlignment'. */
+@property(nonatomic, readonly) BOOL applyAlignmentToAllSimulcastLayers;
+
+/** If YES, the receiver is expected to resample/scale the source texture to the expected output
+    size. */
+@property(nonatomic, readonly) BOOL supportsNativeHandle;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/base/RTCVideoEncoderFactory.h b/third_party/libwebrtc/sdk/objc/base/RTCVideoEncoderFactory.h
new file mode 100644
index 0000000000..a73cd77990
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/base/RTCVideoEncoderFactory.h
@@ -0,0 +1,52 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+#import "RTCVideoCodecInfo.h"
+#import "RTCVideoEncoder.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+/** RTCVideoEncoderFactory is an Objective-C version of
+ webrtc::VideoEncoderFactory::VideoEncoderSelector.
+ */
+RTC_OBJC_EXPORT
+@protocol RTC_OBJC_TYPE
+(RTCVideoEncoderSelector)<NSObject>
+
+    - (void)registerCurrentEncoderInfo : (RTC_OBJC_TYPE(RTCVideoCodecInfo) *)info;
+- (nullable RTC_OBJC_TYPE(RTCVideoCodecInfo) *)encoderForBitrate:(NSInteger)bitrate;
+- (nullable RTC_OBJC_TYPE(RTCVideoCodecInfo) *)encoderForBrokenEncoder;
+
+@optional
+- (nullable RTC_OBJC_TYPE(RTCVideoCodecInfo) *)encoderForResolutionChangeBySize:(CGSize)size;
+
+@end
+
+/** RTCVideoEncoderFactory is an Objective-C version of webrtc::VideoEncoderFactory.
+ */
+RTC_OBJC_EXPORT
+@protocol RTC_OBJC_TYPE
+(RTCVideoEncoderFactory)<NSObject>
+
+    - (nullable id<RTC_OBJC_TYPE(RTCVideoEncoder)>)createEncoder
+    : (RTC_OBJC_TYPE(RTCVideoCodecInfo) *)info;
+- (NSArray<RTC_OBJC_TYPE(RTCVideoCodecInfo) *> *)
+    supportedCodecs;  // TODO(andersc): "supportedFormats" instead?
+
+@optional
+- (NSArray<RTC_OBJC_TYPE(RTCVideoCodecInfo) *> *)implementations;
+- (nullable id<RTC_OBJC_TYPE(RTCVideoEncoderSelector)>)encoderSelector;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/base/RTCVideoEncoderQpThresholds.h b/third_party/libwebrtc/sdk/objc/base/RTCVideoEncoderQpThresholds.h
new file mode 100644
index 0000000000..1a6e9e88ab
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/base/RTCVideoEncoderQpThresholds.h
@@ -0,0 +1,28 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+/** QP thresholds for encoder. Corresponds to webrtc::VideoEncoder::QpThresholds. */
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCVideoEncoderQpThresholds) : NSObject
+
+- (instancetype)initWithThresholdsLow:(NSInteger)low high:(NSInteger)high;
+
+@property(nonatomic, readonly) NSInteger low;
+@property(nonatomic, readonly) NSInteger high;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/base/RTCVideoEncoderQpThresholds.m b/third_party/libwebrtc/sdk/objc/base/RTCVideoEncoderQpThresholds.m
new file mode 100644
index 0000000000..fb7012f44f
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/base/RTCVideoEncoderQpThresholds.m
@@ -0,0 +1,26 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCVideoEncoderQpThresholds.h"
+
+@implementation RTC_OBJC_TYPE (RTCVideoEncoderQpThresholds)
+
+@synthesize low = _low;
+@synthesize high = _high;
+
+- (instancetype)initWithThresholdsLow:(NSInteger)low high:(NSInteger)high {
+  if (self = [super init]) {
+    _low = low;
+    _high = high;
+  }
+  return self;
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/base/RTCVideoEncoderSettings.h b/third_party/libwebrtc/sdk/objc/base/RTCVideoEncoderSettings.h
new file mode 100644
index 0000000000..ae792eab71
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/base/RTCVideoEncoderSettings.h
@@ -0,0 +1,42 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+typedef NS_ENUM(NSUInteger, RTCVideoCodecMode) {
+  RTCVideoCodecModeRealtimeVideo,
+  RTCVideoCodecModeScreensharing,
+};
+
+/** Settings for encoder. Corresponds to webrtc::VideoCodec. */
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCVideoEncoderSettings) : NSObject
+
+@property(nonatomic, strong) NSString *name;
+
+@property(nonatomic, assign) unsigned short width;
+@property(nonatomic, assign) unsigned short height;
+
+@property(nonatomic, assign) unsigned int startBitrate;  // kilobits/sec.
+@property(nonatomic, assign) unsigned int maxBitrate;
+@property(nonatomic, assign) unsigned int minBitrate;
+
+@property(nonatomic, assign) uint32_t maxFramerate;
+
+@property(nonatomic, assign) unsigned int qpMax;
+@property(nonatomic, assign) RTCVideoCodecMode mode;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/base/RTCVideoEncoderSettings.m b/third_party/libwebrtc/sdk/objc/base/RTCVideoEncoderSettings.m
new file mode 100644
index 0000000000..f66cd2cf77
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/base/RTCVideoEncoderSettings.m
@@ -0,0 +1,25 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCVideoEncoderSettings.h"
+
+@implementation RTC_OBJC_TYPE (RTCVideoEncoderSettings)
+
+@synthesize name = _name;
+@synthesize width = _width;
+@synthesize height = _height;
+@synthesize startBitrate = _startBitrate;
+@synthesize maxBitrate = _maxBitrate;
+@synthesize minBitrate = _minBitrate;
+@synthesize maxFramerate = _maxFramerate;
+@synthesize qpMax = _qpMax;
+@synthesize mode = _mode;
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/base/RTCVideoFrame.h b/third_party/libwebrtc/sdk/objc/base/RTCVideoFrame.h
new file mode 100644
index 0000000000..f5638d27cf
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/base/RTCVideoFrame.h
@@ -0,0 +1,86 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <AVFoundation/AVFoundation.h>
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+typedef NS_ENUM(NSInteger, RTCVideoRotation) {
+  RTCVideoRotation_0 = 0,
+  RTCVideoRotation_90 = 90,
+  RTCVideoRotation_180 = 180,
+  RTCVideoRotation_270 = 270,
+};
+
+@protocol RTC_OBJC_TYPE
+(RTCVideoFrameBuffer);
+
+// RTCVideoFrame is an ObjectiveC version of webrtc::VideoFrame.
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCVideoFrame) : NSObject
+
+/** Width without rotation applied. */
+@property(nonatomic, readonly) int width;
+
+/** Height without rotation applied. */
+@property(nonatomic, readonly) int height;
+@property(nonatomic, readonly) RTCVideoRotation rotation;
+
+/** Timestamp in nanoseconds. */
+@property(nonatomic, readonly) int64_t timeStampNs;
+
+/** Timestamp 90 kHz. */
+@property(nonatomic, assign) int32_t timeStamp;
+
+@property(nonatomic, readonly) id<RTC_OBJC_TYPE(RTCVideoFrameBuffer)> buffer;
+
+- (instancetype)init NS_UNAVAILABLE;
+- (instancetype) new NS_UNAVAILABLE;
+
+/** Initialize an RTCVideoFrame from a pixel buffer, rotation, and timestamp.
+ *  Deprecated - initialize with a RTCCVPixelBuffer instead
+ */
+- (instancetype)initWithPixelBuffer:(CVPixelBufferRef)pixelBuffer
+                           rotation:(RTCVideoRotation)rotation
+                        timeStampNs:(int64_t)timeStampNs
+    DEPRECATED_MSG_ATTRIBUTE("use initWithBuffer instead");
+
+/** Initialize an RTCVideoFrame from a pixel buffer combined with cropping and
+ *  scaling. Cropping will be applied first on the pixel buffer, followed by
+ *  scaling to the final resolution of scaledWidth x scaledHeight.
+ */
+- (instancetype)initWithPixelBuffer:(CVPixelBufferRef)pixelBuffer
+                        scaledWidth:(int)scaledWidth
+                       scaledHeight:(int)scaledHeight
+                          cropWidth:(int)cropWidth
+                         cropHeight:(int)cropHeight
+                              cropX:(int)cropX
+                              cropY:(int)cropY
+                           rotation:(RTCVideoRotation)rotation
+                        timeStampNs:(int64_t)timeStampNs
+    DEPRECATED_MSG_ATTRIBUTE("use initWithBuffer instead");
+
+/** Initialize an RTCVideoFrame from a frame buffer, rotation, and timestamp.
+ */
+- (instancetype)initWithBuffer:(id<RTC_OBJC_TYPE(RTCVideoFrameBuffer)>)frameBuffer
+                      rotation:(RTCVideoRotation)rotation
+                   timeStampNs:(int64_t)timeStampNs;
+
+/** Return a frame that is guaranteed to be I420, i.e. it is possible to access
+ *  the YUV data on it.
+ */
+- (RTC_OBJC_TYPE(RTCVideoFrame) *)newI420VideoFrame;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/base/RTCVideoFrame.mm b/third_party/libwebrtc/sdk/objc/base/RTCVideoFrame.mm
new file mode 100644
index 0000000000..e162238d73
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/base/RTCVideoFrame.mm
@@ -0,0 +1,78 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCVideoFrame.h"
+
+#import "RTCI420Buffer.h"
+#import "RTCVideoFrameBuffer.h"
+
+@implementation RTC_OBJC_TYPE (RTCVideoFrame) {
+  RTCVideoRotation _rotation;
+  int64_t _timeStampNs;
+}
+
+@synthesize buffer = _buffer;
+@synthesize timeStamp;
+
+- (int)width {
+  return _buffer.width;
+}
+
+- (int)height {
+  return _buffer.height;
+}
+
+- (RTCVideoRotation)rotation {
+  return _rotation;
+}
+
+- (int64_t)timeStampNs {
+  return _timeStampNs;
+}
+
+- (RTC_OBJC_TYPE(RTCVideoFrame) *)newI420VideoFrame {
+  return [[RTC_OBJC_TYPE(RTCVideoFrame) alloc] initWithBuffer:[_buffer toI420]
+                                                     rotation:_rotation
+                                                  timeStampNs:_timeStampNs];
+}
+
+- (instancetype)initWithPixelBuffer:(CVPixelBufferRef)pixelBuffer
+                           rotation:(RTCVideoRotation)rotation
+                        timeStampNs:(int64_t)timeStampNs {
+  // Deprecated.
+  return nil;
+}
+
+- (instancetype)initWithPixelBuffer:(CVPixelBufferRef)pixelBuffer
+                        scaledWidth:(int)scaledWidth
+                       scaledHeight:(int)scaledHeight
+                          cropWidth:(int)cropWidth
+                         cropHeight:(int)cropHeight
+                              cropX:(int)cropX
+                              cropY:(int)cropY
+                           rotation:(RTCVideoRotation)rotation
+                        timeStampNs:(int64_t)timeStampNs {
+  // Deprecated.
+  return nil;
+}
+
+- (instancetype)initWithBuffer:(id<RTC_OBJC_TYPE(RTCVideoFrameBuffer)>)buffer
+                      rotation:(RTCVideoRotation)rotation
+                   timeStampNs:(int64_t)timeStampNs {
+  if (self = [super init]) {
+    _buffer = buffer;
+    _rotation = rotation;
+    _timeStampNs = timeStampNs;
+  }
+
+  return self;
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/base/RTCVideoFrameBuffer.h b/third_party/libwebrtc/sdk/objc/base/RTCVideoFrameBuffer.h
new file mode 100644
index 0000000000..a36807951b
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/base/RTCVideoFrameBuffer.h
@@ -0,0 +1,40 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <AVFoundation/AVFoundation.h>
+
+#import "RTCMacros.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+@protocol RTC_OBJC_TYPE
+(RTCI420Buffer);
+
+// RTCVideoFrameBuffer is an ObjectiveC version of webrtc::VideoFrameBuffer.
+RTC_OBJC_EXPORT
+@protocol RTC_OBJC_TYPE
+(RTCVideoFrameBuffer)<NSObject>
+
+    @property(nonatomic, readonly) int width;
+@property(nonatomic, readonly) int height;
+
+- (id<RTC_OBJC_TYPE(RTCI420Buffer)>)toI420;
+
+@optional
+- (id<RTC_OBJC_TYPE(RTCVideoFrameBuffer)>)cropAndScaleWith:(int)offsetX
+                                                   offsetY:(int)offsetY
+                                                 cropWidth:(int)cropWidth
+                                                cropHeight:(int)cropHeight
+                                                scaleWidth:(int)scaleWidth
+                                               scaleHeight:(int)scaleHeight;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/base/RTCVideoRenderer.h b/third_party/libwebrtc/sdk/objc/base/RTCVideoRenderer.h
new file mode 100644
index 0000000000..0f763295ad
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/base/RTCVideoRenderer.h
@@ -0,0 +1,43 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+#if TARGET_OS_IPHONE
+#import <UIKit/UIKit.h>
+#endif
+
+#import "RTCMacros.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+@class RTC_OBJC_TYPE(RTCVideoFrame);
+
+RTC_OBJC_EXPORT
+@protocol RTC_OBJC_TYPE
+(RTCVideoRenderer)<NSObject>
+
+    /** The size of the frame. */
+    - (void)setSize : (CGSize)size;
+
+/** The frame to be displayed. */
+- (void)renderFrame:(nullable RTC_OBJC_TYPE(RTCVideoFrame) *)frame;
+
+@end
+
+RTC_OBJC_EXPORT
+@protocol RTC_OBJC_TYPE
+(RTCVideoViewDelegate)
+
+    - (void)videoView : (id<RTC_OBJC_TYPE(RTCVideoRenderer)>)videoView didChangeVideoSize
+    : (CGSize)size;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/base/RTCYUVPlanarBuffer.h b/third_party/libwebrtc/sdk/objc/base/RTCYUVPlanarBuffer.h
new file mode 100644
index 0000000000..be01b915f5
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/base/RTCYUVPlanarBuffer.h
@@ -0,0 +1,46 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <AVFoundation/AVFoundation.h>
+
+#import "RTCMacros.h"
+#import "RTCVideoFrameBuffer.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+/** Protocol for RTCVideoFrameBuffers containing YUV planar data. */
+RTC_OBJC_EXPORT
+@protocol RTC_OBJC_TYPE
+(RTCYUVPlanarBuffer)<RTC_OBJC_TYPE(RTCVideoFrameBuffer)>
+
+    @property(nonatomic, readonly) int chromaWidth;
+@property(nonatomic, readonly) int chromaHeight;
+@property(nonatomic, readonly) const uint8_t *dataY;
+@property(nonatomic, readonly) const uint8_t *dataU;
+@property(nonatomic, readonly) const uint8_t *dataV;
+@property(nonatomic, readonly) int strideY;
+@property(nonatomic, readonly) int strideU;
+@property(nonatomic, readonly) int strideV;
+
+- (instancetype)initWithWidth:(int)width
+                       height:(int)height
+                        dataY:(const uint8_t *)dataY
+                        dataU:(const uint8_t *)dataU
+                        dataV:(const uint8_t *)dataV;
+- (instancetype)initWithWidth:(int)width height:(int)height;
+- (instancetype)initWithWidth:(int)width
+                       height:(int)height
+                      strideY:(int)strideY
+                      strideU:(int)strideU
+                      strideV:(int)strideV;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/components/audio/RTCAudioDevice.h b/third_party/libwebrtc/sdk/objc/components/audio/RTCAudioDevice.h
new file mode 100644
index 0000000000..f445825ff0
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/components/audio/RTCAudioDevice.h
@@ -0,0 +1,308 @@
+/*
+ *  Copyright 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <AudioUnit/AudioUnit.h>
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+typedef OSStatus (^RTC_OBJC_TYPE(RTCAudioDeviceGetPlayoutDataBlock))(
+    AudioUnitRenderActionFlags *_Nonnull actionFlags,
+    const AudioTimeStamp *_Nonnull timestamp,
+    NSInteger inputBusNumber,
+    UInt32 frameCount,
+    AudioBufferList *_Nonnull outputData);
+
+typedef OSStatus (^RTC_OBJC_TYPE(RTCAudioDeviceRenderRecordedDataBlock))(
+    AudioUnitRenderActionFlags *_Nonnull actionFlags,
+    const AudioTimeStamp *_Nonnull timestamp,
+    NSInteger inputBusNumber,
+    UInt32 frameCount,
+    AudioBufferList *_Nonnull inputData,
+    void *_Nullable renderContext);
+
+typedef OSStatus (^RTC_OBJC_TYPE(RTCAudioDeviceDeliverRecordedDataBlock))(
+    AudioUnitRenderActionFlags *_Nonnull actionFlags,
+    const AudioTimeStamp *_Nonnull timestamp,
+    NSInteger inputBusNumber,
+    UInt32 frameCount,
+    const AudioBufferList *_Nullable inputData,
+    void *_Nullable renderContext,
+    NS_NOESCAPE RTC_OBJC_TYPE(RTCAudioDeviceRenderRecordedDataBlock) _Nullable renderBlock);
+
+/**
+ * Delegate object provided by native ADM during RTCAudioDevice initialization.
+ * Provides blocks to poll playback audio samples from native ADM and to feed
+ * recorded audio samples into native ADM.
+ */
+RTC_OBJC_EXPORT @protocol RTC_OBJC_TYPE
+(RTCAudioDeviceDelegate)<NSObject>
+    /**
+     * Implementation of RTCAudioSource should call this block to feed recorded PCM (16-bit integer)
+     * into native ADM. Stereo data is expected to be interleaved starting with the left channel.
+     * Either `inputData` with pre-filled audio data must be provided during block
+     * call or `renderBlock` must be provided which must fill provided audio buffer with recorded
+     * samples.
+     *
+     * NOTE: Implementation of RTCAudioDevice is expected to call the block on the same thread until
+     * `notifyAudioInterrupted` is called. When `notifyAudioInterrupted` is called implementation
+     * can call the block on a different thread.
+     */
+    @property(readonly, nonnull)
+        RTC_OBJC_TYPE(RTCAudioDeviceDeliverRecordedDataBlock) deliverRecordedData;
+
+/**
+ * Provides input sample rate preference as it preferred by native ADM.
+ */
+@property(readonly) double preferredInputSampleRate;
+
+/**
+ * Provides input IO buffer duration preference as it preferred by native ADM.
+ */
+@property(readonly) NSTimeInterval preferredInputIOBufferDuration;
+
+/**
+ * Provides output sample rate preference as it preferred by native ADM.
+ */
+@property(readonly) double preferredOutputSampleRate;
+
+/**
+ * Provides output IO buffer duration preference as it preferred by native ADM.
+ */
+@property(readonly) NSTimeInterval preferredOutputIOBufferDuration;
+
+/**
+ * Implementation of RTCAudioDevice should call this block to request PCM (16-bit integer)
+ * from native ADM to play. Stereo data is interleaved starting with the left channel.
+ *
+ * NOTE: Implementation of RTCAudioDevice is expected to invoke of this block on the
+ * same thread until `notifyAudioInterrupted` is called. When `notifyAudioInterrupted` is called
+ * implementation can call the block from a different thread.
+ */
+@property(readonly, nonnull) RTC_OBJC_TYPE(RTCAudioDeviceGetPlayoutDataBlock) getPlayoutData;
+
+/**
+ * Notifies native ADM that some of the audio input parameters of RTCAudioDevice like
+ * samle rate and/or IO buffer duration and/or IO latency had possibly changed.
+ * Native ADM will adjust its audio input buffer to match current parameters of audio device.
+ *
+ * NOTE: Must be called within block executed via `dispatchAsync` or `dispatchSync`.
+ */
+- (void)notifyAudioInputParametersChange;
+
+/**
+ * Notifies native ADM that some of the audio output parameters of RTCAudioDevice like
+ * samle rate and/or IO buffer duration and/or IO latency had possibly changed.
+ * Native ADM will adjust its audio output buffer to match current parameters of audio device.
+ *
+ * NOTE: Must be called within block executed via `dispatchAsync` or `dispatchSync`.
+ */
+- (void)notifyAudioOutputParametersChange;
+
+/**
+ * Notifies native ADM that audio input is interrupted and further audio playout
+ * and recording might happen on a different thread.
+ *
+ * NOTE: Must be called within block executed via `dispatchAsync` or `dispatchSync`.
+ */
+- (void)notifyAudioInputInterrupted;
+
+/**
+ * Notifies native ADM that audio output is interrupted and further audio playout
+ * and recording might happen on a different thread.
+ *
+ * NOTE: Must be called within block executed via `dispatchAsync` or `dispatchSync`.
+ */
+- (void)notifyAudioOutputInterrupted;
+
+/**
+ * Asynchronously execute block of code within the context of
+ * thread which owns native ADM.
+ *
+ * NOTE: Intended to be used to invoke `notifyAudioInputParametersChange`,
+ * `notifyAudioOutputParametersChange`, `notifyAudioInputInterrupted`,
+ * `notifyAudioOutputInterrupted` on native ADM thread.
+ * Also could be used by `RTCAudioDevice` implementation to tie
+ * mutations of underlying audio objects (AVAudioEngine, AudioUnit, etc)
+ * to the native ADM thread. Could be useful to handle events like audio route change, which
+ * could lead to audio parameters change.
+ */
+- (void)dispatchAsync:(dispatch_block_t)block;
+
+/**
+ * Synchronously execute block of code within the context of
+ * thread which owns native ADM. Allows reentrancy.
+ *
+ * NOTE: Intended to be used to invoke `notifyAudioInputParametersChange`,
+ * `notifyAudioOutputParametersChange`, `notifyAudioInputInterrupted`,
+ * `notifyAudioOutputInterrupted` on native ADM thread and make sure
+ * aforementioned is completed before `dispatchSync` returns. Could be useful
+ * when implementation of `RTCAudioDevice` tie mutation to underlying audio objects (AVAudioEngine,
+ * AudioUnit, etc) to own thread to satisfy requirement that native ADM audio parameters
+ * must be kept in sync with current audio parameters before audio is actually played or recorded.
+ */
+- (void)dispatchSync:(dispatch_block_t)block;
+
+@end
+
+/**
+ * Protocol to abstract platform specific ways to implement playback and recording.
+ *
+ * NOTE: All the members of protocol are called by native ADM from the same thread
+ * between calls to `initializeWithDelegate` and `terminate`.
+ * NOTE: Implementation is fully responsible for configuring application's AVAudioSession.
+ * An example implementation of RTCAudioDevice: https://github.com/mstyura/RTCAudioDevice
+ * TODO(yura.yaroshevich): Implement custom RTCAudioDevice for AppRTCMobile demo app.
+ */
+RTC_OBJC_EXPORT @protocol RTC_OBJC_TYPE
+(RTCAudioDevice)<NSObject>
+
+    /**
+     * Indicates current sample rate of audio recording. Changes to this property
+     * must be notified back to native ADM via `-[RTCAudioDeviceDelegate
+     * notifyAudioParametersChange]`.
+     */
+    @property(readonly) double deviceInputSampleRate;
+
+/**
+ * Indicates current size of record buffer. Changes to this property
+ * must be notified back to native ADM via `-[RTCAudioDeviceDelegate notifyAudioParametersChange]`.
+ */
+@property(readonly) NSTimeInterval inputIOBufferDuration;
+
+/**
+ * Indicates current number of recorded audio channels. Changes to this property
+ * must be notified back to native ADM via `-[RTCAudioDeviceDelegate notifyAudioParametersChange]`.
+ */
+@property(readonly) NSInteger inputNumberOfChannels;
+
+/**
+ * Indicates current input latency
+ */
+@property(readonly) NSTimeInterval inputLatency;
+
+/**
+ * Indicates current sample rate of audio playback. Changes to this property
+ * must be notified back to native ADM via `-[RTCAudioDeviceDelegate notifyAudioParametersChange]`.
+ */
+@property(readonly) double deviceOutputSampleRate;
+
+/**
+ * Indicates current size of playback buffer. Changes to this property
+ * must be notified back to native ADM via `-[RTCAudioDeviceDelegate notifyAudioParametersChange]`.
+ */
+@property(readonly) NSTimeInterval outputIOBufferDuration;
+
+/**
+ * Indicates current number of playback audio channels. Changes to this property
+ * must be notified back to WebRTC via `[RTCAudioDeviceDelegate notifyAudioParametersChange]`.
+ */
+@property(readonly) NSInteger outputNumberOfChannels;
+
+/**
+ * Indicates current output latency
+ */
+@property(readonly) NSTimeInterval outputLatency;
+
+/**
+ * Indicates if invocation of `initializeWithDelegate` required before usage of RTCAudioDevice.
+ * YES indicates that `initializeWithDelegate` was called earlier without subsequent call to
+ * `terminate`. NO indicates that either `initializeWithDelegate` not called or `terminate` called.
+ */
+@property(readonly) BOOL isInitialized;
+
+/**
+ * Initializes RTCAudioDevice with RTCAudioDeviceDelegate.
+ * Implementation must return YES if RTCAudioDevice initialized successfully and NO otherwise.
+ */
+- (BOOL)initializeWithDelegate:(id<RTC_OBJC_TYPE(RTCAudioDeviceDelegate)>)delegate;
+
+/**
+ * De-initializes RTCAudioDevice. Implementation should forget about `delegate` provided in
+ * `initializeWithDelegate`.
+ */
+- (BOOL)terminateDevice;
+
+/**
+ * Property to indicate if `initializePlayout` call required before invocation of `startPlayout`.
+ * YES indicates that `initializePlayout` was successfully invoked earlier or not necessary,
+ * NO indicates that `initializePlayout` invocation required.
+ */
+@property(readonly) BOOL isPlayoutInitialized;
+
+/**
+ * Prepares RTCAudioDevice to play audio.
+ * Called by native ADM before invocation of `startPlayout`.
+ * Implementation is expected to return YES in case of successful playout initialization and NO
+ * otherwise.
+ */
+- (BOOL)initializePlayout;
+
+/**
+ * Property to indicate if RTCAudioDevice should be playing according to
+ * earlier calls of `startPlayout` and `stopPlayout`.
+ */
+@property(readonly) BOOL isPlaying;
+
+/**
+ * Method is called when native ADM wants to play audio.
+ * Implementation is expected to return YES if playback start request
+ * successfully handled and NO otherwise.
+ */
+- (BOOL)startPlayout;
+
+/**
+ * Method is called when native ADM no longer needs to play audio.
+ * Implementation is expected to return YES if playback stop request
+ * successfully handled and NO otherwise.
+ */
+- (BOOL)stopPlayout;
+
+/**
+ * Property to indicate if `initializeRecording` call required before usage of `startRecording`.
+ * YES indicates that `initializeRecording` was successfully invoked earlier or not necessary,
+ * NO indicates that `initializeRecording` invocation required.
+ */
+@property(readonly) BOOL isRecordingInitialized;
+
+/**
+ * Prepares RTCAudioDevice to record audio.
+ * Called by native ADM before invocation of `startRecording`.
+ * Implementation may use this method to prepare resources required to record audio.
+ * Implementation is expected to return YES in case of successful record initialization and NO
+ * otherwise.
+ */
+- (BOOL)initializeRecording;
+
+/**
+ * Property to indicate if RTCAudioDevice should record audio according to
+ * earlier calls to `startRecording` and `stopRecording`.
+ */
+@property(readonly) BOOL isRecording;
+
+/**
+ * Method is called when native ADM wants to record audio.
+ * Implementation is expected to return YES if recording start request
+ * successfully handled and NO otherwise.
+ */
+- (BOOL)startRecording;
+
+/**
+ * Method is called when native ADM no longer needs to record audio.
+ * Implementation is expected to return YES if recording stop request
+ * successfully handled and NO otherwise.
+ */
+- (BOOL)stopRecording;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/components/audio/RTCAudioSession+Configuration.mm b/third_party/libwebrtc/sdk/objc/components/audio/RTCAudioSession+Configuration.mm
new file mode 100644
index 0000000000..449f31e9dd
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/components/audio/RTCAudioSession+Configuration.mm
@@ -0,0 +1,176 @@
+/*
+ *  Copyright 2016 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCAudioSession+Private.h"
+#import "RTCAudioSessionConfiguration.h"
+
+#import "base/RTCLogging.h"
+
+@implementation RTC_OBJC_TYPE (RTCAudioSession)
+(Configuration)
+
+    - (BOOL)setConfiguration : (RTC_OBJC_TYPE(RTCAudioSessionConfiguration) *)configuration error
+    : (NSError **)outError {
+  return [self setConfiguration:configuration
+                         active:NO
+                shouldSetActive:NO
+                          error:outError];
+}
+
+- (BOOL)setConfiguration:(RTC_OBJC_TYPE(RTCAudioSessionConfiguration) *)configuration
+                  active:(BOOL)active
+                   error:(NSError **)outError {
+  return [self setConfiguration:configuration
+                         active:active
+                shouldSetActive:YES
+                          error:outError];
+}
+
+#pragma mark - Private
+
+- (BOOL)setConfiguration:(RTC_OBJC_TYPE(RTCAudioSessionConfiguration) *)configuration
+                  active:(BOOL)active
+         shouldSetActive:(BOOL)shouldSetActive
+                   error:(NSError **)outError {
+  NSParameterAssert(configuration);
+  if (outError) {
+    *outError = nil;
+  }
+
+  // Provide an error even if there isn't one so we can log it. We will not
+  // return immediately on error in this function and instead try to set
+  // everything we can.
+  NSError *error = nil;
+
+  if (self.category != configuration.category ||
+      self.categoryOptions != configuration.categoryOptions) {
+    NSError *categoryError = nil;
+    if (![self setCategory:configuration.category
+               withOptions:configuration.categoryOptions
+                     error:&categoryError]) {
+      RTCLogError(@"Failed to set category: %@",
+                  categoryError.localizedDescription);
+      error = categoryError;
+    } else {
+      RTCLog(@"Set category to: %@", configuration.category);
+    }
+  }
+
+  if (self.mode != configuration.mode) {
+    NSError *modeError = nil;
+    if (![self setMode:configuration.mode error:&modeError]) {
+      RTCLogError(@"Failed to set mode: %@",
+                  modeError.localizedDescription);
+      error = modeError;
+    } else {
+      RTCLog(@"Set mode to: %@", configuration.mode);
+    }
+  }
+
+  // Sometimes category options don't stick after setting mode.
+  if (self.categoryOptions != configuration.categoryOptions) {
+    NSError *categoryError = nil;
+    if (![self setCategory:configuration.category
+               withOptions:configuration.categoryOptions
+                     error:&categoryError]) {
+      RTCLogError(@"Failed to set category options: %@",
+                  categoryError.localizedDescription);
+      error = categoryError;
+    } else {
+      RTCLog(@"Set category options to: %ld",
+             (long)configuration.categoryOptions);
+    }
+  }
+
+  if (self.preferredSampleRate != configuration.sampleRate) {
+    NSError *sampleRateError = nil;
+    if (![self setPreferredSampleRate:configuration.sampleRate
+                                error:&sampleRateError]) {
+      RTCLogError(@"Failed to set preferred sample rate: %@",
+                  sampleRateError.localizedDescription);
+      if (!self.ignoresPreferredAttributeConfigurationErrors) {
+        error = sampleRateError;
+      }
+    } else {
+      RTCLog(@"Set preferred sample rate to: %.2f",
+             configuration.sampleRate);
+    }
+  }
+
+  if (self.preferredIOBufferDuration != configuration.ioBufferDuration) {
+    NSError *bufferDurationError = nil;
+    if (![self setPreferredIOBufferDuration:configuration.ioBufferDuration
+                                      error:&bufferDurationError]) {
+      RTCLogError(@"Failed to set preferred IO buffer duration: %@",
+                  bufferDurationError.localizedDescription);
+      if (!self.ignoresPreferredAttributeConfigurationErrors) {
+        error = bufferDurationError;
+      }
+    } else {
+      RTCLog(@"Set preferred IO buffer duration to: %f",
+             configuration.ioBufferDuration);
+    }
+  }
+
+  if (shouldSetActive) {
+    NSError *activeError = nil;
+    if (![self setActive:active error:&activeError]) {
+      RTCLogError(@"Failed to setActive to %d: %@",
+                  active, activeError.localizedDescription);
+      error = activeError;
+    }
+  }
+
+  if (self.isActive &&
+      // TODO(tkchin): Figure out which category/mode numChannels is valid for.
+      [self.mode isEqualToString:AVAudioSessionModeVoiceChat]) {
+    // Try to set the preferred number of hardware audio channels. These calls
+    // must be done after setting the audio session’s category and mode and
+    // activating the session.
+    NSInteger inputNumberOfChannels = configuration.inputNumberOfChannels;
+    if (self.inputNumberOfChannels != inputNumberOfChannels) {
+      NSError *inputChannelsError = nil;
+      if (![self setPreferredInputNumberOfChannels:inputNumberOfChannels
+                                             error:&inputChannelsError]) {
+       RTCLogError(@"Failed to set preferred input number of channels: %@",
+                   inputChannelsError.localizedDescription);
+       if (!self.ignoresPreferredAttributeConfigurationErrors) {
+         error = inputChannelsError;
+       }
+      } else {
+        RTCLog(@"Set input number of channels to: %ld",
+               (long)inputNumberOfChannels);
+      }
+    }
+    NSInteger outputNumberOfChannels = configuration.outputNumberOfChannels;
+    if (self.outputNumberOfChannels != outputNumberOfChannels) {
+      NSError *outputChannelsError = nil;
+      if (![self setPreferredOutputNumberOfChannels:outputNumberOfChannels
+                                              error:&outputChannelsError]) {
+        RTCLogError(@"Failed to set preferred output number of channels: %@",
+                    outputChannelsError.localizedDescription);
+        if (!self.ignoresPreferredAttributeConfigurationErrors) {
+          error = outputChannelsError;
+        }
+      } else {
+        RTCLog(@"Set output number of channels to: %ld",
+               (long)outputNumberOfChannels);
+      }
+    }
+  }
+
+  if (outError) {
+    *outError = error;
+  }
+
+  return error == nil;
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/components/audio/RTCAudioSession+Private.h b/third_party/libwebrtc/sdk/objc/components/audio/RTCAudioSession+Private.h
new file mode 100644
index 0000000000..2be1b9fb3d
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/components/audio/RTCAudioSession+Private.h
@@ -0,0 +1,95 @@
+/*
+ *  Copyright 2016 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCAudioSession.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+@class RTC_OBJC_TYPE(RTCAudioSessionConfiguration);
+
+@interface RTC_OBJC_TYPE (RTCAudioSession)
+()
+
+    /** Number of times setActive:YES has succeeded without a balanced call to
+     *  setActive:NO.
+     */
+    @property(nonatomic, readonly) int activationCount;
+
+/** The number of times `beginWebRTCSession` was called without a balanced call
+ *  to `endWebRTCSession`.
+ */
+@property(nonatomic, readonly) int webRTCSessionCount;
+
+/** Convenience BOOL that checks useManualAudio and isAudioEnebled. */
+@property(readonly) BOOL canPlayOrRecord;
+
+/** Tracks whether we have been sent an interruption event that hasn't been matched by either an
+ *  interrupted end event or a foreground event.
+ */
+@property(nonatomic, assign) BOOL isInterrupted;
+
+/** Adds the delegate to the list of delegates, and places it at the front of
+ *  the list. This delegate will be notified before other delegates of
+ *  audio events.
+ */
+- (void)pushDelegate:(id<RTC_OBJC_TYPE(RTCAudioSessionDelegate)>)delegate;
+
+/** Signals RTCAudioSession that a WebRTC session is about to begin and
+ *  audio configuration is needed. Will configure the audio session for WebRTC
+ *  if not already configured and if configuration is not delayed.
+ *  Successful calls must be balanced by a call to endWebRTCSession.
+ */
+- (BOOL)beginWebRTCSession:(NSError **)outError;
+
+/** Signals RTCAudioSession that a WebRTC session is about to end and audio
+ *  unconfiguration is needed. Will unconfigure the audio session for WebRTC
+ *  if this is the last unmatched call and if configuration is not delayed.
+ */
+- (BOOL)endWebRTCSession:(NSError **)outError;
+
+/** Configure the audio session for WebRTC. This call will fail if the session
+ *  is already configured. On other failures, we will attempt to restore the
+ *  previously used audio session configuration.
+ *  `lockForConfiguration` must be called first.
+ *  Successful calls to configureWebRTCSession must be matched by calls to
+ *  `unconfigureWebRTCSession`.
+ */
+- (BOOL)configureWebRTCSession:(NSError **)outError;
+
+/** Unconfigures the session for WebRTC. This will attempt to restore the
+ *  audio session to the settings used before `configureWebRTCSession` was
+ *  called.
+ *  `lockForConfiguration` must be called first.
+ */
+- (BOOL)unconfigureWebRTCSession:(NSError **)outError;
+
+/** Returns a configuration error with the given description. */
+- (NSError *)configurationErrorWithDescription:(NSString *)description;
+
+/** Notifies the receiver that a playout glitch was detected. */
+- (void)notifyDidDetectPlayoutGlitch:(int64_t)totalNumberOfGlitches;
+
+/** Notifies the receiver that there was an error when starting an audio unit. */
+- (void)notifyAudioUnitStartFailedWithError:(OSStatus)error;
+
+// Properties and methods for tests.
+- (void)notifyDidBeginInterruption;
+- (void)notifyDidEndInterruptionWithShouldResumeSession:(BOOL)shouldResumeSession;
+- (void)notifyDidChangeRouteWithReason:(AVAudioSessionRouteChangeReason)reason
+                         previousRoute:(AVAudioSessionRouteDescription *)previousRoute;
+- (void)notifyMediaServicesWereLost;
+- (void)notifyMediaServicesWereReset;
+- (void)notifyDidChangeCanPlayOrRecord:(BOOL)canPlayOrRecord;
+- (void)notifyDidStartPlayOrRecord;
+- (void)notifyDidStopPlayOrRecord;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/components/audio/RTCAudioSession.h b/third_party/libwebrtc/sdk/objc/components/audio/RTCAudioSession.h
new file mode 100644
index 0000000000..3b83b27ba5
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/components/audio/RTCAudioSession.h
@@ -0,0 +1,265 @@
+/*
+ *  Copyright 2016 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <AVFoundation/AVFoundation.h>
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+extern NSString *const kRTCAudioSessionErrorDomain;
+/** Method that requires lock was called without lock. */
+extern NSInteger const kRTCAudioSessionErrorLockRequired;
+/** Unknown configuration error occurred. */
+extern NSInteger const kRTCAudioSessionErrorConfiguration;
+
+@class RTC_OBJC_TYPE(RTCAudioSession);
+@class RTC_OBJC_TYPE(RTCAudioSessionConfiguration);
+
+// Surfaces AVAudioSession events. WebRTC will listen directly for notifications
+// from AVAudioSession and handle them before calling these delegate methods,
+// at which point applications can perform additional processing if required.
+RTC_OBJC_EXPORT
+@protocol RTC_OBJC_TYPE
+(RTCAudioSessionDelegate)<NSObject>
+
+    @optional
+/** Called on a system notification thread when AVAudioSession starts an
+ *  interruption event.
+ */
+- (void)audioSessionDidBeginInterruption:(RTC_OBJC_TYPE(RTCAudioSession) *)session;
+
+/** Called on a system notification thread when AVAudioSession ends an
+ *  interruption event.
+ */
+- (void)audioSessionDidEndInterruption:(RTC_OBJC_TYPE(RTCAudioSession) *)session
+                   shouldResumeSession:(BOOL)shouldResumeSession;
+
+/** Called on a system notification thread when AVAudioSession changes the
+ *  route.
+ */
+- (void)audioSessionDidChangeRoute:(RTC_OBJC_TYPE(RTCAudioSession) *)session
+                            reason:(AVAudioSessionRouteChangeReason)reason
+                     previousRoute:(AVAudioSessionRouteDescription *)previousRoute;
+
+/** Called on a system notification thread when AVAudioSession media server
+ *  terminates.
+ */
+- (void)audioSessionMediaServerTerminated:(RTC_OBJC_TYPE(RTCAudioSession) *)session;
+
+/** Called on a system notification thread when AVAudioSession media server
+ *  restarts.
+ */
+- (void)audioSessionMediaServerReset:(RTC_OBJC_TYPE(RTCAudioSession) *)session;
+
+// TODO(tkchin): Maybe handle SilenceSecondaryAudioHintNotification.
+
+- (void)audioSession:(RTC_OBJC_TYPE(RTCAudioSession) *)session
+    didChangeCanPlayOrRecord:(BOOL)canPlayOrRecord;
+
+/** Called on a WebRTC thread when the audio device is notified to begin
+ *  playback or recording.
+ */
+- (void)audioSessionDidStartPlayOrRecord:(RTC_OBJC_TYPE(RTCAudioSession) *)session;
+
+/** Called on a WebRTC thread when the audio device is notified to stop
+ *  playback or recording.
+ */
+- (void)audioSessionDidStopPlayOrRecord:(RTC_OBJC_TYPE(RTCAudioSession) *)session;
+
+/** Called when the AVAudioSession output volume value changes. */
+- (void)audioSession:(RTC_OBJC_TYPE(RTCAudioSession) *)audioSession
+    didChangeOutputVolume:(float)outputVolume;
+
+/** Called when the audio device detects a playout glitch. The argument is the
+ *  number of glitches detected so far in the current audio playout session.
+ */
+- (void)audioSession:(RTC_OBJC_TYPE(RTCAudioSession) *)audioSession
+    didDetectPlayoutGlitch:(int64_t)totalNumberOfGlitches;
+
+/** Called when the audio session is about to change the active state.
+ */
+- (void)audioSession:(RTC_OBJC_TYPE(RTCAudioSession) *)audioSession willSetActive:(BOOL)active;
+
+/** Called after the audio session sucessfully changed the active state.
+ */
+- (void)audioSession:(RTC_OBJC_TYPE(RTCAudioSession) *)audioSession didSetActive:(BOOL)active;
+
+/** Called after the audio session failed to change the active state.
+ */
+- (void)audioSession:(RTC_OBJC_TYPE(RTCAudioSession) *)audioSession
+    failedToSetActive:(BOOL)active
+                error:(NSError *)error;
+
+- (void)audioSession:(RTC_OBJC_TYPE(RTCAudioSession) *)audioSession
+    audioUnitStartFailedWithError:(NSError *)error;
+
+@end
+
+/** This is a protocol used to inform RTCAudioSession when the audio session
+ *  activation state has changed outside of RTCAudioSession. The current known use
+ *  case of this is when CallKit activates the audio session for the application
+ */
+RTC_OBJC_EXPORT
+@protocol RTC_OBJC_TYPE
+(RTCAudioSessionActivationDelegate)<NSObject>
+
+    /** Called when the audio session is activated outside of the app by iOS. */
+    - (void)audioSessionDidActivate : (AVAudioSession *)session;
+
+/** Called when the audio session is deactivated outside of the app by iOS. */
+- (void)audioSessionDidDeactivate:(AVAudioSession *)session;
+
+@end
+
+/** Proxy class for AVAudioSession that adds a locking mechanism similar to
+ *  AVCaptureDevice. This is used to that interleaving configurations between
+ *  WebRTC and the application layer are avoided.
+ *
+ *  RTCAudioSession also coordinates activation so that the audio session is
+ *  activated only once. See `setActive:error:`.
+ */
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCAudioSession) : NSObject <RTC_OBJC_TYPE(RTCAudioSessionActivationDelegate)>
+
+/** Convenience property to access the AVAudioSession singleton. Callers should
+ *  not call setters on AVAudioSession directly, but other method invocations
+ *  are fine.
+ */
+@property(nonatomic, readonly) AVAudioSession *session;
+
+/** Our best guess at whether the session is active based on results of calls to
+ *  AVAudioSession.
+ */
+@property(nonatomic, readonly) BOOL isActive;
+
+/** If YES, WebRTC will not initialize the audio unit automatically when an
+ *  audio track is ready for playout or recording. Instead, applications should
+ *  call setIsAudioEnabled. If NO, WebRTC will initialize the audio unit
+ *  as soon as an audio track is ready for playout or recording.
+ */
+@property(nonatomic, assign) BOOL useManualAudio;
+
+/** This property is only effective if useManualAudio is YES.
+ *  Represents permission for WebRTC to initialize the VoIP audio unit.
+ *  When set to NO, if the VoIP audio unit used by WebRTC is active, it will be
+ *  stopped and uninitialized. This will stop incoming and outgoing audio.
+ *  When set to YES, WebRTC will initialize and start the audio unit when it is
+ *  needed (e.g. due to establishing an audio connection).
+ *  This property was introduced to work around an issue where if an AVPlayer is
+ *  playing audio while the VoIP audio unit is initialized, its audio would be
+ *  either cut off completely or played at a reduced volume. By preventing
+ *  the audio unit from being initialized until after the audio has completed,
+ *  we are able to prevent the abrupt cutoff.
+ */
+@property(nonatomic, assign) BOOL isAudioEnabled;
+
+// Proxy properties.
+@property(readonly) NSString *category;
+@property(readonly) AVAudioSessionCategoryOptions categoryOptions;
+@property(readonly) NSString *mode;
+@property(readonly) BOOL secondaryAudioShouldBeSilencedHint;
+@property(readonly) AVAudioSessionRouteDescription *currentRoute;
+@property(readonly) NSInteger maximumInputNumberOfChannels;
+@property(readonly) NSInteger maximumOutputNumberOfChannels;
+@property(readonly) float inputGain;
+@property(readonly) BOOL inputGainSettable;
+@property(readonly) BOOL inputAvailable;
+@property(readonly, nullable) NSArray<AVAudioSessionDataSourceDescription *> *inputDataSources;
+@property(readonly, nullable) AVAudioSessionDataSourceDescription *inputDataSource;
+@property(readonly, nullable) NSArray<AVAudioSessionDataSourceDescription *> *outputDataSources;
+@property(readonly, nullable) AVAudioSessionDataSourceDescription *outputDataSource;
+@property(readonly) double sampleRate;
+@property(readonly) double preferredSampleRate;
+@property(readonly) NSInteger inputNumberOfChannels;
+@property(readonly) NSInteger outputNumberOfChannels;
+@property(readonly) float outputVolume;
+@property(readonly) NSTimeInterval inputLatency;
+@property(readonly) NSTimeInterval outputLatency;
+@property(readonly) NSTimeInterval IOBufferDuration;
+@property(readonly) NSTimeInterval preferredIOBufferDuration;
+
+/**
+ When YES, calls to -setConfiguration:error: and -setConfiguration:active:error: ignore errors in
+ configuring the audio session's "preferred" attributes (e.g. preferredInputNumberOfChannels).
+ Typically, configurations to preferred attributes are optimizations, and ignoring this type of
+ configuration error allows code flow to continue along the happy path when these optimization are
+ not available. The default value of this property is NO.
+ */
+@property(nonatomic) BOOL ignoresPreferredAttributeConfigurationErrors;
+
+/** Default constructor. */
++ (instancetype)sharedInstance;
+- (instancetype)init NS_UNAVAILABLE;
+
+/** Adds a delegate, which is held weakly. */
+- (void)addDelegate:(id<RTC_OBJC_TYPE(RTCAudioSessionDelegate)>)delegate;
+/** Removes an added delegate. */
+- (void)removeDelegate:(id<RTC_OBJC_TYPE(RTCAudioSessionDelegate)>)delegate;
+
+/** Request exclusive access to the audio session for configuration. This call
+ *  will block if the lock is held by another object.
+ */
+- (void)lockForConfiguration;
+/** Relinquishes exclusive access to the audio session. */
+- (void)unlockForConfiguration;
+
+/** If `active`, activates the audio session if it isn't already active.
+ *  Successful calls must be balanced with a setActive:NO when activation is no
+ *  longer required. If not `active`, deactivates the audio session if one is
+ *  active and this is the last balanced call. When deactivating, the
+ *  AVAudioSessionSetActiveOptionNotifyOthersOnDeactivation option is passed to
+ *  AVAudioSession.
+ */
+- (BOOL)setActive:(BOOL)active error:(NSError **)outError;
+
+// The following methods are proxies for the associated methods on
+// AVAudioSession. `lockForConfiguration` must be called before using them
+// otherwise they will fail with kRTCAudioSessionErrorLockRequired.
+
+- (BOOL)setCategory:(NSString *)category
+        withOptions:(AVAudioSessionCategoryOptions)options
+              error:(NSError **)outError;
+- (BOOL)setMode:(NSString *)mode error:(NSError **)outError;
+- (BOOL)setInputGain:(float)gain error:(NSError **)outError;
+- (BOOL)setPreferredSampleRate:(double)sampleRate error:(NSError **)outError;
+- (BOOL)setPreferredIOBufferDuration:(NSTimeInterval)duration error:(NSError **)outError;
+- (BOOL)setPreferredInputNumberOfChannels:(NSInteger)count error:(NSError **)outError;
+- (BOOL)setPreferredOutputNumberOfChannels:(NSInteger)count error:(NSError **)outError;
+- (BOOL)overrideOutputAudioPort:(AVAudioSessionPortOverride)portOverride error:(NSError **)outError;
+- (BOOL)setPreferredInput:(AVAudioSessionPortDescription *)inPort error:(NSError **)outError;
+- (BOOL)setInputDataSource:(AVAudioSessionDataSourceDescription *)dataSource
+                     error:(NSError **)outError;
+- (BOOL)setOutputDataSource:(AVAudioSessionDataSourceDescription *)dataSource
+                      error:(NSError **)outError;
+@end
+
+@interface RTC_OBJC_TYPE (RTCAudioSession)
+(Configuration)
+
+    /** Applies the configuration to the current session. Attempts to set all
+     *  properties even if previous ones fail. Only the last error will be
+     *  returned.
+     *  `lockForConfiguration` must be called first.
+     */
+    - (BOOL)setConfiguration : (RTC_OBJC_TYPE(RTCAudioSessionConfiguration) *)configuration error
+    : (NSError **)outError;
+
+/** Convenience method that calls both setConfiguration and setActive.
+ *  `lockForConfiguration` must be called first.
+ */
+- (BOOL)setConfiguration:(RTC_OBJC_TYPE(RTCAudioSessionConfiguration) *)configuration
+                  active:(BOOL)active
+                   error:(NSError **)outError;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/components/audio/RTCAudioSession.mm b/third_party/libwebrtc/sdk/objc/components/audio/RTCAudioSession.mm
new file mode 100644
index 0000000000..550a426d36
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/components/audio/RTCAudioSession.mm
@@ -0,0 +1,1000 @@
+/*
+ *  Copyright 2016 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCAudioSession+Private.h"
+
+#import <UIKit/UIKit.h>
+
+#include <atomic>
+#include <vector>
+
+#include "absl/base/attributes.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/synchronization/mutex.h"
+
+#import "RTCAudioSessionConfiguration.h"
+#import "base/RTCLogging.h"
+
+#if !defined(ABSL_HAVE_THREAD_LOCAL)
+#error ABSL_HAVE_THREAD_LOCAL should be defined for MacOS / iOS Targets.
+#endif
+
+NSString *const kRTCAudioSessionErrorDomain = @"org.webrtc.RTC_OBJC_TYPE(RTCAudioSession)";
+NSInteger const kRTCAudioSessionErrorLockRequired = -1;
+NSInteger const kRTCAudioSessionErrorConfiguration = -2;
+NSString * const kRTCAudioSessionOutputVolumeSelector = @"outputVolume";
+
+namespace {
+// Since webrtc::Mutex is not a reentrant lock and cannot check if the mutex is locked,
+// we need a separate variable to check that the mutex is locked in the RTCAudioSession.
+ABSL_CONST_INIT thread_local bool mutex_locked = false;
+}  // namespace
+
+@interface RTC_OBJC_TYPE (RTCAudioSession)
+() @property(nonatomic,
+             readonly) std::vector<__weak id<RTC_OBJC_TYPE(RTCAudioSessionDelegate)> > delegates;
+@end
+
+// This class needs to be thread-safe because it is accessed from many threads.
+// TODO(tkchin): Consider more granular locking. We're not expecting a lot of
+// lock contention so coarse locks should be fine for now.
+@implementation RTC_OBJC_TYPE (RTCAudioSession) {
+  webrtc::Mutex _mutex;
+  AVAudioSession *_session;
+  std::atomic<int> _activationCount;
+  std::atomic<int> _webRTCSessionCount;
+  BOOL _isActive;
+  BOOL _useManualAudio;
+  BOOL _isAudioEnabled;
+  BOOL _canPlayOrRecord;
+  BOOL _isInterrupted;
+}
+
+@synthesize session = _session;
+@synthesize delegates = _delegates;
+@synthesize ignoresPreferredAttributeConfigurationErrors =
+    _ignoresPreferredAttributeConfigurationErrors;
+
++ (instancetype)sharedInstance {
+  static dispatch_once_t onceToken;
+  static RTC_OBJC_TYPE(RTCAudioSession) *sharedInstance = nil;
+  dispatch_once(&onceToken, ^{
+    sharedInstance = [[self alloc] init];
+  });
+  return sharedInstance;
+}
+
+- (instancetype)init {
+  return [self initWithAudioSession:[AVAudioSession sharedInstance]];
+}
+
+/** This initializer provides a way for unit tests to inject a fake/mock audio session. */
+- (instancetype)initWithAudioSession:(id)audioSession {
+  if (self = [super init]) {
+    _session = audioSession;
+
+    NSNotificationCenter *center = [NSNotificationCenter defaultCenter];
+    [center addObserver:self
+               selector:@selector(handleInterruptionNotification:)
+                   name:AVAudioSessionInterruptionNotification
+                 object:nil];
+    [center addObserver:self
+               selector:@selector(handleRouteChangeNotification:)
+                   name:AVAudioSessionRouteChangeNotification
+                 object:nil];
+    [center addObserver:self
+               selector:@selector(handleMediaServicesWereLost:)
+                   name:AVAudioSessionMediaServicesWereLostNotification
+                 object:nil];
+    [center addObserver:self
+               selector:@selector(handleMediaServicesWereReset:)
+                   name:AVAudioSessionMediaServicesWereResetNotification
+                 object:nil];
+    // Posted on the main thread when the primary audio from other applications
+    // starts and stops. Foreground applications may use this notification as a
+    // hint to enable or disable audio that is secondary.
+    [center addObserver:self
+               selector:@selector(handleSilenceSecondaryAudioHintNotification:)
+                   name:AVAudioSessionSilenceSecondaryAudioHintNotification
+                 object:nil];
+    // Also track foreground event in order to deal with interruption ended situation.
+    [center addObserver:self
+               selector:@selector(handleApplicationDidBecomeActive:)
+                   name:UIApplicationDidBecomeActiveNotification
+                 object:nil];
+    [_session addObserver:self
+               forKeyPath:kRTCAudioSessionOutputVolumeSelector
+                  options:NSKeyValueObservingOptionNew | NSKeyValueObservingOptionOld
+                  context:(__bridge void *)RTC_OBJC_TYPE(RTCAudioSession).class];
+
+    RTCLog(@"RTC_OBJC_TYPE(RTCAudioSession) (%p): init.", self);
+  }
+  return self;
+}
+
+- (void)dealloc {
+  [[NSNotificationCenter defaultCenter] removeObserver:self];
+  [_session removeObserver:self
+                forKeyPath:kRTCAudioSessionOutputVolumeSelector
+                   context:(__bridge void *)RTC_OBJC_TYPE(RTCAudioSession).class];
+  RTCLog(@"RTC_OBJC_TYPE(RTCAudioSession) (%p): dealloc.", self);
+}
+
+- (NSString *)description {
+  NSString *format = @"RTC_OBJC_TYPE(RTCAudioSession): {\n"
+                      "  category: %@\n"
+                      "  categoryOptions: %ld\n"
+                      "  mode: %@\n"
+                      "  isActive: %d\n"
+                      "  sampleRate: %.2f\n"
+                      "  IOBufferDuration: %f\n"
+                      "  outputNumberOfChannels: %ld\n"
+                      "  inputNumberOfChannels: %ld\n"
+                      "  outputLatency: %f\n"
+                      "  inputLatency: %f\n"
+                      "  outputVolume: %f\n"
+                      "}";
+  NSString *description = [NSString stringWithFormat:format,
+      self.category, (long)self.categoryOptions, self.mode,
+      self.isActive, self.sampleRate, self.IOBufferDuration,
+      self.outputNumberOfChannels, self.inputNumberOfChannels,
+      self.outputLatency, self.inputLatency, self.outputVolume];
+  return description;
+}
+
+- (void)setIsActive:(BOOL)isActive {
+  @synchronized(self) {
+    _isActive = isActive;
+  }
+}
+
+- (BOOL)isActive {
+  @synchronized(self) {
+    return _isActive;
+  }
+}
+
+- (void)setUseManualAudio:(BOOL)useManualAudio {
+  @synchronized(self) {
+    if (_useManualAudio == useManualAudio) {
+      return;
+    }
+    _useManualAudio = useManualAudio;
+  }
+  [self updateCanPlayOrRecord];
+}
+
+- (BOOL)useManualAudio {
+  @synchronized(self) {
+    return _useManualAudio;
+  }
+}
+
+- (void)setIsAudioEnabled:(BOOL)isAudioEnabled {
+  @synchronized(self) {
+    if (_isAudioEnabled == isAudioEnabled) {
+      return;
+    }
+    _isAudioEnabled = isAudioEnabled;
+  }
+  [self updateCanPlayOrRecord];
+}
+
+- (BOOL)isAudioEnabled {
+  @synchronized(self) {
+    return _isAudioEnabled;
+  }
+}
+
+- (void)setIgnoresPreferredAttributeConfigurationErrors:
+    (BOOL)ignoresPreferredAttributeConfigurationErrors {
+  @synchronized(self) {
+    if (_ignoresPreferredAttributeConfigurationErrors ==
+        ignoresPreferredAttributeConfigurationErrors) {
+      return;
+    }
+    _ignoresPreferredAttributeConfigurationErrors = ignoresPreferredAttributeConfigurationErrors;
+  }
+}
+
+- (BOOL)ignoresPreferredAttributeConfigurationErrors {
+  @synchronized(self) {
+    return _ignoresPreferredAttributeConfigurationErrors;
+  }
+}
+
+// TODO(tkchin): Check for duplicates.
+- (void)addDelegate:(id<RTC_OBJC_TYPE(RTCAudioSessionDelegate)>)delegate {
+  RTCLog(@"Adding delegate: (%p)", delegate);
+  if (!delegate) {
+    return;
+  }
+  @synchronized(self) {
+    _delegates.push_back(delegate);
+    [self removeZeroedDelegates];
+  }
+}
+
+- (void)removeDelegate:(id<RTC_OBJC_TYPE(RTCAudioSessionDelegate)>)delegate {
+  RTCLog(@"Removing delegate: (%p)", delegate);
+  if (!delegate) {
+    return;
+  }
+  @synchronized(self) {
+    _delegates.erase(std::remove(_delegates.begin(),
+                                 _delegates.end(),
+                                 delegate),
+                     _delegates.end());
+    [self removeZeroedDelegates];
+  }
+}
+
+#pragma clang diagnostic push
+#pragma clang diagnostic ignored "-Wthread-safety-analysis"
+
+- (void)lockForConfiguration {
+  RTC_CHECK(!mutex_locked);
+  _mutex.Lock();
+  mutex_locked = true;
+}
+
+- (void)unlockForConfiguration {
+  mutex_locked = false;
+  _mutex.Unlock();
+}
+
+#pragma clang diagnostic pop
+
+#pragma mark - AVAudioSession proxy methods
+
+- (NSString *)category {
+  return self.session.category;
+}
+
+- (AVAudioSessionCategoryOptions)categoryOptions {
+  return self.session.categoryOptions;
+}
+
+- (NSString *)mode {
+  return self.session.mode;
+}
+
+- (BOOL)secondaryAudioShouldBeSilencedHint {
+  return self.session.secondaryAudioShouldBeSilencedHint;
+}
+
+- (AVAudioSessionRouteDescription *)currentRoute {
+  return self.session.currentRoute;
+}
+
+- (NSInteger)maximumInputNumberOfChannels {
+  return self.session.maximumInputNumberOfChannels;
+}
+
+- (NSInteger)maximumOutputNumberOfChannels {
+  return self.session.maximumOutputNumberOfChannels;
+}
+
+- (float)inputGain {
+  return self.session.inputGain;
+}
+
+- (BOOL)inputGainSettable {
+  return self.session.inputGainSettable;
+}
+
+- (BOOL)inputAvailable {
+  return self.session.inputAvailable;
+}
+
+- (NSArray<AVAudioSessionDataSourceDescription *> *)inputDataSources {
+  return self.session.inputDataSources;
+}
+
+- (AVAudioSessionDataSourceDescription *)inputDataSource {
+  return self.session.inputDataSource;
+}
+
+- (NSArray<AVAudioSessionDataSourceDescription *> *)outputDataSources {
+  return self.session.outputDataSources;
+}
+
+- (AVAudioSessionDataSourceDescription *)outputDataSource {
+  return self.session.outputDataSource;
+}
+
+- (double)sampleRate {
+  return self.session.sampleRate;
+}
+
+- (double)preferredSampleRate {
+  return self.session.preferredSampleRate;
+}
+
+- (NSInteger)inputNumberOfChannels {
+  return self.session.inputNumberOfChannels;
+}
+
+- (NSInteger)outputNumberOfChannels {
+  return self.session.outputNumberOfChannels;
+}
+
+- (float)outputVolume {
+  return self.session.outputVolume;
+}
+
+- (NSTimeInterval)inputLatency {
+  return self.session.inputLatency;
+}
+
+- (NSTimeInterval)outputLatency {
+  return self.session.outputLatency;
+}
+
+- (NSTimeInterval)IOBufferDuration {
+  return self.session.IOBufferDuration;
+}
+
+- (NSTimeInterval)preferredIOBufferDuration {
+  return self.session.preferredIOBufferDuration;
+}
+
+- (BOOL)setActive:(BOOL)active
+            error:(NSError **)outError {
+  if (![self checkLock:outError]) {
+    return NO;
+  }
+  int activationCount = _activationCount.load();
+  if (!active && activationCount == 0) {
+    RTCLogWarning(@"Attempting to deactivate without prior activation.");
+  }
+  [self notifyWillSetActive:active];
+  BOOL success = YES;
+  BOOL isActive = self.isActive;
+  // Keep a local error so we can log it.
+  NSError *error = nil;
+  BOOL shouldSetActive =
+      (active && !isActive) || (!active && isActive && activationCount == 1);
+  // Attempt to activate if we're not active.
+  // Attempt to deactivate if we're active and it's the last unbalanced call.
+  if (shouldSetActive) {
+    AVAudioSession *session = self.session;
+    // AVAudioSessionSetActiveOptionNotifyOthersOnDeactivation is used to ensure
+    // that other audio sessions that were interrupted by our session can return
+    // to their active state. It is recommended for VoIP apps to use this
+    // option.
+    AVAudioSessionSetActiveOptions options =
+        active ? 0 : AVAudioSessionSetActiveOptionNotifyOthersOnDeactivation;
+    success = [session setActive:active
+                     withOptions:options
+                           error:&error];
+    if (outError) {
+      *outError = error;
+    }
+  }
+  if (success) {
+    if (active) {
+      if (shouldSetActive) {
+        self.isActive = active;
+        if (self.isInterrupted) {
+          self.isInterrupted = NO;
+          [self notifyDidEndInterruptionWithShouldResumeSession:YES];
+        }
+      }
+      [self incrementActivationCount];
+      [self notifyDidSetActive:active];
+    }
+  } else {
+    RTCLogError(@"Failed to setActive:%d. Error: %@",
+                active, error.localizedDescription);
+    [self notifyFailedToSetActive:active error:error];
+  }
+  // Set isActive and decrement activation count on deactivation
+  // whether or not it succeeded.
+  if (!active) {
+    self.isActive = active;
+    [self notifyDidSetActive:active];
+    [self decrementActivationCount];
+  }
+  RTCLog(@"Number of current activations: %d", _activationCount.load());
+  return success;
+}
+
+- (BOOL)setCategory:(NSString *)category
+        withOptions:(AVAudioSessionCategoryOptions)options
+              error:(NSError **)outError {
+  if (![self checkLock:outError]) {
+    return NO;
+  }
+  return [self.session setCategory:category withOptions:options error:outError];
+}
+
+- (BOOL)setMode:(NSString *)mode error:(NSError **)outError {
+  if (![self checkLock:outError]) {
+    return NO;
+  }
+  return [self.session setMode:mode error:outError];
+}
+
+- (BOOL)setInputGain:(float)gain error:(NSError **)outError {
+  if (![self checkLock:outError]) {
+    return NO;
+  }
+  return [self.session setInputGain:gain error:outError];
+}
+
+- (BOOL)setPreferredSampleRate:(double)sampleRate error:(NSError **)outError {
+  if (![self checkLock:outError]) {
+    return NO;
+  }
+  return [self.session setPreferredSampleRate:sampleRate error:outError];
+}
+
+- (BOOL)setPreferredIOBufferDuration:(NSTimeInterval)duration
+                               error:(NSError **)outError {
+  if (![self checkLock:outError]) {
+    return NO;
+  }
+  return [self.session setPreferredIOBufferDuration:duration error:outError];
+}
+
+- (BOOL)setPreferredInputNumberOfChannels:(NSInteger)count
+                                    error:(NSError **)outError {
+  if (![self checkLock:outError]) {
+    return NO;
+  }
+  return [self.session setPreferredInputNumberOfChannels:count error:outError];
+}
+- (BOOL)setPreferredOutputNumberOfChannels:(NSInteger)count
+                                     error:(NSError **)outError {
+  if (![self checkLock:outError]) {
+    return NO;
+  }
+  return [self.session setPreferredOutputNumberOfChannels:count error:outError];
+}
+
+- (BOOL)overrideOutputAudioPort:(AVAudioSessionPortOverride)portOverride
+                          error:(NSError **)outError {
+  if (![self checkLock:outError]) {
+    return NO;
+  }
+  return [self.session overrideOutputAudioPort:portOverride error:outError];
+}
+
+- (BOOL)setPreferredInput:(AVAudioSessionPortDescription *)inPort
+                    error:(NSError **)outError {
+  if (![self checkLock:outError]) {
+    return NO;
+  }
+  return [self.session setPreferredInput:inPort error:outError];
+}
+
+- (BOOL)setInputDataSource:(AVAudioSessionDataSourceDescription *)dataSource
+                     error:(NSError **)outError {
+  if (![self checkLock:outError]) {
+    return NO;
+  }
+  return [self.session setInputDataSource:dataSource error:outError];
+}
+
+- (BOOL)setOutputDataSource:(AVAudioSessionDataSourceDescription *)dataSource
+                      error:(NSError **)outError {
+  if (![self checkLock:outError]) {
+    return NO;
+  }
+  return [self.session setOutputDataSource:dataSource error:outError];
+}
+
+#pragma mark - Notifications
+
+- (void)handleInterruptionNotification:(NSNotification *)notification {
+  NSNumber* typeNumber =
+      notification.userInfo[AVAudioSessionInterruptionTypeKey];
+  AVAudioSessionInterruptionType type =
+      (AVAudioSessionInterruptionType)typeNumber.unsignedIntegerValue;
+  switch (type) {
+    case AVAudioSessionInterruptionTypeBegan:
+      RTCLog(@"Audio session interruption began.");
+      self.isActive = NO;
+      self.isInterrupted = YES;
+      [self notifyDidBeginInterruption];
+      break;
+    case AVAudioSessionInterruptionTypeEnded: {
+      RTCLog(@"Audio session interruption ended.");
+      self.isInterrupted = NO;
+      [self updateAudioSessionAfterEvent];
+      NSNumber *optionsNumber =
+          notification.userInfo[AVAudioSessionInterruptionOptionKey];
+      AVAudioSessionInterruptionOptions options =
+          optionsNumber.unsignedIntegerValue;
+      BOOL shouldResume =
+          options & AVAudioSessionInterruptionOptionShouldResume;
+      [self notifyDidEndInterruptionWithShouldResumeSession:shouldResume];
+      break;
+    }
+  }
+}
+
+- (void)handleRouteChangeNotification:(NSNotification *)notification {
+  // Get reason for current route change.
+  NSNumber* reasonNumber =
+      notification.userInfo[AVAudioSessionRouteChangeReasonKey];
+  AVAudioSessionRouteChangeReason reason =
+      (AVAudioSessionRouteChangeReason)reasonNumber.unsignedIntegerValue;
+  RTCLog(@"Audio route changed:");
+  switch (reason) {
+    case AVAudioSessionRouteChangeReasonUnknown:
+      RTCLog(@"Audio route changed: ReasonUnknown");
+      break;
+    case AVAudioSessionRouteChangeReasonNewDeviceAvailable:
+      RTCLog(@"Audio route changed: NewDeviceAvailable");
+      break;
+    case AVAudioSessionRouteChangeReasonOldDeviceUnavailable:
+      RTCLog(@"Audio route changed: OldDeviceUnavailable");
+      break;
+    case AVAudioSessionRouteChangeReasonCategoryChange:
+      RTCLog(@"Audio route changed: CategoryChange to :%@",
+             self.session.category);
+      break;
+    case AVAudioSessionRouteChangeReasonOverride:
+      RTCLog(@"Audio route changed: Override");
+      break;
+    case AVAudioSessionRouteChangeReasonWakeFromSleep:
+      RTCLog(@"Audio route changed: WakeFromSleep");
+      break;
+    case AVAudioSessionRouteChangeReasonNoSuitableRouteForCategory:
+      RTCLog(@"Audio route changed: NoSuitableRouteForCategory");
+      break;
+    case AVAudioSessionRouteChangeReasonRouteConfigurationChange:
+      RTCLog(@"Audio route changed: RouteConfigurationChange");
+      break;
+  }
+  AVAudioSessionRouteDescription* previousRoute =
+      notification.userInfo[AVAudioSessionRouteChangePreviousRouteKey];
+  // Log previous route configuration.
+  RTCLog(@"Previous route: %@\nCurrent route:%@",
+         previousRoute, self.session.currentRoute);
+  [self notifyDidChangeRouteWithReason:reason previousRoute:previousRoute];
+}
+
+- (void)handleMediaServicesWereLost:(NSNotification *)notification {
+  RTCLog(@"Media services were lost.");
+  [self updateAudioSessionAfterEvent];
+  [self notifyMediaServicesWereLost];
+}
+
+- (void)handleMediaServicesWereReset:(NSNotification *)notification {
+  RTCLog(@"Media services were reset.");
+  [self updateAudioSessionAfterEvent];
+  [self notifyMediaServicesWereReset];
+}
+
+- (void)handleSilenceSecondaryAudioHintNotification:(NSNotification *)notification {
+  // TODO(henrika): just adding logs here for now until we know if we are ever
+  // see this notification and might be affected by it or if further actions
+  // are required.
+  NSNumber *typeNumber =
+      notification.userInfo[AVAudioSessionSilenceSecondaryAudioHintTypeKey];
+  AVAudioSessionSilenceSecondaryAudioHintType type =
+      (AVAudioSessionSilenceSecondaryAudioHintType)typeNumber.unsignedIntegerValue;
+  switch (type) {
+    case AVAudioSessionSilenceSecondaryAudioHintTypeBegin:
+      RTCLog(@"Another application's primary audio has started.");
+      break;
+    case AVAudioSessionSilenceSecondaryAudioHintTypeEnd:
+      RTCLog(@"Another application's primary audio has stopped.");
+      break;
+  }
+}
+
+- (void)handleApplicationDidBecomeActive:(NSNotification *)notification {
+  BOOL isInterrupted = self.isInterrupted;
+  RTCLog(@"Application became active after an interruption. Treating as interruption "
+          "end. isInterrupted changed from %d to 0.",
+         isInterrupted);
+  if (isInterrupted) {
+    self.isInterrupted = NO;
+    [self updateAudioSessionAfterEvent];
+  }
+  // Always treat application becoming active as an interruption end event.
+  [self notifyDidEndInterruptionWithShouldResumeSession:YES];
+}
+
+#pragma mark - Private
+
++ (NSError *)lockError {
+  NSDictionary *userInfo =
+      @{NSLocalizedDescriptionKey : @"Must call lockForConfiguration before calling this method."};
+  NSError *error = [[NSError alloc] initWithDomain:kRTCAudioSessionErrorDomain
+                                              code:kRTCAudioSessionErrorLockRequired
+                                          userInfo:userInfo];
+  return error;
+}
+
+- (std::vector<__weak id<RTC_OBJC_TYPE(RTCAudioSessionDelegate)> >)delegates {
+  @synchronized(self) {
+    // Note: this returns a copy.
+    return _delegates;
+  }
+}
+
+// TODO(tkchin): check for duplicates.
+- (void)pushDelegate:(id<RTC_OBJC_TYPE(RTCAudioSessionDelegate)>)delegate {
+  @synchronized(self) {
+    _delegates.insert(_delegates.begin(), delegate);
+  }
+}
+
+- (void)removeZeroedDelegates {
+  @synchronized(self) {
+    _delegates.erase(
+        std::remove_if(_delegates.begin(),
+                       _delegates.end(),
+                       [](id delegate) -> bool { return delegate == nil; }),
+        _delegates.end());
+  }
+}
+
+- (int)activationCount {
+  return _activationCount.load();
+}
+
+- (int)incrementActivationCount {
+  RTCLog(@"Incrementing activation count.");
+  return _activationCount.fetch_add(1) + 1;
+}
+
+- (NSInteger)decrementActivationCount {
+  RTCLog(@"Decrementing activation count.");
+  return _activationCount.fetch_sub(1) - 1;
+}
+
+- (int)webRTCSessionCount {
+  return _webRTCSessionCount.load();
+}
+
+- (BOOL)canPlayOrRecord {
+  return !self.useManualAudio || self.isAudioEnabled;
+}
+
+- (BOOL)isInterrupted {
+  @synchronized(self) {
+    return _isInterrupted;
+  }
+}
+
+- (void)setIsInterrupted:(BOOL)isInterrupted {
+  @synchronized(self) {
+    if (_isInterrupted == isInterrupted) {
+      return;
+   }
+   _isInterrupted = isInterrupted;
+  }
+}
+
+- (BOOL)checkLock:(NSError **)outError {
+  if (!mutex_locked) {
+    if (outError) {
+      *outError = [RTC_OBJC_TYPE(RTCAudioSession) lockError];
+    }
+    return NO;
+  }
+  return YES;
+}
+
+- (BOOL)beginWebRTCSession:(NSError **)outError {
+  if (outError) {
+    *outError = nil;
+  }
+  _webRTCSessionCount.fetch_add(1);
+  [self notifyDidStartPlayOrRecord];
+  return YES;
+}
+
+- (BOOL)endWebRTCSession:(NSError **)outError {
+  if (outError) {
+    *outError = nil;
+  }
+  _webRTCSessionCount.fetch_sub(1);
+  [self notifyDidStopPlayOrRecord];
+  return YES;
+}
+
+- (BOOL)configureWebRTCSession:(NSError **)outError {
+  if (outError) {
+    *outError = nil;
+  }
+  RTCLog(@"Configuring audio session for WebRTC.");
+
+  // Configure the AVAudioSession and activate it.
+  // Provide an error even if there isn't one so we can log it.
+  NSError *error = nil;
+  RTC_OBJC_TYPE(RTCAudioSessionConfiguration) *webRTCConfig =
+      [RTC_OBJC_TYPE(RTCAudioSessionConfiguration) webRTCConfiguration];
+  if (![self setConfiguration:webRTCConfig active:YES error:&error]) {
+    RTCLogError(@"Failed to set WebRTC audio configuration: %@",
+                error.localizedDescription);
+    // Do not call setActive:NO if setActive:YES failed.
+    if (outError) {
+      *outError = error;
+    }
+    return NO;
+  }
+
+  // Ensure that the device currently supports audio input.
+  // TODO(tkchin): Figure out if this is really necessary.
+  if (!self.inputAvailable) {
+    RTCLogError(@"No audio input path is available!");
+    [self unconfigureWebRTCSession:nil];
+    if (outError) {
+      *outError = [self configurationErrorWithDescription:@"No input path."];
+    }
+    return NO;
+  }
+
+  // It can happen (e.g. in combination with BT devices) that the attempt to set
+  // the preferred sample rate for WebRTC (48kHz) fails. If so, make a new
+  // configuration attempt using the sample rate that worked using the active
+  // audio session. A typical case is that only 8 or 16kHz can be set, e.g. in
+  // combination with BT headsets. Using this "trick" seems to avoid a state
+  // where Core Audio asks for a different number of audio frames than what the
+  // session's I/O buffer duration corresponds to.
+  // TODO(henrika): this fix resolves bugs.webrtc.org/6004 but it has only been
+  // tested on a limited set of iOS devices and BT devices.
+  double sessionSampleRate = self.sampleRate;
+  double preferredSampleRate = webRTCConfig.sampleRate;
+  if (sessionSampleRate != preferredSampleRate) {
+    RTCLogWarning(
+        @"Current sample rate (%.2f) is not the preferred rate (%.2f)",
+        sessionSampleRate, preferredSampleRate);
+    if (![self setPreferredSampleRate:sessionSampleRate
+                                error:&error]) {
+      RTCLogError(@"Failed to set preferred sample rate: %@",
+                  error.localizedDescription);
+      if (outError) {
+        *outError = error;
+      }
+    }
+  }
+
+  return YES;
+}
+
+- (BOOL)unconfigureWebRTCSession:(NSError **)outError {
+  if (outError) {
+    *outError = nil;
+  }
+  RTCLog(@"Unconfiguring audio session for WebRTC.");
+  [self setActive:NO error:outError];
+
+  return YES;
+}
+
+- (NSError *)configurationErrorWithDescription:(NSString *)description {
+  NSDictionary* userInfo = @{
+    NSLocalizedDescriptionKey: description,
+  };
+  return [[NSError alloc] initWithDomain:kRTCAudioSessionErrorDomain
+                                    code:kRTCAudioSessionErrorConfiguration
+                                userInfo:userInfo];
+}
+
+- (void)updateAudioSessionAfterEvent {
+  BOOL shouldActivate = self.activationCount > 0;
+  AVAudioSessionSetActiveOptions options = shouldActivate ?
+      0 : AVAudioSessionSetActiveOptionNotifyOthersOnDeactivation;
+  NSError *error = nil;
+  if ([self.session setActive:shouldActivate
+                  withOptions:options
+                        error:&error]) {
+    self.isActive = shouldActivate;
+  } else {
+    RTCLogError(@"Failed to set session active to %d. Error:%@",
+                shouldActivate, error.localizedDescription);
+  }
+}
+
+- (void)updateCanPlayOrRecord {
+  BOOL canPlayOrRecord = NO;
+  BOOL shouldNotify = NO;
+  @synchronized(self) {
+    canPlayOrRecord = !self.useManualAudio || self.isAudioEnabled;
+    if (_canPlayOrRecord == canPlayOrRecord) {
+      return;
+    }
+    _canPlayOrRecord = canPlayOrRecord;
+    shouldNotify = YES;
+  }
+  if (shouldNotify) {
+    [self notifyDidChangeCanPlayOrRecord:canPlayOrRecord];
+  }
+}
+
+- (void)audioSessionDidActivate:(AVAudioSession *)session {
+  if (_session != session) {
+    RTCLogError(@"audioSessionDidActivate called on different AVAudioSession");
+  }
+  RTCLog(@"Audio session was externally activated.");
+  [self incrementActivationCount];
+  self.isActive = YES;
+  // When a CallKit call begins, it's possible that we receive an interruption
+  // begin without a corresponding end. Since we know that we have an activated
+  // audio session at this point, just clear any saved interruption flag since
+  // the app may never be foregrounded during the duration of the call.
+  if (self.isInterrupted) {
+    RTCLog(@"Clearing interrupted state due to external activation.");
+    self.isInterrupted = NO;
+  }
+  // Treat external audio session activation as an end interruption event.
+  [self notifyDidEndInterruptionWithShouldResumeSession:YES];
+}
+
+- (void)audioSessionDidDeactivate:(AVAudioSession *)session {
+  if (_session != session) {
+    RTCLogError(@"audioSessionDidDeactivate called on different AVAudioSession");
+  }
+  RTCLog(@"Audio session was externally deactivated.");
+  self.isActive = NO;
+  [self decrementActivationCount];
+}
+
+- (void)observeValueForKeyPath:(NSString *)keyPath
+                      ofObject:(id)object
+                        change:(NSDictionary *)change
+                       context:(void *)context {
+  if (context == (__bridge void *)RTC_OBJC_TYPE(RTCAudioSession).class) {
+    if (object == _session) {
+      NSNumber *newVolume = change[NSKeyValueChangeNewKey];
+      RTCLog(@"OutputVolumeDidChange to %f", newVolume.floatValue);
+      [self notifyDidChangeOutputVolume:newVolume.floatValue];
+    }
+  } else {
+    [super observeValueForKeyPath:keyPath
+                         ofObject:object
+                           change:change
+                          context:context];
+  }
+}
+
+- (void)notifyAudioUnitStartFailedWithError:(OSStatus)error {
+  for (auto delegate : self.delegates) {
+    SEL sel = @selector(audioSession:audioUnitStartFailedWithError:);
+    if ([delegate respondsToSelector:sel]) {
+      [delegate audioSession:self
+          audioUnitStartFailedWithError:[NSError errorWithDomain:kRTCAudioSessionErrorDomain
+                                                            code:error
+                                                        userInfo:nil]];
+    }
+  }
+}
+
+- (void)notifyDidBeginInterruption {
+  for (auto delegate : self.delegates) {
+    SEL sel = @selector(audioSessionDidBeginInterruption:);
+    if ([delegate respondsToSelector:sel]) {
+      [delegate audioSessionDidBeginInterruption:self];
+    }
+  }
+}
+
+- (void)notifyDidEndInterruptionWithShouldResumeSession:
+    (BOOL)shouldResumeSession {
+  for (auto delegate : self.delegates) {
+    SEL sel = @selector(audioSessionDidEndInterruption:shouldResumeSession:);
+    if ([delegate respondsToSelector:sel]) {
+      [delegate audioSessionDidEndInterruption:self
+                           shouldResumeSession:shouldResumeSession];
+    }
+  }
+}
+
+- (void)notifyDidChangeRouteWithReason:(AVAudioSessionRouteChangeReason)reason
+    previousRoute:(AVAudioSessionRouteDescription *)previousRoute {
+  for (auto delegate : self.delegates) {
+    SEL sel = @selector(audioSessionDidChangeRoute:reason:previousRoute:);
+    if ([delegate respondsToSelector:sel]) {
+      [delegate audioSessionDidChangeRoute:self
+                                    reason:reason
+                             previousRoute:previousRoute];
+    }
+  }
+}
+
+- (void)notifyMediaServicesWereLost {
+  for (auto delegate : self.delegates) {
+    SEL sel = @selector(audioSessionMediaServerTerminated:);
+    if ([delegate respondsToSelector:sel]) {
+      [delegate audioSessionMediaServerTerminated:self];
+    }
+  }
+}
+
+- (void)notifyMediaServicesWereReset {
+  for (auto delegate : self.delegates) {
+    SEL sel = @selector(audioSessionMediaServerReset:);
+    if ([delegate respondsToSelector:sel]) {
+      [delegate audioSessionMediaServerReset:self];
+    }
+  }
+}
+
+- (void)notifyDidChangeCanPlayOrRecord:(BOOL)canPlayOrRecord {
+  for (auto delegate : self.delegates) {
+    SEL sel = @selector(audioSession:didChangeCanPlayOrRecord:);
+    if ([delegate respondsToSelector:sel]) {
+      [delegate audioSession:self didChangeCanPlayOrRecord:canPlayOrRecord];
+    }
+  }
+}
+
+- (void)notifyDidStartPlayOrRecord {
+  for (auto delegate : self.delegates) {
+    SEL sel = @selector(audioSessionDidStartPlayOrRecord:);
+    if ([delegate respondsToSelector:sel]) {
+      [delegate audioSessionDidStartPlayOrRecord:self];
+    }
+  }
+}
+
+- (void)notifyDidStopPlayOrRecord {
+  for (auto delegate : self.delegates) {
+    SEL sel = @selector(audioSessionDidStopPlayOrRecord:);
+    if ([delegate respondsToSelector:sel]) {
+      [delegate audioSessionDidStopPlayOrRecord:self];
+    }
+  }
+}
+
+- (void)notifyDidChangeOutputVolume:(float)volume {
+  for (auto delegate : self.delegates) {
+    SEL sel = @selector(audioSession:didChangeOutputVolume:);
+    if ([delegate respondsToSelector:sel]) {
+      [delegate audioSession:self didChangeOutputVolume:volume];
+    }
+  }
+}
+
+- (void)notifyDidDetectPlayoutGlitch:(int64_t)totalNumberOfGlitches {
+  for (auto delegate : self.delegates) {
+    SEL sel = @selector(audioSession:didDetectPlayoutGlitch:);
+    if ([delegate respondsToSelector:sel]) {
+      [delegate audioSession:self didDetectPlayoutGlitch:totalNumberOfGlitches];
+    }
+  }
+}
+
+- (void)notifyWillSetActive:(BOOL)active {
+  for (id delegate : self.delegates) {
+    SEL sel = @selector(audioSession:willSetActive:);
+    if ([delegate respondsToSelector:sel]) {
+      [delegate audioSession:self willSetActive:active];
+    }
+  }
+}
+
+- (void)notifyDidSetActive:(BOOL)active {
+  for (id delegate : self.delegates) {
+    SEL sel = @selector(audioSession:didSetActive:);
+    if ([delegate respondsToSelector:sel]) {
+      [delegate audioSession:self didSetActive:active];
+    }
+  }
+}
+
+- (void)notifyFailedToSetActive:(BOOL)active error:(NSError *)error {
+  for (id delegate : self.delegates) {
+    SEL sel = @selector(audioSession:failedToSetActive:error:);
+    if ([delegate respondsToSelector:sel]) {
+      [delegate audioSession:self failedToSetActive:active error:error];
+    }
+  }
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/components/audio/RTCAudioSessionConfiguration.h b/third_party/libwebrtc/sdk/objc/components/audio/RTCAudioSessionConfiguration.h
new file mode 100644
index 0000000000..4582b80557
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/components/audio/RTCAudioSessionConfiguration.h
@@ -0,0 +1,48 @@
+/*
+ *  Copyright 2016 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <AVFoundation/AVFoundation.h>
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+RTC_EXTERN const int kRTCAudioSessionPreferredNumberOfChannels;
+RTC_EXTERN const double kRTCAudioSessionHighPerformanceSampleRate;
+RTC_EXTERN const double kRTCAudioSessionLowComplexitySampleRate;
+RTC_EXTERN const double kRTCAudioSessionHighPerformanceIOBufferDuration;
+RTC_EXTERN const double kRTCAudioSessionLowComplexityIOBufferDuration;
+
+// Struct to hold configuration values.
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCAudioSessionConfiguration) : NSObject
+
+@property(nonatomic, strong) NSString *category;
+@property(nonatomic, assign) AVAudioSessionCategoryOptions categoryOptions;
+@property(nonatomic, strong) NSString *mode;
+@property(nonatomic, assign) double sampleRate;
+@property(nonatomic, assign) NSTimeInterval ioBufferDuration;
+@property(nonatomic, assign) NSInteger inputNumberOfChannels;
+@property(nonatomic, assign) NSInteger outputNumberOfChannels;
+
+/** Initializes configuration to defaults. */
+- (instancetype)init NS_DESIGNATED_INITIALIZER;
+
+/** Returns the current configuration of the audio session. */
++ (instancetype)currentConfiguration;
+/** Returns the configuration that WebRTC needs. */
++ (instancetype)webRTCConfiguration;
+/** Provide a way to override the default configuration. */
++ (void)setWebRTCConfiguration:(RTC_OBJC_TYPE(RTCAudioSessionConfiguration) *)configuration;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/components/audio/RTCAudioSessionConfiguration.m b/third_party/libwebrtc/sdk/objc/components/audio/RTCAudioSessionConfiguration.m
new file mode 100644
index 0000000000..39e9ac13ec
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/components/audio/RTCAudioSessionConfiguration.m
@@ -0,0 +1,133 @@
+/*
+ *  Copyright 2016 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCAudioSessionConfiguration.h"
+#import "RTCAudioSession.h"
+
+#import "helpers/RTCDispatcher.h"
+#import "helpers/UIDevice+RTCDevice.h"
+
+// Try to use mono to save resources. Also avoids channel format conversion
+// in the I/O audio unit. Initial tests have shown that it is possible to use
+// mono natively for built-in microphones and for BT headsets but not for
+// wired headsets. Wired headsets only support stereo as native channel format
+// but it is a low cost operation to do a format conversion to mono in the
+// audio unit. Hence, we will not hit a RTC_CHECK in
+// VerifyAudioParametersForActiveAudioSession() for a mismatch between the
+// preferred number of channels and the actual number of channels.
+const int kRTCAudioSessionPreferredNumberOfChannels = 1;
+
+// Preferred hardware sample rate (unit is in Hertz). The client sample rate
+// will be set to this value as well to avoid resampling the the audio unit's
+// format converter. Note that, some devices, e.g. BT headsets, only supports
+// 8000Hz as native sample rate.
+const double kRTCAudioSessionHighPerformanceSampleRate = 48000.0;
+
+// A lower sample rate will be used for devices with only one core
+// (e.g. iPhone 4). The goal is to reduce the CPU load of the application.
+const double kRTCAudioSessionLowComplexitySampleRate = 16000.0;
+
+// Use a hardware I/O buffer size (unit is in seconds) that matches the 10ms
+// size used by WebRTC. The exact actual size will differ between devices.
+// Example: using 48kHz on iPhone 6 results in a native buffer size of
+// ~10.6667ms or 512 audio frames per buffer. The FineAudioBuffer instance will
+// take care of any buffering required to convert between native buffers and
+// buffers used by WebRTC. It is beneficial for the performance if the native
+// size is as an even multiple of 10ms as possible since it results in "clean"
+// callback sequence without bursts of callbacks back to back.
+const double kRTCAudioSessionHighPerformanceIOBufferDuration = 0.02;
+
+// Use a larger buffer size on devices with only one core (e.g. iPhone 4).
+// It will result in a lower CPU consumption at the cost of a larger latency.
+// The size of 60ms is based on instrumentation that shows a significant
+// reduction in CPU load compared with 10ms on low-end devices.
+// TODO(henrika): monitor this size and determine if it should be modified.
+const double kRTCAudioSessionLowComplexityIOBufferDuration = 0.06;
+
+static RTC_OBJC_TYPE(RTCAudioSessionConfiguration) *gWebRTCConfiguration = nil;
+
+@implementation RTC_OBJC_TYPE (RTCAudioSessionConfiguration)
+
+@synthesize category = _category;
+@synthesize categoryOptions = _categoryOptions;
+@synthesize mode = _mode;
+@synthesize sampleRate = _sampleRate;
+@synthesize ioBufferDuration = _ioBufferDuration;
+@synthesize inputNumberOfChannels = _inputNumberOfChannels;
+@synthesize outputNumberOfChannels = _outputNumberOfChannels;
+
+- (instancetype)init {
+  if (self = [super init]) {
+    // Use a category which supports simultaneous recording and playback.
+    // By default, using this category implies that our app’s audio is
+    // nonmixable, hence activating the session will interrupt any other
+    // audio sessions which are also nonmixable.
+    _category = AVAudioSessionCategoryPlayAndRecord;
+    _categoryOptions = AVAudioSessionCategoryOptionAllowBluetooth;
+
+    // Specify mode for two-way voice communication (e.g. VoIP).
+    _mode = AVAudioSessionModeVoiceChat;
+
+    // Set the session's sample rate or the hardware sample rate.
+    // It is essential that we use the same sample rate as stream format
+    // to ensure that the I/O unit does not have to do sample rate conversion.
+    // Set the preferred audio I/O buffer duration, in seconds.
+    NSUInteger processorCount = [NSProcessInfo processInfo].processorCount;
+    // Use best sample rate and buffer duration if the CPU has more than one
+    // core.
+    if (processorCount > 1 && [UIDevice deviceType] != RTCDeviceTypeIPhone4S) {
+      _sampleRate = kRTCAudioSessionHighPerformanceSampleRate;
+      _ioBufferDuration = kRTCAudioSessionHighPerformanceIOBufferDuration;
+    } else {
+      _sampleRate = kRTCAudioSessionLowComplexitySampleRate;
+      _ioBufferDuration = kRTCAudioSessionLowComplexityIOBufferDuration;
+    }
+
+    // We try to use mono in both directions to save resources and format
+    // conversions in the audio unit. Some devices does only support stereo;
+    // e.g. wired headset on iPhone 6.
+    // TODO(henrika): add support for stereo if needed.
+    _inputNumberOfChannels = kRTCAudioSessionPreferredNumberOfChannels;
+    _outputNumberOfChannels = kRTCAudioSessionPreferredNumberOfChannels;
+  }
+  return self;
+}
+
++ (void)initialize {
+  gWebRTCConfiguration = [[self alloc] init];
+}
+
++ (instancetype)currentConfiguration {
+  RTC_OBJC_TYPE(RTCAudioSession) *session = [RTC_OBJC_TYPE(RTCAudioSession) sharedInstance];
+  RTC_OBJC_TYPE(RTCAudioSessionConfiguration) *config =
+      [[RTC_OBJC_TYPE(RTCAudioSessionConfiguration) alloc] init];
+  config.category = session.category;
+  config.categoryOptions = session.categoryOptions;
+  config.mode = session.mode;
+  config.sampleRate = session.sampleRate;
+  config.ioBufferDuration = session.IOBufferDuration;
+  config.inputNumberOfChannels = session.inputNumberOfChannels;
+  config.outputNumberOfChannels = session.outputNumberOfChannels;
+  return config;
+}
+
++ (instancetype)webRTCConfiguration {
+  @synchronized(self) {
+    return (RTC_OBJC_TYPE(RTCAudioSessionConfiguration) *)gWebRTCConfiguration;
+  }
+}
+
++ (void)setWebRTCConfiguration:(RTC_OBJC_TYPE(RTCAudioSessionConfiguration) *)configuration {
+  @synchronized(self) {
+    gWebRTCConfiguration = configuration;
+  }
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/components/audio/RTCNativeAudioSessionDelegateAdapter.h b/third_party/libwebrtc/sdk/objc/components/audio/RTCNativeAudioSessionDelegateAdapter.h
new file mode 100644
index 0000000000..6a75f01479
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/components/audio/RTCNativeAudioSessionDelegateAdapter.h
@@ -0,0 +1,33 @@
+/*
+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCAudioSession.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+namespace webrtc {
+class AudioSessionObserver;
+}
+
+/** Adapter that forwards RTCAudioSessionDelegate calls to the appropriate
+ *  methods on the AudioSessionObserver.
+ */
+@interface RTCNativeAudioSessionDelegateAdapter : NSObject <RTC_OBJC_TYPE (RTCAudioSessionDelegate)>
+
+- (instancetype)init NS_UNAVAILABLE;
+
+/** `observer` is a raw pointer and should be kept alive
+ *  for this object's lifetime.
+ */
+- (instancetype)initWithObserver:(webrtc::AudioSessionObserver *)observer NS_DESIGNATED_INITIALIZER;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/components/audio/RTCNativeAudioSessionDelegateAdapter.mm b/third_party/libwebrtc/sdk/objc/components/audio/RTCNativeAudioSessionDelegateAdapter.mm
new file mode 100644
index 0000000000..daddf314a4
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/components/audio/RTCNativeAudioSessionDelegateAdapter.mm
@@ -0,0 +1,89 @@
+/*
+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCNativeAudioSessionDelegateAdapter.h"
+
+#include "sdk/objc/native/src/audio/audio_session_observer.h"
+
+#import "base/RTCLogging.h"
+
+@implementation RTCNativeAudioSessionDelegateAdapter {
+  webrtc::AudioSessionObserver *_observer;
+}
+
+- (instancetype)initWithObserver:(webrtc::AudioSessionObserver *)observer {
+  RTC_DCHECK(observer);
+  if (self = [super init]) {
+    _observer = observer;
+  }
+  return self;
+}
+
+#pragma mark - RTC_OBJC_TYPE(RTCAudioSessionDelegate)
+
+- (void)audioSessionDidBeginInterruption:(RTC_OBJC_TYPE(RTCAudioSession) *)session {
+  _observer->OnInterruptionBegin();
+}
+
+- (void)audioSessionDidEndInterruption:(RTC_OBJC_TYPE(RTCAudioSession) *)session
+                   shouldResumeSession:(BOOL)shouldResumeSession {
+  _observer->OnInterruptionEnd();
+}
+
+- (void)audioSessionDidChangeRoute:(RTC_OBJC_TYPE(RTCAudioSession) *)session
+                            reason:(AVAudioSessionRouteChangeReason)reason
+                     previousRoute:(AVAudioSessionRouteDescription *)previousRoute {
+  switch (reason) {
+    case AVAudioSessionRouteChangeReasonUnknown:
+    case AVAudioSessionRouteChangeReasonNewDeviceAvailable:
+    case AVAudioSessionRouteChangeReasonOldDeviceUnavailable:
+    case AVAudioSessionRouteChangeReasonCategoryChange:
+      // It turns out that we see a category change (at least in iOS 9.2)
+      // when making a switch from a BT device to e.g. Speaker using the
+      // iOS Control Center and that we therefore must check if the sample
+      // rate has changed. And if so is the case, restart the audio unit.
+    case AVAudioSessionRouteChangeReasonOverride:
+    case AVAudioSessionRouteChangeReasonWakeFromSleep:
+    case AVAudioSessionRouteChangeReasonNoSuitableRouteForCategory:
+      _observer->OnValidRouteChange();
+      break;
+    case AVAudioSessionRouteChangeReasonRouteConfigurationChange:
+      // The set of input and output ports has not changed, but their
+      // configuration has, e.g., a port’s selected data source has
+      // changed. Ignore this type of route change since we are focusing
+      // on detecting headset changes.
+      RTCLog(@"Ignoring RouteConfigurationChange");
+      break;
+  }
+}
+
+- (void)audioSessionMediaServerTerminated:(RTC_OBJC_TYPE(RTCAudioSession) *)session {
+}
+
+- (void)audioSessionMediaServerReset:(RTC_OBJC_TYPE(RTCAudioSession) *)session {
+}
+
+- (void)audioSession:(RTC_OBJC_TYPE(RTCAudioSession) *)session
+    didChangeCanPlayOrRecord:(BOOL)canPlayOrRecord {
+  _observer->OnCanPlayOrRecordChange(canPlayOrRecord);
+}
+
+- (void)audioSessionDidStartPlayOrRecord:(RTC_OBJC_TYPE(RTCAudioSession) *)session {
+}
+
+- (void)audioSessionDidStopPlayOrRecord:(RTC_OBJC_TYPE(RTCAudioSession) *)session {
+}
+
+- (void)audioSession:(RTC_OBJC_TYPE(RTCAudioSession) *)audioSession
+    didChangeOutputVolume:(float)outputVolume {
+  _observer->OnChangedOutputVolume();
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/components/capturer/RTCCameraVideoCapturer.h b/third_party/libwebrtc/sdk/objc/components/capturer/RTCCameraVideoCapturer.h
new file mode 100644
index 0000000000..370bfa70f0
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/components/capturer/RTCCameraVideoCapturer.h
@@ -0,0 +1,56 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <AVFoundation/AVFoundation.h>
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+#import "RTCVideoCapturer.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+RTC_OBJC_EXPORT
+// Camera capture that implements RTCVideoCapturer. Delivers frames to a
+// RTCVideoCapturerDelegate (usually RTCVideoSource).
+NS_EXTENSION_UNAVAILABLE_IOS("Camera not available in app extensions.")
+@interface RTC_OBJC_TYPE (RTCCameraVideoCapturer) : RTC_OBJC_TYPE(RTCVideoCapturer)
+
+// Capture session that is used for capturing. Valid from initialization to dealloc.
+@property(readonly, nonatomic) AVCaptureSession *captureSession;
+
+// Returns list of available capture devices that support video capture.
++ (NSArray<AVCaptureDevice *> *)captureDevices;
+// Returns list of formats that are supported by this class for this device.
++ (NSArray<AVCaptureDeviceFormat *> *)supportedFormatsForDevice:(AVCaptureDevice *)device;
+
+// Returns the most efficient supported output pixel format for this capturer.
+- (FourCharCode)preferredOutputPixelFormat;
+
+// Starts the capture session asynchronously and notifies callback on completion.
+// The device will capture video in the format given in the `format` parameter. If the pixel format
+// in `format` is supported by the WebRTC pipeline, the same pixel format will be used for the
+// output. Otherwise, the format returned by `preferredOutputPixelFormat` will be used.
+- (void)startCaptureWithDevice:(AVCaptureDevice *)device
+                        format:(AVCaptureDeviceFormat *)format
+                           fps:(NSInteger)fps
+             completionHandler:(nullable void (^)(NSError *_Nullable))completionHandler;
+// Stops the capture session asynchronously and notifies callback on completion.
+- (void)stopCaptureWithCompletionHandler:(nullable void (^)(void))completionHandler;
+
+// Starts the capture session asynchronously.
+- (void)startCaptureWithDevice:(AVCaptureDevice *)device
+                        format:(AVCaptureDeviceFormat *)format
+                           fps:(NSInteger)fps;
+// Stops the capture session asynchronously.
+- (void)stopCapture;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/components/capturer/RTCCameraVideoCapturer.m b/third_party/libwebrtc/sdk/objc/components/capturer/RTCCameraVideoCapturer.m
new file mode 100644
index 0000000000..98d3cf9f45
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/components/capturer/RTCCameraVideoCapturer.m
@@ -0,0 +1,535 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCCameraVideoCapturer.h"
+#import "base/RTCLogging.h"
+#import "base/RTCVideoFrameBuffer.h"
+#import "components/video_frame_buffer/RTCCVPixelBuffer.h"
+
+#if TARGET_OS_IPHONE
+#import "helpers/UIDevice+RTCDevice.h"
+#endif
+
+#import "helpers/AVCaptureSession+DevicePosition.h"
+#import "helpers/RTCDispatcher+Private.h"
+#include "rtc_base/system/gcd_helpers.h"
+
+const int64_t kNanosecondsPerSecond = 1000000000;
+
+@interface RTC_OBJC_TYPE (RTCCameraVideoCapturer)
+()<AVCaptureVideoDataOutputSampleBufferDelegate> @property(nonatomic,
+                                                           readonly) dispatch_queue_t frameQueue;
+@property(nonatomic, strong) AVCaptureDevice *currentDevice;
+@property(nonatomic, assign) BOOL hasRetriedOnFatalError;
+@property(nonatomic, assign) BOOL isRunning;
+// Will the session be running once all asynchronous operations have been completed?
+@property(nonatomic, assign) BOOL willBeRunning;
+@end
+
+@implementation RTC_OBJC_TYPE (RTCCameraVideoCapturer) {
+  AVCaptureVideoDataOutput *_videoDataOutput;
+  AVCaptureSession *_captureSession;
+  FourCharCode _preferredOutputPixelFormat;
+  FourCharCode _outputPixelFormat;
+  RTCVideoRotation _rotation;
+#if TARGET_OS_IPHONE
+  UIDeviceOrientation _orientation;
+  BOOL _generatingOrientationNotifications;
+#endif
+}
+
+@synthesize frameQueue = _frameQueue;
+@synthesize captureSession = _captureSession;
+@synthesize currentDevice = _currentDevice;
+@synthesize hasRetriedOnFatalError = _hasRetriedOnFatalError;
+@synthesize isRunning = _isRunning;
+@synthesize willBeRunning = _willBeRunning;
+
+- (instancetype)init {
+  return [self initWithDelegate:nil captureSession:[[AVCaptureSession alloc] init]];
+}
+
+- (instancetype)initWithDelegate:(__weak id<RTC_OBJC_TYPE(RTCVideoCapturerDelegate)>)delegate {
+  return [self initWithDelegate:delegate captureSession:[[AVCaptureSession alloc] init]];
+}
+
+// This initializer is used for testing.
+- (instancetype)initWithDelegate:(__weak id<RTC_OBJC_TYPE(RTCVideoCapturerDelegate)>)delegate
+                  captureSession:(AVCaptureSession *)captureSession {
+  if (self = [super initWithDelegate:delegate]) {
+    // Create the capture session and all relevant inputs and outputs. We need
+    // to do this in init because the application may want the capture session
+    // before we start the capturer for e.g. AVCapturePreviewLayer. All objects
+    // created here are retained until dealloc and never recreated.
+    if (![self setupCaptureSession:captureSession]) {
+      return nil;
+    }
+    NSNotificationCenter *center = [NSNotificationCenter defaultCenter];
+#if TARGET_OS_IPHONE
+    _orientation = UIDeviceOrientationPortrait;
+    _rotation = RTCVideoRotation_90;
+    [center addObserver:self
+               selector:@selector(deviceOrientationDidChange:)
+                   name:UIDeviceOrientationDidChangeNotification
+                 object:nil];
+    [center addObserver:self
+               selector:@selector(handleCaptureSessionInterruption:)
+                   name:AVCaptureSessionWasInterruptedNotification
+                 object:_captureSession];
+    [center addObserver:self
+               selector:@selector(handleCaptureSessionInterruptionEnded:)
+                   name:AVCaptureSessionInterruptionEndedNotification
+                 object:_captureSession];
+    [center addObserver:self
+               selector:@selector(handleApplicationDidBecomeActive:)
+                   name:UIApplicationDidBecomeActiveNotification
+                 object:[UIApplication sharedApplication]];
+#endif
+    [center addObserver:self
+               selector:@selector(handleCaptureSessionRuntimeError:)
+                   name:AVCaptureSessionRuntimeErrorNotification
+                 object:_captureSession];
+    [center addObserver:self
+               selector:@selector(handleCaptureSessionDidStartRunning:)
+                   name:AVCaptureSessionDidStartRunningNotification
+                 object:_captureSession];
+    [center addObserver:self
+               selector:@selector(handleCaptureSessionDidStopRunning:)
+                   name:AVCaptureSessionDidStopRunningNotification
+                 object:_captureSession];
+  }
+  return self;
+}
+
+- (void)dealloc {
+  NSAssert(!_willBeRunning,
+           @"Session was still running in RTC_OBJC_TYPE(RTCCameraVideoCapturer) dealloc. Forgot to "
+           @"call stopCapture?");
+  [[NSNotificationCenter defaultCenter] removeObserver:self];
+}
+
++ (NSArray<AVCaptureDevice *> *)captureDevices {
+#if defined(WEBRTC_IOS) && defined(__IPHONE_10_0) && \
+    __IPHONE_OS_VERSION_MIN_REQUIRED >= __IPHONE_10_0
+  AVCaptureDeviceDiscoverySession *session = [AVCaptureDeviceDiscoverySession
+      discoverySessionWithDeviceTypes:@[ AVCaptureDeviceTypeBuiltInWideAngleCamera ]
+                            mediaType:AVMediaTypeVideo
+                             position:AVCaptureDevicePositionUnspecified];
+  return session.devices;
+#else
+  return [AVCaptureDevice devicesWithMediaType:AVMediaTypeVideo];
+#endif
+}
+
++ (NSArray<AVCaptureDeviceFormat *> *)supportedFormatsForDevice:(AVCaptureDevice *)device {
+  // Support opening the device in any format. We make sure it's converted to a format we
+  // can handle, if needed, in the method `-setupVideoDataOutput`.
+  return device.formats;
+}
+
+- (FourCharCode)preferredOutputPixelFormat {
+  return _preferredOutputPixelFormat;
+}
+
+- (void)startCaptureWithDevice:(AVCaptureDevice *)device
+                        format:(AVCaptureDeviceFormat *)format
+                           fps:(NSInteger)fps {
+  [self startCaptureWithDevice:device format:format fps:fps completionHandler:nil];
+}
+
+- (void)stopCapture {
+  [self stopCaptureWithCompletionHandler:nil];
+}
+
+- (void)startCaptureWithDevice:(AVCaptureDevice *)device
+                        format:(AVCaptureDeviceFormat *)format
+                           fps:(NSInteger)fps
+             completionHandler:(nullable void (^)(NSError *_Nullable error))completionHandler {
+  _willBeRunning = YES;
+  [RTC_OBJC_TYPE(RTCDispatcher)
+      dispatchAsyncOnType:RTCDispatcherTypeCaptureSession
+                    block:^{
+                      RTCLogInfo("startCaptureWithDevice %@ @ %ld fps", format, (long)fps);
+
+#if TARGET_OS_IPHONE
+                      dispatch_async(dispatch_get_main_queue(), ^{
+                        if (!self->_generatingOrientationNotifications) {
+                          [[UIDevice currentDevice] beginGeneratingDeviceOrientationNotifications];
+                          self->_generatingOrientationNotifications = YES;
+                        }
+                      });
+#endif
+
+                      self.currentDevice = device;
+
+                      NSError *error = nil;
+                      if (![self.currentDevice lockForConfiguration:&error]) {
+                        RTCLogError(@"Failed to lock device %@. Error: %@",
+                                    self.currentDevice,
+                                    error.userInfo);
+                        if (completionHandler) {
+                          completionHandler(error);
+                        }
+                        self.willBeRunning = NO;
+                        return;
+                      }
+                      [self reconfigureCaptureSessionInput];
+                      [self updateOrientation];
+                      [self updateDeviceCaptureFormat:format fps:fps];
+                      [self updateVideoDataOutputPixelFormat:format];
+                      [self.captureSession startRunning];
+                      [self.currentDevice unlockForConfiguration];
+                      self.isRunning = YES;
+                      if (completionHandler) {
+                        completionHandler(nil);
+                      }
+                    }];
+}
+
+- (void)stopCaptureWithCompletionHandler:(nullable void (^)(void))completionHandler {
+  _willBeRunning = NO;
+  [RTC_OBJC_TYPE(RTCDispatcher)
+      dispatchAsyncOnType:RTCDispatcherTypeCaptureSession
+                    block:^{
+                      RTCLogInfo("Stop");
+                      self.currentDevice = nil;
+                      for (AVCaptureDeviceInput *oldInput in [self.captureSession.inputs copy]) {
+                        [self.captureSession removeInput:oldInput];
+                      }
+                      [self.captureSession stopRunning];
+
+#if TARGET_OS_IPHONE
+                      dispatch_async(dispatch_get_main_queue(), ^{
+                        if (self->_generatingOrientationNotifications) {
+                          [[UIDevice currentDevice] endGeneratingDeviceOrientationNotifications];
+                          self->_generatingOrientationNotifications = NO;
+                        }
+                      });
+#endif
+                      self.isRunning = NO;
+                      if (completionHandler) {
+                        completionHandler();
+                      }
+                    }];
+}
+
+#pragma mark iOS notifications
+
+#if TARGET_OS_IPHONE
+- (void)deviceOrientationDidChange:(NSNotification *)notification {
+  [RTC_OBJC_TYPE(RTCDispatcher) dispatchAsyncOnType:RTCDispatcherTypeCaptureSession
+                                              block:^{
+                                                [self updateOrientation];
+                                              }];
+}
+#endif
+
+#pragma mark AVCaptureVideoDataOutputSampleBufferDelegate
+
+- (void)captureOutput:(AVCaptureOutput *)captureOutput
+    didOutputSampleBuffer:(CMSampleBufferRef)sampleBuffer
+           fromConnection:(AVCaptureConnection *)connection {
+  NSParameterAssert(captureOutput == _videoDataOutput);
+
+  if (CMSampleBufferGetNumSamples(sampleBuffer) != 1 || !CMSampleBufferIsValid(sampleBuffer) ||
+      !CMSampleBufferDataIsReady(sampleBuffer)) {
+    return;
+  }
+
+  CVPixelBufferRef pixelBuffer = CMSampleBufferGetImageBuffer(sampleBuffer);
+  if (pixelBuffer == nil) {
+    return;
+  }
+
+#if TARGET_OS_IPHONE
+  // Default to portrait orientation on iPhone.
+  BOOL usingFrontCamera = NO;
+  // Check the image's EXIF for the camera the image came from as the image could have been
+  // delayed as we set alwaysDiscardsLateVideoFrames to NO.
+  AVCaptureDevicePosition cameraPosition =
+      [AVCaptureSession devicePositionForSampleBuffer:sampleBuffer];
+  if (cameraPosition != AVCaptureDevicePositionUnspecified) {
+    usingFrontCamera = AVCaptureDevicePositionFront == cameraPosition;
+  } else {
+    AVCaptureDeviceInput *deviceInput =
+        (AVCaptureDeviceInput *)((AVCaptureInputPort *)connection.inputPorts.firstObject).input;
+    usingFrontCamera = AVCaptureDevicePositionFront == deviceInput.device.position;
+  }
+  switch (_orientation) {
+    case UIDeviceOrientationPortrait:
+      _rotation = RTCVideoRotation_90;
+      break;
+    case UIDeviceOrientationPortraitUpsideDown:
+      _rotation = RTCVideoRotation_270;
+      break;
+    case UIDeviceOrientationLandscapeLeft:
+      _rotation = usingFrontCamera ? RTCVideoRotation_180 : RTCVideoRotation_0;
+      break;
+    case UIDeviceOrientationLandscapeRight:
+      _rotation = usingFrontCamera ? RTCVideoRotation_0 : RTCVideoRotation_180;
+      break;
+    case UIDeviceOrientationFaceUp:
+    case UIDeviceOrientationFaceDown:
+    case UIDeviceOrientationUnknown:
+      // Ignore.
+      break;
+  }
+#else
+  // No rotation on Mac.
+  _rotation = RTCVideoRotation_0;
+#endif
+
+  RTC_OBJC_TYPE(RTCCVPixelBuffer) *rtcPixelBuffer =
+      [[RTC_OBJC_TYPE(RTCCVPixelBuffer) alloc] initWithPixelBuffer:pixelBuffer];
+  int64_t timeStampNs = CMTimeGetSeconds(CMSampleBufferGetPresentationTimeStamp(sampleBuffer)) *
+      kNanosecondsPerSecond;
+  RTC_OBJC_TYPE(RTCVideoFrame) *videoFrame =
+      [[RTC_OBJC_TYPE(RTCVideoFrame) alloc] initWithBuffer:rtcPixelBuffer
+                                                  rotation:_rotation
+                                               timeStampNs:timeStampNs];
+  [self.delegate capturer:self didCaptureVideoFrame:videoFrame];
+}
+
+- (void)captureOutput:(AVCaptureOutput *)captureOutput
+    didDropSampleBuffer:(CMSampleBufferRef)sampleBuffer
+         fromConnection:(AVCaptureConnection *)connection {
+#if TARGET_OS_IPHONE
+  CFStringRef droppedReason =
+      CMGetAttachment(sampleBuffer, kCMSampleBufferAttachmentKey_DroppedFrameReason, nil);
+#else
+  // DroppedFrameReason unavailable on macOS.
+  CFStringRef droppedReason = nil;
+#endif
+  RTCLogError(@"Dropped sample buffer. Reason: %@", (__bridge NSString *)droppedReason);
+}
+
+#pragma mark - AVCaptureSession notifications
+
+- (void)handleCaptureSessionInterruption:(NSNotification *)notification {
+  NSString *reasonString = nil;
+#if TARGET_OS_IPHONE
+  NSNumber *reason = notification.userInfo[AVCaptureSessionInterruptionReasonKey];
+  if (reason) {
+    switch (reason.intValue) {
+      case AVCaptureSessionInterruptionReasonVideoDeviceNotAvailableInBackground:
+        reasonString = @"VideoDeviceNotAvailableInBackground";
+        break;
+      case AVCaptureSessionInterruptionReasonAudioDeviceInUseByAnotherClient:
+        reasonString = @"AudioDeviceInUseByAnotherClient";
+        break;
+      case AVCaptureSessionInterruptionReasonVideoDeviceInUseByAnotherClient:
+        reasonString = @"VideoDeviceInUseByAnotherClient";
+        break;
+      case AVCaptureSessionInterruptionReasonVideoDeviceNotAvailableWithMultipleForegroundApps:
+        reasonString = @"VideoDeviceNotAvailableWithMultipleForegroundApps";
+        break;
+    }
+  }
+#endif
+  RTCLog(@"Capture session interrupted: %@", reasonString);
+}
+
+- (void)handleCaptureSessionInterruptionEnded:(NSNotification *)notification {
+  RTCLog(@"Capture session interruption ended.");
+}
+
+- (void)handleCaptureSessionRuntimeError:(NSNotification *)notification {
+  NSError *error = [notification.userInfo objectForKey:AVCaptureSessionErrorKey];
+  RTCLogError(@"Capture session runtime error: %@", error);
+
+  [RTC_OBJC_TYPE(RTCDispatcher) dispatchAsyncOnType:RTCDispatcherTypeCaptureSession
+                                              block:^{
+#if TARGET_OS_IPHONE
+                                                if (error.code == AVErrorMediaServicesWereReset) {
+                                                  [self handleNonFatalError];
+                                                } else {
+                                                  [self handleFatalError];
+                                                }
+#else
+        [self handleFatalError];
+#endif
+                                              }];
+}
+
+- (void)handleCaptureSessionDidStartRunning:(NSNotification *)notification {
+  RTCLog(@"Capture session started.");
+
+  [RTC_OBJC_TYPE(RTCDispatcher) dispatchAsyncOnType:RTCDispatcherTypeCaptureSession
+                                              block:^{
+                                                // If we successfully restarted after an unknown
+                                                // error, allow future retries on fatal errors.
+                                                self.hasRetriedOnFatalError = NO;
+                                              }];
+}
+
+- (void)handleCaptureSessionDidStopRunning:(NSNotification *)notification {
+  RTCLog(@"Capture session stopped.");
+}
+
+- (void)handleFatalError {
+  [RTC_OBJC_TYPE(RTCDispatcher)
+      dispatchAsyncOnType:RTCDispatcherTypeCaptureSession
+                    block:^{
+                      if (!self.hasRetriedOnFatalError) {
+                        RTCLogWarning(@"Attempting to recover from fatal capture error.");
+                        [self handleNonFatalError];
+                        self.hasRetriedOnFatalError = YES;
+                      } else {
+                        RTCLogError(@"Previous fatal error recovery failed.");
+                      }
+                    }];
+}
+
+- (void)handleNonFatalError {
+  [RTC_OBJC_TYPE(RTCDispatcher) dispatchAsyncOnType:RTCDispatcherTypeCaptureSession
+                                              block:^{
+                                                RTCLog(@"Restarting capture session after error.");
+                                                if (self.isRunning) {
+                                                  [self.captureSession startRunning];
+                                                }
+                                              }];
+}
+
+#if TARGET_OS_IPHONE
+
+#pragma mark - UIApplication notifications
+
+- (void)handleApplicationDidBecomeActive:(NSNotification *)notification {
+  [RTC_OBJC_TYPE(RTCDispatcher)
+      dispatchAsyncOnType:RTCDispatcherTypeCaptureSession
+                    block:^{
+                      if (self.isRunning && !self.captureSession.isRunning) {
+                        RTCLog(@"Restarting capture session on active.");
+                        [self.captureSession startRunning];
+                      }
+                    }];
+}
+
+#endif  // TARGET_OS_IPHONE
+
+#pragma mark - Private
+
+- (dispatch_queue_t)frameQueue {
+  if (!_frameQueue) {
+    _frameQueue = RTCDispatchQueueCreateWithTarget(
+        "org.webrtc.cameravideocapturer.video",
+        DISPATCH_QUEUE_SERIAL,
+        dispatch_get_global_queue(DISPATCH_QUEUE_PRIORITY_HIGH, 0));
+  }
+  return _frameQueue;
+}
+
+- (BOOL)setupCaptureSession:(AVCaptureSession *)captureSession {
+  NSAssert(_captureSession == nil, @"Setup capture session called twice.");
+  _captureSession = captureSession;
+#if defined(WEBRTC_IOS)
+  _captureSession.sessionPreset = AVCaptureSessionPresetInputPriority;
+  _captureSession.usesApplicationAudioSession = NO;
+#endif
+  [self setupVideoDataOutput];
+  // Add the output.
+  if (![_captureSession canAddOutput:_videoDataOutput]) {
+    RTCLogError(@"Video data output unsupported.");
+    return NO;
+  }
+  [_captureSession addOutput:_videoDataOutput];
+
+  return YES;
+}
+
+- (void)setupVideoDataOutput {
+  NSAssert(_videoDataOutput == nil, @"Setup video data output called twice.");
+  AVCaptureVideoDataOutput *videoDataOutput = [[AVCaptureVideoDataOutput alloc] init];
+
+  // `videoDataOutput.availableVideoCVPixelFormatTypes` returns the pixel formats supported by the
+  // device with the most efficient output format first. Find the first format that we support.
+  NSSet<NSNumber *> *supportedPixelFormats =
+      [RTC_OBJC_TYPE(RTCCVPixelBuffer) supportedPixelFormats];
+  NSMutableOrderedSet *availablePixelFormats =
+      [NSMutableOrderedSet orderedSetWithArray:videoDataOutput.availableVideoCVPixelFormatTypes];
+  [availablePixelFormats intersectSet:supportedPixelFormats];
+  NSNumber *pixelFormat = availablePixelFormats.firstObject;
+  NSAssert(pixelFormat, @"Output device has no supported formats.");
+
+  _preferredOutputPixelFormat = [pixelFormat unsignedIntValue];
+  _outputPixelFormat = _preferredOutputPixelFormat;
+  videoDataOutput.videoSettings = @{(NSString *)kCVPixelBufferPixelFormatTypeKey : pixelFormat};
+  videoDataOutput.alwaysDiscardsLateVideoFrames = NO;
+  [videoDataOutput setSampleBufferDelegate:self queue:self.frameQueue];
+  _videoDataOutput = videoDataOutput;
+}
+
+- (void)updateVideoDataOutputPixelFormat:(AVCaptureDeviceFormat *)format {
+  FourCharCode mediaSubType = CMFormatDescriptionGetMediaSubType(format.formatDescription);
+  if (![[RTC_OBJC_TYPE(RTCCVPixelBuffer) supportedPixelFormats] containsObject:@(mediaSubType)]) {
+    mediaSubType = _preferredOutputPixelFormat;
+  }
+
+  if (mediaSubType != _outputPixelFormat) {
+    _outputPixelFormat = mediaSubType;
+  }
+
+  // Update videoSettings with dimensions, as some virtual cameras, e.g. Snap Camera, may not work
+  // otherwise.
+  CMVideoDimensions dimensions = CMVideoFormatDescriptionGetDimensions(format.formatDescription);
+  _videoDataOutput.videoSettings = @{
+    (id)kCVPixelBufferWidthKey : @(dimensions.width),
+    (id)kCVPixelBufferHeightKey : @(dimensions.height),
+    (id)kCVPixelBufferPixelFormatTypeKey : @(_outputPixelFormat),
+  };
+}
+
+#pragma mark - Private, called inside capture queue
+
+- (void)updateDeviceCaptureFormat:(AVCaptureDeviceFormat *)format fps:(NSInteger)fps {
+  NSAssert([RTC_OBJC_TYPE(RTCDispatcher) isOnQueueForType:RTCDispatcherTypeCaptureSession],
+           @"updateDeviceCaptureFormat must be called on the capture queue.");
+  @try {
+    _currentDevice.activeFormat = format;
+    _currentDevice.activeVideoMinFrameDuration = CMTimeMake(1, fps);
+  } @catch (NSException *exception) {
+    RTCLogError(@"Failed to set active format!\n User info:%@", exception.userInfo);
+    return;
+  }
+}
+
+- (void)reconfigureCaptureSessionInput {
+  NSAssert([RTC_OBJC_TYPE(RTCDispatcher) isOnQueueForType:RTCDispatcherTypeCaptureSession],
+           @"reconfigureCaptureSessionInput must be called on the capture queue.");
+  NSError *error = nil;
+  AVCaptureDeviceInput *input =
+      [AVCaptureDeviceInput deviceInputWithDevice:_currentDevice error:&error];
+  if (!input) {
+    RTCLogError(@"Failed to create front camera input: %@", error.localizedDescription);
+    return;
+  }
+  [_captureSession beginConfiguration];
+  for (AVCaptureDeviceInput *oldInput in [_captureSession.inputs copy]) {
+    [_captureSession removeInput:oldInput];
+  }
+  if ([_captureSession canAddInput:input]) {
+    [_captureSession addInput:input];
+  } else {
+    RTCLogError(@"Cannot add camera as an input to the session.");
+  }
+  [_captureSession commitConfiguration];
+}
+
+- (void)updateOrientation {
+  NSAssert([RTC_OBJC_TYPE(RTCDispatcher) isOnQueueForType:RTCDispatcherTypeCaptureSession],
+           @"updateOrientation must be called on the capture queue.");
+#if TARGET_OS_IPHONE
+  _orientation = [UIDevice currentDevice].orientation;
+#endif
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/components/capturer/RTCFileVideoCapturer.h b/third_party/libwebrtc/sdk/objc/components/capturer/RTCFileVideoCapturer.h
new file mode 100644
index 0000000000..19262c64cf
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/components/capturer/RTCFileVideoCapturer.h
@@ -0,0 +1,51 @@
+/*
+ *  Copyright 2017 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCVideoCapturer.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+/**
+ * Error passing block.
+ */
+typedef void (^RTCFileVideoCapturerErrorBlock)(NSError *error);
+
+/**
+ * Captures buffers from bundled video file.
+ *
+ * See @c RTCVideoCapturer for more info on capturers.
+ */
+RTC_OBJC_EXPORT
+
+NS_CLASS_AVAILABLE_IOS(10)
+@interface RTC_OBJC_TYPE (RTCFileVideoCapturer) : RTC_OBJC_TYPE(RTCVideoCapturer)
+
+/**
+ * Starts asynchronous capture of frames from video file.
+ *
+ * Capturing is not started if error occurs. Underlying error will be
+ * relayed in the errorBlock if one is provided.
+ * Successfully captured video frames will be passed to the delegate.
+ *
+ * @param nameOfFile The name of the bundled video file to be read.
+ * @errorBlock block to be executed upon error.
+ */
+- (void)startCapturingFromFileNamed:(NSString *)nameOfFile
+                            onError:(__nullable RTCFileVideoCapturerErrorBlock)errorBlock;
+
+/**
+ * Immediately stops capture.
+ */
+- (void)stopCapture;
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/components/capturer/RTCFileVideoCapturer.m b/third_party/libwebrtc/sdk/objc/components/capturer/RTCFileVideoCapturer.m
new file mode 100644
index 0000000000..bcf1506259
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/components/capturer/RTCFileVideoCapturer.m
@@ -0,0 +1,215 @@
+/**
+ *  Copyright 2017 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCFileVideoCapturer.h"
+
+#import "base/RTCLogging.h"
+#import "base/RTCVideoFrameBuffer.h"
+#import "components/video_frame_buffer/RTCCVPixelBuffer.h"
+#include "rtc_base/system/gcd_helpers.h"
+
+NSString *const kRTCFileVideoCapturerErrorDomain =
+    @"org.webrtc.RTC_OBJC_TYPE(RTCFileVideoCapturer)";
+
+typedef NS_ENUM(NSInteger, RTCFileVideoCapturerErrorCode) {
+  RTCFileVideoCapturerErrorCode_CapturerRunning = 2000,
+  RTCFileVideoCapturerErrorCode_FileNotFound
+};
+
+typedef NS_ENUM(NSInteger, RTCFileVideoCapturerStatus) {
+  RTCFileVideoCapturerStatusNotInitialized,
+  RTCFileVideoCapturerStatusStarted,
+  RTCFileVideoCapturerStatusStopped
+};
+
+@interface RTC_OBJC_TYPE (RTCFileVideoCapturer)
+() @property(nonatomic, assign) CMTime lastPresentationTime;
+@property(nonatomic, strong) NSURL *fileURL;
+@end
+
+@implementation RTC_OBJC_TYPE (RTCFileVideoCapturer) {
+  AVAssetReader *_reader;
+  AVAssetReaderTrackOutput *_outTrack;
+  RTCFileVideoCapturerStatus _status;
+  dispatch_queue_t _frameQueue;
+}
+
+@synthesize lastPresentationTime = _lastPresentationTime;
+@synthesize fileURL = _fileURL;
+
+- (void)startCapturingFromFileNamed:(NSString *)nameOfFile
+                            onError:(RTCFileVideoCapturerErrorBlock)errorBlock {
+  if (_status == RTCFileVideoCapturerStatusStarted) {
+    NSError *error =
+        [NSError errorWithDomain:kRTCFileVideoCapturerErrorDomain
+                            code:RTCFileVideoCapturerErrorCode_CapturerRunning
+                        userInfo:@{NSUnderlyingErrorKey : @"Capturer has been started."}];
+
+    errorBlock(error);
+    return;
+  } else {
+    _status = RTCFileVideoCapturerStatusStarted;
+  }
+
+  dispatch_async(dispatch_get_global_queue(DISPATCH_QUEUE_PRIORITY_DEFAULT, 0), ^{
+    NSString *pathForFile = [self pathForFileName:nameOfFile];
+    if (!pathForFile) {
+      NSString *errorString =
+          [NSString stringWithFormat:@"File %@ not found in bundle", nameOfFile];
+      NSError *error = [NSError errorWithDomain:kRTCFileVideoCapturerErrorDomain
+                                           code:RTCFileVideoCapturerErrorCode_FileNotFound
+                                       userInfo:@{NSUnderlyingErrorKey : errorString}];
+      errorBlock(error);
+      return;
+    }
+
+    self.lastPresentationTime = CMTimeMake(0, 0);
+
+    self.fileURL = [NSURL fileURLWithPath:pathForFile];
+    [self setupReaderOnError:errorBlock];
+  });
+}
+
+- (void)setupReaderOnError:(RTCFileVideoCapturerErrorBlock)errorBlock {
+  AVURLAsset *asset = [AVURLAsset URLAssetWithURL:_fileURL options:nil];
+
+  NSArray *allTracks = [asset tracksWithMediaType:AVMediaTypeVideo];
+  NSError *error = nil;
+
+  _reader = [[AVAssetReader alloc] initWithAsset:asset error:&error];
+  if (error) {
+    errorBlock(error);
+    return;
+  }
+
+  NSDictionary *options = @{
+    (NSString *)kCVPixelBufferPixelFormatTypeKey : @(kCVPixelFormatType_420YpCbCr8BiPlanarFullRange)
+  };
+  _outTrack =
+      [[AVAssetReaderTrackOutput alloc] initWithTrack:allTracks.firstObject outputSettings:options];
+  [_reader addOutput:_outTrack];
+
+  [_reader startReading];
+  RTCLog(@"File capturer started reading");
+  [self readNextBuffer];
+}
+- (void)stopCapture {
+  _status = RTCFileVideoCapturerStatusStopped;
+  RTCLog(@"File capturer stopped.");
+}
+
+#pragma mark - Private
+
+- (nullable NSString *)pathForFileName:(NSString *)fileName {
+  NSArray *nameComponents = [fileName componentsSeparatedByString:@"."];
+  if (nameComponents.count != 2) {
+    return nil;
+  }
+
+  NSString *path =
+      [[NSBundle mainBundle] pathForResource:nameComponents[0] ofType:nameComponents[1]];
+  return path;
+}
+
+- (dispatch_queue_t)frameQueue {
+  if (!_frameQueue) {
+    _frameQueue = RTCDispatchQueueCreateWithTarget(
+        "org.webrtc.filecapturer.video",
+        DISPATCH_QUEUE_SERIAL,
+        dispatch_get_global_queue(DISPATCH_QUEUE_PRIORITY_BACKGROUND, 0));
+  }
+  return _frameQueue;
+}
+
+- (void)readNextBuffer {
+  if (_status == RTCFileVideoCapturerStatusStopped) {
+    [_reader cancelReading];
+    _reader = nil;
+    return;
+  }
+
+  if (_reader.status == AVAssetReaderStatusCompleted) {
+    [_reader cancelReading];
+    _reader = nil;
+    [self setupReaderOnError:nil];
+    return;
+  }
+
+  CMSampleBufferRef sampleBuffer = [_outTrack copyNextSampleBuffer];
+  if (!sampleBuffer) {
+    [self readNextBuffer];
+    return;
+  }
+  if (CMSampleBufferGetNumSamples(sampleBuffer) != 1 || !CMSampleBufferIsValid(sampleBuffer) ||
+      !CMSampleBufferDataIsReady(sampleBuffer)) {
+    CFRelease(sampleBuffer);
+    [self readNextBuffer];
+    return;
+  }
+
+  [self publishSampleBuffer:sampleBuffer];
+}
+
+- (void)publishSampleBuffer:(CMSampleBufferRef)sampleBuffer {
+  CMTime presentationTime = CMSampleBufferGetPresentationTimeStamp(sampleBuffer);
+  Float64 presentationDifference =
+      CMTimeGetSeconds(CMTimeSubtract(presentationTime, _lastPresentationTime));
+  _lastPresentationTime = presentationTime;
+  int64_t presentationDifferenceRound = lroundf(presentationDifference * NSEC_PER_SEC);
+
+  __block dispatch_source_t timer = [self createStrictTimer];
+  // Strict timer that will fire `presentationDifferenceRound` ns from now and never again.
+  dispatch_source_set_timer(timer,
+                            dispatch_time(DISPATCH_TIME_NOW, presentationDifferenceRound),
+                            DISPATCH_TIME_FOREVER,
+                            0);
+  dispatch_source_set_event_handler(timer, ^{
+    dispatch_source_cancel(timer);
+    timer = nil;
+
+    CVPixelBufferRef pixelBuffer = CMSampleBufferGetImageBuffer(sampleBuffer);
+    if (!pixelBuffer) {
+      CFRelease(sampleBuffer);
+      dispatch_async(dispatch_get_global_queue(DISPATCH_QUEUE_PRIORITY_DEFAULT, 0), ^{
+        [self readNextBuffer];
+      });
+      return;
+    }
+
+    RTC_OBJC_TYPE(RTCCVPixelBuffer) *rtcPixelBuffer =
+        [[RTC_OBJC_TYPE(RTCCVPixelBuffer) alloc] initWithPixelBuffer:pixelBuffer];
+    NSTimeInterval timeStampSeconds = CACurrentMediaTime();
+    int64_t timeStampNs = lroundf(timeStampSeconds * NSEC_PER_SEC);
+    RTC_OBJC_TYPE(RTCVideoFrame) *videoFrame =
+        [[RTC_OBJC_TYPE(RTCVideoFrame) alloc] initWithBuffer:rtcPixelBuffer
+                                                    rotation:0
+                                                 timeStampNs:timeStampNs];
+    CFRelease(sampleBuffer);
+
+    dispatch_async(dispatch_get_global_queue(DISPATCH_QUEUE_PRIORITY_DEFAULT, 0), ^{
+      [self readNextBuffer];
+    });
+
+    [self.delegate capturer:self didCaptureVideoFrame:videoFrame];
+  });
+  dispatch_activate(timer);
+}
+
+- (dispatch_source_t)createStrictTimer {
+  dispatch_source_t timer = dispatch_source_create(
+      DISPATCH_SOURCE_TYPE_TIMER, 0, DISPATCH_TIMER_STRICT, [self frameQueue]);
+  return timer;
+}
+
+- (void)dealloc {
+  [self stopCapture];
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/components/network/RTCNetworkMonitor+Private.h b/third_party/libwebrtc/sdk/objc/components/network/RTCNetworkMonitor+Private.h
new file mode 100644
index 0000000000..b5c786be18
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/components/network/RTCNetworkMonitor+Private.h
@@ -0,0 +1,26 @@
+/*
+ *  Copyright 2020 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCNetworkMonitor.h"
+
+#include "sdk/objc/native/src/network_monitor_observer.h"
+
+@interface RTCNetworkMonitor ()
+
+/** `observer` is a raw pointer and should be kept alive
+ *  for this object's lifetime.
+ */
+- (instancetype)initWithObserver:(webrtc::NetworkMonitorObserver *)observer
+    NS_DESIGNATED_INITIALIZER;
+
+/** Stops the receiver from posting updates to `observer`. */
+- (void)stop;
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/components/network/RTCNetworkMonitor.h b/third_party/libwebrtc/sdk/objc/components/network/RTCNetworkMonitor.h
new file mode 100644
index 0000000000..21d22f5463
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/components/network/RTCNetworkMonitor.h
@@ -0,0 +1,24 @@
+/*
+ *  Copyright 2020 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+NS_ASSUME_NONNULL_BEGIN
+
+/** Listens for NWPathMonitor updates and forwards the results to a C++
+ *  observer.
+ */
+@interface RTCNetworkMonitor : NSObject
+
+- (instancetype)init NS_UNAVAILABLE;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/components/network/RTCNetworkMonitor.mm b/third_party/libwebrtc/sdk/objc/components/network/RTCNetworkMonitor.mm
new file mode 100644
index 0000000000..7e75b2b4c0
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/components/network/RTCNetworkMonitor.mm
@@ -0,0 +1,126 @@
+/*
+ *  Copyright 2020 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCNetworkMonitor+Private.h"
+
+#import <Network/Network.h>
+
+#import "base/RTCLogging.h"
+#import "helpers/RTCDispatcher+Private.h"
+
+#include "rtc_base/string_utils.h"
+
+namespace {
+
+rtc::AdapterType AdapterTypeFromInterfaceType(nw_interface_type_t interfaceType) {
+  rtc::AdapterType adapterType = rtc::ADAPTER_TYPE_UNKNOWN;
+  switch (interfaceType) {
+    case nw_interface_type_other:
+      adapterType = rtc::ADAPTER_TYPE_UNKNOWN;
+      break;
+    case nw_interface_type_wifi:
+      adapterType = rtc::ADAPTER_TYPE_WIFI;
+      break;
+    case nw_interface_type_cellular:
+      adapterType = rtc::ADAPTER_TYPE_CELLULAR;
+      break;
+    case nw_interface_type_wired:
+      adapterType = rtc::ADAPTER_TYPE_ETHERNET;
+      break;
+    case nw_interface_type_loopback:
+      adapterType = rtc::ADAPTER_TYPE_LOOPBACK;
+      break;
+    default:
+      adapterType = rtc::ADAPTER_TYPE_UNKNOWN;
+      break;
+  }
+  return adapterType;
+}
+
+}  // namespace
+
+@implementation RTCNetworkMonitor {
+  webrtc::NetworkMonitorObserver *_observer;
+  nw_path_monitor_t _pathMonitor;
+  dispatch_queue_t _monitorQueue;
+}
+
+- (instancetype)initWithObserver:(webrtc::NetworkMonitorObserver *)observer {
+  RTC_DCHECK(observer);
+  if (self = [super init]) {
+    _observer = observer;
+    if (@available(iOS 12, *)) {
+      _pathMonitor = nw_path_monitor_create();
+      if (_pathMonitor == nil) {
+        RTCLog(@"nw_path_monitor_create failed.");
+        return nil;
+      }
+      RTCLog(@"NW path monitor created.");
+      __weak RTCNetworkMonitor *weakSelf = self;
+      nw_path_monitor_set_update_handler(_pathMonitor, ^(nw_path_t path) {
+        if (weakSelf == nil) {
+          return;
+        }
+        RTCNetworkMonitor *strongSelf = weakSelf;
+        RTCLog(@"NW path monitor: updated.");
+        nw_path_status_t status = nw_path_get_status(path);
+        if (status == nw_path_status_invalid) {
+          RTCLog(@"NW path monitor status: invalid.");
+        } else if (status == nw_path_status_unsatisfied) {
+          RTCLog(@"NW path monitor status: unsatisfied.");
+        } else if (status == nw_path_status_satisfied) {
+          RTCLog(@"NW path monitor status: satisfied.");
+        } else if (status == nw_path_status_satisfiable) {
+          RTCLog(@"NW path monitor status: satisfiable.");
+        }
+        std::map<std::string, rtc::AdapterType, rtc::AbslStringViewCmp> *map =
+            new std::map<std::string, rtc::AdapterType, rtc::AbslStringViewCmp>();
+        nw_path_enumerate_interfaces(
+            path, (nw_path_enumerate_interfaces_block_t) ^ (nw_interface_t interface) {
+              const char *name = nw_interface_get_name(interface);
+              nw_interface_type_t interfaceType = nw_interface_get_type(interface);
+              RTCLog(@"NW path monitor available interface: %s", name);
+              rtc::AdapterType adapterType = AdapterTypeFromInterfaceType(interfaceType);
+              map->insert(std::pair<std::string, rtc::AdapterType>(name, adapterType));
+            });
+        @synchronized(strongSelf) {
+          webrtc::NetworkMonitorObserver *observer = strongSelf->_observer;
+          if (observer) {
+            observer->OnPathUpdate(std::move(*map));
+          }
+        }
+        delete map;
+      });
+      nw_path_monitor_set_queue(
+          _pathMonitor,
+          [RTC_OBJC_TYPE(RTCDispatcher) dispatchQueueForType:RTCDispatcherTypeNetworkMonitor]);
+      nw_path_monitor_start(_pathMonitor);
+    }
+  }
+  return self;
+}
+
+- (void)cancel {
+  if (@available(iOS 12, *)) {
+    nw_path_monitor_cancel(_pathMonitor);
+  }
+}
+- (void)stop {
+  [self cancel];
+  @synchronized(self) {
+    _observer = nil;
+  }
+}
+
+- (void)dealloc {
+  [self cancel];
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/components/renderer/metal/RTCMTLI420Renderer.h b/third_party/libwebrtc/sdk/objc/components/renderer/metal/RTCMTLI420Renderer.h
new file mode 100644
index 0000000000..e5987fe22a
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/components/renderer/metal/RTCMTLI420Renderer.h
@@ -0,0 +1,17 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMTLRenderer.h"
+
+NS_AVAILABLE(10_11, 9_0)
+@interface RTCMTLI420Renderer : RTCMTLRenderer
+@end
diff --git a/third_party/libwebrtc/sdk/objc/components/renderer/metal/RTCMTLI420Renderer.mm b/third_party/libwebrtc/sdk/objc/components/renderer/metal/RTCMTLI420Renderer.mm
new file mode 100644
index 0000000000..f4c76fa313
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/components/renderer/metal/RTCMTLI420Renderer.mm
@@ -0,0 +1,170 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCMTLI420Renderer.h"
+
+#import <Metal/Metal.h>
+#import <MetalKit/MetalKit.h>
+
+#import "base/RTCI420Buffer.h"
+#import "base/RTCLogging.h"
+#import "base/RTCVideoFrame.h"
+#import "base/RTCVideoFrameBuffer.h"
+
+#import "RTCMTLRenderer+Private.h"
+
+static NSString *const shaderSource = MTL_STRINGIFY(
+    using namespace metal;
+
+    typedef struct {
+      packed_float2 position;
+      packed_float2 texcoord;
+    } Vertex;
+
+    typedef struct {
+      float4 position[[position]];
+      float2 texcoord;
+    } Varyings;
+
+    vertex Varyings vertexPassthrough(constant Vertex *verticies[[buffer(0)]],
+                                      unsigned int vid[[vertex_id]]) {
+      Varyings out;
+      constant Vertex &v = verticies[vid];
+      out.position = float4(float2(v.position), 0.0, 1.0);
+      out.texcoord = v.texcoord;
+
+      return out;
+    }
+
+    fragment half4 fragmentColorConversion(
+        Varyings in[[stage_in]],
+        texture2d<float, access::sample> textureY[[texture(0)]],
+        texture2d<float, access::sample> textureU[[texture(1)]],
+        texture2d<float, access::sample> textureV[[texture(2)]]) {
+      constexpr sampler s(address::clamp_to_edge, filter::linear);
+      float y;
+      float u;
+      float v;
+      float r;
+      float g;
+      float b;
+      // Conversion for YUV to rgb from http://www.fourcc.org/fccyvrgb.php
+      y = textureY.sample(s, in.texcoord).r;
+      u = textureU.sample(s, in.texcoord).r;
+      v = textureV.sample(s, in.texcoord).r;
+      u = u - 0.5;
+      v = v - 0.5;
+      r = y + 1.403 * v;
+      g = y - 0.344 * u - 0.714 * v;
+      b = y + 1.770 * u;
+
+      float4 out = float4(r, g, b, 1.0);
+
+      return half4(out);
+    });
+
+@implementation RTCMTLI420Renderer {
+  // Textures.
+  id<MTLTexture> _yTexture;
+  id<MTLTexture> _uTexture;
+  id<MTLTexture> _vTexture;
+
+  MTLTextureDescriptor *_descriptor;
+  MTLTextureDescriptor *_chromaDescriptor;
+
+  int _width;
+  int _height;
+  int _chromaWidth;
+  int _chromaHeight;
+}
+
+#pragma mark - Virtual
+
+- (NSString *)shaderSource {
+  return shaderSource;
+}
+
+- (void)getWidth:(nonnull int *)width
+          height:(nonnull int *)height
+       cropWidth:(nonnull int *)cropWidth
+      cropHeight:(nonnull int *)cropHeight
+           cropX:(nonnull int *)cropX
+           cropY:(nonnull int *)cropY
+         ofFrame:(nonnull RTC_OBJC_TYPE(RTCVideoFrame) *)frame {
+  *width = frame.width;
+  *height = frame.height;
+  *cropWidth = frame.width;
+  *cropHeight = frame.height;
+  *cropX = 0;
+  *cropY = 0;
+}
+
+- (BOOL)setupTexturesForFrame:(nonnull RTC_OBJC_TYPE(RTCVideoFrame) *)frame {
+  if (![super setupTexturesForFrame:frame]) {
+    return NO;
+  }
+
+  id<MTLDevice> device = [self currentMetalDevice];
+  if (!device) {
+    return NO;
+  }
+
+  id<RTC_OBJC_TYPE(RTCI420Buffer)> buffer = [frame.buffer toI420];
+
+  // Luma (y) texture.
+  if (!_descriptor || _width != frame.width || _height != frame.height) {
+    _width = frame.width;
+    _height = frame.height;
+    _descriptor = [MTLTextureDescriptor texture2DDescriptorWithPixelFormat:MTLPixelFormatR8Unorm
+                                                                     width:_width
+                                                                    height:_height
+                                                                 mipmapped:NO];
+    _descriptor.usage = MTLTextureUsageShaderRead;
+    _yTexture = [device newTextureWithDescriptor:_descriptor];
+  }
+
+  // Chroma (u,v) textures
+  [_yTexture replaceRegion:MTLRegionMake2D(0, 0, _width, _height)
+               mipmapLevel:0
+                 withBytes:buffer.dataY
+               bytesPerRow:buffer.strideY];
+
+  if (!_chromaDescriptor || _chromaWidth != frame.width / 2 || _chromaHeight != frame.height / 2) {
+    _chromaWidth = frame.width / 2;
+    _chromaHeight = frame.height / 2;
+    _chromaDescriptor =
+        [MTLTextureDescriptor texture2DDescriptorWithPixelFormat:MTLPixelFormatR8Unorm
+                                                           width:_chromaWidth
+                                                          height:_chromaHeight
+                                                       mipmapped:NO];
+    _chromaDescriptor.usage = MTLTextureUsageShaderRead;
+    _uTexture = [device newTextureWithDescriptor:_chromaDescriptor];
+    _vTexture = [device newTextureWithDescriptor:_chromaDescriptor];
+  }
+
+  [_uTexture replaceRegion:MTLRegionMake2D(0, 0, _chromaWidth, _chromaHeight)
+               mipmapLevel:0
+                 withBytes:buffer.dataU
+               bytesPerRow:buffer.strideU];
+  [_vTexture replaceRegion:MTLRegionMake2D(0, 0, _chromaWidth, _chromaHeight)
+               mipmapLevel:0
+                 withBytes:buffer.dataV
+               bytesPerRow:buffer.strideV];
+
+  return (_uTexture != nil) && (_yTexture != nil) && (_vTexture != nil);
+}
+
+- (void)uploadTexturesToRenderEncoder:(id<MTLRenderCommandEncoder>)renderEncoder {
+  [renderEncoder setFragmentTexture:_yTexture atIndex:0];
+  [renderEncoder setFragmentTexture:_uTexture atIndex:1];
+  [renderEncoder setFragmentTexture:_vTexture atIndex:2];
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/components/renderer/metal/RTCMTLNSVideoView.h b/third_party/libwebrtc/sdk/objc/components/renderer/metal/RTCMTLNSVideoView.h
new file mode 100644
index 0000000000..f70e2ad5ee
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/components/renderer/metal/RTCMTLNSVideoView.h
@@ -0,0 +1,24 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <AppKit/AppKit.h>
+
+#import "RTCVideoRenderer.h"
+
+NS_AVAILABLE_MAC(10.11)
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCMTLNSVideoView) : NSView <RTC_OBJC_TYPE(RTCVideoRenderer)>
+
+@property(nonatomic, weak) id<RTC_OBJC_TYPE(RTCVideoViewDelegate)> delegate;
+
++ (BOOL)isMetalAvailable;
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/components/renderer/metal/RTCMTLNSVideoView.m b/third_party/libwebrtc/sdk/objc/components/renderer/metal/RTCMTLNSVideoView.m
new file mode 100644
index 0000000000..625fb1caa7
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/components/renderer/metal/RTCMTLNSVideoView.m
@@ -0,0 +1,122 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCMTLNSVideoView.h"
+
+#import <Metal/Metal.h>
+#import <MetalKit/MetalKit.h>
+
+#import "base/RTCVideoFrame.h"
+
+#import "RTCMTLI420Renderer.h"
+
+@interface RTC_OBJC_TYPE (RTCMTLNSVideoView)
+()<MTKViewDelegate> @property(nonatomic) id<RTCMTLRenderer> renderer;
+@property(nonatomic, strong) MTKView *metalView;
+@property(atomic, strong) RTC_OBJC_TYPE(RTCVideoFrame) * videoFrame;
+@end
+
+@implementation RTC_OBJC_TYPE (RTCMTLNSVideoView) {
+  id<RTCMTLRenderer> _renderer;
+}
+
+@synthesize delegate = _delegate;
+@synthesize renderer = _renderer;
+@synthesize metalView = _metalView;
+@synthesize videoFrame = _videoFrame;
+
+- (instancetype)initWithFrame:(CGRect)frameRect {
+  self = [super initWithFrame:frameRect];
+  if (self) {
+    [self configure];
+  }
+  return self;
+}
+
+- (instancetype)initWithCoder:(NSCoder *)aCoder {
+  self = [super initWithCoder:aCoder];
+  if (self) {
+    [self configure];
+  }
+  return self;
+}
+
+#pragma mark - Private
+
++ (BOOL)isMetalAvailable {
+  return [MTLCopyAllDevices() count] > 0;
+}
+
+- (void)configure {
+  if ([[self class] isMetalAvailable]) {
+    _metalView = [[MTKView alloc] initWithFrame:self.bounds];
+    [self addSubview:_metalView];
+    _metalView.layerContentsPlacement = NSViewLayerContentsPlacementScaleProportionallyToFit;
+    _metalView.translatesAutoresizingMaskIntoConstraints = NO;
+    _metalView.framebufferOnly = YES;
+    _metalView.delegate = self;
+
+    _renderer = [[RTCMTLI420Renderer alloc] init];
+    if (![(RTCMTLI420Renderer *)_renderer addRenderingDestination:_metalView]) {
+      _renderer = nil;
+    };
+  }
+}
+
+- (void)updateConstraints {
+  NSDictionary *views = NSDictionaryOfVariableBindings(_metalView);
+
+  NSArray *constraintsHorizontal =
+      [NSLayoutConstraint constraintsWithVisualFormat:@"H:|-0-[_metalView]-0-|"
+                                              options:0
+                                              metrics:nil
+                                                views:views];
+  [self addConstraints:constraintsHorizontal];
+
+  NSArray *constraintsVertical =
+      [NSLayoutConstraint constraintsWithVisualFormat:@"V:|-0-[_metalView]-0-|"
+                                              options:0
+                                              metrics:nil
+                                                views:views];
+  [self addConstraints:constraintsVertical];
+  [super updateConstraints];
+}
+
+#pragma mark - MTKViewDelegate methods
+- (void)drawInMTKView:(nonnull MTKView *)view {
+  if (self.videoFrame == nil) {
+    return;
+  }
+  if (view == self.metalView) {
+    [_renderer drawFrame:self.videoFrame];
+  }
+}
+
+- (void)mtkView:(MTKView *)view drawableSizeWillChange:(CGSize)size {
+}
+
+#pragma mark - RTC_OBJC_TYPE(RTCVideoRenderer)
+
+- (void)setSize:(CGSize)size {
+  _metalView.drawableSize = size;
+  dispatch_async(dispatch_get_main_queue(), ^{
+    [self.delegate videoView:self didChangeVideoSize:size];
+  });
+  [_metalView draw];
+}
+
+- (void)renderFrame:(nullable RTC_OBJC_TYPE(RTCVideoFrame) *)frame {
+  if (frame == nil) {
+    return;
+  }
+  self.videoFrame = [frame newI420VideoFrame];
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/components/renderer/metal/RTCMTLNV12Renderer.h b/third_party/libwebrtc/sdk/objc/components/renderer/metal/RTCMTLNV12Renderer.h
new file mode 100644
index 0000000000..866b7ea17e
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/components/renderer/metal/RTCMTLNV12Renderer.h
@@ -0,0 +1,18 @@
+/*
+ *  Copyright 2017 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMTLRenderer.h"
+
+NS_AVAILABLE(10_11, 9_0)
+@interface RTCMTLNV12Renderer : RTCMTLRenderer
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/components/renderer/metal/RTCMTLNV12Renderer.mm b/third_party/libwebrtc/sdk/objc/components/renderer/metal/RTCMTLNV12Renderer.mm
new file mode 100644
index 0000000000..7b037c6dbc
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/components/renderer/metal/RTCMTLNV12Renderer.mm
@@ -0,0 +1,164 @@
+/*
+ *  Copyright 2017 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCMTLNV12Renderer.h"
+
+#import <Metal/Metal.h>
+#import <MetalKit/MetalKit.h>
+
+#import "RTCMTLRenderer+Private.h"
+#import "base/RTCLogging.h"
+#import "base/RTCVideoFrame.h"
+#import "base/RTCVideoFrameBuffer.h"
+#import "components/video_frame_buffer/RTCCVPixelBuffer.h"
+
+#include "rtc_base/checks.h"
+
+static NSString *const shaderSource = MTL_STRINGIFY(
+    using namespace metal;
+
+    typedef struct {
+      packed_float2 position;
+      packed_float2 texcoord;
+    } Vertex;
+
+    typedef struct {
+      float4 position[[position]];
+      float2 texcoord;
+    } Varyings;
+
+    vertex Varyings vertexPassthrough(constant Vertex *verticies[[buffer(0)]],
+                                      unsigned int vid[[vertex_id]]) {
+      Varyings out;
+      constant Vertex &v = verticies[vid];
+      out.position = float4(float2(v.position), 0.0, 1.0);
+      out.texcoord = v.texcoord;
+      return out;
+    }
+
+    // Receiving YCrCb textures.
+    fragment half4 fragmentColorConversion(
+        Varyings in[[stage_in]],
+        texture2d<float, access::sample> textureY[[texture(0)]],
+        texture2d<float, access::sample> textureCbCr[[texture(1)]]) {
+      constexpr sampler s(address::clamp_to_edge, filter::linear);
+      float y;
+      float2 uv;
+      y = textureY.sample(s, in.texcoord).r;
+      uv = textureCbCr.sample(s, in.texcoord).rg - float2(0.5, 0.5);
+
+      // Conversion for YUV to rgb from http://www.fourcc.org/fccyvrgb.php
+      float4 out = float4(y + 1.403 * uv.y, y - 0.344 * uv.x - 0.714 * uv.y, y + 1.770 * uv.x, 1.0);
+
+      return half4(out);
+    });
+
+@implementation RTCMTLNV12Renderer {
+  // Textures.
+  CVMetalTextureCacheRef _textureCache;
+  id<MTLTexture> _yTexture;
+  id<MTLTexture> _CrCbTexture;
+}
+
+- (BOOL)addRenderingDestination:(__kindof MTKView *)view {
+  if ([super addRenderingDestination:view]) {
+    return [self initializeTextureCache];
+  }
+  return NO;
+}
+
+- (BOOL)initializeTextureCache {
+  CVReturn status = CVMetalTextureCacheCreate(kCFAllocatorDefault, nil, [self currentMetalDevice],
+                                              nil, &_textureCache);
+  if (status != kCVReturnSuccess) {
+    RTCLogError(@"Metal: Failed to initialize metal texture cache. Return status is %d", status);
+    return NO;
+  }
+
+  return YES;
+}
+
+- (NSString *)shaderSource {
+  return shaderSource;
+}
+
+- (void)getWidth:(nonnull int *)width
+          height:(nonnull int *)height
+       cropWidth:(nonnull int *)cropWidth
+      cropHeight:(nonnull int *)cropHeight
+           cropX:(nonnull int *)cropX
+           cropY:(nonnull int *)cropY
+         ofFrame:(nonnull RTC_OBJC_TYPE(RTCVideoFrame) *)frame {
+  RTC_OBJC_TYPE(RTCCVPixelBuffer) *pixelBuffer = (RTC_OBJC_TYPE(RTCCVPixelBuffer) *)frame.buffer;
+  *width = CVPixelBufferGetWidth(pixelBuffer.pixelBuffer);
+  *height = CVPixelBufferGetHeight(pixelBuffer.pixelBuffer);
+  *cropWidth = pixelBuffer.cropWidth;
+  *cropHeight = pixelBuffer.cropHeight;
+  *cropX = pixelBuffer.cropX;
+  *cropY = pixelBuffer.cropY;
+}
+
+- (BOOL)setupTexturesForFrame:(nonnull RTC_OBJC_TYPE(RTCVideoFrame) *)frame {
+  RTC_DCHECK([frame.buffer isKindOfClass:[RTC_OBJC_TYPE(RTCCVPixelBuffer) class]]);
+  if (![super setupTexturesForFrame:frame]) {
+    return NO;
+  }
+  CVPixelBufferRef pixelBuffer = ((RTC_OBJC_TYPE(RTCCVPixelBuffer) *)frame.buffer).pixelBuffer;
+
+  id<MTLTexture> lumaTexture = nil;
+  id<MTLTexture> chromaTexture = nil;
+  CVMetalTextureRef outTexture = nullptr;
+
+  // Luma (y) texture.
+  int lumaWidth = CVPixelBufferGetWidthOfPlane(pixelBuffer, 0);
+  int lumaHeight = CVPixelBufferGetHeightOfPlane(pixelBuffer, 0);
+
+  int indexPlane = 0;
+  CVReturn result = CVMetalTextureCacheCreateTextureFromImage(
+      kCFAllocatorDefault, _textureCache, pixelBuffer, nil, MTLPixelFormatR8Unorm, lumaWidth,
+      lumaHeight, indexPlane, &outTexture);
+
+  if (result == kCVReturnSuccess) {
+    lumaTexture = CVMetalTextureGetTexture(outTexture);
+  }
+
+  // Same as CFRelease except it can be passed NULL without crashing.
+  CVBufferRelease(outTexture);
+  outTexture = nullptr;
+
+  // Chroma (CrCb) texture.
+  indexPlane = 1;
+  result = CVMetalTextureCacheCreateTextureFromImage(
+      kCFAllocatorDefault, _textureCache, pixelBuffer, nil, MTLPixelFormatRG8Unorm, lumaWidth / 2,
+      lumaHeight / 2, indexPlane, &outTexture);
+  if (result == kCVReturnSuccess) {
+    chromaTexture = CVMetalTextureGetTexture(outTexture);
+  }
+  CVBufferRelease(outTexture);
+
+  if (lumaTexture != nil && chromaTexture != nil) {
+    _yTexture = lumaTexture;
+    _CrCbTexture = chromaTexture;
+    return YES;
+  }
+  return NO;
+}
+
+- (void)uploadTexturesToRenderEncoder:(id<MTLRenderCommandEncoder>)renderEncoder {
+  [renderEncoder setFragmentTexture:_yTexture atIndex:0];
+  [renderEncoder setFragmentTexture:_CrCbTexture atIndex:1];
+}
+
+- (void)dealloc {
+  if (_textureCache) {
+    CFRelease(_textureCache);
+  }
+}
+@end
diff --git a/third_party/libwebrtc/sdk/objc/components/renderer/metal/RTCMTLRGBRenderer.h b/third_party/libwebrtc/sdk/objc/components/renderer/metal/RTCMTLRGBRenderer.h
new file mode 100644
index 0000000000..9db422cd22
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/components/renderer/metal/RTCMTLRGBRenderer.h
@@ -0,0 +1,22 @@
+/*
+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMTLRenderer.h"
+
+/** @abstract RGB/BGR renderer.
+ *  @discussion This renderer handles both kCVPixelFormatType_32BGRA and
+ * kCVPixelFormatType_32ARGB.
+ */
+NS_AVAILABLE(10_11, 9_0)
+@interface RTCMTLRGBRenderer : RTCMTLRenderer
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/components/renderer/metal/RTCMTLRGBRenderer.mm b/third_party/libwebrtc/sdk/objc/components/renderer/metal/RTCMTLRGBRenderer.mm
new file mode 100644
index 0000000000..e5dc4ef80a
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/components/renderer/metal/RTCMTLRGBRenderer.mm
@@ -0,0 +1,164 @@
+/*
+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCMTLRGBRenderer.h"
+
+#import <Metal/Metal.h>
+#import <MetalKit/MetalKit.h>
+
+#import "RTCMTLRenderer+Private.h"
+#import "base/RTCLogging.h"
+#import "base/RTCVideoFrame.h"
+#import "base/RTCVideoFrameBuffer.h"
+#import "components/video_frame_buffer/RTCCVPixelBuffer.h"
+
+#include "rtc_base/checks.h"
+
+static NSString *const shaderSource = MTL_STRINGIFY(
+    using namespace metal;
+
+    typedef struct {
+      packed_float2 position;
+      packed_float2 texcoord;
+    } Vertex;
+
+    typedef struct {
+      float4 position[[position]];
+      float2 texcoord;
+    } VertexIO;
+
+    vertex VertexIO vertexPassthrough(constant Vertex *verticies[[buffer(0)]],
+                                      uint vid[[vertex_id]]) {
+      VertexIO out;
+      constant Vertex &v = verticies[vid];
+      out.position = float4(float2(v.position), 0.0, 1.0);
+      out.texcoord = v.texcoord;
+      return out;
+    }
+
+    fragment half4 fragmentColorConversion(VertexIO in[[stage_in]],
+                                           texture2d<half, access::sample> texture[[texture(0)]],
+                                           constant bool &isARGB[[buffer(0)]]) {
+      constexpr sampler s(address::clamp_to_edge, filter::linear);
+
+      half4 out = texture.sample(s, in.texcoord);
+      if (isARGB) {
+        out = half4(out.g, out.b, out.a, out.r);
+      }
+
+      return out;
+    });
+
+@implementation RTCMTLRGBRenderer {
+  // Textures.
+  CVMetalTextureCacheRef _textureCache;
+  id<MTLTexture> _texture;
+
+  // Uniforms.
+  id<MTLBuffer> _uniformsBuffer;
+}
+
+- (BOOL)addRenderingDestination:(__kindof MTKView *)view {
+  if ([super addRenderingDestination:view]) {
+    return [self initializeTextureCache];
+  }
+  return NO;
+}
+
+- (BOOL)initializeTextureCache {
+  CVReturn status = CVMetalTextureCacheCreate(kCFAllocatorDefault, nil, [self currentMetalDevice],
+                                              nil, &_textureCache);
+  if (status != kCVReturnSuccess) {
+    RTCLogError(@"Metal: Failed to initialize metal texture cache. Return status is %d", status);
+    return NO;
+  }
+
+  return YES;
+}
+
+- (NSString *)shaderSource {
+  return shaderSource;
+}
+
+- (void)getWidth:(nonnull int *)width
+          height:(nonnull int *)height
+       cropWidth:(nonnull int *)cropWidth
+      cropHeight:(nonnull int *)cropHeight
+           cropX:(nonnull int *)cropX
+           cropY:(nonnull int *)cropY
+         ofFrame:(nonnull RTC_OBJC_TYPE(RTCVideoFrame) *)frame {
+  RTC_OBJC_TYPE(RTCCVPixelBuffer) *pixelBuffer = (RTC_OBJC_TYPE(RTCCVPixelBuffer) *)frame.buffer;
+  *width = CVPixelBufferGetWidth(pixelBuffer.pixelBuffer);
+  *height = CVPixelBufferGetHeight(pixelBuffer.pixelBuffer);
+  *cropWidth = pixelBuffer.cropWidth;
+  *cropHeight = pixelBuffer.cropHeight;
+  *cropX = pixelBuffer.cropX;
+  *cropY = pixelBuffer.cropY;
+}
+
+- (BOOL)setupTexturesForFrame:(nonnull RTC_OBJC_TYPE(RTCVideoFrame) *)frame {
+  RTC_DCHECK([frame.buffer isKindOfClass:[RTC_OBJC_TYPE(RTCCVPixelBuffer) class]]);
+  if (![super setupTexturesForFrame:frame]) {
+    return NO;
+  }
+  CVPixelBufferRef pixelBuffer = ((RTC_OBJC_TYPE(RTCCVPixelBuffer) *)frame.buffer).pixelBuffer;
+
+  id<MTLTexture> gpuTexture = nil;
+  CVMetalTextureRef textureOut = nullptr;
+  bool isARGB;
+
+  int width = CVPixelBufferGetWidth(pixelBuffer);
+  int height = CVPixelBufferGetHeight(pixelBuffer);
+  OSType pixelFormat = CVPixelBufferGetPixelFormatType(pixelBuffer);
+
+  MTLPixelFormat mtlPixelFormat;
+  if (pixelFormat == kCVPixelFormatType_32BGRA) {
+    mtlPixelFormat = MTLPixelFormatBGRA8Unorm;
+    isARGB = false;
+  } else if (pixelFormat == kCVPixelFormatType_32ARGB) {
+    mtlPixelFormat = MTLPixelFormatRGBA8Unorm;
+    isARGB = true;
+  } else {
+    RTC_DCHECK_NOTREACHED();
+    return NO;
+  }
+
+  CVReturn result = CVMetalTextureCacheCreateTextureFromImage(
+                kCFAllocatorDefault, _textureCache, pixelBuffer, nil, mtlPixelFormat,
+                width, height, 0, &textureOut);
+  if (result == kCVReturnSuccess) {
+    gpuTexture = CVMetalTextureGetTexture(textureOut);
+  }
+  CVBufferRelease(textureOut);
+
+  if (gpuTexture != nil) {
+    _texture = gpuTexture;
+    _uniformsBuffer =
+        [[self currentMetalDevice] newBufferWithBytes:&isARGB
+                                               length:sizeof(isARGB)
+                                              options:MTLResourceCPUCacheModeDefaultCache];
+    return YES;
+  }
+
+  return NO;
+}
+
+- (void)uploadTexturesToRenderEncoder:(id<MTLRenderCommandEncoder>)renderEncoder {
+  [renderEncoder setFragmentTexture:_texture atIndex:0];
+  [renderEncoder setFragmentBuffer:_uniformsBuffer offset:0 atIndex:0];
+}
+
+- (void)dealloc {
+  if (_textureCache) {
+    CFRelease(_textureCache);
+  }
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/components/renderer/metal/RTCMTLRenderer+Private.h b/third_party/libwebrtc/sdk/objc/components/renderer/metal/RTCMTLRenderer+Private.h
new file mode 100644
index 0000000000..916d4d4430
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/components/renderer/metal/RTCMTLRenderer+Private.h
@@ -0,0 +1,33 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Metal/Metal.h>
+
+#import "RTCMTLRenderer.h"
+
+#define MTL_STRINGIFY(s) @ #s
+
+NS_ASSUME_NONNULL_BEGIN
+
+@interface RTCMTLRenderer (Private)
+- (nullable id<MTLDevice>)currentMetalDevice;
+- (NSString *)shaderSource;
+- (BOOL)setupTexturesForFrame:(nonnull RTC_OBJC_TYPE(RTCVideoFrame) *)frame;
+- (void)uploadTexturesToRenderEncoder:(id<MTLRenderCommandEncoder>)renderEncoder;
+- (void)getWidth:(nonnull int *)width
+          height:(nonnull int *)height
+       cropWidth:(nonnull int *)cropWidth
+      cropHeight:(nonnull int *)cropHeight
+           cropX:(nonnull int *)cropX
+           cropY:(nonnull int *)cropY
+         ofFrame:(nonnull RTC_OBJC_TYPE(RTCVideoFrame) *)frame;
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/components/renderer/metal/RTCMTLRenderer.h b/third_party/libwebrtc/sdk/objc/components/renderer/metal/RTCMTLRenderer.h
new file mode 100644
index 0000000000..aa31545973
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/components/renderer/metal/RTCMTLRenderer.h
@@ -0,0 +1,61 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+#if TARGET_OS_IPHONE
+#import <UIKit/UIKit.h>
+#else
+#import <AppKit/AppKit.h>
+#endif
+
+#import "base/RTCVideoFrame.h"
+
+NS_ASSUME_NONNULL_BEGIN
+/**
+ * Protocol defining ability to render RTCVideoFrame in Metal enabled views.
+ */
+@protocol RTCMTLRenderer <NSObject>
+
+/**
+ * Method to be implemented to perform actual rendering of the provided frame.
+ *
+ * @param frame The frame to be rendered.
+ */
+- (void)drawFrame:(RTC_OBJC_TYPE(RTCVideoFrame) *)frame;
+
+/**
+ * Sets the provided view as rendering destination if possible.
+ *
+ * If not possible method returns NO and callers of the method are responisble for performing
+ * cleanups.
+ */
+
+#if TARGET_OS_IOS
+- (BOOL)addRenderingDestination:(__kindof UIView *)view;
+#else
+- (BOOL)addRenderingDestination:(__kindof NSView *)view;
+#endif
+
+@end
+
+/**
+ * Implementation of RTCMTLRenderer protocol.
+ */
+NS_AVAILABLE(10_11, 9_0)
+@interface RTCMTLRenderer : NSObject <RTCMTLRenderer>
+
+/** @abstract   A wrapped RTCVideoRotation, or nil.
+    @discussion When not nil, the rotation of the actual frame is ignored when rendering.
+ */
+@property(atomic, nullable) NSValue *rotationOverride;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/components/renderer/metal/RTCMTLRenderer.mm b/third_party/libwebrtc/sdk/objc/components/renderer/metal/RTCMTLRenderer.mm
new file mode 100644
index 0000000000..410590a7b1
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/components/renderer/metal/RTCMTLRenderer.mm
@@ -0,0 +1,328 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCMTLRenderer+Private.h"
+
+#import <Metal/Metal.h>
+#import <MetalKit/MetalKit.h>
+
+#import "base/RTCLogging.h"
+#import "base/RTCVideoFrame.h"
+#import "base/RTCVideoFrameBuffer.h"
+
+#include "api/video/video_rotation.h"
+#include "rtc_base/checks.h"
+
+// As defined in shaderSource.
+static NSString *const vertexFunctionName = @"vertexPassthrough";
+static NSString *const fragmentFunctionName = @"fragmentColorConversion";
+
+static NSString *const pipelineDescriptorLabel = @"RTCPipeline";
+static NSString *const commandBufferLabel = @"RTCCommandBuffer";
+static NSString *const renderEncoderLabel = @"RTCEncoder";
+static NSString *const renderEncoderDebugGroup = @"RTCDrawFrame";
+
+// Computes the texture coordinates given rotation and cropping.
+static inline void getCubeVertexData(int cropX,
+                                     int cropY,
+                                     int cropWidth,
+                                     int cropHeight,
+                                     size_t frameWidth,
+                                     size_t frameHeight,
+                                     RTCVideoRotation rotation,
+                                     float *buffer) {
+  // The computed values are the adjusted texture coordinates, in [0..1].
+  // For the left and top, 0.0 means no cropping and e.g. 0.2 means we're skipping 20% of the
+  // left/top edge.
+  // For the right and bottom, 1.0 means no cropping and e.g. 0.8 means we're skipping 20% of the
+  // right/bottom edge (i.e. render up to 80% of the width/height).
+  float cropLeft = cropX / (float)frameWidth;
+  float cropRight = (cropX + cropWidth) / (float)frameWidth;
+  float cropTop = cropY / (float)frameHeight;
+  float cropBottom = (cropY + cropHeight) / (float)frameHeight;
+
+  // These arrays map the view coordinates to texture coordinates, taking cropping and rotation
+  // into account. The first two columns are view coordinates, the last two are texture coordinates.
+  switch (rotation) {
+    case RTCVideoRotation_0: {
+      float values[16] = {-1.0, -1.0, cropLeft, cropBottom,
+                           1.0, -1.0, cropRight, cropBottom,
+                          -1.0,  1.0, cropLeft, cropTop,
+                           1.0,  1.0, cropRight, cropTop};
+      memcpy(buffer, &values, sizeof(values));
+    } break;
+    case RTCVideoRotation_90: {
+      float values[16] = {-1.0, -1.0, cropRight, cropBottom,
+                           1.0, -1.0, cropRight, cropTop,
+                          -1.0,  1.0, cropLeft, cropBottom,
+                           1.0,  1.0, cropLeft, cropTop};
+      memcpy(buffer, &values, sizeof(values));
+    } break;
+    case RTCVideoRotation_180: {
+      float values[16] = {-1.0, -1.0, cropRight, cropTop,
+                           1.0, -1.0, cropLeft, cropTop,
+                          -1.0,  1.0, cropRight, cropBottom,
+                           1.0,  1.0, cropLeft, cropBottom};
+      memcpy(buffer, &values, sizeof(values));
+    } break;
+    case RTCVideoRotation_270: {
+      float values[16] = {-1.0, -1.0, cropLeft, cropTop,
+                           1.0, -1.0, cropLeft, cropBottom,
+                          -1.0, 1.0, cropRight, cropTop,
+                           1.0, 1.0, cropRight, cropBottom};
+      memcpy(buffer, &values, sizeof(values));
+    } break;
+  }
+}
+
+// The max number of command buffers in flight (submitted to GPU).
+// For now setting it up to 1.
+// In future we might use triple buffering method if it improves performance.
+static const NSInteger kMaxInflightBuffers = 1;
+
+@implementation RTCMTLRenderer {
+  __kindof MTKView *_view;
+
+  // Controller.
+  dispatch_semaphore_t _inflight_semaphore;
+
+  // Renderer.
+  id<MTLDevice> _device;
+  id<MTLCommandQueue> _commandQueue;
+  id<MTLLibrary> _defaultLibrary;
+  id<MTLRenderPipelineState> _pipelineState;
+
+  // Buffers.
+  id<MTLBuffer> _vertexBuffer;
+
+  // Values affecting the vertex buffer. Stored for comparison to avoid unnecessary recreation.
+  int _oldFrameWidth;
+  int _oldFrameHeight;
+  int _oldCropWidth;
+  int _oldCropHeight;
+  int _oldCropX;
+  int _oldCropY;
+  RTCVideoRotation _oldRotation;
+}
+
+@synthesize rotationOverride = _rotationOverride;
+
+- (instancetype)init {
+  if (self = [super init]) {
+    _inflight_semaphore = dispatch_semaphore_create(kMaxInflightBuffers);
+  }
+
+  return self;
+}
+
+- (BOOL)addRenderingDestination:(__kindof MTKView *)view {
+  return [self setupWithView:view];
+}
+
+#pragma mark - Private
+
+- (BOOL)setupWithView:(__kindof MTKView *)view {
+  BOOL success = NO;
+  if ([self setupMetal]) {
+    _view = view;
+    view.device = _device;
+    view.preferredFramesPerSecond = 30;
+    view.autoResizeDrawable = NO;
+
+    [self loadAssets];
+
+    float vertexBufferArray[16] = {0};
+    _vertexBuffer = [_device newBufferWithBytes:vertexBufferArray
+                                         length:sizeof(vertexBufferArray)
+                                        options:MTLResourceCPUCacheModeWriteCombined];
+    success = YES;
+  }
+  return success;
+}
+#pragma mark - Inheritance
+
+- (id<MTLDevice>)currentMetalDevice {
+  return _device;
+}
+
+- (NSString *)shaderSource {
+  RTC_DCHECK_NOTREACHED() << "Virtual method not implemented in subclass.";
+  return nil;
+}
+
+- (void)uploadTexturesToRenderEncoder:(id<MTLRenderCommandEncoder>)renderEncoder {
+  RTC_DCHECK_NOTREACHED() << "Virtual method not implemented in subclass.";
+}
+
+- (void)getWidth:(int *)width
+          height:(int *)height
+       cropWidth:(int *)cropWidth
+      cropHeight:(int *)cropHeight
+           cropX:(int *)cropX
+           cropY:(int *)cropY
+         ofFrame:(nonnull RTC_OBJC_TYPE(RTCVideoFrame) *)frame {
+  RTC_DCHECK_NOTREACHED() << "Virtual method not implemented in subclass.";
+}
+
+- (BOOL)setupTexturesForFrame:(nonnull RTC_OBJC_TYPE(RTCVideoFrame) *)frame {
+  // Apply rotation override if set.
+  RTCVideoRotation rotation;
+  NSValue *rotationOverride = self.rotationOverride;
+  if (rotationOverride) {
+#if defined(__IPHONE_11_0) && defined(__IPHONE_OS_VERSION_MAX_ALLOWED) && \
+    (__IPHONE_OS_VERSION_MAX_ALLOWED >= __IPHONE_11_0)
+    if (@available(iOS 11, *)) {
+      [rotationOverride getValue:&rotation size:sizeof(rotation)];
+    } else
+#endif
+    {
+      [rotationOverride getValue:&rotation];
+    }
+  } else {
+    rotation = frame.rotation;
+  }
+
+  int frameWidth, frameHeight, cropWidth, cropHeight, cropX, cropY;
+  [self getWidth:&frameWidth
+          height:&frameHeight
+       cropWidth:&cropWidth
+      cropHeight:&cropHeight
+           cropX:&cropX
+           cropY:&cropY
+         ofFrame:frame];
+
+  // Recompute the texture cropping and recreate vertexBuffer if necessary.
+  if (cropX != _oldCropX || cropY != _oldCropY || cropWidth != _oldCropWidth ||
+      cropHeight != _oldCropHeight || rotation != _oldRotation || frameWidth != _oldFrameWidth ||
+      frameHeight != _oldFrameHeight) {
+    getCubeVertexData(cropX,
+                      cropY,
+                      cropWidth,
+                      cropHeight,
+                      frameWidth,
+                      frameHeight,
+                      rotation,
+                      (float *)_vertexBuffer.contents);
+    _oldCropX = cropX;
+    _oldCropY = cropY;
+    _oldCropWidth = cropWidth;
+    _oldCropHeight = cropHeight;
+    _oldRotation = rotation;
+    _oldFrameWidth = frameWidth;
+    _oldFrameHeight = frameHeight;
+  }
+
+  return YES;
+}
+
+#pragma mark - GPU methods
+
+- (BOOL)setupMetal {
+  // Set the view to use the default device.
+  _device = MTLCreateSystemDefaultDevice();
+  if (!_device) {
+    return NO;
+  }
+
+  // Create a new command queue.
+  _commandQueue = [_device newCommandQueue];
+
+  // Load metal library from source.
+  NSError *libraryError = nil;
+  NSString *shaderSource = [self shaderSource];
+
+  id<MTLLibrary> sourceLibrary =
+      [_device newLibraryWithSource:shaderSource options:NULL error:&libraryError];
+
+  if (libraryError) {
+    RTCLogError(@"Metal: Library with source failed\n%@", libraryError);
+    return NO;
+  }
+
+  if (!sourceLibrary) {
+    RTCLogError(@"Metal: Failed to load library. %@", libraryError);
+    return NO;
+  }
+  _defaultLibrary = sourceLibrary;
+
+  return YES;
+}
+
+- (void)loadAssets {
+  id<MTLFunction> vertexFunction = [_defaultLibrary newFunctionWithName:vertexFunctionName];
+  id<MTLFunction> fragmentFunction = [_defaultLibrary newFunctionWithName:fragmentFunctionName];
+
+  MTLRenderPipelineDescriptor *pipelineDescriptor = [[MTLRenderPipelineDescriptor alloc] init];
+  pipelineDescriptor.label = pipelineDescriptorLabel;
+  pipelineDescriptor.vertexFunction = vertexFunction;
+  pipelineDescriptor.fragmentFunction = fragmentFunction;
+  pipelineDescriptor.colorAttachments[0].pixelFormat = _view.colorPixelFormat;
+  pipelineDescriptor.depthAttachmentPixelFormat = MTLPixelFormatInvalid;
+  NSError *error = nil;
+  _pipelineState = [_device newRenderPipelineStateWithDescriptor:pipelineDescriptor error:&error];
+
+  if (!_pipelineState) {
+    RTCLogError(@"Metal: Failed to create pipeline state. %@", error);
+  }
+}
+
+- (void)render {
+  id<MTLCommandBuffer> commandBuffer = [_commandQueue commandBuffer];
+  commandBuffer.label = commandBufferLabel;
+
+  __block dispatch_semaphore_t block_semaphore = _inflight_semaphore;
+  [commandBuffer addCompletedHandler:^(id<MTLCommandBuffer> _Nonnull) {
+    // GPU work completed.
+    dispatch_semaphore_signal(block_semaphore);
+  }];
+
+  MTLRenderPassDescriptor *renderPassDescriptor = _view.currentRenderPassDescriptor;
+  if (renderPassDescriptor) {  // Valid drawable.
+    id<MTLRenderCommandEncoder> renderEncoder =
+        [commandBuffer renderCommandEncoderWithDescriptor:renderPassDescriptor];
+    renderEncoder.label = renderEncoderLabel;
+
+    // Set context state.
+    [renderEncoder pushDebugGroup:renderEncoderDebugGroup];
+    [renderEncoder setRenderPipelineState:_pipelineState];
+    [renderEncoder setVertexBuffer:_vertexBuffer offset:0 atIndex:0];
+    [self uploadTexturesToRenderEncoder:renderEncoder];
+
+    [renderEncoder drawPrimitives:MTLPrimitiveTypeTriangleStrip
+                      vertexStart:0
+                      vertexCount:4
+                    instanceCount:1];
+    [renderEncoder popDebugGroup];
+    [renderEncoder endEncoding];
+
+    [commandBuffer presentDrawable:_view.currentDrawable];
+  }
+
+  // CPU work is completed, GPU work can be started.
+  [commandBuffer commit];
+}
+
+#pragma mark - RTCMTLRenderer
+
+- (void)drawFrame:(RTC_OBJC_TYPE(RTCVideoFrame) *)frame {
+  @autoreleasepool {
+    // Wait until the inflight (curently sent to GPU) command buffer
+    // has completed the GPU work.
+    dispatch_semaphore_wait(_inflight_semaphore, DISPATCH_TIME_FOREVER);
+
+    if ([self setupTexturesForFrame:frame]) {
+      [self render];
+    } else {
+      dispatch_semaphore_signal(_inflight_semaphore);
+    }
+  }
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/components/renderer/metal/RTCMTLVideoView.h b/third_party/libwebrtc/sdk/objc/components/renderer/metal/RTCMTLVideoView.h
new file mode 100644
index 0000000000..3320d12076
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/components/renderer/metal/RTCMTLVideoView.h
@@ -0,0 +1,44 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+#import "RTCVideoFrame.h"
+#import "RTCVideoRenderer.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+/**
+ * RTCMTLVideoView is thin wrapper around MTKView.
+ *
+ * It has id<RTCVideoRenderer> property that renders video frames in the view's
+ * bounds using Metal.
+ */
+NS_CLASS_AVAILABLE_IOS(9)
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCMTLVideoView) : UIView<RTC_OBJC_TYPE(RTCVideoRenderer)>
+
+@property(nonatomic, weak) id<RTC_OBJC_TYPE(RTCVideoViewDelegate)> delegate;
+
+@property(nonatomic) UIViewContentMode videoContentMode;
+
+/** @abstract Enables/disables rendering.
+ */
+@property(nonatomic, getter=isEnabled) BOOL enabled;
+
+/** @abstract Wrapped RTCVideoRotation, or nil.
+ */
+@property(nonatomic, nullable) NSValue* rotationOverride;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/components/renderer/metal/RTCMTLVideoView.m b/third_party/libwebrtc/sdk/objc/components/renderer/metal/RTCMTLVideoView.m
new file mode 100644
index 0000000000..c5d9e4385f
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/components/renderer/metal/RTCMTLVideoView.m
@@ -0,0 +1,265 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCMTLVideoView.h"
+
+#import <Metal/Metal.h>
+#import <MetalKit/MetalKit.h>
+
+#import "base/RTCLogging.h"
+#import "base/RTCVideoFrame.h"
+#import "base/RTCVideoFrameBuffer.h"
+#import "components/video_frame_buffer/RTCCVPixelBuffer.h"
+
+#import "RTCMTLI420Renderer.h"
+#import "RTCMTLNV12Renderer.h"
+#import "RTCMTLRGBRenderer.h"
+
+// To avoid unreconized symbol linker errors, we're taking advantage of the objc runtime.
+// Linking errors occur when compiling for architectures that don't support Metal.
+#define MTKViewClass NSClassFromString(@"MTKView")
+#define RTCMTLNV12RendererClass NSClassFromString(@"RTCMTLNV12Renderer")
+#define RTCMTLI420RendererClass NSClassFromString(@"RTCMTLI420Renderer")
+#define RTCMTLRGBRendererClass NSClassFromString(@"RTCMTLRGBRenderer")
+
+@interface RTC_OBJC_TYPE (RTCMTLVideoView)
+()<MTKViewDelegate> @property(nonatomic) RTCMTLI420Renderer *rendererI420;
+@property(nonatomic) RTCMTLNV12Renderer *rendererNV12;
+@property(nonatomic) RTCMTLRGBRenderer *rendererRGB;
+@property(nonatomic) MTKView *metalView;
+@property(atomic) RTC_OBJC_TYPE(RTCVideoFrame) * videoFrame;
+@property(nonatomic) CGSize videoFrameSize;
+@property(nonatomic) int64_t lastFrameTimeNs;
+@end
+
+@implementation RTC_OBJC_TYPE (RTCMTLVideoView)
+
+@synthesize delegate = _delegate;
+@synthesize rendererI420 = _rendererI420;
+@synthesize rendererNV12 = _rendererNV12;
+@synthesize rendererRGB = _rendererRGB;
+@synthesize metalView = _metalView;
+@synthesize videoFrame = _videoFrame;
+@synthesize videoFrameSize = _videoFrameSize;
+@synthesize lastFrameTimeNs = _lastFrameTimeNs;
+@synthesize rotationOverride = _rotationOverride;
+
+- (instancetype)initWithFrame:(CGRect)frameRect {
+  self = [super initWithFrame:frameRect];
+  if (self) {
+    [self configure];
+  }
+  return self;
+}
+
+- (instancetype)initWithCoder:(NSCoder *)aCoder {
+  self = [super initWithCoder:aCoder];
+  if (self) {
+    [self configure];
+  }
+  return self;
+}
+
+- (BOOL)isEnabled {
+  return !self.metalView.paused;
+}
+
+- (void)setEnabled:(BOOL)enabled {
+  self.metalView.paused = !enabled;
+}
+
+- (UIViewContentMode)videoContentMode {
+  return self.metalView.contentMode;
+}
+
+- (void)setVideoContentMode:(UIViewContentMode)mode {
+  self.metalView.contentMode = mode;
+}
+
+#pragma mark - Private
+
++ (BOOL)isMetalAvailable {
+  return MTLCreateSystemDefaultDevice() != nil;
+}
+
++ (MTKView *)createMetalView:(CGRect)frame {
+  return [[MTKViewClass alloc] initWithFrame:frame];
+}
+
++ (RTCMTLNV12Renderer *)createNV12Renderer {
+  return [[RTCMTLNV12RendererClass alloc] init];
+}
+
++ (RTCMTLI420Renderer *)createI420Renderer {
+  return [[RTCMTLI420RendererClass alloc] init];
+}
+
++ (RTCMTLRGBRenderer *)createRGBRenderer {
+  return [[RTCMTLRGBRenderer alloc] init];
+}
+
+- (void)configure {
+  NSAssert([RTC_OBJC_TYPE(RTCMTLVideoView) isMetalAvailable],
+           @"Metal not availiable on this device");
+
+  self.metalView = [RTC_OBJC_TYPE(RTCMTLVideoView) createMetalView:self.bounds];
+  self.metalView.delegate = self;
+  self.metalView.contentMode = UIViewContentModeScaleAspectFill;
+  [self addSubview:self.metalView];
+  self.videoFrameSize = CGSizeZero;
+}
+
+- (void)setMultipleTouchEnabled:(BOOL)multipleTouchEnabled {
+    [super setMultipleTouchEnabled:multipleTouchEnabled];
+    self.metalView.multipleTouchEnabled = multipleTouchEnabled;
+}
+
+- (void)layoutSubviews {
+  [super layoutSubviews];
+
+  CGRect bounds = self.bounds;
+  self.metalView.frame = bounds;
+  if (!CGSizeEqualToSize(self.videoFrameSize, CGSizeZero)) {
+    self.metalView.drawableSize = [self drawableSize];
+  } else {
+    self.metalView.drawableSize = bounds.size;
+  }
+}
+
+#pragma mark - MTKViewDelegate methods
+
+- (void)drawInMTKView:(nonnull MTKView *)view {
+  NSAssert(view == self.metalView, @"Receiving draw callbacks from foreign instance.");
+  RTC_OBJC_TYPE(RTCVideoFrame) *videoFrame = self.videoFrame;
+  // Skip rendering if we've already rendered this frame.
+  if (!videoFrame || videoFrame.width <= 0 || videoFrame.height <= 0 ||
+      videoFrame.timeStampNs == self.lastFrameTimeNs) {
+    return;
+  }
+
+  if (CGRectIsEmpty(view.bounds)) {
+    return;
+  }
+
+  RTCMTLRenderer *renderer;
+  if ([videoFrame.buffer isKindOfClass:[RTC_OBJC_TYPE(RTCCVPixelBuffer) class]]) {
+    RTC_OBJC_TYPE(RTCCVPixelBuffer) *buffer = (RTC_OBJC_TYPE(RTCCVPixelBuffer) *)videoFrame.buffer;
+    const OSType pixelFormat = CVPixelBufferGetPixelFormatType(buffer.pixelBuffer);
+    if (pixelFormat == kCVPixelFormatType_32BGRA || pixelFormat == kCVPixelFormatType_32ARGB) {
+      if (!self.rendererRGB) {
+        self.rendererRGB = [RTC_OBJC_TYPE(RTCMTLVideoView) createRGBRenderer];
+        if (![self.rendererRGB addRenderingDestination:self.metalView]) {
+          self.rendererRGB = nil;
+          RTCLogError(@"Failed to create RGB renderer");
+          return;
+        }
+      }
+      renderer = self.rendererRGB;
+    } else {
+      if (!self.rendererNV12) {
+        self.rendererNV12 = [RTC_OBJC_TYPE(RTCMTLVideoView) createNV12Renderer];
+        if (![self.rendererNV12 addRenderingDestination:self.metalView]) {
+          self.rendererNV12 = nil;
+          RTCLogError(@"Failed to create NV12 renderer");
+          return;
+        }
+      }
+      renderer = self.rendererNV12;
+    }
+  } else {
+    if (!self.rendererI420) {
+      self.rendererI420 = [RTC_OBJC_TYPE(RTCMTLVideoView) createI420Renderer];
+      if (![self.rendererI420 addRenderingDestination:self.metalView]) {
+        self.rendererI420 = nil;
+        RTCLogError(@"Failed to create I420 renderer");
+        return;
+      }
+    }
+    renderer = self.rendererI420;
+  }
+
+  renderer.rotationOverride = self.rotationOverride;
+
+  [renderer drawFrame:videoFrame];
+  self.lastFrameTimeNs = videoFrame.timeStampNs;
+}
+
+- (void)mtkView:(MTKView *)view drawableSizeWillChange:(CGSize)size {
+}
+
+#pragma mark -
+
+- (void)setRotationOverride:(NSValue *)rotationOverride {
+  _rotationOverride = rotationOverride;
+
+  self.metalView.drawableSize = [self drawableSize];
+  [self setNeedsLayout];
+}
+
+- (RTCVideoRotation)frameRotation {
+  if (self.rotationOverride) {
+    RTCVideoRotation rotation;
+    if (@available(iOS 11, *)) {
+      [self.rotationOverride getValue:&rotation size:sizeof(rotation)];
+    } else {
+      [self.rotationOverride getValue:&rotation];
+    }
+    return rotation;
+  }
+
+  return self.videoFrame.rotation;
+}
+
+- (CGSize)drawableSize {
+  // Flip width/height if the rotations are not the same.
+  CGSize videoFrameSize = self.videoFrameSize;
+  RTCVideoRotation frameRotation = [self frameRotation];
+
+  BOOL useLandscape =
+      (frameRotation == RTCVideoRotation_0) || (frameRotation == RTCVideoRotation_180);
+  BOOL sizeIsLandscape = (self.videoFrame.rotation == RTCVideoRotation_0) ||
+      (self.videoFrame.rotation == RTCVideoRotation_180);
+
+  if (useLandscape == sizeIsLandscape) {
+    return videoFrameSize;
+  } else {
+    return CGSizeMake(videoFrameSize.height, videoFrameSize.width);
+  }
+}
+
+#pragma mark - RTC_OBJC_TYPE(RTCVideoRenderer)
+
+- (void)setSize:(CGSize)size {
+  __weak RTC_OBJC_TYPE(RTCMTLVideoView) *weakSelf = self;
+  dispatch_async(dispatch_get_main_queue(), ^{
+    RTC_OBJC_TYPE(RTCMTLVideoView) *strongSelf = weakSelf;
+
+    strongSelf.videoFrameSize = size;
+    CGSize drawableSize = [strongSelf drawableSize];
+
+    strongSelf.metalView.drawableSize = drawableSize;
+    [strongSelf setNeedsLayout];
+    [strongSelf.delegate videoView:self didChangeVideoSize:size];
+  });
+}
+
+- (void)renderFrame:(nullable RTC_OBJC_TYPE(RTCVideoFrame) *)frame {
+  if (!self.isEnabled) {
+    return;
+  }
+
+  if (frame == nil) {
+    RTCLogInfo(@"Incoming frame is nil. Exiting render callback.");
+    return;
+  }
+  self.videoFrame = frame;
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/components/renderer/opengl/RTCDefaultShader.h b/third_party/libwebrtc/sdk/objc/components/renderer/opengl/RTCDefaultShader.h
new file mode 100644
index 0000000000..71a073ab21
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/components/renderer/opengl/RTCDefaultShader.h
@@ -0,0 +1,23 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCVideoViewShading.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+/** Default RTCVideoViewShading that will be used in RTCNSGLVideoView
+ *  and RTCEAGLVideoView if no external shader is specified. This shader will render
+ *  the video in a rectangle without any color or geometric transformations.
+ */
+@interface RTCDefaultShader : NSObject <RTC_OBJC_TYPE (RTCVideoViewShading)>
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/components/renderer/opengl/RTCDefaultShader.mm b/third_party/libwebrtc/sdk/objc/components/renderer/opengl/RTCDefaultShader.mm
new file mode 100644
index 0000000000..51dca3223d
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/components/renderer/opengl/RTCDefaultShader.mm
@@ -0,0 +1,207 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCDefaultShader.h"
+
+#if TARGET_OS_IPHONE
+#import <OpenGLES/ES3/gl.h>
+#else
+#import <OpenGL/gl3.h>
+#endif
+
+#import "RTCOpenGLDefines.h"
+#import "RTCShader.h"
+#import "base/RTCLogging.h"
+
+#include "absl/types/optional.h"
+
+static const int kYTextureUnit = 0;
+static const int kUTextureUnit = 1;
+static const int kVTextureUnit = 2;
+static const int kUvTextureUnit = 1;
+
+// Fragment shader converts YUV values from input textures into a final RGB
+// pixel. The conversion formula is from http://www.fourcc.org/fccyvrgb.php.
+static const char kI420FragmentShaderSource[] =
+  SHADER_VERSION
+  "precision highp float;"
+  FRAGMENT_SHADER_IN " vec2 v_texcoord;\n"
+  "uniform lowp sampler2D s_textureY;\n"
+  "uniform lowp sampler2D s_textureU;\n"
+  "uniform lowp sampler2D s_textureV;\n"
+  FRAGMENT_SHADER_OUT
+  "void main() {\n"
+  "    float y, u, v, r, g, b;\n"
+  "    y = " FRAGMENT_SHADER_TEXTURE "(s_textureY, v_texcoord).r;\n"
+  "    u = " FRAGMENT_SHADER_TEXTURE "(s_textureU, v_texcoord).r;\n"
+  "    v = " FRAGMENT_SHADER_TEXTURE "(s_textureV, v_texcoord).r;\n"
+  "    u = u - 0.5;\n"
+  "    v = v - 0.5;\n"
+  "    r = y + 1.403 * v;\n"
+  "    g = y - 0.344 * u - 0.714 * v;\n"
+  "    b = y + 1.770 * u;\n"
+  "    " FRAGMENT_SHADER_COLOR " = vec4(r, g, b, 1.0);\n"
+  "  }\n";
+
+static const char kNV12FragmentShaderSource[] =
+  SHADER_VERSION
+  "precision mediump float;"
+  FRAGMENT_SHADER_IN " vec2 v_texcoord;\n"
+  "uniform lowp sampler2D s_textureY;\n"
+  "uniform lowp sampler2D s_textureUV;\n"
+  FRAGMENT_SHADER_OUT
+  "void main() {\n"
+  "    mediump float y;\n"
+  "    mediump vec2 uv;\n"
+  "    y = " FRAGMENT_SHADER_TEXTURE "(s_textureY, v_texcoord).r;\n"
+  "    uv = " FRAGMENT_SHADER_TEXTURE "(s_textureUV, v_texcoord).ra -\n"
+  "        vec2(0.5, 0.5);\n"
+  "    " FRAGMENT_SHADER_COLOR " = vec4(y + 1.403 * uv.y,\n"
+  "                                     y - 0.344 * uv.x - 0.714 * uv.y,\n"
+  "                                     y + 1.770 * uv.x,\n"
+  "                                     1.0);\n"
+  "  }\n";
+
+@implementation RTCDefaultShader {
+  GLuint _vertexBuffer;
+  GLuint _vertexArray;
+  // Store current rotation and only upload new vertex data when rotation changes.
+  absl::optional<RTCVideoRotation> _currentRotation;
+
+  GLuint _i420Program;
+  GLuint _nv12Program;
+}
+
+- (void)dealloc {
+  glDeleteProgram(_i420Program);
+  glDeleteProgram(_nv12Program);
+  glDeleteBuffers(1, &_vertexBuffer);
+  glDeleteVertexArrays(1, &_vertexArray);
+}
+
+- (BOOL)createAndSetupI420Program {
+  NSAssert(!_i420Program, @"I420 program already created");
+  _i420Program = RTCCreateProgramFromFragmentSource(kI420FragmentShaderSource);
+  if (!_i420Program) {
+    return NO;
+  }
+  GLint ySampler = glGetUniformLocation(_i420Program, "s_textureY");
+  GLint uSampler = glGetUniformLocation(_i420Program, "s_textureU");
+  GLint vSampler = glGetUniformLocation(_i420Program, "s_textureV");
+
+  if (ySampler < 0 || uSampler < 0 || vSampler < 0) {
+    RTCLog(@"Failed to get uniform variable locations in I420 shader");
+    glDeleteProgram(_i420Program);
+    _i420Program = 0;
+    return NO;
+  }
+
+  glUseProgram(_i420Program);
+  glUniform1i(ySampler, kYTextureUnit);
+  glUniform1i(uSampler, kUTextureUnit);
+  glUniform1i(vSampler, kVTextureUnit);
+
+  return YES;
+}
+
+- (BOOL)createAndSetupNV12Program {
+  NSAssert(!_nv12Program, @"NV12 program already created");
+  _nv12Program = RTCCreateProgramFromFragmentSource(kNV12FragmentShaderSource);
+  if (!_nv12Program) {
+    return NO;
+  }
+  GLint ySampler = glGetUniformLocation(_nv12Program, "s_textureY");
+  GLint uvSampler = glGetUniformLocation(_nv12Program, "s_textureUV");
+
+  if (ySampler < 0 || uvSampler < 0) {
+    RTCLog(@"Failed to get uniform variable locations in NV12 shader");
+    glDeleteProgram(_nv12Program);
+    _nv12Program = 0;
+    return NO;
+  }
+
+  glUseProgram(_nv12Program);
+  glUniform1i(ySampler, kYTextureUnit);
+  glUniform1i(uvSampler, kUvTextureUnit);
+
+  return YES;
+}
+
+- (BOOL)prepareVertexBufferWithRotation:(RTCVideoRotation)rotation {
+  if (!_vertexBuffer && !RTCCreateVertexBuffer(&_vertexBuffer, &_vertexArray)) {
+    RTCLog(@"Failed to setup vertex buffer");
+    return NO;
+  }
+#if !TARGET_OS_IPHONE
+  glBindVertexArray(_vertexArray);
+#endif
+  glBindBuffer(GL_ARRAY_BUFFER, _vertexBuffer);
+  if (!_currentRotation || rotation != *_currentRotation) {
+    _currentRotation = absl::optional<RTCVideoRotation>(rotation);
+    RTCSetVertexData(*_currentRotation);
+  }
+  return YES;
+}
+
+- (void)applyShadingForFrameWithWidth:(int)width
+                               height:(int)height
+                             rotation:(RTCVideoRotation)rotation
+                               yPlane:(GLuint)yPlane
+                               uPlane:(GLuint)uPlane
+                               vPlane:(GLuint)vPlane {
+  if (![self prepareVertexBufferWithRotation:rotation]) {
+    return;
+  }
+
+  if (!_i420Program && ![self createAndSetupI420Program]) {
+    RTCLog(@"Failed to setup I420 program");
+    return;
+  }
+
+  glUseProgram(_i420Program);
+
+  glActiveTexture(static_cast<GLenum>(GL_TEXTURE0 + kYTextureUnit));
+  glBindTexture(GL_TEXTURE_2D, yPlane);
+
+  glActiveTexture(static_cast<GLenum>(GL_TEXTURE0 + kUTextureUnit));
+  glBindTexture(GL_TEXTURE_2D, uPlane);
+
+  glActiveTexture(static_cast<GLenum>(GL_TEXTURE0 + kVTextureUnit));
+  glBindTexture(GL_TEXTURE_2D, vPlane);
+
+  glDrawArrays(GL_TRIANGLE_FAN, 0, 4);
+}
+
+- (void)applyShadingForFrameWithWidth:(int)width
+                               height:(int)height
+                             rotation:(RTCVideoRotation)rotation
+                               yPlane:(GLuint)yPlane
+                              uvPlane:(GLuint)uvPlane {
+  if (![self prepareVertexBufferWithRotation:rotation]) {
+    return;
+  }
+
+  if (!_nv12Program && ![self createAndSetupNV12Program]) {
+    RTCLog(@"Failed to setup NV12 shader");
+    return;
+  }
+
+  glUseProgram(_nv12Program);
+
+  glActiveTexture(static_cast<GLenum>(GL_TEXTURE0 + kYTextureUnit));
+  glBindTexture(GL_TEXTURE_2D, yPlane);
+
+  glActiveTexture(static_cast<GLenum>(GL_TEXTURE0 + kUvTextureUnit));
+  glBindTexture(GL_TEXTURE_2D, uvPlane);
+
+  glDrawArrays(GL_TRIANGLE_FAN, 0, 4);
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/components/renderer/opengl/RTCDisplayLinkTimer.h b/third_party/libwebrtc/sdk/objc/components/renderer/opengl/RTCDisplayLinkTimer.h
new file mode 100644
index 0000000000..b78501e9e6
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/components/renderer/opengl/RTCDisplayLinkTimer.h
@@ -0,0 +1,24 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+// RTCDisplayLinkTimer wraps a CADisplayLink and is set to fire every two screen
+// refreshes, which should be 30fps. We wrap the display link in order to avoid
+// a retain cycle since CADisplayLink takes a strong reference onto its target.
+// The timer is paused by default.
+@interface RTCDisplayLinkTimer : NSObject
+
+@property(nonatomic) BOOL isPaused;
+
+- (instancetype)initWithTimerHandler:(void (^)(void))timerHandler;
+- (void)invalidate;
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/components/renderer/opengl/RTCDisplayLinkTimer.m b/third_party/libwebrtc/sdk/objc/components/renderer/opengl/RTCDisplayLinkTimer.m
new file mode 100644
index 0000000000..906bb898d6
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/components/renderer/opengl/RTCDisplayLinkTimer.m
@@ -0,0 +1,59 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCDisplayLinkTimer.h"
+
+#import <UIKit/UIKit.h>
+
+@implementation RTCDisplayLinkTimer {
+  CADisplayLink *_displayLink;
+  void (^_timerHandler)(void);
+}
+
+- (instancetype)initWithTimerHandler:(void (^)(void))timerHandler {
+  NSParameterAssert(timerHandler);
+  if (self = [super init]) {
+    _timerHandler = timerHandler;
+    _displayLink =
+        [CADisplayLink displayLinkWithTarget:self
+                                    selector:@selector(displayLinkDidFire:)];
+    _displayLink.paused = YES;
+#if __IPHONE_OS_VERSION_MIN_REQUIRED >= __IPHONE_10_0
+    _displayLink.preferredFramesPerSecond = 30;
+#else
+    [_displayLink setFrameInterval:2];
+#endif
+    [_displayLink addToRunLoop:[NSRunLoop currentRunLoop]
+                       forMode:NSRunLoopCommonModes];
+  }
+  return self;
+}
+
+- (void)dealloc {
+  [self invalidate];
+}
+
+- (BOOL)isPaused {
+  return _displayLink.paused;
+}
+
+- (void)setIsPaused:(BOOL)isPaused {
+  _displayLink.paused = isPaused;
+}
+
+- (void)invalidate {
+  [_displayLink invalidate];
+}
+
+- (void)displayLinkDidFire:(CADisplayLink *)displayLink {
+  _timerHandler();
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/components/renderer/opengl/RTCEAGLVideoView.h b/third_party/libwebrtc/sdk/objc/components/renderer/opengl/RTCEAGLVideoView.h
new file mode 100644
index 0000000000..24b26cd602
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/components/renderer/opengl/RTCEAGLVideoView.h
@@ -0,0 +1,45 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+#import <UIKit/UIKit.h>
+
+#import "RTCMacros.h"
+#import "RTCVideoRenderer.h"
+#import "RTCVideoViewShading.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+@class RTC_OBJC_TYPE(RTCEAGLVideoView);
+
+/**
+ * RTCEAGLVideoView is an RTCVideoRenderer which renders video frames
+ * in its bounds using OpenGLES 2.0 or OpenGLES 3.0.
+ */
+RTC_OBJC_EXPORT
+NS_EXTENSION_UNAVAILABLE_IOS("Rendering not available in app extensions.")
+@interface RTC_OBJC_TYPE (RTCEAGLVideoView) : UIView <RTC_OBJC_TYPE(RTCVideoRenderer)>
+
+@property(nonatomic, weak) id<RTC_OBJC_TYPE(RTCVideoViewDelegate)> delegate;
+
+- (instancetype)initWithFrame:(CGRect)frame
+                       shader:(id<RTC_OBJC_TYPE(RTCVideoViewShading)>)shader
+    NS_DESIGNATED_INITIALIZER;
+
+- (instancetype)initWithCoder:(NSCoder *)aDecoder
+                       shader:(id<RTC_OBJC_TYPE(RTCVideoViewShading)>)shader
+    NS_DESIGNATED_INITIALIZER;
+
+/** @abstract Wrapped RTCVideoRotation, or nil.
+ */
+@property(nonatomic, nullable) NSValue *rotationOverride;
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/components/renderer/opengl/RTCEAGLVideoView.m b/third_party/libwebrtc/sdk/objc/components/renderer/opengl/RTCEAGLVideoView.m
new file mode 100644
index 0000000000..89e62d2ce7
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/components/renderer/opengl/RTCEAGLVideoView.m
@@ -0,0 +1,295 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCEAGLVideoView.h"
+
+#import <GLKit/GLKit.h>
+
+#import "RTCDefaultShader.h"
+#import "RTCDisplayLinkTimer.h"
+#import "RTCI420TextureCache.h"
+#import "RTCNV12TextureCache.h"
+#import "base/RTCLogging.h"
+#import "base/RTCVideoFrame.h"
+#import "base/RTCVideoFrameBuffer.h"
+#import "components/video_frame_buffer/RTCCVPixelBuffer.h"
+
+// RTC_OBJC_TYPE(RTCEAGLVideoView) wraps a GLKView which is setup with
+// enableSetNeedsDisplay = NO for the purpose of gaining control of
+// exactly when to call -[GLKView display]. This need for extra
+// control is required to avoid triggering method calls on GLKView
+// that results in attempting to bind the underlying render buffer
+// when the drawable size would be empty which would result in the
+// error GL_FRAMEBUFFER_INCOMPLETE_ATTACHMENT. -[GLKView display] is
+// the method that will trigger the binding of the render
+// buffer. Because the standard behaviour of -[UIView setNeedsDisplay]
+// is disabled for the reasons above, the RTC_OBJC_TYPE(RTCEAGLVideoView) maintains
+// its own `isDirty` flag.
+
+@interface RTC_OBJC_TYPE (RTCEAGLVideoView)
+()<GLKViewDelegate>
+    // `videoFrame` is set when we receive a frame from a worker thread and is read
+    // from the display link callback so atomicity is required.
+    @property(atomic, strong) RTC_OBJC_TYPE(RTCVideoFrame) * videoFrame;
+@property(nonatomic, readonly) GLKView *glkView;
+@end
+
+@implementation RTC_OBJC_TYPE (RTCEAGLVideoView) {
+  RTCDisplayLinkTimer *_timer;
+  EAGLContext *_glContext;
+  // This flag should only be set and read on the main thread (e.g. by
+  // setNeedsDisplay)
+  BOOL _isDirty;
+  id<RTC_OBJC_TYPE(RTCVideoViewShading)> _shader;
+  RTCNV12TextureCache *_nv12TextureCache;
+  RTCI420TextureCache *_i420TextureCache;
+  // As timestamps should be unique between frames, will store last
+  // drawn frame timestamp instead of the whole frame to reduce memory usage.
+  int64_t _lastDrawnFrameTimeStampNs;
+}
+
+@synthesize delegate = _delegate;
+@synthesize videoFrame = _videoFrame;
+@synthesize glkView = _glkView;
+@synthesize rotationOverride = _rotationOverride;
+
+- (instancetype)initWithFrame:(CGRect)frame {
+  return [self initWithFrame:frame shader:[[RTCDefaultShader alloc] init]];
+}
+
+- (instancetype)initWithCoder:(NSCoder *)aDecoder {
+  return [self initWithCoder:aDecoder shader:[[RTCDefaultShader alloc] init]];
+}
+
+- (instancetype)initWithFrame:(CGRect)frame shader:(id<RTC_OBJC_TYPE(RTCVideoViewShading)>)shader {
+  if (self = [super initWithFrame:frame]) {
+    _shader = shader;
+    if (![self configure]) {
+      return nil;
+    }
+  }
+  return self;
+}
+
+- (instancetype)initWithCoder:(NSCoder *)aDecoder
+                       shader:(id<RTC_OBJC_TYPE(RTCVideoViewShading)>)shader {
+  if (self = [super initWithCoder:aDecoder]) {
+    _shader = shader;
+    if (![self configure]) {
+      return nil;
+    }
+  }
+  return self;
+}
+
+- (BOOL)configure {
+  EAGLContext *glContext =
+    [[EAGLContext alloc] initWithAPI:kEAGLRenderingAPIOpenGLES3];
+  if (!glContext) {
+    glContext = [[EAGLContext alloc] initWithAPI:kEAGLRenderingAPIOpenGLES2];
+  }
+  if (!glContext) {
+    RTCLogError(@"Failed to create EAGLContext");
+    return NO;
+  }
+  _glContext = glContext;
+
+  // GLKView manages a framebuffer for us.
+  _glkView = [[GLKView alloc] initWithFrame:CGRectZero
+                                    context:_glContext];
+  _glkView.drawableColorFormat = GLKViewDrawableColorFormatRGBA8888;
+  _glkView.drawableDepthFormat = GLKViewDrawableDepthFormatNone;
+  _glkView.drawableStencilFormat = GLKViewDrawableStencilFormatNone;
+  _glkView.drawableMultisample = GLKViewDrawableMultisampleNone;
+  _glkView.delegate = self;
+  _glkView.layer.masksToBounds = YES;
+  _glkView.enableSetNeedsDisplay = NO;
+  [self addSubview:_glkView];
+
+  // Listen to application state in order to clean up OpenGL before app goes
+  // away.
+  NSNotificationCenter *notificationCenter =
+    [NSNotificationCenter defaultCenter];
+  [notificationCenter addObserver:self
+                         selector:@selector(willResignActive)
+                             name:UIApplicationWillResignActiveNotification
+                           object:nil];
+  [notificationCenter addObserver:self
+                         selector:@selector(didBecomeActive)
+                             name:UIApplicationDidBecomeActiveNotification
+                           object:nil];
+
+  // Frames are received on a separate thread, so we poll for current frame
+  // using a refresh rate proportional to screen refresh frequency. This
+  // occurs on the main thread.
+  __weak RTC_OBJC_TYPE(RTCEAGLVideoView) *weakSelf = self;
+  _timer = [[RTCDisplayLinkTimer alloc] initWithTimerHandler:^{
+    RTC_OBJC_TYPE(RTCEAGLVideoView) *strongSelf = weakSelf;
+    [strongSelf displayLinkTimerDidFire];
+  }];
+  if ([[UIApplication sharedApplication] applicationState] == UIApplicationStateActive) {
+    [self setupGL];
+  }
+  return YES;
+}
+
+- (void)setMultipleTouchEnabled:(BOOL)multipleTouchEnabled {
+    [super setMultipleTouchEnabled:multipleTouchEnabled];
+    _glkView.multipleTouchEnabled = multipleTouchEnabled;
+}
+
+- (void)dealloc {
+  [[NSNotificationCenter defaultCenter] removeObserver:self];
+  UIApplicationState appState =
+      [UIApplication sharedApplication].applicationState;
+  if (appState == UIApplicationStateActive) {
+    [self teardownGL];
+  }
+  [_timer invalidate];
+  [self ensureGLContext];
+  _shader = nil;
+  if (_glContext && [EAGLContext currentContext] == _glContext) {
+    [EAGLContext setCurrentContext:nil];
+  }
+}
+
+#pragma mark - UIView
+
+- (void)setNeedsDisplay {
+  [super setNeedsDisplay];
+  _isDirty = YES;
+}
+
+- (void)setNeedsDisplayInRect:(CGRect)rect {
+  [super setNeedsDisplayInRect:rect];
+  _isDirty = YES;
+}
+
+- (void)layoutSubviews {
+  [super layoutSubviews];
+  _glkView.frame = self.bounds;
+}
+
+#pragma mark - GLKViewDelegate
+
+// This method is called when the GLKView's content is dirty and needs to be
+// redrawn. This occurs on main thread.
+- (void)glkView:(GLKView *)view drawInRect:(CGRect)rect {
+  // The renderer will draw the frame to the framebuffer corresponding to the
+  // one used by `view`.
+  RTC_OBJC_TYPE(RTCVideoFrame) *frame = self.videoFrame;
+  if (!frame || frame.timeStampNs == _lastDrawnFrameTimeStampNs) {
+    return;
+  }
+  RTCVideoRotation rotation = frame.rotation;
+  if(_rotationOverride != nil) {
+    [_rotationOverride getValue: &rotation];
+  }
+  [self ensureGLContext];
+  glClear(GL_COLOR_BUFFER_BIT);
+  if ([frame.buffer isKindOfClass:[RTC_OBJC_TYPE(RTCCVPixelBuffer) class]]) {
+    if (!_nv12TextureCache) {
+      _nv12TextureCache = [[RTCNV12TextureCache alloc] initWithContext:_glContext];
+    }
+    if (_nv12TextureCache) {
+      [_nv12TextureCache uploadFrameToTextures:frame];
+      [_shader applyShadingForFrameWithWidth:frame.width
+                                      height:frame.height
+                                    rotation:rotation
+                                      yPlane:_nv12TextureCache.yTexture
+                                     uvPlane:_nv12TextureCache.uvTexture];
+      [_nv12TextureCache releaseTextures];
+
+      _lastDrawnFrameTimeStampNs = self.videoFrame.timeStampNs;
+    }
+  } else {
+    if (!_i420TextureCache) {
+      _i420TextureCache = [[RTCI420TextureCache alloc] initWithContext:_glContext];
+    }
+    [_i420TextureCache uploadFrameToTextures:frame];
+    [_shader applyShadingForFrameWithWidth:frame.width
+                                    height:frame.height
+                                  rotation:rotation
+                                    yPlane:_i420TextureCache.yTexture
+                                    uPlane:_i420TextureCache.uTexture
+                                    vPlane:_i420TextureCache.vTexture];
+
+    _lastDrawnFrameTimeStampNs = self.videoFrame.timeStampNs;
+  }
+}
+
+#pragma mark - RTC_OBJC_TYPE(RTCVideoRenderer)
+
+// These methods may be called on non-main thread.
+- (void)setSize:(CGSize)size {
+  __weak RTC_OBJC_TYPE(RTCEAGLVideoView) *weakSelf = self;
+  dispatch_async(dispatch_get_main_queue(), ^{
+    RTC_OBJC_TYPE(RTCEAGLVideoView) *strongSelf = weakSelf;
+    [strongSelf.delegate videoView:strongSelf didChangeVideoSize:size];
+  });
+}
+
+- (void)renderFrame:(RTC_OBJC_TYPE(RTCVideoFrame) *)frame {
+  self.videoFrame = frame;
+}
+
+#pragma mark - Private
+
+- (void)displayLinkTimerDidFire {
+  // Don't render unless video frame have changed or the view content
+  // has explicitly been marked dirty.
+  if (!_isDirty && _lastDrawnFrameTimeStampNs == self.videoFrame.timeStampNs) {
+    return;
+  }
+
+  // Always reset isDirty at this point, even if -[GLKView display]
+  // won't be called in the case the drawable size is empty.
+  _isDirty = NO;
+
+  // Only call -[GLKView display] if the drawable size is
+  // non-empty. Calling display will make the GLKView setup its
+  // render buffer if necessary, but that will fail with error
+  // GL_FRAMEBUFFER_INCOMPLETE_ATTACHMENT if size is empty.
+  if (self.bounds.size.width > 0 && self.bounds.size.height > 0) {
+    [_glkView display];
+  }
+}
+
+- (void)setupGL {
+  [self ensureGLContext];
+  glDisable(GL_DITHER);
+  _timer.isPaused = NO;
+}
+
+- (void)teardownGL {
+  self.videoFrame = nil;
+  _timer.isPaused = YES;
+  [_glkView deleteDrawable];
+  [self ensureGLContext];
+  _nv12TextureCache = nil;
+  _i420TextureCache = nil;
+}
+
+- (void)didBecomeActive {
+  [self setupGL];
+}
+
+- (void)willResignActive {
+  [self teardownGL];
+}
+
+- (void)ensureGLContext {
+  NSAssert(_glContext, @"context shouldn't be nil");
+  if ([EAGLContext currentContext] != _glContext) {
+    [EAGLContext setCurrentContext:_glContext];
+  }
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/components/renderer/opengl/RTCI420TextureCache.h b/third_party/libwebrtc/sdk/objc/components/renderer/opengl/RTCI420TextureCache.h
new file mode 100644
index 0000000000..9fdcc5a695
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/components/renderer/opengl/RTCI420TextureCache.h
@@ -0,0 +1,25 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCOpenGLDefines.h"
+#import "base/RTCVideoFrame.h"
+
+@interface RTCI420TextureCache : NSObject
+
+@property(nonatomic, readonly) GLuint yTexture;
+@property(nonatomic, readonly) GLuint uTexture;
+@property(nonatomic, readonly) GLuint vTexture;
+
+- (instancetype)init NS_UNAVAILABLE;
+- (instancetype)initWithContext:(GlContextType *)context NS_DESIGNATED_INITIALIZER;
+
+- (void)uploadFrameToTextures:(RTC_OBJC_TYPE(RTCVideoFrame) *)frame;
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/components/renderer/opengl/RTCI420TextureCache.mm b/third_party/libwebrtc/sdk/objc/components/renderer/opengl/RTCI420TextureCache.mm
new file mode 100644
index 0000000000..5dccd4bf6a
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/components/renderer/opengl/RTCI420TextureCache.mm
@@ -0,0 +1,157 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCI420TextureCache.h"
+
+#if TARGET_OS_IPHONE
+#import <OpenGLES/ES3/gl.h>
+#else
+#import <OpenGL/gl3.h>
+#endif
+
+#import "base/RTCI420Buffer.h"
+#import "base/RTCVideoFrameBuffer.h"
+
+#include <vector>
+
+// Two sets of 3 textures are used here, one for each of the Y, U and V planes. Having two sets
+// alleviates CPU blockage in the event that the GPU is asked to render to a texture that is already
+// in use.
+static const GLsizei kNumTextureSets = 2;
+static const GLsizei kNumTexturesPerSet = 3;
+static const GLsizei kNumTextures = kNumTexturesPerSet * kNumTextureSets;
+
+@implementation RTCI420TextureCache {
+  BOOL _hasUnpackRowLength;
+  GLint _currentTextureSet;
+  // Handles for OpenGL constructs.
+  GLuint _textures[kNumTextures];
+  // Used to create a non-padded plane for GPU upload when we receive padded frames.
+  std::vector<uint8_t> _planeBuffer;
+}
+
+- (GLuint)yTexture {
+  return _textures[_currentTextureSet * kNumTexturesPerSet];
+}
+
+- (GLuint)uTexture {
+  return _textures[_currentTextureSet * kNumTexturesPerSet + 1];
+}
+
+- (GLuint)vTexture {
+  return _textures[_currentTextureSet * kNumTexturesPerSet + 2];
+}
+
+- (instancetype)initWithContext:(GlContextType *)context {
+  if (self = [super init]) {
+#if TARGET_OS_IPHONE
+    _hasUnpackRowLength = (context.API == kEAGLRenderingAPIOpenGLES3);
+#else
+    _hasUnpackRowLength = YES;
+#endif
+    glPixelStorei(GL_UNPACK_ALIGNMENT, 1);
+
+    [self setupTextures];
+  }
+  return self;
+}
+
+- (void)dealloc {
+  glDeleteTextures(kNumTextures, _textures);
+}
+
+- (void)setupTextures {
+  glGenTextures(kNumTextures, _textures);
+  // Set parameters for each of the textures we created.
+  for (GLsizei i = 0; i < kNumTextures; i++) {
+    glBindTexture(GL_TEXTURE_2D, _textures[i]);
+    glTexParameteri(GL_TEXTURE_2D, GL_TEXTURE_MIN_FILTER, GL_LINEAR);
+    glTexParameteri(GL_TEXTURE_2D, GL_TEXTURE_MAG_FILTER, GL_LINEAR);
+    glTexParameteri(GL_TEXTURE_2D, GL_TEXTURE_WRAP_S, GL_CLAMP_TO_EDGE);
+    glTexParameteri(GL_TEXTURE_2D, GL_TEXTURE_WRAP_T, GL_CLAMP_TO_EDGE);
+  }
+}
+
+- (void)uploadPlane:(const uint8_t *)plane
+            texture:(GLuint)texture
+              width:(size_t)width
+             height:(size_t)height
+             stride:(int32_t)stride {
+  glBindTexture(GL_TEXTURE_2D, texture);
+
+  const uint8_t *uploadPlane = plane;
+  if ((size_t)stride != width) {
+   if (_hasUnpackRowLength) {
+      // GLES3 allows us to specify stride.
+      glPixelStorei(GL_UNPACK_ROW_LENGTH, stride);
+      glTexImage2D(GL_TEXTURE_2D,
+                   0,
+                   RTC_PIXEL_FORMAT,
+                   static_cast<GLsizei>(width),
+                   static_cast<GLsizei>(height),
+                   0,
+                   RTC_PIXEL_FORMAT,
+                   GL_UNSIGNED_BYTE,
+                   uploadPlane);
+      glPixelStorei(GL_UNPACK_ROW_LENGTH, 0);
+      return;
+    } else {
+      // Make an unpadded copy and upload that instead. Quick profiling showed
+      // that this is faster than uploading row by row using glTexSubImage2D.
+      uint8_t *unpaddedPlane = _planeBuffer.data();
+      for (size_t y = 0; y < height; ++y) {
+        memcpy(unpaddedPlane + y * width, plane + y * stride, width);
+      }
+      uploadPlane = unpaddedPlane;
+    }
+  }
+  glTexImage2D(GL_TEXTURE_2D,
+               0,
+               RTC_PIXEL_FORMAT,
+               static_cast<GLsizei>(width),
+               static_cast<GLsizei>(height),
+               0,
+               RTC_PIXEL_FORMAT,
+               GL_UNSIGNED_BYTE,
+               uploadPlane);
+}
+
+- (void)uploadFrameToTextures:(RTC_OBJC_TYPE(RTCVideoFrame) *)frame {
+  _currentTextureSet = (_currentTextureSet + 1) % kNumTextureSets;
+
+  id<RTC_OBJC_TYPE(RTCI420Buffer)> buffer = [frame.buffer toI420];
+
+  const int chromaWidth = buffer.chromaWidth;
+  const int chromaHeight = buffer.chromaHeight;
+  if (buffer.strideY != frame.width || buffer.strideU != chromaWidth ||
+      buffer.strideV != chromaWidth) {
+    _planeBuffer.resize(buffer.width * buffer.height);
+  }
+
+  [self uploadPlane:buffer.dataY
+            texture:self.yTexture
+              width:buffer.width
+             height:buffer.height
+             stride:buffer.strideY];
+
+  [self uploadPlane:buffer.dataU
+            texture:self.uTexture
+              width:chromaWidth
+             height:chromaHeight
+             stride:buffer.strideU];
+
+  [self uploadPlane:buffer.dataV
+            texture:self.vTexture
+              width:chromaWidth
+             height:chromaHeight
+             stride:buffer.strideV];
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/components/renderer/opengl/RTCNSGLVideoView.h b/third_party/libwebrtc/sdk/objc/components/renderer/opengl/RTCNSGLVideoView.h
new file mode 100644
index 0000000000..c9ee986f88
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/components/renderer/opengl/RTCNSGLVideoView.h
@@ -0,0 +1,42 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#if !TARGET_OS_IPHONE
+
+#import <AppKit/NSOpenGLView.h>
+
+#import "RTCVideoRenderer.h"
+#import "RTCVideoViewShading.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+@class RTC_OBJC_TYPE(RTCNSGLVideoView);
+
+RTC_OBJC_EXPORT
+@protocol RTC_OBJC_TYPE
+(RTCNSGLVideoViewDelegate)<RTC_OBJC_TYPE(RTCVideoViewDelegate)> @end
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCNSGLVideoView) : NSOpenGLView <RTC_OBJC_TYPE(RTCVideoRenderer)>
+
+@property(nonatomic, weak) id<RTC_OBJC_TYPE(RTCVideoViewDelegate)> delegate;
+
+- (instancetype)initWithFrame:(NSRect)frameRect
+                  pixelFormat:(NSOpenGLPixelFormat *)format
+                       shader:(id<RTC_OBJC_TYPE(RTCVideoViewShading)>)shader
+    NS_DESIGNATED_INITIALIZER;
+
+@end
+
+NS_ASSUME_NONNULL_END
+
+#endif
diff --git a/third_party/libwebrtc/sdk/objc/components/renderer/opengl/RTCNSGLVideoView.m b/third_party/libwebrtc/sdk/objc/components/renderer/opengl/RTCNSGLVideoView.m
new file mode 100644
index 0000000000..168c73126f
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/components/renderer/opengl/RTCNSGLVideoView.m
@@ -0,0 +1,199 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#if !TARGET_OS_IPHONE
+
+#import "RTCNSGLVideoView.h"
+
+#import <AppKit/NSOpenGL.h>
+#import <CoreVideo/CVDisplayLink.h>
+#import <OpenGL/gl3.h>
+
+#import "RTCDefaultShader.h"
+#import "RTCI420TextureCache.h"
+#import "base/RTCLogging.h"
+#import "base/RTCVideoFrame.h"
+
+@interface RTC_OBJC_TYPE (RTCNSGLVideoView)
+()
+    // `videoFrame` is set when we receive a frame from a worker thread and is read
+    // from the display link callback so atomicity is required.
+    @property(atomic, strong) RTC_OBJC_TYPE(RTCVideoFrame) *
+    videoFrame;
+@property(atomic, strong) RTCI420TextureCache *i420TextureCache;
+
+- (void)drawFrame;
+@end
+
+static CVReturn OnDisplayLinkFired(CVDisplayLinkRef displayLink,
+                                   const CVTimeStamp *now,
+                                   const CVTimeStamp *outputTime,
+                                   CVOptionFlags flagsIn,
+                                   CVOptionFlags *flagsOut,
+                                   void *displayLinkContext) {
+  RTC_OBJC_TYPE(RTCNSGLVideoView) *view =
+      (__bridge RTC_OBJC_TYPE(RTCNSGLVideoView) *)displayLinkContext;
+  [view drawFrame];
+  return kCVReturnSuccess;
+}
+
+@implementation RTC_OBJC_TYPE (RTCNSGLVideoView) {
+  CVDisplayLinkRef _displayLink;
+  RTC_OBJC_TYPE(RTCVideoFrame) * _lastDrawnFrame;
+  id<RTC_OBJC_TYPE(RTCVideoViewShading)> _shader;
+}
+
+@synthesize delegate = _delegate;
+@synthesize videoFrame = _videoFrame;
+@synthesize i420TextureCache = _i420TextureCache;
+
+- (instancetype)initWithFrame:(NSRect)frame pixelFormat:(NSOpenGLPixelFormat *)format {
+  return [self initWithFrame:frame pixelFormat:format shader:[[RTCDefaultShader alloc] init]];
+}
+
+- (instancetype)initWithFrame:(NSRect)frame
+                  pixelFormat:(NSOpenGLPixelFormat *)format
+                       shader:(id<RTC_OBJC_TYPE(RTCVideoViewShading)>)shader {
+  if (self = [super initWithFrame:frame pixelFormat:format]) {
+    _shader = shader;
+  }
+  return self;
+}
+
+- (void)dealloc {
+  [self teardownDisplayLink];
+}
+
+- (void)drawRect:(NSRect)rect {
+  [self drawFrame];
+}
+
+- (void)reshape {
+  [super reshape];
+  NSRect frame = [self frame];
+  [self ensureGLContext];
+  CGLLockContext([[self openGLContext] CGLContextObj]);
+  glViewport(0, 0, frame.size.width, frame.size.height);
+  CGLUnlockContext([[self openGLContext] CGLContextObj]);
+}
+
+- (void)lockFocus {
+  NSOpenGLContext *context = [self openGLContext];
+  [super lockFocus];
+  if ([context view] != self) {
+    [context setView:self];
+  }
+  [context makeCurrentContext];
+}
+
+- (void)prepareOpenGL {
+  [super prepareOpenGL];
+  [self ensureGLContext];
+  glDisable(GL_DITHER);
+  [self setupDisplayLink];
+}
+
+- (void)clearGLContext {
+  [self ensureGLContext];
+  self.i420TextureCache = nil;
+  [super clearGLContext];
+}
+
+#pragma mark - RTC_OBJC_TYPE(RTCVideoRenderer)
+
+// These methods may be called on non-main thread.
+- (void)setSize:(CGSize)size {
+  dispatch_async(dispatch_get_main_queue(), ^{
+    [self.delegate videoView:self didChangeVideoSize:size];
+  });
+}
+
+- (void)renderFrame:(RTC_OBJC_TYPE(RTCVideoFrame) *)frame {
+  self.videoFrame = frame;
+}
+
+#pragma mark - Private
+
+- (void)drawFrame {
+  RTC_OBJC_TYPE(RTCVideoFrame) *frame = self.videoFrame;
+  if (!frame || frame == _lastDrawnFrame) {
+    return;
+  }
+  // This method may be called from CVDisplayLink callback which isn't on the
+  // main thread so we have to lock the GL context before drawing.
+  NSOpenGLContext *context = [self openGLContext];
+  CGLLockContext([context CGLContextObj]);
+
+  [self ensureGLContext];
+  glClear(GL_COLOR_BUFFER_BIT);
+
+  // Rendering native CVPixelBuffer is not supported on OS X.
+  // TODO(magjed): Add support for NV12 texture cache on OS X.
+  frame = [frame newI420VideoFrame];
+  if (!self.i420TextureCache) {
+    self.i420TextureCache = [[RTCI420TextureCache alloc] initWithContext:context];
+  }
+  RTCI420TextureCache *i420TextureCache = self.i420TextureCache;
+  if (i420TextureCache) {
+    [i420TextureCache uploadFrameToTextures:frame];
+    [_shader applyShadingForFrameWithWidth:frame.width
+                                    height:frame.height
+                                  rotation:frame.rotation
+                                    yPlane:i420TextureCache.yTexture
+                                    uPlane:i420TextureCache.uTexture
+                                    vPlane:i420TextureCache.vTexture];
+    [context flushBuffer];
+    _lastDrawnFrame = frame;
+  }
+  CGLUnlockContext([context CGLContextObj]);
+}
+
+- (void)setupDisplayLink {
+  if (_displayLink) {
+    return;
+  }
+  // Synchronize buffer swaps with vertical refresh rate.
+  GLint swapInt = 1;
+  [[self openGLContext] setValues:&swapInt forParameter:NSOpenGLCPSwapInterval];
+
+  // Create display link.
+  CVDisplayLinkCreateWithActiveCGDisplays(&_displayLink);
+  CVDisplayLinkSetOutputCallback(_displayLink,
+                                 &OnDisplayLinkFired,
+                                 (__bridge void *)self);
+  // Set the display link for the current renderer.
+  CGLContextObj cglContext = [[self openGLContext] CGLContextObj];
+  CGLPixelFormatObj cglPixelFormat = [[self pixelFormat] CGLPixelFormatObj];
+  CVDisplayLinkSetCurrentCGDisplayFromOpenGLContext(
+      _displayLink, cglContext, cglPixelFormat);
+  CVDisplayLinkStart(_displayLink);
+}
+
+- (void)teardownDisplayLink {
+  if (!_displayLink) {
+    return;
+  }
+  CVDisplayLinkRelease(_displayLink);
+  _displayLink = NULL;
+}
+
+- (void)ensureGLContext {
+  NSOpenGLContext* context = [self openGLContext];
+  NSAssert(context, @"context shouldn't be nil");
+  if ([NSOpenGLContext currentContext] != context) {
+    [context makeCurrentContext];
+  }
+}
+
+@end
+
+#endif  // !TARGET_OS_IPHONE
diff --git a/third_party/libwebrtc/sdk/objc/components/renderer/opengl/RTCNV12TextureCache.h b/third_party/libwebrtc/sdk/objc/components/renderer/opengl/RTCNV12TextureCache.h
new file mode 100644
index 0000000000..f202b836b5
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/components/renderer/opengl/RTCNV12TextureCache.h
@@ -0,0 +1,33 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <GLKit/GLKit.h>
+
+#import "base/RTCMacros.h"
+
+@class RTC_OBJC_TYPE(RTCVideoFrame);
+
+NS_ASSUME_NONNULL_BEGIN
+
+@interface RTCNV12TextureCache : NSObject
+
+@property(nonatomic, readonly) GLuint yTexture;
+@property(nonatomic, readonly) GLuint uvTexture;
+
+- (instancetype)init NS_UNAVAILABLE;
+- (nullable instancetype)initWithContext:(EAGLContext *)context NS_DESIGNATED_INITIALIZER;
+
+- (BOOL)uploadFrameToTextures:(RTC_OBJC_TYPE(RTCVideoFrame) *)frame;
+
+- (void)releaseTextures;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/components/renderer/opengl/RTCNV12TextureCache.m b/third_party/libwebrtc/sdk/objc/components/renderer/opengl/RTCNV12TextureCache.m
new file mode 100644
index 0000000000..a520ac45b4
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/components/renderer/opengl/RTCNV12TextureCache.m
@@ -0,0 +1,113 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCNV12TextureCache.h"
+
+#import "base/RTCVideoFrame.h"
+#import "base/RTCVideoFrameBuffer.h"
+#import "components/video_frame_buffer/RTCCVPixelBuffer.h"
+
+@implementation RTCNV12TextureCache {
+  CVOpenGLESTextureCacheRef _textureCache;
+  CVOpenGLESTextureRef _yTextureRef;
+  CVOpenGLESTextureRef _uvTextureRef;
+}
+
+- (GLuint)yTexture {
+  return CVOpenGLESTextureGetName(_yTextureRef);
+}
+
+- (GLuint)uvTexture {
+  return CVOpenGLESTextureGetName(_uvTextureRef);
+}
+
+- (instancetype)initWithContext:(EAGLContext *)context {
+  if (self = [super init]) {
+    CVReturn ret = CVOpenGLESTextureCacheCreate(
+        kCFAllocatorDefault, NULL,
+#if COREVIDEO_USE_EAGLCONTEXT_CLASS_IN_API
+        context,
+#else
+        (__bridge void *)context,
+#endif
+        NULL, &_textureCache);
+    if (ret != kCVReturnSuccess) {
+      self = nil;
+    }
+  }
+  return self;
+}
+
+- (BOOL)loadTexture:(CVOpenGLESTextureRef *)textureOut
+        pixelBuffer:(CVPixelBufferRef)pixelBuffer
+         planeIndex:(int)planeIndex
+        pixelFormat:(GLenum)pixelFormat {
+  const int width = CVPixelBufferGetWidthOfPlane(pixelBuffer, planeIndex);
+  const int height = CVPixelBufferGetHeightOfPlane(pixelBuffer, planeIndex);
+
+  if (*textureOut) {
+    CFRelease(*textureOut);
+    *textureOut = nil;
+  }
+  CVReturn ret = CVOpenGLESTextureCacheCreateTextureFromImage(
+      kCFAllocatorDefault, _textureCache, pixelBuffer, NULL, GL_TEXTURE_2D, pixelFormat, width,
+      height, pixelFormat, GL_UNSIGNED_BYTE, planeIndex, textureOut);
+  if (ret != kCVReturnSuccess) {
+    if (*textureOut) {
+      CFRelease(*textureOut);
+      *textureOut = nil;
+    }
+    return NO;
+  }
+  NSAssert(CVOpenGLESTextureGetTarget(*textureOut) == GL_TEXTURE_2D,
+           @"Unexpected GLES texture target");
+  glBindTexture(GL_TEXTURE_2D, CVOpenGLESTextureGetName(*textureOut));
+  glTexParameteri(GL_TEXTURE_2D, GL_TEXTURE_MIN_FILTER, GL_LINEAR);
+  glTexParameteri(GL_TEXTURE_2D, GL_TEXTURE_MAG_FILTER, GL_LINEAR);
+  glTexParameteri(GL_TEXTURE_2D, GL_TEXTURE_WRAP_S, GL_CLAMP_TO_EDGE);
+  glTexParameteri(GL_TEXTURE_2D, GL_TEXTURE_WRAP_T, GL_CLAMP_TO_EDGE);
+  return YES;
+}
+
+- (BOOL)uploadFrameToTextures:(RTC_OBJC_TYPE(RTCVideoFrame) *)frame {
+  NSAssert([frame.buffer isKindOfClass:[RTC_OBJC_TYPE(RTCCVPixelBuffer) class]],
+           @"frame must be CVPixelBuffer backed");
+  RTC_OBJC_TYPE(RTCCVPixelBuffer) *rtcPixelBuffer = (RTC_OBJC_TYPE(RTCCVPixelBuffer) *)frame.buffer;
+  CVPixelBufferRef pixelBuffer = rtcPixelBuffer.pixelBuffer;
+  return [self loadTexture:&_yTextureRef
+               pixelBuffer:pixelBuffer
+                planeIndex:0
+               pixelFormat:GL_LUMINANCE] &&
+         [self loadTexture:&_uvTextureRef
+               pixelBuffer:pixelBuffer
+                planeIndex:1
+               pixelFormat:GL_LUMINANCE_ALPHA];
+}
+
+- (void)releaseTextures {
+  if (_uvTextureRef) {
+    CFRelease(_uvTextureRef);
+    _uvTextureRef = nil;
+  }
+  if (_yTextureRef) {
+    CFRelease(_yTextureRef);
+    _yTextureRef = nil;
+  }
+}
+
+- (void)dealloc {
+  [self releaseTextures];
+  if (_textureCache) {
+    CFRelease(_textureCache);
+    _textureCache = nil;
+  }
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/components/renderer/opengl/RTCOpenGLDefines.h b/third_party/libwebrtc/sdk/objc/components/renderer/opengl/RTCOpenGLDefines.h
new file mode 100644
index 0000000000..4088535861
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/components/renderer/opengl/RTCOpenGLDefines.h
@@ -0,0 +1,37 @@
+/*
+ *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#if TARGET_OS_IPHONE
+#define RTC_PIXEL_FORMAT GL_LUMINANCE
+#define SHADER_VERSION
+#define VERTEX_SHADER_IN "attribute"
+#define VERTEX_SHADER_OUT "varying"
+#define FRAGMENT_SHADER_IN "varying"
+#define FRAGMENT_SHADER_OUT
+#define FRAGMENT_SHADER_COLOR "gl_FragColor"
+#define FRAGMENT_SHADER_TEXTURE "texture2D"
+
+@class EAGLContext;
+typedef EAGLContext GlContextType;
+#else
+#define RTC_PIXEL_FORMAT GL_RED
+#define SHADER_VERSION "#version 150\n"
+#define VERTEX_SHADER_IN "in"
+#define VERTEX_SHADER_OUT "out"
+#define FRAGMENT_SHADER_IN "in"
+#define FRAGMENT_SHADER_OUT "out vec4 fragColor;\n"
+#define FRAGMENT_SHADER_COLOR "fragColor"
+#define FRAGMENT_SHADER_TEXTURE "texture"
+
+@class NSOpenGLContext;
+typedef NSOpenGLContext GlContextType;
+#endif
diff --git a/third_party/libwebrtc/sdk/objc/components/renderer/opengl/RTCShader.h b/third_party/libwebrtc/sdk/objc/components/renderer/opengl/RTCShader.h
new file mode 100644
index 0000000000..d1b91fb643
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/components/renderer/opengl/RTCShader.h
@@ -0,0 +1,21 @@
+/*
+ *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "base/RTCVideoFrame.h"
+
+RTC_EXTERN const char kRTCVertexShaderSource[];
+
+RTC_EXTERN GLuint RTCCreateShader(GLenum type, const GLchar* source);
+RTC_EXTERN GLuint RTCCreateProgram(GLuint vertexShader, GLuint fragmentShader);
+RTC_EXTERN GLuint
+RTCCreateProgramFromFragmentSource(const char fragmentShaderSource[]);
+RTC_EXTERN BOOL RTCCreateVertexBuffer(GLuint* vertexBuffer,
+                                      GLuint* vertexArray);
+RTC_EXTERN void RTCSetVertexData(RTCVideoRotation rotation);
diff --git a/third_party/libwebrtc/sdk/objc/components/renderer/opengl/RTCShader.mm b/third_party/libwebrtc/sdk/objc/components/renderer/opengl/RTCShader.mm
new file mode 100644
index 0000000000..8eccd7fbec
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/components/renderer/opengl/RTCShader.mm
@@ -0,0 +1,189 @@
+/*
+ *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCShader.h"
+
+#if TARGET_OS_IPHONE
+#import <OpenGLES/ES3/gl.h>
+#else
+#import <OpenGL/gl3.h>
+#endif
+
+#include <algorithm>
+#include <array>
+#include <memory>
+
+#import "RTCOpenGLDefines.h"
+
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+// Vertex shader doesn't do anything except pass coordinates through.
+const char kRTCVertexShaderSource[] =
+  SHADER_VERSION
+  VERTEX_SHADER_IN " vec2 position;\n"
+  VERTEX_SHADER_IN " vec2 texcoord;\n"
+  VERTEX_SHADER_OUT " vec2 v_texcoord;\n"
+  "void main() {\n"
+  "    gl_Position = vec4(position.x, position.y, 0.0, 1.0);\n"
+  "    v_texcoord = texcoord;\n"
+  "}\n";
+
+// Compiles a shader of the given `type` with GLSL source `source` and returns
+// the shader handle or 0 on error.
+GLuint RTCCreateShader(GLenum type, const GLchar *source) {
+  GLuint shader = glCreateShader(type);
+  if (!shader) {
+    return 0;
+  }
+  glShaderSource(shader, 1, &source, NULL);
+  glCompileShader(shader);
+  GLint compileStatus = GL_FALSE;
+  glGetShaderiv(shader, GL_COMPILE_STATUS, &compileStatus);
+  if (compileStatus == GL_FALSE) {
+    GLint logLength = 0;
+    // The null termination character is included in the returned log length.
+    glGetShaderiv(shader, GL_INFO_LOG_LENGTH, &logLength);
+    if (logLength > 0) {
+      std::unique_ptr<char[]> compileLog(new char[logLength]);
+      // The returned string is null terminated.
+      glGetShaderInfoLog(shader, logLength, NULL, compileLog.get());
+      RTC_LOG(LS_ERROR) << "Shader compile error: " << compileLog.get();
+    }
+    glDeleteShader(shader);
+    shader = 0;
+  }
+  return shader;
+}
+
+// Links a shader program with the given vertex and fragment shaders and
+// returns the program handle or 0 on error.
+GLuint RTCCreateProgram(GLuint vertexShader, GLuint fragmentShader) {
+  if (vertexShader == 0 || fragmentShader == 0) {
+    return 0;
+  }
+  GLuint program = glCreateProgram();
+  if (!program) {
+    return 0;
+  }
+  glAttachShader(program, vertexShader);
+  glAttachShader(program, fragmentShader);
+  glLinkProgram(program);
+  GLint linkStatus = GL_FALSE;
+  glGetProgramiv(program, GL_LINK_STATUS, &linkStatus);
+  if (linkStatus == GL_FALSE) {
+    glDeleteProgram(program);
+    program = 0;
+  }
+  return program;
+}
+
+// Creates and links a shader program with the given fragment shader source and
+// a plain vertex shader. Returns the program handle or 0 on error.
+GLuint RTCCreateProgramFromFragmentSource(const char fragmentShaderSource[]) {
+  GLuint vertexShader = RTCCreateShader(GL_VERTEX_SHADER, kRTCVertexShaderSource);
+  RTC_CHECK(vertexShader) << "failed to create vertex shader";
+  GLuint fragmentShader =
+      RTCCreateShader(GL_FRAGMENT_SHADER, fragmentShaderSource);
+  RTC_CHECK(fragmentShader) << "failed to create fragment shader";
+  GLuint program = RTCCreateProgram(vertexShader, fragmentShader);
+  // Shaders are created only to generate program.
+  if (vertexShader) {
+    glDeleteShader(vertexShader);
+  }
+  if (fragmentShader) {
+    glDeleteShader(fragmentShader);
+  }
+
+  // Set vertex shader variables 'position' and 'texcoord' in program.
+  GLint position = glGetAttribLocation(program, "position");
+  GLint texcoord = glGetAttribLocation(program, "texcoord");
+  if (position < 0 || texcoord < 0) {
+    glDeleteProgram(program);
+    return 0;
+  }
+
+  // Read position attribute with size of 2 and stride of 4 beginning at the start of the array. The
+  // last argument indicates offset of data within the vertex buffer.
+  glVertexAttribPointer(position, 2, GL_FLOAT, GL_FALSE, 4 * sizeof(GLfloat), (void *)0);
+  glEnableVertexAttribArray(position);
+
+  // Read texcoord attribute  with size of 2 and stride of 4 beginning at the first texcoord in the
+  // array. The last argument indicates offset of data within the vertex buffer.
+  glVertexAttribPointer(
+      texcoord, 2, GL_FLOAT, GL_FALSE, 4 * sizeof(GLfloat), (void *)(2 * sizeof(GLfloat)));
+  glEnableVertexAttribArray(texcoord);
+
+  return program;
+}
+
+BOOL RTCCreateVertexBuffer(GLuint *vertexBuffer, GLuint *vertexArray) {
+#if !TARGET_OS_IPHONE
+  glGenVertexArrays(1, vertexArray);
+  if (*vertexArray == 0) {
+    return NO;
+  }
+  glBindVertexArray(*vertexArray);
+#endif
+  glGenBuffers(1, vertexBuffer);
+  if (*vertexBuffer == 0) {
+    glDeleteVertexArrays(1, vertexArray);
+    return NO;
+  }
+  glBindBuffer(GL_ARRAY_BUFFER, *vertexBuffer);
+  glBufferData(GL_ARRAY_BUFFER, 4 * 4 * sizeof(GLfloat), NULL, GL_DYNAMIC_DRAW);
+  return YES;
+}
+
+// Set vertex data to the currently bound vertex buffer.
+void RTCSetVertexData(RTCVideoRotation rotation) {
+  // When modelview and projection matrices are identity (default) the world is
+  // contained in the square around origin with unit size 2. Drawing to these
+  // coordinates is equivalent to drawing to the entire screen. The texture is
+  // stretched over that square using texture coordinates (u, v) that range
+  // from (0, 0) to (1, 1) inclusive. Texture coordinates are flipped vertically
+  // here because the incoming frame has origin in upper left hand corner but
+  // OpenGL expects origin in bottom left corner.
+  std::array<std::array<GLfloat, 2>, 4> UVCoords = {{
+      {{0, 1}},  // Lower left.
+      {{1, 1}},  // Lower right.
+      {{1, 0}},  // Upper right.
+      {{0, 0}},  // Upper left.
+  }};
+
+  // Rotate the UV coordinates.
+  int rotation_offset;
+  switch (rotation) {
+    case RTCVideoRotation_0:
+      rotation_offset = 0;
+      break;
+    case RTCVideoRotation_90:
+      rotation_offset = 1;
+      break;
+    case RTCVideoRotation_180:
+      rotation_offset = 2;
+      break;
+    case RTCVideoRotation_270:
+      rotation_offset = 3;
+      break;
+  }
+  std::rotate(UVCoords.begin(), UVCoords.begin() + rotation_offset,
+              UVCoords.end());
+
+  const GLfloat gVertices[] = {
+      // X, Y, U, V.
+      -1, -1, UVCoords[0][0], UVCoords[0][1],
+       1, -1, UVCoords[1][0], UVCoords[1][1],
+       1,  1, UVCoords[2][0], UVCoords[2][1],
+      -1,  1, UVCoords[3][0], UVCoords[3][1],
+  };
+
+  glBufferSubData(GL_ARRAY_BUFFER, 0, sizeof(gVertices), gVertices);
+}
diff --git a/third_party/libwebrtc/sdk/objc/components/renderer/opengl/RTCVideoViewShading.h b/third_party/libwebrtc/sdk/objc/components/renderer/opengl/RTCVideoViewShading.h
new file mode 100644
index 0000000000..9df30a8fa0
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/components/renderer/opengl/RTCVideoViewShading.h
@@ -0,0 +1,39 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCVideoFrame.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+/**
+ * RTCVideoViewShading provides a way for apps to customize the OpenGL(ES shaders
+ * used in rendering for the RTCEAGLVideoView/RTCNSGLVideoView.
+ */
+RTC_OBJC_EXPORT
+@protocol RTC_OBJC_TYPE
+(RTCVideoViewShading)<NSObject>
+
+    /** Callback for I420 frames. Each plane is given as a texture. */
+    - (void)applyShadingForFrameWithWidth : (int)width height : (int)height rotation
+    : (RTCVideoRotation)rotation yPlane : (GLuint)yPlane uPlane : (GLuint)uPlane vPlane
+    : (GLuint)vPlane;
+
+/** Callback for NV12 frames. Each plane is given as a texture. */
+- (void)applyShadingForFrameWithWidth:(int)width
+                               height:(int)height
+                             rotation:(RTCVideoRotation)rotation
+                               yPlane:(GLuint)yPlane
+                              uvPlane:(GLuint)uvPlane;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/components/video_codec/RTCCodecSpecificInfoH264+Private.h b/third_party/libwebrtc/sdk/objc/components/video_codec/RTCCodecSpecificInfoH264+Private.h
new file mode 100644
index 0000000000..a0cd8515d1
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/components/video_codec/RTCCodecSpecificInfoH264+Private.h
@@ -0,0 +1,25 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCCodecSpecificInfoH264.h"
+
+#include "modules/video_coding/include/video_codec_interface.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+/* Interfaces for converting to/from internal C++ formats. */
+@interface RTC_OBJC_TYPE (RTCCodecSpecificInfoH264)
+()
+
+    - (webrtc::CodecSpecificInfo)nativeCodecSpecificInfo;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/components/video_codec/RTCCodecSpecificInfoH264.h b/third_party/libwebrtc/sdk/objc/components/video_codec/RTCCodecSpecificInfoH264.h
new file mode 100644
index 0000000000..ae3003a115
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/components/video_codec/RTCCodecSpecificInfoH264.h
@@ -0,0 +1,27 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCCodecSpecificInfo.h"
+#import "RTCMacros.h"
+
+/** Class for H264 specific config. */
+typedef NS_ENUM(NSUInteger, RTCH264PacketizationMode) {
+  RTCH264PacketizationModeNonInterleaved = 0,  // Mode 1 - STAP-A, FU-A is allowed
+  RTCH264PacketizationModeSingleNalUnit        // Mode 0 - only single NALU allowed
+};
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCCodecSpecificInfoH264) : NSObject <RTC_OBJC_TYPE(RTCCodecSpecificInfo)>
+
+@property(nonatomic, assign) RTCH264PacketizationMode packetizationMode;
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/components/video_codec/RTCCodecSpecificInfoH264.mm b/third_party/libwebrtc/sdk/objc/components/video_codec/RTCCodecSpecificInfoH264.mm
new file mode 100644
index 0000000000..e38ed307b3
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/components/video_codec/RTCCodecSpecificInfoH264.mm
@@ -0,0 +1,29 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCCodecSpecificInfoH264+Private.h"
+
+#import "RTCH264ProfileLevelId.h"
+
+// H264 specific settings.
+@implementation RTC_OBJC_TYPE (RTCCodecSpecificInfoH264)
+
+@synthesize packetizationMode = _packetizationMode;
+
+- (webrtc::CodecSpecificInfo)nativeCodecSpecificInfo {
+  webrtc::CodecSpecificInfo codecSpecificInfo;
+  codecSpecificInfo.codecType = webrtc::kVideoCodecH264;
+  codecSpecificInfo.codecSpecific.H264.packetization_mode =
+      (webrtc::H264PacketizationMode)_packetizationMode;
+
+  return codecSpecificInfo;
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/components/video_codec/RTCDefaultVideoDecoderFactory.h b/third_party/libwebrtc/sdk/objc/components/video_codec/RTCDefaultVideoDecoderFactory.h
new file mode 100644
index 0000000000..de5a9c4684
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/components/video_codec/RTCDefaultVideoDecoderFactory.h
@@ -0,0 +1,26 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+#import "RTCVideoDecoderFactory.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+/** This decoder factory include support for all codecs bundled with WebRTC. If using custom
+ *  codecs, create custom implementations of RTCVideoEncoderFactory and
+ *  RTCVideoDecoderFactory.
+ */
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCDefaultVideoDecoderFactory) : NSObject <RTC_OBJC_TYPE(RTCVideoDecoderFactory)>
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/components/video_codec/RTCDefaultVideoDecoderFactory.m b/third_party/libwebrtc/sdk/objc/components/video_codec/RTCDefaultVideoDecoderFactory.m
new file mode 100644
index 0000000000..6e3baa8750
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/components/video_codec/RTCDefaultVideoDecoderFactory.m
@@ -0,0 +1,85 @@
+/*
+ *  Copyright 2017 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCDefaultVideoDecoderFactory.h"
+
+#import "RTCH264ProfileLevelId.h"
+#import "RTCVideoDecoderH264.h"
+#import "api/video_codec/RTCVideoCodecConstants.h"
+#import "api/video_codec/RTCVideoDecoderVP8.h"
+#import "api/video_codec/RTCVideoDecoderVP9.h"
+#import "base/RTCVideoCodecInfo.h"
+
+#if defined(RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY)
+#import "api/video_codec/RTCVideoDecoderAV1.h"  // nogncheck
+#endif
+
+@implementation RTC_OBJC_TYPE (RTCDefaultVideoDecoderFactory)
+
+- (NSArray<RTC_OBJC_TYPE(RTCVideoCodecInfo) *> *)supportedCodecs {
+  NSDictionary<NSString *, NSString *> *constrainedHighParams = @{
+    @"profile-level-id" : kRTCMaxSupportedH264ProfileLevelConstrainedHigh,
+    @"level-asymmetry-allowed" : @"1",
+    @"packetization-mode" : @"1",
+  };
+  RTC_OBJC_TYPE(RTCVideoCodecInfo) *constrainedHighInfo =
+      [[RTC_OBJC_TYPE(RTCVideoCodecInfo) alloc] initWithName:kRTCVideoCodecH264Name
+                                                  parameters:constrainedHighParams];
+
+  NSDictionary<NSString *, NSString *> *constrainedBaselineParams = @{
+    @"profile-level-id" : kRTCMaxSupportedH264ProfileLevelConstrainedBaseline,
+    @"level-asymmetry-allowed" : @"1",
+    @"packetization-mode" : @"1",
+  };
+  RTC_OBJC_TYPE(RTCVideoCodecInfo) *constrainedBaselineInfo =
+      [[RTC_OBJC_TYPE(RTCVideoCodecInfo) alloc] initWithName:kRTCVideoCodecH264Name
+                                                  parameters:constrainedBaselineParams];
+
+  RTC_OBJC_TYPE(RTCVideoCodecInfo) *vp8Info =
+      [[RTC_OBJC_TYPE(RTCVideoCodecInfo) alloc] initWithName:kRTCVideoCodecVp8Name];
+
+  NSMutableArray<RTC_OBJC_TYPE(RTCVideoCodecInfo) *> *result = [@[
+    constrainedHighInfo,
+    constrainedBaselineInfo,
+    vp8Info,
+  ] mutableCopy];
+
+  if ([RTC_OBJC_TYPE(RTCVideoDecoderVP9) isSupported]) {
+    [result
+        addObject:[[RTC_OBJC_TYPE(RTCVideoCodecInfo) alloc] initWithName:kRTCVideoCodecVp9Name]];
+  }
+
+#if defined(RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY)
+  [result addObject:[[RTC_OBJC_TYPE(RTCVideoCodecInfo) alloc] initWithName:kRTCVideoCodecAv1Name]];
+#endif
+
+  return result;
+}
+
+- (id<RTC_OBJC_TYPE(RTCVideoDecoder)>)createDecoder:(RTC_OBJC_TYPE(RTCVideoCodecInfo) *)info {
+  if ([info.name isEqualToString:kRTCVideoCodecH264Name]) {
+    return [[RTC_OBJC_TYPE(RTCVideoDecoderH264) alloc] init];
+  } else if ([info.name isEqualToString:kRTCVideoCodecVp8Name]) {
+    return [RTC_OBJC_TYPE(RTCVideoDecoderVP8) vp8Decoder];
+  } else if ([info.name isEqualToString:kRTCVideoCodecVp9Name] &&
+             [RTC_OBJC_TYPE(RTCVideoDecoderVP9) isSupported]) {
+    return [RTC_OBJC_TYPE(RTCVideoDecoderVP9) vp9Decoder];
+  }
+
+#if defined(RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY)
+  if ([info.name isEqualToString:kRTCVideoCodecAv1Name]) {
+    return [RTC_OBJC_TYPE(RTCVideoDecoderAV1) av1Decoder];
+  }
+#endif
+
+  return nil;
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/components/video_codec/RTCDefaultVideoEncoderFactory.h b/third_party/libwebrtc/sdk/objc/components/video_codec/RTCDefaultVideoEncoderFactory.h
new file mode 100644
index 0000000000..92ab40c95b
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/components/video_codec/RTCDefaultVideoEncoderFactory.h
@@ -0,0 +1,31 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+#import "RTCVideoEncoderFactory.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+/** This encoder factory include support for all codecs bundled with WebRTC. If using custom
+ *  codecs, create custom implementations of RTCVideoEncoderFactory and
+ *  RTCVideoDecoderFactory.
+ */
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCDefaultVideoEncoderFactory) : NSObject <RTC_OBJC_TYPE(RTCVideoEncoderFactory)>
+
+@property(nonatomic, retain) RTC_OBJC_TYPE(RTCVideoCodecInfo) *preferredCodec;
+
++ (NSArray<RTC_OBJC_TYPE(RTCVideoCodecInfo) *> *)supportedCodecs;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/components/video_codec/RTCDefaultVideoEncoderFactory.m b/third_party/libwebrtc/sdk/objc/components/video_codec/RTCDefaultVideoEncoderFactory.m
new file mode 100644
index 0000000000..8de55bde4a
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/components/video_codec/RTCDefaultVideoEncoderFactory.m
@@ -0,0 +1,102 @@
+/*
+ *  Copyright 2017 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCDefaultVideoEncoderFactory.h"
+
+#import "RTCH264ProfileLevelId.h"
+#import "RTCVideoEncoderH264.h"
+#import "api/video_codec/RTCVideoCodecConstants.h"
+#import "api/video_codec/RTCVideoEncoderVP8.h"
+#import "api/video_codec/RTCVideoEncoderVP9.h"
+#import "base/RTCVideoCodecInfo.h"
+
+#if defined(RTC_USE_LIBAOM_AV1_ENCODER)
+#import "api/video_codec/RTCVideoEncoderAV1.h"  // nogncheck
+#endif
+
+@implementation RTC_OBJC_TYPE (RTCDefaultVideoEncoderFactory)
+
+@synthesize preferredCodec;
+
++ (NSArray<RTC_OBJC_TYPE(RTCVideoCodecInfo) *> *)supportedCodecs {
+  NSDictionary<NSString *, NSString *> *constrainedHighParams = @{
+    @"profile-level-id" : kRTCMaxSupportedH264ProfileLevelConstrainedHigh,
+    @"level-asymmetry-allowed" : @"1",
+    @"packetization-mode" : @"1",
+  };
+  RTC_OBJC_TYPE(RTCVideoCodecInfo) *constrainedHighInfo =
+      [[RTC_OBJC_TYPE(RTCVideoCodecInfo) alloc] initWithName:kRTCVideoCodecH264Name
+                                                  parameters:constrainedHighParams];
+
+  NSDictionary<NSString *, NSString *> *constrainedBaselineParams = @{
+    @"profile-level-id" : kRTCMaxSupportedH264ProfileLevelConstrainedBaseline,
+    @"level-asymmetry-allowed" : @"1",
+    @"packetization-mode" : @"1",
+  };
+  RTC_OBJC_TYPE(RTCVideoCodecInfo) *constrainedBaselineInfo =
+      [[RTC_OBJC_TYPE(RTCVideoCodecInfo) alloc] initWithName:kRTCVideoCodecH264Name
+                                                  parameters:constrainedBaselineParams];
+
+  RTC_OBJC_TYPE(RTCVideoCodecInfo) *vp8Info =
+      [[RTC_OBJC_TYPE(RTCVideoCodecInfo) alloc] initWithName:kRTCVideoCodecVp8Name];
+
+  NSMutableArray<RTC_OBJC_TYPE(RTCVideoCodecInfo) *> *result = [@[
+    constrainedHighInfo,
+    constrainedBaselineInfo,
+    vp8Info,
+  ] mutableCopy];
+
+  if ([RTC_OBJC_TYPE(RTCVideoEncoderVP9) isSupported]) {
+    [result
+        addObject:[[RTC_OBJC_TYPE(RTCVideoCodecInfo) alloc] initWithName:kRTCVideoCodecVp9Name]];
+  }
+
+#if defined(RTC_USE_LIBAOM_AV1_ENCODER)
+  [result addObject:[[RTC_OBJC_TYPE(RTCVideoCodecInfo) alloc] initWithName:kRTCVideoCodecAv1Name]];
+#endif
+
+  return result;
+}
+
+- (id<RTC_OBJC_TYPE(RTCVideoEncoder)>)createEncoder:(RTC_OBJC_TYPE(RTCVideoCodecInfo) *)info {
+  if ([info.name isEqualToString:kRTCVideoCodecH264Name]) {
+    return [[RTC_OBJC_TYPE(RTCVideoEncoderH264) alloc] initWithCodecInfo:info];
+  } else if ([info.name isEqualToString:kRTCVideoCodecVp8Name]) {
+    return [RTC_OBJC_TYPE(RTCVideoEncoderVP8) vp8Encoder];
+  } else if ([info.name isEqualToString:kRTCVideoCodecVp9Name] &&
+             [RTC_OBJC_TYPE(RTCVideoEncoderVP9) isSupported]) {
+    return [RTC_OBJC_TYPE(RTCVideoEncoderVP9) vp9Encoder];
+  }
+
+#if defined(RTC_USE_LIBAOM_AV1_ENCODER)
+  if ([info.name isEqualToString:kRTCVideoCodecAv1Name]) {
+    return [RTC_OBJC_TYPE(RTCVideoEncoderAV1) av1Encoder];
+  }
+#endif
+
+  return nil;
+}
+
+- (NSArray<RTC_OBJC_TYPE(RTCVideoCodecInfo) *> *)supportedCodecs {
+  NSMutableArray<RTC_OBJC_TYPE(RTCVideoCodecInfo) *> *codecs =
+      [[[self class] supportedCodecs] mutableCopy];
+
+  NSMutableArray<RTC_OBJC_TYPE(RTCVideoCodecInfo) *> *orderedCodecs = [NSMutableArray array];
+  NSUInteger index = [codecs indexOfObject:self.preferredCodec];
+  if (index != NSNotFound) {
+    [orderedCodecs addObject:[codecs objectAtIndex:index]];
+    [codecs removeObjectAtIndex:index];
+  }
+  [orderedCodecs addObjectsFromArray:codecs];
+
+  return [orderedCodecs copy];
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/components/video_codec/RTCH264ProfileLevelId.h b/third_party/libwebrtc/sdk/objc/components/video_codec/RTCH264ProfileLevelId.h
new file mode 100644
index 0000000000..dac7bb5610
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/components/video_codec/RTCH264ProfileLevelId.h
@@ -0,0 +1,60 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+RTC_OBJC_EXPORT extern NSString *const kRTCVideoCodecH264Name;
+RTC_OBJC_EXPORT extern NSString *const kRTCLevel31ConstrainedHigh;
+RTC_OBJC_EXPORT extern NSString *const kRTCLevel31ConstrainedBaseline;
+RTC_OBJC_EXPORT extern NSString *const kRTCMaxSupportedH264ProfileLevelConstrainedHigh;
+RTC_OBJC_EXPORT extern NSString *const kRTCMaxSupportedH264ProfileLevelConstrainedBaseline;
+
+/** H264 Profiles and levels. */
+typedef NS_ENUM(NSUInteger, RTCH264Profile) {
+  RTCH264ProfileConstrainedBaseline,
+  RTCH264ProfileBaseline,
+  RTCH264ProfileMain,
+  RTCH264ProfileConstrainedHigh,
+  RTCH264ProfileHigh,
+};
+
+typedef NS_ENUM(NSUInteger, RTCH264Level) {
+  RTCH264Level1_b = 0,
+  RTCH264Level1 = 10,
+  RTCH264Level1_1 = 11,
+  RTCH264Level1_2 = 12,
+  RTCH264Level1_3 = 13,
+  RTCH264Level2 = 20,
+  RTCH264Level2_1 = 21,
+  RTCH264Level2_2 = 22,
+  RTCH264Level3 = 30,
+  RTCH264Level3_1 = 31,
+  RTCH264Level3_2 = 32,
+  RTCH264Level4 = 40,
+  RTCH264Level4_1 = 41,
+  RTCH264Level4_2 = 42,
+  RTCH264Level5 = 50,
+  RTCH264Level5_1 = 51,
+  RTCH264Level5_2 = 52
+};
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCH264ProfileLevelId) : NSObject
+
+@property(nonatomic, readonly) RTCH264Profile profile;
+@property(nonatomic, readonly) RTCH264Level level;
+@property(nonatomic, readonly) NSString *hexString;
+
+- (instancetype)initWithHexString:(NSString *)hexString;
+- (instancetype)initWithProfile:(RTCH264Profile)profile level:(RTCH264Level)level;
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/components/video_codec/RTCH264ProfileLevelId.mm b/third_party/libwebrtc/sdk/objc/components/video_codec/RTCH264ProfileLevelId.mm
new file mode 100644
index 0000000000..f0ef3ec232
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/components/video_codec/RTCH264ProfileLevelId.mm
@@ -0,0 +1,120 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ *
+ */
+
+#import "RTCH264ProfileLevelId.h"
+
+#import "helpers/NSString+StdString.h"
+#if defined(WEBRTC_IOS)
+#import "UIDevice+H264Profile.h"
+#endif
+
+#include "api/video_codecs/h264_profile_level_id.h"
+#include "media/base/media_constants.h"
+
+namespace {
+
+NSString *MaxSupportedProfileLevelConstrainedHigh();
+NSString *MaxSupportedProfileLevelConstrainedBaseline();
+
+}  // namespace
+
+NSString *const kRTCVideoCodecH264Name = @(cricket::kH264CodecName);
+NSString *const kRTCLevel31ConstrainedHigh = @"640c1f";
+NSString *const kRTCLevel31ConstrainedBaseline = @"42e01f";
+NSString *const kRTCMaxSupportedH264ProfileLevelConstrainedHigh =
+    MaxSupportedProfileLevelConstrainedHigh();
+NSString *const kRTCMaxSupportedH264ProfileLevelConstrainedBaseline =
+    MaxSupportedProfileLevelConstrainedBaseline();
+
+namespace {
+
+#if defined(WEBRTC_IOS)
+
+NSString *MaxSupportedLevelForProfile(webrtc::H264Profile profile) {
+  const absl::optional<webrtc::H264ProfileLevelId> profileLevelId =
+      [UIDevice maxSupportedH264Profile];
+  if (profileLevelId && profileLevelId->profile >= profile) {
+    const absl::optional<std::string> profileString =
+        H264ProfileLevelIdToString(webrtc::H264ProfileLevelId(profile, profileLevelId->level));
+    if (profileString) {
+      return [NSString stringForStdString:*profileString];
+    }
+  }
+  return nil;
+}
+#endif
+
+NSString *MaxSupportedProfileLevelConstrainedBaseline() {
+#if defined(WEBRTC_IOS)
+  NSString *profile = MaxSupportedLevelForProfile(webrtc::H264Profile::kProfileConstrainedBaseline);
+  if (profile != nil) {
+    return profile;
+  }
+#endif
+  return kRTCLevel31ConstrainedBaseline;
+}
+
+NSString *MaxSupportedProfileLevelConstrainedHigh() {
+#if defined(WEBRTC_IOS)
+  NSString *profile = MaxSupportedLevelForProfile(webrtc::H264Profile::kProfileConstrainedHigh);
+  if (profile != nil) {
+    return profile;
+  }
+#endif
+  return kRTCLevel31ConstrainedHigh;
+}
+
+}  // namespace
+
+@interface RTC_OBJC_TYPE (RTCH264ProfileLevelId)
+()
+
+    @property(nonatomic, assign) RTCH264Profile profile;
+@property(nonatomic, assign) RTCH264Level level;
+@property(nonatomic, strong) NSString *hexString;
+
+@end
+
+@implementation RTC_OBJC_TYPE (RTCH264ProfileLevelId)
+
+@synthesize profile = _profile;
+@synthesize level = _level;
+@synthesize hexString = _hexString;
+
+- (instancetype)initWithHexString:(NSString *)hexString {
+  if (self = [super init]) {
+    self.hexString = hexString;
+
+    absl::optional<webrtc::H264ProfileLevelId> profile_level_id =
+        webrtc::ParseH264ProfileLevelId([hexString cStringUsingEncoding:NSUTF8StringEncoding]);
+    if (profile_level_id.has_value()) {
+      self.profile = static_cast<RTCH264Profile>(profile_level_id->profile);
+      self.level = static_cast<RTCH264Level>(profile_level_id->level);
+    }
+  }
+  return self;
+}
+
+- (instancetype)initWithProfile:(RTCH264Profile)profile level:(RTCH264Level)level {
+  if (self = [super init]) {
+    self.profile = profile;
+    self.level = level;
+
+    absl::optional<std::string> hex_string =
+        webrtc::H264ProfileLevelIdToString(webrtc::H264ProfileLevelId(
+            static_cast<webrtc::H264Profile>(profile), static_cast<webrtc::H264Level>(level)));
+    self.hexString =
+        [NSString stringWithCString:hex_string.value_or("").c_str() encoding:NSUTF8StringEncoding];
+  }
+  return self;
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/components/video_codec/RTCVideoDecoderFactoryH264.h b/third_party/libwebrtc/sdk/objc/components/video_codec/RTCVideoDecoderFactoryH264.h
new file mode 100644
index 0000000000..88bacbbdfe
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/components/video_codec/RTCVideoDecoderFactoryH264.h
@@ -0,0 +1,18 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+#import "RTCVideoDecoderFactory.h"
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCVideoDecoderFactoryH264) : NSObject <RTC_OBJC_TYPE(RTCVideoDecoderFactory)>
+@end
diff --git a/third_party/libwebrtc/sdk/objc/components/video_codec/RTCVideoDecoderFactoryH264.m b/third_party/libwebrtc/sdk/objc/components/video_codec/RTCVideoDecoderFactoryH264.m
new file mode 100644
index 0000000000..bdae19d687
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/components/video_codec/RTCVideoDecoderFactoryH264.m
@@ -0,0 +1,49 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCVideoDecoderFactoryH264.h"
+
+#import "RTCH264ProfileLevelId.h"
+#import "RTCVideoDecoderH264.h"
+
+@implementation RTC_OBJC_TYPE (RTCVideoDecoderFactoryH264)
+
+- (NSArray<RTC_OBJC_TYPE(RTCVideoCodecInfo) *> *)supportedCodecs {
+  NSMutableArray<RTC_OBJC_TYPE(RTCVideoCodecInfo) *> *codecs = [NSMutableArray array];
+  NSString *codecName = kRTCVideoCodecH264Name;
+
+  NSDictionary<NSString *, NSString *> *constrainedHighParams = @{
+    @"profile-level-id" : kRTCMaxSupportedH264ProfileLevelConstrainedHigh,
+    @"level-asymmetry-allowed" : @"1",
+    @"packetization-mode" : @"1",
+  };
+  RTC_OBJC_TYPE(RTCVideoCodecInfo) *constrainedHighInfo =
+      [[RTC_OBJC_TYPE(RTCVideoCodecInfo) alloc] initWithName:codecName
+                                                  parameters:constrainedHighParams];
+  [codecs addObject:constrainedHighInfo];
+
+  NSDictionary<NSString *, NSString *> *constrainedBaselineParams = @{
+    @"profile-level-id" : kRTCMaxSupportedH264ProfileLevelConstrainedBaseline,
+    @"level-asymmetry-allowed" : @"1",
+    @"packetization-mode" : @"1",
+  };
+  RTC_OBJC_TYPE(RTCVideoCodecInfo) *constrainedBaselineInfo =
+      [[RTC_OBJC_TYPE(RTCVideoCodecInfo) alloc] initWithName:codecName
+                                                  parameters:constrainedBaselineParams];
+  [codecs addObject:constrainedBaselineInfo];
+
+  return [codecs copy];
+}
+
+- (id<RTC_OBJC_TYPE(RTCVideoDecoder)>)createDecoder:(RTC_OBJC_TYPE(RTCVideoCodecInfo) *)info {
+  return [[RTC_OBJC_TYPE(RTCVideoDecoderH264) alloc] init];
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/components/video_codec/RTCVideoDecoderH264.h b/third_party/libwebrtc/sdk/objc/components/video_codec/RTCVideoDecoderH264.h
new file mode 100644
index 0000000000..a12e4212a7
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/components/video_codec/RTCVideoDecoderH264.h
@@ -0,0 +1,18 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+#import "RTCVideoDecoder.h"
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCVideoDecoderH264) : NSObject <RTC_OBJC_TYPE(RTCVideoDecoder)>
+@end
diff --git a/third_party/libwebrtc/sdk/objc/components/video_codec/RTCVideoDecoderH264.mm b/third_party/libwebrtc/sdk/objc/components/video_codec/RTCVideoDecoderH264.mm
new file mode 100644
index 0000000000..09e642bc37
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/components/video_codec/RTCVideoDecoderH264.mm
@@ -0,0 +1,276 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ *
+ */
+
+#import "RTCVideoDecoderH264.h"
+
+#import <VideoToolbox/VideoToolbox.h>
+
+#import "base/RTCVideoFrame.h"
+#import "base/RTCVideoFrameBuffer.h"
+#import "components/video_frame_buffer/RTCCVPixelBuffer.h"
+#import "helpers.h"
+#import "helpers/scoped_cftyperef.h"
+
+#if defined(WEBRTC_IOS)
+#import "helpers/UIDevice+RTCDevice.h"
+#endif
+
+#include "modules/video_coding/include/video_error_codes.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/time_utils.h"
+#include "sdk/objc/components/video_codec/nalu_rewriter.h"
+
+// Struct that we pass to the decoder per frame to decode. We receive it again
+// in the decoder callback.
+struct RTCFrameDecodeParams {
+  RTCFrameDecodeParams(RTCVideoDecoderCallback cb, int64_t ts) : callback(cb), timestamp(ts) {}
+  RTCVideoDecoderCallback callback;
+  int64_t timestamp;
+};
+
+@interface RTC_OBJC_TYPE (RTCVideoDecoderH264)
+() - (void)setError : (OSStatus)error;
+@end
+
+// This is the callback function that VideoToolbox calls when decode is
+// complete.
+void decompressionOutputCallback(void *decoderRef,
+                                 void *params,
+                                 OSStatus status,
+                                 VTDecodeInfoFlags infoFlags,
+                                 CVImageBufferRef imageBuffer,
+                                 CMTime timestamp,
+                                 CMTime duration) {
+  std::unique_ptr<RTCFrameDecodeParams> decodeParams(
+      reinterpret_cast<RTCFrameDecodeParams *>(params));
+  if (status != noErr) {
+    RTC_OBJC_TYPE(RTCVideoDecoderH264) *decoder =
+        (__bridge RTC_OBJC_TYPE(RTCVideoDecoderH264) *)decoderRef;
+    [decoder setError:status];
+    RTC_LOG(LS_ERROR) << "Failed to decode frame. Status: " << status;
+    return;
+  }
+  // TODO(tkchin): Handle CVO properly.
+  RTC_OBJC_TYPE(RTCCVPixelBuffer) *frameBuffer =
+      [[RTC_OBJC_TYPE(RTCCVPixelBuffer) alloc] initWithPixelBuffer:imageBuffer];
+  RTC_OBJC_TYPE(RTCVideoFrame) *decodedFrame = [[RTC_OBJC_TYPE(RTCVideoFrame) alloc]
+      initWithBuffer:frameBuffer
+            rotation:RTCVideoRotation_0
+         timeStampNs:CMTimeGetSeconds(timestamp) * rtc::kNumNanosecsPerSec];
+  decodedFrame.timeStamp = decodeParams->timestamp;
+  decodeParams->callback(decodedFrame);
+}
+
+// Decoder.
+@implementation RTC_OBJC_TYPE (RTCVideoDecoderH264) {
+  CMVideoFormatDescriptionRef _videoFormat;
+  CMMemoryPoolRef _memoryPool;
+  VTDecompressionSessionRef _decompressionSession;
+  RTCVideoDecoderCallback _callback;
+  OSStatus _error;
+}
+
+- (instancetype)init {
+  self = [super init];
+  if (self) {
+    _memoryPool = CMMemoryPoolCreate(nil);
+  }
+  return self;
+}
+
+- (void)dealloc {
+  CMMemoryPoolInvalidate(_memoryPool);
+  CFRelease(_memoryPool);
+  [self destroyDecompressionSession];
+  [self setVideoFormat:nullptr];
+}
+
+- (NSInteger)startDecodeWithNumberOfCores:(int)numberOfCores {
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+- (NSInteger)decode:(RTC_OBJC_TYPE(RTCEncodedImage) *)inputImage
+        missingFrames:(BOOL)missingFrames
+    codecSpecificInfo:(nullable id<RTC_OBJC_TYPE(RTCCodecSpecificInfo)>)info
+         renderTimeMs:(int64_t)renderTimeMs {
+  RTC_DCHECK(inputImage.buffer);
+
+  if (_error != noErr) {
+    RTC_LOG(LS_WARNING) << "Last frame decode failed.";
+    _error = noErr;
+    return WEBRTC_VIDEO_CODEC_ERROR;
+  }
+
+  rtc::ScopedCFTypeRef<CMVideoFormatDescriptionRef> inputFormat =
+      rtc::ScopedCF(webrtc::CreateVideoFormatDescription((uint8_t *)inputImage.buffer.bytes,
+                                                         inputImage.buffer.length));
+  if (inputFormat) {
+    // Check if the video format has changed, and reinitialize decoder if
+    // needed.
+    if (!CMFormatDescriptionEqual(inputFormat.get(), _videoFormat)) {
+      [self setVideoFormat:inputFormat.get()];
+      int resetDecompressionSessionError = [self resetDecompressionSession];
+      if (resetDecompressionSessionError != WEBRTC_VIDEO_CODEC_OK) {
+        return resetDecompressionSessionError;
+      }
+    }
+  }
+  if (!_videoFormat) {
+    // We received a frame but we don't have format information so we can't
+    // decode it.
+    // This can happen after backgrounding. We need to wait for the next
+    // sps/pps before we can resume so we request a keyframe by returning an
+    // error.
+    RTC_LOG(LS_WARNING) << "Missing video format. Frame with sps/pps required.";
+    return WEBRTC_VIDEO_CODEC_ERROR;
+  }
+  CMSampleBufferRef sampleBuffer = nullptr;
+  if (!webrtc::H264AnnexBBufferToCMSampleBuffer((uint8_t *)inputImage.buffer.bytes,
+                                                inputImage.buffer.length,
+                                                _videoFormat,
+                                                &sampleBuffer,
+                                                _memoryPool)) {
+    return WEBRTC_VIDEO_CODEC_ERROR;
+  }
+  RTC_DCHECK(sampleBuffer);
+  VTDecodeFrameFlags decodeFlags = kVTDecodeFrame_EnableAsynchronousDecompression;
+  std::unique_ptr<RTCFrameDecodeParams> frameDecodeParams;
+  frameDecodeParams.reset(new RTCFrameDecodeParams(_callback, inputImage.timeStamp));
+  OSStatus status = VTDecompressionSessionDecodeFrame(
+      _decompressionSession, sampleBuffer, decodeFlags, frameDecodeParams.release(), nullptr);
+#if defined(WEBRTC_IOS)
+  // Re-initialize the decoder if we have an invalid session while the app is
+  // active or decoder malfunctions and retry the decode request.
+  if ((status == kVTInvalidSessionErr || status == kVTVideoDecoderMalfunctionErr) &&
+      [self resetDecompressionSession] == WEBRTC_VIDEO_CODEC_OK) {
+    RTC_LOG(LS_INFO) << "Failed to decode frame with code: " << status
+                     << " retrying decode after decompression session reset";
+    frameDecodeParams.reset(new RTCFrameDecodeParams(_callback, inputImage.timeStamp));
+    status = VTDecompressionSessionDecodeFrame(
+        _decompressionSession, sampleBuffer, decodeFlags, frameDecodeParams.release(), nullptr);
+  }
+#endif
+  CFRelease(sampleBuffer);
+  if (status != noErr) {
+    RTC_LOG(LS_ERROR) << "Failed to decode frame with code: " << status;
+    return WEBRTC_VIDEO_CODEC_ERROR;
+  }
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+- (void)setCallback:(RTCVideoDecoderCallback)callback {
+  _callback = callback;
+}
+
+- (void)setError:(OSStatus)error {
+  _error = error;
+}
+
+- (NSInteger)releaseDecoder {
+  // Need to invalidate the session so that callbacks no longer occur and it
+  // is safe to null out the callback.
+  [self destroyDecompressionSession];
+  [self setVideoFormat:nullptr];
+  _callback = nullptr;
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+#pragma mark - Private
+
+- (int)resetDecompressionSession {
+  [self destroyDecompressionSession];
+
+  // Need to wait for the first SPS to initialize decoder.
+  if (!_videoFormat) {
+    return WEBRTC_VIDEO_CODEC_OK;
+  }
+
+  // Set keys for OpenGL and IOSurface compatibilty, which makes the encoder
+  // create pixel buffers with GPU backed memory. The intent here is to pass
+  // the pixel buffers directly so we avoid a texture upload later during
+  // rendering. This currently is moot because we are converting back to an
+  // I420 frame after decode, but eventually we will be able to plumb
+  // CVPixelBuffers directly to the renderer.
+  // TODO(tkchin): Maybe only set OpenGL/IOSurface keys if we know that that
+  // we can pass CVPixelBuffers as native handles in decoder output.
+  NSDictionary *attributes = @{
+#if defined(WEBRTC_IOS) && (TARGET_OS_MACCATALYST || TARGET_OS_SIMULATOR)
+    (NSString *)kCVPixelBufferMetalCompatibilityKey : @(YES),
+#elif defined(WEBRTC_IOS)
+    (NSString *)kCVPixelBufferOpenGLESCompatibilityKey : @(YES),
+#elif defined(WEBRTC_MAC) && !defined(WEBRTC_ARCH_ARM64)
+    (NSString *)kCVPixelBufferOpenGLCompatibilityKey : @(YES),
+#endif
+#if !(TARGET_OS_SIMULATOR)
+    (NSString *)kCVPixelBufferIOSurfacePropertiesKey : @{},
+#endif
+    (NSString *)
+    kCVPixelBufferPixelFormatTypeKey : @(kCVPixelFormatType_420YpCbCr8BiPlanarFullRange),
+  };
+
+  VTDecompressionOutputCallbackRecord record = {
+      decompressionOutputCallback, (__bridge void *)self,
+  };
+  OSStatus status = VTDecompressionSessionCreate(nullptr,
+                                                 _videoFormat,
+                                                 nullptr,
+                                                 (__bridge CFDictionaryRef)attributes,
+                                                 &record,
+                                                 &_decompressionSession);
+  if (status != noErr) {
+    RTC_LOG(LS_ERROR) << "Failed to create decompression session: " << status;
+    [self destroyDecompressionSession];
+    return WEBRTC_VIDEO_CODEC_ERROR;
+  }
+  [self configureDecompressionSession];
+
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+- (void)configureDecompressionSession {
+  RTC_DCHECK(_decompressionSession);
+#if defined(WEBRTC_IOS)
+  VTSessionSetProperty(_decompressionSession, kVTDecompressionPropertyKey_RealTime, kCFBooleanTrue);
+#endif
+}
+
+- (void)destroyDecompressionSession {
+  if (_decompressionSession) {
+#if defined(WEBRTC_IOS)
+    if ([UIDevice isIOS11OrLater]) {
+      VTDecompressionSessionWaitForAsynchronousFrames(_decompressionSession);
+    }
+#endif
+    VTDecompressionSessionInvalidate(_decompressionSession);
+    CFRelease(_decompressionSession);
+    _decompressionSession = nullptr;
+  }
+}
+
+- (void)setVideoFormat:(CMVideoFormatDescriptionRef)videoFormat {
+  if (_videoFormat == videoFormat) {
+    return;
+  }
+  if (_videoFormat) {
+    CFRelease(_videoFormat);
+  }
+  _videoFormat = videoFormat;
+  if (_videoFormat) {
+    CFRetain(_videoFormat);
+  }
+}
+
+- (NSString *)implementationName {
+  return @"VideoToolbox";
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/components/video_codec/RTCVideoEncoderFactoryH264.h b/third_party/libwebrtc/sdk/objc/components/video_codec/RTCVideoEncoderFactoryH264.h
new file mode 100644
index 0000000000..45fc4be2ea
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/components/video_codec/RTCVideoEncoderFactoryH264.h
@@ -0,0 +1,18 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+#import "RTCVideoEncoderFactory.h"
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCVideoEncoderFactoryH264) : NSObject <RTC_OBJC_TYPE(RTCVideoEncoderFactory)>
+@end
diff --git a/third_party/libwebrtc/sdk/objc/components/video_codec/RTCVideoEncoderFactoryH264.m b/third_party/libwebrtc/sdk/objc/components/video_codec/RTCVideoEncoderFactoryH264.m
new file mode 100644
index 0000000000..9843849307
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/components/video_codec/RTCVideoEncoderFactoryH264.m
@@ -0,0 +1,49 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCVideoEncoderFactoryH264.h"
+
+#import "RTCH264ProfileLevelId.h"
+#import "RTCVideoEncoderH264.h"
+
+@implementation RTC_OBJC_TYPE (RTCVideoEncoderFactoryH264)
+
+- (NSArray<RTC_OBJC_TYPE(RTCVideoCodecInfo) *> *)supportedCodecs {
+  NSMutableArray<RTC_OBJC_TYPE(RTCVideoCodecInfo) *> *codecs = [NSMutableArray array];
+  NSString *codecName = kRTCVideoCodecH264Name;
+
+  NSDictionary<NSString *, NSString *> *constrainedHighParams = @{
+    @"profile-level-id" : kRTCMaxSupportedH264ProfileLevelConstrainedHigh,
+    @"level-asymmetry-allowed" : @"1",
+    @"packetization-mode" : @"1",
+  };
+  RTC_OBJC_TYPE(RTCVideoCodecInfo) *constrainedHighInfo =
+      [[RTC_OBJC_TYPE(RTCVideoCodecInfo) alloc] initWithName:codecName
+                                                  parameters:constrainedHighParams];
+  [codecs addObject:constrainedHighInfo];
+
+  NSDictionary<NSString *, NSString *> *constrainedBaselineParams = @{
+    @"profile-level-id" : kRTCMaxSupportedH264ProfileLevelConstrainedBaseline,
+    @"level-asymmetry-allowed" : @"1",
+    @"packetization-mode" : @"1",
+  };
+  RTC_OBJC_TYPE(RTCVideoCodecInfo) *constrainedBaselineInfo =
+      [[RTC_OBJC_TYPE(RTCVideoCodecInfo) alloc] initWithName:codecName
+                                                  parameters:constrainedBaselineParams];
+  [codecs addObject:constrainedBaselineInfo];
+
+  return [codecs copy];
+}
+
+- (id<RTC_OBJC_TYPE(RTCVideoEncoder)>)createEncoder:(RTC_OBJC_TYPE(RTCVideoCodecInfo) *)info {
+  return [[RTC_OBJC_TYPE(RTCVideoEncoderH264) alloc] initWithCodecInfo:info];
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/components/video_codec/RTCVideoEncoderH264.h b/third_party/libwebrtc/sdk/objc/components/video_codec/RTCVideoEncoderH264.h
new file mode 100644
index 0000000000..9f4f4c7c8d
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/components/video_codec/RTCVideoEncoderH264.h
@@ -0,0 +1,22 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+#import "RTCVideoCodecInfo.h"
+#import "RTCVideoEncoder.h"
+
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCVideoEncoderH264) : NSObject <RTC_OBJC_TYPE(RTCVideoEncoder)>
+
+- (instancetype)initWithCodecInfo:(RTC_OBJC_TYPE(RTCVideoCodecInfo) *)codecInfo;
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/components/video_codec/RTCVideoEncoderH264.mm b/third_party/libwebrtc/sdk/objc/components/video_codec/RTCVideoEncoderH264.mm
new file mode 100644
index 0000000000..2160d79ae5
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/components/video_codec/RTCVideoEncoderH264.mm
@@ -0,0 +1,828 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ *
+ */
+
+#import "RTCVideoEncoderH264.h"
+
+#import <VideoToolbox/VideoToolbox.h>
+#include <vector>
+
+#if defined(WEBRTC_IOS)
+#import "helpers/UIDevice+RTCDevice.h"
+#endif
+#import "RTCCodecSpecificInfoH264.h"
+#import "RTCH264ProfileLevelId.h"
+#import "api/peerconnection/RTCVideoCodecInfo+Private.h"
+#import "base/RTCCodecSpecificInfo.h"
+#import "base/RTCI420Buffer.h"
+#import "base/RTCVideoEncoder.h"
+#import "base/RTCVideoFrame.h"
+#import "base/RTCVideoFrameBuffer.h"
+#import "components/video_frame_buffer/RTCCVPixelBuffer.h"
+#import "helpers.h"
+
+#include "api/video_codecs/h264_profile_level_id.h"
+#include "common_video/h264/h264_bitstream_parser.h"
+#include "common_video/include/bitrate_adjuster.h"
+#include "modules/video_coding/include/video_error_codes.h"
+#include "rtc_base/buffer.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/time_utils.h"
+#include "sdk/objc/components/video_codec/nalu_rewriter.h"
+#include "third_party/libyuv/include/libyuv/convert_from.h"
+
+@interface RTC_OBJC_TYPE (RTCVideoEncoderH264)
+()
+
+    - (void)frameWasEncoded : (OSStatus)status flags : (VTEncodeInfoFlags)infoFlags sampleBuffer
+    : (CMSampleBufferRef)sampleBuffer codecSpecificInfo
+    : (id<RTC_OBJC_TYPE(RTCCodecSpecificInfo)>)codecSpecificInfo width : (int32_t)width height
+    : (int32_t)height renderTimeMs : (int64_t)renderTimeMs timestamp : (uint32_t)timestamp rotation
+    : (RTCVideoRotation)rotation;
+
+@end
+
+namespace {  // anonymous namespace
+
+// The ratio between kVTCompressionPropertyKey_DataRateLimits and
+// kVTCompressionPropertyKey_AverageBitRate. The data rate limit is set higher
+// than the average bit rate to avoid undershooting the target.
+const float kLimitToAverageBitRateFactor = 1.5f;
+// These thresholds deviate from the default h264 QP thresholds, as they
+// have been found to work better on devices that support VideoToolbox
+const int kLowH264QpThreshold = 28;
+const int kHighH264QpThreshold = 39;
+
+const OSType kNV12PixelFormat = kCVPixelFormatType_420YpCbCr8BiPlanarFullRange;
+
+// Struct that we pass to the encoder per frame to encode. We receive it again
+// in the encoder callback.
+struct RTCFrameEncodeParams {
+  RTCFrameEncodeParams(RTC_OBJC_TYPE(RTCVideoEncoderH264) * e,
+                       RTC_OBJC_TYPE(RTCCodecSpecificInfoH264) * csi,
+                       int32_t w,
+                       int32_t h,
+                       int64_t rtms,
+                       uint32_t ts,
+                       RTCVideoRotation r)
+      : encoder(e), width(w), height(h), render_time_ms(rtms), timestamp(ts), rotation(r) {
+    if (csi) {
+      codecSpecificInfo = csi;
+    } else {
+      codecSpecificInfo = [[RTC_OBJC_TYPE(RTCCodecSpecificInfoH264) alloc] init];
+    }
+  }
+
+  RTC_OBJC_TYPE(RTCVideoEncoderH264) * encoder;
+  RTC_OBJC_TYPE(RTCCodecSpecificInfoH264) * codecSpecificInfo;
+  int32_t width;
+  int32_t height;
+  int64_t render_time_ms;
+  uint32_t timestamp;
+  RTCVideoRotation rotation;
+};
+
+// We receive I420Frames as input, but we need to feed CVPixelBuffers into the
+// encoder. This performs the copy and format conversion.
+// TODO(tkchin): See if encoder will accept i420 frames and compare performance.
+bool CopyVideoFrameToNV12PixelBuffer(id<RTC_OBJC_TYPE(RTCI420Buffer)> frameBuffer,
+                                     CVPixelBufferRef pixelBuffer) {
+  RTC_DCHECK(pixelBuffer);
+  RTC_DCHECK_EQ(CVPixelBufferGetPixelFormatType(pixelBuffer), kNV12PixelFormat);
+  RTC_DCHECK_EQ(CVPixelBufferGetHeightOfPlane(pixelBuffer, 0), frameBuffer.height);
+  RTC_DCHECK_EQ(CVPixelBufferGetWidthOfPlane(pixelBuffer, 0), frameBuffer.width);
+
+  CVReturn cvRet = CVPixelBufferLockBaseAddress(pixelBuffer, 0);
+  if (cvRet != kCVReturnSuccess) {
+    RTC_LOG(LS_ERROR) << "Failed to lock base address: " << cvRet;
+    return false;
+  }
+  uint8_t *dstY = reinterpret_cast<uint8_t *>(CVPixelBufferGetBaseAddressOfPlane(pixelBuffer, 0));
+  int dstStrideY = CVPixelBufferGetBytesPerRowOfPlane(pixelBuffer, 0);
+  uint8_t *dstUV = reinterpret_cast<uint8_t *>(CVPixelBufferGetBaseAddressOfPlane(pixelBuffer, 1));
+  int dstStrideUV = CVPixelBufferGetBytesPerRowOfPlane(pixelBuffer, 1);
+  // Convert I420 to NV12.
+  int ret = libyuv::I420ToNV12(frameBuffer.dataY,
+                               frameBuffer.strideY,
+                               frameBuffer.dataU,
+                               frameBuffer.strideU,
+                               frameBuffer.dataV,
+                               frameBuffer.strideV,
+                               dstY,
+                               dstStrideY,
+                               dstUV,
+                               dstStrideUV,
+                               frameBuffer.width,
+                               frameBuffer.height);
+  CVPixelBufferUnlockBaseAddress(pixelBuffer, 0);
+  if (ret) {
+    RTC_LOG(LS_ERROR) << "Error converting I420 VideoFrame to NV12 :" << ret;
+    return false;
+  }
+  return true;
+}
+
+CVPixelBufferRef CreatePixelBuffer(VTCompressionSessionRef compression_session) {
+  if (!compression_session) {
+    RTC_LOG(LS_ERROR) << "Failed to get compression session.";
+    return nullptr;
+  }
+  CVPixelBufferPoolRef pixel_buffer_pool =
+      VTCompressionSessionGetPixelBufferPool(compression_session);
+
+  if (!pixel_buffer_pool) {
+    RTC_LOG(LS_ERROR) << "Failed to get pixel buffer pool.";
+    return nullptr;
+  }
+  CVPixelBufferRef pixel_buffer;
+  CVReturn ret = CVPixelBufferPoolCreatePixelBuffer(nullptr, pixel_buffer_pool, &pixel_buffer);
+  if (ret != kCVReturnSuccess) {
+    RTC_LOG(LS_ERROR) << "Failed to create pixel buffer: " << ret;
+    // We probably want to drop frames here, since failure probably means
+    // that the pool is empty.
+    return nullptr;
+  }
+  return pixel_buffer;
+}
+
+// This is the callback function that VideoToolbox calls when encode is
+// complete. From inspection this happens on its own queue.
+void compressionOutputCallback(void *encoder,
+                               void *params,
+                               OSStatus status,
+                               VTEncodeInfoFlags infoFlags,
+                               CMSampleBufferRef sampleBuffer) {
+  if (!params) {
+    // If there are pending callbacks when the encoder is destroyed, this can happen.
+    return;
+  }
+  std::unique_ptr<RTCFrameEncodeParams> encodeParams(
+      reinterpret_cast<RTCFrameEncodeParams *>(params));
+  [encodeParams->encoder frameWasEncoded:status
+                                   flags:infoFlags
+                            sampleBuffer:sampleBuffer
+                       codecSpecificInfo:encodeParams->codecSpecificInfo
+                                   width:encodeParams->width
+                                  height:encodeParams->height
+                            renderTimeMs:encodeParams->render_time_ms
+                               timestamp:encodeParams->timestamp
+                                rotation:encodeParams->rotation];
+}
+
+// Extract VideoToolbox profile out of the webrtc::SdpVideoFormat. If there is
+// no specific VideoToolbox profile for the specified level, AutoLevel will be
+// returned. The user must initialize the encoder with a resolution and
+// framerate conforming to the selected H264 level regardless.
+CFStringRef ExtractProfile(const webrtc::H264ProfileLevelId &profile_level_id) {
+  switch (profile_level_id.profile) {
+    case webrtc::H264Profile::kProfileConstrainedBaseline:
+    case webrtc::H264Profile::kProfileBaseline:
+      switch (profile_level_id.level) {
+        case webrtc::H264Level::kLevel3:
+          return kVTProfileLevel_H264_Baseline_3_0;
+        case webrtc::H264Level::kLevel3_1:
+          return kVTProfileLevel_H264_Baseline_3_1;
+        case webrtc::H264Level::kLevel3_2:
+          return kVTProfileLevel_H264_Baseline_3_2;
+        case webrtc::H264Level::kLevel4:
+          return kVTProfileLevel_H264_Baseline_4_0;
+        case webrtc::H264Level::kLevel4_1:
+          return kVTProfileLevel_H264_Baseline_4_1;
+        case webrtc::H264Level::kLevel4_2:
+          return kVTProfileLevel_H264_Baseline_4_2;
+        case webrtc::H264Level::kLevel5:
+          return kVTProfileLevel_H264_Baseline_5_0;
+        case webrtc::H264Level::kLevel5_1:
+          return kVTProfileLevel_H264_Baseline_5_1;
+        case webrtc::H264Level::kLevel5_2:
+          return kVTProfileLevel_H264_Baseline_5_2;
+        case webrtc::H264Level::kLevel1:
+        case webrtc::H264Level::kLevel1_b:
+        case webrtc::H264Level::kLevel1_1:
+        case webrtc::H264Level::kLevel1_2:
+        case webrtc::H264Level::kLevel1_3:
+        case webrtc::H264Level::kLevel2:
+        case webrtc::H264Level::kLevel2_1:
+        case webrtc::H264Level::kLevel2_2:
+          return kVTProfileLevel_H264_Baseline_AutoLevel;
+      }
+
+    case webrtc::H264Profile::kProfileMain:
+      switch (profile_level_id.level) {
+        case webrtc::H264Level::kLevel3:
+          return kVTProfileLevel_H264_Main_3_0;
+        case webrtc::H264Level::kLevel3_1:
+          return kVTProfileLevel_H264_Main_3_1;
+        case webrtc::H264Level::kLevel3_2:
+          return kVTProfileLevel_H264_Main_3_2;
+        case webrtc::H264Level::kLevel4:
+          return kVTProfileLevel_H264_Main_4_0;
+        case webrtc::H264Level::kLevel4_1:
+          return kVTProfileLevel_H264_Main_4_1;
+        case webrtc::H264Level::kLevel4_2:
+          return kVTProfileLevel_H264_Main_4_2;
+        case webrtc::H264Level::kLevel5:
+          return kVTProfileLevel_H264_Main_5_0;
+        case webrtc::H264Level::kLevel5_1:
+          return kVTProfileLevel_H264_Main_5_1;
+        case webrtc::H264Level::kLevel5_2:
+          return kVTProfileLevel_H264_Main_5_2;
+        case webrtc::H264Level::kLevel1:
+        case webrtc::H264Level::kLevel1_b:
+        case webrtc::H264Level::kLevel1_1:
+        case webrtc::H264Level::kLevel1_2:
+        case webrtc::H264Level::kLevel1_3:
+        case webrtc::H264Level::kLevel2:
+        case webrtc::H264Level::kLevel2_1:
+        case webrtc::H264Level::kLevel2_2:
+          return kVTProfileLevel_H264_Main_AutoLevel;
+      }
+
+    case webrtc::H264Profile::kProfileConstrainedHigh:
+    case webrtc::H264Profile::kProfileHigh:
+    case webrtc::H264Profile::kProfilePredictiveHigh444:
+      switch (profile_level_id.level) {
+        case webrtc::H264Level::kLevel3:
+          return kVTProfileLevel_H264_High_3_0;
+        case webrtc::H264Level::kLevel3_1:
+          return kVTProfileLevel_H264_High_3_1;
+        case webrtc::H264Level::kLevel3_2:
+          return kVTProfileLevel_H264_High_3_2;
+        case webrtc::H264Level::kLevel4:
+          return kVTProfileLevel_H264_High_4_0;
+        case webrtc::H264Level::kLevel4_1:
+          return kVTProfileLevel_H264_High_4_1;
+        case webrtc::H264Level::kLevel4_2:
+          return kVTProfileLevel_H264_High_4_2;
+        case webrtc::H264Level::kLevel5:
+          return kVTProfileLevel_H264_High_5_0;
+        case webrtc::H264Level::kLevel5_1:
+          return kVTProfileLevel_H264_High_5_1;
+        case webrtc::H264Level::kLevel5_2:
+          return kVTProfileLevel_H264_High_5_2;
+        case webrtc::H264Level::kLevel1:
+        case webrtc::H264Level::kLevel1_b:
+        case webrtc::H264Level::kLevel1_1:
+        case webrtc::H264Level::kLevel1_2:
+        case webrtc::H264Level::kLevel1_3:
+        case webrtc::H264Level::kLevel2:
+        case webrtc::H264Level::kLevel2_1:
+        case webrtc::H264Level::kLevel2_2:
+          return kVTProfileLevel_H264_High_AutoLevel;
+      }
+  }
+}
+
+// The function returns the max allowed sample rate (pixels per second) that
+// can be processed by given encoder with `profile_level_id`.
+// See https://www.itu.int/rec/dologin_pub.asp?lang=e&id=T-REC-H.264-201610-S!!PDF-E&type=items
+// for details.
+NSUInteger GetMaxSampleRate(const webrtc::H264ProfileLevelId &profile_level_id) {
+  switch (profile_level_id.level) {
+    case webrtc::H264Level::kLevel3:
+      return 10368000;
+    case webrtc::H264Level::kLevel3_1:
+      return 27648000;
+    case webrtc::H264Level::kLevel3_2:
+      return 55296000;
+    case webrtc::H264Level::kLevel4:
+    case webrtc::H264Level::kLevel4_1:
+      return 62914560;
+    case webrtc::H264Level::kLevel4_2:
+      return 133693440;
+    case webrtc::H264Level::kLevel5:
+      return 150994944;
+    case webrtc::H264Level::kLevel5_1:
+      return 251658240;
+    case webrtc::H264Level::kLevel5_2:
+      return 530841600;
+    case webrtc::H264Level::kLevel1:
+    case webrtc::H264Level::kLevel1_b:
+    case webrtc::H264Level::kLevel1_1:
+    case webrtc::H264Level::kLevel1_2:
+    case webrtc::H264Level::kLevel1_3:
+    case webrtc::H264Level::kLevel2:
+    case webrtc::H264Level::kLevel2_1:
+    case webrtc::H264Level::kLevel2_2:
+      // Zero means auto rate setting.
+      return 0;
+  }
+}
+}  // namespace
+
+@implementation RTC_OBJC_TYPE (RTCVideoEncoderH264) {
+  RTC_OBJC_TYPE(RTCVideoCodecInfo) * _codecInfo;
+  std::unique_ptr<webrtc::BitrateAdjuster> _bitrateAdjuster;
+  uint32_t _targetBitrateBps;
+  uint32_t _encoderBitrateBps;
+  uint32_t _encoderFrameRate;
+  uint32_t _maxAllowedFrameRate;
+  RTCH264PacketizationMode _packetizationMode;
+  absl::optional<webrtc::H264ProfileLevelId> _profile_level_id;
+  RTCVideoEncoderCallback _callback;
+  int32_t _width;
+  int32_t _height;
+  VTCompressionSessionRef _compressionSession;
+  RTCVideoCodecMode _mode;
+
+  webrtc::H264BitstreamParser _h264BitstreamParser;
+  std::vector<uint8_t> _frameScaleBuffer;
+}
+
+// .5 is set as a mininum to prevent overcompensating for large temporary
+// overshoots. We don't want to degrade video quality too badly.
+// .95 is set to prevent oscillations. When a lower bitrate is set on the
+// encoder than previously set, its output seems to have a brief period of
+// drastically reduced bitrate, so we want to avoid that. In steady state
+// conditions, 0.95 seems to give us better overall bitrate over long periods
+// of time.
+- (instancetype)initWithCodecInfo:(RTC_OBJC_TYPE(RTCVideoCodecInfo) *)codecInfo {
+  if (self = [super init]) {
+    _codecInfo = codecInfo;
+    _bitrateAdjuster.reset(new webrtc::BitrateAdjuster(.5, .95));
+    _packetizationMode = RTCH264PacketizationModeNonInterleaved;
+    _profile_level_id =
+        webrtc::ParseSdpForH264ProfileLevelId([codecInfo nativeSdpVideoFormat].parameters);
+    RTC_DCHECK(_profile_level_id);
+    RTC_LOG(LS_INFO) << "Using profile " << CFStringToString(ExtractProfile(*_profile_level_id));
+    RTC_CHECK([codecInfo.name isEqualToString:kRTCVideoCodecH264Name]);
+  }
+  return self;
+}
+
+- (void)dealloc {
+  [self destroyCompressionSession];
+}
+
+- (NSInteger)startEncodeWithSettings:(RTC_OBJC_TYPE(RTCVideoEncoderSettings) *)settings
+                       numberOfCores:(int)numberOfCores {
+  RTC_DCHECK(settings);
+  RTC_DCHECK([settings.name isEqualToString:kRTCVideoCodecH264Name]);
+
+  _width = settings.width;
+  _height = settings.height;
+  _mode = settings.mode;
+
+  uint32_t aligned_width = (((_width + 15) >> 4) << 4);
+  uint32_t aligned_height = (((_height + 15) >> 4) << 4);
+  _maxAllowedFrameRate = static_cast<uint32_t>(GetMaxSampleRate(*_profile_level_id) /
+                                               (aligned_width * aligned_height));
+
+  // We can only set average bitrate on the HW encoder.
+  _targetBitrateBps = settings.startBitrate * 1000;  // startBitrate is in kbps.
+  _bitrateAdjuster->SetTargetBitrateBps(_targetBitrateBps);
+  _encoderFrameRate = MIN(settings.maxFramerate, _maxAllowedFrameRate);
+  if (settings.maxFramerate > _maxAllowedFrameRate && _maxAllowedFrameRate > 0) {
+    RTC_LOG(LS_WARNING) << "Initial encoder frame rate setting " << settings.maxFramerate
+                        << " is larger than the "
+                        << "maximal allowed frame rate " << _maxAllowedFrameRate << ".";
+  }
+
+  // TODO(tkchin): Try setting payload size via
+  // kVTCompressionPropertyKey_MaxH264SliceBytes.
+
+  return [self resetCompressionSessionWithPixelFormat:kNV12PixelFormat];
+}
+
+- (NSInteger)encode:(RTC_OBJC_TYPE(RTCVideoFrame) *)frame
+    codecSpecificInfo:(nullable id<RTC_OBJC_TYPE(RTCCodecSpecificInfo)>)codecSpecificInfo
+           frameTypes:(NSArray<NSNumber *> *)frameTypes {
+  if (!_callback || !_compressionSession) {
+    return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
+  }
+  BOOL isKeyframeRequired = NO;
+
+  // Get a pixel buffer from the pool and copy frame data over.
+  if ([self resetCompressionSessionIfNeededWithFrame:frame]) {
+    isKeyframeRequired = YES;
+  }
+
+  CVPixelBufferRef pixelBuffer = nullptr;
+  if ([frame.buffer isKindOfClass:[RTC_OBJC_TYPE(RTCCVPixelBuffer) class]]) {
+    // Native frame buffer
+    RTC_OBJC_TYPE(RTCCVPixelBuffer) *rtcPixelBuffer =
+        (RTC_OBJC_TYPE(RTCCVPixelBuffer) *)frame.buffer;
+    if (![rtcPixelBuffer requiresCropping]) {
+      // This pixel buffer might have a higher resolution than what the
+      // compression session is configured to. The compression session can
+      // handle that and will output encoded frames in the configured
+      // resolution regardless of the input pixel buffer resolution.
+      pixelBuffer = rtcPixelBuffer.pixelBuffer;
+      CVBufferRetain(pixelBuffer);
+    } else {
+      // Cropping required, we need to crop and scale to a new pixel buffer.
+      pixelBuffer = CreatePixelBuffer(_compressionSession);
+      if (!pixelBuffer) {
+        return WEBRTC_VIDEO_CODEC_ERROR;
+      }
+      int dstWidth = CVPixelBufferGetWidth(pixelBuffer);
+      int dstHeight = CVPixelBufferGetHeight(pixelBuffer);
+      if ([rtcPixelBuffer requiresScalingToWidth:dstWidth height:dstHeight]) {
+        int size =
+            [rtcPixelBuffer bufferSizeForCroppingAndScalingToWidth:dstWidth height:dstHeight];
+        _frameScaleBuffer.resize(size);
+      } else {
+        _frameScaleBuffer.clear();
+      }
+      _frameScaleBuffer.shrink_to_fit();
+      if (![rtcPixelBuffer cropAndScaleTo:pixelBuffer withTempBuffer:_frameScaleBuffer.data()]) {
+        CVBufferRelease(pixelBuffer);
+        return WEBRTC_VIDEO_CODEC_ERROR;
+      }
+    }
+  }
+
+  if (!pixelBuffer) {
+    // We did not have a native frame buffer
+    RTC_DCHECK_EQ(frame.width, _width);
+    RTC_DCHECK_EQ(frame.height, _height);
+
+    pixelBuffer = CreatePixelBuffer(_compressionSession);
+    if (!pixelBuffer) {
+      return WEBRTC_VIDEO_CODEC_ERROR;
+    }
+    RTC_DCHECK(pixelBuffer);
+    if (!CopyVideoFrameToNV12PixelBuffer([frame.buffer toI420], pixelBuffer)) {
+      RTC_LOG(LS_ERROR) << "Failed to copy frame data.";
+      CVBufferRelease(pixelBuffer);
+      return WEBRTC_VIDEO_CODEC_ERROR;
+    }
+  }
+
+  // Check if we need a keyframe.
+  if (!isKeyframeRequired && frameTypes) {
+    for (NSNumber *frameType in frameTypes) {
+      if ((RTCFrameType)frameType.intValue == RTCFrameTypeVideoFrameKey) {
+        isKeyframeRequired = YES;
+        break;
+      }
+    }
+  }
+
+  CMTime presentationTimeStamp = CMTimeMake(frame.timeStampNs / rtc::kNumNanosecsPerMillisec, 1000);
+  CFDictionaryRef frameProperties = nullptr;
+  if (isKeyframeRequired) {
+    CFTypeRef keys[] = {kVTEncodeFrameOptionKey_ForceKeyFrame};
+    CFTypeRef values[] = {kCFBooleanTrue};
+    frameProperties = CreateCFTypeDictionary(keys, values, 1);
+  }
+
+  std::unique_ptr<RTCFrameEncodeParams> encodeParams;
+  encodeParams.reset(new RTCFrameEncodeParams(self,
+                                              codecSpecificInfo,
+                                              _width,
+                                              _height,
+                                              frame.timeStampNs / rtc::kNumNanosecsPerMillisec,
+                                              frame.timeStamp,
+                                              frame.rotation));
+  encodeParams->codecSpecificInfo.packetizationMode = _packetizationMode;
+
+  // Update the bitrate if needed.
+  [self setBitrateBps:_bitrateAdjuster->GetAdjustedBitrateBps() frameRate:_encoderFrameRate];
+
+  OSStatus status = VTCompressionSessionEncodeFrame(_compressionSession,
+                                                    pixelBuffer,
+                                                    presentationTimeStamp,
+                                                    kCMTimeInvalid,
+                                                    frameProperties,
+                                                    encodeParams.release(),
+                                                    nullptr);
+  if (frameProperties) {
+    CFRelease(frameProperties);
+  }
+  if (pixelBuffer) {
+    CVBufferRelease(pixelBuffer);
+  }
+
+  if (status == kVTInvalidSessionErr) {
+    // This error occurs when entering foreground after backgrounding the app.
+    RTC_LOG(LS_ERROR) << "Invalid compression session, resetting.";
+    [self resetCompressionSessionWithPixelFormat:[self pixelFormatOfFrame:frame]];
+
+    return WEBRTC_VIDEO_CODEC_NO_OUTPUT;
+  } else if (status == kVTVideoEncoderMalfunctionErr) {
+    // Sometimes the encoder malfunctions and needs to be restarted.
+    RTC_LOG(LS_ERROR)
+        << "Encountered video encoder malfunction error. Resetting compression session.";
+    [self resetCompressionSessionWithPixelFormat:[self pixelFormatOfFrame:frame]];
+
+    return WEBRTC_VIDEO_CODEC_NO_OUTPUT;
+  } else if (status != noErr) {
+    RTC_LOG(LS_ERROR) << "Failed to encode frame with code: " << status;
+    return WEBRTC_VIDEO_CODEC_ERROR;
+  }
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+- (void)setCallback:(RTCVideoEncoderCallback)callback {
+  _callback = callback;
+}
+
+- (int)setBitrate:(uint32_t)bitrateKbit framerate:(uint32_t)framerate {
+  _targetBitrateBps = 1000 * bitrateKbit;
+  _bitrateAdjuster->SetTargetBitrateBps(_targetBitrateBps);
+  if (framerate > _maxAllowedFrameRate && _maxAllowedFrameRate > 0) {
+    RTC_LOG(LS_WARNING) << "Encoder frame rate setting " << framerate << " is larger than the "
+                        << "maximal allowed frame rate " << _maxAllowedFrameRate << ".";
+  }
+  framerate = MIN(framerate, _maxAllowedFrameRate);
+  [self setBitrateBps:_bitrateAdjuster->GetAdjustedBitrateBps() frameRate:framerate];
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+- (NSInteger)resolutionAlignment {
+  return 1;
+}
+
+- (BOOL)applyAlignmentToAllSimulcastLayers {
+  return NO;
+}
+
+- (BOOL)supportsNativeHandle {
+  return YES;
+}
+
+#pragma mark - Private
+
+- (NSInteger)releaseEncoder {
+  // Need to destroy so that the session is invalidated and won't use the
+  // callback anymore. Do not remove callback until the session is invalidated
+  // since async encoder callbacks can occur until invalidation.
+  [self destroyCompressionSession];
+  _callback = nullptr;
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+- (OSType)pixelFormatOfFrame:(RTC_OBJC_TYPE(RTCVideoFrame) *)frame {
+  // Use NV12 for non-native frames.
+  if ([frame.buffer isKindOfClass:[RTC_OBJC_TYPE(RTCCVPixelBuffer) class]]) {
+    RTC_OBJC_TYPE(RTCCVPixelBuffer) *rtcPixelBuffer =
+        (RTC_OBJC_TYPE(RTCCVPixelBuffer) *)frame.buffer;
+    return CVPixelBufferGetPixelFormatType(rtcPixelBuffer.pixelBuffer);
+  }
+
+  return kNV12PixelFormat;
+}
+
+- (BOOL)resetCompressionSessionIfNeededWithFrame:(RTC_OBJC_TYPE(RTCVideoFrame) *)frame {
+  BOOL resetCompressionSession = NO;
+
+  // If we're capturing native frames in another pixel format than the compression session is
+  // configured with, make sure the compression session is reset using the correct pixel format.
+  OSType framePixelFormat = [self pixelFormatOfFrame:frame];
+
+  if (_compressionSession) {
+    // The pool attribute `kCVPixelBufferPixelFormatTypeKey` can contain either an array of pixel
+    // formats or a single pixel format.
+
+    CVPixelBufferPoolRef pixelBufferPool =
+        VTCompressionSessionGetPixelBufferPool(_compressionSession);
+    if (!pixelBufferPool) {
+      return NO;
+    }
+
+    NSDictionary *poolAttributes =
+        (__bridge NSDictionary *)CVPixelBufferPoolGetPixelBufferAttributes(pixelBufferPool);
+    id pixelFormats =
+        [poolAttributes objectForKey:(__bridge NSString *)kCVPixelBufferPixelFormatTypeKey];
+    NSArray<NSNumber *> *compressionSessionPixelFormats = nil;
+    if ([pixelFormats isKindOfClass:[NSArray class]]) {
+      compressionSessionPixelFormats = (NSArray *)pixelFormats;
+    } else if ([pixelFormats isKindOfClass:[NSNumber class]]) {
+      compressionSessionPixelFormats = @[ (NSNumber *)pixelFormats ];
+    }
+
+    if (![compressionSessionPixelFormats
+            containsObject:[NSNumber numberWithLong:framePixelFormat]]) {
+      resetCompressionSession = YES;
+      RTC_LOG(LS_INFO) << "Resetting compression session due to non-matching pixel format.";
+    }
+  } else {
+    resetCompressionSession = YES;
+  }
+
+  if (resetCompressionSession) {
+    [self resetCompressionSessionWithPixelFormat:framePixelFormat];
+  }
+  return resetCompressionSession;
+}
+
+- (int)resetCompressionSessionWithPixelFormat:(OSType)framePixelFormat {
+  [self destroyCompressionSession];
+
+  // Set source image buffer attributes. These attributes will be present on
+  // buffers retrieved from the encoder's pixel buffer pool.
+  NSDictionary *sourceAttributes = @{
+#if defined(WEBRTC_IOS) && (TARGET_OS_MACCATALYST || TARGET_OS_SIMULATOR)
+    (NSString *)kCVPixelBufferMetalCompatibilityKey : @(YES),
+#elif defined(WEBRTC_IOS)
+    (NSString *)kCVPixelBufferOpenGLESCompatibilityKey : @(YES),
+#elif defined(WEBRTC_MAC) && !defined(WEBRTC_ARCH_ARM64)
+    (NSString *)kCVPixelBufferOpenGLCompatibilityKey : @(YES),
+#endif
+    (NSString *)kCVPixelBufferIOSurfacePropertiesKey : @{},
+    (NSString *)kCVPixelBufferPixelFormatTypeKey : @(framePixelFormat),
+  };
+
+  NSDictionary *encoder_specs;
+#if defined(WEBRTC_MAC) && !defined(WEBRTC_IOS)
+  // Currently hw accl is supported above 360p on mac, below 360p
+  // the compression session will be created with hw accl disabled.
+  encoder_specs = @{
+    (NSString *)kVTVideoEncoderSpecification_EnableHardwareAcceleratedVideoEncoder : @(YES),
+  };
+
+#endif
+  OSStatus status = VTCompressionSessionCreate(
+      nullptr,  // use default allocator
+      _width,
+      _height,
+      kCMVideoCodecType_H264,
+      (__bridge CFDictionaryRef)encoder_specs,  // use hardware accelerated encoder if available
+      (__bridge CFDictionaryRef)sourceAttributes,
+      nullptr,  // use default compressed data allocator
+      compressionOutputCallback,
+      nullptr,
+      &_compressionSession);
+  if (status != noErr) {
+    RTC_LOG(LS_ERROR) << "Failed to create compression session: " << status;
+    return WEBRTC_VIDEO_CODEC_ERROR;
+  }
+#if defined(WEBRTC_MAC) && !defined(WEBRTC_IOS)
+  CFBooleanRef hwaccl_enabled = nullptr;
+  status = VTSessionCopyProperty(_compressionSession,
+                                 kVTCompressionPropertyKey_UsingHardwareAcceleratedVideoEncoder,
+                                 nullptr,
+                                 &hwaccl_enabled);
+  if (status == noErr && (CFBooleanGetValue(hwaccl_enabled))) {
+    RTC_LOG(LS_INFO) << "Compression session created with hw accl enabled";
+  } else {
+    RTC_LOG(LS_INFO) << "Compression session created with hw accl disabled";
+  }
+#endif
+  [self configureCompressionSession];
+
+  return WEBRTC_VIDEO_CODEC_OK;
+}
+
+- (void)configureCompressionSession {
+  RTC_DCHECK(_compressionSession);
+  SetVTSessionProperty(_compressionSession, kVTCompressionPropertyKey_RealTime, true);
+  SetVTSessionProperty(_compressionSession,
+                       kVTCompressionPropertyKey_ProfileLevel,
+                       ExtractProfile(*_profile_level_id));
+  SetVTSessionProperty(_compressionSession, kVTCompressionPropertyKey_AllowFrameReordering, false);
+  [self setEncoderBitrateBps:_targetBitrateBps frameRate:_encoderFrameRate];
+  // TODO(tkchin): Look at entropy mode and colorspace matrices.
+  // TODO(tkchin): Investigate to see if there's any way to make this work.
+  // May need it to interop with Android. Currently this call just fails.
+  // On inspecting encoder output on iOS8, this value is set to 6.
+  // internal::SetVTSessionProperty(compression_session_,
+  //     kVTCompressionPropertyKey_MaxFrameDelayCount,
+  //     1);
+
+  // Set a relatively large value for keyframe emission (7200 frames or 4 minutes).
+  SetVTSessionProperty(_compressionSession, kVTCompressionPropertyKey_MaxKeyFrameInterval, 7200);
+  SetVTSessionProperty(
+      _compressionSession, kVTCompressionPropertyKey_MaxKeyFrameIntervalDuration, 240);
+}
+
+- (void)destroyCompressionSession {
+  if (_compressionSession) {
+    VTCompressionSessionInvalidate(_compressionSession);
+    CFRelease(_compressionSession);
+    _compressionSession = nullptr;
+  }
+}
+
+- (NSString *)implementationName {
+  return @"VideoToolbox";
+}
+
+- (void)setBitrateBps:(uint32_t)bitrateBps frameRate:(uint32_t)frameRate {
+  if (_encoderBitrateBps != bitrateBps || _encoderFrameRate != frameRate) {
+    [self setEncoderBitrateBps:bitrateBps frameRate:frameRate];
+  }
+}
+
+- (void)setEncoderBitrateBps:(uint32_t)bitrateBps frameRate:(uint32_t)frameRate {
+  if (_compressionSession) {
+    SetVTSessionProperty(_compressionSession, kVTCompressionPropertyKey_AverageBitRate, bitrateBps);
+
+    // With zero `_maxAllowedFrameRate`, we fall back to automatic frame rate detection.
+    if (_maxAllowedFrameRate > 0) {
+      SetVTSessionProperty(
+          _compressionSession, kVTCompressionPropertyKey_ExpectedFrameRate, frameRate);
+    }
+
+    // TODO(tkchin): Add a helper method to set array value.
+    int64_t dataLimitBytesPerSecondValue =
+        static_cast<int64_t>(bitrateBps * kLimitToAverageBitRateFactor / 8);
+    CFNumberRef bytesPerSecond =
+        CFNumberCreate(kCFAllocatorDefault, kCFNumberSInt64Type, &dataLimitBytesPerSecondValue);
+    int64_t oneSecondValue = 1;
+    CFNumberRef oneSecond =
+        CFNumberCreate(kCFAllocatorDefault, kCFNumberSInt64Type, &oneSecondValue);
+    const void *nums[2] = {bytesPerSecond, oneSecond};
+    CFArrayRef dataRateLimits = CFArrayCreate(nullptr, nums, 2, &kCFTypeArrayCallBacks);
+    OSStatus status = VTSessionSetProperty(
+        _compressionSession, kVTCompressionPropertyKey_DataRateLimits, dataRateLimits);
+    if (bytesPerSecond) {
+      CFRelease(bytesPerSecond);
+    }
+    if (oneSecond) {
+      CFRelease(oneSecond);
+    }
+    if (dataRateLimits) {
+      CFRelease(dataRateLimits);
+    }
+    if (status != noErr) {
+      RTC_LOG(LS_ERROR) << "Failed to set data rate limit with code: " << status;
+    }
+
+    _encoderBitrateBps = bitrateBps;
+    _encoderFrameRate = frameRate;
+  }
+}
+
+- (void)frameWasEncoded:(OSStatus)status
+                  flags:(VTEncodeInfoFlags)infoFlags
+           sampleBuffer:(CMSampleBufferRef)sampleBuffer
+      codecSpecificInfo:(id<RTC_OBJC_TYPE(RTCCodecSpecificInfo)>)codecSpecificInfo
+                  width:(int32_t)width
+                 height:(int32_t)height
+           renderTimeMs:(int64_t)renderTimeMs
+              timestamp:(uint32_t)timestamp
+               rotation:(RTCVideoRotation)rotation {
+  RTCVideoEncoderCallback callback = _callback;
+  if (!callback) {
+    return;
+  }
+  if (status != noErr) {
+    RTC_LOG(LS_ERROR) << "H264 encode failed with code: " << status;
+    return;
+  }
+  if (infoFlags & kVTEncodeInfo_FrameDropped) {
+    RTC_LOG(LS_INFO) << "H264 encode dropped frame.";
+    return;
+  }
+
+  BOOL isKeyframe = NO;
+  CFArrayRef attachments = CMSampleBufferGetSampleAttachmentsArray(sampleBuffer, 0);
+  if (attachments != nullptr && CFArrayGetCount(attachments)) {
+    CFDictionaryRef attachment =
+        static_cast<CFDictionaryRef>(CFArrayGetValueAtIndex(attachments, 0));
+    isKeyframe = !CFDictionaryContainsKey(attachment, kCMSampleAttachmentKey_NotSync);
+  }
+
+  if (isKeyframe) {
+    RTC_LOG(LS_INFO) << "Generated keyframe";
+  }
+
+  __block std::unique_ptr<rtc::Buffer> buffer = std::make_unique<rtc::Buffer>();
+  if (!webrtc::H264CMSampleBufferToAnnexBBuffer(sampleBuffer, isKeyframe, buffer.get())) {
+    return;
+  }
+
+  RTC_OBJC_TYPE(RTCEncodedImage) *frame = [[RTC_OBJC_TYPE(RTCEncodedImage) alloc] init];
+  // This assumes ownership of `buffer` and is responsible for freeing it when done.
+  frame.buffer = [[NSData alloc] initWithBytesNoCopy:buffer->data()
+                                              length:buffer->size()
+                                         deallocator:^(void *bytes, NSUInteger size) {
+                                           buffer.reset();
+                                         }];
+  frame.encodedWidth = width;
+  frame.encodedHeight = height;
+  frame.frameType = isKeyframe ? RTCFrameTypeVideoFrameKey : RTCFrameTypeVideoFrameDelta;
+  frame.captureTimeMs = renderTimeMs;
+  frame.timeStamp = timestamp;
+  frame.rotation = rotation;
+  frame.contentType = (_mode == RTCVideoCodecModeScreensharing) ? RTCVideoContentTypeScreenshare :
+                                                                  RTCVideoContentTypeUnspecified;
+  frame.flags = webrtc::VideoSendTiming::kInvalid;
+
+  _h264BitstreamParser.ParseBitstream(*buffer);
+  frame.qp = @(_h264BitstreamParser.GetLastSliceQp().value_or(-1));
+
+  BOOL res = callback(frame, codecSpecificInfo);
+  if (!res) {
+    RTC_LOG(LS_ERROR) << "Encode callback failed";
+    return;
+  }
+  _bitrateAdjuster->Update(frame.buffer.length);
+}
+
+- (nullable RTC_OBJC_TYPE(RTCVideoEncoderQpThresholds) *)scalingSettings {
+  return [[RTC_OBJC_TYPE(RTCVideoEncoderQpThresholds) alloc]
+      initWithThresholdsLow:kLowH264QpThreshold
+                       high:kHighH264QpThreshold];
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/components/video_codec/UIDevice+H264Profile.h b/third_party/libwebrtc/sdk/objc/components/video_codec/UIDevice+H264Profile.h
new file mode 100644
index 0000000000..a51debb9fa
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/components/video_codec/UIDevice+H264Profile.h
@@ -0,0 +1,19 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <UIKit/UIKit.h>
+
+#include "api/video_codecs/h264_profile_level_id.h"
+
+@interface UIDevice (H264Profile)
+
++ (absl::optional<webrtc::H264ProfileLevelId>)maxSupportedH264Profile;
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/components/video_codec/UIDevice+H264Profile.mm b/third_party/libwebrtc/sdk/objc/components/video_codec/UIDevice+H264Profile.mm
new file mode 100644
index 0000000000..0ef6a8d77c
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/components/video_codec/UIDevice+H264Profile.mm
@@ -0,0 +1,205 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "UIDevice+H264Profile.h"
+#import "helpers/UIDevice+RTCDevice.h"
+
+#include <algorithm>
+
+namespace {
+
+using namespace webrtc;
+
+struct SupportedH264Profile {
+  const RTCDeviceType deviceType;
+  const H264ProfileLevelId profile;
+};
+
+constexpr SupportedH264Profile kH264MaxSupportedProfiles[] = {
+    // iPhones with at least iOS 9
+    {RTCDeviceTypeIPhone13ProMax,
+     {H264Profile::kProfileHigh, H264Level::kLevel5_2}},  // https://support.apple.com/kb/SP848
+    {RTCDeviceTypeIPhone13Pro,
+     {H264Profile::kProfileHigh, H264Level::kLevel5_2}},  // https://support.apple.com/kb/SP852
+    {RTCDeviceTypeIPhone13,
+     {H264Profile::kProfileHigh, H264Level::kLevel5_2}},  // https://support.apple.com/kb/SP851
+    {RTCDeviceTypeIPhone13Mini,
+     {H264Profile::kProfileHigh, H264Level::kLevel5_2}},  // https://support.apple.com/kb/SP847
+    {RTCDeviceTypeIPhoneSE2Gen,
+     {H264Profile::kProfileHigh, H264Level::kLevel5_2}},  // https://support.apple.com/kb/SP820
+    {RTCDeviceTypeIPhone12ProMax,
+     {H264Profile::kProfileHigh, H264Level::kLevel5_2}},  // https://support.apple.com/kb/SP832
+    {RTCDeviceTypeIPhone12Pro,
+     {H264Profile::kProfileHigh, H264Level::kLevel5_2}},  // https://support.apple.com/kb/SP831
+    {RTCDeviceTypeIPhone12,
+     {H264Profile::kProfileHigh, H264Level::kLevel5_2}},  // https://support.apple.com/kb/SP830
+    {RTCDeviceTypeIPhone12Mini,
+     {H264Profile::kProfileHigh, H264Level::kLevel5_2}},  // https://support.apple.com/kb/SP829
+    {RTCDeviceTypeIPhone11ProMax,
+     {H264Profile::kProfileHigh, H264Level::kLevel5_2}},  // https://support.apple.com/kb/SP806
+    {RTCDeviceTypeIPhone11Pro,
+     {H264Profile::kProfileHigh, H264Level::kLevel5_2}},  // https://support.apple.com/kb/SP805
+    {RTCDeviceTypeIPhone11,
+     {H264Profile::kProfileHigh, H264Level::kLevel5_2}},  // https://support.apple.com/kb/SP804
+    {RTCDeviceTypeIPhoneXS,
+     {H264Profile::kProfileHigh, H264Level::kLevel5_2}},  // https://support.apple.com/kb/SP779
+    {RTCDeviceTypeIPhoneXSMax,
+     {H264Profile::kProfileHigh, H264Level::kLevel5_2}},  // https://support.apple.com/kb/SP780
+    {RTCDeviceTypeIPhoneXR,
+     {H264Profile::kProfileHigh, H264Level::kLevel5_2}},  // https://support.apple.com/kb/SP781
+    {RTCDeviceTypeIPhoneX,
+     {H264Profile::kProfileHigh, H264Level::kLevel5_2}},  // https://support.apple.com/kb/SP770
+    {RTCDeviceTypeIPhone8,
+     {H264Profile::kProfileHigh, H264Level::kLevel5_2}},  // https://support.apple.com/kb/SP767
+    {RTCDeviceTypeIPhone8Plus,
+     {H264Profile::kProfileHigh, H264Level::kLevel5_2}},  // https://support.apple.com/kb/SP768
+    {RTCDeviceTypeIPhone7,
+     {H264Profile::kProfileHigh, H264Level::kLevel5_1}},  // https://support.apple.com/kb/SP743
+    {RTCDeviceTypeIPhone7Plus,
+     {H264Profile::kProfileHigh, H264Level::kLevel5_1}},  // https://support.apple.com/kb/SP744
+    {RTCDeviceTypeIPhoneSE,
+     {H264Profile::kProfileHigh, H264Level::kLevel4_2}},  // https://support.apple.com/kb/SP738
+    {RTCDeviceTypeIPhone6S,
+     {H264Profile::kProfileHigh, H264Level::kLevel4_2}},  // https://support.apple.com/kb/SP726
+    {RTCDeviceTypeIPhone6SPlus,
+     {H264Profile::kProfileHigh, H264Level::kLevel4_2}},  // https://support.apple.com/kb/SP727
+    {RTCDeviceTypeIPhone6,
+     {H264Profile::kProfileHigh, H264Level::kLevel4_2}},  // https://support.apple.com/kb/SP705
+    {RTCDeviceTypeIPhone6Plus,
+     {H264Profile::kProfileHigh, H264Level::kLevel4_2}},  // https://support.apple.com/kb/SP706
+    {RTCDeviceTypeIPhone5SGSM,
+     {H264Profile::kProfileHigh, H264Level::kLevel4_2}},  // https://support.apple.com/kb/SP685
+    {RTCDeviceTypeIPhone5SGSM_CDMA,
+     {H264Profile::kProfileHigh, H264Level::kLevel4_2}},  // https://support.apple.com/kb/SP685
+    {RTCDeviceTypeIPhone5GSM,
+     {H264Profile::kProfileHigh, H264Level::kLevel4_1}},  // https://support.apple.com/kb/SP655
+    {RTCDeviceTypeIPhone5GSM_CDMA,
+     {H264Profile::kProfileHigh, H264Level::kLevel4_1}},  // https://support.apple.com/kb/SP655
+    {RTCDeviceTypeIPhone5CGSM,
+     {H264Profile::kProfileHigh, H264Level::kLevel4_1}},  // https://support.apple.com/kb/SP684
+    {RTCDeviceTypeIPhone5CGSM_CDMA,
+     {H264Profile::kProfileHigh, H264Level::kLevel4_1}},  // https://support.apple.com/kb/SP684
+    {RTCDeviceTypeIPhone4S,
+     {H264Profile::kProfileHigh, H264Level::kLevel4_1}},  // https://support.apple.com/kb/SP643
+
+    // iPods with at least iOS 9
+    {RTCDeviceTypeIPodTouch7G,
+     {H264Profile::kProfileMain, H264Level::kLevel4_1}},  // https://support.apple.com/kb/SP796
+    {RTCDeviceTypeIPodTouch6G,
+     {H264Profile::kProfileMain, H264Level::kLevel4_1}},  // https://support.apple.com/kb/SP720
+    {RTCDeviceTypeIPodTouch5G,
+     {H264Profile::kProfileMain, H264Level::kLevel3_1}},  // https://support.apple.com/kb/SP657
+
+    // iPads with at least iOS 9
+    {RTCDeviceTypeIPadAir4Gen,
+     {H264Profile::kProfileHigh, H264Level::kLevel4_1}},  // https://support.apple.com/kb/SP828
+    {RTCDeviceTypeIPad8,
+     {H264Profile::kProfileHigh, H264Level::kLevel4_1}},  // https://support.apple.com/kb/SP822
+    {RTCDeviceTypeIPadPro4Gen12Inch,
+     {H264Profile::kProfileHigh, H264Level::kLevel4_1}},  // https://support.apple.com/kb/SP815
+    {RTCDeviceTypeIPadPro4Gen11Inch,
+     {H264Profile::kProfileHigh, H264Level::kLevel4_1}},  // https://support.apple.com/kb/SP814
+    {RTCDeviceTypeIPadAir3Gen,
+     {H264Profile::kProfileHigh, H264Level::kLevel4_1}},  // https://support.apple.com/kb/SP787
+    {RTCDeviceTypeIPadMini5Gen,
+     {H264Profile::kProfileHigh, H264Level::kLevel4_1}},  // https://support.apple.com/kb/SP788
+    {RTCDeviceTypeIPadPro3Gen12Inch,
+     {H264Profile::kProfileHigh, H264Level::kLevel5_2}},  // https://support.apple.com/kb/SP785
+    {RTCDeviceTypeIPadPro3Gen11Inch,
+     {H264Profile::kProfileHigh, H264Level::kLevel5_2}},  // https://support.apple.com/kb/SP784
+    {RTCDeviceTypeIPad7Gen10Inch,
+     {H264Profile::kProfileHigh, H264Level::kLevel4_1}},  // https://support.apple.com/kb/SP807
+    {RTCDeviceTypeIPad2Wifi,
+     {H264Profile::kProfileHigh, H264Level::kLevel4_1}},  // https://support.apple.com/kb/SP622
+    {RTCDeviceTypeIPad2GSM,
+     {H264Profile::kProfileHigh, H264Level::kLevel4_1}},  // https://support.apple.com/kb/SP622
+    {RTCDeviceTypeIPad2CDMA,
+     {H264Profile::kProfileHigh, H264Level::kLevel4_1}},  // https://support.apple.com/kb/SP622
+    {RTCDeviceTypeIPad2Wifi2,
+     {H264Profile::kProfileHigh, H264Level::kLevel4_1}},  // https://support.apple.com/kb/SP622
+    {RTCDeviceTypeIPadMiniWifi,
+     {H264Profile::kProfileHigh, H264Level::kLevel4_1}},  // https://support.apple.com/kb/SP661
+    {RTCDeviceTypeIPadMiniGSM,
+     {H264Profile::kProfileHigh, H264Level::kLevel4_1}},  // https://support.apple.com/kb/SP661
+    {RTCDeviceTypeIPadMiniGSM_CDMA,
+     {H264Profile::kProfileHigh, H264Level::kLevel4_1}},  // https://support.apple.com/kb/SP661
+    {RTCDeviceTypeIPad3Wifi,
+     {H264Profile::kProfileHigh, H264Level::kLevel4_1}},  // https://support.apple.com/kb/SP647
+    {RTCDeviceTypeIPad3GSM_CDMA,
+     {H264Profile::kProfileHigh, H264Level::kLevel4_1}},  // https://support.apple.com/kb/SP647
+    {RTCDeviceTypeIPad3GSM,
+     {H264Profile::kProfileHigh, H264Level::kLevel4_1}},  // https://support.apple.com/kb/SP647
+    {RTCDeviceTypeIPad4Wifi,
+     {H264Profile::kProfileHigh, H264Level::kLevel4_1}},  // https://support.apple.com/kb/SP662
+    {RTCDeviceTypeIPad4GSM,
+     {H264Profile::kProfileHigh, H264Level::kLevel4_1}},  // https://support.apple.com/kb/SP662
+    {RTCDeviceTypeIPad4GSM_CDMA,
+     {H264Profile::kProfileHigh, H264Level::kLevel4_1}},  // https://support.apple.com/kb/SP662
+    {RTCDeviceTypeIPad5,
+     {H264Profile::kProfileHigh, H264Level::kLevel4_2}},  // https://support.apple.com/kb/SP751
+    {RTCDeviceTypeIPad6,
+     {H264Profile::kProfileHigh, H264Level::kLevel4_2}},  // https://support.apple.com/kb/SP774
+    {RTCDeviceTypeIPadAirWifi,
+     {H264Profile::kProfileHigh, H264Level::kLevel4_2}},  // https://support.apple.com/kb/SP692
+    {RTCDeviceTypeIPadAirCellular,
+     {H264Profile::kProfileHigh, H264Level::kLevel4_2}},  // https://support.apple.com/kb/SP692
+    {RTCDeviceTypeIPadAirWifiCellular,
+     {H264Profile::kProfileHigh, H264Level::kLevel4_2}},  // https://support.apple.com/kb/SP692
+    {RTCDeviceTypeIPadAir2,
+     {H264Profile::kProfileHigh, H264Level::kLevel4_2}},  // https://support.apple.com/kb/SP708
+    {RTCDeviceTypeIPadMini2GWifi,
+     {H264Profile::kProfileHigh, H264Level::kLevel4_2}},  // https://support.apple.com/kb/SP693
+    {RTCDeviceTypeIPadMini2GCellular,
+     {H264Profile::kProfileHigh, H264Level::kLevel4_2}},  // https://support.apple.com/kb/SP693
+    {RTCDeviceTypeIPadMini2GWifiCellular,
+     {H264Profile::kProfileHigh, H264Level::kLevel4_2}},  // https://support.apple.com/kb/SP693
+    {RTCDeviceTypeIPadMini3,
+     {H264Profile::kProfileHigh, H264Level::kLevel4_2}},  // https://support.apple.com/kb/SP709
+    {RTCDeviceTypeIPadMini4,
+     {H264Profile::kProfileHigh, H264Level::kLevel4_2}},  // https://support.apple.com/kb/SP725
+    {RTCDeviceTypeIPadPro9Inch,
+     {H264Profile::kProfileHigh, H264Level::kLevel4_2}},  // https://support.apple.com/kb/SP739
+    {RTCDeviceTypeIPadPro12Inch,
+     {H264Profile::kProfileHigh, H264Level::kLevel4_2}},  // https://support.apple.com/kb/sp723
+    {RTCDeviceTypeIPadPro12Inch2,
+     {H264Profile::kProfileHigh, H264Level::kLevel4_2}},  // https://support.apple.com/kb/SP761
+    {RTCDeviceTypeIPadPro10Inch,
+     {H264Profile::kProfileHigh, H264Level::kLevel4_2}},  // https://support.apple.com/kb/SP762
+    {RTCDeviceTypeIPadMini6,
+     {H264Profile::kProfileHigh, H264Level::kLevel5_2}},  // https://support.apple.com/kb/SP850
+    {RTCDeviceTypeIPad9,
+     {H264Profile::kProfileHigh, H264Level::kLevel4_2}},  // https://support.apple.com/kb/SP849
+    {RTCDeviceTypeIPadPro5Gen12Inch,
+     {H264Profile::kProfileHigh, H264Level::kLevel5_2}},  // https://support.apple.com/kb/SP844
+    {RTCDeviceTypeIPadPro5Gen11Inch,
+     {H264Profile::kProfileHigh, H264Level::kLevel5_2}},  // https://support.apple.com/kb/SP843
+};
+
+absl::optional<H264ProfileLevelId> FindMaxSupportedProfileForDevice(RTCDeviceType deviceType) {
+  const auto* result = std::find_if(std::begin(kH264MaxSupportedProfiles),
+                                    std::end(kH264MaxSupportedProfiles),
+                                    [deviceType](const SupportedH264Profile& supportedProfile) {
+                                      return supportedProfile.deviceType == deviceType;
+                                    });
+  if (result != std::end(kH264MaxSupportedProfiles)) {
+    return result->profile;
+  }
+  return absl::nullopt;
+}
+
+}  // namespace
+
+@implementation UIDevice (H264Profile)
+
++ (absl::optional<webrtc::H264ProfileLevelId>)maxSupportedH264Profile {
+  return FindMaxSupportedProfileForDevice([self deviceType]);
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/components/video_codec/helpers.cc b/third_party/libwebrtc/sdk/objc/components/video_codec/helpers.cc
new file mode 100644
index 0000000000..ac957f1b49
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/components/video_codec/helpers.cc
@@ -0,0 +1,90 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ *
+ */
+
+#include "helpers.h"
+
+#include <string>
+
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+// Copies characters from a CFStringRef into a std::string.
+std::string CFStringToString(const CFStringRef cf_string) {
+  RTC_DCHECK(cf_string);
+  std::string std_string;
+  // Get the size needed for UTF8 plus terminating character.
+  size_t buffer_size =
+      CFStringGetMaximumSizeForEncoding(CFStringGetLength(cf_string),
+                                        kCFStringEncodingUTF8) +
+      1;
+  std::unique_ptr<char[]> buffer(new char[buffer_size]);
+  if (CFStringGetCString(cf_string, buffer.get(), buffer_size,
+                         kCFStringEncodingUTF8)) {
+    // Copy over the characters.
+    std_string.assign(buffer.get());
+  }
+  return std_string;
+}
+
+// Convenience function for setting a VT property.
+void SetVTSessionProperty(VTSessionRef session,
+                          CFStringRef key,
+                          int32_t value) {
+  CFNumberRef cfNum =
+      CFNumberCreate(kCFAllocatorDefault, kCFNumberSInt32Type, &value);
+  OSStatus status = VTSessionSetProperty(session, key, cfNum);
+  CFRelease(cfNum);
+  if (status != noErr) {
+    std::string key_string = CFStringToString(key);
+    RTC_LOG(LS_ERROR) << "VTSessionSetProperty failed to set: " << key_string
+                      << " to " << value << ": " << status;
+  }
+}
+
+// Convenience function for setting a VT property.
+void SetVTSessionProperty(VTSessionRef session,
+                          CFStringRef key,
+                          uint32_t value) {
+  int64_t value_64 = value;
+  CFNumberRef cfNum =
+      CFNumberCreate(kCFAllocatorDefault, kCFNumberSInt64Type, &value_64);
+  OSStatus status = VTSessionSetProperty(session, key, cfNum);
+  CFRelease(cfNum);
+  if (status != noErr) {
+    std::string key_string = CFStringToString(key);
+    RTC_LOG(LS_ERROR) << "VTSessionSetProperty failed to set: " << key_string
+                      << " to " << value << ": " << status;
+  }
+}
+
+// Convenience function for setting a VT property.
+void SetVTSessionProperty(VTSessionRef session, CFStringRef key, bool value) {
+  CFBooleanRef cf_bool = (value) ? kCFBooleanTrue : kCFBooleanFalse;
+  OSStatus status = VTSessionSetProperty(session, key, cf_bool);
+  if (status != noErr) {
+    std::string key_string = CFStringToString(key);
+    RTC_LOG(LS_ERROR) << "VTSessionSetProperty failed to set: " << key_string
+                      << " to " << value << ": " << status;
+  }
+}
+
+// Convenience function for setting a VT property.
+void SetVTSessionProperty(VTSessionRef session,
+                          CFStringRef key,
+                          CFStringRef value) {
+  OSStatus status = VTSessionSetProperty(session, key, value);
+  if (status != noErr) {
+    std::string key_string = CFStringToString(key);
+    std::string val_string = CFStringToString(value);
+    RTC_LOG(LS_ERROR) << "VTSessionSetProperty failed to set: " << key_string
+                      << " to " << val_string << ": " << status;
+  }
+}
diff --git a/third_party/libwebrtc/sdk/objc/components/video_codec/helpers.h b/third_party/libwebrtc/sdk/objc/components/video_codec/helpers.h
new file mode 100644
index 0000000000..7c9ef1cd87
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/components/video_codec/helpers.h
@@ -0,0 +1,47 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ *
+ */
+
+#ifndef SDK_OBJC_FRAMEWORK_CLASSES_VIDEOTOOLBOX_HELPERS_H_
+#define SDK_OBJC_FRAMEWORK_CLASSES_VIDEOTOOLBOX_HELPERS_H_
+
+#include <CoreFoundation/CoreFoundation.h>
+#include <VideoToolbox/VideoToolbox.h>
+#include <string>
+
+// Convenience function for creating a dictionary.
+inline CFDictionaryRef CreateCFTypeDictionary(CFTypeRef* keys,
+                                              CFTypeRef* values,
+                                              size_t size) {
+  return CFDictionaryCreate(kCFAllocatorDefault, keys, values, size,
+                            &kCFTypeDictionaryKeyCallBacks,
+                            &kCFTypeDictionaryValueCallBacks);
+}
+
+// Copies characters from a CFStringRef into a std::string.
+std::string CFStringToString(CFStringRef cf_string);
+
+// Convenience function for setting a VT property.
+void SetVTSessionProperty(VTSessionRef session, CFStringRef key, int32_t value);
+
+// Convenience function for setting a VT property.
+void SetVTSessionProperty(VTSessionRef session,
+                          CFStringRef key,
+                          uint32_t value);
+
+// Convenience function for setting a VT property.
+void SetVTSessionProperty(VTSessionRef session, CFStringRef key, bool value);
+
+// Convenience function for setting a VT property.
+void SetVTSessionProperty(VTSessionRef session,
+                          CFStringRef key,
+                          CFStringRef value);
+
+#endif  // SDK_OBJC_FRAMEWORK_CLASSES_VIDEOTOOLBOX_HELPERS_H_
diff --git a/third_party/libwebrtc/sdk/objc/components/video_codec/nalu_rewriter.cc b/third_party/libwebrtc/sdk/objc/components/video_codec/nalu_rewriter.cc
new file mode 100644
index 0000000000..b7330e1f9c
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/components/video_codec/nalu_rewriter.cc
@@ -0,0 +1,327 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ *
+ */
+
+#include "sdk/objc/components/video_codec/nalu_rewriter.h"
+
+#include <CoreFoundation/CoreFoundation.h>
+#include <memory>
+#include <vector>
+
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+
+using H264::kAud;
+using H264::kSps;
+using H264::NaluIndex;
+using H264::NaluType;
+using H264::ParseNaluType;
+
+const char kAnnexBHeaderBytes[4] = {0, 0, 0, 1};
+const size_t kAvccHeaderByteSize = sizeof(uint32_t);
+
+bool H264CMSampleBufferToAnnexBBuffer(CMSampleBufferRef avcc_sample_buffer,
+                                      bool is_keyframe,
+                                      rtc::Buffer* annexb_buffer) {
+  RTC_DCHECK(avcc_sample_buffer);
+
+  // Get format description from the sample buffer.
+  CMVideoFormatDescriptionRef description =
+      CMSampleBufferGetFormatDescription(avcc_sample_buffer);
+  if (description == nullptr) {
+    RTC_LOG(LS_ERROR) << "Failed to get sample buffer's description.";
+    return false;
+  }
+
+  // Get parameter set information.
+  int nalu_header_size = 0;
+  size_t param_set_count = 0;
+  OSStatus status = CMVideoFormatDescriptionGetH264ParameterSetAtIndex(
+      description, 0, nullptr, nullptr, &param_set_count, &nalu_header_size);
+  if (status != noErr) {
+    RTC_LOG(LS_ERROR) << "Failed to get parameter set.";
+    return false;
+  }
+  RTC_CHECK_EQ(nalu_header_size, kAvccHeaderByteSize);
+  RTC_DCHECK_EQ(param_set_count, 2);
+
+  // Truncate any previous data in the buffer without changing its capacity.
+  annexb_buffer->SetSize(0);
+
+  // Place all parameter sets at the front of buffer.
+  if (is_keyframe) {
+    size_t param_set_size = 0;
+    const uint8_t* param_set = nullptr;
+    for (size_t i = 0; i < param_set_count; ++i) {
+      status = CMVideoFormatDescriptionGetH264ParameterSetAtIndex(
+          description, i, &param_set, &param_set_size, nullptr, nullptr);
+      if (status != noErr) {
+        RTC_LOG(LS_ERROR) << "Failed to get parameter set.";
+        return false;
+      }
+      // Update buffer.
+      annexb_buffer->AppendData(kAnnexBHeaderBytes, sizeof(kAnnexBHeaderBytes));
+      annexb_buffer->AppendData(reinterpret_cast<const char*>(param_set),
+                                param_set_size);
+    }
+  }
+
+  // Get block buffer from the sample buffer.
+  CMBlockBufferRef block_buffer =
+      CMSampleBufferGetDataBuffer(avcc_sample_buffer);
+  if (block_buffer == nullptr) {
+    RTC_LOG(LS_ERROR) << "Failed to get sample buffer's block buffer.";
+    return false;
+  }
+  CMBlockBufferRef contiguous_buffer = nullptr;
+  // Make sure block buffer is contiguous.
+  if (!CMBlockBufferIsRangeContiguous(block_buffer, 0, 0)) {
+    status = CMBlockBufferCreateContiguous(
+        nullptr, block_buffer, nullptr, nullptr, 0, 0, 0, &contiguous_buffer);
+    if (status != noErr) {
+      RTC_LOG(LS_ERROR) << "Failed to flatten non-contiguous block buffer: "
+                        << status;
+      return false;
+    }
+  } else {
+    contiguous_buffer = block_buffer;
+    // Retain to make cleanup easier.
+    CFRetain(contiguous_buffer);
+    block_buffer = nullptr;
+  }
+
+  // Now copy the actual data.
+  char* data_ptr = nullptr;
+  size_t block_buffer_size = CMBlockBufferGetDataLength(contiguous_buffer);
+  status = CMBlockBufferGetDataPointer(contiguous_buffer, 0, nullptr, nullptr,
+                                       &data_ptr);
+  if (status != noErr) {
+    RTC_LOG(LS_ERROR) << "Failed to get block buffer data.";
+    CFRelease(contiguous_buffer);
+    return false;
+  }
+  size_t bytes_remaining = block_buffer_size;
+  while (bytes_remaining > 0) {
+    // The size type here must match `nalu_header_size`, we expect 4 bytes.
+    // Read the length of the next packet of data. Must convert from big endian
+    // to host endian.
+    RTC_DCHECK_GE(bytes_remaining, (size_t)nalu_header_size);
+    uint32_t* uint32_data_ptr = reinterpret_cast<uint32_t*>(data_ptr);
+    uint32_t packet_size = CFSwapInt32BigToHost(*uint32_data_ptr);
+    // Update buffer.
+    annexb_buffer->AppendData(kAnnexBHeaderBytes, sizeof(kAnnexBHeaderBytes));
+    annexb_buffer->AppendData(data_ptr + nalu_header_size, packet_size);
+
+    size_t bytes_written = packet_size + sizeof(kAnnexBHeaderBytes);
+    bytes_remaining -= bytes_written;
+    data_ptr += bytes_written;
+  }
+  RTC_DCHECK_EQ(bytes_remaining, (size_t)0);
+
+  CFRelease(contiguous_buffer);
+  return true;
+}
+
+bool H264AnnexBBufferToCMSampleBuffer(const uint8_t* annexb_buffer,
+                                      size_t annexb_buffer_size,
+                                      CMVideoFormatDescriptionRef video_format,
+                                      CMSampleBufferRef* out_sample_buffer,
+                                      CMMemoryPoolRef memory_pool) {
+  RTC_DCHECK(annexb_buffer);
+  RTC_DCHECK(out_sample_buffer);
+  RTC_DCHECK(video_format);
+  *out_sample_buffer = nullptr;
+
+  AnnexBBufferReader reader(annexb_buffer, annexb_buffer_size);
+  if (reader.SeekToNextNaluOfType(kSps)) {
+    // Buffer contains an SPS NALU - skip it and the following PPS
+    const uint8_t* data;
+    size_t data_len;
+    if (!reader.ReadNalu(&data, &data_len)) {
+      RTC_LOG(LS_ERROR) << "Failed to read SPS";
+      return false;
+    }
+    if (!reader.ReadNalu(&data, &data_len)) {
+      RTC_LOG(LS_ERROR) << "Failed to read PPS";
+      return false;
+    }
+  } else {
+    // No SPS NALU - start reading from the first NALU in the buffer
+    reader.SeekToStart();
+  }
+
+  // Allocate memory as a block buffer.
+  CMBlockBufferRef block_buffer = nullptr;
+  CFAllocatorRef block_allocator = CMMemoryPoolGetAllocator(memory_pool);
+  OSStatus status = CMBlockBufferCreateWithMemoryBlock(
+      kCFAllocatorDefault, nullptr, reader.BytesRemaining(), block_allocator,
+      nullptr, 0, reader.BytesRemaining(), kCMBlockBufferAssureMemoryNowFlag,
+      &block_buffer);
+  if (status != kCMBlockBufferNoErr) {
+    RTC_LOG(LS_ERROR) << "Failed to create block buffer.";
+    return false;
+  }
+
+  // Make sure block buffer is contiguous.
+  CMBlockBufferRef contiguous_buffer = nullptr;
+  if (!CMBlockBufferIsRangeContiguous(block_buffer, 0, 0)) {
+    status = CMBlockBufferCreateContiguous(kCFAllocatorDefault, block_buffer,
+                                           block_allocator, nullptr, 0, 0, 0,
+                                           &contiguous_buffer);
+    if (status != noErr) {
+      RTC_LOG(LS_ERROR) << "Failed to flatten non-contiguous block buffer: "
+                        << status;
+      CFRelease(block_buffer);
+      return false;
+    }
+  } else {
+    contiguous_buffer = block_buffer;
+    block_buffer = nullptr;
+  }
+
+  // Get a raw pointer into allocated memory.
+  size_t block_buffer_size = 0;
+  char* data_ptr = nullptr;
+  status = CMBlockBufferGetDataPointer(contiguous_buffer, 0, nullptr,
+                                       &block_buffer_size, &data_ptr);
+  if (status != kCMBlockBufferNoErr) {
+    RTC_LOG(LS_ERROR) << "Failed to get block buffer data pointer.";
+    CFRelease(contiguous_buffer);
+    return false;
+  }
+  RTC_DCHECK(block_buffer_size == reader.BytesRemaining());
+
+  // Write Avcc NALUs into block buffer memory.
+  AvccBufferWriter writer(reinterpret_cast<uint8_t*>(data_ptr),
+                          block_buffer_size);
+  while (reader.BytesRemaining() > 0) {
+    const uint8_t* nalu_data_ptr = nullptr;
+    size_t nalu_data_size = 0;
+    if (reader.ReadNalu(&nalu_data_ptr, &nalu_data_size)) {
+      writer.WriteNalu(nalu_data_ptr, nalu_data_size);
+    }
+  }
+
+  // Create sample buffer.
+  status = CMSampleBufferCreate(kCFAllocatorDefault, contiguous_buffer, true,
+                                nullptr, nullptr, video_format, 1, 0, nullptr,
+                                0, nullptr, out_sample_buffer);
+  if (status != noErr) {
+    RTC_LOG(LS_ERROR) << "Failed to create sample buffer.";
+    CFRelease(contiguous_buffer);
+    return false;
+  }
+  CFRelease(contiguous_buffer);
+  return true;
+}
+
+CMVideoFormatDescriptionRef CreateVideoFormatDescription(
+    const uint8_t* annexb_buffer,
+    size_t annexb_buffer_size) {
+  const uint8_t* param_set_ptrs[2] = {};
+  size_t param_set_sizes[2] = {};
+  AnnexBBufferReader reader(annexb_buffer, annexb_buffer_size);
+  // Skip everyting before the SPS, then read the SPS and PPS
+  if (!reader.SeekToNextNaluOfType(kSps)) {
+    return nullptr;
+  }
+  if (!reader.ReadNalu(&param_set_ptrs[0], &param_set_sizes[0])) {
+    RTC_LOG(LS_ERROR) << "Failed to read SPS";
+    return nullptr;
+  }
+  if (!reader.ReadNalu(&param_set_ptrs[1], &param_set_sizes[1])) {
+    RTC_LOG(LS_ERROR) << "Failed to read PPS";
+    return nullptr;
+  }
+
+  // Parse the SPS and PPS into a CMVideoFormatDescription.
+  CMVideoFormatDescriptionRef description = nullptr;
+  OSStatus status = CMVideoFormatDescriptionCreateFromH264ParameterSets(
+      kCFAllocatorDefault, 2, param_set_ptrs, param_set_sizes, 4, &description);
+  if (status != noErr) {
+    RTC_LOG(LS_ERROR) << "Failed to create video format description.";
+    return nullptr;
+  }
+  return description;
+}
+
+AnnexBBufferReader::AnnexBBufferReader(const uint8_t* annexb_buffer,
+                                       size_t length)
+    : start_(annexb_buffer), length_(length) {
+  RTC_DCHECK(annexb_buffer);
+  offsets_ = H264::FindNaluIndices(annexb_buffer, length);
+  offset_ = offsets_.begin();
+}
+
+AnnexBBufferReader::~AnnexBBufferReader() = default;
+
+bool AnnexBBufferReader::ReadNalu(const uint8_t** out_nalu,
+                                  size_t* out_length) {
+  RTC_DCHECK(out_nalu);
+  RTC_DCHECK(out_length);
+  *out_nalu = nullptr;
+  *out_length = 0;
+
+  if (offset_ == offsets_.end()) {
+    return false;
+  }
+  *out_nalu = start_ + offset_->payload_start_offset;
+  *out_length = offset_->payload_size;
+  ++offset_;
+  return true;
+}
+
+size_t AnnexBBufferReader::BytesRemaining() const {
+  if (offset_ == offsets_.end()) {
+    return 0;
+  }
+  return length_ - offset_->start_offset;
+}
+
+void AnnexBBufferReader::SeekToStart() {
+  offset_ = offsets_.begin();
+}
+
+bool AnnexBBufferReader::SeekToNextNaluOfType(NaluType type) {
+  for (; offset_ != offsets_.end(); ++offset_) {
+    if (offset_->payload_size < 1)
+      continue;
+    if (ParseNaluType(*(start_ + offset_->payload_start_offset)) == type)
+      return true;
+  }
+  return false;
+}
+AvccBufferWriter::AvccBufferWriter(uint8_t* const avcc_buffer, size_t length)
+    : start_(avcc_buffer), offset_(0), length_(length) {
+  RTC_DCHECK(avcc_buffer);
+}
+
+bool AvccBufferWriter::WriteNalu(const uint8_t* data, size_t data_size) {
+  // Check if we can write this length of data.
+  if (data_size + kAvccHeaderByteSize > BytesRemaining()) {
+    return false;
+  }
+  // Write length header, which needs to be big endian.
+  uint32_t big_endian_length = CFSwapInt32HostToBig(data_size);
+  memcpy(start_ + offset_, &big_endian_length, sizeof(big_endian_length));
+  offset_ += sizeof(big_endian_length);
+  // Write data.
+  memcpy(start_ + offset_, data, data_size);
+  offset_ += data_size;
+  return true;
+}
+
+size_t AvccBufferWriter::BytesRemaining() const {
+  return length_ - offset_;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/sdk/objc/components/video_codec/nalu_rewriter.h b/third_party/libwebrtc/sdk/objc/components/video_codec/nalu_rewriter.h
new file mode 100644
index 0000000000..c6474971e2
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/components/video_codec/nalu_rewriter.h
@@ -0,0 +1,113 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ *
+ */
+
+#ifndef SDK_OBJC_FRAMEWORK_CLASSES_VIDEOTOOLBOX_NALU_REWRITER_H_
+#define SDK_OBJC_FRAMEWORK_CLASSES_VIDEOTOOLBOX_NALU_REWRITER_H_
+
+#include "modules/video_coding/codecs/h264/include/h264.h"
+
+#include <CoreMedia/CoreMedia.h>
+#include <vector>
+
+#include "common_video/h264/h264_common.h"
+#include "rtc_base/buffer.h"
+
+using webrtc::H264::NaluIndex;
+
+namespace webrtc {
+
+// Converts a sample buffer emitted from the VideoToolbox encoder into a buffer
+// suitable for RTP. The sample buffer is in avcc format whereas the rtp buffer
+// needs to be in Annex B format. Data is written directly to `annexb_buffer`.
+bool H264CMSampleBufferToAnnexBBuffer(CMSampleBufferRef avcc_sample_buffer,
+                                      bool is_keyframe,
+                                      rtc::Buffer* annexb_buffer);
+
+// Converts a buffer received from RTP into a sample buffer suitable for the
+// VideoToolbox decoder. The RTP buffer is in annex b format whereas the sample
+// buffer is in avcc format.
+// If `is_keyframe` is true then `video_format` is ignored since the format will
+// be read from the buffer. Otherwise `video_format` must be provided.
+// Caller is responsible for releasing the created sample buffer.
+bool H264AnnexBBufferToCMSampleBuffer(const uint8_t* annexb_buffer,
+                                      size_t annexb_buffer_size,
+                                      CMVideoFormatDescriptionRef video_format,
+                                      CMSampleBufferRef* out_sample_buffer,
+                                      CMMemoryPoolRef memory_pool);
+
+// Returns a video format description created from the sps/pps information in
+// the Annex B buffer. If there is no such information, nullptr is returned.
+// The caller is responsible for releasing the description.
+CMVideoFormatDescriptionRef CreateVideoFormatDescription(
+    const uint8_t* annexb_buffer,
+    size_t annexb_buffer_size);
+
+// Helper class for reading NALUs from an RTP Annex B buffer.
+class AnnexBBufferReader final {
+ public:
+  AnnexBBufferReader(const uint8_t* annexb_buffer, size_t length);
+  ~AnnexBBufferReader();
+  AnnexBBufferReader(const AnnexBBufferReader& other) = delete;
+  void operator=(const AnnexBBufferReader& other) = delete;
+
+  // Returns a pointer to the beginning of the next NALU slice without the
+  // header bytes and its length. Returns false if no more slices remain.
+  bool ReadNalu(const uint8_t** out_nalu, size_t* out_length);
+
+  // Returns the number of unread NALU bytes, including the size of the header.
+  // If the buffer has no remaining NALUs this will return zero.
+  size_t BytesRemaining() const;
+
+  // Reset the reader to start reading from the first NALU
+  void SeekToStart();
+
+  // Seek to the next position that holds a NALU of the desired type,
+  // or the end if no such NALU is found.
+  // Return true if a NALU of the desired type is found, false if we
+  // reached the end instead
+  bool SeekToNextNaluOfType(H264::NaluType type);
+
+ private:
+  // Returns the the next offset that contains NALU data.
+  size_t FindNextNaluHeader(const uint8_t* start,
+                            size_t length,
+                            size_t offset) const;
+
+  const uint8_t* const start_;
+  std::vector<NaluIndex> offsets_;
+  std::vector<NaluIndex>::iterator offset_;
+  const size_t length_;
+};
+
+// Helper class for writing NALUs using avcc format into a buffer.
+class AvccBufferWriter final {
+ public:
+  AvccBufferWriter(uint8_t* const avcc_buffer, size_t length);
+  ~AvccBufferWriter() {}
+  AvccBufferWriter(const AvccBufferWriter& other) = delete;
+  void operator=(const AvccBufferWriter& other) = delete;
+
+  // Writes the data slice into the buffer. Returns false if there isn't
+  // enough space left.
+  bool WriteNalu(const uint8_t* data, size_t data_size);
+
+  // Returns the unused bytes in the buffer.
+  size_t BytesRemaining() const;
+
+ private:
+  uint8_t* const start_;
+  size_t offset_;
+  const size_t length_;
+};
+
+}  // namespace webrtc
+
+#endif  // SDK_OBJC_FRAMEWORK_CLASSES_VIDEOTOOLBOX_NALU_REWRITER_H_
diff --git a/third_party/libwebrtc/sdk/objc/components/video_frame_buffer/RTCCVPixelBuffer.h b/third_party/libwebrtc/sdk/objc/components/video_frame_buffer/RTCCVPixelBuffer.h
new file mode 100644
index 0000000000..664d9bb904
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/components/video_frame_buffer/RTCCVPixelBuffer.h
@@ -0,0 +1,52 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <AVFoundation/AVFoundation.h>
+
+#import "RTCMacros.h"
+#import "RTCVideoFrameBuffer.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+/** RTCVideoFrameBuffer containing a CVPixelBufferRef */
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCCVPixelBuffer) : NSObject <RTC_OBJC_TYPE(RTCVideoFrameBuffer)>
+
+@property(nonatomic, readonly) CVPixelBufferRef pixelBuffer;
+@property(nonatomic, readonly) int cropX;
+@property(nonatomic, readonly) int cropY;
+@property(nonatomic, readonly) int cropWidth;
+@property(nonatomic, readonly) int cropHeight;
+
++ (NSSet<NSNumber *> *)supportedPixelFormats;
+
+- (instancetype)initWithPixelBuffer:(CVPixelBufferRef)pixelBuffer;
+- (instancetype)initWithPixelBuffer:(CVPixelBufferRef)pixelBuffer
+                       adaptedWidth:(int)adaptedWidth
+                      adaptedHeight:(int)adaptedHeight
+                          cropWidth:(int)cropWidth
+                         cropHeight:(int)cropHeight
+                              cropX:(int)cropX
+                              cropY:(int)cropY;
+
+- (BOOL)requiresCropping;
+- (BOOL)requiresScalingToWidth:(int)width height:(int)height;
+- (int)bufferSizeForCroppingAndScalingToWidth:(int)width height:(int)height;
+
+/** The minimum size of the `tmpBuffer` must be the number of bytes returned from the
+ * bufferSizeForCroppingAndScalingToWidth:height: method.
+ * If that size is 0, the `tmpBuffer` may be nil.
+ */
+- (BOOL)cropAndScaleTo:(CVPixelBufferRef)outputPixelBuffer
+        withTempBuffer:(nullable uint8_t *)tmpBuffer;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/components/video_frame_buffer/RTCCVPixelBuffer.mm b/third_party/libwebrtc/sdk/objc/components/video_frame_buffer/RTCCVPixelBuffer.mm
new file mode 100644
index 0000000000..1a9b672d1a
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/components/video_frame_buffer/RTCCVPixelBuffer.mm
@@ -0,0 +1,367 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCCVPixelBuffer.h"
+
+#import "api/video_frame_buffer/RTCNativeMutableI420Buffer.h"
+
+#include "common_video/libyuv/include/webrtc_libyuv.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "third_party/libyuv/include/libyuv.h"
+
+#if !defined(NDEBUG) && defined(WEBRTC_IOS)
+#import <UIKit/UIKit.h>
+#import <VideoToolbox/VideoToolbox.h>
+#endif
+
+@implementation RTC_OBJC_TYPE (RTCCVPixelBuffer) {
+  int _width;
+  int _height;
+  int _bufferWidth;
+  int _bufferHeight;
+  int _cropWidth;
+  int _cropHeight;
+}
+
+@synthesize pixelBuffer = _pixelBuffer;
+@synthesize cropX = _cropX;
+@synthesize cropY = _cropY;
+@synthesize cropWidth = _cropWidth;
+@synthesize cropHeight = _cropHeight;
+
++ (NSSet<NSNumber*>*)supportedPixelFormats {
+  return [NSSet setWithObjects:@(kCVPixelFormatType_420YpCbCr8BiPlanarFullRange),
+                               @(kCVPixelFormatType_420YpCbCr8BiPlanarVideoRange),
+                               @(kCVPixelFormatType_32BGRA),
+                               @(kCVPixelFormatType_32ARGB),
+                               nil];
+}
+
+- (instancetype)initWithPixelBuffer:(CVPixelBufferRef)pixelBuffer {
+  return [self initWithPixelBuffer:pixelBuffer
+                      adaptedWidth:CVPixelBufferGetWidth(pixelBuffer)
+                     adaptedHeight:CVPixelBufferGetHeight(pixelBuffer)
+                         cropWidth:CVPixelBufferGetWidth(pixelBuffer)
+                        cropHeight:CVPixelBufferGetHeight(pixelBuffer)
+                             cropX:0
+                             cropY:0];
+}
+
+- (instancetype)initWithPixelBuffer:(CVPixelBufferRef)pixelBuffer
+                       adaptedWidth:(int)adaptedWidth
+                      adaptedHeight:(int)adaptedHeight
+                          cropWidth:(int)cropWidth
+                         cropHeight:(int)cropHeight
+                              cropX:(int)cropX
+                              cropY:(int)cropY {
+  if (self = [super init]) {
+    _width = adaptedWidth;
+    _height = adaptedHeight;
+    _pixelBuffer = pixelBuffer;
+    _bufferWidth = CVPixelBufferGetWidth(_pixelBuffer);
+    _bufferHeight = CVPixelBufferGetHeight(_pixelBuffer);
+    _cropWidth = cropWidth;
+    _cropHeight = cropHeight;
+    // Can only crop at even pixels.
+    _cropX = cropX & ~1;
+    _cropY = cropY & ~1;
+    CVBufferRetain(_pixelBuffer);
+  }
+
+  return self;
+}
+
+- (void)dealloc {
+  CVBufferRelease(_pixelBuffer);
+}
+
+- (int)width {
+  return _width;
+}
+
+- (int)height {
+  return _height;
+}
+
+- (BOOL)requiresCropping {
+  return _cropWidth != _bufferWidth || _cropHeight != _bufferHeight;
+}
+
+- (BOOL)requiresScalingToWidth:(int)width height:(int)height {
+  return _cropWidth != width || _cropHeight != height;
+}
+
+- (int)bufferSizeForCroppingAndScalingToWidth:(int)width height:(int)height {
+  const OSType srcPixelFormat = CVPixelBufferGetPixelFormatType(_pixelBuffer);
+  switch (srcPixelFormat) {
+    case kCVPixelFormatType_420YpCbCr8BiPlanarFullRange:
+    case kCVPixelFormatType_420YpCbCr8BiPlanarVideoRange: {
+      int srcChromaWidth = (_cropWidth + 1) / 2;
+      int srcChromaHeight = (_cropHeight + 1) / 2;
+      int dstChromaWidth = (width + 1) / 2;
+      int dstChromaHeight = (height + 1) / 2;
+
+      return srcChromaWidth * srcChromaHeight * 2 + dstChromaWidth * dstChromaHeight * 2;
+    }
+    case kCVPixelFormatType_32BGRA:
+    case kCVPixelFormatType_32ARGB: {
+      return 0;  // Scaling RGBA frames does not require a temporary buffer.
+    }
+  }
+  RTC_DCHECK_NOTREACHED() << "Unsupported pixel format.";
+  return 0;
+}
+
+- (BOOL)cropAndScaleTo:(CVPixelBufferRef)outputPixelBuffer
+        withTempBuffer:(nullable uint8_t*)tmpBuffer {
+  const OSType srcPixelFormat = CVPixelBufferGetPixelFormatType(_pixelBuffer);
+  const OSType dstPixelFormat = CVPixelBufferGetPixelFormatType(outputPixelBuffer);
+
+  switch (srcPixelFormat) {
+    case kCVPixelFormatType_420YpCbCr8BiPlanarFullRange:
+    case kCVPixelFormatType_420YpCbCr8BiPlanarVideoRange: {
+      size_t dstWidth = CVPixelBufferGetWidth(outputPixelBuffer);
+      size_t dstHeight = CVPixelBufferGetHeight(outputPixelBuffer);
+      if (dstWidth > 0 && dstHeight > 0) {
+        RTC_DCHECK(dstPixelFormat == kCVPixelFormatType_420YpCbCr8BiPlanarFullRange ||
+                   dstPixelFormat == kCVPixelFormatType_420YpCbCr8BiPlanarVideoRange);
+        if ([self requiresScalingToWidth:dstWidth height:dstHeight]) {
+          RTC_DCHECK(tmpBuffer);
+        }
+        [self cropAndScaleNV12To:outputPixelBuffer withTempBuffer:tmpBuffer];
+      }
+      break;
+    }
+    case kCVPixelFormatType_32BGRA:
+    case kCVPixelFormatType_32ARGB: {
+      RTC_DCHECK(srcPixelFormat == dstPixelFormat);
+      [self cropAndScaleARGBTo:outputPixelBuffer];
+      break;
+    }
+    default: {
+      RTC_DCHECK_NOTREACHED() << "Unsupported pixel format.";
+    }
+  }
+
+  return YES;
+}
+- (id<RTC_OBJC_TYPE(RTCVideoFrameBuffer)>)cropAndScaleWith:(int)offsetX
+                                                   offsetY:(int)offsetY
+                                                 cropWidth:(int)cropWidth
+                                                cropHeight:(int)cropHeight
+                                                scaleWidth:(int)scaleWidth
+                                               scaleHeight:(int)scaleHeight {
+  return [[RTC_OBJC_TYPE(RTCCVPixelBuffer) alloc]
+      initWithPixelBuffer:_pixelBuffer
+             adaptedWidth:scaleWidth
+            adaptedHeight:scaleHeight
+                cropWidth:cropWidth * _cropWidth / _width
+               cropHeight:cropHeight * _cropHeight / _height
+                    cropX:_cropX + offsetX * _cropWidth / _width
+                    cropY:_cropY + offsetY * _cropHeight / _height];
+}
+
+- (id<RTC_OBJC_TYPE(RTCI420Buffer)>)toI420 {
+  const OSType pixelFormat = CVPixelBufferGetPixelFormatType(_pixelBuffer);
+
+  CVPixelBufferLockBaseAddress(_pixelBuffer, kCVPixelBufferLock_ReadOnly);
+
+  RTC_OBJC_TYPE(RTCMutableI420Buffer)* i420Buffer =
+      [[RTC_OBJC_TYPE(RTCMutableI420Buffer) alloc] initWithWidth:[self width] height:[self height]];
+
+  switch (pixelFormat) {
+    case kCVPixelFormatType_420YpCbCr8BiPlanarFullRange:
+    case kCVPixelFormatType_420YpCbCr8BiPlanarVideoRange: {
+      const uint8_t* srcY =
+          static_cast<uint8_t*>(CVPixelBufferGetBaseAddressOfPlane(_pixelBuffer, 0));
+      const int srcYStride = CVPixelBufferGetBytesPerRowOfPlane(_pixelBuffer, 0);
+      const uint8_t* srcUV =
+          static_cast<uint8_t*>(CVPixelBufferGetBaseAddressOfPlane(_pixelBuffer, 1));
+      const int srcUVStride = CVPixelBufferGetBytesPerRowOfPlane(_pixelBuffer, 1);
+
+      // Crop just by modifying pointers.
+      srcY += srcYStride * _cropY + _cropX;
+      srcUV += srcUVStride * (_cropY / 2) + _cropX;
+
+      // TODO(magjed): Use a frame buffer pool.
+      webrtc::NV12ToI420Scaler nv12ToI420Scaler;
+      nv12ToI420Scaler.NV12ToI420Scale(srcY,
+                                       srcYStride,
+                                       srcUV,
+                                       srcUVStride,
+                                       _cropWidth,
+                                       _cropHeight,
+                                       i420Buffer.mutableDataY,
+                                       i420Buffer.strideY,
+                                       i420Buffer.mutableDataU,
+                                       i420Buffer.strideU,
+                                       i420Buffer.mutableDataV,
+                                       i420Buffer.strideV,
+                                       i420Buffer.width,
+                                       i420Buffer.height);
+      break;
+    }
+    case kCVPixelFormatType_32BGRA:
+    case kCVPixelFormatType_32ARGB: {
+      CVPixelBufferRef scaledPixelBuffer = NULL;
+      CVPixelBufferRef sourcePixelBuffer = NULL;
+      if ([self requiresCropping] ||
+          [self requiresScalingToWidth:i420Buffer.width height:i420Buffer.height]) {
+        CVPixelBufferCreate(
+            NULL, i420Buffer.width, i420Buffer.height, pixelFormat, NULL, &scaledPixelBuffer);
+        [self cropAndScaleTo:scaledPixelBuffer withTempBuffer:NULL];
+
+        CVPixelBufferLockBaseAddress(scaledPixelBuffer, kCVPixelBufferLock_ReadOnly);
+        sourcePixelBuffer = scaledPixelBuffer;
+      } else {
+        sourcePixelBuffer = _pixelBuffer;
+      }
+      const uint8_t* src = static_cast<uint8_t*>(CVPixelBufferGetBaseAddress(sourcePixelBuffer));
+      const size_t bytesPerRow = CVPixelBufferGetBytesPerRow(sourcePixelBuffer);
+
+      if (pixelFormat == kCVPixelFormatType_32BGRA) {
+        // Corresponds to libyuv::FOURCC_ARGB
+        libyuv::ARGBToI420(src,
+                           bytesPerRow,
+                           i420Buffer.mutableDataY,
+                           i420Buffer.strideY,
+                           i420Buffer.mutableDataU,
+                           i420Buffer.strideU,
+                           i420Buffer.mutableDataV,
+                           i420Buffer.strideV,
+                           i420Buffer.width,
+                           i420Buffer.height);
+      } else if (pixelFormat == kCVPixelFormatType_32ARGB) {
+        // Corresponds to libyuv::FOURCC_BGRA
+        libyuv::BGRAToI420(src,
+                           bytesPerRow,
+                           i420Buffer.mutableDataY,
+                           i420Buffer.strideY,
+                           i420Buffer.mutableDataU,
+                           i420Buffer.strideU,
+                           i420Buffer.mutableDataV,
+                           i420Buffer.strideV,
+                           i420Buffer.width,
+                           i420Buffer.height);
+      }
+
+      if (scaledPixelBuffer) {
+        CVPixelBufferUnlockBaseAddress(scaledPixelBuffer, kCVPixelBufferLock_ReadOnly);
+        CVBufferRelease(scaledPixelBuffer);
+      }
+      break;
+    }
+    default: {
+      RTC_DCHECK_NOTREACHED() << "Unsupported pixel format.";
+    }
+  }
+
+  CVPixelBufferUnlockBaseAddress(_pixelBuffer, kCVPixelBufferLock_ReadOnly);
+
+  return i420Buffer;
+}
+
+#pragma mark - Debugging
+
+#if !defined(NDEBUG) && defined(WEBRTC_IOS)
+- (id)debugQuickLookObject {
+  CGImageRef cgImage;
+  VTCreateCGImageFromCVPixelBuffer(_pixelBuffer, NULL, &cgImage);
+  UIImage *image = [UIImage imageWithCGImage:cgImage scale:1.0 orientation:UIImageOrientationUp];
+  CGImageRelease(cgImage);
+  return image;
+}
+#endif
+
+#pragma mark - Private
+
+- (void)cropAndScaleNV12To:(CVPixelBufferRef)outputPixelBuffer withTempBuffer:(uint8_t*)tmpBuffer {
+  // Prepare output pointers.
+  CVReturn cvRet = CVPixelBufferLockBaseAddress(outputPixelBuffer, 0);
+  if (cvRet != kCVReturnSuccess) {
+    RTC_LOG(LS_ERROR) << "Failed to lock base address: " << cvRet;
+  }
+  const int dstWidth = CVPixelBufferGetWidth(outputPixelBuffer);
+  const int dstHeight = CVPixelBufferGetHeight(outputPixelBuffer);
+  uint8_t* dstY =
+      reinterpret_cast<uint8_t*>(CVPixelBufferGetBaseAddressOfPlane(outputPixelBuffer, 0));
+  const int dstYStride = CVPixelBufferGetBytesPerRowOfPlane(outputPixelBuffer, 0);
+  uint8_t* dstUV =
+      reinterpret_cast<uint8_t*>(CVPixelBufferGetBaseAddressOfPlane(outputPixelBuffer, 1));
+  const int dstUVStride = CVPixelBufferGetBytesPerRowOfPlane(outputPixelBuffer, 1);
+
+  // Prepare source pointers.
+  CVPixelBufferLockBaseAddress(_pixelBuffer, kCVPixelBufferLock_ReadOnly);
+  const uint8_t* srcY = static_cast<uint8_t*>(CVPixelBufferGetBaseAddressOfPlane(_pixelBuffer, 0));
+  const int srcYStride = CVPixelBufferGetBytesPerRowOfPlane(_pixelBuffer, 0);
+  const uint8_t* srcUV = static_cast<uint8_t*>(CVPixelBufferGetBaseAddressOfPlane(_pixelBuffer, 1));
+  const int srcUVStride = CVPixelBufferGetBytesPerRowOfPlane(_pixelBuffer, 1);
+
+  // Crop just by modifying pointers.
+  srcY += srcYStride * _cropY + _cropX;
+  srcUV += srcUVStride * (_cropY / 2) + _cropX;
+
+  webrtc::NV12Scale(tmpBuffer,
+                    srcY,
+                    srcYStride,
+                    srcUV,
+                    srcUVStride,
+                    _cropWidth,
+                    _cropHeight,
+                    dstY,
+                    dstYStride,
+                    dstUV,
+                    dstUVStride,
+                    dstWidth,
+                    dstHeight);
+
+  CVPixelBufferUnlockBaseAddress(_pixelBuffer, kCVPixelBufferLock_ReadOnly);
+  CVPixelBufferUnlockBaseAddress(outputPixelBuffer, 0);
+}
+
+- (void)cropAndScaleARGBTo:(CVPixelBufferRef)outputPixelBuffer {
+  // Prepare output pointers.
+  CVReturn cvRet = CVPixelBufferLockBaseAddress(outputPixelBuffer, 0);
+  if (cvRet != kCVReturnSuccess) {
+    RTC_LOG(LS_ERROR) << "Failed to lock base address: " << cvRet;
+  }
+  const int dstWidth = CVPixelBufferGetWidth(outputPixelBuffer);
+  const int dstHeight = CVPixelBufferGetHeight(outputPixelBuffer);
+
+  uint8_t* dst = reinterpret_cast<uint8_t*>(CVPixelBufferGetBaseAddress(outputPixelBuffer));
+  const int dstStride = CVPixelBufferGetBytesPerRow(outputPixelBuffer);
+
+  // Prepare source pointers.
+  CVPixelBufferLockBaseAddress(_pixelBuffer, kCVPixelBufferLock_ReadOnly);
+  const uint8_t* src = static_cast<uint8_t*>(CVPixelBufferGetBaseAddress(_pixelBuffer));
+  const int srcStride = CVPixelBufferGetBytesPerRow(_pixelBuffer);
+
+  // Crop just by modifying pointers. Need to ensure that src pointer points to a byte corresponding
+  // to the start of a new pixel (byte with B for BGRA) so that libyuv scales correctly.
+  const int bytesPerPixel = 4;
+  src += srcStride * _cropY + (_cropX * bytesPerPixel);
+
+  // kCVPixelFormatType_32BGRA corresponds to libyuv::FOURCC_ARGB
+  libyuv::ARGBScale(src,
+                    srcStride,
+                    _cropWidth,
+                    _cropHeight,
+                    dst,
+                    dstStride,
+                    dstWidth,
+                    dstHeight,
+                    libyuv::kFilterBox);
+
+  CVPixelBufferUnlockBaseAddress(_pixelBuffer, kCVPixelBufferLock_ReadOnly);
+  CVPixelBufferUnlockBaseAddress(outputPixelBuffer, 0);
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/helpers/AVCaptureSession+DevicePosition.h b/third_party/libwebrtc/sdk/objc/helpers/AVCaptureSession+DevicePosition.h
new file mode 100644
index 0000000000..32ab6877f0
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/helpers/AVCaptureSession+DevicePosition.h
@@ -0,0 +1,23 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <AVFoundation/AVFoundation.h>
+#import <CoreMedia/CoreMedia.h>
+
+NS_ASSUME_NONNULL_BEGIN
+
+@interface AVCaptureSession (DevicePosition)
+
+// Check the image's EXIF for the camera the image came from.
++ (AVCaptureDevicePosition)devicePositionForSampleBuffer:(CMSampleBufferRef)sampleBuffer;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/helpers/AVCaptureSession+DevicePosition.mm b/third_party/libwebrtc/sdk/objc/helpers/AVCaptureSession+DevicePosition.mm
new file mode 100644
index 0000000000..0814ecc6c5
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/helpers/AVCaptureSession+DevicePosition.mm
@@ -0,0 +1,51 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "AVCaptureSession+DevicePosition.h"
+
+BOOL CFStringContainsString(CFStringRef theString, CFStringRef stringToFind) {
+  return CFStringFindWithOptions(theString,
+                                 stringToFind,
+                                 CFRangeMake(0, CFStringGetLength(theString)),
+                                 kCFCompareCaseInsensitive,
+                                 nil);
+}
+
+@implementation AVCaptureSession (DevicePosition)
+
++ (AVCaptureDevicePosition)devicePositionForSampleBuffer:(CMSampleBufferRef)sampleBuffer {
+  // Check the image's EXIF for the camera the image came from.
+  AVCaptureDevicePosition cameraPosition = AVCaptureDevicePositionUnspecified;
+  CFDictionaryRef attachments = CMCopyDictionaryOfAttachments(
+      kCFAllocatorDefault, sampleBuffer, kCMAttachmentMode_ShouldPropagate);
+  if (attachments) {
+    int size = CFDictionaryGetCount(attachments);
+    if (size > 0) {
+      CFDictionaryRef cfExifDictVal = nil;
+      if (CFDictionaryGetValueIfPresent(
+              attachments, (const void *)CFSTR("{Exif}"), (const void **)&cfExifDictVal)) {
+        CFStringRef cfLensModelStrVal;
+        if (CFDictionaryGetValueIfPresent(cfExifDictVal,
+                                          (const void *)CFSTR("LensModel"),
+                                          (const void **)&cfLensModelStrVal)) {
+          if (CFStringContainsString(cfLensModelStrVal, CFSTR("front"))) {
+            cameraPosition = AVCaptureDevicePositionFront;
+          } else if (CFStringContainsString(cfLensModelStrVal, CFSTR("back"))) {
+            cameraPosition = AVCaptureDevicePositionBack;
+          }
+        }
+      }
+    }
+    CFRelease(attachments);
+  }
+  return cameraPosition;
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/helpers/NSString+StdString.h b/third_party/libwebrtc/sdk/objc/helpers/NSString+StdString.h
new file mode 100644
index 0000000000..b0324e8a19
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/helpers/NSString+StdString.h
@@ -0,0 +1,34 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#include <string>
+
+#include "absl/strings/string_view.h"
+
+NS_ASSUME_NONNULL_BEGIN
+
+@interface NSString (StdString)
+
+@property(nonatomic, readonly) std::string stdString;
+
++ (std::string)stdStringForString:(NSString *)nsString;
++ (NSString *)stringForStdString:(const std::string &)stdString;
+
+@end
+
+@interface NSString (AbslStringView)
+
++ (NSString *)stringForAbslStringView:(const absl::string_view)abslStringView;
+
+@end
+
+NS_ASSUME_NONNULL_END
diff --git a/third_party/libwebrtc/sdk/objc/helpers/NSString+StdString.mm b/third_party/libwebrtc/sdk/objc/helpers/NSString+StdString.mm
new file mode 100644
index 0000000000..c98432c445
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/helpers/NSString+StdString.mm
@@ -0,0 +1,45 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "NSString+StdString.h"
+
+#include "absl/strings/string_view.h"
+
+@implementation NSString (StdString)
+
+- (std::string)stdString {
+  return [NSString stdStringForString:self];
+}
+
++ (std::string)stdStringForString:(NSString *)nsString {
+  NSData *charData = [nsString dataUsingEncoding:NSUTF8StringEncoding];
+  return std::string(reinterpret_cast<const char *>(charData.bytes),
+                     charData.length);
+}
+
++ (NSString *)stringForStdString:(const std::string&)stdString {
+  // std::string may contain null termination character so we construct
+  // using length.
+  return [[NSString alloc] initWithBytes:stdString.data()
+                                  length:stdString.length()
+                                encoding:NSUTF8StringEncoding];
+}
+
+@end
+
+@implementation NSString (AbslStringView)
+
++ (NSString *)stringForAbslStringView:(const absl::string_view)abslStringView {
+  return [[NSString alloc] initWithBytes:abslStringView.data()
+                                  length:abslStringView.length()
+                                encoding:NSUTF8StringEncoding];
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/helpers/RTCCameraPreviewView.h b/third_party/libwebrtc/sdk/objc/helpers/RTCCameraPreviewView.h
new file mode 100644
index 0000000000..db9b15a45c
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/helpers/RTCCameraPreviewView.h
@@ -0,0 +1,30 @@
+/*
+ *  Copyright 2015 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+#import <UIKit/UIKit.h>
+
+#import "RTCMacros.h"
+
+@class AVCaptureSession;
+
+/** RTCCameraPreviewView is a view that renders local video from an
+ *  AVCaptureSession.
+ */
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCCameraPreviewView) : UIView
+
+/** The capture session being rendered in the view. Capture session
+ *  is assigned to AVCaptureVideoPreviewLayer async in the same
+ *  queue that the AVCaptureSession is started/stopped.
+ */
+@property(nonatomic, strong) AVCaptureSession* captureSession;
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/helpers/RTCCameraPreviewView.m b/third_party/libwebrtc/sdk/objc/helpers/RTCCameraPreviewView.m
new file mode 100644
index 0000000000..12e87d8d64
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/helpers/RTCCameraPreviewView.m
@@ -0,0 +1,123 @@
+/*
+ *  Copyright 2015 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCCameraPreviewView.h"
+
+#import <AVFoundation/AVFoundation.h>
+#import <UIKit/UIKit.h>
+
+#import "RTCDispatcher+Private.h"
+
+@implementation RTC_OBJC_TYPE (RTCCameraPreviewView)
+
+@synthesize captureSession = _captureSession;
+
++ (Class)layerClass {
+  return [AVCaptureVideoPreviewLayer class];
+}
+
+- (instancetype)initWithFrame:(CGRect)aRect {
+  self = [super initWithFrame:aRect];
+  if (self) {
+    [self addOrientationObserver];
+  }
+  return self;
+}
+
+- (instancetype)initWithCoder:(NSCoder*)aDecoder {
+  self = [super initWithCoder:aDecoder];
+  if (self) {
+    [self addOrientationObserver];
+  }
+  return self;
+}
+
+- (void)dealloc {
+  [self removeOrientationObserver];
+}
+
+- (void)setCaptureSession:(AVCaptureSession *)captureSession {
+  if (_captureSession == captureSession) {
+    return;
+  }
+  _captureSession = captureSession;
+  [RTC_OBJC_TYPE(RTCDispatcher)
+      dispatchAsyncOnType:RTCDispatcherTypeMain
+                    block:^{
+                      AVCaptureVideoPreviewLayer *previewLayer = [self previewLayer];
+                      [RTC_OBJC_TYPE(RTCDispatcher)
+                          dispatchAsyncOnType:RTCDispatcherTypeCaptureSession
+                                        block:^{
+                                          previewLayer.session = captureSession;
+                                          [RTC_OBJC_TYPE(RTCDispatcher)
+                                              dispatchAsyncOnType:RTCDispatcherTypeMain
+                                                            block:^{
+                                                              [self setCorrectVideoOrientation];
+                                                            }];
+                                        }];
+                    }];
+}
+
+- (void)layoutSubviews {
+  [super layoutSubviews];
+
+  // Update the video orientation based on the device orientation.
+  [self setCorrectVideoOrientation];
+}
+
+-(void)orientationChanged:(NSNotification *)notification {
+  [self setCorrectVideoOrientation];
+}
+
+- (void)setCorrectVideoOrientation {
+  // Get current device orientation.
+  UIDeviceOrientation deviceOrientation = [UIDevice currentDevice].orientation;
+  AVCaptureVideoPreviewLayer *previewLayer = [self previewLayer];
+
+  // First check if we are allowed to set the video orientation.
+  if (previewLayer.connection.isVideoOrientationSupported) {
+    // Set the video orientation based on device orientation.
+    if (deviceOrientation == UIInterfaceOrientationPortraitUpsideDown) {
+      previewLayer.connection.videoOrientation =
+          AVCaptureVideoOrientationPortraitUpsideDown;
+    } else if (deviceOrientation == UIInterfaceOrientationLandscapeRight) {
+      previewLayer.connection.videoOrientation =
+          AVCaptureVideoOrientationLandscapeRight;
+    } else if (deviceOrientation == UIInterfaceOrientationLandscapeLeft) {
+      previewLayer.connection.videoOrientation =
+          AVCaptureVideoOrientationLandscapeLeft;
+    } else if (deviceOrientation == UIInterfaceOrientationPortrait) {
+      previewLayer.connection.videoOrientation =
+          AVCaptureVideoOrientationPortrait;
+    }
+    // If device orientation switches to FaceUp or FaceDown, don't change video orientation.
+  }
+}
+
+#pragma mark - Private
+
+- (void)addOrientationObserver {
+  [[NSNotificationCenter defaultCenter] addObserver:self
+                                            selector:@selector(orientationChanged:)
+                                                name:UIDeviceOrientationDidChangeNotification
+                                              object:nil];
+}
+
+- (void)removeOrientationObserver {
+  [[NSNotificationCenter defaultCenter] removeObserver:self
+                                                  name:UIDeviceOrientationDidChangeNotification
+                                                object:nil];
+}
+
+- (AVCaptureVideoPreviewLayer *)previewLayer {
+  return (AVCaptureVideoPreviewLayer *)self.layer;
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/helpers/RTCDispatcher+Private.h b/third_party/libwebrtc/sdk/objc/helpers/RTCDispatcher+Private.h
new file mode 100644
index 0000000000..195c651790
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/helpers/RTCDispatcher+Private.h
@@ -0,0 +1,18 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCDispatcher.h"
+
+@interface RTC_OBJC_TYPE (RTCDispatcher)
+()
+
+    + (dispatch_queue_t)dispatchQueueForType : (RTCDispatcherQueueType)dispatchType;
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/helpers/RTCDispatcher.h b/third_party/libwebrtc/sdk/objc/helpers/RTCDispatcher.h
new file mode 100644
index 0000000000..e148af6dea
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/helpers/RTCDispatcher.h
@@ -0,0 +1,46 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+
+#import "RTCMacros.h"
+
+typedef NS_ENUM(NSInteger, RTCDispatcherQueueType) {
+  // Main dispatcher queue.
+  RTCDispatcherTypeMain,
+  // Used for starting/stopping AVCaptureSession, and assigning
+  // capture session to AVCaptureVideoPreviewLayer.
+  RTCDispatcherTypeCaptureSession,
+  // Used for operations on AVAudioSession.
+  RTCDispatcherTypeAudioSession,
+  // Used for operations on NWPathMonitor.
+  RTCDispatcherTypeNetworkMonitor,
+};
+
+/** Dispatcher that asynchronously dispatches blocks to a specific
+ *  shared dispatch queue.
+ */
+RTC_OBJC_EXPORT
+@interface RTC_OBJC_TYPE (RTCDispatcher) : NSObject
+
+- (instancetype)init NS_UNAVAILABLE;
+
+/** Dispatch the block asynchronously on the queue for dispatchType.
+ *  @param dispatchType The queue type to dispatch on.
+ *  @param block The block to dispatch asynchronously.
+ */
++ (void)dispatchAsyncOnType:(RTCDispatcherQueueType)dispatchType block:(dispatch_block_t)block;
+
+/** Returns YES if run on queue for the dispatchType otherwise NO.
+ *  Useful for asserting that a method is run on a correct queue.
+ */
++ (BOOL)isOnQueueForType:(RTCDispatcherQueueType)dispatchType;
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/helpers/RTCDispatcher.m b/third_party/libwebrtc/sdk/objc/helpers/RTCDispatcher.m
new file mode 100644
index 0000000000..4df19bc297
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/helpers/RTCDispatcher.m
@@ -0,0 +1,65 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "RTCDispatcher+Private.h"
+
+static dispatch_queue_t kAudioSessionQueue = nil;
+static dispatch_queue_t kCaptureSessionQueue = nil;
+static dispatch_queue_t kNetworkMonitorQueue = nil;
+
+@implementation RTC_OBJC_TYPE (RTCDispatcher)
+
++ (void)initialize {
+  static dispatch_once_t onceToken;
+  dispatch_once(&onceToken, ^{
+    kAudioSessionQueue = dispatch_queue_create(
+        "org.webrtc.RTCDispatcherAudioSession",
+        DISPATCH_QUEUE_SERIAL);
+    kCaptureSessionQueue = dispatch_queue_create(
+        "org.webrtc.RTCDispatcherCaptureSession",
+        DISPATCH_QUEUE_SERIAL);
+    kNetworkMonitorQueue =
+        dispatch_queue_create("org.webrtc.RTCDispatcherNetworkMonitor", DISPATCH_QUEUE_SERIAL);
+  });
+}
+
++ (void)dispatchAsyncOnType:(RTCDispatcherQueueType)dispatchType
+                      block:(dispatch_block_t)block {
+  dispatch_queue_t queue = [self dispatchQueueForType:dispatchType];
+  dispatch_async(queue, block);
+}
+
++ (BOOL)isOnQueueForType:(RTCDispatcherQueueType)dispatchType {
+  dispatch_queue_t targetQueue = [self dispatchQueueForType:dispatchType];
+  const char* targetLabel = dispatch_queue_get_label(targetQueue);
+  const char* currentLabel = dispatch_queue_get_label(DISPATCH_CURRENT_QUEUE_LABEL);
+
+  NSAssert(strlen(targetLabel) > 0, @"Label is required for the target queue.");
+  NSAssert(strlen(currentLabel) > 0, @"Label is required for the current queue.");
+
+  return strcmp(targetLabel, currentLabel) == 0;
+}
+
+#pragma mark - Private
+
++ (dispatch_queue_t)dispatchQueueForType:(RTCDispatcherQueueType)dispatchType {
+  switch (dispatchType) {
+    case RTCDispatcherTypeMain:
+      return dispatch_get_main_queue();
+    case RTCDispatcherTypeCaptureSession:
+      return kCaptureSessionQueue;
+    case RTCDispatcherTypeAudioSession:
+      return kAudioSessionQueue;
+    case RTCDispatcherTypeNetworkMonitor:
+      return kNetworkMonitorQueue;
+  }
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/helpers/UIDevice+RTCDevice.h b/third_party/libwebrtc/sdk/objc/helpers/UIDevice+RTCDevice.h
new file mode 100644
index 0000000000..ab477e2ada
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/helpers/UIDevice+RTCDevice.h
@@ -0,0 +1,111 @@
+/*
+ *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <UIKit/UIKit.h>
+
+typedef NS_ENUM(NSInteger, RTCDeviceType) {
+  RTCDeviceTypeUnknown,
+  RTCDeviceTypeIPhone1G,
+  RTCDeviceTypeIPhone3G,
+  RTCDeviceTypeIPhone3GS,
+  RTCDeviceTypeIPhone4,
+  RTCDeviceTypeIPhone4Verizon,
+  RTCDeviceTypeIPhone4S,
+  RTCDeviceTypeIPhone5GSM,
+  RTCDeviceTypeIPhone5GSM_CDMA,
+  RTCDeviceTypeIPhone5CGSM,
+  RTCDeviceTypeIPhone5CGSM_CDMA,
+  RTCDeviceTypeIPhone5SGSM,
+  RTCDeviceTypeIPhone5SGSM_CDMA,
+  RTCDeviceTypeIPhone6Plus,
+  RTCDeviceTypeIPhone6,
+  RTCDeviceTypeIPhone6S,
+  RTCDeviceTypeIPhone6SPlus,
+  RTCDeviceTypeIPhone7,
+  RTCDeviceTypeIPhone7Plus,
+  RTCDeviceTypeIPhoneSE,
+  RTCDeviceTypeIPhone8,
+  RTCDeviceTypeIPhone8Plus,
+  RTCDeviceTypeIPhoneX,
+  RTCDeviceTypeIPhoneXS,
+  RTCDeviceTypeIPhoneXSMax,
+  RTCDeviceTypeIPhoneXR,
+  RTCDeviceTypeIPhone11,
+  RTCDeviceTypeIPhone11Pro,
+  RTCDeviceTypeIPhone11ProMax,
+  RTCDeviceTypeIPhone12Mini,
+  RTCDeviceTypeIPhone12,
+  RTCDeviceTypeIPhone12Pro,
+  RTCDeviceTypeIPhone12ProMax,
+  RTCDeviceTypeIPhoneSE2Gen,
+  RTCDeviceTypeIPhone13,
+  RTCDeviceTypeIPhone13Mini,
+  RTCDeviceTypeIPhone13Pro,
+  RTCDeviceTypeIPhone13ProMax,
+
+  RTCDeviceTypeIPodTouch1G,
+  RTCDeviceTypeIPodTouch2G,
+  RTCDeviceTypeIPodTouch3G,
+  RTCDeviceTypeIPodTouch4G,
+  RTCDeviceTypeIPodTouch5G,
+  RTCDeviceTypeIPodTouch6G,
+  RTCDeviceTypeIPodTouch7G,
+  RTCDeviceTypeIPad,
+  RTCDeviceTypeIPad2Wifi,
+  RTCDeviceTypeIPad2GSM,
+  RTCDeviceTypeIPad2CDMA,
+  RTCDeviceTypeIPad2Wifi2,
+  RTCDeviceTypeIPadMiniWifi,
+  RTCDeviceTypeIPadMiniGSM,
+  RTCDeviceTypeIPadMiniGSM_CDMA,
+  RTCDeviceTypeIPad3Wifi,
+  RTCDeviceTypeIPad3GSM_CDMA,
+  RTCDeviceTypeIPad3GSM,
+  RTCDeviceTypeIPad4Wifi,
+  RTCDeviceTypeIPad4GSM,
+  RTCDeviceTypeIPad4GSM_CDMA,
+  RTCDeviceTypeIPad5,
+  RTCDeviceTypeIPad6,
+  RTCDeviceTypeIPadAirWifi,
+  RTCDeviceTypeIPadAirCellular,
+  RTCDeviceTypeIPadAirWifiCellular,
+  RTCDeviceTypeIPadAir2,
+  RTCDeviceTypeIPadMini2GWifi,
+  RTCDeviceTypeIPadMini2GCellular,
+  RTCDeviceTypeIPadMini2GWifiCellular,
+  RTCDeviceTypeIPadMini3,
+  RTCDeviceTypeIPadMini4,
+  RTCDeviceTypeIPadPro9Inch,
+  RTCDeviceTypeIPadPro12Inch,
+  RTCDeviceTypeIPadPro12Inch2,
+  RTCDeviceTypeIPadPro10Inch,
+  RTCDeviceTypeIPad7Gen10Inch,
+  RTCDeviceTypeIPadPro3Gen11Inch,
+  RTCDeviceTypeIPadPro3Gen12Inch,
+  RTCDeviceTypeIPadPro4Gen11Inch,
+  RTCDeviceTypeIPadPro4Gen12Inch,
+  RTCDeviceTypeIPadMini5Gen,
+  RTCDeviceTypeIPadAir3Gen,
+  RTCDeviceTypeIPad8,
+  RTCDeviceTypeIPad9,
+  RTCDeviceTypeIPadMini6,
+  RTCDeviceTypeIPadAir4Gen,
+  RTCDeviceTypeIPadPro5Gen11Inch,
+  RTCDeviceTypeIPadPro5Gen12Inch,
+  RTCDeviceTypeSimulatori386,
+  RTCDeviceTypeSimulatorx86_64,
+};
+
+@interface UIDevice (RTCDevice)
+
++ (RTCDeviceType)deviceType;
++ (BOOL)isIOS11OrLater;
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/helpers/UIDevice+RTCDevice.mm b/third_party/libwebrtc/sdk/objc/helpers/UIDevice+RTCDevice.mm
new file mode 100644
index 0000000000..77a5d79f79
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/helpers/UIDevice+RTCDevice.mm
@@ -0,0 +1,171 @@
+/*
+ *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "UIDevice+RTCDevice.h"
+
+#import <sys/utsname.h>
+#include <memory>
+
+@implementation UIDevice (RTCDevice)
+
++ (RTCDeviceType)deviceType {
+  NSDictionary *machineNameToType = @{
+    @"iPhone1,1" : @(RTCDeviceTypeIPhone1G),
+    @"iPhone1,2" : @(RTCDeviceTypeIPhone3G),
+    @"iPhone2,1" : @(RTCDeviceTypeIPhone3GS),
+    @"iPhone3,1" : @(RTCDeviceTypeIPhone4),
+    @"iPhone3,2" : @(RTCDeviceTypeIPhone4),
+    @"iPhone3,3" : @(RTCDeviceTypeIPhone4Verizon),
+    @"iPhone4,1" : @(RTCDeviceTypeIPhone4S),
+    @"iPhone5,1" : @(RTCDeviceTypeIPhone5GSM),
+    @"iPhone5,2" : @(RTCDeviceTypeIPhone5GSM_CDMA),
+    @"iPhone5,3" : @(RTCDeviceTypeIPhone5CGSM),
+    @"iPhone5,4" : @(RTCDeviceTypeIPhone5CGSM_CDMA),
+    @"iPhone6,1" : @(RTCDeviceTypeIPhone5SGSM),
+    @"iPhone6,2" : @(RTCDeviceTypeIPhone5SGSM_CDMA),
+    @"iPhone7,1" : @(RTCDeviceTypeIPhone6Plus),
+    @"iPhone7,2" : @(RTCDeviceTypeIPhone6),
+    @"iPhone8,1" : @(RTCDeviceTypeIPhone6S),
+    @"iPhone8,2" : @(RTCDeviceTypeIPhone6SPlus),
+    @"iPhone8,4" : @(RTCDeviceTypeIPhoneSE),
+    @"iPhone9,1" : @(RTCDeviceTypeIPhone7),
+    @"iPhone9,2" : @(RTCDeviceTypeIPhone7Plus),
+    @"iPhone9,3" : @(RTCDeviceTypeIPhone7),
+    @"iPhone9,4" : @(RTCDeviceTypeIPhone7Plus),
+    @"iPhone10,1" : @(RTCDeviceTypeIPhone8),
+    @"iPhone10,2" : @(RTCDeviceTypeIPhone8Plus),
+    @"iPhone10,3" : @(RTCDeviceTypeIPhoneX),
+    @"iPhone10,4" : @(RTCDeviceTypeIPhone8),
+    @"iPhone10,5" : @(RTCDeviceTypeIPhone8Plus),
+    @"iPhone10,6" : @(RTCDeviceTypeIPhoneX),
+    @"iPhone11,2" : @(RTCDeviceTypeIPhoneXS),
+    @"iPhone11,4" : @(RTCDeviceTypeIPhoneXSMax),
+    @"iPhone11,6" : @(RTCDeviceTypeIPhoneXSMax),
+    @"iPhone11,8" : @(RTCDeviceTypeIPhoneXR),
+    @"iPhone12,1" : @(RTCDeviceTypeIPhone11),
+    @"iPhone12,3" : @(RTCDeviceTypeIPhone11Pro),
+    @"iPhone12,5" : @(RTCDeviceTypeIPhone11ProMax),
+    @"iPhone12,8" : @(RTCDeviceTypeIPhoneSE2Gen),
+    @"iPhone13,1" : @(RTCDeviceTypeIPhone12Mini),
+    @"iPhone13,2" : @(RTCDeviceTypeIPhone12),
+    @"iPhone13,3" : @(RTCDeviceTypeIPhone12Pro),
+    @"iPhone13,4" : @(RTCDeviceTypeIPhone12ProMax),
+    @"iPhone14,5" : @(RTCDeviceTypeIPhone13),
+    @"iPhone14,4" : @(RTCDeviceTypeIPhone13Mini),
+    @"iPhone14,2" : @(RTCDeviceTypeIPhone13Pro),
+    @"iPhone14,3" : @(RTCDeviceTypeIPhone13ProMax),
+    @"iPod1,1" : @(RTCDeviceTypeIPodTouch1G),
+    @"iPod2,1" : @(RTCDeviceTypeIPodTouch2G),
+    @"iPod3,1" : @(RTCDeviceTypeIPodTouch3G),
+    @"iPod4,1" : @(RTCDeviceTypeIPodTouch4G),
+    @"iPod5,1" : @(RTCDeviceTypeIPodTouch5G),
+    @"iPod7,1" : @(RTCDeviceTypeIPodTouch6G),
+    @"iPod9,1" : @(RTCDeviceTypeIPodTouch7G),
+    @"iPad1,1" : @(RTCDeviceTypeIPad),
+    @"iPad2,1" : @(RTCDeviceTypeIPad2Wifi),
+    @"iPad2,2" : @(RTCDeviceTypeIPad2GSM),
+    @"iPad2,3" : @(RTCDeviceTypeIPad2CDMA),
+    @"iPad2,4" : @(RTCDeviceTypeIPad2Wifi2),
+    @"iPad2,5" : @(RTCDeviceTypeIPadMiniWifi),
+    @"iPad2,6" : @(RTCDeviceTypeIPadMiniGSM),
+    @"iPad2,7" : @(RTCDeviceTypeIPadMiniGSM_CDMA),
+    @"iPad3,1" : @(RTCDeviceTypeIPad3Wifi),
+    @"iPad3,2" : @(RTCDeviceTypeIPad3GSM_CDMA),
+    @"iPad3,3" : @(RTCDeviceTypeIPad3GSM),
+    @"iPad3,4" : @(RTCDeviceTypeIPad4Wifi),
+    @"iPad3,5" : @(RTCDeviceTypeIPad4GSM),
+    @"iPad3,6" : @(RTCDeviceTypeIPad4GSM_CDMA),
+    @"iPad4,1" : @(RTCDeviceTypeIPadAirWifi),
+    @"iPad4,2" : @(RTCDeviceTypeIPadAirCellular),
+    @"iPad4,3" : @(RTCDeviceTypeIPadAirWifiCellular),
+    @"iPad4,4" : @(RTCDeviceTypeIPadMini2GWifi),
+    @"iPad4,5" : @(RTCDeviceTypeIPadMini2GCellular),
+    @"iPad4,6" : @(RTCDeviceTypeIPadMini2GWifiCellular),
+    @"iPad4,7" : @(RTCDeviceTypeIPadMini3),
+    @"iPad4,8" : @(RTCDeviceTypeIPadMini3),
+    @"iPad4,9" : @(RTCDeviceTypeIPadMini3),
+    @"iPad5,1" : @(RTCDeviceTypeIPadMini4),
+    @"iPad5,2" : @(RTCDeviceTypeIPadMini4),
+    @"iPad5,3" : @(RTCDeviceTypeIPadAir2),
+    @"iPad5,4" : @(RTCDeviceTypeIPadAir2),
+    @"iPad6,3" : @(RTCDeviceTypeIPadPro9Inch),
+    @"iPad6,4" : @(RTCDeviceTypeIPadPro9Inch),
+    @"iPad6,7" : @(RTCDeviceTypeIPadPro12Inch),
+    @"iPad6,8" : @(RTCDeviceTypeIPadPro12Inch),
+    @"iPad6,11" : @(RTCDeviceTypeIPad5),
+    @"iPad6,12" : @(RTCDeviceTypeIPad5),
+    @"iPad7,1" : @(RTCDeviceTypeIPadPro12Inch2),
+    @"iPad7,2" : @(RTCDeviceTypeIPadPro12Inch2),
+    @"iPad7,3" : @(RTCDeviceTypeIPadPro10Inch),
+    @"iPad7,4" : @(RTCDeviceTypeIPadPro10Inch),
+    @"iPad7,5" : @(RTCDeviceTypeIPad6),
+    @"iPad7,6" : @(RTCDeviceTypeIPad6),
+    @"iPad7,11" : @(RTCDeviceTypeIPad7Gen10Inch),
+    @"iPad7,12" : @(RTCDeviceTypeIPad7Gen10Inch),
+    @"iPad8,1" : @(RTCDeviceTypeIPadPro3Gen11Inch),
+    @"iPad8,2" : @(RTCDeviceTypeIPadPro3Gen11Inch),
+    @"iPad8,3" : @(RTCDeviceTypeIPadPro3Gen11Inch),
+    @"iPad8,4" : @(RTCDeviceTypeIPadPro3Gen11Inch),
+    @"iPad8,5" : @(RTCDeviceTypeIPadPro3Gen12Inch),
+    @"iPad8,6" : @(RTCDeviceTypeIPadPro3Gen12Inch),
+    @"iPad8,7" : @(RTCDeviceTypeIPadPro3Gen12Inch),
+    @"iPad8,8" : @(RTCDeviceTypeIPadPro3Gen12Inch),
+    @"iPad8,9" : @(RTCDeviceTypeIPadPro4Gen11Inch),
+    @"iPad8,10" : @(RTCDeviceTypeIPadPro4Gen11Inch),
+    @"iPad8,11" : @(RTCDeviceTypeIPadPro4Gen12Inch),
+    @"iPad8,12" : @(RTCDeviceTypeIPadPro4Gen12Inch),
+    @"iPad11,1" : @(RTCDeviceTypeIPadMini5Gen),
+    @"iPad11,2" : @(RTCDeviceTypeIPadMini5Gen),
+    @"iPad11,3" : @(RTCDeviceTypeIPadAir3Gen),
+    @"iPad11,4" : @(RTCDeviceTypeIPadAir3Gen),
+    @"iPad11,6" : @(RTCDeviceTypeIPad8),
+    @"iPad11,7" : @(RTCDeviceTypeIPad8),
+    @"iPad12,1" : @(RTCDeviceTypeIPad9),
+    @"iPad12,2" : @(RTCDeviceTypeIPad9),
+    @"iPad13,1" : @(RTCDeviceTypeIPadAir4Gen),
+    @"iPad13,2" : @(RTCDeviceTypeIPadAir4Gen),
+    @"iPad13,4" : @(RTCDeviceTypeIPadPro5Gen11Inch),
+    @"iPad13,5" : @(RTCDeviceTypeIPadPro5Gen11Inch),
+    @"iPad13,6" : @(RTCDeviceTypeIPadPro5Gen11Inch),
+    @"iPad13,7" : @(RTCDeviceTypeIPadPro5Gen11Inch),
+    @"iPad13,8" : @(RTCDeviceTypeIPadPro5Gen12Inch),
+    @"iPad13,9" : @(RTCDeviceTypeIPadPro5Gen12Inch),
+    @"iPad13,10" : @(RTCDeviceTypeIPadPro5Gen12Inch),
+    @"iPad13,11" : @(RTCDeviceTypeIPadPro5Gen12Inch),
+    @"iPad14,1" : @(RTCDeviceTypeIPadMini6),
+    @"iPad14,2" : @(RTCDeviceTypeIPadMini6),
+    @"i386" : @(RTCDeviceTypeSimulatori386),
+    @"x86_64" : @(RTCDeviceTypeSimulatorx86_64),
+  };
+
+  RTCDeviceType deviceType = RTCDeviceTypeUnknown;
+  NSNumber *typeNumber = machineNameToType[[self machineName]];
+  if (typeNumber) {
+    deviceType = static_cast<RTCDeviceType>(typeNumber.integerValue);
+  }
+  return deviceType;
+}
+
++ (NSString *)machineName {
+  struct utsname systemInfo;
+  uname(&systemInfo);
+  return [[NSString alloc] initWithCString:systemInfo.machine
+                                  encoding:NSUTF8StringEncoding];
+}
+
++ (double)currentDeviceSystemVersion {
+  return [self currentDevice].systemVersion.doubleValue;
+}
+
++ (BOOL)isIOS11OrLater {
+  return [self currentDeviceSystemVersion] >= 11.0;
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/helpers/noop.mm b/third_party/libwebrtc/sdk/objc/helpers/noop.mm
new file mode 100644
index 0000000000..16a8e6d5c1
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/helpers/noop.mm
@@ -0,0 +1,13 @@
+/*
+ *  Copyright 2015 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+// This file is only needed to make ninja happy on some platforms.
+// On some platforms it is not possible to link an rtc_static_library
+// without any source file listed in the GN target.
diff --git a/third_party/libwebrtc/sdk/objc/helpers/scoped_cftyperef.h b/third_party/libwebrtc/sdk/objc/helpers/scoped_cftyperef.h
new file mode 100644
index 0000000000..092f02b3af
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/helpers/scoped_cftyperef.h
@@ -0,0 +1,116 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ *
+ */
+
+#ifndef SDK_OBJC_HELPERS_SCOPED_CFTYPEREF_H_
+#define SDK_OBJC_HELPERS_SCOPED_CFTYPEREF_H_
+
+#include <CoreFoundation/CoreFoundation.h>
+namespace rtc {
+
+// RETAIN: ScopedTypeRef should retain the object when it takes
+// ownership.
+// ASSUME: Assume the object already has already been retained.
+// ScopedTypeRef takes over ownership.
+enum class RetainPolicy { RETAIN, ASSUME };
+
+namespace internal {
+template <typename T>
+struct CFTypeRefTraits {
+  static T InvalidValue() { return nullptr; }
+  static void Release(T ref) { CFRelease(ref); }
+  static T Retain(T ref) {
+    CFRetain(ref);
+    return ref;
+  }
+};
+
+template <typename T, typename Traits>
+class ScopedTypeRef {
+ public:
+  ScopedTypeRef() : ptr_(Traits::InvalidValue()) {}
+  explicit ScopedTypeRef(T ptr) : ptr_(ptr) {}
+  ScopedTypeRef(T ptr, RetainPolicy policy) : ScopedTypeRef(ptr) {
+    if (ptr_ && policy == RetainPolicy::RETAIN)
+      Traits::Retain(ptr_);
+  }
+
+  ScopedTypeRef(const ScopedTypeRef<T, Traits>& rhs) : ptr_(rhs.ptr_) {
+    if (ptr_)
+      ptr_ = Traits::Retain(ptr_);
+  }
+
+  ~ScopedTypeRef() {
+    if (ptr_) {
+      Traits::Release(ptr_);
+    }
+  }
+
+  T get() const { return ptr_; }
+  T operator->() const { return ptr_; }
+  explicit operator bool() const { return ptr_; }
+
+  bool operator!() const { return !ptr_; }
+
+  ScopedTypeRef& operator=(const T& rhs) {
+    if (ptr_)
+      Traits::Release(ptr_);
+    ptr_ = rhs;
+    return *this;
+  }
+
+  ScopedTypeRef& operator=(const ScopedTypeRef<T, Traits>& rhs) {
+    reset(rhs.get(), RetainPolicy::RETAIN);
+    return *this;
+  }
+
+  // This is intended to take ownership of objects that are
+  // created by pass-by-pointer initializers.
+  T* InitializeInto() {
+    RTC_DCHECK(!ptr_);
+    return &ptr_;
+  }
+
+  void reset(T ptr, RetainPolicy policy = RetainPolicy::ASSUME) {
+    if (ptr && policy == RetainPolicy::RETAIN)
+      Traits::Retain(ptr);
+    if (ptr_)
+      Traits::Release(ptr_);
+    ptr_ = ptr;
+  }
+
+  T release() {
+    T temp = ptr_;
+    ptr_ = Traits::InvalidValue();
+    return temp;
+  }
+
+ private:
+  T ptr_;
+};
+}  // namespace internal
+
+template <typename T>
+using ScopedCFTypeRef =
+    internal::ScopedTypeRef<T, internal::CFTypeRefTraits<T>>;
+
+template <typename T>
+static ScopedCFTypeRef<T> AdoptCF(T cftype) {
+  return ScopedCFTypeRef<T>(cftype, RetainPolicy::RETAIN);
+}
+
+template <typename T>
+static ScopedCFTypeRef<T> ScopedCF(T cftype) {
+  return ScopedCFTypeRef<T>(cftype);
+}
+
+}  // namespace rtc
+
+#endif  // SDK_OBJC_HELPERS_SCOPED_CFTYPEREF_H_
diff --git a/third_party/libwebrtc/sdk/objc/native/api/audio_device_module.h b/third_party/libwebrtc/sdk/objc/native/api/audio_device_module.h
new file mode 100644
index 0000000000..3405469709
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/native/api/audio_device_module.h
@@ -0,0 +1,30 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef SDK_OBJC_NATIVE_API_AUDIO_DEVICE_MODULE_H_
+#define SDK_OBJC_NATIVE_API_AUDIO_DEVICE_MODULE_H_
+
+#include <memory>
+
+#include "modules/audio_device/include/audio_device.h"
+
+namespace webrtc {
+
+// If `bypass_voice_processing` is true, WebRTC will attempt to disable hardware
+// audio processing on iOS.
+// Warning: Setting `bypass_voice_processing` will have unpredictable
+// consequences for the audio path in the device. It is not advisable to use in
+// most scenarios.
+rtc::scoped_refptr<AudioDeviceModule> CreateAudioDeviceModule(
+    bool bypass_voice_processing = false);
+
+}  // namespace webrtc
+
+#endif  // SDK_OBJC_NATIVE_API_AUDIO_DEVICE_MODULE_H_
diff --git a/third_party/libwebrtc/sdk/objc/native/api/audio_device_module.mm b/third_party/libwebrtc/sdk/objc/native/api/audio_device_module.mm
new file mode 100644
index 0000000000..4e7b681e69
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/native/api/audio_device_module.mm
@@ -0,0 +1,29 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "audio_device_module.h"
+
+#include "api/make_ref_counted.h"
+#include "rtc_base/logging.h"
+
+#include "sdk/objc/native/src/audio/audio_device_module_ios.h"
+
+namespace webrtc {
+
+rtc::scoped_refptr<AudioDeviceModule> CreateAudioDeviceModule(bool bypass_voice_processing) {
+  RTC_DLOG(LS_INFO) << __FUNCTION__;
+#if defined(WEBRTC_IOS)
+  return rtc::make_ref_counted<ios_adm::AudioDeviceModuleIOS>(bypass_voice_processing);
+#else
+  RTC_LOG(LS_ERROR) << "current platform is not supported => this module will self destruct!";
+  return nullptr;
+#endif
+}
+}
diff --git a/third_party/libwebrtc/sdk/objc/native/api/network_monitor_factory.h b/third_party/libwebrtc/sdk/objc/native/api/network_monitor_factory.h
new file mode 100644
index 0000000000..903c66893d
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/native/api/network_monitor_factory.h
@@ -0,0 +1,24 @@
+/*
+ *  Copyright 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef SDK_OBJC_NATIVE_API_NETWORK_MONITOR_FACTORY_H_
+#define SDK_OBJC_NATIVE_API_NETWORK_MONITOR_FACTORY_H_
+
+#include <memory>
+
+#include "rtc_base/network_monitor_factory.h"
+
+namespace webrtc {
+
+std::unique_ptr<rtc::NetworkMonitorFactory> CreateNetworkMonitorFactory();
+
+}  // namespace webrtc
+
+#endif  // SDK_OBJC_NATIVE_API_NETWORK_MONITOR_FACTORY_H_
diff --git a/third_party/libwebrtc/sdk/objc/native/api/network_monitor_factory.mm b/third_party/libwebrtc/sdk/objc/native/api/network_monitor_factory.mm
new file mode 100644
index 0000000000..acde634b1d
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/native/api/network_monitor_factory.mm
@@ -0,0 +1,30 @@
+/*
+ *  Copyright 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "network_monitor_factory.h"
+
+#if defined(WEBRTC_IOS)
+#include "sdk/objc/native/src/objc_network_monitor.h"
+#endif
+
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+
+std::unique_ptr<rtc::NetworkMonitorFactory> CreateNetworkMonitorFactory() {
+  RTC_DLOG(LS_INFO) << __FUNCTION__;
+#if defined(WEBRTC_IOS)
+  return std::make_unique<ObjCNetworkMonitorFactory>();
+#else
+  return nullptr;
+#endif
+}
+
+}
diff --git a/third_party/libwebrtc/sdk/objc/native/api/objc_audio_device_module.h b/third_party/libwebrtc/sdk/objc/native/api/objc_audio_device_module.h
new file mode 100644
index 0000000000..0fe2dda4a0
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/native/api/objc_audio_device_module.h
@@ -0,0 +1,24 @@
+/*
+ *  Copyright 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef SDK_OBJC_NATIVE_API_OBJC_AUDIO_DEVICE_MODULE_H_
+#define SDK_OBJC_NATIVE_API_OBJC_AUDIO_DEVICE_MODULE_H_
+
+#import "components/audio/RTCAudioDevice.h"
+#include "modules/audio_device/include/audio_device.h"
+
+namespace webrtc {
+
+rtc::scoped_refptr<AudioDeviceModule> CreateAudioDeviceModule(
+    id<RTC_OBJC_TYPE(RTCAudioDevice)> audio_device);
+
+}  // namespace webrtc
+
+#endif  // SDK_OBJC_NATIVE_API_OBJC_AUDIO_DEVICE_MODULE_H_
diff --git a/third_party/libwebrtc/sdk/objc/native/api/objc_audio_device_module.mm b/third_party/libwebrtc/sdk/objc/native/api/objc_audio_device_module.mm
new file mode 100644
index 0000000000..76edd45605
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/native/api/objc_audio_device_module.mm
@@ -0,0 +1,26 @@
+/*
+ *  Copyright 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "objc_audio_device_module.h"
+
+#include "api/make_ref_counted.h"
+#include "rtc_base/logging.h"
+
+#include "sdk/objc/native/src/objc_audio_device.h"
+
+namespace webrtc {
+
+rtc::scoped_refptr<AudioDeviceModule> CreateAudioDeviceModule(
+    id<RTC_OBJC_TYPE(RTCAudioDevice)> audio_device) {
+  RTC_DLOG(LS_INFO) << __FUNCTION__;
+  return rtc::make_ref_counted<objc_adm::ObjCAudioDeviceModule>(audio_device);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/sdk/objc/native/api/ssl_certificate_verifier.h b/third_party/libwebrtc/sdk/objc/native/api/ssl_certificate_verifier.h
new file mode 100644
index 0000000000..35ab1be9a8
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/native/api/ssl_certificate_verifier.h
@@ -0,0 +1,26 @@
+/*
+ *  Copyright 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef SDK_OBJC_NATIVE_API_SSL_CERTIFICATE_VERIFIER_H_
+#define SDK_OBJC_NATIVE_API_SSL_CERTIFICATE_VERIFIER_H_
+
+#include <memory>
+
+#import "RTCSSLCertificateVerifier.h"
+#include "rtc_base/ssl_certificate.h"
+
+namespace webrtc {
+
+std::unique_ptr<rtc::SSLCertificateVerifier> ObjCToNativeCertificateVerifier(
+    id<RTC_OBJC_TYPE(RTCSSLCertificateVerifier)> objc_certificate_verifier);
+
+}  // namespace webrtc
+
+#endif  // SDK_OBJC_NATIVE_API_SSL_CERTIFICATE_VERIFIER_H_
diff --git a/third_party/libwebrtc/sdk/objc/native/api/ssl_certificate_verifier.mm b/third_party/libwebrtc/sdk/objc/native/api/ssl_certificate_verifier.mm
new file mode 100644
index 0000000000..4437402b9c
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/native/api/ssl_certificate_verifier.mm
@@ -0,0 +1,48 @@
+/*
+ *  Copyright 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "ssl_certificate_verifier.h"
+
+#include "rtc_base/buffer.h"
+
+namespace {
+
+class SSLCertificateVerifierAdapter final : public rtc::SSLCertificateVerifier {
+ public:
+  SSLCertificateVerifierAdapter(
+      id<RTC_OBJC_TYPE(RTCSSLCertificateVerifier)> objc_certificate_verifier)
+      : objc_certificate_verifier_(objc_certificate_verifier) {
+    RTC_DCHECK(objc_certificate_verifier_ != nil);
+  }
+
+  bool Verify(const rtc::SSLCertificate& certificate) override {
+    @autoreleasepool {
+      rtc::Buffer der_buffer;
+      certificate.ToDER(&der_buffer);
+      NSData* serialized_certificate = [[NSData alloc] initWithBytes:der_buffer.data()
+                                                              length:der_buffer.size()];
+      return [objc_certificate_verifier_ verify:serialized_certificate];
+    }
+  }
+
+ private:
+  id<RTC_OBJC_TYPE(RTCSSLCertificateVerifier)> objc_certificate_verifier_;
+};
+
+}
+
+namespace webrtc {
+
+std::unique_ptr<rtc::SSLCertificateVerifier> ObjCToNativeCertificateVerifier(
+    id<RTC_OBJC_TYPE(RTCSSLCertificateVerifier)> objc_certificate_verifier) {
+  return std::make_unique<SSLCertificateVerifierAdapter>(objc_certificate_verifier);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/sdk/objc/native/api/video_capturer.h b/third_party/libwebrtc/sdk/objc/native/api/video_capturer.h
new file mode 100644
index 0000000000..9847d8148b
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/native/api/video_capturer.h
@@ -0,0 +1,29 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef SDK_OBJC_NATIVE_API_VIDEO_CAPTURER_H_
+#define SDK_OBJC_NATIVE_API_VIDEO_CAPTURER_H_
+
+#import "base/RTCVideoCapturer.h"
+
+#include "api/media_stream_interface.h"
+#include "api/scoped_refptr.h"
+#include "rtc_base/thread.h"
+
+namespace webrtc {
+
+rtc::scoped_refptr<webrtc::VideoTrackSourceInterface> ObjCToNativeVideoCapturer(
+    RTC_OBJC_TYPE(RTCVideoCapturer) * objc_video_capturer,
+    rtc::Thread* signaling_thread,
+    rtc::Thread* worker_thread);
+
+}  // namespace webrtc
+
+#endif  // SDK_OBJC_NATIVE_API_VIDEO_CAPTURER_H_
diff --git a/third_party/libwebrtc/sdk/objc/native/api/video_capturer.mm b/third_party/libwebrtc/sdk/objc/native/api/video_capturer.mm
new file mode 100644
index 0000000000..a7260ab802
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/native/api/video_capturer.mm
@@ -0,0 +1,35 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "sdk/objc/native/api/video_capturer.h"
+
+#include "absl/memory/memory.h"
+#include "api/video_track_source_proxy_factory.h"
+#include "sdk/objc/native/src/objc_video_track_source.h"
+
+namespace webrtc {
+
+rtc::scoped_refptr<webrtc::VideoTrackSourceInterface> ObjCToNativeVideoCapturer(
+    RTC_OBJC_TYPE(RTCVideoCapturer) * objc_video_capturer,
+    rtc::Thread *signaling_thread,
+    rtc::Thread *worker_thread) {
+  RTCObjCVideoSourceAdapter *adapter = [[RTCObjCVideoSourceAdapter alloc] init];
+  rtc::scoped_refptr<webrtc::ObjCVideoTrackSource> objc_video_track_source =
+      rtc::make_ref_counted<webrtc::ObjCVideoTrackSource>(adapter);
+  rtc::scoped_refptr<webrtc::VideoTrackSourceInterface> video_source =
+      webrtc::CreateVideoTrackSourceProxy(
+          signaling_thread, worker_thread, objc_video_track_source.get());
+
+  objc_video_capturer.delegate = adapter;
+
+  return video_source;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/sdk/objc/native/api/video_decoder_factory.h b/third_party/libwebrtc/sdk/objc/native/api/video_decoder_factory.h
new file mode 100644
index 0000000000..03d8af3cfe
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/native/api/video_decoder_factory.h
@@ -0,0 +1,27 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef SDK_OBJC_NATIVE_API_VIDEO_DECODER_FACTORY_H_
+#define SDK_OBJC_NATIVE_API_VIDEO_DECODER_FACTORY_H_
+
+#include <memory>
+
+#import "base/RTCVideoDecoderFactory.h"
+
+#include "api/video_codecs/video_decoder_factory.h"
+
+namespace webrtc {
+
+std::unique_ptr<VideoDecoderFactory> ObjCToNativeVideoDecoderFactory(
+    id<RTC_OBJC_TYPE(RTCVideoDecoderFactory)> objc_video_decoder_factory);
+
+}  // namespace webrtc
+
+#endif  // SDK_OBJC_NATIVE_API_VIDEO_DECODER_FACTORY_H_
diff --git a/third_party/libwebrtc/sdk/objc/native/api/video_decoder_factory.mm b/third_party/libwebrtc/sdk/objc/native/api/video_decoder_factory.mm
new file mode 100644
index 0000000000..d418f2fe6f
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/native/api/video_decoder_factory.mm
@@ -0,0 +1,24 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "sdk/objc/native/api/video_decoder_factory.h"
+
+#include <memory>
+
+#include "sdk/objc/native/src/objc_video_decoder_factory.h"
+
+namespace webrtc {
+
+std::unique_ptr<VideoDecoderFactory> ObjCToNativeVideoDecoderFactory(
+    id<RTC_OBJC_TYPE(RTCVideoDecoderFactory)> objc_video_decoder_factory) {
+  return std::make_unique<ObjCVideoDecoderFactory>(objc_video_decoder_factory);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/sdk/objc/native/api/video_encoder_factory.h b/third_party/libwebrtc/sdk/objc/native/api/video_encoder_factory.h
new file mode 100644
index 0000000000..6e551b288d
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/native/api/video_encoder_factory.h
@@ -0,0 +1,27 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef SDK_OBJC_NATIVE_API_VIDEO_ENCODER_FACTORY_H_
+#define SDK_OBJC_NATIVE_API_VIDEO_ENCODER_FACTORY_H_
+
+#include <memory>
+
+#import "base/RTCVideoEncoderFactory.h"
+
+#include "api/video_codecs/video_encoder_factory.h"
+
+namespace webrtc {
+
+std::unique_ptr<VideoEncoderFactory> ObjCToNativeVideoEncoderFactory(
+    id<RTC_OBJC_TYPE(RTCVideoEncoderFactory)> objc_video_encoder_factory);
+
+}  // namespace webrtc
+
+#endif  // SDK_OBJC_NATIVE_API_VIDEO_ENCODER_FACTORY_H_
diff --git a/third_party/libwebrtc/sdk/objc/native/api/video_encoder_factory.mm b/third_party/libwebrtc/sdk/objc/native/api/video_encoder_factory.mm
new file mode 100644
index 0000000000..6fa5563f75
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/native/api/video_encoder_factory.mm
@@ -0,0 +1,24 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "sdk/objc/native/api/video_encoder_factory.h"
+
+#include <memory>
+
+#include "sdk/objc/native/src/objc_video_encoder_factory.h"
+
+namespace webrtc {
+
+std::unique_ptr<VideoEncoderFactory> ObjCToNativeVideoEncoderFactory(
+    id<RTC_OBJC_TYPE(RTCVideoEncoderFactory)> objc_video_encoder_factory) {
+  return std::make_unique<ObjCVideoEncoderFactory>(objc_video_encoder_factory);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/sdk/objc/native/api/video_frame.h b/third_party/libwebrtc/sdk/objc/native/api/video_frame.h
new file mode 100644
index 0000000000..b4416ffabe
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/native/api/video_frame.h
@@ -0,0 +1,24 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef SDK_OBJC_NATIVE_API_VIDEO_FRAME_H_
+#define SDK_OBJC_NATIVE_API_VIDEO_FRAME_H_
+
+#import "base/RTCVideoFrame.h"
+
+#include "api/video/video_frame.h"
+
+namespace webrtc {
+
+RTC_OBJC_TYPE(RTCVideoFrame) * NativeToObjCVideoFrame(const VideoFrame& frame);
+
+}  // namespace webrtc
+
+#endif  // SDK_OBJC_NATIVE_API_VIDEO_FRAME_H_
diff --git a/third_party/libwebrtc/sdk/objc/native/api/video_frame.mm b/third_party/libwebrtc/sdk/objc/native/api/video_frame.mm
new file mode 100644
index 0000000000..b82994fd5f
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/native/api/video_frame.mm
@@ -0,0 +1,21 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "sdk/objc/native/api/video_frame.h"
+
+#include "sdk/objc/native/src/objc_video_frame.h"
+
+namespace webrtc {
+
+RTC_OBJC_TYPE(RTCVideoFrame) * NativeToObjCVideoFrame(const VideoFrame& frame) {
+  return ToObjCVideoFrame(frame);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/sdk/objc/native/api/video_frame_buffer.h b/third_party/libwebrtc/sdk/objc/native/api/video_frame_buffer.h
new file mode 100644
index 0000000000..204d65d850
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/native/api/video_frame_buffer.h
@@ -0,0 +1,29 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef SDK_OBJC_NATIVE_API_VIDEO_FRAME_BUFFER_H_
+#define SDK_OBJC_NATIVE_API_VIDEO_FRAME_BUFFER_H_
+
+#import "base/RTCVideoFrameBuffer.h"
+
+#include "api/scoped_refptr.h"
+#include "common_video/include/video_frame_buffer.h"
+
+namespace webrtc {
+
+rtc::scoped_refptr<VideoFrameBuffer> ObjCToNativeVideoFrameBuffer(
+    id<RTC_OBJC_TYPE(RTCVideoFrameBuffer)> objc_video_frame_buffer);
+
+id<RTC_OBJC_TYPE(RTCVideoFrameBuffer)> NativeToObjCVideoFrameBuffer(
+    const rtc::scoped_refptr<VideoFrameBuffer>& buffer);
+
+}  // namespace webrtc
+
+#endif  // SDK_OBJC_NATIVE_API_VIDEO_FRAME_BUFFER_H_
diff --git a/third_party/libwebrtc/sdk/objc/native/api/video_frame_buffer.mm b/third_party/libwebrtc/sdk/objc/native/api/video_frame_buffer.mm
new file mode 100644
index 0000000000..4fe9037bce
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/native/api/video_frame_buffer.mm
@@ -0,0 +1,28 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "sdk/objc/native/api/video_frame_buffer.h"
+
+#include "api/make_ref_counted.h"
+#include "sdk/objc/native/src/objc_frame_buffer.h"
+
+namespace webrtc {
+
+rtc::scoped_refptr<VideoFrameBuffer> ObjCToNativeVideoFrameBuffer(
+    id<RTC_OBJC_TYPE(RTCVideoFrameBuffer)> objc_video_frame_buffer) {
+  return rtc::make_ref_counted<ObjCFrameBuffer>(objc_video_frame_buffer);
+}
+
+id<RTC_OBJC_TYPE(RTCVideoFrameBuffer)> NativeToObjCVideoFrameBuffer(
+    const rtc::scoped_refptr<VideoFrameBuffer> &buffer) {
+  return ToObjCVideoFrameBuffer(buffer);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/sdk/objc/native/api/video_renderer.h b/third_party/libwebrtc/sdk/objc/native/api/video_renderer.h
new file mode 100644
index 0000000000..04796b8049
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/native/api/video_renderer.h
@@ -0,0 +1,28 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef SDK_OBJC_NATIVE_API_VIDEO_RENDERER_H_
+#define SDK_OBJC_NATIVE_API_VIDEO_RENDERER_H_
+
+#import "base/RTCVideoRenderer.h"
+
+#include <memory>
+
+#include "api/video/video_frame.h"
+#include "api/video/video_sink_interface.h"
+
+namespace webrtc {
+
+std::unique_ptr<rtc::VideoSinkInterface<VideoFrame>> ObjCToNativeVideoRenderer(
+    id<RTC_OBJC_TYPE(RTCVideoRenderer)> objc_video_renderer);
+
+}  // namespace webrtc
+
+#endif  // SDK_OBJC_NATIVE_API_VIDEO_RENDERER_H_
diff --git a/third_party/libwebrtc/sdk/objc/native/api/video_renderer.mm b/third_party/libwebrtc/sdk/objc/native/api/video_renderer.mm
new file mode 100644
index 0000000000..e92d47d1e3
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/native/api/video_renderer.mm
@@ -0,0 +1,24 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "sdk/objc/native/api/video_renderer.h"
+
+#include <memory>
+
+#include "sdk/objc/native/src/objc_video_renderer.h"
+
+namespace webrtc {
+
+std::unique_ptr<rtc::VideoSinkInterface<VideoFrame>> ObjCToNativeVideoRenderer(
+    id<RTC_OBJC_TYPE(RTCVideoRenderer)> objc_video_renderer) {
+  return std::make_unique<ObjCVideoRenderer>(objc_video_renderer);
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/sdk/objc/native/src/audio/audio_device_ios.h b/third_party/libwebrtc/sdk/objc/native/src/audio/audio_device_ios.h
new file mode 100644
index 0000000000..a86acb56fe
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/native/src/audio/audio_device_ios.h
@@ -0,0 +1,306 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef SDK_OBJC_NATIVE_SRC_AUDIO_AUDIO_DEVICE_IOS_H_
+#define SDK_OBJC_NATIVE_SRC_AUDIO_AUDIO_DEVICE_IOS_H_
+
+#include <atomic>
+#include <memory>
+
+#include "api/scoped_refptr.h"
+#include "api/sequence_checker.h"
+#include "api/task_queue/pending_task_safety_flag.h"
+#include "audio_session_observer.h"
+#include "modules/audio_device/audio_device_generic.h"
+#include "rtc_base/buffer.h"
+#include "rtc_base/thread.h"
+#include "rtc_base/thread_annotations.h"
+#include "sdk/objc/base/RTCMacros.h"
+#include "voice_processing_audio_unit.h"
+
+RTC_FWD_DECL_OBJC_CLASS(RTCNativeAudioSessionDelegateAdapter);
+
+namespace webrtc {
+
+class FineAudioBuffer;
+
+namespace ios_adm {
+
+// Implements full duplex 16-bit mono PCM audio support for iOS using a
+// Voice-Processing (VP) I/O audio unit in Core Audio. The VP I/O audio unit
+// supports audio echo cancellation. It also adds automatic gain control,
+// adjustment of voice-processing quality and muting.
+//
+// An instance must be created and destroyed on one and the same thread.
+// All supported public methods must also be called on the same thread.
+// A thread checker will RTC_DCHECK if any supported method is called on an
+// invalid thread.
+//
+// Recorded audio will be delivered on a real-time internal I/O thread in the
+// audio unit. The audio unit will also ask for audio data to play out on this
+// same thread.
+class AudioDeviceIOS : public AudioDeviceGeneric,
+                       public AudioSessionObserver,
+                       public VoiceProcessingAudioUnitObserver {
+ public:
+  explicit AudioDeviceIOS(bool bypass_voice_processing);
+  ~AudioDeviceIOS() override;
+
+  void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) override;
+
+  InitStatus Init() override;
+  int32_t Terminate() override;
+  bool Initialized() const override;
+
+  int32_t InitPlayout() override;
+  bool PlayoutIsInitialized() const override;
+
+  int32_t InitRecording() override;
+  bool RecordingIsInitialized() const override;
+
+  int32_t StartPlayout() override;
+  int32_t StopPlayout() override;
+  bool Playing() const override;
+
+  int32_t StartRecording() override;
+  int32_t StopRecording() override;
+  bool Recording() const override;
+
+  // These methods returns hard-coded delay values and not dynamic delay
+  // estimates. The reason is that iOS supports a built-in AEC and the WebRTC
+  // AEC will always be disabled in the Libjingle layer to avoid running two
+  // AEC implementations at the same time. And, it saves resources to avoid
+  // updating these delay values continuously.
+  // TODO(henrika): it would be possible to mark these two methods as not
+  // implemented since they are only called for A/V-sync purposes today and
+  // A/V-sync is not supported on iOS. However, we avoid adding error messages
+  // the log by using these dummy implementations instead.
+  int32_t PlayoutDelay(uint16_t& delayMS) const override;
+
+  // No implementation for playout underrun on iOS. We override it to avoid a
+  // periodic log that it isn't available from the base class.
+  int32_t GetPlayoutUnderrunCount() const override { return -1; }
+
+  // Native audio parameters stored during construction.
+  // These methods are unique for the iOS implementation.
+  int GetPlayoutAudioParameters(AudioParameters* params) const override;
+  int GetRecordAudioParameters(AudioParameters* params) const override;
+
+  // These methods are currently not fully implemented on iOS:
+
+  // See audio_device_not_implemented.cc for trivial implementations.
+  int32_t ActiveAudioLayer(
+      AudioDeviceModule::AudioLayer& audioLayer) const override;
+  int32_t PlayoutIsAvailable(bool& available) override;
+  int32_t RecordingIsAvailable(bool& available) override;
+  int16_t PlayoutDevices() override;
+  int16_t RecordingDevices() override;
+  int32_t PlayoutDeviceName(uint16_t index,
+                            char name[kAdmMaxDeviceNameSize],
+                            char guid[kAdmMaxGuidSize]) override;
+  int32_t RecordingDeviceName(uint16_t index,
+                              char name[kAdmMaxDeviceNameSize],
+                              char guid[kAdmMaxGuidSize]) override;
+  int32_t SetPlayoutDevice(uint16_t index) override;
+  int32_t SetPlayoutDevice(
+      AudioDeviceModule::WindowsDeviceType device) override;
+  int32_t SetRecordingDevice(uint16_t index) override;
+  int32_t SetRecordingDevice(
+      AudioDeviceModule::WindowsDeviceType device) override;
+  int32_t InitSpeaker() override;
+  bool SpeakerIsInitialized() const override;
+  int32_t InitMicrophone() override;
+  bool MicrophoneIsInitialized() const override;
+  int32_t SpeakerVolumeIsAvailable(bool& available) override;
+  int32_t SetSpeakerVolume(uint32_t volume) override;
+  int32_t SpeakerVolume(uint32_t& volume) const override;
+  int32_t MaxSpeakerVolume(uint32_t& maxVolume) const override;
+  int32_t MinSpeakerVolume(uint32_t& minVolume) const override;
+  int32_t MicrophoneVolumeIsAvailable(bool& available) override;
+  int32_t SetMicrophoneVolume(uint32_t volume) override;
+  int32_t MicrophoneVolume(uint32_t& volume) const override;
+  int32_t MaxMicrophoneVolume(uint32_t& maxVolume) const override;
+  int32_t MinMicrophoneVolume(uint32_t& minVolume) const override;
+  int32_t MicrophoneMuteIsAvailable(bool& available) override;
+  int32_t SetMicrophoneMute(bool enable) override;
+  int32_t MicrophoneMute(bool& enabled) const override;
+  int32_t SpeakerMuteIsAvailable(bool& available) override;
+  int32_t SetSpeakerMute(bool enable) override;
+  int32_t SpeakerMute(bool& enabled) const override;
+  int32_t StereoPlayoutIsAvailable(bool& available) override;
+  int32_t SetStereoPlayout(bool enable) override;
+  int32_t StereoPlayout(bool& enabled) const override;
+  int32_t StereoRecordingIsAvailable(bool& available) override;
+  int32_t SetStereoRecording(bool enable) override;
+  int32_t StereoRecording(bool& enabled) const override;
+
+  // AudioSessionObserver methods. May be called from any thread.
+  void OnInterruptionBegin() override;
+  void OnInterruptionEnd() override;
+  void OnValidRouteChange() override;
+  void OnCanPlayOrRecordChange(bool can_play_or_record) override;
+  void OnChangedOutputVolume() override;
+
+  // VoiceProcessingAudioUnitObserver methods.
+  OSStatus OnDeliverRecordedData(AudioUnitRenderActionFlags* flags,
+                                 const AudioTimeStamp* time_stamp,
+                                 UInt32 bus_number,
+                                 UInt32 num_frames,
+                                 AudioBufferList* io_data) override;
+  OSStatus OnGetPlayoutData(AudioUnitRenderActionFlags* flags,
+                            const AudioTimeStamp* time_stamp,
+                            UInt32 bus_number,
+                            UInt32 num_frames,
+                            AudioBufferList* io_data) override;
+
+  bool IsInterrupted();
+
+ private:
+  // Called by the relevant AudioSessionObserver methods on `thread_`.
+  void HandleInterruptionBegin();
+  void HandleInterruptionEnd();
+  void HandleValidRouteChange();
+  void HandleCanPlayOrRecordChange(bool can_play_or_record);
+  void HandleSampleRateChange();
+  void HandlePlayoutGlitchDetected();
+  void HandleOutputVolumeChange();
+
+  // Uses current `playout_parameters_` and `record_parameters_` to inform the
+  // audio device buffer (ADB) about our internal audio parameters.
+  void UpdateAudioDeviceBuffer();
+
+  // Since the preferred audio parameters are only hints to the OS, the actual
+  // values may be different once the AVAudioSession has been activated.
+  // This method asks for the current hardware parameters and takes actions
+  // if they should differ from what we have asked for initially. It also
+  // defines `playout_parameters_` and `record_parameters_`.
+  void SetupAudioBuffersForActiveAudioSession();
+
+  // Creates the audio unit.
+  bool CreateAudioUnit();
+
+  // Updates the audio unit state based on current state.
+  void UpdateAudioUnit(bool can_play_or_record);
+
+  // Configures the audio session for WebRTC.
+  bool ConfigureAudioSession();
+
+  // Like above, but requires caller to already hold session lock.
+  bool ConfigureAudioSessionLocked();
+
+  // Unconfigures the audio session.
+  void UnconfigureAudioSession();
+
+  // Activates our audio session, creates and initializes the voice-processing
+  // audio unit and verifies that we got the preferred native audio parameters.
+  bool InitPlayOrRecord();
+
+  // Closes and deletes the voice-processing I/O unit.
+  void ShutdownPlayOrRecord();
+
+  // Resets thread-checkers before a call is restarted.
+  void PrepareForNewStart();
+
+  // Determines whether voice processing should be enabled or disabled.
+  const bool bypass_voice_processing_;
+
+  // Native I/O audio thread checker.
+  SequenceChecker io_thread_checker_;
+
+  // Thread that this object is created on.
+  rtc::Thread* thread_;
+
+  // Raw pointer handle provided to us in AttachAudioBuffer(). Owned by the
+  // AudioDeviceModuleImpl class and called by AudioDeviceModule::Create().
+  // The AudioDeviceBuffer is a member of the AudioDeviceModuleImpl instance
+  // and therefore outlives this object.
+  AudioDeviceBuffer* audio_device_buffer_;
+
+  // Contains audio parameters (sample rate, #channels, buffer size etc.) for
+  // the playout and recording sides. These structure is set in two steps:
+  // first, native sample rate and #channels are defined in Init(). Next, the
+  // audio session is activated and we verify that the preferred parameters
+  // were granted by the OS. At this stage it is also possible to add a third
+  // component to the parameters; the native I/O buffer duration.
+  // A RTC_CHECK will be hit if we for some reason fail to open an audio session
+  // using the specified parameters.
+  AudioParameters playout_parameters_;
+  AudioParameters record_parameters_;
+
+  // The AudioUnit used to play and record audio.
+  std::unique_ptr<VoiceProcessingAudioUnit> audio_unit_;
+
+  // FineAudioBuffer takes an AudioDeviceBuffer which delivers audio data
+  // in chunks of 10ms. It then allows for this data to be pulled in
+  // a finer or coarser granularity. I.e. interacting with this class instead
+  // of directly with the AudioDeviceBuffer one can ask for any number of
+  // audio data samples. Is also supports a similar scheme for the recording
+  // side.
+  // Example: native buffer size can be 128 audio frames at 16kHz sample rate.
+  // WebRTC will provide 480 audio frames per 10ms but iOS asks for 128
+  // in each callback (one every 8ms). This class can then ask for 128 and the
+  // FineAudioBuffer will ask WebRTC for new data only when needed and also
+  // cache non-utilized audio between callbacks. On the recording side, iOS
+  // can provide audio data frames of size 128 and these are accumulated until
+  // enough data to supply one 10ms call exists. This 10ms chunk is then sent
+  // to WebRTC and the remaining part is stored.
+  std::unique_ptr<FineAudioBuffer> fine_audio_buffer_;
+
+  // Temporary storage for recorded data. AudioUnitRender() renders into this
+  // array as soon as a frame of the desired buffer size has been recorded.
+  // On real iOS devices, the size will be fixed and set once. For iOS
+  // simulators, the size can vary from callback to callback and the size
+  // will be changed dynamically to account for this behavior.
+  rtc::BufferT<int16_t> record_audio_buffer_;
+
+  // Set to 1 when recording is active and 0 otherwise.
+  std::atomic<int> recording_;
+
+  // Set to 1 when playout is active and 0 otherwise.
+  std::atomic<int> playing_;
+
+  // Set to true after successful call to Init(), false otherwise.
+  bool initialized_ RTC_GUARDED_BY(thread_);
+
+  // Set to true after successful call to InitRecording() or InitPlayout(),
+  // false otherwise.
+  bool audio_is_initialized_;
+
+  // Set to true if audio session is interrupted, false otherwise.
+  bool is_interrupted_;
+
+  // Audio interruption observer instance.
+  RTCNativeAudioSessionDelegateAdapter* audio_session_observer_
+      RTC_GUARDED_BY(thread_);
+
+  // Set to true if we've activated the audio session.
+  bool has_configured_session_ RTC_GUARDED_BY(thread_);
+
+  // Counts number of detected audio glitches on the playout side.
+  int64_t num_detected_playout_glitches_ RTC_GUARDED_BY(thread_);
+  int64_t last_playout_time_ RTC_GUARDED_BY(io_thread_checker_);
+
+  // Counts number of playout callbacks per call.
+  // The value is updated on the native I/O thread and later read on the
+  // creating `thread_` but at this stage no audio is active.
+  // Hence, it is a "thread safe" design and no lock is needed.
+  int64_t num_playout_callbacks_;
+
+  // Contains the time for when the last output volume change was detected.
+  int64_t last_output_volume_change_time_ RTC_GUARDED_BY(thread_);
+
+  // Avoids running pending task after `this` is Terminated.
+  rtc::scoped_refptr<PendingTaskSafetyFlag> safety_ =
+      PendingTaskSafetyFlag::Create();
+};
+}  // namespace ios_adm
+}  // namespace webrtc
+
+#endif  // SDK_OBJC_NATIVE_SRC_AUDIO_AUDIO_DEVICE_IOS_H_
diff --git a/third_party/libwebrtc/sdk/objc/native/src/audio/audio_device_ios.mm b/third_party/libwebrtc/sdk/objc/native/src/audio/audio_device_ios.mm
new file mode 100644
index 0000000000..dd2c11bdd2
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/native/src/audio/audio_device_ios.mm
@@ -0,0 +1,1127 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <AVFoundation/AVFoundation.h>
+#import <Foundation/Foundation.h>
+
+#include "audio_device_ios.h"
+
+#include <cmath>
+
+#include "api/array_view.h"
+#include "api/task_queue/pending_task_safety_flag.h"
+#include "helpers.h"
+#include "modules/audio_device/fine_audio_buffer.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/thread.h"
+#include "rtc_base/thread_annotations.h"
+#include "rtc_base/time_utils.h"
+#include "system_wrappers/include/field_trial.h"
+#include "system_wrappers/include/metrics.h"
+
+#import "base/RTCLogging.h"
+#import "components/audio/RTCAudioSession+Private.h"
+#import "components/audio/RTCAudioSession.h"
+#import "components/audio/RTCAudioSessionConfiguration.h"
+#import "components/audio/RTCNativeAudioSessionDelegateAdapter.h"
+
+namespace webrtc {
+namespace ios_adm {
+
+#define LOGI() RTC_LOG(LS_INFO) << "AudioDeviceIOS::"
+
+#define LOG_AND_RETURN_IF_ERROR(error, message)    \
+  do {                                             \
+    OSStatus err = error;                          \
+    if (err) {                                     \
+      RTC_LOG(LS_ERROR) << message << ": " << err; \
+      return false;                                \
+    }                                              \
+  } while (0)
+
+#define LOG_IF_ERROR(error, message)               \
+  do {                                             \
+    OSStatus err = error;                          \
+    if (err) {                                     \
+      RTC_LOG(LS_ERROR) << message << ": " << err; \
+    }                                              \
+  } while (0)
+
+// Hardcoded delay estimates based on real measurements.
+// TODO(henrika): these value is not used in combination with built-in AEC.
+// Can most likely be removed.
+const UInt16 kFixedPlayoutDelayEstimate = 30;
+const UInt16 kFixedRecordDelayEstimate = 30;
+
+using ios::CheckAndLogError;
+
+#if !defined(NDEBUG)
+// Returns true when the code runs on a device simulator.
+static bool DeviceIsSimulator() {
+  return ios::GetDeviceName() == "x86_64";
+}
+
+// Helper method that logs essential device information strings.
+static void LogDeviceInfo() {
+  RTC_LOG(LS_INFO) << "LogDeviceInfo";
+  @autoreleasepool {
+    RTC_LOG(LS_INFO) << " system name: " << ios::GetSystemName();
+    RTC_LOG(LS_INFO) << " system version: " << ios::GetSystemVersionAsString();
+    RTC_LOG(LS_INFO) << " device type: " << ios::GetDeviceType();
+    RTC_LOG(LS_INFO) << " device name: " << ios::GetDeviceName();
+    RTC_LOG(LS_INFO) << " process name: " << ios::GetProcessName();
+    RTC_LOG(LS_INFO) << " process ID: " << ios::GetProcessID();
+    RTC_LOG(LS_INFO) << " OS version: " << ios::GetOSVersionString();
+    RTC_LOG(LS_INFO) << " processing cores: " << ios::GetProcessorCount();
+    RTC_LOG(LS_INFO) << " low power mode: " << ios::GetLowPowerModeEnabled();
+#if TARGET_IPHONE_SIMULATOR
+    RTC_LOG(LS_INFO) << " TARGET_IPHONE_SIMULATOR is defined";
+#endif
+    RTC_LOG(LS_INFO) << " DeviceIsSimulator: " << DeviceIsSimulator();
+  }
+}
+#endif  // !defined(NDEBUG)
+
+AudioDeviceIOS::AudioDeviceIOS(bool bypass_voice_processing)
+    : bypass_voice_processing_(bypass_voice_processing),
+      audio_device_buffer_(nullptr),
+      audio_unit_(nullptr),
+      recording_(0),
+      playing_(0),
+      initialized_(false),
+      audio_is_initialized_(false),
+      is_interrupted_(false),
+      has_configured_session_(false),
+      num_detected_playout_glitches_(0),
+      last_playout_time_(0),
+      num_playout_callbacks_(0),
+      last_output_volume_change_time_(0) {
+  LOGI() << "ctor" << ios::GetCurrentThreadDescription()
+         << ",bypass_voice_processing=" << bypass_voice_processing_;
+  io_thread_checker_.Detach();
+  thread_ = rtc::Thread::Current();
+
+  audio_session_observer_ = [[RTCNativeAudioSessionDelegateAdapter alloc] initWithObserver:this];
+}
+
+AudioDeviceIOS::~AudioDeviceIOS() {
+  RTC_DCHECK_RUN_ON(thread_);
+  LOGI() << "~dtor" << ios::GetCurrentThreadDescription();
+  safety_->SetNotAlive();
+  Terminate();
+  audio_session_observer_ = nil;
+}
+
+void AudioDeviceIOS::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) {
+  LOGI() << "AttachAudioBuffer";
+  RTC_DCHECK(audioBuffer);
+  RTC_DCHECK_RUN_ON(thread_);
+  audio_device_buffer_ = audioBuffer;
+}
+
+AudioDeviceGeneric::InitStatus AudioDeviceIOS::Init() {
+  LOGI() << "Init";
+  io_thread_checker_.Detach();
+
+  RTC_DCHECK_RUN_ON(thread_);
+  if (initialized_) {
+    return InitStatus::OK;
+  }
+#if !defined(NDEBUG)
+  LogDeviceInfo();
+#endif
+  // Store the preferred sample rate and preferred number of channels already
+  // here. They have not been set and confirmed yet since configureForWebRTC
+  // is not called until audio is about to start. However, it makes sense to
+  // store the parameters now and then verify at a later stage.
+  RTC_OBJC_TYPE(RTCAudioSessionConfiguration)* config =
+      [RTC_OBJC_TYPE(RTCAudioSessionConfiguration) webRTCConfiguration];
+  playout_parameters_.reset(config.sampleRate, config.outputNumberOfChannels);
+  record_parameters_.reset(config.sampleRate, config.inputNumberOfChannels);
+  // Ensure that the audio device buffer (ADB) knows about the internal audio
+  // parameters. Note that, even if we are unable to get a mono audio session,
+  // we will always tell the I/O audio unit to do a channel format conversion
+  // to guarantee mono on the "input side" of the audio unit.
+  UpdateAudioDeviceBuffer();
+  initialized_ = true;
+  return InitStatus::OK;
+}
+
+int32_t AudioDeviceIOS::Terminate() {
+  LOGI() << "Terminate";
+  RTC_DCHECK_RUN_ON(thread_);
+  if (!initialized_) {
+    return 0;
+  }
+  StopPlayout();
+  StopRecording();
+  initialized_ = false;
+  return 0;
+}
+
+bool AudioDeviceIOS::Initialized() const {
+  RTC_DCHECK_RUN_ON(thread_);
+  return initialized_;
+}
+
+int32_t AudioDeviceIOS::InitPlayout() {
+  LOGI() << "InitPlayout";
+  RTC_DCHECK_RUN_ON(thread_);
+  RTC_DCHECK(initialized_);
+  RTC_DCHECK(!audio_is_initialized_);
+  RTC_DCHECK(!playing_.load());
+  if (!audio_is_initialized_) {
+    if (!InitPlayOrRecord()) {
+      RTC_LOG_F(LS_ERROR) << "InitPlayOrRecord failed for InitPlayout!";
+      return -1;
+    }
+  }
+  audio_is_initialized_ = true;
+  return 0;
+}
+
+bool AudioDeviceIOS::PlayoutIsInitialized() const {
+  RTC_DCHECK_RUN_ON(thread_);
+  return audio_is_initialized_;
+}
+
+bool AudioDeviceIOS::RecordingIsInitialized() const {
+  RTC_DCHECK_RUN_ON(thread_);
+  return audio_is_initialized_;
+}
+
+int32_t AudioDeviceIOS::InitRecording() {
+  LOGI() << "InitRecording";
+  RTC_DCHECK_RUN_ON(thread_);
+  RTC_DCHECK(initialized_);
+  RTC_DCHECK(!audio_is_initialized_);
+  RTC_DCHECK(!recording_.load());
+  if (!audio_is_initialized_) {
+    if (!InitPlayOrRecord()) {
+      RTC_LOG_F(LS_ERROR) << "InitPlayOrRecord failed for InitRecording!";
+      return -1;
+    }
+  }
+  audio_is_initialized_ = true;
+  return 0;
+}
+
+int32_t AudioDeviceIOS::StartPlayout() {
+  LOGI() << "StartPlayout";
+  RTC_DCHECK_RUN_ON(thread_);
+  RTC_DCHECK(audio_is_initialized_);
+  RTC_DCHECK(!playing_.load());
+  RTC_DCHECK(audio_unit_);
+  if (fine_audio_buffer_) {
+    fine_audio_buffer_->ResetPlayout();
+  }
+  if (!recording_.load() && audio_unit_->GetState() == VoiceProcessingAudioUnit::kInitialized) {
+    OSStatus result = audio_unit_->Start();
+    if (result != noErr) {
+      RTC_OBJC_TYPE(RTCAudioSession)* session = [RTC_OBJC_TYPE(RTCAudioSession) sharedInstance];
+      [session notifyAudioUnitStartFailedWithError:result];
+      RTCLogError(@"StartPlayout failed to start audio unit, reason %d", result);
+      return -1;
+    }
+    RTC_LOG(LS_INFO) << "Voice-Processing I/O audio unit is now started";
+  }
+  playing_.store(1, std::memory_order_release);
+  num_playout_callbacks_ = 0;
+  num_detected_playout_glitches_ = 0;
+  return 0;
+}
+
+int32_t AudioDeviceIOS::StopPlayout() {
+  LOGI() << "StopPlayout";
+  RTC_DCHECK_RUN_ON(thread_);
+  if (!audio_is_initialized_ || !playing_.load()) {
+    return 0;
+  }
+  if (!recording_.load()) {
+    ShutdownPlayOrRecord();
+    audio_is_initialized_ = false;
+  }
+  playing_.store(0, std::memory_order_release);
+
+  // Derive average number of calls to OnGetPlayoutData() between detected
+  // audio glitches and add the result to a histogram.
+  int average_number_of_playout_callbacks_between_glitches = 100000;
+  RTC_DCHECK_GE(num_playout_callbacks_, num_detected_playout_glitches_);
+  if (num_detected_playout_glitches_ > 0) {
+    average_number_of_playout_callbacks_between_glitches =
+        num_playout_callbacks_ / num_detected_playout_glitches_;
+  }
+  RTC_HISTOGRAM_COUNTS_100000("WebRTC.Audio.AveragePlayoutCallbacksBetweenGlitches",
+                              average_number_of_playout_callbacks_between_glitches);
+  RTCLog(@"Average number of playout callbacks between glitches: %d",
+         average_number_of_playout_callbacks_between_glitches);
+  return 0;
+}
+
+bool AudioDeviceIOS::Playing() const {
+  return playing_.load();
+}
+
+int32_t AudioDeviceIOS::StartRecording() {
+  LOGI() << "StartRecording";
+  RTC_DCHECK_RUN_ON(thread_);
+  RTC_DCHECK(audio_is_initialized_);
+  RTC_DCHECK(!recording_.load());
+  RTC_DCHECK(audio_unit_);
+  if (fine_audio_buffer_) {
+    fine_audio_buffer_->ResetRecord();
+  }
+  if (!playing_.load() && audio_unit_->GetState() == VoiceProcessingAudioUnit::kInitialized) {
+    OSStatus result = audio_unit_->Start();
+    if (result != noErr) {
+      RTC_OBJC_TYPE(RTCAudioSession)* session = [RTC_OBJC_TYPE(RTCAudioSession) sharedInstance];
+      [session notifyAudioUnitStartFailedWithError:result];
+      RTCLogError(@"StartRecording failed to start audio unit, reason %d", result);
+      return -1;
+    }
+    RTC_LOG(LS_INFO) << "Voice-Processing I/O audio unit is now started";
+  }
+  recording_.store(1, std::memory_order_release);
+  return 0;
+}
+
+int32_t AudioDeviceIOS::StopRecording() {
+  LOGI() << "StopRecording";
+  RTC_DCHECK_RUN_ON(thread_);
+  if (!audio_is_initialized_ || !recording_.load()) {
+    return 0;
+  }
+  if (!playing_.load()) {
+    ShutdownPlayOrRecord();
+    audio_is_initialized_ = false;
+  }
+  recording_.store(0, std::memory_order_release);
+  return 0;
+}
+
+bool AudioDeviceIOS::Recording() const {
+  return recording_.load();
+}
+
+int32_t AudioDeviceIOS::PlayoutDelay(uint16_t& delayMS) const {
+  delayMS = kFixedPlayoutDelayEstimate;
+  return 0;
+}
+
+int AudioDeviceIOS::GetPlayoutAudioParameters(AudioParameters* params) const {
+  LOGI() << "GetPlayoutAudioParameters";
+  RTC_DCHECK(playout_parameters_.is_valid());
+  RTC_DCHECK_RUN_ON(thread_);
+  *params = playout_parameters_;
+  return 0;
+}
+
+int AudioDeviceIOS::GetRecordAudioParameters(AudioParameters* params) const {
+  LOGI() << "GetRecordAudioParameters";
+  RTC_DCHECK(record_parameters_.is_valid());
+  RTC_DCHECK_RUN_ON(thread_);
+  *params = record_parameters_;
+  return 0;
+}
+
+void AudioDeviceIOS::OnInterruptionBegin() {
+  RTC_DCHECK(thread_);
+  LOGI() << "OnInterruptionBegin";
+  thread_->PostTask(SafeTask(safety_, [this] { HandleInterruptionBegin(); }));
+}
+
+void AudioDeviceIOS::OnInterruptionEnd() {
+  RTC_DCHECK(thread_);
+  LOGI() << "OnInterruptionEnd";
+  thread_->PostTask(SafeTask(safety_, [this] { HandleInterruptionEnd(); }));
+}
+
+void AudioDeviceIOS::OnValidRouteChange() {
+  RTC_DCHECK(thread_);
+  thread_->PostTask(SafeTask(safety_, [this] { HandleValidRouteChange(); }));
+}
+
+void AudioDeviceIOS::OnCanPlayOrRecordChange(bool can_play_or_record) {
+  RTC_DCHECK(thread_);
+  thread_->PostTask(SafeTask(
+      safety_, [this, can_play_or_record] { HandleCanPlayOrRecordChange(can_play_or_record); }));
+}
+
+void AudioDeviceIOS::OnChangedOutputVolume() {
+  RTC_DCHECK(thread_);
+  thread_->PostTask(SafeTask(safety_, [this] { HandleOutputVolumeChange(); }));
+}
+
+OSStatus AudioDeviceIOS::OnDeliverRecordedData(AudioUnitRenderActionFlags* flags,
+                                               const AudioTimeStamp* time_stamp,
+                                               UInt32 bus_number,
+                                               UInt32 num_frames,
+                                               AudioBufferList* /* io_data */) {
+  RTC_DCHECK_RUN_ON(&io_thread_checker_);
+  OSStatus result = noErr;
+  // Simply return if recording is not enabled.
+  if (!recording_.load(std::memory_order_acquire)) return result;
+
+  // Set the size of our own audio buffer and clear it first to avoid copying
+  // in combination with potential reallocations.
+  // On real iOS devices, the size will only be set once (at first callback).
+  record_audio_buffer_.Clear();
+  record_audio_buffer_.SetSize(num_frames);
+
+  // Allocate AudioBuffers to be used as storage for the received audio.
+  // The AudioBufferList structure works as a placeholder for the
+  // AudioBuffer structure, which holds a pointer to the actual data buffer
+  // in `record_audio_buffer_`. Recorded audio will be rendered into this memory
+  // at each input callback when calling AudioUnitRender().
+  AudioBufferList audio_buffer_list;
+  audio_buffer_list.mNumberBuffers = 1;
+  AudioBuffer* audio_buffer = &audio_buffer_list.mBuffers[0];
+  audio_buffer->mNumberChannels = record_parameters_.channels();
+  audio_buffer->mDataByteSize =
+      record_audio_buffer_.size() * VoiceProcessingAudioUnit::kBytesPerSample;
+  audio_buffer->mData = reinterpret_cast<int8_t*>(record_audio_buffer_.data());
+
+  // Obtain the recorded audio samples by initiating a rendering cycle.
+  // Since it happens on the input bus, the `io_data` parameter is a reference
+  // to the preallocated audio buffer list that the audio unit renders into.
+  // We can make the audio unit provide a buffer instead in io_data, but we
+  // currently just use our own.
+  // TODO(henrika): should error handling be improved?
+  result = audio_unit_->Render(flags, time_stamp, bus_number, num_frames, &audio_buffer_list);
+  if (result != noErr) {
+    RTCLogError(@"Failed to render audio.");
+    return result;
+  }
+
+  // Get a pointer to the recorded audio and send it to the WebRTC ADB.
+  // Use the FineAudioBuffer instance to convert between native buffer size
+  // and the 10ms buffer size used by WebRTC.
+  fine_audio_buffer_->DeliverRecordedData(record_audio_buffer_, kFixedRecordDelayEstimate);
+  return noErr;
+}
+
+OSStatus AudioDeviceIOS::OnGetPlayoutData(AudioUnitRenderActionFlags* flags,
+                                          const AudioTimeStamp* time_stamp,
+                                          UInt32 bus_number,
+                                          UInt32 num_frames,
+                                          AudioBufferList* io_data) {
+  RTC_DCHECK_RUN_ON(&io_thread_checker_);
+  // Verify 16-bit, noninterleaved mono PCM signal format.
+  RTC_DCHECK_EQ(1, io_data->mNumberBuffers);
+  AudioBuffer* audio_buffer = &io_data->mBuffers[0];
+  RTC_DCHECK_EQ(1, audio_buffer->mNumberChannels);
+
+  // Produce silence and give audio unit a hint about it if playout is not
+  // activated.
+  if (!playing_.load(std::memory_order_acquire)) {
+    const size_t size_in_bytes = audio_buffer->mDataByteSize;
+    RTC_CHECK_EQ(size_in_bytes / VoiceProcessingAudioUnit::kBytesPerSample, num_frames);
+    *flags |= kAudioUnitRenderAction_OutputIsSilence;
+    memset(static_cast<int8_t*>(audio_buffer->mData), 0, size_in_bytes);
+    return noErr;
+  }
+
+  // Measure time since last call to OnGetPlayoutData() and see if it is larger
+  // than a well defined threshold which depends on the current IO buffer size.
+  // If so, we have an indication of a glitch in the output audio since the
+  // core audio layer will most likely run dry in this state.
+  ++num_playout_callbacks_;
+  const int64_t now_time = rtc::TimeMillis();
+  if (time_stamp->mSampleTime != num_frames) {
+    const int64_t delta_time = now_time - last_playout_time_;
+    const int glitch_threshold = 1.6 * playout_parameters_.GetBufferSizeInMilliseconds();
+    if (delta_time > glitch_threshold) {
+      RTCLogWarning(@"Possible playout audio glitch detected.\n"
+                     "  Time since last OnGetPlayoutData was %lld ms.\n",
+                    delta_time);
+      // Exclude extreme delta values since they do most likely not correspond
+      // to a real glitch. Instead, the most probable cause is that a headset
+      // has been plugged in or out. There are more direct ways to detect
+      // audio device changes (see HandleValidRouteChange()) but experiments
+      // show that using it leads to more complex implementations.
+      // TODO(henrika): more tests might be needed to come up with an even
+      // better upper limit.
+      if (glitch_threshold < 120 && delta_time > 120) {
+        RTCLog(@"Glitch warning is ignored. Probably caused by device switch.");
+      } else {
+        thread_->PostTask(SafeTask(safety_, [this] { HandlePlayoutGlitchDetected(); }));
+      }
+    }
+  }
+  last_playout_time_ = now_time;
+
+  // Read decoded 16-bit PCM samples from WebRTC (using a size that matches
+  // the native I/O audio unit) and copy the result to the audio buffer in the
+  // `io_data` destination.
+  fine_audio_buffer_->GetPlayoutData(
+      rtc::ArrayView<int16_t>(static_cast<int16_t*>(audio_buffer->mData), num_frames),
+      kFixedPlayoutDelayEstimate);
+  return noErr;
+}
+
+void AudioDeviceIOS::HandleInterruptionBegin() {
+  RTC_DCHECK_RUN_ON(thread_);
+  RTCLog(@"Interruption begin. IsInterrupted changed from %d to 1.", is_interrupted_);
+  if (audio_unit_ && audio_unit_->GetState() == VoiceProcessingAudioUnit::kStarted) {
+    RTCLog(@"Stopping the audio unit due to interruption begin.");
+    if (!audio_unit_->Stop()) {
+      RTCLogError(@"Failed to stop the audio unit for interruption begin.");
+    }
+    PrepareForNewStart();
+  }
+  is_interrupted_ = true;
+}
+
+void AudioDeviceIOS::HandleInterruptionEnd() {
+  RTC_DCHECK_RUN_ON(thread_);
+  RTCLog(@"Interruption ended. IsInterrupted changed from %d to 0. "
+          "Updating audio unit state.",
+         is_interrupted_);
+  is_interrupted_ = false;
+  if (!audio_unit_) return;
+  if (webrtc::field_trial::IsEnabled("WebRTC-Audio-iOS-Holding")) {
+    // Work around an issue where audio does not restart properly after an interruption
+    // by restarting the audio unit when the interruption ends.
+    if (audio_unit_->GetState() == VoiceProcessingAudioUnit::kStarted) {
+      audio_unit_->Stop();
+      PrepareForNewStart();
+    }
+    if (audio_unit_->GetState() == VoiceProcessingAudioUnit::kInitialized) {
+      audio_unit_->Uninitialize();
+    }
+    // Allocate new buffers given the potentially new stream format.
+    SetupAudioBuffersForActiveAudioSession();
+  }
+  UpdateAudioUnit([RTC_OBJC_TYPE(RTCAudioSession) sharedInstance].canPlayOrRecord);
+}
+
+void AudioDeviceIOS::HandleValidRouteChange() {
+  RTC_DCHECK_RUN_ON(thread_);
+  RTC_OBJC_TYPE(RTCAudioSession)* session = [RTC_OBJC_TYPE(RTCAudioSession) sharedInstance];
+  RTCLog(@"%@", session);
+  HandleSampleRateChange();
+}
+
+void AudioDeviceIOS::HandleCanPlayOrRecordChange(bool can_play_or_record) {
+  RTCLog(@"Handling CanPlayOrRecord change to: %d", can_play_or_record);
+  UpdateAudioUnit(can_play_or_record);
+}
+
+void AudioDeviceIOS::HandleSampleRateChange() {
+  RTC_DCHECK_RUN_ON(thread_);
+  RTCLog(@"Handling sample rate change.");
+
+  // Don't do anything if we're interrupted.
+  if (is_interrupted_) {
+    RTCLog(@"Ignoring sample rate change due to interruption.");
+    return;
+  }
+
+  // If we don't have an audio unit yet, or the audio unit is uninitialized,
+  // there is no work to do.
+  if (!audio_unit_ || audio_unit_->GetState() < VoiceProcessingAudioUnit::kInitialized) {
+    return;
+  }
+
+  // The audio unit is already initialized or started.
+  // Check to see if the sample rate or buffer size has changed.
+  RTC_OBJC_TYPE(RTCAudioSession)* session = [RTC_OBJC_TYPE(RTCAudioSession) sharedInstance];
+  const double new_sample_rate = session.sampleRate;
+  const NSTimeInterval session_buffer_duration = session.IOBufferDuration;
+  const size_t new_frames_per_buffer =
+      static_cast<size_t>(new_sample_rate * session_buffer_duration + .5);
+  const double current_sample_rate = playout_parameters_.sample_rate();
+  const size_t current_frames_per_buffer = playout_parameters_.frames_per_buffer();
+  RTCLog(@"Handling playout sample rate change:\n"
+          "  Session sample rate: %f frames_per_buffer: %lu\n"
+          "  ADM sample rate: %f frames_per_buffer: %lu",
+         new_sample_rate,
+         (unsigned long)new_frames_per_buffer,
+         current_sample_rate,
+         (unsigned long)current_frames_per_buffer);
+
+  // Sample rate and buffer size are the same, no work to do.
+  if (std::abs(current_sample_rate - new_sample_rate) <= DBL_EPSILON &&
+      current_frames_per_buffer == new_frames_per_buffer) {
+    RTCLog(@"Ignoring sample rate change since audio parameters are intact.");
+    return;
+  }
+
+  // Extra sanity check to ensure that the new sample rate is valid.
+  if (new_sample_rate <= 0.0) {
+    RTCLogError(@"Sample rate is invalid: %f", new_sample_rate);
+    return;
+  }
+
+  // We need to adjust our format and buffer sizes.
+  // The stream format is about to be changed and it requires that we first
+  // stop and uninitialize the audio unit to deallocate its resources.
+  RTCLog(@"Stopping and uninitializing audio unit to adjust buffers.");
+  bool restart_audio_unit = false;
+  if (audio_unit_->GetState() == VoiceProcessingAudioUnit::kStarted) {
+    audio_unit_->Stop();
+    restart_audio_unit = true;
+    PrepareForNewStart();
+  }
+  if (audio_unit_->GetState() == VoiceProcessingAudioUnit::kInitialized) {
+    audio_unit_->Uninitialize();
+  }
+
+  // Allocate new buffers given the new stream format.
+  SetupAudioBuffersForActiveAudioSession();
+
+  // Initialize the audio unit again with the new sample rate.
+  if (!audio_unit_->Initialize(playout_parameters_.sample_rate())) {
+    RTCLogError(@"Failed to initialize the audio unit with sample rate: %d",
+                playout_parameters_.sample_rate());
+    return;
+  }
+
+  // Restart the audio unit if it was already running.
+  if (restart_audio_unit) {
+    OSStatus result = audio_unit_->Start();
+    if (result != noErr) {
+      RTC_OBJC_TYPE(RTCAudioSession)* session = [RTC_OBJC_TYPE(RTCAudioSession) sharedInstance];
+      [session notifyAudioUnitStartFailedWithError:result];
+      RTCLogError(@"Failed to start audio unit with sample rate: %d, reason %d",
+                  playout_parameters_.sample_rate(),
+                  result);
+      return;
+    }
+  }
+  RTCLog(@"Successfully handled sample rate change.");
+}
+
+void AudioDeviceIOS::HandlePlayoutGlitchDetected() {
+  RTC_DCHECK_RUN_ON(thread_);
+  // Don't update metrics if we're interrupted since a "glitch" is expected
+  // in this state.
+  if (is_interrupted_) {
+    RTCLog(@"Ignoring audio glitch due to interruption.");
+    return;
+  }
+  // Avoid doing glitch detection for two seconds after a volume change
+  // has been detected to reduce the risk of false alarm.
+  if (last_output_volume_change_time_ > 0 &&
+      rtc::TimeSince(last_output_volume_change_time_) < 2000) {
+    RTCLog(@"Ignoring audio glitch due to recent output volume change.");
+    return;
+  }
+  num_detected_playout_glitches_++;
+  RTCLog(@"Number of detected playout glitches: %lld", num_detected_playout_glitches_);
+
+  int64_t glitch_count = num_detected_playout_glitches_;
+  dispatch_async(dispatch_get_main_queue(), ^{
+    RTC_OBJC_TYPE(RTCAudioSession)* session = [RTC_OBJC_TYPE(RTCAudioSession) sharedInstance];
+    [session notifyDidDetectPlayoutGlitch:glitch_count];
+  });
+}
+
+void AudioDeviceIOS::HandleOutputVolumeChange() {
+  RTC_DCHECK_RUN_ON(thread_);
+  RTCLog(@"Output volume change detected.");
+  // Store time of this detection so it can be used to defer detection of
+  // glitches too close in time to this event.
+  last_output_volume_change_time_ = rtc::TimeMillis();
+}
+
+void AudioDeviceIOS::UpdateAudioDeviceBuffer() {
+  LOGI() << "UpdateAudioDevicebuffer";
+  // AttachAudioBuffer() is called at construction by the main class but check
+  // just in case.
+  RTC_DCHECK(audio_device_buffer_) << "AttachAudioBuffer must be called first";
+  RTC_DCHECK_GT(playout_parameters_.sample_rate(), 0);
+  RTC_DCHECK_GT(record_parameters_.sample_rate(), 0);
+  RTC_DCHECK_EQ(playout_parameters_.channels(), 1);
+  RTC_DCHECK_EQ(record_parameters_.channels(), 1);
+  // Inform the audio device buffer (ADB) about the new audio format.
+  audio_device_buffer_->SetPlayoutSampleRate(playout_parameters_.sample_rate());
+  audio_device_buffer_->SetPlayoutChannels(playout_parameters_.channels());
+  audio_device_buffer_->SetRecordingSampleRate(record_parameters_.sample_rate());
+  audio_device_buffer_->SetRecordingChannels(record_parameters_.channels());
+}
+
+void AudioDeviceIOS::SetupAudioBuffersForActiveAudioSession() {
+  LOGI() << "SetupAudioBuffersForActiveAudioSession";
+  // Verify the current values once the audio session has been activated.
+  RTC_OBJC_TYPE(RTCAudioSession)* session = [RTC_OBJC_TYPE(RTCAudioSession) sharedInstance];
+  double sample_rate = session.sampleRate;
+  NSTimeInterval io_buffer_duration = session.IOBufferDuration;
+  RTCLog(@"%@", session);
+
+  // Log a warning message for the case when we are unable to set the preferred
+  // hardware sample rate but continue and use the non-ideal sample rate after
+  // reinitializing the audio parameters. Most BT headsets only support 8kHz or
+  // 16kHz.
+  RTC_OBJC_TYPE(RTCAudioSessionConfiguration)* webRTCConfig =
+      [RTC_OBJC_TYPE(RTCAudioSessionConfiguration) webRTCConfiguration];
+  if (sample_rate != webRTCConfig.sampleRate) {
+    RTC_LOG(LS_WARNING) << "Unable to set the preferred sample rate";
+  }
+
+  // Crash reports indicates that it can happen in rare cases that the reported
+  // sample rate is less than or equal to zero. If that happens and if a valid
+  // sample rate has already been set during initialization, the best guess we
+  // can do is to reuse the current sample rate.
+  if (sample_rate <= DBL_EPSILON && playout_parameters_.sample_rate() > 0) {
+    RTCLogError(@"Reported rate is invalid: %f. "
+                 "Using %d as sample rate instead.",
+                sample_rate, playout_parameters_.sample_rate());
+    sample_rate = playout_parameters_.sample_rate();
+  }
+
+  // At this stage, we also know the exact IO buffer duration and can add
+  // that info to the existing audio parameters where it is converted into
+  // number of audio frames.
+  // Example: IO buffer size = 0.008 seconds <=> 128 audio frames at 16kHz.
+  // Hence, 128 is the size we expect to see in upcoming render callbacks.
+  playout_parameters_.reset(sample_rate, playout_parameters_.channels(), io_buffer_duration);
+  RTC_DCHECK(playout_parameters_.is_complete());
+  record_parameters_.reset(sample_rate, record_parameters_.channels(), io_buffer_duration);
+  RTC_DCHECK(record_parameters_.is_complete());
+  RTC_LOG(LS_INFO) << " frames per I/O buffer: " << playout_parameters_.frames_per_buffer();
+  RTC_LOG(LS_INFO) << " bytes per I/O buffer: " << playout_parameters_.GetBytesPerBuffer();
+  RTC_DCHECK_EQ(playout_parameters_.GetBytesPerBuffer(), record_parameters_.GetBytesPerBuffer());
+
+  // Update the ADB parameters since the sample rate might have changed.
+  UpdateAudioDeviceBuffer();
+
+  // Create a modified audio buffer class which allows us to ask for,
+  // or deliver, any number of samples (and not only multiple of 10ms) to match
+  // the native audio unit buffer size.
+  RTC_DCHECK(audio_device_buffer_);
+  fine_audio_buffer_.reset(new FineAudioBuffer(audio_device_buffer_));
+}
+
+bool AudioDeviceIOS::CreateAudioUnit() {
+  RTC_DCHECK(!audio_unit_);
+
+  audio_unit_.reset(new VoiceProcessingAudioUnit(bypass_voice_processing_, this));
+  if (!audio_unit_->Init()) {
+    audio_unit_.reset();
+    return false;
+  }
+
+  return true;
+}
+
+void AudioDeviceIOS::UpdateAudioUnit(bool can_play_or_record) {
+  RTC_DCHECK_RUN_ON(thread_);
+  RTCLog(@"Updating audio unit state. CanPlayOrRecord=%d IsInterrupted=%d",
+         can_play_or_record,
+         is_interrupted_);
+
+  if (is_interrupted_) {
+    RTCLog(@"Ignoring audio unit update due to interruption.");
+    return;
+  }
+
+  // If we're not initialized we don't need to do anything. Audio unit will
+  // be initialized on initialization.
+  if (!audio_is_initialized_) return;
+
+  // If we're initialized, we must have an audio unit.
+  RTC_DCHECK(audio_unit_);
+
+  bool should_initialize_audio_unit = false;
+  bool should_uninitialize_audio_unit = false;
+  bool should_start_audio_unit = false;
+  bool should_stop_audio_unit = false;
+
+  switch (audio_unit_->GetState()) {
+    case VoiceProcessingAudioUnit::kInitRequired:
+      RTCLog(@"VPAU state: InitRequired");
+      RTC_DCHECK_NOTREACHED();
+      break;
+    case VoiceProcessingAudioUnit::kUninitialized:
+      RTCLog(@"VPAU state: Uninitialized");
+      should_initialize_audio_unit = can_play_or_record;
+      should_start_audio_unit =
+          should_initialize_audio_unit && (playing_.load() || recording_.load());
+      break;
+    case VoiceProcessingAudioUnit::kInitialized:
+      RTCLog(@"VPAU state: Initialized");
+      should_start_audio_unit = can_play_or_record && (playing_.load() || recording_.load());
+      should_uninitialize_audio_unit = !can_play_or_record;
+      break;
+    case VoiceProcessingAudioUnit::kStarted:
+      RTCLog(@"VPAU state: Started");
+      RTC_DCHECK(playing_.load() || recording_.load());
+      should_stop_audio_unit = !can_play_or_record;
+      should_uninitialize_audio_unit = should_stop_audio_unit;
+      break;
+  }
+
+  if (should_initialize_audio_unit) {
+    RTCLog(@"Initializing audio unit for UpdateAudioUnit");
+    ConfigureAudioSession();
+    SetupAudioBuffersForActiveAudioSession();
+    if (!audio_unit_->Initialize(playout_parameters_.sample_rate())) {
+      RTCLogError(@"Failed to initialize audio unit.");
+      return;
+    }
+  }
+
+  if (should_start_audio_unit) {
+    RTCLog(@"Starting audio unit for UpdateAudioUnit");
+    // Log session settings before trying to start audio streaming.
+    RTC_OBJC_TYPE(RTCAudioSession)* session = [RTC_OBJC_TYPE(RTCAudioSession) sharedInstance];
+    RTCLog(@"%@", session);
+    OSStatus result = audio_unit_->Start();
+    if (result != noErr) {
+      [session notifyAudioUnitStartFailedWithError:result];
+      RTCLogError(@"Failed to start audio unit, reason %d", result);
+      return;
+    }
+  }
+
+  if (should_stop_audio_unit) {
+    RTCLog(@"Stopping audio unit for UpdateAudioUnit");
+    if (!audio_unit_->Stop()) {
+      RTCLogError(@"Failed to stop audio unit.");
+      PrepareForNewStart();
+      return;
+    }
+    PrepareForNewStart();
+  }
+
+  if (should_uninitialize_audio_unit) {
+    RTCLog(@"Uninitializing audio unit for UpdateAudioUnit");
+    audio_unit_->Uninitialize();
+    UnconfigureAudioSession();
+  }
+}
+
+bool AudioDeviceIOS::ConfigureAudioSession() {
+  RTC_DCHECK_RUN_ON(thread_);
+  RTCLog(@"Configuring audio session.");
+  if (has_configured_session_) {
+    RTCLogWarning(@"Audio session already configured.");
+    return false;
+  }
+  RTC_OBJC_TYPE(RTCAudioSession)* session = [RTC_OBJC_TYPE(RTCAudioSession) sharedInstance];
+  [session lockForConfiguration];
+  bool success = [session configureWebRTCSession:nil];
+  [session unlockForConfiguration];
+  if (success) {
+    has_configured_session_ = true;
+    RTCLog(@"Configured audio session.");
+  } else {
+    RTCLog(@"Failed to configure audio session.");
+  }
+  return success;
+}
+
+bool AudioDeviceIOS::ConfigureAudioSessionLocked() {
+  RTC_DCHECK_RUN_ON(thread_);
+  RTCLog(@"Configuring audio session.");
+  if (has_configured_session_) {
+    RTCLogWarning(@"Audio session already configured.");
+    return false;
+  }
+  RTC_OBJC_TYPE(RTCAudioSession)* session = [RTC_OBJC_TYPE(RTCAudioSession) sharedInstance];
+  bool success = [session configureWebRTCSession:nil];
+  if (success) {
+    has_configured_session_ = true;
+    RTCLog(@"Configured audio session.");
+  } else {
+    RTCLog(@"Failed to configure audio session.");
+  }
+  return success;
+}
+
+void AudioDeviceIOS::UnconfigureAudioSession() {
+  RTC_DCHECK_RUN_ON(thread_);
+  RTCLog(@"Unconfiguring audio session.");
+  if (!has_configured_session_) {
+    RTCLogWarning(@"Audio session already unconfigured.");
+    return;
+  }
+  RTC_OBJC_TYPE(RTCAudioSession)* session = [RTC_OBJC_TYPE(RTCAudioSession) sharedInstance];
+  [session lockForConfiguration];
+  [session unconfigureWebRTCSession:nil];
+  [session endWebRTCSession:nil];
+  [session unlockForConfiguration];
+  has_configured_session_ = false;
+  RTCLog(@"Unconfigured audio session.");
+}
+
+bool AudioDeviceIOS::InitPlayOrRecord() {
+  LOGI() << "InitPlayOrRecord";
+  RTC_DCHECK_RUN_ON(thread_);
+
+  // There should be no audio unit at this point.
+  if (!CreateAudioUnit()) {
+    return false;
+  }
+
+  RTC_OBJC_TYPE(RTCAudioSession)* session = [RTC_OBJC_TYPE(RTCAudioSession) sharedInstance];
+  // Subscribe to audio session events.
+  [session pushDelegate:audio_session_observer_];
+  is_interrupted_ = session.isInterrupted ? true : false;
+
+  // Lock the session to make configuration changes.
+  [session lockForConfiguration];
+  NSError* error = nil;
+  if (![session beginWebRTCSession:&error]) {
+    [session unlockForConfiguration];
+    RTCLogError(@"Failed to begin WebRTC session: %@", error.localizedDescription);
+    audio_unit_.reset();
+    return false;
+  }
+
+  // If we are ready to play or record, and if the audio session can be
+  // configured, then initialize the audio unit.
+  if (session.canPlayOrRecord) {
+    if (!ConfigureAudioSessionLocked()) {
+      // One possible reason for failure is if an attempt was made to use the
+      // audio session during or after a Media Services failure.
+      // See AVAudioSessionErrorCodeMediaServicesFailed for details.
+      [session unlockForConfiguration];
+      audio_unit_.reset();
+      return false;
+    }
+    SetupAudioBuffersForActiveAudioSession();
+    audio_unit_->Initialize(playout_parameters_.sample_rate());
+  }
+
+  // Release the lock.
+  [session unlockForConfiguration];
+  return true;
+}
+
+void AudioDeviceIOS::ShutdownPlayOrRecord() {
+  LOGI() << "ShutdownPlayOrRecord";
+  RTC_DCHECK_RUN_ON(thread_);
+
+  // Stop the audio unit to prevent any additional audio callbacks.
+  audio_unit_->Stop();
+
+  // Close and delete the voice-processing I/O unit.
+  audio_unit_.reset();
+
+  // Detach thread checker for the AURemoteIO::IOThread to ensure that the
+  // next session uses a fresh thread id.
+  io_thread_checker_.Detach();
+
+  // Remove audio session notification observers.
+  RTC_OBJC_TYPE(RTCAudioSession)* session = [RTC_OBJC_TYPE(RTCAudioSession) sharedInstance];
+  [session removeDelegate:audio_session_observer_];
+
+  // All I/O should be stopped or paused prior to deactivating the audio
+  // session, hence we deactivate as last action.
+  UnconfigureAudioSession();
+}
+
+void AudioDeviceIOS::PrepareForNewStart() {
+  LOGI() << "PrepareForNewStart";
+  // The audio unit has been stopped and preparations are needed for an upcoming
+  // restart. It will result in audio callbacks from a new native I/O thread
+  // which means that we must detach thread checkers here to be prepared for an
+  // upcoming new audio stream.
+  io_thread_checker_.Detach();
+}
+
+bool AudioDeviceIOS::IsInterrupted() {
+  return is_interrupted_;
+}
+
+#pragma mark - Not Implemented
+
+int32_t AudioDeviceIOS::ActiveAudioLayer(AudioDeviceModule::AudioLayer& audioLayer) const {
+  audioLayer = AudioDeviceModule::kPlatformDefaultAudio;
+  return 0;
+}
+
+int16_t AudioDeviceIOS::PlayoutDevices() {
+  // TODO(henrika): improve.
+  RTC_LOG_F(LS_WARNING) << "Not implemented";
+  return (int16_t)1;
+}
+
+int16_t AudioDeviceIOS::RecordingDevices() {
+  // TODO(henrika): improve.
+  RTC_LOG_F(LS_WARNING) << "Not implemented";
+  return (int16_t)1;
+}
+
+int32_t AudioDeviceIOS::InitSpeaker() {
+  return 0;
+}
+
+bool AudioDeviceIOS::SpeakerIsInitialized() const {
+  return true;
+}
+
+int32_t AudioDeviceIOS::SpeakerVolumeIsAvailable(bool& available) {
+  available = false;
+  return 0;
+}
+
+int32_t AudioDeviceIOS::SetSpeakerVolume(uint32_t volume) {
+  RTC_DCHECK_NOTREACHED() << "Not implemented";
+  return -1;
+}
+
+int32_t AudioDeviceIOS::SpeakerVolume(uint32_t& volume) const {
+  RTC_DCHECK_NOTREACHED() << "Not implemented";
+  return -1;
+}
+
+int32_t AudioDeviceIOS::MaxSpeakerVolume(uint32_t& maxVolume) const {
+  RTC_DCHECK_NOTREACHED() << "Not implemented";
+  return -1;
+}
+
+int32_t AudioDeviceIOS::MinSpeakerVolume(uint32_t& minVolume) const {
+  RTC_DCHECK_NOTREACHED() << "Not implemented";
+  return -1;
+}
+
+int32_t AudioDeviceIOS::SpeakerMuteIsAvailable(bool& available) {
+  available = false;
+  return 0;
+}
+
+int32_t AudioDeviceIOS::SetSpeakerMute(bool enable) {
+  RTC_DCHECK_NOTREACHED() << "Not implemented";
+  return -1;
+}
+
+int32_t AudioDeviceIOS::SpeakerMute(bool& enabled) const {
+  RTC_DCHECK_NOTREACHED() << "Not implemented";
+  return -1;
+}
+
+int32_t AudioDeviceIOS::SetPlayoutDevice(uint16_t index) {
+  RTC_LOG_F(LS_WARNING) << "Not implemented";
+  return 0;
+}
+
+int32_t AudioDeviceIOS::SetPlayoutDevice(AudioDeviceModule::WindowsDeviceType) {
+  RTC_DCHECK_NOTREACHED() << "Not implemented";
+  return -1;
+}
+
+int32_t AudioDeviceIOS::InitMicrophone() {
+  return 0;
+}
+
+bool AudioDeviceIOS::MicrophoneIsInitialized() const {
+  return true;
+}
+
+int32_t AudioDeviceIOS::MicrophoneMuteIsAvailable(bool& available) {
+  available = false;
+  return 0;
+}
+
+int32_t AudioDeviceIOS::SetMicrophoneMute(bool enable) {
+  RTC_DCHECK_NOTREACHED() << "Not implemented";
+  return -1;
+}
+
+int32_t AudioDeviceIOS::MicrophoneMute(bool& enabled) const {
+  RTC_DCHECK_NOTREACHED() << "Not implemented";
+  return -1;
+}
+
+int32_t AudioDeviceIOS::StereoRecordingIsAvailable(bool& available) {
+  available = false;
+  return 0;
+}
+
+int32_t AudioDeviceIOS::SetStereoRecording(bool enable) {
+  RTC_LOG_F(LS_WARNING) << "Not implemented";
+  return -1;
+}
+
+int32_t AudioDeviceIOS::StereoRecording(bool& enabled) const {
+  enabled = false;
+  return 0;
+}
+
+int32_t AudioDeviceIOS::StereoPlayoutIsAvailable(bool& available) {
+  available = false;
+  return 0;
+}
+
+int32_t AudioDeviceIOS::SetStereoPlayout(bool enable) {
+  RTC_LOG_F(LS_WARNING) << "Not implemented";
+  return -1;
+}
+
+int32_t AudioDeviceIOS::StereoPlayout(bool& enabled) const {
+  enabled = false;
+  return 0;
+}
+
+int32_t AudioDeviceIOS::MicrophoneVolumeIsAvailable(bool& available) {
+  available = false;
+  return 0;
+}
+
+int32_t AudioDeviceIOS::SetMicrophoneVolume(uint32_t volume) {
+  RTC_DCHECK_NOTREACHED() << "Not implemented";
+  return -1;
+}
+
+int32_t AudioDeviceIOS::MicrophoneVolume(uint32_t& volume) const {
+  RTC_DCHECK_NOTREACHED() << "Not implemented";
+  return -1;
+}
+
+int32_t AudioDeviceIOS::MaxMicrophoneVolume(uint32_t& maxVolume) const {
+  RTC_DCHECK_NOTREACHED() << "Not implemented";
+  return -1;
+}
+
+int32_t AudioDeviceIOS::MinMicrophoneVolume(uint32_t& minVolume) const {
+  RTC_DCHECK_NOTREACHED() << "Not implemented";
+  return -1;
+}
+
+int32_t AudioDeviceIOS::PlayoutDeviceName(uint16_t index,
+                                          char name[kAdmMaxDeviceNameSize],
+                                          char guid[kAdmMaxGuidSize]) {
+  RTC_DCHECK_NOTREACHED() << "Not implemented";
+  return -1;
+}
+
+int32_t AudioDeviceIOS::RecordingDeviceName(uint16_t index,
+                                            char name[kAdmMaxDeviceNameSize],
+                                            char guid[kAdmMaxGuidSize]) {
+  RTC_DCHECK_NOTREACHED() << "Not implemented";
+  return -1;
+}
+
+int32_t AudioDeviceIOS::SetRecordingDevice(uint16_t index) {
+  RTC_LOG_F(LS_WARNING) << "Not implemented";
+  return 0;
+}
+
+int32_t AudioDeviceIOS::SetRecordingDevice(AudioDeviceModule::WindowsDeviceType) {
+  RTC_DCHECK_NOTREACHED() << "Not implemented";
+  return -1;
+}
+
+int32_t AudioDeviceIOS::PlayoutIsAvailable(bool& available) {
+  available = true;
+  return 0;
+}
+
+int32_t AudioDeviceIOS::RecordingIsAvailable(bool& available) {
+  available = true;
+  return 0;
+}
+
+}  // namespace ios_adm
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/sdk/objc/native/src/audio/audio_device_module_ios.h b/third_party/libwebrtc/sdk/objc/native/src/audio/audio_device_module_ios.h
new file mode 100644
index 0000000000..9bcf114e32
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/native/src/audio/audio_device_module_ios.h
@@ -0,0 +1,143 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef SDK_OBJC_NATIVE_SRC_AUDIO_AUDIO_DEVICE_MODULE_IOS_H_
+#define SDK_OBJC_NATIVE_SRC_AUDIO_AUDIO_DEVICE_MODULE_IOS_H_
+
+#include <memory>
+
+#include "audio_device_ios.h"
+
+#include "api/task_queue/task_queue_factory.h"
+#include "modules/audio_device/audio_device_buffer.h"
+#include "modules/audio_device/include/audio_device.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+class AudioDeviceGeneric;
+
+namespace ios_adm {
+
+class AudioDeviceModuleIOS : public AudioDeviceModule {
+ public:
+  int32_t AttachAudioBuffer();
+
+  explicit AudioDeviceModuleIOS(bool bypass_voice_processing);
+  ~AudioDeviceModuleIOS() override;
+
+  // Retrieve the currently utilized audio layer
+  int32_t ActiveAudioLayer(AudioLayer* audioLayer) const override;
+
+  // Full-duplex transportation of PCM audio
+  int32_t RegisterAudioCallback(AudioTransport* audioCallback) override;
+
+  // Main initializaton and termination
+  int32_t Init() override;
+  int32_t Terminate() override;
+  bool Initialized() const override;
+
+  // Device enumeration
+  int16_t PlayoutDevices() override;
+  int16_t RecordingDevices() override;
+  int32_t PlayoutDeviceName(uint16_t index,
+                            char name[kAdmMaxDeviceNameSize],
+                            char guid[kAdmMaxGuidSize]) override;
+  int32_t RecordingDeviceName(uint16_t index,
+                              char name[kAdmMaxDeviceNameSize],
+                              char guid[kAdmMaxGuidSize]) override;
+
+  // Device selection
+  int32_t SetPlayoutDevice(uint16_t index) override;
+  int32_t SetPlayoutDevice(WindowsDeviceType device) override;
+  int32_t SetRecordingDevice(uint16_t index) override;
+  int32_t SetRecordingDevice(WindowsDeviceType device) override;
+
+  // Audio transport initialization
+  int32_t PlayoutIsAvailable(bool* available) override;
+  int32_t InitPlayout() override;
+  bool PlayoutIsInitialized() const override;
+  int32_t RecordingIsAvailable(bool* available) override;
+  int32_t InitRecording() override;
+  bool RecordingIsInitialized() const override;
+
+  // Audio transport control
+  int32_t StartPlayout() override;
+  int32_t StopPlayout() override;
+  bool Playing() const override;
+  int32_t StartRecording() override;
+  int32_t StopRecording() override;
+  bool Recording() const override;
+
+  // Audio mixer initialization
+  int32_t InitSpeaker() override;
+  bool SpeakerIsInitialized() const override;
+  int32_t InitMicrophone() override;
+  bool MicrophoneIsInitialized() const override;
+
+  // Speaker volume controls
+  int32_t SpeakerVolumeIsAvailable(bool* available) override;
+  int32_t SetSpeakerVolume(uint32_t volume) override;
+  int32_t SpeakerVolume(uint32_t* volume) const override;
+  int32_t MaxSpeakerVolume(uint32_t* maxVolume) const override;
+  int32_t MinSpeakerVolume(uint32_t* minVolume) const override;
+
+  // Microphone volume controls
+  int32_t MicrophoneVolumeIsAvailable(bool* available) override;
+  int32_t SetMicrophoneVolume(uint32_t volume) override;
+  int32_t MicrophoneVolume(uint32_t* volume) const override;
+  int32_t MaxMicrophoneVolume(uint32_t* maxVolume) const override;
+  int32_t MinMicrophoneVolume(uint32_t* minVolume) const override;
+
+  // Speaker mute control
+  int32_t SpeakerMuteIsAvailable(bool* available) override;
+  int32_t SetSpeakerMute(bool enable) override;
+  int32_t SpeakerMute(bool* enabled) const override;
+
+  // Microphone mute control
+  int32_t MicrophoneMuteIsAvailable(bool* available) override;
+  int32_t SetMicrophoneMute(bool enable) override;
+  int32_t MicrophoneMute(bool* enabled) const override;
+
+  // Stereo support
+  int32_t StereoPlayoutIsAvailable(bool* available) const override;
+  int32_t SetStereoPlayout(bool enable) override;
+  int32_t StereoPlayout(bool* enabled) const override;
+  int32_t StereoRecordingIsAvailable(bool* available) const override;
+  int32_t SetStereoRecording(bool enable) override;
+  int32_t StereoRecording(bool* enabled) const override;
+
+  // Delay information and control
+  int32_t PlayoutDelay(uint16_t* delayMS) const override;
+
+  bool BuiltInAECIsAvailable() const override;
+  int32_t EnableBuiltInAEC(bool enable) override;
+  bool BuiltInAGCIsAvailable() const override;
+  int32_t EnableBuiltInAGC(bool enable) override;
+  bool BuiltInNSIsAvailable() const override;
+  int32_t EnableBuiltInNS(bool enable) override;
+
+  int32_t GetPlayoutUnderrunCount() const override;
+
+#if defined(WEBRTC_IOS)
+  int GetPlayoutAudioParameters(AudioParameters* params) const override;
+  int GetRecordAudioParameters(AudioParameters* params) const override;
+#endif  // WEBRTC_IOS
+ private:
+  const bool bypass_voice_processing_;
+  bool initialized_ = false;
+  const std::unique_ptr<TaskQueueFactory> task_queue_factory_;
+  std::unique_ptr<AudioDeviceIOS> audio_device_;
+  std::unique_ptr<AudioDeviceBuffer> audio_device_buffer_;
+};
+}  // namespace ios_adm
+}  // namespace webrtc
+
+#endif  // SDK_OBJC_NATIVE_SRC_AUDIO_AUDIO_DEVICE_MODULE_IOS_H_
diff --git a/third_party/libwebrtc/sdk/objc/native/src/audio/audio_device_module_ios.mm b/third_party/libwebrtc/sdk/objc/native/src/audio/audio_device_module_ios.mm
new file mode 100644
index 0000000000..5effef3abd
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/native/src/audio/audio_device_module_ios.mm
@@ -0,0 +1,669 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "audio_device_module_ios.h"
+
+#include "api/task_queue/default_task_queue_factory.h"
+#include "modules/audio_device/audio_device_config.h"
+#include "modules/audio_device/audio_device_generic.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/ref_count.h"
+#include "system_wrappers/include/metrics.h"
+
+#if defined(WEBRTC_IOS)
+#include "audio_device_ios.h"
+#endif
+
+#define CHECKinitialized_() \
+  {                         \
+    if (!initialized_) {    \
+      return -1;            \
+    };                      \
+  }
+
+#define CHECKinitialized__BOOL() \
+  {                              \
+    if (!initialized_) {         \
+      return false;              \
+    };                           \
+  }
+
+namespace webrtc {
+namespace ios_adm {
+
+AudioDeviceModuleIOS::AudioDeviceModuleIOS(bool bypass_voice_processing)
+    : bypass_voice_processing_(bypass_voice_processing),
+      task_queue_factory_(CreateDefaultTaskQueueFactory()) {
+  RTC_LOG(LS_INFO) << "current platform is IOS";
+  RTC_LOG(LS_INFO) << "iPhone Audio APIs will be utilized.";
+}
+
+  int32_t AudioDeviceModuleIOS::AttachAudioBuffer() {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    audio_device_->AttachAudioBuffer(audio_device_buffer_.get());
+    return 0;
+  }
+
+  AudioDeviceModuleIOS::~AudioDeviceModuleIOS() {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+  }
+
+  int32_t AudioDeviceModuleIOS::ActiveAudioLayer(AudioLayer* audioLayer) const {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    AudioLayer activeAudio;
+    if (audio_device_->ActiveAudioLayer(activeAudio) == -1) {
+      return -1;
+    }
+    *audioLayer = activeAudio;
+    return 0;
+  }
+
+  int32_t AudioDeviceModuleIOS::Init() {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    if (initialized_)
+      return 0;
+
+    audio_device_buffer_.reset(new webrtc::AudioDeviceBuffer(task_queue_factory_.get()));
+    audio_device_.reset(new ios_adm::AudioDeviceIOS(bypass_voice_processing_));
+    RTC_CHECK(audio_device_);
+
+    this->AttachAudioBuffer();
+
+    AudioDeviceGeneric::InitStatus status = audio_device_->Init();
+    RTC_HISTOGRAM_ENUMERATION(
+        "WebRTC.Audio.InitializationResult", static_cast<int>(status),
+        static_cast<int>(AudioDeviceGeneric::InitStatus::NUM_STATUSES));
+    if (status != AudioDeviceGeneric::InitStatus::OK) {
+      RTC_LOG(LS_ERROR) << "Audio device initialization failed.";
+      return -1;
+    }
+    initialized_ = true;
+    return 0;
+  }
+
+  int32_t AudioDeviceModuleIOS::Terminate() {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    if (!initialized_)
+      return 0;
+    if (audio_device_->Terminate() == -1) {
+      return -1;
+    }
+    initialized_ = false;
+    return 0;
+  }
+
+  bool AudioDeviceModuleIOS::Initialized() const {
+    RTC_DLOG(LS_INFO) << __FUNCTION__ << ": " << initialized_;
+    return initialized_;
+  }
+
+  int32_t AudioDeviceModuleIOS::InitSpeaker() {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    CHECKinitialized_();
+    return audio_device_->InitSpeaker();
+  }
+
+  int32_t AudioDeviceModuleIOS::InitMicrophone() {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    CHECKinitialized_();
+    return audio_device_->InitMicrophone();
+  }
+
+  int32_t AudioDeviceModuleIOS::SpeakerVolumeIsAvailable(bool* available) {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    CHECKinitialized_();
+    bool isAvailable = false;
+    if (audio_device_->SpeakerVolumeIsAvailable(isAvailable) == -1) {
+      return -1;
+    }
+    *available = isAvailable;
+    RTC_DLOG(LS_INFO) << "output: " << isAvailable;
+    return 0;
+  }
+
+  int32_t AudioDeviceModuleIOS::SetSpeakerVolume(uint32_t volume) {
+    RTC_DLOG(LS_INFO) << __FUNCTION__ << "(" << volume << ")";
+    CHECKinitialized_();
+    return audio_device_->SetSpeakerVolume(volume);
+  }
+
+  int32_t AudioDeviceModuleIOS::SpeakerVolume(uint32_t* volume) const {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    CHECKinitialized_();
+    uint32_t level = 0;
+    if (audio_device_->SpeakerVolume(level) == -1) {
+      return -1;
+    }
+    *volume = level;
+    RTC_DLOG(LS_INFO) << "output: " << *volume;
+    return 0;
+  }
+
+  bool AudioDeviceModuleIOS::SpeakerIsInitialized() const {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    CHECKinitialized__BOOL();
+    bool isInitialized = audio_device_->SpeakerIsInitialized();
+    RTC_DLOG(LS_INFO) << "output: " << isInitialized;
+    return isInitialized;
+  }
+
+  bool AudioDeviceModuleIOS::MicrophoneIsInitialized() const {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    CHECKinitialized__BOOL();
+    bool isInitialized = audio_device_->MicrophoneIsInitialized();
+    RTC_DLOG(LS_INFO) << "output: " << isInitialized;
+    return isInitialized;
+  }
+
+  int32_t AudioDeviceModuleIOS::MaxSpeakerVolume(uint32_t* maxVolume) const {
+    CHECKinitialized_();
+    uint32_t maxVol = 0;
+    if (audio_device_->MaxSpeakerVolume(maxVol) == -1) {
+      return -1;
+    }
+    *maxVolume = maxVol;
+    return 0;
+  }
+
+  int32_t AudioDeviceModuleIOS::MinSpeakerVolume(uint32_t* minVolume) const {
+    CHECKinitialized_();
+    uint32_t minVol = 0;
+    if (audio_device_->MinSpeakerVolume(minVol) == -1) {
+      return -1;
+    }
+    *minVolume = minVol;
+    return 0;
+  }
+
+  int32_t AudioDeviceModuleIOS::SpeakerMuteIsAvailable(bool* available) {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    CHECKinitialized_();
+    bool isAvailable = false;
+    if (audio_device_->SpeakerMuteIsAvailable(isAvailable) == -1) {
+      return -1;
+    }
+    *available = isAvailable;
+    RTC_DLOG(LS_INFO) << "output: " << isAvailable;
+    return 0;
+  }
+
+  int32_t AudioDeviceModuleIOS::SetSpeakerMute(bool enable) {
+    RTC_DLOG(LS_INFO) << __FUNCTION__ << "(" << enable << ")";
+    CHECKinitialized_();
+    return audio_device_->SetSpeakerMute(enable);
+  }
+
+  int32_t AudioDeviceModuleIOS::SpeakerMute(bool* enabled) const {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    CHECKinitialized_();
+    bool muted = false;
+    if (audio_device_->SpeakerMute(muted) == -1) {
+      return -1;
+    }
+    *enabled = muted;
+    RTC_DLOG(LS_INFO) << "output: " << muted;
+    return 0;
+  }
+
+  int32_t AudioDeviceModuleIOS::MicrophoneMuteIsAvailable(bool* available) {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    CHECKinitialized_();
+    bool isAvailable = false;
+    if (audio_device_->MicrophoneMuteIsAvailable(isAvailable) == -1) {
+      return -1;
+    }
+    *available = isAvailable;
+    RTC_DLOG(LS_INFO) << "output: " << isAvailable;
+    return 0;
+  }
+
+  int32_t AudioDeviceModuleIOS::SetMicrophoneMute(bool enable) {
+    RTC_DLOG(LS_INFO) << __FUNCTION__ << "(" << enable << ")";
+    CHECKinitialized_();
+    return (audio_device_->SetMicrophoneMute(enable));
+  }
+
+  int32_t AudioDeviceModuleIOS::MicrophoneMute(bool* enabled) const {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    CHECKinitialized_();
+    bool muted = false;
+    if (audio_device_->MicrophoneMute(muted) == -1) {
+      return -1;
+    }
+    *enabled = muted;
+    RTC_DLOG(LS_INFO) << "output: " << muted;
+    return 0;
+  }
+
+  int32_t AudioDeviceModuleIOS::MicrophoneVolumeIsAvailable(bool* available) {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    CHECKinitialized_();
+    bool isAvailable = false;
+    if (audio_device_->MicrophoneVolumeIsAvailable(isAvailable) == -1) {
+      return -1;
+    }
+    *available = isAvailable;
+    RTC_DLOG(LS_INFO) << "output: " << isAvailable;
+    return 0;
+  }
+
+  int32_t AudioDeviceModuleIOS::SetMicrophoneVolume(uint32_t volume) {
+    RTC_DLOG(LS_INFO) << __FUNCTION__ << "(" << volume << ")";
+    CHECKinitialized_();
+    return (audio_device_->SetMicrophoneVolume(volume));
+  }
+
+  int32_t AudioDeviceModuleIOS::MicrophoneVolume(uint32_t* volume) const {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    CHECKinitialized_();
+    uint32_t level = 0;
+    if (audio_device_->MicrophoneVolume(level) == -1) {
+      return -1;
+    }
+    *volume = level;
+    RTC_DLOG(LS_INFO) << "output: " << *volume;
+    return 0;
+  }
+
+  int32_t AudioDeviceModuleIOS::StereoRecordingIsAvailable(
+      bool* available) const {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    CHECKinitialized_();
+    bool isAvailable = false;
+    if (audio_device_->StereoRecordingIsAvailable(isAvailable) == -1) {
+      return -1;
+    }
+    *available = isAvailable;
+    RTC_DLOG(LS_INFO) << "output: " << isAvailable;
+    return 0;
+  }
+
+  int32_t AudioDeviceModuleIOS::SetStereoRecording(bool enable) {
+    RTC_DLOG(LS_INFO) << __FUNCTION__ << "(" << enable << ")";
+    CHECKinitialized_();
+    if (enable) {
+      RTC_LOG(LS_WARNING) << "recording in stereo is not supported";
+    }
+    return -1;
+  }
+
+  int32_t AudioDeviceModuleIOS::StereoRecording(bool* enabled) const {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    CHECKinitialized_();
+    bool stereo = false;
+    if (audio_device_->StereoRecording(stereo) == -1) {
+      return -1;
+    }
+    *enabled = stereo;
+    RTC_DLOG(LS_INFO) << "output: " << stereo;
+    return 0;
+  }
+
+  int32_t AudioDeviceModuleIOS::StereoPlayoutIsAvailable(bool* available) const {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    CHECKinitialized_();
+    bool isAvailable = false;
+    if (audio_device_->StereoPlayoutIsAvailable(isAvailable) == -1) {
+      return -1;
+    }
+    *available = isAvailable;
+    RTC_DLOG(LS_INFO) << "output: " << isAvailable;
+    return 0;
+  }
+
+  int32_t AudioDeviceModuleIOS::SetStereoPlayout(bool enable) {
+    RTC_DLOG(LS_INFO) << __FUNCTION__ << "(" << enable << ")";
+    CHECKinitialized_();
+    if (audio_device_->PlayoutIsInitialized()) {
+      RTC_LOG(LS_ERROR) << "unable to set stereo mode while playing side is initialized";
+      return -1;
+    }
+    if (audio_device_->SetStereoPlayout(enable)) {
+      RTC_LOG(LS_WARNING) << "stereo playout is not supported";
+      return -1;
+    }
+    int8_t nChannels(1);
+    if (enable) {
+      nChannels = 2;
+    }
+    audio_device_buffer_.get()->SetPlayoutChannels(nChannels);
+    return 0;
+  }
+
+  int32_t AudioDeviceModuleIOS::StereoPlayout(bool* enabled) const {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    CHECKinitialized_();
+    bool stereo = false;
+    if (audio_device_->StereoPlayout(stereo) == -1) {
+      return -1;
+    }
+    *enabled = stereo;
+    RTC_DLOG(LS_INFO) << "output: " << stereo;
+    return 0;
+  }
+
+  int32_t AudioDeviceModuleIOS::PlayoutIsAvailable(bool* available) {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    CHECKinitialized_();
+    bool isAvailable = false;
+    if (audio_device_->PlayoutIsAvailable(isAvailable) == -1) {
+      return -1;
+    }
+    *available = isAvailable;
+    RTC_DLOG(LS_INFO) << "output: " << isAvailable;
+    return 0;
+  }
+
+  int32_t AudioDeviceModuleIOS::RecordingIsAvailable(bool* available) {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    CHECKinitialized_();
+    bool isAvailable = false;
+    if (audio_device_->RecordingIsAvailable(isAvailable) == -1) {
+      return -1;
+    }
+    *available = isAvailable;
+    RTC_DLOG(LS_INFO) << "output: " << isAvailable;
+    return 0;
+  }
+
+  int32_t AudioDeviceModuleIOS::MaxMicrophoneVolume(uint32_t* maxVolume) const {
+    CHECKinitialized_();
+    uint32_t maxVol(0);
+    if (audio_device_->MaxMicrophoneVolume(maxVol) == -1) {
+      return -1;
+    }
+    *maxVolume = maxVol;
+    return 0;
+  }
+
+  int32_t AudioDeviceModuleIOS::MinMicrophoneVolume(uint32_t* minVolume) const {
+    CHECKinitialized_();
+    uint32_t minVol(0);
+    if (audio_device_->MinMicrophoneVolume(minVol) == -1) {
+      return -1;
+    }
+    *minVolume = minVol;
+    return 0;
+  }
+
+  int16_t AudioDeviceModuleIOS::PlayoutDevices() {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    CHECKinitialized_();
+    uint16_t nPlayoutDevices = audio_device_->PlayoutDevices();
+    RTC_DLOG(LS_INFO) << "output: " << nPlayoutDevices;
+    return (int16_t)(nPlayoutDevices);
+  }
+
+  int32_t AudioDeviceModuleIOS::SetPlayoutDevice(uint16_t index) {
+    RTC_DLOG(LS_INFO) << __FUNCTION__ << "(" << index << ")";
+    CHECKinitialized_();
+    return audio_device_->SetPlayoutDevice(index);
+  }
+
+  int32_t AudioDeviceModuleIOS::SetPlayoutDevice(WindowsDeviceType device) {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    CHECKinitialized_();
+    return audio_device_->SetPlayoutDevice(device);
+  }
+
+  int32_t AudioDeviceModuleIOS::PlayoutDeviceName(
+      uint16_t index,
+      char name[kAdmMaxDeviceNameSize],
+      char guid[kAdmMaxGuidSize]) {
+    RTC_DLOG(LS_INFO) << __FUNCTION__ << "(" << index << ", ...)";
+    CHECKinitialized_();
+    if (name == NULL) {
+      return -1;
+    }
+    if (audio_device_->PlayoutDeviceName(index, name, guid) == -1) {
+      return -1;
+    }
+    if (name != NULL) {
+      RTC_DLOG(LS_INFO) << "output: name = " << name;
+    }
+    if (guid != NULL) {
+      RTC_DLOG(LS_INFO) << "output: guid = " << guid;
+    }
+    return 0;
+  }
+
+  int32_t AudioDeviceModuleIOS::RecordingDeviceName(
+      uint16_t index,
+      char name[kAdmMaxDeviceNameSize],
+      char guid[kAdmMaxGuidSize]) {
+    RTC_DLOG(LS_INFO) << __FUNCTION__ << "(" << index << ", ...)";
+    CHECKinitialized_();
+    if (name == NULL) {
+      return -1;
+    }
+    if (audio_device_->RecordingDeviceName(index, name, guid) == -1) {
+      return -1;
+    }
+    if (name != NULL) {
+      RTC_DLOG(LS_INFO) << "output: name = " << name;
+    }
+    if (guid != NULL) {
+      RTC_DLOG(LS_INFO) << "output: guid = " << guid;
+    }
+    return 0;
+  }
+
+  int16_t AudioDeviceModuleIOS::RecordingDevices() {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    CHECKinitialized_();
+    uint16_t nRecordingDevices = audio_device_->RecordingDevices();
+    RTC_DLOG(LS_INFO) << "output: " << nRecordingDevices;
+    return (int16_t)nRecordingDevices;
+  }
+
+  int32_t AudioDeviceModuleIOS::SetRecordingDevice(uint16_t index) {
+    RTC_DLOG(LS_INFO) << __FUNCTION__ << "(" << index << ")";
+    CHECKinitialized_();
+    return audio_device_->SetRecordingDevice(index);
+  }
+
+  int32_t AudioDeviceModuleIOS::SetRecordingDevice(WindowsDeviceType device) {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    CHECKinitialized_();
+    return audio_device_->SetRecordingDevice(device);
+  }
+
+  int32_t AudioDeviceModuleIOS::InitPlayout() {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    CHECKinitialized_();
+    if (PlayoutIsInitialized()) {
+      return 0;
+    }
+    int32_t result = audio_device_->InitPlayout();
+    RTC_DLOG(LS_INFO) << "output: " << result;
+    RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.InitPlayoutSuccess",
+                          static_cast<int>(result == 0));
+    return result;
+  }
+
+  int32_t AudioDeviceModuleIOS::InitRecording() {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    CHECKinitialized_();
+    if (RecordingIsInitialized()) {
+      return 0;
+    }
+    int32_t result = audio_device_->InitRecording();
+    RTC_DLOG(LS_INFO) << "output: " << result;
+    RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.InitRecordingSuccess",
+                          static_cast<int>(result == 0));
+    return result;
+  }
+
+  bool AudioDeviceModuleIOS::PlayoutIsInitialized() const {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    CHECKinitialized__BOOL();
+    return audio_device_->PlayoutIsInitialized();
+  }
+
+  bool AudioDeviceModuleIOS::RecordingIsInitialized() const {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    CHECKinitialized__BOOL();
+    return audio_device_->RecordingIsInitialized();
+  }
+
+  int32_t AudioDeviceModuleIOS::StartPlayout() {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    CHECKinitialized_();
+    if (Playing()) {
+      return 0;
+    }
+    audio_device_buffer_.get()->StartPlayout();
+    int32_t result = audio_device_->StartPlayout();
+    RTC_DLOG(LS_INFO) << "output: " << result;
+    RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.StartPlayoutSuccess",
+                          static_cast<int>(result == 0));
+    return result;
+  }
+
+  int32_t AudioDeviceModuleIOS::StopPlayout() {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    CHECKinitialized_();
+    int32_t result = audio_device_->StopPlayout();
+    audio_device_buffer_.get()->StopPlayout();
+    RTC_DLOG(LS_INFO) << "output: " << result;
+    RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.StopPlayoutSuccess",
+                          static_cast<int>(result == 0));
+    return result;
+  }
+
+  bool AudioDeviceModuleIOS::Playing() const {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    CHECKinitialized__BOOL();
+    return audio_device_->Playing();
+  }
+
+  int32_t AudioDeviceModuleIOS::StartRecording() {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    CHECKinitialized_();
+    if (Recording()) {
+      return 0;
+    }
+    audio_device_buffer_.get()->StartRecording();
+    int32_t result = audio_device_->StartRecording();
+    RTC_DLOG(LS_INFO) << "output: " << result;
+    RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.StartRecordingSuccess",
+                          static_cast<int>(result == 0));
+    return result;
+  }
+
+  int32_t AudioDeviceModuleIOS::StopRecording() {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    CHECKinitialized_();
+    int32_t result = audio_device_->StopRecording();
+    audio_device_buffer_.get()->StopRecording();
+    RTC_DLOG(LS_INFO) << "output: " << result;
+    RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.StopRecordingSuccess",
+                          static_cast<int>(result == 0));
+    return result;
+  }
+
+  bool AudioDeviceModuleIOS::Recording() const {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    CHECKinitialized__BOOL();
+    return audio_device_->Recording();
+  }
+
+  int32_t AudioDeviceModuleIOS::RegisterAudioCallback(
+      AudioTransport* audioCallback) {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    return audio_device_buffer_.get()->RegisterAudioCallback(audioCallback);
+  }
+
+  int32_t AudioDeviceModuleIOS::PlayoutDelay(uint16_t* delayMS) const {
+    CHECKinitialized_();
+    uint16_t delay = 0;
+    if (audio_device_->PlayoutDelay(delay) == -1) {
+      RTC_LOG(LS_ERROR) << "failed to retrieve the playout delay";
+      return -1;
+    }
+    *delayMS = delay;
+    return 0;
+  }
+
+  bool AudioDeviceModuleIOS::BuiltInAECIsAvailable() const {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    CHECKinitialized__BOOL();
+    bool isAvailable = audio_device_->BuiltInAECIsAvailable();
+    RTC_DLOG(LS_INFO) << "output: " << isAvailable;
+    return isAvailable;
+  }
+
+  int32_t AudioDeviceModuleIOS::EnableBuiltInAEC(bool enable) {
+    RTC_DLOG(LS_INFO) << __FUNCTION__ << "(" << enable << ")";
+    CHECKinitialized_();
+    int32_t ok = audio_device_->EnableBuiltInAEC(enable);
+    RTC_DLOG(LS_INFO) << "output: " << ok;
+    return ok;
+  }
+
+  bool AudioDeviceModuleIOS::BuiltInAGCIsAvailable() const {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    CHECKinitialized__BOOL();
+    bool isAvailable = audio_device_->BuiltInAGCIsAvailable();
+    RTC_DLOG(LS_INFO) << "output: " << isAvailable;
+    return isAvailable;
+  }
+
+  int32_t AudioDeviceModuleIOS::EnableBuiltInAGC(bool enable) {
+    RTC_DLOG(LS_INFO) << __FUNCTION__ << "(" << enable << ")";
+    CHECKinitialized_();
+    int32_t ok = audio_device_->EnableBuiltInAGC(enable);
+    RTC_DLOG(LS_INFO) << "output: " << ok;
+    return ok;
+  }
+
+  bool AudioDeviceModuleIOS::BuiltInNSIsAvailable() const {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    CHECKinitialized__BOOL();
+    bool isAvailable = audio_device_->BuiltInNSIsAvailable();
+    RTC_DLOG(LS_INFO) << "output: " << isAvailable;
+    return isAvailable;
+  }
+
+  int32_t AudioDeviceModuleIOS::EnableBuiltInNS(bool enable) {
+    RTC_DLOG(LS_INFO) << __FUNCTION__ << "(" << enable << ")";
+    CHECKinitialized_();
+    int32_t ok = audio_device_->EnableBuiltInNS(enable);
+    RTC_DLOG(LS_INFO) << "output: " << ok;
+    return ok;
+  }
+
+  int32_t AudioDeviceModuleIOS::GetPlayoutUnderrunCount() const {
+    // Don't log here, as this method can be called very often.
+    CHECKinitialized_();
+    int32_t ok = audio_device_->GetPlayoutUnderrunCount();
+    return ok;
+  }
+
+#if defined(WEBRTC_IOS)
+  int AudioDeviceModuleIOS::GetPlayoutAudioParameters(
+      AudioParameters* params) const {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    int r = audio_device_->GetPlayoutAudioParameters(params);
+    RTC_DLOG(LS_INFO) << "output: " << r;
+    return r;
+  }
+
+  int AudioDeviceModuleIOS::GetRecordAudioParameters(
+      AudioParameters* params) const {
+    RTC_DLOG(LS_INFO) << __FUNCTION__;
+    int r = audio_device_->GetRecordAudioParameters(params);
+    RTC_DLOG(LS_INFO) << "output: " << r;
+    return r;
+  }
+#endif  // WEBRTC_IOS
+}
+}
diff --git a/third_party/libwebrtc/sdk/objc/native/src/audio/audio_session_observer.h b/third_party/libwebrtc/sdk/objc/native/src/audio/audio_session_observer.h
new file mode 100644
index 0000000000..f7c44c8184
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/native/src/audio/audio_session_observer.h
@@ -0,0 +1,41 @@
+/*
+ *  Copyright 2016 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef SDK_OBJC_NATIVE_SRC_AUDIO_AUDIO_SESSION_OBSERVER_H_
+#define SDK_OBJC_NATIVE_SRC_AUDIO_AUDIO_SESSION_OBSERVER_H_
+
+#include "rtc_base/thread.h"
+
+namespace webrtc {
+
+// Observer interface for listening to AVAudioSession events.
+class AudioSessionObserver {
+ public:
+  // Called when audio session interruption begins.
+  virtual void OnInterruptionBegin() = 0;
+
+  // Called when audio session interruption ends.
+  virtual void OnInterruptionEnd() = 0;
+
+  // Called when audio route changes.
+  virtual void OnValidRouteChange() = 0;
+
+  // Called when the ability to play or record changes.
+  virtual void OnCanPlayOrRecordChange(bool can_play_or_record) = 0;
+
+  virtual void OnChangedOutputVolume() = 0;
+
+ protected:
+  virtual ~AudioSessionObserver() {}
+};
+
+}  // namespace webrtc
+
+#endif  //  SDK_OBJC_NATIVE_SRC_AUDIO_AUDIO_SESSION_OBSERVER_H_
diff --git a/third_party/libwebrtc/sdk/objc/native/src/audio/helpers.h b/third_party/libwebrtc/sdk/objc/native/src/audio/helpers.h
new file mode 100644
index 0000000000..12464ac897
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/native/src/audio/helpers.h
@@ -0,0 +1,79 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef SDK_OBJC_NATIVE_SRC_AUDIO_HELPERS_H_
+#define SDK_OBJC_NATIVE_SRC_AUDIO_HELPERS_H_
+
+#import <Foundation/Foundation.h>
+#include <objc/objc.h>
+
+#include <string>
+
+namespace webrtc {
+namespace ios {
+
+bool CheckAndLogError(BOOL success, NSError* error);
+
+NSString* NSStringFromStdString(const std::string& stdString);
+std::string StdStringFromNSString(NSString* nsString);
+
+// Return thread ID as a string.
+std::string GetThreadId();
+
+// Return thread ID as string suitable for debug logging.
+std::string GetThreadInfo();
+
+// Returns [NSThread currentThread] description as string.
+// Example: <NSThread: 0x170066d80>{number = 1, name = main}
+std::string GetCurrentThreadDescription();
+
+#if defined(WEBRTC_IOS)
+// Returns the current name of the operating system.
+std::string GetSystemName();
+
+// Returns the current version of the operating system as a string.
+std::string GetSystemVersionAsString();
+
+// Returns the version of the operating system in double representation.
+// Uses a cached value of the system version.
+double GetSystemVersion();
+
+// Returns the device type.
+// Examples: ”iPhone” and ”iPod touch”.
+std::string GetDeviceType();
+#endif  // defined(WEBRTC_IOS)
+
+// Returns a more detailed device name.
+// Examples: "iPhone 5s (GSM)" and "iPhone 6 Plus".
+std::string GetDeviceName();
+
+// Returns the name of the process. Does not uniquely identify the process.
+std::string GetProcessName();
+
+// Returns the identifier of the process (often called process ID).
+int GetProcessID();
+
+// Returns a string containing the version of the operating system on which the
+// process is executing. The string is string is human readable, localized, and
+// is appropriate for displaying to the user.
+std::string GetOSVersionString();
+
+// Returns the number of processing cores available on the device.
+int GetProcessorCount();
+
+#if defined(WEBRTC_IOS)
+// Indicates whether Low Power Mode is enabled on the iOS device.
+bool GetLowPowerModeEnabled();
+#endif
+
+}  // namespace ios
+}  // namespace webrtc
+
+#endif  // SDK_OBJC_NATIVE_SRC_AUDIO_HELPERS_H_
diff --git a/third_party/libwebrtc/sdk/objc/native/src/audio/helpers.mm b/third_party/libwebrtc/sdk/objc/native/src/audio/helpers.mm
new file mode 100644
index 0000000000..cd0469656a
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/native/src/audio/helpers.mm
@@ -0,0 +1,109 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+#import <sys/sysctl.h>
+#if defined(WEBRTC_IOS)
+#import <UIKit/UIKit.h>
+#endif
+
+#include <memory>
+
+#include "helpers.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+namespace ios {
+
+NSString* NSStringFromStdString(const std::string& stdString) {
+  // std::string may contain null termination character so we construct
+  // using length.
+  return [[NSString alloc] initWithBytes:stdString.data()
+                                  length:stdString.length()
+                                encoding:NSUTF8StringEncoding];
+}
+
+std::string StdStringFromNSString(NSString* nsString) {
+  NSData* charData = [nsString dataUsingEncoding:NSUTF8StringEncoding];
+  return std::string(reinterpret_cast<const char*>([charData bytes]),
+                     [charData length]);
+}
+
+bool CheckAndLogError(BOOL success, NSError* error) {
+  if (!success) {
+    NSString* msg =
+        [NSString stringWithFormat:@"Error: %ld, %@, %@", (long)error.code,
+                                   error.localizedDescription,
+                                   error.localizedFailureReason];
+    RTC_LOG(LS_ERROR) << StdStringFromNSString(msg);
+    return false;
+  }
+  return true;
+}
+
+// TODO(henrika): see if it is possible to move to GetThreadName in
+// platform_thread.h and base it on pthread methods instead.
+std::string GetCurrentThreadDescription() {
+  NSString* name = [NSString stringWithFormat:@"%@", [NSThread currentThread]];
+  return StdStringFromNSString(name);
+}
+
+#if defined(WEBRTC_IOS)
+std::string GetSystemName() {
+  NSString* osName = [[UIDevice currentDevice] systemName];
+  return StdStringFromNSString(osName);
+}
+
+std::string GetSystemVersionAsString() {
+  NSString* osVersion = [[UIDevice currentDevice] systemVersion];
+  return StdStringFromNSString(osVersion);
+}
+
+std::string GetDeviceType() {
+  NSString* deviceModel = [[UIDevice currentDevice] model];
+  return StdStringFromNSString(deviceModel);
+}
+
+bool GetLowPowerModeEnabled() {
+  return [NSProcessInfo processInfo].lowPowerModeEnabled;
+}
+#endif
+
+std::string GetDeviceName() {
+  size_t size;
+  sysctlbyname("hw.machine", NULL, &size, NULL, 0);
+  std::unique_ptr<char[]> machine;
+  machine.reset(new char[size]);
+  sysctlbyname("hw.machine", machine.get(), &size, NULL, 0);
+  return std::string(machine.get());
+}
+
+std::string GetProcessName() {
+  NSString* processName = [NSProcessInfo processInfo].processName;
+  return StdStringFromNSString(processName);
+}
+
+int GetProcessID() {
+  return [NSProcessInfo processInfo].processIdentifier;
+}
+
+std::string GetOSVersionString() {
+  NSString* osVersion =
+      [NSProcessInfo processInfo].operatingSystemVersionString;
+  return StdStringFromNSString(osVersion);
+}
+
+int GetProcessorCount() {
+  return [NSProcessInfo processInfo].processorCount;
+}
+
+}  // namespace ios
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/sdk/objc/native/src/audio/voice_processing_audio_unit.h b/third_party/libwebrtc/sdk/objc/native/src/audio/voice_processing_audio_unit.h
new file mode 100644
index 0000000000..ed9dd98568
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/native/src/audio/voice_processing_audio_unit.h
@@ -0,0 +1,141 @@
+/*
+ *  Copyright 2016 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef SDK_OBJC_NATIVE_SRC_AUDIO_VOICE_PROCESSING_AUDIO_UNIT_H_
+#define SDK_OBJC_NATIVE_SRC_AUDIO_VOICE_PROCESSING_AUDIO_UNIT_H_
+
+#include <AudioUnit/AudioUnit.h>
+
+namespace webrtc {
+namespace ios_adm {
+
+class VoiceProcessingAudioUnitObserver {
+ public:
+  // Callback function called on a real-time priority I/O thread from the audio
+  // unit. This method is used to signal that recorded audio is available.
+  virtual OSStatus OnDeliverRecordedData(AudioUnitRenderActionFlags* flags,
+                                         const AudioTimeStamp* time_stamp,
+                                         UInt32 bus_number,
+                                         UInt32 num_frames,
+                                         AudioBufferList* io_data) = 0;
+
+  // Callback function called on a real-time priority I/O thread from the audio
+  // unit. This method is used to provide audio samples to the audio unit.
+  virtual OSStatus OnGetPlayoutData(AudioUnitRenderActionFlags* io_action_flags,
+                                    const AudioTimeStamp* time_stamp,
+                                    UInt32 bus_number,
+                                    UInt32 num_frames,
+                                    AudioBufferList* io_data) = 0;
+
+ protected:
+  ~VoiceProcessingAudioUnitObserver() {}
+};
+
+// Convenience class to abstract away the management of a Voice Processing
+// I/O Audio Unit. The Voice Processing I/O unit has the same characteristics
+// as the Remote I/O unit (supports full duplex low-latency audio input and
+// output) and adds AEC for for two-way duplex communication. It also adds AGC,
+// adjustment of voice-processing quality, and muting. Hence, ideal for
+// VoIP applications.
+class VoiceProcessingAudioUnit {
+ public:
+  VoiceProcessingAudioUnit(bool bypass_voice_processing,
+                           VoiceProcessingAudioUnitObserver* observer);
+  ~VoiceProcessingAudioUnit();
+
+  // TODO(tkchin): enum for state and state checking.
+  enum State : int32_t {
+    // Init() should be called.
+    kInitRequired,
+    // Audio unit created but not initialized.
+    kUninitialized,
+    // Initialized but not started. Equivalent to stopped.
+    kInitialized,
+    // Initialized and started.
+    kStarted,
+  };
+
+  // Number of bytes per audio sample for 16-bit signed integer representation.
+  static const UInt32 kBytesPerSample;
+
+  // Initializes this class by creating the underlying audio unit instance.
+  // Creates a Voice-Processing I/O unit and configures it for full-duplex
+  // audio. The selected stream format is selected to avoid internal resampling
+  // and to match the 10ms callback rate for WebRTC as well as possible.
+  // Does not intialize the audio unit.
+  bool Init();
+
+  VoiceProcessingAudioUnit::State GetState() const;
+
+  // Initializes the underlying audio unit with the given sample rate.
+  bool Initialize(Float64 sample_rate);
+
+  // Starts the underlying audio unit.
+  OSStatus Start();
+
+  // Stops the underlying audio unit.
+  bool Stop();
+
+  // Uninitializes the underlying audio unit.
+  bool Uninitialize();
+
+  // Calls render on the underlying audio unit.
+  OSStatus Render(AudioUnitRenderActionFlags* flags,
+                  const AudioTimeStamp* time_stamp,
+                  UInt32 output_bus_number,
+                  UInt32 num_frames,
+                  AudioBufferList* io_data);
+
+ private:
+  // The C API used to set callbacks requires static functions. When these are
+  // called, they will invoke the relevant instance method by casting
+  // in_ref_con to VoiceProcessingAudioUnit*.
+  static OSStatus OnGetPlayoutData(void* in_ref_con,
+                                   AudioUnitRenderActionFlags* flags,
+                                   const AudioTimeStamp* time_stamp,
+                                   UInt32 bus_number,
+                                   UInt32 num_frames,
+                                   AudioBufferList* io_data);
+  static OSStatus OnDeliverRecordedData(void* in_ref_con,
+                                        AudioUnitRenderActionFlags* flags,
+                                        const AudioTimeStamp* time_stamp,
+                                        UInt32 bus_number,
+                                        UInt32 num_frames,
+                                        AudioBufferList* io_data);
+
+  // Notifies observer that samples are needed for playback.
+  OSStatus NotifyGetPlayoutData(AudioUnitRenderActionFlags* flags,
+                                const AudioTimeStamp* time_stamp,
+                                UInt32 bus_number,
+                                UInt32 num_frames,
+                                AudioBufferList* io_data);
+  // Notifies observer that recorded samples are available for render.
+  OSStatus NotifyDeliverRecordedData(AudioUnitRenderActionFlags* flags,
+                                     const AudioTimeStamp* time_stamp,
+                                     UInt32 bus_number,
+                                     UInt32 num_frames,
+                                     AudioBufferList* io_data);
+
+  // Returns the predetermined format with a specific sample rate. See
+  // implementation file for details on format.
+  AudioStreamBasicDescription GetFormat(Float64 sample_rate) const;
+
+  // Deletes the underlying audio unit.
+  void DisposeAudioUnit();
+
+  const bool bypass_voice_processing_;
+  VoiceProcessingAudioUnitObserver* observer_;
+  AudioUnit vpio_unit_;
+  VoiceProcessingAudioUnit::State state_;
+};
+}  // namespace ios_adm
+}  // namespace webrtc
+
+#endif  // SDK_OBJC_NATIVE_SRC_AUDIO_VOICE_PROCESSING_AUDIO_UNIT_H_
diff --git a/third_party/libwebrtc/sdk/objc/native/src/audio/voice_processing_audio_unit.mm b/third_party/libwebrtc/sdk/objc/native/src/audio/voice_processing_audio_unit.mm
new file mode 100644
index 0000000000..3905b6857a
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/native/src/audio/voice_processing_audio_unit.mm
@@ -0,0 +1,488 @@
+/*
+ *  Copyright 2016 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "voice_processing_audio_unit.h"
+
+#include "rtc_base/checks.h"
+#include "system_wrappers/include/metrics.h"
+
+#import "base/RTCLogging.h"
+#import "sdk/objc/components/audio/RTCAudioSessionConfiguration.h"
+
+#if !defined(NDEBUG)
+static void LogStreamDescription(AudioStreamBasicDescription description) {
+  char formatIdString[5];
+  UInt32 formatId = CFSwapInt32HostToBig(description.mFormatID);
+  bcopy(&formatId, formatIdString, 4);
+  formatIdString[4] = '\0';
+  RTCLog(@"AudioStreamBasicDescription: {\n"
+          "  mSampleRate: %.2f\n"
+          "  formatIDString: %s\n"
+          "  mFormatFlags: 0x%X\n"
+          "  mBytesPerPacket: %u\n"
+          "  mFramesPerPacket: %u\n"
+          "  mBytesPerFrame: %u\n"
+          "  mChannelsPerFrame: %u\n"
+          "  mBitsPerChannel: %u\n"
+          "  mReserved: %u\n}",
+         description.mSampleRate, formatIdString,
+         static_cast<unsigned int>(description.mFormatFlags),
+         static_cast<unsigned int>(description.mBytesPerPacket),
+         static_cast<unsigned int>(description.mFramesPerPacket),
+         static_cast<unsigned int>(description.mBytesPerFrame),
+         static_cast<unsigned int>(description.mChannelsPerFrame),
+         static_cast<unsigned int>(description.mBitsPerChannel),
+         static_cast<unsigned int>(description.mReserved));
+}
+#endif
+
+namespace webrtc {
+namespace ios_adm {
+
+// Calls to AudioUnitInitialize() can fail if called back-to-back on different
+// ADM instances. A fall-back solution is to allow multiple sequential calls
+// with as small delay between each. This factor sets the max number of allowed
+// initialization attempts.
+static const int kMaxNumberOfAudioUnitInitializeAttempts = 5;
+// A VP I/O unit's bus 1 connects to input hardware (microphone).
+static const AudioUnitElement kInputBus = 1;
+// A VP I/O unit's bus 0 connects to output hardware (speaker).
+static const AudioUnitElement kOutputBus = 0;
+
+// Returns the automatic gain control (AGC) state on the processed microphone
+// signal. Should be on by default for Voice Processing audio units.
+static OSStatus GetAGCState(AudioUnit audio_unit, UInt32* enabled) {
+  RTC_DCHECK(audio_unit);
+  UInt32 size = sizeof(*enabled);
+  OSStatus result = AudioUnitGetProperty(audio_unit,
+                                         kAUVoiceIOProperty_VoiceProcessingEnableAGC,
+                                         kAudioUnitScope_Global,
+                                         kInputBus,
+                                         enabled,
+                                         &size);
+  RTCLog(@"VPIO unit AGC: %u", static_cast<unsigned int>(*enabled));
+  return result;
+}
+
+VoiceProcessingAudioUnit::VoiceProcessingAudioUnit(bool bypass_voice_processing,
+                                                   VoiceProcessingAudioUnitObserver* observer)
+    : bypass_voice_processing_(bypass_voice_processing),
+      observer_(observer),
+      vpio_unit_(nullptr),
+      state_(kInitRequired) {
+  RTC_DCHECK(observer);
+}
+
+VoiceProcessingAudioUnit::~VoiceProcessingAudioUnit() {
+  DisposeAudioUnit();
+}
+
+const UInt32 VoiceProcessingAudioUnit::kBytesPerSample = 2;
+
+bool VoiceProcessingAudioUnit::Init() {
+  RTC_DCHECK_EQ(state_, kInitRequired);
+
+  // Create an audio component description to identify the Voice Processing
+  // I/O audio unit.
+  AudioComponentDescription vpio_unit_description;
+  vpio_unit_description.componentType = kAudioUnitType_Output;
+  vpio_unit_description.componentSubType = kAudioUnitSubType_VoiceProcessingIO;
+  vpio_unit_description.componentManufacturer = kAudioUnitManufacturer_Apple;
+  vpio_unit_description.componentFlags = 0;
+  vpio_unit_description.componentFlagsMask = 0;
+
+  // Obtain an audio unit instance given the description.
+  AudioComponent found_vpio_unit_ref =
+      AudioComponentFindNext(nullptr, &vpio_unit_description);
+
+  // Create a Voice Processing IO audio unit.
+  OSStatus result = noErr;
+  result = AudioComponentInstanceNew(found_vpio_unit_ref, &vpio_unit_);
+  if (result != noErr) {
+    vpio_unit_ = nullptr;
+    RTCLogError(@"AudioComponentInstanceNew failed. Error=%ld.", (long)result);
+    return false;
+  }
+
+  // Enable input on the input scope of the input element.
+  UInt32 enable_input = 1;
+  result = AudioUnitSetProperty(vpio_unit_, kAudioOutputUnitProperty_EnableIO,
+                                kAudioUnitScope_Input, kInputBus, &enable_input,
+                                sizeof(enable_input));
+  if (result != noErr) {
+    DisposeAudioUnit();
+    RTCLogError(@"Failed to enable input on input scope of input element. "
+                 "Error=%ld.",
+                (long)result);
+    return false;
+  }
+
+  // Enable output on the output scope of the output element.
+  UInt32 enable_output = 1;
+  result = AudioUnitSetProperty(vpio_unit_, kAudioOutputUnitProperty_EnableIO,
+                                kAudioUnitScope_Output, kOutputBus,
+                                &enable_output, sizeof(enable_output));
+  if (result != noErr) {
+    DisposeAudioUnit();
+    RTCLogError(@"Failed to enable output on output scope of output element. "
+                 "Error=%ld.",
+                (long)result);
+    return false;
+  }
+
+  // Specify the callback function that provides audio samples to the audio
+  // unit.
+  AURenderCallbackStruct render_callback;
+  render_callback.inputProc = OnGetPlayoutData;
+  render_callback.inputProcRefCon = this;
+  result = AudioUnitSetProperty(
+      vpio_unit_, kAudioUnitProperty_SetRenderCallback, kAudioUnitScope_Input,
+      kOutputBus, &render_callback, sizeof(render_callback));
+  if (result != noErr) {
+    DisposeAudioUnit();
+    RTCLogError(@"Failed to specify the render callback on the output bus. "
+                 "Error=%ld.",
+                (long)result);
+    return false;
+  }
+
+  // Disable AU buffer allocation for the recorder, we allocate our own.
+  // TODO(henrika): not sure that it actually saves resource to make this call.
+  UInt32 flag = 0;
+  result = AudioUnitSetProperty(
+      vpio_unit_, kAudioUnitProperty_ShouldAllocateBuffer,
+      kAudioUnitScope_Output, kInputBus, &flag, sizeof(flag));
+  if (result != noErr) {
+    DisposeAudioUnit();
+    RTCLogError(@"Failed to disable buffer allocation on the input bus. "
+                 "Error=%ld.",
+                (long)result);
+    return false;
+  }
+
+  // Specify the callback to be called by the I/O thread to us when input audio
+  // is available. The recorded samples can then be obtained by calling the
+  // AudioUnitRender() method.
+  AURenderCallbackStruct input_callback;
+  input_callback.inputProc = OnDeliverRecordedData;
+  input_callback.inputProcRefCon = this;
+  result = AudioUnitSetProperty(vpio_unit_,
+                                kAudioOutputUnitProperty_SetInputCallback,
+                                kAudioUnitScope_Global, kInputBus,
+                                &input_callback, sizeof(input_callback));
+  if (result != noErr) {
+    DisposeAudioUnit();
+    RTCLogError(@"Failed to specify the input callback on the input bus. "
+                 "Error=%ld.",
+                (long)result);
+    return false;
+  }
+
+  state_ = kUninitialized;
+  return true;
+}
+
+VoiceProcessingAudioUnit::State VoiceProcessingAudioUnit::GetState() const {
+  return state_;
+}
+
+bool VoiceProcessingAudioUnit::Initialize(Float64 sample_rate) {
+  RTC_DCHECK_GE(state_, kUninitialized);
+  RTCLog(@"Initializing audio unit with sample rate: %f", sample_rate);
+
+  OSStatus result = noErr;
+  AudioStreamBasicDescription format = GetFormat(sample_rate);
+  UInt32 size = sizeof(format);
+#if !defined(NDEBUG)
+  LogStreamDescription(format);
+#endif
+
+  // Set the format on the output scope of the input element/bus.
+  result =
+      AudioUnitSetProperty(vpio_unit_, kAudioUnitProperty_StreamFormat,
+                           kAudioUnitScope_Output, kInputBus, &format, size);
+  if (result != noErr) {
+    RTCLogError(@"Failed to set format on output scope of input bus. "
+                 "Error=%ld.",
+                (long)result);
+    return false;
+  }
+
+  // Set the format on the input scope of the output element/bus.
+  result =
+      AudioUnitSetProperty(vpio_unit_, kAudioUnitProperty_StreamFormat,
+                           kAudioUnitScope_Input, kOutputBus, &format, size);
+  if (result != noErr) {
+    RTCLogError(@"Failed to set format on input scope of output bus. "
+                 "Error=%ld.",
+                (long)result);
+    return false;
+  }
+
+  // Initialize the Voice Processing I/O unit instance.
+  // Calls to AudioUnitInitialize() can fail if called back-to-back on
+  // different ADM instances. The error message in this case is -66635 which is
+  // undocumented. Tests have shown that calling AudioUnitInitialize a second
+  // time, after a short sleep, avoids this issue.
+  // See webrtc:5166 for details.
+  int failed_initalize_attempts = 0;
+  result = AudioUnitInitialize(vpio_unit_);
+  while (result != noErr) {
+    RTCLogError(@"Failed to initialize the Voice Processing I/O unit. "
+                 "Error=%ld.",
+                (long)result);
+    ++failed_initalize_attempts;
+    if (failed_initalize_attempts == kMaxNumberOfAudioUnitInitializeAttempts) {
+      // Max number of initialization attempts exceeded, hence abort.
+      RTCLogError(@"Too many initialization attempts.");
+      return false;
+    }
+    RTCLog(@"Pause 100ms and try audio unit initialization again...");
+    [NSThread sleepForTimeInterval:0.1f];
+    result = AudioUnitInitialize(vpio_unit_);
+  }
+  if (result == noErr) {
+    RTCLog(@"Voice Processing I/O unit is now initialized.");
+  }
+
+  if (bypass_voice_processing_) {
+    // Attempt to disable builtin voice processing.
+    UInt32 toggle = 1;
+    result = AudioUnitSetProperty(vpio_unit_,
+                                  kAUVoiceIOProperty_BypassVoiceProcessing,
+                                  kAudioUnitScope_Global,
+                                  kInputBus,
+                                  &toggle,
+                                  sizeof(toggle));
+    if (result == noErr) {
+      RTCLog(@"Successfully bypassed voice processing.");
+    } else {
+      RTCLogError(@"Failed to bypass voice processing. Error=%ld.", (long)result);
+    }
+    state_ = kInitialized;
+    return true;
+  }
+
+  // AGC should be enabled by default for Voice Processing I/O units but it is
+  // checked below and enabled explicitly if needed. This scheme is used
+  // to be absolutely sure that the AGC is enabled since we have seen cases
+  // where only zeros are recorded and a disabled AGC could be one of the
+  // reasons why it happens.
+  int agc_was_enabled_by_default = 0;
+  UInt32 agc_is_enabled = 0;
+  result = GetAGCState(vpio_unit_, &agc_is_enabled);
+  if (result != noErr) {
+    RTCLogError(@"Failed to get AGC state (1st attempt). "
+                 "Error=%ld.",
+                (long)result);
+    // Example of error code: kAudioUnitErr_NoConnection (-10876).
+    // All error codes related to audio units are negative and are therefore
+    // converted into a postive value to match the UMA APIs.
+    RTC_HISTOGRAM_COUNTS_SPARSE_100000(
+        "WebRTC.Audio.GetAGCStateErrorCode1", (-1) * result);
+  } else if (agc_is_enabled) {
+    // Remember that the AGC was enabled by default. Will be used in UMA.
+    agc_was_enabled_by_default = 1;
+  } else {
+    // AGC was initially disabled => try to enable it explicitly.
+    UInt32 enable_agc = 1;
+    result =
+        AudioUnitSetProperty(vpio_unit_,
+                             kAUVoiceIOProperty_VoiceProcessingEnableAGC,
+                             kAudioUnitScope_Global, kInputBus, &enable_agc,
+                             sizeof(enable_agc));
+    if (result != noErr) {
+      RTCLogError(@"Failed to enable the built-in AGC. "
+                   "Error=%ld.",
+                  (long)result);
+      RTC_HISTOGRAM_COUNTS_SPARSE_100000(
+          "WebRTC.Audio.SetAGCStateErrorCode", (-1) * result);
+    }
+    result = GetAGCState(vpio_unit_, &agc_is_enabled);
+    if (result != noErr) {
+      RTCLogError(@"Failed to get AGC state (2nd attempt). "
+                   "Error=%ld.",
+                  (long)result);
+      RTC_HISTOGRAM_COUNTS_SPARSE_100000(
+          "WebRTC.Audio.GetAGCStateErrorCode2", (-1) * result);
+    }
+  }
+
+  // Track if the built-in AGC was enabled by default (as it should) or not.
+  RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.BuiltInAGCWasEnabledByDefault",
+                        agc_was_enabled_by_default);
+  RTCLog(@"WebRTC.Audio.BuiltInAGCWasEnabledByDefault: %d",
+         agc_was_enabled_by_default);
+  // As a final step, add an UMA histogram for tracking the AGC state.
+  // At this stage, the AGC should be enabled, and if it is not, more work is
+  // needed to find out the root cause.
+  RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.BuiltInAGCIsEnabled", agc_is_enabled);
+  RTCLog(@"WebRTC.Audio.BuiltInAGCIsEnabled: %u",
+         static_cast<unsigned int>(agc_is_enabled));
+
+  state_ = kInitialized;
+  return true;
+}
+
+OSStatus VoiceProcessingAudioUnit::Start() {
+  RTC_DCHECK_GE(state_, kUninitialized);
+  RTCLog(@"Starting audio unit.");
+
+  OSStatus result = AudioOutputUnitStart(vpio_unit_);
+  if (result != noErr) {
+    RTCLogError(@"Failed to start audio unit. Error=%ld", (long)result);
+    return result;
+  } else {
+    RTCLog(@"Started audio unit");
+  }
+  state_ = kStarted;
+  return noErr;
+}
+
+bool VoiceProcessingAudioUnit::Stop() {
+  RTC_DCHECK_GE(state_, kUninitialized);
+  RTCLog(@"Stopping audio unit.");
+
+  OSStatus result = AudioOutputUnitStop(vpio_unit_);
+  if (result != noErr) {
+    RTCLogError(@"Failed to stop audio unit. Error=%ld", (long)result);
+    return false;
+  } else {
+    RTCLog(@"Stopped audio unit");
+  }
+
+  state_ = kInitialized;
+  return true;
+}
+
+bool VoiceProcessingAudioUnit::Uninitialize() {
+  RTC_DCHECK_GE(state_, kUninitialized);
+  RTCLog(@"Unintializing audio unit.");
+
+  OSStatus result = AudioUnitUninitialize(vpio_unit_);
+  if (result != noErr) {
+    RTCLogError(@"Failed to uninitialize audio unit. Error=%ld", (long)result);
+    return false;
+  } else {
+    RTCLog(@"Uninitialized audio unit.");
+  }
+
+  state_ = kUninitialized;
+  return true;
+}
+
+OSStatus VoiceProcessingAudioUnit::Render(AudioUnitRenderActionFlags* flags,
+                                          const AudioTimeStamp* time_stamp,
+                                          UInt32 output_bus_number,
+                                          UInt32 num_frames,
+                                          AudioBufferList* io_data) {
+  RTC_DCHECK(vpio_unit_) << "Init() not called.";
+
+  OSStatus result = AudioUnitRender(vpio_unit_, flags, time_stamp,
+                                    output_bus_number, num_frames, io_data);
+  if (result != noErr) {
+    RTCLogError(@"Failed to render audio unit. Error=%ld", (long)result);
+  }
+  return result;
+}
+
+OSStatus VoiceProcessingAudioUnit::OnGetPlayoutData(
+    void* in_ref_con,
+    AudioUnitRenderActionFlags* flags,
+    const AudioTimeStamp* time_stamp,
+    UInt32 bus_number,
+    UInt32 num_frames,
+    AudioBufferList* io_data) {
+  VoiceProcessingAudioUnit* audio_unit =
+      static_cast<VoiceProcessingAudioUnit*>(in_ref_con);
+  return audio_unit->NotifyGetPlayoutData(flags, time_stamp, bus_number,
+                                          num_frames, io_data);
+}
+
+OSStatus VoiceProcessingAudioUnit::OnDeliverRecordedData(
+    void* in_ref_con,
+    AudioUnitRenderActionFlags* flags,
+    const AudioTimeStamp* time_stamp,
+    UInt32 bus_number,
+    UInt32 num_frames,
+    AudioBufferList* io_data) {
+  VoiceProcessingAudioUnit* audio_unit =
+      static_cast<VoiceProcessingAudioUnit*>(in_ref_con);
+  return audio_unit->NotifyDeliverRecordedData(flags, time_stamp, bus_number,
+                                               num_frames, io_data);
+}
+
+OSStatus VoiceProcessingAudioUnit::NotifyGetPlayoutData(
+    AudioUnitRenderActionFlags* flags,
+    const AudioTimeStamp* time_stamp,
+    UInt32 bus_number,
+    UInt32 num_frames,
+    AudioBufferList* io_data) {
+  return observer_->OnGetPlayoutData(flags, time_stamp, bus_number, num_frames,
+                                     io_data);
+}
+
+OSStatus VoiceProcessingAudioUnit::NotifyDeliverRecordedData(
+    AudioUnitRenderActionFlags* flags,
+    const AudioTimeStamp* time_stamp,
+    UInt32 bus_number,
+    UInt32 num_frames,
+    AudioBufferList* io_data) {
+  return observer_->OnDeliverRecordedData(flags, time_stamp, bus_number,
+                                          num_frames, io_data);
+}
+
+AudioStreamBasicDescription VoiceProcessingAudioUnit::GetFormat(
+    Float64 sample_rate) const {
+  // Set the application formats for input and output:
+  // - use same format in both directions
+  // - avoid resampling in the I/O unit by using the hardware sample rate
+  // - linear PCM => noncompressed audio data format with one frame per packet
+  // - no need to specify interleaving since only mono is supported
+  AudioStreamBasicDescription format;
+  RTC_DCHECK_EQ(1, kRTCAudioSessionPreferredNumberOfChannels);
+  format.mSampleRate = sample_rate;
+  format.mFormatID = kAudioFormatLinearPCM;
+  format.mFormatFlags =
+      kLinearPCMFormatFlagIsSignedInteger | kLinearPCMFormatFlagIsPacked;
+  format.mBytesPerPacket = kBytesPerSample;
+  format.mFramesPerPacket = 1;  // uncompressed.
+  format.mBytesPerFrame = kBytesPerSample;
+  format.mChannelsPerFrame = kRTCAudioSessionPreferredNumberOfChannels;
+  format.mBitsPerChannel = 8 * kBytesPerSample;
+  return format;
+}
+
+void VoiceProcessingAudioUnit::DisposeAudioUnit() {
+  if (vpio_unit_) {
+    switch (state_) {
+      case kStarted:
+        Stop();
+        [[fallthrough]];
+      case kInitialized:
+        Uninitialize();
+        break;
+      case kUninitialized:
+      case kInitRequired:
+        break;
+    }
+
+    RTCLog(@"Disposing audio unit.");
+    OSStatus result = AudioComponentInstanceDispose(vpio_unit_);
+    if (result != noErr) {
+      RTCLogError(@"AudioComponentInstanceDispose failed. Error=%ld.",
+                  (long)result);
+    }
+    vpio_unit_ = nullptr;
+  }
+}
+
+}  // namespace ios_adm
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/sdk/objc/native/src/network_monitor_observer.h b/third_party/libwebrtc/sdk/objc/native/src/network_monitor_observer.h
new file mode 100644
index 0000000000..7c411a1db1
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/native/src/network_monitor_observer.h
@@ -0,0 +1,42 @@
+/*
+ *  Copyright 2020 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef SDK_OBJC_NATIVE_SRC_NETWORK_MONITOR_OBSERVER_H_
+#define SDK_OBJC_NATIVE_SRC_NETWORK_MONITOR_OBSERVER_H_
+
+#include <map>
+#include <string>
+
+#include "absl/strings/string_view.h"
+#include "rtc_base/network_constants.h"
+#include "rtc_base/string_utils.h"
+#include "rtc_base/thread.h"
+
+namespace webrtc {
+
+// Observer interface for listening to NWPathMonitor updates.
+class NetworkMonitorObserver {
+ public:
+  // Called when a path update occurs, on network monitor dispatch queue.
+  //
+  // `adapter_type_by_name` is a map from interface name (i.e. "pdp_ip0") to
+  // adapter type, for all available interfaces on the current path. If an
+  // interface name isn't present it can be assumed to be unavailable.
+  virtual void OnPathUpdate(
+      std::map<std::string, rtc::AdapterType, rtc::AbslStringViewCmp>
+          adapter_type_by_name) = 0;
+
+ protected:
+  virtual ~NetworkMonitorObserver() {}
+};
+
+}  // namespace webrtc
+
+#endif  //  SDK_OBJC_NATIVE_SRC_AUDIO_AUDIO_SESSION_OBSERVER_H_
diff --git a/third_party/libwebrtc/sdk/objc/native/src/objc_audio_device.h b/third_party/libwebrtc/sdk/objc/native/src/objc_audio_device.h
new file mode 100644
index 0000000000..fcfe7a6e8b
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/native/src/objc_audio_device.h
@@ -0,0 +1,277 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef SDK_OBJC_NATIVE_SRC_OBJC_AUDIO_DEVICE_H_
+#define SDK_OBJC_NATIVE_SRC_OBJC_AUDIO_DEVICE_H_
+
+#include <memory>
+
+#import "components/audio/RTCAudioDevice.h"
+
+#include "modules/audio_device/audio_device_buffer.h"
+#include "modules/audio_device/include/audio_device.h"
+#include "rtc_base/thread.h"
+
+@class ObjCAudioDeviceDelegate;
+
+namespace webrtc {
+
+class FineAudioBuffer;
+
+namespace objc_adm {
+
+class ObjCAudioDeviceModule : public AudioDeviceModule {
+ public:
+  explicit ObjCAudioDeviceModule(id<RTC_OBJC_TYPE(RTCAudioDevice)> audio_device);
+  ~ObjCAudioDeviceModule() override;
+
+  // Retrieve the currently utilized audio layer
+  int32_t ActiveAudioLayer(AudioLayer* audioLayer) const override;
+
+  // Full-duplex transportation of PCM audio
+  int32_t RegisterAudioCallback(AudioTransport* audioCallback) override;
+
+  // Main initialization and termination
+  int32_t Init() override;
+  int32_t Terminate() override;
+  bool Initialized() const override;
+
+  // Device enumeration
+  int16_t PlayoutDevices() override;
+  int16_t RecordingDevices() override;
+  int32_t PlayoutDeviceName(uint16_t index,
+                            char name[kAdmMaxDeviceNameSize],
+                            char guid[kAdmMaxGuidSize]) override;
+  int32_t RecordingDeviceName(uint16_t index,
+                              char name[kAdmMaxDeviceNameSize],
+                              char guid[kAdmMaxGuidSize]) override;
+
+  // Device selection
+  int32_t SetPlayoutDevice(uint16_t index) override;
+  int32_t SetPlayoutDevice(WindowsDeviceType device) override;
+  int32_t SetRecordingDevice(uint16_t index) override;
+  int32_t SetRecordingDevice(WindowsDeviceType device) override;
+
+  // Audio transport initialization
+  int32_t PlayoutIsAvailable(bool* available) override;
+  int32_t InitPlayout() override;
+  bool PlayoutIsInitialized() const override;
+  int32_t RecordingIsAvailable(bool* available) override;
+  int32_t InitRecording() override;
+  bool RecordingIsInitialized() const override;
+
+  // Audio transport control
+  int32_t StartPlayout() override;
+  int32_t StopPlayout() override;
+  bool Playing() const override;
+  int32_t StartRecording() override;
+  int32_t StopRecording() override;
+  bool Recording() const override;
+
+  // Audio mixer initialization
+  int32_t InitSpeaker() override;
+  bool SpeakerIsInitialized() const override;
+  int32_t InitMicrophone() override;
+  bool MicrophoneIsInitialized() const override;
+
+  // Speaker volume controls
+  int32_t SpeakerVolumeIsAvailable(bool* available) override;
+  int32_t SetSpeakerVolume(uint32_t volume) override;
+  int32_t SpeakerVolume(uint32_t* volume) const override;
+  int32_t MaxSpeakerVolume(uint32_t* maxVolume) const override;
+  int32_t MinSpeakerVolume(uint32_t* minVolume) const override;
+
+  // Microphone volume controls
+  int32_t MicrophoneVolumeIsAvailable(bool* available) override;
+  int32_t SetMicrophoneVolume(uint32_t volume) override;
+  int32_t MicrophoneVolume(uint32_t* volume) const override;
+  int32_t MaxMicrophoneVolume(uint32_t* maxVolume) const override;
+  int32_t MinMicrophoneVolume(uint32_t* minVolume) const override;
+
+  // Speaker mute control
+  int32_t SpeakerMuteIsAvailable(bool* available) override;
+  int32_t SetSpeakerMute(bool enable) override;
+  int32_t SpeakerMute(bool* enabled) const override;
+
+  // Microphone mute control
+  int32_t MicrophoneMuteIsAvailable(bool* available) override;
+  int32_t SetMicrophoneMute(bool enable) override;
+  int32_t MicrophoneMute(bool* enabled) const override;
+
+  // Stereo support
+  int32_t StereoPlayoutIsAvailable(bool* available) const override;
+  int32_t SetStereoPlayout(bool enable) override;
+  int32_t StereoPlayout(bool* enabled) const override;
+  int32_t StereoRecordingIsAvailable(bool* available) const override;
+  int32_t SetStereoRecording(bool enable) override;
+  int32_t StereoRecording(bool* enabled) const override;
+
+  // Playout delay
+  int32_t PlayoutDelay(uint16_t* delayMS) const override;
+
+  // Only supported on Android.
+  bool BuiltInAECIsAvailable() const override;
+  bool BuiltInAGCIsAvailable() const override;
+  bool BuiltInNSIsAvailable() const override;
+
+  // Enables the built-in audio effects. Only supported on Android.
+  int32_t EnableBuiltInAEC(bool enable) override;
+  int32_t EnableBuiltInAGC(bool enable) override;
+  int32_t EnableBuiltInNS(bool enable) override;
+
+  // Play underrun count. Only supported on Android.
+  int32_t GetPlayoutUnderrunCount() const override;
+
+#if defined(WEBRTC_IOS)
+  int GetPlayoutAudioParameters(AudioParameters* params) const override;
+  int GetRecordAudioParameters(AudioParameters* params) const override;
+#endif  // WEBRTC_IOS
+
+ public:
+  OSStatus OnDeliverRecordedData(AudioUnitRenderActionFlags* flags,
+                                 const AudioTimeStamp* time_stamp,
+                                 NSInteger bus_number,
+                                 UInt32 num_frames,
+                                 const AudioBufferList* io_data,
+                                 void* render_context,
+                                 RTC_OBJC_TYPE(RTCAudioDeviceRenderRecordedDataBlock) render_block);
+
+  OSStatus OnGetPlayoutData(AudioUnitRenderActionFlags* flags,
+                            const AudioTimeStamp* time_stamp,
+                            NSInteger bus_number,
+                            UInt32 num_frames,
+                            AudioBufferList* io_data);
+
+  // Notifies `ObjCAudioDeviceModule` that at least one of the audio input
+  // parameters or audio input latency of `RTCAudioDevice` has changed. It necessary to
+  // update `record_parameters_` with current audio parameter of `RTCAudioDevice`
+  // via `UpdateAudioParameters` and if parameters are actually change then
+  // ADB parameters are updated with `UpdateInputAudioDeviceBuffer`. Audio input latency
+  // stored in `cached_recording_delay_ms_` is also updated with current latency
+  // of `RTCAudioDevice`.
+  void HandleAudioInputParametersChange();
+
+  // Same as `HandleAudioInputParametersChange` but should be called when audio output
+  // parameters of `RTCAudioDevice` has changed.
+  void HandleAudioOutputParametersChange();
+
+  // Notifies `ObjCAudioDeviceModule` about audio input interruption happen due to
+  // any reason so `ObjCAudioDeviceModule` is can prepare to restart of audio IO.
+  void HandleAudioInputInterrupted();
+
+  // Same as `ObjCAudioDeviceModule` but should be called when audio output
+  // is interrupted.
+  void HandleAudioOutputInterrupted();
+
+ private:
+  // Update our audio parameters if they are different from current device audio parameters
+  // Returns true when our parameters are update, false - otherwise.
+  // `ObjCAudioDeviceModule` has audio device buffer (ADB) which has audio parameters
+  // of playout & recording. The ADB is configured to work with specific sample rate & channel
+  // count. `ObjCAudioDeviceModule` stores audio parameters which were used to configure ADB in the
+  // fields `playout_parameters_` and `recording_parameters_`.
+  // `RTCAudioDevice` protocol has its own audio parameters exposed as individual properties.
+  // `RTCAudioDevice` audio parameters might change when playout/recording is already in progress,
+  // for example, when device is switched. `RTCAudioDevice` audio parameters must be kept in sync
+  // with ADB audio parameters. This method is invoked when `RTCAudioDevice` reports that it's audio
+  // parameters (`device_params`) are changed and it detects if there any difference with our
+  // current audio parameters (`params`). Our parameters are updated in case of actual change and
+  // method returns true. In case of actual change there is follow-up call to either
+  // `UpdateOutputAudioDeviceBuffer` or `UpdateInputAudioDeviceBuffer` to apply updated
+  // `playout_parameters_` or `recording_parameters_` to ADB.
+
+  bool UpdateAudioParameters(AudioParameters& params, const AudioParameters& device_params);
+
+  // Update our cached audio latency with device latency. Device latency is reported by
+  // `RTCAudioDevice` object. Whenever latency is changed, `RTCAudioDevice` is obliged to notify ADM
+  // about the change via `HandleAudioInputParametersChange` or `HandleAudioOutputParametersChange`.
+  // Current device IO latency is cached in the atomic field and used from audio IO thread
+  // to be reported to audio device buffer. It is highly recommended by Apple not to call any
+  // ObjC methods from audio IO thread, that is why implementation relies on caching latency
+  // into a field and being notified when latency is changed, which is the case when device
+  // is switched.
+  void UpdateAudioDelay(std::atomic<int>& delay_ms, const NSTimeInterval device_latency);
+
+  // Uses current `playout_parameters_` to inform the audio device buffer (ADB)
+  // about our internal audio parameters.
+  void UpdateOutputAudioDeviceBuffer();
+
+  // Uses current `record_parameters_` to inform the audio device buffer (ADB)
+  // about our internal audio parameters.
+  void UpdateInputAudioDeviceBuffer();
+
+ private:
+  id<RTC_OBJC_TYPE(RTCAudioDevice)> audio_device_;
+
+  const std::unique_ptr<TaskQueueFactory> task_queue_factory_;
+
+  // AudioDeviceBuffer is a buffer to consume audio recorded by `RTCAudioDevice`
+  // and provide audio to be played via `RTCAudioDevice`.
+  // Audio PCMs could have different sample rate and channels count, but expected
+  // to be in 16-bit integer interleaved linear PCM format.
+  // The current parameters ADB configured to work with is stored in field
+  // `playout_parameters_` for playout and `record_parameters_` for recording.
+  // These parameters and ADB must kept in sync with `RTCAudioDevice` audio parameters.
+  std::unique_ptr<AudioDeviceBuffer> audio_device_buffer_;
+
+  // Set to 1 when recording is active and 0 otherwise.
+  std::atomic<bool> recording_ = false;
+
+  // Set to 1 when playout is active and 0 otherwise.
+  std::atomic<bool> playing_ = false;
+
+  // Stores cached value of `RTCAudioDevice outputLatency` to be used from
+  // audio IO thread. Latency is updated on audio output parameters change.
+  std::atomic<int> cached_playout_delay_ms_ = 0;
+
+  // Same as `cached_playout_delay_ms_` but for audio input
+  std::atomic<int> cached_recording_delay_ms_ = 0;
+
+  // Thread that is initialized audio device module.
+  rtc::Thread* thread_;
+
+  // Ensures that methods are called from the same thread as this object is
+  // initialized on.
+  SequenceChecker thread_checker_;
+
+  // I/O audio thread checker.
+  SequenceChecker io_playout_thread_checker_;
+  SequenceChecker io_record_thread_checker_;
+
+  bool is_initialized_ RTC_GUARDED_BY(thread_checker_) = false;
+  bool is_playout_initialized_ RTC_GUARDED_BY(thread_checker_) = false;
+  bool is_recording_initialized_ RTC_GUARDED_BY(thread_checker_) = false;
+
+  // Contains audio parameters (sample rate, #channels, buffer size etc.) for
+  // the playout and recording sides.
+  AudioParameters playout_parameters_;
+  AudioParameters record_parameters_;
+
+  // `FineAudioBuffer` takes an `AudioDeviceBuffer` which delivers audio data
+  // in chunks of 10ms. `RTCAudioDevice` might deliver recorded data in
+  // chunks which are not 10ms long. `FineAudioBuffer` implements adaptation
+  // from undetermined chunk size to 10ms chunks.
+  std::unique_ptr<FineAudioBuffer> record_fine_audio_buffer_;
+
+  // Same as `record_fine_audio_buffer_` but for audio output.
+  std::unique_ptr<FineAudioBuffer> playout_fine_audio_buffer_;
+
+  // Temporary storage for recorded data.
+  rtc::BufferT<int16_t> record_audio_buffer_;
+
+  // Delegate object provided to RTCAudioDevice during initialization
+  ObjCAudioDeviceDelegate* audio_device_delegate_;
+};
+
+}  // namespace objc_adm
+
+}  // namespace webrtc
+
+#endif  // SDK_OBJC_NATIVE_SRC_OBJC_AUDIO_DEVICE_H_
diff --git a/third_party/libwebrtc/sdk/objc/native/src/objc_audio_device.mm b/third_party/libwebrtc/sdk/objc/native/src/objc_audio_device.mm
new file mode 100644
index 0000000000..d629fae20f
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/native/src/objc_audio_device.mm
@@ -0,0 +1,711 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "objc_audio_device.h"
+#include "objc_audio_device_delegate.h"
+
+#import "components/audio/RTCAudioDevice.h"
+#include "modules/audio_device/fine_audio_buffer.h"
+
+#include "api/task_queue/default_task_queue_factory.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/numerics/safe_minmax.h"
+#include "rtc_base/time_utils.h"
+
+namespace {
+
+webrtc::AudioParameters RecordParameters(id<RTC_OBJC_TYPE(RTCAudioDevice)> audio_device) {
+  const double sample_rate = static_cast<int>([audio_device deviceInputSampleRate]);
+  const size_t channels = static_cast<size_t>([audio_device inputNumberOfChannels]);
+  const size_t frames_per_buffer =
+      static_cast<size_t>(sample_rate * [audio_device inputIOBufferDuration] + .5);
+  return webrtc::AudioParameters(sample_rate, channels, frames_per_buffer);
+}
+
+webrtc::AudioParameters PlayoutParameters(id<RTC_OBJC_TYPE(RTCAudioDevice)> audio_device) {
+  const double sample_rate = static_cast<int>([audio_device deviceOutputSampleRate]);
+  const size_t channels = static_cast<size_t>([audio_device outputNumberOfChannels]);
+  const size_t frames_per_buffer =
+      static_cast<size_t>(sample_rate * [audio_device outputIOBufferDuration] + .5);
+  return webrtc::AudioParameters(sample_rate, channels, frames_per_buffer);
+}
+
+}  // namespace
+
+namespace webrtc {
+namespace objc_adm {
+
+ObjCAudioDeviceModule::ObjCAudioDeviceModule(id<RTC_OBJC_TYPE(RTCAudioDevice)> audio_device)
+    : audio_device_(audio_device), task_queue_factory_(CreateDefaultTaskQueueFactory()) {
+  RTC_DLOG_F(LS_VERBOSE) << "";
+  RTC_DCHECK(audio_device_);
+  thread_checker_.Detach();
+  io_playout_thread_checker_.Detach();
+  io_record_thread_checker_.Detach();
+}
+
+ObjCAudioDeviceModule::~ObjCAudioDeviceModule() {
+  RTC_DLOG_F(LS_VERBOSE) << "";
+}
+
+int32_t ObjCAudioDeviceModule::RegisterAudioCallback(AudioTransport* audioCallback) {
+  RTC_DLOG_F(LS_VERBOSE) << "";
+  RTC_DCHECK(audio_device_buffer_);
+  return audio_device_buffer_->RegisterAudioCallback(audioCallback);
+}
+
+int32_t ObjCAudioDeviceModule::Init() {
+  RTC_DLOG_F(LS_VERBOSE) << "";
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+
+  if (Initialized()) {
+    RTC_LOG_F(LS_INFO) << "Already initialized";
+    return 0;
+  }
+  io_playout_thread_checker_.Detach();
+  io_record_thread_checker_.Detach();
+
+  thread_ = rtc::Thread::Current();
+  audio_device_buffer_.reset(new webrtc::AudioDeviceBuffer(task_queue_factory_.get()));
+
+  if (![audio_device_ isInitialized]) {
+    if (audio_device_delegate_ == nil) {
+      audio_device_delegate_ = [[ObjCAudioDeviceDelegate alloc]
+          initWithAudioDeviceModule:rtc::scoped_refptr<ObjCAudioDeviceModule>(this)
+                  audioDeviceThread:thread_];
+    }
+
+    if (![audio_device_ initializeWithDelegate:audio_device_delegate_]) {
+      RTC_LOG_F(LS_WARNING) << "Failed to initialize audio device";
+      [audio_device_delegate_ resetAudioDeviceModule];
+      audio_device_delegate_ = nil;
+      return -1;
+    }
+  }
+
+  playout_parameters_.reset([audio_device_delegate_ preferredOutputSampleRate], 1);
+  UpdateOutputAudioDeviceBuffer();
+
+  record_parameters_.reset([audio_device_delegate_ preferredInputSampleRate], 1);
+  UpdateInputAudioDeviceBuffer();
+
+  is_initialized_ = true;
+
+  RTC_LOG_F(LS_INFO) << "Did initialize";
+  return 0;
+}
+
+int32_t ObjCAudioDeviceModule::Terminate() {
+  RTC_DLOG_F(LS_VERBOSE) << "";
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+
+  if (!Initialized()) {
+    RTC_LOG_F(LS_INFO) << "Not initialized";
+    return 0;
+  }
+
+  if ([audio_device_ isInitialized]) {
+    if (![audio_device_ terminateDevice]) {
+      RTC_LOG_F(LS_ERROR) << "Failed to terminate audio device";
+      return -1;
+    }
+  }
+
+  if (audio_device_delegate_ != nil) {
+    [audio_device_delegate_ resetAudioDeviceModule];
+    audio_device_delegate_ = nil;
+  }
+
+  is_initialized_ = false;
+  is_playout_initialized_ = false;
+  is_recording_initialized_ = false;
+  thread_ = nullptr;
+
+  RTC_LOG_F(LS_INFO) << "Did terminate";
+  return 0;
+}
+
+bool ObjCAudioDeviceModule::Initialized() const {
+  RTC_DLOG_F(LS_VERBOSE) << "";
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+  return is_initialized_ && [audio_device_ isInitialized];
+}
+
+int32_t ObjCAudioDeviceModule::PlayoutIsAvailable(bool* available) {
+  RTC_DLOG_F(LS_VERBOSE) << "";
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+  *available = Initialized();
+  return 0;
+}
+
+bool ObjCAudioDeviceModule::PlayoutIsInitialized() const {
+  RTC_DLOG_F(LS_VERBOSE) << "";
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+  return Initialized() && is_playout_initialized_ && [audio_device_ isPlayoutInitialized];
+}
+
+int32_t ObjCAudioDeviceModule::InitPlayout() {
+  RTC_DLOG_F(LS_VERBOSE) << "";
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+  if (!Initialized()) {
+    return -1;
+  }
+  if (PlayoutIsInitialized()) {
+    return 0;
+  }
+  RTC_DCHECK(!playing_.load());
+
+  if (![audio_device_ isPlayoutInitialized]) {
+    if (![audio_device_ initializePlayout]) {
+      RTC_LOG_F(LS_ERROR) << "Failed to initialize audio device playout";
+      return -1;
+    }
+  }
+
+  if (UpdateAudioParameters(playout_parameters_, PlayoutParameters(audio_device_))) {
+    UpdateOutputAudioDeviceBuffer();
+  }
+
+  is_playout_initialized_ = true;
+  RTC_LOG_F(LS_INFO) << "Did initialize playout";
+  return 0;
+}
+
+bool ObjCAudioDeviceModule::Playing() const {
+  RTC_DLOG_F(LS_VERBOSE) << "";
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+  return playing_.load() && [audio_device_ isPlaying];
+}
+
+int32_t ObjCAudioDeviceModule::StartPlayout() {
+  RTC_DLOG_F(LS_VERBOSE) << "";
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+  if (!PlayoutIsInitialized()) {
+    return -1;
+  }
+  if (Playing()) {
+    return 0;
+  }
+
+  audio_device_buffer_->StartPlayout();
+  if (playout_fine_audio_buffer_) {
+    playout_fine_audio_buffer_->ResetPlayout();
+  }
+  if (![audio_device_ startPlayout]) {
+    RTC_LOG_F(LS_ERROR) << "Failed to start audio device playout";
+    return -1;
+  }
+  playing_.store(true, std::memory_order_release);
+  RTC_LOG_F(LS_INFO) << "Did start playout";
+  return 0;
+}
+
+int32_t ObjCAudioDeviceModule::StopPlayout() {
+  RTC_DLOG_F(LS_VERBOSE) << "";
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+
+  if (![audio_device_ stopPlayout]) {
+    RTC_LOG_F(LS_WARNING) << "Failed to stop playout";
+    return -1;
+  }
+
+  audio_device_buffer_->StopPlayout();
+  playing_.store(false, std::memory_order_release);
+  RTC_LOG_F(LS_INFO) << "Did stop playout";
+  return 0;
+}
+
+int32_t ObjCAudioDeviceModule::PlayoutDelay(uint16_t* delayMS) const {
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+  *delayMS = static_cast<uint16_t>(rtc::SafeClamp<int>(
+      cached_playout_delay_ms_.load(), 0, std::numeric_limits<uint16_t>::max()));
+  return 0;
+}
+
+int32_t ObjCAudioDeviceModule::RecordingIsAvailable(bool* available) {
+  RTC_DLOG_F(LS_VERBOSE) << "";
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+  *available = Initialized();
+  return 0;
+}
+
+bool ObjCAudioDeviceModule::RecordingIsInitialized() const {
+  RTC_DLOG_F(LS_VERBOSE) << "";
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+  return Initialized() && is_recording_initialized_ && [audio_device_ isRecordingInitialized];
+}
+
+int32_t ObjCAudioDeviceModule::InitRecording() {
+  RTC_DLOG_F(LS_VERBOSE) << "";
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+  if (!Initialized()) {
+    return -1;
+  }
+  if (RecordingIsInitialized()) {
+    return 0;
+  }
+  RTC_DCHECK(!recording_.load());
+
+  if (![audio_device_ isRecordingInitialized]) {
+    if (![audio_device_ initializeRecording]) {
+      RTC_LOG_F(LS_ERROR) << "Failed to initialize audio device recording";
+      return -1;
+    }
+  }
+
+  if (UpdateAudioParameters(record_parameters_, RecordParameters(audio_device_))) {
+    UpdateInputAudioDeviceBuffer();
+  }
+
+  is_recording_initialized_ = true;
+  RTC_LOG_F(LS_INFO) << "Did initialize recording";
+  return 0;
+}
+
+bool ObjCAudioDeviceModule::Recording() const {
+  RTC_DLOG_F(LS_VERBOSE) << "";
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+  return recording_.load() && [audio_device_ isRecording];
+}
+
+int32_t ObjCAudioDeviceModule::StartRecording() {
+  RTC_DLOG_F(LS_VERBOSE) << "";
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+  if (!RecordingIsInitialized()) {
+    return -1;
+  }
+  if (Recording()) {
+    return 0;
+  }
+
+  audio_device_buffer_->StartRecording();
+  if (record_fine_audio_buffer_) {
+    record_fine_audio_buffer_->ResetRecord();
+  }
+
+  if (![audio_device_ startRecording]) {
+    RTC_LOG_F(LS_ERROR) << "Failed to start audio device recording";
+    return -1;
+  }
+  recording_.store(true, std::memory_order_release);
+  RTC_LOG_F(LS_INFO) << "Did start recording";
+  return 0;
+}
+
+int32_t ObjCAudioDeviceModule::StopRecording() {
+  RTC_DLOG_F(LS_VERBOSE) << "";
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+
+  if (![audio_device_ stopRecording]) {
+    RTC_LOG_F(LS_WARNING) << "Failed to stop recording";
+    return -1;
+  }
+  audio_device_buffer_->StopRecording();
+  recording_.store(false, std::memory_order_release);
+  RTC_LOG_F(LS_INFO) << "Did stop recording";
+  return 0;
+}
+
+#if defined(WEBRTC_IOS)
+
+int ObjCAudioDeviceModule::GetPlayoutAudioParameters(AudioParameters* params) const {
+  RTC_DLOG_F(LS_VERBOSE) << "";
+  RTC_DCHECK(playout_parameters_.is_valid());
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+  *params = playout_parameters_;
+  return 0;
+}
+
+int ObjCAudioDeviceModule::GetRecordAudioParameters(AudioParameters* params) const {
+  RTC_DLOG_F(LS_VERBOSE) << "";
+  RTC_DCHECK(record_parameters_.is_valid());
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+  *params = record_parameters_;
+  return 0;
+}
+
+#endif  // WEBRTC_IOS
+
+void ObjCAudioDeviceModule::UpdateOutputAudioDeviceBuffer() {
+  RTC_DLOG_F(LS_VERBOSE) << "";
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+  RTC_DCHECK(audio_device_buffer_) << "AttachAudioBuffer must be called first";
+
+  RTC_DCHECK_GT(playout_parameters_.sample_rate(), 0);
+  RTC_DCHECK(playout_parameters_.channels() == 1 || playout_parameters_.channels() == 2);
+
+  audio_device_buffer_->SetPlayoutSampleRate(playout_parameters_.sample_rate());
+  audio_device_buffer_->SetPlayoutChannels(playout_parameters_.channels());
+  playout_fine_audio_buffer_.reset(new FineAudioBuffer(audio_device_buffer_.get()));
+}
+
+void ObjCAudioDeviceModule::UpdateInputAudioDeviceBuffer() {
+  RTC_DLOG_F(LS_VERBOSE) << "";
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+  RTC_DCHECK(audio_device_buffer_) << "AttachAudioBuffer must be called first";
+
+  RTC_DCHECK_GT(record_parameters_.sample_rate(), 0);
+  RTC_DCHECK(record_parameters_.channels() == 1 || record_parameters_.channels() == 2);
+
+  audio_device_buffer_->SetRecordingSampleRate(record_parameters_.sample_rate());
+  audio_device_buffer_->SetRecordingChannels(record_parameters_.channels());
+  record_fine_audio_buffer_.reset(new FineAudioBuffer(audio_device_buffer_.get()));
+}
+
+void ObjCAudioDeviceModule::UpdateAudioDelay(std::atomic<int>& delay_ms,
+                                             const NSTimeInterval device_latency) {
+  RTC_DLOG_F(LS_VERBOSE) << "";
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+  int latency_ms = static_cast<int>(rtc::kNumMillisecsPerSec * device_latency);
+  if (latency_ms <= 0) {
+    return;
+  }
+  const int old_latency_ms = delay_ms.exchange(latency_ms);
+  if (old_latency_ms != latency_ms) {
+    RTC_LOG_F(LS_INFO) << "Did change audio IO latency from: " << old_latency_ms
+                       << " ms to: " << latency_ms << " ms";
+  }
+}
+
+bool ObjCAudioDeviceModule::UpdateAudioParameters(AudioParameters& params,
+                                                  const AudioParameters& device_params) {
+  RTC_DLOG_F(LS_VERBOSE) << "";
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+  if (!device_params.is_complete()) {
+    RTC_LOG_F(LS_INFO) << "Device params are incomplete: " << device_params.ToString();
+    return false;
+  }
+  if (params.channels() == device_params.channels() &&
+      params.frames_per_buffer() == device_params.frames_per_buffer() &&
+      params.sample_rate() == device_params.sample_rate()) {
+    RTC_LOG_F(LS_INFO) << "Device params: " << device_params.ToString()
+                       << " are not different from: " << params.ToString();
+    return false;
+  }
+
+  RTC_LOG_F(LS_INFO) << "Audio params will be changed from: " << params.ToString()
+                     << " to: " << device_params.ToString();
+  params.reset(
+      device_params.sample_rate(), device_params.channels(), device_params.frames_per_buffer());
+  return true;
+}
+
+OSStatus ObjCAudioDeviceModule::OnDeliverRecordedData(
+    AudioUnitRenderActionFlags* flags,
+    const AudioTimeStamp* time_stamp,
+    NSInteger bus_number,
+    UInt32 num_frames,
+    const AudioBufferList* io_data,
+    void* render_context,
+    RTC_OBJC_TYPE(RTCAudioDeviceRenderRecordedDataBlock) render_block) {
+  RTC_DCHECK_RUN_ON(&io_record_thread_checker_);
+  OSStatus result = noErr;
+  // Simply return if recording is not enabled.
+  if (!recording_.load()) return result;
+
+  if (io_data != nullptr) {
+    // AudioBuffer already fullfilled with audio data
+    RTC_DCHECK_EQ(1, io_data->mNumberBuffers);
+    const AudioBuffer* audio_buffer = &io_data->mBuffers[0];
+    RTC_DCHECK(audio_buffer->mNumberChannels == 1 || audio_buffer->mNumberChannels == 2);
+
+    record_fine_audio_buffer_->DeliverRecordedData(
+        rtc::ArrayView<const int16_t>(static_cast<int16_t*>(audio_buffer->mData), num_frames),
+        cached_recording_delay_ms_.load());
+    return noErr;
+  }
+  RTC_DCHECK(render_block != nullptr) << "Either io_data or render_block must be provided";
+
+  // Set the size of our own audio buffer and clear it first to avoid copying
+  // in combination with potential reallocations.
+  // On real iOS devices, the size will only be set once (at first callback).
+  const int channels_count = record_parameters_.channels();
+  record_audio_buffer_.Clear();
+  record_audio_buffer_.SetSize(num_frames * channels_count);
+
+  // Allocate AudioBuffers to be used as storage for the received audio.
+  // The AudioBufferList structure works as a placeholder for the
+  // AudioBuffer structure, which holds a pointer to the actual data buffer
+  // in `record_audio_buffer_`. Recorded audio will be rendered into this memory
+  // at each input callback when calling `render_block`.
+  AudioBufferList audio_buffer_list;
+  audio_buffer_list.mNumberBuffers = 1;
+  AudioBuffer* audio_buffer = &audio_buffer_list.mBuffers[0];
+  audio_buffer->mNumberChannels = channels_count;
+  audio_buffer->mDataByteSize =
+      record_audio_buffer_.size() * sizeof(decltype(record_audio_buffer_)::value_type);
+  audio_buffer->mData = reinterpret_cast<int8_t*>(record_audio_buffer_.data());
+
+  // Obtain the recorded audio samples by initiating a rendering cycle into own buffer.
+  result =
+      render_block(flags, time_stamp, bus_number, num_frames, &audio_buffer_list, render_context);
+  if (result != noErr) {
+    RTC_LOG_F(LS_ERROR) << "Failed to render audio: " << result;
+    return result;
+  }
+
+  // Get a pointer to the recorded audio and send it to the WebRTC ADB.
+  // Use the FineAudioBuffer instance to convert between native buffer size
+  // and the 10ms buffer size used by WebRTC.
+  record_fine_audio_buffer_->DeliverRecordedData(record_audio_buffer_,
+                                                 cached_recording_delay_ms_.load());
+  return noErr;
+}
+
+OSStatus ObjCAudioDeviceModule::OnGetPlayoutData(AudioUnitRenderActionFlags* flags,
+                                                 const AudioTimeStamp* time_stamp,
+                                                 NSInteger bus_number,
+                                                 UInt32 num_frames,
+                                                 AudioBufferList* io_data) {
+  RTC_DCHECK_RUN_ON(&io_playout_thread_checker_);
+  // Verify 16-bit, noninterleaved mono or stereo PCM signal format.
+  RTC_DCHECK_EQ(1, io_data->mNumberBuffers);
+  AudioBuffer* audio_buffer = &io_data->mBuffers[0];
+  RTC_DCHECK(audio_buffer->mNumberChannels == 1 || audio_buffer->mNumberChannels == 2);
+  RTC_DCHECK_EQ(audio_buffer->mDataByteSize,
+                sizeof(int16_t) * num_frames * audio_buffer->mNumberChannels);
+
+  // Produce silence and give player a hint about it if playout is not
+  // activated.
+  if (!playing_.load()) {
+    *flags |= kAudioUnitRenderAction_OutputIsSilence;
+    memset(static_cast<int8_t*>(audio_buffer->mData), 0, audio_buffer->mDataByteSize);
+    return noErr;
+  }
+
+  // Read decoded 16-bit PCM samples from WebRTC into the
+  // `io_data` destination buffer.
+  playout_fine_audio_buffer_->GetPlayoutData(
+      rtc::ArrayView<int16_t>(static_cast<int16_t*>(audio_buffer->mData),
+                              num_frames * audio_buffer->mNumberChannels),
+      cached_playout_delay_ms_.load());
+
+  return noErr;
+}
+
+void ObjCAudioDeviceModule::HandleAudioInputInterrupted() {
+  RTC_DLOG_F(LS_VERBOSE) << "";
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+  io_record_thread_checker_.Detach();
+}
+
+void ObjCAudioDeviceModule::HandleAudioOutputInterrupted() {
+  RTC_DLOG_F(LS_VERBOSE) << "";
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+  io_playout_thread_checker_.Detach();
+}
+
+void ObjCAudioDeviceModule::HandleAudioInputParametersChange() {
+  RTC_DLOG_F(LS_VERBOSE) << "";
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+
+  if (UpdateAudioParameters(record_parameters_, RecordParameters(audio_device_))) {
+    UpdateInputAudioDeviceBuffer();
+  }
+
+  UpdateAudioDelay(cached_recording_delay_ms_, [audio_device_ inputLatency]);
+}
+
+void ObjCAudioDeviceModule::HandleAudioOutputParametersChange() {
+  RTC_DLOG_F(LS_VERBOSE) << "";
+  RTC_DCHECK_RUN_ON(&thread_checker_);
+
+  if (UpdateAudioParameters(playout_parameters_, PlayoutParameters(audio_device_))) {
+    UpdateOutputAudioDeviceBuffer();
+  }
+
+  UpdateAudioDelay(cached_playout_delay_ms_, [audio_device_ outputLatency]);
+}
+
+#pragma mark - Not implemented/Not relevant methods from AudioDeviceModule
+
+int32_t ObjCAudioDeviceModule::ActiveAudioLayer(AudioLayer* audioLayer) const {
+  return -1;
+}
+
+int16_t ObjCAudioDeviceModule::PlayoutDevices() {
+  return 0;
+}
+
+int16_t ObjCAudioDeviceModule::RecordingDevices() {
+  return 0;
+}
+
+int32_t ObjCAudioDeviceModule::PlayoutDeviceName(uint16_t index,
+                                                 char name[kAdmMaxDeviceNameSize],
+                                                 char guid[kAdmMaxGuidSize]) {
+  return -1;
+}
+
+int32_t ObjCAudioDeviceModule::RecordingDeviceName(uint16_t index,
+                                                   char name[kAdmMaxDeviceNameSize],
+                                                   char guid[kAdmMaxGuidSize]) {
+  return -1;
+}
+
+int32_t ObjCAudioDeviceModule::SetPlayoutDevice(uint16_t index) {
+  return 0;
+}
+
+int32_t ObjCAudioDeviceModule::SetPlayoutDevice(WindowsDeviceType device) {
+  return -1;
+}
+
+int32_t ObjCAudioDeviceModule::SetRecordingDevice(uint16_t index) {
+  return 0;
+}
+
+int32_t ObjCAudioDeviceModule::SetRecordingDevice(WindowsDeviceType device) {
+  return -1;
+}
+
+int32_t ObjCAudioDeviceModule::InitSpeaker() {
+  return 0;
+}
+
+bool ObjCAudioDeviceModule::SpeakerIsInitialized() const {
+  return true;
+}
+
+int32_t ObjCAudioDeviceModule::InitMicrophone() {
+  return 0;
+}
+
+bool ObjCAudioDeviceModule::MicrophoneIsInitialized() const {
+  return true;
+}
+
+int32_t ObjCAudioDeviceModule::SpeakerVolumeIsAvailable(bool* available) {
+  *available = false;
+  return 0;
+}
+
+int32_t ObjCAudioDeviceModule::SetSpeakerVolume(uint32_t volume) {
+  return -1;
+}
+
+int32_t ObjCAudioDeviceModule::SpeakerVolume(uint32_t* volume) const {
+  return -1;
+}
+
+int32_t ObjCAudioDeviceModule::MaxSpeakerVolume(uint32_t* maxVolume) const {
+  return -1;
+}
+
+int32_t ObjCAudioDeviceModule::MinSpeakerVolume(uint32_t* minVolume) const {
+  return -1;
+}
+
+int32_t ObjCAudioDeviceModule::SpeakerMuteIsAvailable(bool* available) {
+  *available = false;
+  return 0;
+}
+
+int32_t ObjCAudioDeviceModule::SetSpeakerMute(bool enable) {
+  return -1;
+}
+
+int32_t ObjCAudioDeviceModule::SpeakerMute(bool* enabled) const {
+  return -1;
+}
+
+int32_t ObjCAudioDeviceModule::MicrophoneMuteIsAvailable(bool* available) {
+  *available = false;
+  return 0;
+}
+
+int32_t ObjCAudioDeviceModule::SetMicrophoneMute(bool enable) {
+  return -1;
+}
+
+int32_t ObjCAudioDeviceModule::MicrophoneMute(bool* enabled) const {
+  return -1;
+}
+
+int32_t ObjCAudioDeviceModule::MicrophoneVolumeIsAvailable(bool* available) {
+  *available = false;
+  return 0;
+}
+
+int32_t ObjCAudioDeviceModule::SetMicrophoneVolume(uint32_t volume) {
+  return -1;
+}
+
+int32_t ObjCAudioDeviceModule::MicrophoneVolume(uint32_t* volume) const {
+  return -1;
+}
+
+int32_t ObjCAudioDeviceModule::MaxMicrophoneVolume(uint32_t* maxVolume) const {
+  return -1;
+}
+
+int32_t ObjCAudioDeviceModule::MinMicrophoneVolume(uint32_t* minVolume) const {
+  return -1;
+}
+
+int32_t ObjCAudioDeviceModule::StereoPlayoutIsAvailable(bool* available) const {
+  *available = false;
+  return 0;
+}
+
+int32_t ObjCAudioDeviceModule::SetStereoPlayout(bool enable) {
+  return -1;
+}
+
+int32_t ObjCAudioDeviceModule::StereoPlayout(bool* enabled) const {
+  *enabled = false;
+  return 0;
+}
+
+int32_t ObjCAudioDeviceModule::StereoRecordingIsAvailable(bool* available) const {
+  *available = false;
+  return 0;
+}
+
+int32_t ObjCAudioDeviceModule::SetStereoRecording(bool enable) {
+  return -1;
+}
+
+int32_t ObjCAudioDeviceModule::StereoRecording(bool* enabled) const {
+  *enabled = false;
+  return 0;
+}
+
+bool ObjCAudioDeviceModule::BuiltInAECIsAvailable() const {
+  return false;
+}
+
+int32_t ObjCAudioDeviceModule::EnableBuiltInAEC(bool enable) {
+  return 0;
+}
+
+bool ObjCAudioDeviceModule::BuiltInAGCIsAvailable() const {
+  return false;
+}
+
+int32_t ObjCAudioDeviceModule::EnableBuiltInAGC(bool enable) {
+  return 0;
+}
+
+bool ObjCAudioDeviceModule::BuiltInNSIsAvailable() const {
+  return false;
+}
+
+int32_t ObjCAudioDeviceModule::EnableBuiltInNS(bool enable) {
+  return 0;
+}
+
+int32_t ObjCAudioDeviceModule::GetPlayoutUnderrunCount() const {
+  return -1;
+}
+
+}  // namespace objc_adm
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/sdk/objc/native/src/objc_audio_device_delegate.h b/third_party/libwebrtc/sdk/objc/native/src/objc_audio_device_delegate.h
new file mode 100644
index 0000000000..3af079dad9
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/native/src/objc_audio_device_delegate.h
@@ -0,0 +1,35 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef SDK_OBJC_NATIVE_SRC_OBJC_AUDIO_DEVICE_DELEGATE_H_
+#define SDK_OBJC_NATIVE_SRC_OBJC_AUDIO_DEVICE_DELEGATE_H_
+
+#include "api/scoped_refptr.h"
+#include "rtc_base/thread.h"
+
+#import "components/audio/RTCAudioDevice.h"
+
+namespace webrtc {
+namespace objc_adm {
+class ObjCAudioDeviceModule;
+}  // namespace objc_adm
+}  // namespace webrtc
+
+@interface ObjCAudioDeviceDelegate : NSObject <RTC_OBJC_TYPE (RTCAudioDeviceDelegate)>
+
+- (instancetype)initWithAudioDeviceModule:
+                    (rtc::scoped_refptr<webrtc::objc_adm::ObjCAudioDeviceModule>)audioDeviceModule
+                        audioDeviceThread:(rtc::Thread*)thread;
+
+- (void)resetAudioDeviceModule;
+
+@end
+
+#endif  // SDK_OBJC_NATIVE_SRC_OBJC_AUDIO_DEVICE_DELEGATE_H_
diff --git a/third_party/libwebrtc/sdk/objc/native/src/objc_audio_device_delegate.mm b/third_party/libwebrtc/sdk/objc/native/src/objc_audio_device_delegate.mm
new file mode 100644
index 0000000000..156d6326a4
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/native/src/objc_audio_device_delegate.mm
@@ -0,0 +1,194 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <AudioUnit/AudioUnit.h>
+#import <Foundation/Foundation.h>
+
+#import "objc_audio_device.h"
+#import "objc_audio_device_delegate.h"
+
+#include "api/make_ref_counted.h"
+#include "api/ref_counted_base.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/thread.h"
+
+namespace {
+
+constexpr double kPreferredInputSampleRate = 48000.0;
+constexpr double kPreferredOutputSampleRate = 48000.0;
+
+// WebRTC processes audio in chunks of 10ms. Preferring 20ms audio chunks
+// is a compromize between performance and power consumption.
+constexpr NSTimeInterval kPeferredInputIOBufferDuration = 0.02;
+constexpr NSTimeInterval kPeferredOutputIOBufferDuration = 0.02;
+
+class AudioDeviceDelegateImpl final : public rtc::RefCountedNonVirtual<AudioDeviceDelegateImpl> {
+ public:
+  AudioDeviceDelegateImpl(
+      rtc::scoped_refptr<webrtc::objc_adm::ObjCAudioDeviceModule> audio_device_module,
+      rtc::Thread* thread)
+      : audio_device_module_(audio_device_module), thread_(thread) {
+    RTC_DCHECK(audio_device_module_);
+    RTC_DCHECK(thread_);
+  }
+
+  webrtc::objc_adm::ObjCAudioDeviceModule* audio_device_module() const {
+    return audio_device_module_.get();
+  }
+
+  rtc::Thread* thread() const { return thread_; }
+
+  void reset_audio_device_module() { audio_device_module_ = nullptr; }
+
+ private:
+  rtc::scoped_refptr<webrtc::objc_adm::ObjCAudioDeviceModule> audio_device_module_;
+  rtc::Thread* thread_;
+};
+
+}  // namespace
+
+@implementation ObjCAudioDeviceDelegate {
+  rtc::scoped_refptr<AudioDeviceDelegateImpl> impl_;
+}
+
+@synthesize getPlayoutData = getPlayoutData_;
+
+@synthesize deliverRecordedData = deliverRecordedData_;
+
+@synthesize preferredInputSampleRate = preferredInputSampleRate_;
+
+@synthesize preferredInputIOBufferDuration = preferredInputIOBufferDuration_;
+
+@synthesize preferredOutputSampleRate = preferredOutputSampleRate_;
+
+@synthesize preferredOutputIOBufferDuration = preferredOutputIOBufferDuration_;
+
+- (instancetype)initWithAudioDeviceModule:
+                    (rtc::scoped_refptr<webrtc::objc_adm::ObjCAudioDeviceModule>)audioDeviceModule
+                        audioDeviceThread:(rtc::Thread*)thread {
+  RTC_DCHECK_RUN_ON(thread);
+  if (self = [super init]) {
+    impl_ = rtc::make_ref_counted<AudioDeviceDelegateImpl>(audioDeviceModule, thread);
+    preferredInputSampleRate_ = kPreferredInputSampleRate;
+    preferredInputIOBufferDuration_ = kPeferredInputIOBufferDuration;
+    preferredOutputSampleRate_ = kPreferredOutputSampleRate;
+    preferredOutputIOBufferDuration_ = kPeferredOutputIOBufferDuration;
+
+    rtc::scoped_refptr<AudioDeviceDelegateImpl> playout_delegate = impl_;
+    getPlayoutData_ = ^OSStatus(AudioUnitRenderActionFlags* _Nonnull actionFlags,
+                                const AudioTimeStamp* _Nonnull timestamp,
+                                NSInteger inputBusNumber,
+                                UInt32 frameCount,
+                                AudioBufferList* _Nonnull outputData) {
+      webrtc::objc_adm::ObjCAudioDeviceModule* audio_device =
+          playout_delegate->audio_device_module();
+      if (audio_device) {
+        return audio_device->OnGetPlayoutData(
+            actionFlags, timestamp, inputBusNumber, frameCount, outputData);
+      } else {
+        *actionFlags |= kAudioUnitRenderAction_OutputIsSilence;
+        RTC_LOG(LS_VERBOSE) << "No alive audio device";
+        return noErr;
+      }
+    };
+
+    rtc::scoped_refptr<AudioDeviceDelegateImpl> record_delegate = impl_;
+    deliverRecordedData_ =
+        ^OSStatus(AudioUnitRenderActionFlags* _Nonnull actionFlags,
+                  const AudioTimeStamp* _Nonnull timestamp,
+                  NSInteger inputBusNumber,
+                  UInt32 frameCount,
+                  const AudioBufferList* _Nullable inputData,
+                  void* renderContext,
+                  RTC_OBJC_TYPE(RTCAudioDeviceRenderRecordedDataBlock) _Nullable renderBlock) {
+          webrtc::objc_adm::ObjCAudioDeviceModule* audio_device =
+              record_delegate->audio_device_module();
+          if (audio_device) {
+            return audio_device->OnDeliverRecordedData(actionFlags,
+                                                       timestamp,
+                                                       inputBusNumber,
+                                                       frameCount,
+                                                       inputData,
+                                                       renderContext,
+                                                       renderBlock);
+          } else {
+            RTC_LOG(LS_VERBOSE) << "No alive audio device";
+            return noErr;
+          }
+        };
+  }
+  return self;
+}
+
+- (void)notifyAudioInputParametersChange {
+  RTC_DCHECK_RUN_ON(impl_->thread());
+  webrtc::objc_adm::ObjCAudioDeviceModule* audio_device_module = impl_->audio_device_module();
+  if (audio_device_module) {
+    audio_device_module->HandleAudioInputParametersChange();
+  }
+}
+
+- (void)notifyAudioOutputParametersChange {
+  RTC_DCHECK_RUN_ON(impl_->thread());
+  webrtc::objc_adm::ObjCAudioDeviceModule* audio_device_module = impl_->audio_device_module();
+  if (audio_device_module) {
+    audio_device_module->HandleAudioOutputParametersChange();
+  }
+}
+
+- (void)notifyAudioInputInterrupted {
+  RTC_DCHECK_RUN_ON(impl_->thread());
+  webrtc::objc_adm::ObjCAudioDeviceModule* audio_device_module = impl_->audio_device_module();
+  if (audio_device_module) {
+    audio_device_module->HandleAudioInputInterrupted();
+  }
+}
+
+- (void)notifyAudioOutputInterrupted {
+  RTC_DCHECK_RUN_ON(impl_->thread());
+  webrtc::objc_adm::ObjCAudioDeviceModule* audio_device_module = impl_->audio_device_module();
+  if (audio_device_module) {
+    audio_device_module->HandleAudioOutputInterrupted();
+  }
+}
+
+- (void)dispatchAsync:(dispatch_block_t)block {
+  rtc::Thread* thread = impl_->thread();
+  RTC_DCHECK(thread);
+  thread->PostTask([block] {
+    @autoreleasepool {
+      block();
+    }
+  });
+}
+
+- (void)dispatchSync:(dispatch_block_t)block {
+  rtc::Thread* thread = impl_->thread();
+  RTC_DCHECK(thread);
+  if (thread->IsCurrent()) {
+    @autoreleasepool {
+      block();
+    }
+  } else {
+    thread->BlockingCall([block] {
+      @autoreleasepool {
+        block();
+      }
+    });
+  }
+}
+
+- (void)resetAudioDeviceModule {
+  RTC_DCHECK_RUN_ON(impl_->thread());
+  impl_->reset_audio_device_module();
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/native/src/objc_frame_buffer.h b/third_party/libwebrtc/sdk/objc/native/src/objc_frame_buffer.h
new file mode 100644
index 0000000000..944690c8bc
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/native/src/objc_frame_buffer.h
@@ -0,0 +1,56 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef SDK_OBJC_NATIVE_SRC_OBJC_FRAME_BUFFER_H_
+#define SDK_OBJC_NATIVE_SRC_OBJC_FRAME_BUFFER_H_
+
+#import <CoreVideo/CoreVideo.h>
+
+#import "base/RTCMacros.h"
+
+#include "common_video/include/video_frame_buffer.h"
+
+@protocol RTC_OBJC_TYPE
+(RTCVideoFrameBuffer);
+
+namespace webrtc {
+
+class ObjCFrameBuffer : public VideoFrameBuffer {
+ public:
+  explicit ObjCFrameBuffer(id<RTC_OBJC_TYPE(RTCVideoFrameBuffer)>);
+  ~ObjCFrameBuffer() override;
+
+  Type type() const override;
+
+  int width() const override;
+  int height() const override;
+
+  rtc::scoped_refptr<I420BufferInterface> ToI420() override;
+  rtc::scoped_refptr<VideoFrameBuffer> CropAndScale(int offset_x,
+                                                    int offset_y,
+                                                    int crop_width,
+                                                    int crop_height,
+                                                    int scaled_width,
+                                                    int scaled_height) override;
+
+  id<RTC_OBJC_TYPE(RTCVideoFrameBuffer)> wrapped_frame_buffer() const;
+
+ private:
+  id<RTC_OBJC_TYPE(RTCVideoFrameBuffer)> frame_buffer_;
+  int width_;
+  int height_;
+};
+
+id<RTC_OBJC_TYPE(RTCVideoFrameBuffer)> ToObjCVideoFrameBuffer(
+    const rtc::scoped_refptr<VideoFrameBuffer>& buffer);
+
+}  // namespace webrtc
+
+#endif  // SDK_OBJC_NATIVE_SRC_OBJC_FRAME_BUFFER_H_
diff --git a/third_party/libwebrtc/sdk/objc/native/src/objc_frame_buffer.mm b/third_party/libwebrtc/sdk/objc/native/src/objc_frame_buffer.mm
new file mode 100644
index 0000000000..00e4b4be85
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/native/src/objc_frame_buffer.mm
@@ -0,0 +1,107 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "sdk/objc/native/src/objc_frame_buffer.h"
+
+#include "api/make_ref_counted.h"
+#import "base/RTCVideoFrameBuffer.h"
+#import "sdk/objc/api/video_frame_buffer/RTCNativeI420Buffer+Private.h"
+
+namespace webrtc {
+
+namespace {
+
+/** ObjCFrameBuffer that conforms to I420BufferInterface by wrapping RTC_OBJC_TYPE(RTCI420Buffer) */
+class ObjCI420FrameBuffer : public I420BufferInterface {
+ public:
+  explicit ObjCI420FrameBuffer(id<RTC_OBJC_TYPE(RTCI420Buffer)> frame_buffer)
+      : frame_buffer_(frame_buffer), width_(frame_buffer.width), height_(frame_buffer.height) {}
+  ~ObjCI420FrameBuffer() override {}
+
+  int width() const override { return width_; }
+
+  int height() const override { return height_; }
+
+  const uint8_t* DataY() const override { return frame_buffer_.dataY; }
+
+  const uint8_t* DataU() const override { return frame_buffer_.dataU; }
+
+  const uint8_t* DataV() const override { return frame_buffer_.dataV; }
+
+  int StrideY() const override { return frame_buffer_.strideY; }
+
+  int StrideU() const override { return frame_buffer_.strideU; }
+
+  int StrideV() const override { return frame_buffer_.strideV; }
+
+ private:
+  id<RTC_OBJC_TYPE(RTCI420Buffer)> frame_buffer_;
+  int width_;
+  int height_;
+};
+
+}  // namespace
+
+ObjCFrameBuffer::ObjCFrameBuffer(id<RTC_OBJC_TYPE(RTCVideoFrameBuffer)> frame_buffer)
+    : frame_buffer_(frame_buffer), width_(frame_buffer.width), height_(frame_buffer.height) {}
+
+ObjCFrameBuffer::~ObjCFrameBuffer() {}
+
+VideoFrameBuffer::Type ObjCFrameBuffer::type() const {
+  return Type::kNative;
+}
+
+int ObjCFrameBuffer::width() const {
+  return width_;
+}
+
+int ObjCFrameBuffer::height() const {
+  return height_;
+}
+
+rtc::scoped_refptr<I420BufferInterface> ObjCFrameBuffer::ToI420() {
+  return rtc::make_ref_counted<ObjCI420FrameBuffer>([frame_buffer_ toI420]);
+}
+
+rtc::scoped_refptr<VideoFrameBuffer> ObjCFrameBuffer::CropAndScale(int offset_x,
+                                                                   int offset_y,
+                                                                   int crop_width,
+                                                                   int crop_height,
+                                                                   int scaled_width,
+                                                                   int scaled_height) {
+  if ([frame_buffer_ respondsToSelector:@selector
+                     (cropAndScaleWith:offsetY:cropWidth:cropHeight:scaleWidth:scaleHeight:)]) {
+    return rtc::make_ref_counted<ObjCFrameBuffer>([frame_buffer_ cropAndScaleWith:offset_x
+                                                                          offsetY:offset_y
+                                                                        cropWidth:crop_width
+                                                                       cropHeight:crop_height
+                                                                       scaleWidth:scaled_width
+                                                                      scaleHeight:scaled_height]);
+  }
+
+  // Use the default implementation.
+  return VideoFrameBuffer::CropAndScale(
+      offset_x, offset_y, crop_width, crop_height, scaled_width, scaled_height);
+}
+
+id<RTC_OBJC_TYPE(RTCVideoFrameBuffer)> ObjCFrameBuffer::wrapped_frame_buffer() const {
+  return frame_buffer_;
+}
+
+id<RTC_OBJC_TYPE(RTCVideoFrameBuffer)> ToObjCVideoFrameBuffer(
+    const rtc::scoped_refptr<VideoFrameBuffer>& buffer) {
+  if (buffer->type() == VideoFrameBuffer::Type::kNative) {
+    return static_cast<ObjCFrameBuffer*>(buffer.get())->wrapped_frame_buffer();
+  } else {
+    return [[RTC_OBJC_TYPE(RTCI420Buffer) alloc] initWithFrameBuffer:buffer->ToI420()];
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/sdk/objc/native/src/objc_network_monitor.h b/third_party/libwebrtc/sdk/objc/native/src/objc_network_monitor.h
new file mode 100644
index 0000000000..709e9dfbe5
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/native/src/objc_network_monitor.h
@@ -0,0 +1,67 @@
+/*
+ *  Copyright 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef SDK_OBJC_NATIVE_SRC_OBJC_NETWORK_MONITOR_H_
+#define SDK_OBJC_NATIVE_SRC_OBJC_NETWORK_MONITOR_H_
+
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "api/field_trials_view.h"
+#include "api/sequence_checker.h"
+#include "api/task_queue/pending_task_safety_flag.h"
+#include "rtc_base/network_monitor.h"
+#include "rtc_base/network_monitor_factory.h"
+#include "rtc_base/string_utils.h"
+#include "rtc_base/thread.h"
+#include "rtc_base/thread_annotations.h"
+#include "sdk/objc/components/network/RTCNetworkMonitor+Private.h"
+#include "sdk/objc/native/src/network_monitor_observer.h"
+
+namespace webrtc {
+
+class ObjCNetworkMonitorFactory : public rtc::NetworkMonitorFactory {
+ public:
+  ObjCNetworkMonitorFactory() = default;
+  ~ObjCNetworkMonitorFactory() override = default;
+
+  rtc::NetworkMonitorInterface* CreateNetworkMonitor(
+      const FieldTrialsView& field_trials) override;
+};
+
+class ObjCNetworkMonitor : public rtc::NetworkMonitorInterface,
+                           public NetworkMonitorObserver {
+ public:
+  ObjCNetworkMonitor();
+  ~ObjCNetworkMonitor() override;
+
+  void Start() override;
+  void Stop() override;
+
+  InterfaceInfo GetInterfaceInfo(absl::string_view interface_name) override;
+
+  // NetworkMonitorObserver override.
+  // Fans out updates to observers on the correct thread.
+  void OnPathUpdate(
+      std::map<std::string, rtc::AdapterType, rtc::AbslStringViewCmp>
+          adapter_type_by_name) override;
+
+ private:
+  rtc::Thread* thread_ = nullptr;
+  bool started_ = false;
+  std::map<std::string, rtc::AdapterType, rtc::AbslStringViewCmp>
+      adapter_type_by_name_ RTC_GUARDED_BY(thread_);
+  rtc::scoped_refptr<PendingTaskSafetyFlag> safety_flag_;
+  RTCNetworkMonitor* network_monitor_ = nil;
+};
+
+}  // namespace webrtc
+
+#endif  // SDK_OBJC_NATIVE_SRC_OBJC_NETWORK_MONITOR_H_
diff --git a/third_party/libwebrtc/sdk/objc/native/src/objc_network_monitor.mm b/third_party/libwebrtc/sdk/objc/native/src/objc_network_monitor.mm
new file mode 100644
index 0000000000..535548c64c
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/native/src/objc_network_monitor.mm
@@ -0,0 +1,95 @@
+/*
+ *  Copyright 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "sdk/objc/native/src/objc_network_monitor.h"
+#include "absl/strings/string_view.h"
+
+#include <algorithm>
+
+#include "rtc_base/logging.h"
+#include "rtc_base/string_utils.h"
+
+namespace webrtc {
+
+rtc::NetworkMonitorInterface* ObjCNetworkMonitorFactory::CreateNetworkMonitor(
+    const FieldTrialsView& field_trials) {
+  return new ObjCNetworkMonitor();
+}
+
+ObjCNetworkMonitor::ObjCNetworkMonitor() {
+  safety_flag_ = PendingTaskSafetyFlag::Create();
+}
+
+ObjCNetworkMonitor::~ObjCNetworkMonitor() {
+  [network_monitor_ stop];
+  network_monitor_ = nil;
+}
+
+void ObjCNetworkMonitor::Start() {
+  if (started_) {
+    return;
+  }
+  thread_ = rtc::Thread::Current();
+  RTC_DCHECK_RUN_ON(thread_);
+  safety_flag_->SetAlive();
+  network_monitor_ = [[RTCNetworkMonitor alloc] initWithObserver:this];
+  if (network_monitor_ == nil) {
+    RTC_LOG(LS_WARNING) << "Failed to create RTCNetworkMonitor; not available on this OS?";
+  }
+  started_ = true;
+}
+
+void ObjCNetworkMonitor::Stop() {
+  RTC_DCHECK_RUN_ON(thread_);
+  if (!started_) {
+    return;
+  }
+  safety_flag_->SetNotAlive();
+  [network_monitor_ stop];
+  network_monitor_ = nil;
+  started_ = false;
+}
+
+rtc::NetworkMonitorInterface::InterfaceInfo ObjCNetworkMonitor::GetInterfaceInfo(
+    absl::string_view interface_name) {
+  RTC_DCHECK_RUN_ON(thread_);
+  if (adapter_type_by_name_.empty()) {
+    // If we have no path update, assume everything's available, because it's
+    // preferable for WebRTC to try all interfaces rather than none at all.
+    return {
+        .adapter_type = rtc::ADAPTER_TYPE_UNKNOWN,
+        .available = true,
+    };
+  }
+  auto iter = adapter_type_by_name_.find(interface_name);
+  if (iter == adapter_type_by_name_.end()) {
+    return {
+        .adapter_type = rtc::ADAPTER_TYPE_UNKNOWN,
+        .available = false,
+    };
+  }
+
+  return {
+      .adapter_type = iter->second,
+      .available = true,
+  };
+}
+
+void ObjCNetworkMonitor::OnPathUpdate(
+    std::map<std::string, rtc::AdapterType, rtc::AbslStringViewCmp> adapter_type_by_name) {
+  RTC_DCHECK(network_monitor_ != nil);
+  thread_->PostTask(SafeTask(safety_flag_, [this, adapter_type_by_name] {
+    RTC_DCHECK_RUN_ON(thread_);
+    adapter_type_by_name_ = adapter_type_by_name;
+    InvokeNetworksChangedCallback();
+  }));
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/sdk/objc/native/src/objc_video_decoder_factory.h b/third_party/libwebrtc/sdk/objc/native/src/objc_video_decoder_factory.h
new file mode 100644
index 0000000000..30ad8c2a4b
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/native/src/objc_video_decoder_factory.h
@@ -0,0 +1,41 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef SDK_OBJC_NATIVE_SRC_OBJC_VIDEO_DECODER_FACTORY_H_
+#define SDK_OBJC_NATIVE_SRC_OBJC_VIDEO_DECODER_FACTORY_H_
+
+#import "base/RTCMacros.h"
+
+#include "api/video_codecs/video_decoder_factory.h"
+#include "media/base/codec.h"
+
+@protocol RTC_OBJC_TYPE
+(RTCVideoDecoderFactory);
+
+namespace webrtc {
+
+class ObjCVideoDecoderFactory : public VideoDecoderFactory {
+ public:
+  explicit ObjCVideoDecoderFactory(id<RTC_OBJC_TYPE(RTCVideoDecoderFactory)>);
+  ~ObjCVideoDecoderFactory() override;
+
+  id<RTC_OBJC_TYPE(RTCVideoDecoderFactory)> wrapped_decoder_factory() const;
+
+  std::vector<SdpVideoFormat> GetSupportedFormats() const override;
+  std::unique_ptr<VideoDecoder> CreateVideoDecoder(
+      const SdpVideoFormat& format) override;
+
+ private:
+  id<RTC_OBJC_TYPE(RTCVideoDecoderFactory)> decoder_factory_;
+};
+
+}  // namespace webrtc
+
+#endif  // SDK_OBJC_NATIVE_SRC_OBJC_VIDEO_DECODER_FACTORY_H_
diff --git a/third_party/libwebrtc/sdk/objc/native/src/objc_video_decoder_factory.mm b/third_party/libwebrtc/sdk/objc/native/src/objc_video_decoder_factory.mm
new file mode 100644
index 0000000000..da3b302275
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/native/src/objc_video_decoder_factory.mm
@@ -0,0 +1,123 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "sdk/objc/native/src/objc_video_decoder_factory.h"
+
+#import "base/RTCMacros.h"
+#import "base/RTCVideoDecoder.h"
+#import "base/RTCVideoDecoderFactory.h"
+#import "base/RTCVideoFrame.h"
+#import "base/RTCVideoFrameBuffer.h"
+#import "components/video_codec/RTCCodecSpecificInfoH264.h"
+#import "sdk/objc/api/peerconnection/RTCEncodedImage+Private.h"
+#import "sdk/objc/api/peerconnection/RTCVideoCodecInfo+Private.h"
+#import "sdk/objc/api/video_codec/RTCWrappedNativeVideoDecoder.h"
+#import "sdk/objc/helpers/NSString+StdString.h"
+
+#include "api/video_codecs/sdp_video_format.h"
+#include "api/video_codecs/video_decoder.h"
+#include "modules/video_coding/include/video_codec_interface.h"
+#include "modules/video_coding/include/video_error_codes.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/time_utils.h"
+#include "sdk/objc/native/src/objc_frame_buffer.h"
+
+namespace webrtc {
+
+namespace {
+class ObjCVideoDecoder : public VideoDecoder {
+ public:
+  ObjCVideoDecoder(id<RTC_OBJC_TYPE(RTCVideoDecoder)> decoder)
+      : decoder_(decoder), implementation_name_([decoder implementationName].stdString) {}
+
+  bool Configure(const Settings &settings) override {
+    return
+        [decoder_ startDecodeWithNumberOfCores:settings.number_of_cores()] == WEBRTC_VIDEO_CODEC_OK;
+  }
+
+  int32_t Decode(const EncodedImage &input_image,
+                 bool missing_frames,
+                 int64_t render_time_ms = -1) override {
+    RTC_OBJC_TYPE(RTCEncodedImage) *encodedImage =
+        [[RTC_OBJC_TYPE(RTCEncodedImage) alloc] initWithNativeEncodedImage:input_image];
+
+    return [decoder_ decode:encodedImage
+              missingFrames:missing_frames
+          codecSpecificInfo:nil
+               renderTimeMs:render_time_ms];
+  }
+
+  int32_t RegisterDecodeCompleteCallback(DecodedImageCallback *callback) override {
+    [decoder_ setCallback:^(RTC_OBJC_TYPE(RTCVideoFrame) * frame) {
+      const auto buffer = rtc::make_ref_counted<ObjCFrameBuffer>(frame.buffer);
+      VideoFrame videoFrame =
+          VideoFrame::Builder()
+              .set_video_frame_buffer(buffer)
+              .set_timestamp_rtp((uint32_t)(frame.timeStampNs / rtc::kNumNanosecsPerMicrosec))
+              .set_timestamp_ms(0)
+              .set_rotation((VideoRotation)frame.rotation)
+              .build();
+      videoFrame.set_timestamp(frame.timeStamp);
+
+      callback->Decoded(videoFrame);
+    }];
+
+    return WEBRTC_VIDEO_CODEC_OK;
+  }
+
+  int32_t Release() override { return [decoder_ releaseDecoder]; }
+
+  const char *ImplementationName() const override { return implementation_name_.c_str(); }
+
+ private:
+  id<RTC_OBJC_TYPE(RTCVideoDecoder)> decoder_;
+  const std::string implementation_name_;
+};
+}  // namespace
+
+ObjCVideoDecoderFactory::ObjCVideoDecoderFactory(
+    id<RTC_OBJC_TYPE(RTCVideoDecoderFactory)> decoder_factory)
+    : decoder_factory_(decoder_factory) {}
+
+ObjCVideoDecoderFactory::~ObjCVideoDecoderFactory() {}
+
+id<RTC_OBJC_TYPE(RTCVideoDecoderFactory)> ObjCVideoDecoderFactory::wrapped_decoder_factory() const {
+  return decoder_factory_;
+}
+
+std::unique_ptr<VideoDecoder> ObjCVideoDecoderFactory::CreateVideoDecoder(
+    const SdpVideoFormat &format) {
+  NSString *codecName = [NSString stringWithUTF8String:format.name.c_str()];
+  for (RTC_OBJC_TYPE(RTCVideoCodecInfo) * codecInfo in decoder_factory_.supportedCodecs) {
+    if ([codecName isEqualToString:codecInfo.name]) {
+      id<RTC_OBJC_TYPE(RTCVideoDecoder)> decoder = [decoder_factory_ createDecoder:codecInfo];
+
+      if ([decoder isKindOfClass:[RTC_OBJC_TYPE(RTCWrappedNativeVideoDecoder) class]]) {
+        return [(RTC_OBJC_TYPE(RTCWrappedNativeVideoDecoder) *)decoder releaseWrappedDecoder];
+      } else {
+        return std::unique_ptr<ObjCVideoDecoder>(new ObjCVideoDecoder(decoder));
+      }
+    }
+  }
+
+  return nullptr;
+}
+
+std::vector<SdpVideoFormat> ObjCVideoDecoderFactory::GetSupportedFormats() const {
+  std::vector<SdpVideoFormat> supported_formats;
+  for (RTC_OBJC_TYPE(RTCVideoCodecInfo) * supportedCodec in decoder_factory_.supportedCodecs) {
+    SdpVideoFormat format = [supportedCodec nativeSdpVideoFormat];
+    supported_formats.push_back(format);
+  }
+
+  return supported_formats;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/sdk/objc/native/src/objc_video_encoder_factory.h b/third_party/libwebrtc/sdk/objc/native/src/objc_video_encoder_factory.h
new file mode 100644
index 0000000000..38db5e6ae7
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/native/src/objc_video_encoder_factory.h
@@ -0,0 +1,44 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef SDK_OBJC_NATIVE_SRC_OBJC_VIDEO_ENCODER_FACTORY_H_
+#define SDK_OBJC_NATIVE_SRC_OBJC_VIDEO_ENCODER_FACTORY_H_
+
+#import <Foundation/Foundation.h>
+
+#import "base/RTCMacros.h"
+
+#include "api/video_codecs/video_encoder_factory.h"
+
+@protocol RTC_OBJC_TYPE
+(RTCVideoEncoderFactory);
+
+namespace webrtc {
+
+class ObjCVideoEncoderFactory : public VideoEncoderFactory {
+ public:
+  explicit ObjCVideoEncoderFactory(id<RTC_OBJC_TYPE(RTCVideoEncoderFactory)>);
+  ~ObjCVideoEncoderFactory() override;
+
+  id<RTC_OBJC_TYPE(RTCVideoEncoderFactory)> wrapped_encoder_factory() const;
+
+  std::vector<SdpVideoFormat> GetSupportedFormats() const override;
+  std::vector<SdpVideoFormat> GetImplementations() const override;
+  std::unique_ptr<VideoEncoder> CreateVideoEncoder(
+      const SdpVideoFormat& format) override;
+  std::unique_ptr<EncoderSelectorInterface> GetEncoderSelector() const override;
+
+ private:
+  id<RTC_OBJC_TYPE(RTCVideoEncoderFactory)> encoder_factory_;
+};
+
+}  // namespace webrtc
+
+#endif  // SDK_OBJC_NATIVE_SRC_OBJC_VIDEO_ENCODER_FACTORY_H_
diff --git a/third_party/libwebrtc/sdk/objc/native/src/objc_video_encoder_factory.mm b/third_party/libwebrtc/sdk/objc/native/src/objc_video_encoder_factory.mm
new file mode 100644
index 0000000000..d4ea79cc88
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/native/src/objc_video_encoder_factory.mm
@@ -0,0 +1,209 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "sdk/objc/native/src/objc_video_encoder_factory.h"
+
+#include <string>
+
+#import "base/RTCMacros.h"
+#import "base/RTCVideoEncoder.h"
+#import "base/RTCVideoEncoderFactory.h"
+#import "components/video_codec/RTCCodecSpecificInfoH264+Private.h"
+#import "sdk/objc/api/peerconnection/RTCEncodedImage+Private.h"
+#import "sdk/objc/api/peerconnection/RTCVideoCodecInfo+Private.h"
+#import "sdk/objc/api/peerconnection/RTCVideoEncoderSettings+Private.h"
+#import "sdk/objc/api/video_codec/RTCVideoCodecConstants.h"
+#import "sdk/objc/api/video_codec/RTCWrappedNativeVideoEncoder.h"
+#import "sdk/objc/helpers/NSString+StdString.h"
+
+#include "api/video/video_frame.h"
+#include "api/video_codecs/sdp_video_format.h"
+#include "api/video_codecs/video_encoder.h"
+#include "modules/video_coding/include/video_codec_interface.h"
+#include "modules/video_coding/include/video_error_codes.h"
+#include "rtc_base/logging.h"
+#include "sdk/objc/native/src/objc_video_frame.h"
+
+namespace webrtc {
+
+namespace {
+
+class ObjCVideoEncoder : public VideoEncoder {
+ public:
+  ObjCVideoEncoder(id<RTC_OBJC_TYPE(RTCVideoEncoder)> encoder)
+      : encoder_(encoder), implementation_name_([encoder implementationName].stdString) {}
+
+  int32_t InitEncode(const VideoCodec *codec_settings, const Settings &encoder_settings) override {
+    RTC_OBJC_TYPE(RTCVideoEncoderSettings) *settings =
+        [[RTC_OBJC_TYPE(RTCVideoEncoderSettings) alloc] initWithNativeVideoCodec:codec_settings];
+    return [encoder_ startEncodeWithSettings:settings
+                               numberOfCores:encoder_settings.number_of_cores];
+  }
+
+  int32_t RegisterEncodeCompleteCallback(EncodedImageCallback *callback) override {
+    if (callback) {
+      [encoder_ setCallback:^BOOL(RTC_OBJC_TYPE(RTCEncodedImage) * _Nonnull frame,
+                                  id<RTC_OBJC_TYPE(RTCCodecSpecificInfo)> _Nonnull info) {
+        EncodedImage encodedImage = [frame nativeEncodedImage];
+
+        // Handle types that can be converted into one of CodecSpecificInfo's hard coded cases.
+        CodecSpecificInfo codecSpecificInfo;
+        if ([info isKindOfClass:[RTC_OBJC_TYPE(RTCCodecSpecificInfoH264) class]]) {
+          codecSpecificInfo =
+              [(RTC_OBJC_TYPE(RTCCodecSpecificInfoH264) *)info nativeCodecSpecificInfo];
+        }
+
+        EncodedImageCallback::Result res = callback->OnEncodedImage(encodedImage, &codecSpecificInfo);
+        return res.error == EncodedImageCallback::Result::OK;
+      }];
+    } else {
+      [encoder_ setCallback:nil];
+    }
+    return WEBRTC_VIDEO_CODEC_OK;
+  }
+
+  int32_t Release() override { return [encoder_ releaseEncoder]; }
+
+  int32_t Encode(const VideoFrame &frame,
+                 const std::vector<VideoFrameType> *frame_types) override {
+    NSMutableArray<NSNumber *> *rtcFrameTypes = [NSMutableArray array];
+    for (size_t i = 0; i < frame_types->size(); ++i) {
+      [rtcFrameTypes addObject:@(RTCFrameType(frame_types->at(i)))];
+    }
+
+    return [encoder_ encode:ToObjCVideoFrame(frame)
+          codecSpecificInfo:nil
+                 frameTypes:rtcFrameTypes];
+  }
+
+  void SetRates(const RateControlParameters &parameters) override {
+    const uint32_t bitrate = parameters.bitrate.get_sum_kbps();
+    const uint32_t framerate = static_cast<uint32_t>(parameters.framerate_fps + 0.5);
+    [encoder_ setBitrate:bitrate framerate:framerate];
+  }
+
+  VideoEncoder::EncoderInfo GetEncoderInfo() const override {
+    EncoderInfo info;
+    info.implementation_name = implementation_name_;
+
+    RTC_OBJC_TYPE(RTCVideoEncoderQpThresholds) *qp_thresholds = [encoder_ scalingSettings];
+    info.scaling_settings = qp_thresholds ? ScalingSettings(qp_thresholds.low, qp_thresholds.high) :
+                                            ScalingSettings::kOff;
+
+    info.requested_resolution_alignment = encoder_.resolutionAlignment > 0 ?: 1;
+    info.apply_alignment_to_all_simulcast_layers = encoder_.applyAlignmentToAllSimulcastLayers;
+    info.supports_native_handle = encoder_.supportsNativeHandle;
+    info.is_hardware_accelerated = true;
+    return info;
+  }
+
+ private:
+  id<RTC_OBJC_TYPE(RTCVideoEncoder)> encoder_;
+  const std::string implementation_name_;
+};
+
+class ObjcVideoEncoderSelector : public VideoEncoderFactory::EncoderSelectorInterface {
+ public:
+  ObjcVideoEncoderSelector(id<RTC_OBJC_TYPE(RTCVideoEncoderSelector)> selector) {
+    selector_ = selector;
+  }
+  void OnCurrentEncoder(const SdpVideoFormat &format) override {
+    RTC_OBJC_TYPE(RTCVideoCodecInfo) *info =
+        [[RTC_OBJC_TYPE(RTCVideoCodecInfo) alloc] initWithNativeSdpVideoFormat:format];
+    [selector_ registerCurrentEncoderInfo:info];
+  }
+  absl::optional<SdpVideoFormat> OnEncoderBroken() override {
+    RTC_OBJC_TYPE(RTCVideoCodecInfo) *info = [selector_ encoderForBrokenEncoder];
+    if (info) {
+      return [info nativeSdpVideoFormat];
+    }
+    return absl::nullopt;
+  }
+  absl::optional<SdpVideoFormat> OnAvailableBitrate(const DataRate &rate) override {
+    RTC_OBJC_TYPE(RTCVideoCodecInfo) *info = [selector_ encoderForBitrate:rate.kbps<NSInteger>()];
+    if (info) {
+      return [info nativeSdpVideoFormat];
+    }
+    return absl::nullopt;
+  }
+
+  absl::optional<SdpVideoFormat> OnResolutionChange(const RenderResolution &resolution) override {
+    if ([selector_ respondsToSelector:@selector(encoderForResolutionChangeBySize:)]) {
+      RTC_OBJC_TYPE(RTCVideoCodecInfo) *info = [selector_
+          encoderForResolutionChangeBySize:CGSizeMake(resolution.Width(), resolution.Height())];
+      if (info) {
+        return [info nativeSdpVideoFormat];
+      }
+    }
+    return absl::nullopt;
+  }
+
+ private:
+  id<RTC_OBJC_TYPE(RTCVideoEncoderSelector)> selector_;
+};
+
+}  // namespace
+
+ObjCVideoEncoderFactory::ObjCVideoEncoderFactory(
+    id<RTC_OBJC_TYPE(RTCVideoEncoderFactory)> encoder_factory)
+    : encoder_factory_(encoder_factory) {}
+
+ObjCVideoEncoderFactory::~ObjCVideoEncoderFactory() {}
+
+id<RTC_OBJC_TYPE(RTCVideoEncoderFactory)> ObjCVideoEncoderFactory::wrapped_encoder_factory() const {
+  return encoder_factory_;
+}
+
+std::vector<SdpVideoFormat> ObjCVideoEncoderFactory::GetSupportedFormats() const {
+  std::vector<SdpVideoFormat> supported_formats;
+  for (RTC_OBJC_TYPE(RTCVideoCodecInfo) * supportedCodec in [encoder_factory_ supportedCodecs]) {
+    SdpVideoFormat format = [supportedCodec nativeSdpVideoFormat];
+    supported_formats.push_back(format);
+  }
+
+  return supported_formats;
+}
+
+std::vector<SdpVideoFormat> ObjCVideoEncoderFactory::GetImplementations() const {
+  if ([encoder_factory_ respondsToSelector:@selector(implementations)]) {
+    std::vector<SdpVideoFormat> supported_formats;
+    for (RTC_OBJC_TYPE(RTCVideoCodecInfo) * supportedCodec in [encoder_factory_ implementations]) {
+      SdpVideoFormat format = [supportedCodec nativeSdpVideoFormat];
+      supported_formats.push_back(format);
+    }
+    return supported_formats;
+  }
+  return GetSupportedFormats();
+}
+
+std::unique_ptr<VideoEncoder> ObjCVideoEncoderFactory::CreateVideoEncoder(
+    const SdpVideoFormat &format) {
+  RTC_OBJC_TYPE(RTCVideoCodecInfo) *info =
+      [[RTC_OBJC_TYPE(RTCVideoCodecInfo) alloc] initWithNativeSdpVideoFormat:format];
+  id<RTC_OBJC_TYPE(RTCVideoEncoder)> encoder = [encoder_factory_ createEncoder:info];
+  if ([encoder isKindOfClass:[RTC_OBJC_TYPE(RTCWrappedNativeVideoEncoder) class]]) {
+    return [(RTC_OBJC_TYPE(RTCWrappedNativeVideoEncoder) *)encoder releaseWrappedEncoder];
+  } else {
+    return std::unique_ptr<ObjCVideoEncoder>(new ObjCVideoEncoder(encoder));
+  }
+}
+
+std::unique_ptr<VideoEncoderFactory::EncoderSelectorInterface>
+    ObjCVideoEncoderFactory::GetEncoderSelector() const {
+  if ([encoder_factory_ respondsToSelector:@selector(encoderSelector)]) {
+    id<RTC_OBJC_TYPE(RTCVideoEncoderSelector)> selector = [encoder_factory_ encoderSelector];
+    if (selector) {
+      return absl::make_unique<ObjcVideoEncoderSelector>(selector);
+    }
+  }
+  return nullptr;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/sdk/objc/native/src/objc_video_frame.h b/third_party/libwebrtc/sdk/objc/native/src/objc_video_frame.h
new file mode 100644
index 0000000000..c2931cb2f8
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/native/src/objc_video_frame.h
@@ -0,0 +1,24 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef SDK_OBJC_NATIVE_SRC_OBJC_VIDEO_FRAME_H_
+#define SDK_OBJC_NATIVE_SRC_OBJC_VIDEO_FRAME_H_
+
+#import "base/RTCVideoFrame.h"
+
+#include "api/video/video_frame.h"
+
+namespace webrtc {
+
+RTC_OBJC_TYPE(RTCVideoFrame) * ToObjCVideoFrame(const VideoFrame& frame);
+
+}  // namespace webrtc
+
+#endif  // SDK_OBJC_NATIVE_SRC_OBJC_VIDEO_FRAME_H_
diff --git a/third_party/libwebrtc/sdk/objc/native/src/objc_video_frame.mm b/third_party/libwebrtc/sdk/objc/native/src/objc_video_frame.mm
new file mode 100644
index 0000000000..2e8ce6153e
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/native/src/objc_video_frame.mm
@@ -0,0 +1,28 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "sdk/objc/native/src/objc_video_frame.h"
+
+#include "rtc_base/time_utils.h"
+#include "sdk/objc/native/src/objc_frame_buffer.h"
+
+namespace webrtc {
+
+RTC_OBJC_TYPE(RTCVideoFrame) * ToObjCVideoFrame(const VideoFrame &frame) {
+  RTC_OBJC_TYPE(RTCVideoFrame) *videoFrame = [[RTC_OBJC_TYPE(RTCVideoFrame) alloc]
+      initWithBuffer:ToObjCVideoFrameBuffer(frame.video_frame_buffer())
+            rotation:RTCVideoRotation(frame.rotation())
+         timeStampNs:frame.timestamp_us() * rtc::kNumNanosecsPerMicrosec];
+  videoFrame.timeStamp = frame.timestamp();
+
+  return videoFrame;
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/sdk/objc/native/src/objc_video_renderer.h b/third_party/libwebrtc/sdk/objc/native/src/objc_video_renderer.h
new file mode 100644
index 0000000000..f9c35eae96
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/native/src/objc_video_renderer.h
@@ -0,0 +1,39 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef SDK_OBJC_NATIVE_SRC_OBJC_VIDEO_RENDERER_H_
+#define SDK_OBJC_NATIVE_SRC_OBJC_VIDEO_RENDERER_H_
+
+#import <CoreGraphics/CoreGraphics.h>
+#import <Foundation/Foundation.h>
+
+#import "base/RTCMacros.h"
+
+#include "api/video/video_frame.h"
+#include "api/video/video_sink_interface.h"
+
+@protocol RTC_OBJC_TYPE
+(RTCVideoRenderer);
+
+namespace webrtc {
+
+class ObjCVideoRenderer : public rtc::VideoSinkInterface<VideoFrame> {
+ public:
+  ObjCVideoRenderer(id<RTC_OBJC_TYPE(RTCVideoRenderer)> renderer);
+  void OnFrame(const VideoFrame& nativeVideoFrame) override;
+
+ private:
+  id<RTC_OBJC_TYPE(RTCVideoRenderer)> renderer_;
+  CGSize size_;
+};
+
+}  // namespace webrtc
+
+#endif  // SDK_OBJC_NATIVE_SRC_OBJC_VIDEO_RENDERER_H_
diff --git a/third_party/libwebrtc/sdk/objc/native/src/objc_video_renderer.mm b/third_party/libwebrtc/sdk/objc/native/src/objc_video_renderer.mm
new file mode 100644
index 0000000000..4a9b647ec3
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/native/src/objc_video_renderer.mm
@@ -0,0 +1,38 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "sdk/objc/native/src/objc_video_renderer.h"
+
+#import "base/RTCMacros.h"
+#import "base/RTCVideoFrame.h"
+#import "base/RTCVideoRenderer.h"
+
+#include "sdk/objc/native/src/objc_video_frame.h"
+
+namespace webrtc {
+
+ObjCVideoRenderer::ObjCVideoRenderer(id<RTC_OBJC_TYPE(RTCVideoRenderer)> renderer)
+    : renderer_(renderer), size_(CGSizeZero) {}
+
+void ObjCVideoRenderer::OnFrame(const VideoFrame& nativeVideoFrame) {
+  RTC_OBJC_TYPE(RTCVideoFrame)* videoFrame = ToObjCVideoFrame(nativeVideoFrame);
+
+  CGSize current_size = (videoFrame.rotation % 180 == 0) ?
+      CGSizeMake(videoFrame.width, videoFrame.height) :
+      CGSizeMake(videoFrame.height, videoFrame.width);
+
+  if (!CGSizeEqualToSize(size_, current_size)) {
+    size_ = current_size;
+    [renderer_ setSize:size_];
+  }
+  [renderer_ renderFrame:videoFrame];
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/sdk/objc/native/src/objc_video_track_source.h b/third_party/libwebrtc/sdk/objc/native/src/objc_video_track_source.h
new file mode 100644
index 0000000000..19a3d6db43
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/native/src/objc_video_track_source.h
@@ -0,0 +1,59 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef SDK_OBJC_CLASSES_VIDEO_OBJC_VIDEO_TRACK_SOURCE_H_
+#define SDK_OBJC_CLASSES_VIDEO_OBJC_VIDEO_TRACK_SOURCE_H_
+
+#import "base/RTCVideoCapturer.h"
+
+#include "base/RTCMacros.h"
+#include "media/base/adapted_video_track_source.h"
+#include "rtc_base/timestamp_aligner.h"
+
+RTC_FWD_DECL_OBJC_CLASS(RTC_OBJC_TYPE(RTCVideoFrame));
+
+@interface RTCObjCVideoSourceAdapter : NSObject <RTC_OBJC_TYPE (RTCVideoCapturerDelegate)>
+@end
+
+namespace webrtc {
+
+class ObjCVideoTrackSource : public rtc::AdaptedVideoTrackSource {
+ public:
+  ObjCVideoTrackSource();
+  explicit ObjCVideoTrackSource(bool is_screencast);
+  explicit ObjCVideoTrackSource(RTCObjCVideoSourceAdapter* adapter);
+
+  bool is_screencast() const override;
+
+  // Indicates that the encoder should denoise video before encoding it.
+  // If it is not set, the default configuration is used which is different
+  // depending on video codec.
+  absl::optional<bool> needs_denoising() const override;
+
+  SourceState state() const override;
+
+  bool remote() const override;
+
+  void OnCapturedFrame(RTC_OBJC_TYPE(RTCVideoFrame) * frame);
+
+  // Called by RTCVideoSource.
+  void OnOutputFormatRequest(int width, int height, int fps);
+
+ private:
+  rtc::VideoBroadcaster broadcaster_;
+  rtc::TimestampAligner timestamp_aligner_;
+
+  RTCObjCVideoSourceAdapter* adapter_;
+  bool is_screencast_;
+};
+
+}  // namespace webrtc
+
+#endif  // SDK_OBJC_CLASSES_VIDEO_OBJC_VIDEO_TRACK_SOURCE_H_
diff --git a/third_party/libwebrtc/sdk/objc/native/src/objc_video_track_source.mm b/third_party/libwebrtc/sdk/objc/native/src/objc_video_track_source.mm
new file mode 100644
index 0000000000..7937e90505
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/native/src/objc_video_track_source.mm
@@ -0,0 +1,131 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "sdk/objc/native/src/objc_video_track_source.h"
+
+#import "base/RTCVideoFrame.h"
+#import "base/RTCVideoFrameBuffer.h"
+#import "components/video_frame_buffer/RTCCVPixelBuffer.h"
+
+#include "api/video/i420_buffer.h"
+#include "sdk/objc/native/src/objc_frame_buffer.h"
+
+@interface RTCObjCVideoSourceAdapter ()
+@property(nonatomic) webrtc::ObjCVideoTrackSource *objCVideoTrackSource;
+@end
+
+@implementation RTCObjCVideoSourceAdapter
+
+@synthesize objCVideoTrackSource = _objCVideoTrackSource;
+
+- (void)capturer:(RTC_OBJC_TYPE(RTCVideoCapturer) *)capturer
+    didCaptureVideoFrame:(RTC_OBJC_TYPE(RTCVideoFrame) *)frame {
+  _objCVideoTrackSource->OnCapturedFrame(frame);
+}
+
+@end
+
+namespace webrtc {
+
+ObjCVideoTrackSource::ObjCVideoTrackSource() : ObjCVideoTrackSource(false) {}
+
+ObjCVideoTrackSource::ObjCVideoTrackSource(bool is_screencast)
+    : AdaptedVideoTrackSource(/* required resolution alignment */ 2),
+      is_screencast_(is_screencast) {}
+
+ObjCVideoTrackSource::ObjCVideoTrackSource(RTCObjCVideoSourceAdapter *adapter) : adapter_(adapter) {
+  adapter_.objCVideoTrackSource = this;
+}
+
+bool ObjCVideoTrackSource::is_screencast() const {
+  return is_screencast_;
+}
+
+absl::optional<bool> ObjCVideoTrackSource::needs_denoising() const {
+  return false;
+}
+
+MediaSourceInterface::SourceState ObjCVideoTrackSource::state() const {
+  return SourceState::kLive;
+}
+
+bool ObjCVideoTrackSource::remote() const {
+  return false;
+}
+
+void ObjCVideoTrackSource::OnOutputFormatRequest(int width, int height, int fps) {
+  cricket::VideoFormat format(width, height, cricket::VideoFormat::FpsToInterval(fps), 0);
+  video_adapter()->OnOutputFormatRequest(format);
+}
+
+void ObjCVideoTrackSource::OnCapturedFrame(RTC_OBJC_TYPE(RTCVideoFrame) * frame) {
+  const int64_t timestamp_us = frame.timeStampNs / rtc::kNumNanosecsPerMicrosec;
+  const int64_t translated_timestamp_us =
+      timestamp_aligner_.TranslateTimestamp(timestamp_us, rtc::TimeMicros());
+
+  int adapted_width;
+  int adapted_height;
+  int crop_width;
+  int crop_height;
+  int crop_x;
+  int crop_y;
+  if (!AdaptFrame(frame.width,
+                  frame.height,
+                  timestamp_us,
+                  &adapted_width,
+                  &adapted_height,
+                  &crop_width,
+                  &crop_height,
+                  &crop_x,
+                  &crop_y)) {
+    return;
+  }
+
+  rtc::scoped_refptr<VideoFrameBuffer> buffer;
+  if (adapted_width == frame.width && adapted_height == frame.height) {
+    // No adaption - optimized path.
+    buffer = rtc::make_ref_counted<ObjCFrameBuffer>(frame.buffer);
+  } else if ([frame.buffer isKindOfClass:[RTC_OBJC_TYPE(RTCCVPixelBuffer) class]]) {
+    // Adapted CVPixelBuffer frame.
+    RTC_OBJC_TYPE(RTCCVPixelBuffer) *rtcPixelBuffer =
+        (RTC_OBJC_TYPE(RTCCVPixelBuffer) *)frame.buffer;
+    buffer = rtc::make_ref_counted<ObjCFrameBuffer>([[RTC_OBJC_TYPE(RTCCVPixelBuffer) alloc]
+        initWithPixelBuffer:rtcPixelBuffer.pixelBuffer
+               adaptedWidth:adapted_width
+              adaptedHeight:adapted_height
+                  cropWidth:crop_width
+                 cropHeight:crop_height
+                      cropX:crop_x + rtcPixelBuffer.cropX
+                      cropY:crop_y + rtcPixelBuffer.cropY]);
+  } else {
+    // Adapted I420 frame.
+    // TODO(magjed): Optimize this I420 path.
+    rtc::scoped_refptr<I420Buffer> i420_buffer = I420Buffer::Create(adapted_width, adapted_height);
+    buffer = rtc::make_ref_counted<ObjCFrameBuffer>(frame.buffer);
+    i420_buffer->CropAndScaleFrom(*buffer->ToI420(), crop_x, crop_y, crop_width, crop_height);
+    buffer = i420_buffer;
+  }
+
+  // Applying rotation is only supported for legacy reasons and performance is
+  // not critical here.
+  VideoRotation rotation = static_cast<VideoRotation>(frame.rotation);
+  if (apply_rotation() && rotation != kVideoRotation_0) {
+    buffer = I420Buffer::Rotate(*buffer->ToI420(), rotation);
+    rotation = kVideoRotation_0;
+  }
+
+  OnFrame(VideoFrame::Builder()
+              .set_video_frame_buffer(buffer)
+              .set_rotation(rotation)
+              .set_timestamp_us(translated_timestamp_us)
+              .build());
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/sdk/objc/unittests/ObjCVideoTrackSource_xctest.mm b/third_party/libwebrtc/sdk/objc/unittests/ObjCVideoTrackSource_xctest.mm
new file mode 100644
index 0000000000..4c8bf348f4
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/unittests/ObjCVideoTrackSource_xctest.mm
@@ -0,0 +1,469 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+#import <XCTest/XCTest.h>
+
+#include "sdk/objc/native/src/objc_video_track_source.h"
+
+#import "api/video_frame_buffer/RTCNativeI420Buffer+Private.h"
+#import "base/RTCVideoFrame.h"
+#import "base/RTCVideoFrameBuffer.h"
+#import "components/video_frame_buffer/RTCCVPixelBuffer.h"
+#import "frame_buffer_helpers.h"
+
+#include "api/scoped_refptr.h"
+#include "common_video/libyuv/include/webrtc_libyuv.h"
+#include "media/base/fake_video_renderer.h"
+#include "sdk/objc/native/api/video_frame.h"
+
+typedef void (^VideoSinkCallback)(RTC_OBJC_TYPE(RTCVideoFrame) *);
+
+namespace {
+
+class ObjCCallbackVideoSink : public rtc::VideoSinkInterface<webrtc::VideoFrame> {
+ public:
+  ObjCCallbackVideoSink(VideoSinkCallback callback) : callback_(callback) {}
+
+  void OnFrame(const webrtc::VideoFrame &frame) override {
+    callback_(NativeToObjCVideoFrame(frame));
+  }
+
+ private:
+  VideoSinkCallback callback_;
+};
+
+}  // namespace
+
+@interface ObjCVideoTrackSourceTests : XCTestCase
+@end
+
+@implementation ObjCVideoTrackSourceTests {
+  rtc::scoped_refptr<webrtc::ObjCVideoTrackSource> _video_source;
+}
+
+- (void)setUp {
+  _video_source = rtc::make_ref_counted<webrtc::ObjCVideoTrackSource>();
+}
+
+- (void)tearDown {
+  _video_source = NULL;
+}
+
+- (void)testOnCapturedFrameAdaptsFrame {
+  CVPixelBufferRef pixelBufferRef = NULL;
+  CVPixelBufferCreate(
+      NULL, 720, 1280, kCVPixelFormatType_420YpCbCr8BiPlanarVideoRange, NULL, &pixelBufferRef);
+
+  RTC_OBJC_TYPE(RTCCVPixelBuffer) *buffer =
+      [[RTC_OBJC_TYPE(RTCCVPixelBuffer) alloc] initWithPixelBuffer:pixelBufferRef];
+
+  RTC_OBJC_TYPE(RTCVideoFrame) *frame =
+      [[RTC_OBJC_TYPE(RTCVideoFrame) alloc] initWithBuffer:buffer
+                                                  rotation:RTCVideoRotation_0
+                                               timeStampNs:0];
+
+  cricket::FakeVideoRenderer *video_renderer = new cricket::FakeVideoRenderer();
+  const rtc::VideoSinkWants video_sink_wants;
+  rtc::VideoSourceInterface<webrtc::VideoFrame> *video_source_interface = _video_source.get();
+  video_source_interface->AddOrUpdateSink(video_renderer, video_sink_wants);
+
+  _video_source->OnOutputFormatRequest(640, 360, 30);
+  _video_source->OnCapturedFrame(frame);
+
+  XCTAssertEqual(video_renderer->num_rendered_frames(), 1);
+  XCTAssertEqual(video_renderer->width(), 360);
+  XCTAssertEqual(video_renderer->height(), 640);
+
+  CVBufferRelease(pixelBufferRef);
+}
+
+- (void)testOnCapturedFrameAdaptsFrameWithAlignment {
+  // Requesting to adapt 1280x720 to 912x514 gives 639x360 without alignment. The 639 causes issues
+  // with some hardware encoders (e.g. HEVC) so in this test we verify that the alignment is set and
+  // respected.
+
+  CVPixelBufferRef pixelBufferRef = NULL;
+  CVPixelBufferCreate(
+      NULL, 720, 1280, kCVPixelFormatType_420YpCbCr8BiPlanarVideoRange, NULL, &pixelBufferRef);
+
+  RTC_OBJC_TYPE(RTCCVPixelBuffer) *buffer =
+      [[RTC_OBJC_TYPE(RTCCVPixelBuffer) alloc] initWithPixelBuffer:pixelBufferRef];
+
+  RTC_OBJC_TYPE(RTCVideoFrame) *frame =
+      [[RTC_OBJC_TYPE(RTCVideoFrame) alloc] initWithBuffer:buffer
+                                                  rotation:RTCVideoRotation_0
+                                               timeStampNs:0];
+
+  cricket::FakeVideoRenderer *video_renderer = new cricket::FakeVideoRenderer();
+  const rtc::VideoSinkWants video_sink_wants;
+  rtc::VideoSourceInterface<webrtc::VideoFrame> *video_source_interface = _video_source.get();
+  video_source_interface->AddOrUpdateSink(video_renderer, video_sink_wants);
+
+  _video_source->OnOutputFormatRequest(912, 514, 30);
+  _video_source->OnCapturedFrame(frame);
+
+  XCTAssertEqual(video_renderer->num_rendered_frames(), 1);
+  XCTAssertEqual(video_renderer->width(), 360);
+  XCTAssertEqual(video_renderer->height(), 640);
+
+  CVBufferRelease(pixelBufferRef);
+}
+
+- (void)testOnCapturedFrameAdaptationResultsInCommonResolutions {
+  // Some of the most common resolutions used in the wild are 640x360, 480x270 and 320x180.
+  // Make sure that we properly scale down to exactly these resolutions.
+  CVPixelBufferRef pixelBufferRef = NULL;
+  CVPixelBufferCreate(
+      NULL, 720, 1280, kCVPixelFormatType_420YpCbCr8BiPlanarVideoRange, NULL, &pixelBufferRef);
+
+  RTC_OBJC_TYPE(RTCCVPixelBuffer) *buffer =
+      [[RTC_OBJC_TYPE(RTCCVPixelBuffer) alloc] initWithPixelBuffer:pixelBufferRef];
+
+  RTC_OBJC_TYPE(RTCVideoFrame) *frame =
+      [[RTC_OBJC_TYPE(RTCVideoFrame) alloc] initWithBuffer:buffer
+                                                  rotation:RTCVideoRotation_0
+                                               timeStampNs:0];
+
+  cricket::FakeVideoRenderer *video_renderer = new cricket::FakeVideoRenderer();
+  const rtc::VideoSinkWants video_sink_wants;
+  rtc::VideoSourceInterface<webrtc::VideoFrame> *video_source_interface = _video_source.get();
+  video_source_interface->AddOrUpdateSink(video_renderer, video_sink_wants);
+
+  _video_source->OnOutputFormatRequest(640, 360, 30);
+  _video_source->OnCapturedFrame(frame);
+
+  XCTAssertEqual(video_renderer->num_rendered_frames(), 1);
+  XCTAssertEqual(video_renderer->width(), 360);
+  XCTAssertEqual(video_renderer->height(), 640);
+
+  _video_source->OnOutputFormatRequest(480, 270, 30);
+  _video_source->OnCapturedFrame(frame);
+
+  XCTAssertEqual(video_renderer->num_rendered_frames(), 2);
+  XCTAssertEqual(video_renderer->width(), 270);
+  XCTAssertEqual(video_renderer->height(), 480);
+
+  _video_source->OnOutputFormatRequest(320, 180, 30);
+  _video_source->OnCapturedFrame(frame);
+
+  XCTAssertEqual(video_renderer->num_rendered_frames(), 3);
+  XCTAssertEqual(video_renderer->width(), 180);
+  XCTAssertEqual(video_renderer->height(), 320);
+
+  CVBufferRelease(pixelBufferRef);
+}
+
+- (void)testOnCapturedFrameWithoutAdaptation {
+  CVPixelBufferRef pixelBufferRef = NULL;
+  CVPixelBufferCreate(
+      NULL, 360, 640, kCVPixelFormatType_420YpCbCr8BiPlanarVideoRange, NULL, &pixelBufferRef);
+
+  RTC_OBJC_TYPE(RTCCVPixelBuffer) *buffer =
+      [[RTC_OBJC_TYPE(RTCCVPixelBuffer) alloc] initWithPixelBuffer:pixelBufferRef];
+  RTC_OBJC_TYPE(RTCVideoFrame) *frame =
+      [[RTC_OBJC_TYPE(RTCVideoFrame) alloc] initWithBuffer:buffer
+                                                  rotation:RTCVideoRotation_0
+                                               timeStampNs:0];
+
+  XCTestExpectation *callbackExpectation = [self expectationWithDescription:@"videoSinkCallback"];
+  ObjCCallbackVideoSink callback_video_sink(^void(RTC_OBJC_TYPE(RTCVideoFrame) * outputFrame) {
+    XCTAssertEqual(frame.width, outputFrame.width);
+    XCTAssertEqual(frame.height, outputFrame.height);
+
+    RTC_OBJC_TYPE(RTCCVPixelBuffer) *outputBuffer = outputFrame.buffer;
+    XCTAssertEqual(buffer.cropX, outputBuffer.cropX);
+    XCTAssertEqual(buffer.cropY, outputBuffer.cropY);
+    XCTAssertEqual(buffer.pixelBuffer, outputBuffer.pixelBuffer);
+
+    [callbackExpectation fulfill];
+  });
+
+  const rtc::VideoSinkWants video_sink_wants;
+  rtc::VideoSourceInterface<webrtc::VideoFrame> *video_source_interface = _video_source.get();
+  video_source_interface->AddOrUpdateSink(&callback_video_sink, video_sink_wants);
+
+  _video_source->OnOutputFormatRequest(640, 360, 30);
+  _video_source->OnCapturedFrame(frame);
+
+  [self waitForExpectations:@[ callbackExpectation ] timeout:10.0];
+  CVBufferRelease(pixelBufferRef);
+}
+
+- (void)testOnCapturedFrameCVPixelBufferNeedsAdaptation {
+  CVPixelBufferRef pixelBufferRef = NULL;
+  CVPixelBufferCreate(
+      NULL, 720, 1280, kCVPixelFormatType_420YpCbCr8BiPlanarVideoRange, NULL, &pixelBufferRef);
+
+  RTC_OBJC_TYPE(RTCCVPixelBuffer) *buffer =
+      [[RTC_OBJC_TYPE(RTCCVPixelBuffer) alloc] initWithPixelBuffer:pixelBufferRef];
+  RTC_OBJC_TYPE(RTCVideoFrame) *frame =
+      [[RTC_OBJC_TYPE(RTCVideoFrame) alloc] initWithBuffer:buffer
+                                                  rotation:RTCVideoRotation_0
+                                               timeStampNs:0];
+
+  XCTestExpectation *callbackExpectation = [self expectationWithDescription:@"videoSinkCallback"];
+  ObjCCallbackVideoSink callback_video_sink(^void(RTC_OBJC_TYPE(RTCVideoFrame) * outputFrame) {
+    XCTAssertEqual(outputFrame.width, 360);
+    XCTAssertEqual(outputFrame.height, 640);
+
+    RTC_OBJC_TYPE(RTCCVPixelBuffer) *outputBuffer = outputFrame.buffer;
+    XCTAssertEqual(outputBuffer.cropX, 0);
+    XCTAssertEqual(outputBuffer.cropY, 0);
+    XCTAssertEqual(buffer.pixelBuffer, outputBuffer.pixelBuffer);
+
+    [callbackExpectation fulfill];
+  });
+
+  const rtc::VideoSinkWants video_sink_wants;
+  rtc::VideoSourceInterface<webrtc::VideoFrame> *video_source_interface = _video_source.get();
+  video_source_interface->AddOrUpdateSink(&callback_video_sink, video_sink_wants);
+
+  _video_source->OnOutputFormatRequest(640, 360, 30);
+  _video_source->OnCapturedFrame(frame);
+
+  [self waitForExpectations:@[ callbackExpectation ] timeout:10.0];
+  CVBufferRelease(pixelBufferRef);
+}
+
+- (void)testOnCapturedFrameCVPixelBufferNeedsCropping {
+  CVPixelBufferRef pixelBufferRef = NULL;
+  CVPixelBufferCreate(
+      NULL, 380, 640, kCVPixelFormatType_420YpCbCr8BiPlanarVideoRange, NULL, &pixelBufferRef);
+
+  RTC_OBJC_TYPE(RTCCVPixelBuffer) *buffer =
+      [[RTC_OBJC_TYPE(RTCCVPixelBuffer) alloc] initWithPixelBuffer:pixelBufferRef];
+  RTC_OBJC_TYPE(RTCVideoFrame) *frame =
+      [[RTC_OBJC_TYPE(RTCVideoFrame) alloc] initWithBuffer:buffer
+                                                  rotation:RTCVideoRotation_0
+                                               timeStampNs:0];
+
+  XCTestExpectation *callbackExpectation = [self expectationWithDescription:@"videoSinkCallback"];
+  ObjCCallbackVideoSink callback_video_sink(^void(RTC_OBJC_TYPE(RTCVideoFrame) * outputFrame) {
+    XCTAssertEqual(outputFrame.width, 360);
+    XCTAssertEqual(outputFrame.height, 640);
+
+    RTC_OBJC_TYPE(RTCCVPixelBuffer) *outputBuffer = outputFrame.buffer;
+    XCTAssertEqual(outputBuffer.cropX, 10);
+    XCTAssertEqual(outputBuffer.cropY, 0);
+    XCTAssertEqual(buffer.pixelBuffer, outputBuffer.pixelBuffer);
+
+    [callbackExpectation fulfill];
+  });
+
+  const rtc::VideoSinkWants video_sink_wants;
+  rtc::VideoSourceInterface<webrtc::VideoFrame> *video_source_interface = _video_source.get();
+  video_source_interface->AddOrUpdateSink(&callback_video_sink, video_sink_wants);
+
+  _video_source->OnOutputFormatRequest(640, 360, 30);
+  _video_source->OnCapturedFrame(frame);
+
+  [self waitForExpectations:@[ callbackExpectation ] timeout:10.0];
+  CVBufferRelease(pixelBufferRef);
+}
+
+- (void)testOnCapturedFramePreAdaptedCVPixelBufferNeedsAdaptation {
+  CVPixelBufferRef pixelBufferRef = NULL;
+  CVPixelBufferCreate(
+      NULL, 720, 1280, kCVPixelFormatType_420YpCbCr8BiPlanarVideoRange, NULL, &pixelBufferRef);
+
+  // Create a frame that's already adapted down.
+  RTC_OBJC_TYPE(RTCCVPixelBuffer) *buffer =
+      [[RTC_OBJC_TYPE(RTCCVPixelBuffer) alloc] initWithPixelBuffer:pixelBufferRef
+                                                      adaptedWidth:640
+                                                     adaptedHeight:360
+                                                         cropWidth:720
+                                                        cropHeight:1280
+                                                             cropX:0
+                                                             cropY:0];
+  RTC_OBJC_TYPE(RTCVideoFrame) *frame =
+      [[RTC_OBJC_TYPE(RTCVideoFrame) alloc] initWithBuffer:buffer
+                                                  rotation:RTCVideoRotation_0
+                                               timeStampNs:0];
+
+  XCTestExpectation *callbackExpectation = [self expectationWithDescription:@"videoSinkCallback"];
+  ObjCCallbackVideoSink callback_video_sink(^void(RTC_OBJC_TYPE(RTCVideoFrame) * outputFrame) {
+    XCTAssertEqual(outputFrame.width, 480);
+    XCTAssertEqual(outputFrame.height, 270);
+
+    RTC_OBJC_TYPE(RTCCVPixelBuffer) *outputBuffer = outputFrame.buffer;
+    XCTAssertEqual(outputBuffer.cropX, 0);
+    XCTAssertEqual(outputBuffer.cropY, 0);
+    XCTAssertEqual(outputBuffer.cropWidth, 640);
+    XCTAssertEqual(outputBuffer.cropHeight, 360);
+    XCTAssertEqual(buffer.pixelBuffer, outputBuffer.pixelBuffer);
+
+    [callbackExpectation fulfill];
+  });
+
+  const rtc::VideoSinkWants video_sink_wants;
+  rtc::VideoSourceInterface<webrtc::VideoFrame> *video_source_interface = _video_source.get();
+  video_source_interface->AddOrUpdateSink(&callback_video_sink, video_sink_wants);
+
+  _video_source->OnOutputFormatRequest(480, 270, 30);
+  _video_source->OnCapturedFrame(frame);
+
+  [self waitForExpectations:@[ callbackExpectation ] timeout:10.0];
+  CVBufferRelease(pixelBufferRef);
+}
+
+- (void)testOnCapturedFramePreCroppedCVPixelBufferNeedsCropping {
+  CVPixelBufferRef pixelBufferRef = NULL;
+  CVPixelBufferCreate(
+      NULL, 380, 640, kCVPixelFormatType_420YpCbCr8BiPlanarVideoRange, NULL, &pixelBufferRef);
+
+  RTC_OBJC_TYPE(RTCCVPixelBuffer) *buffer =
+      [[RTC_OBJC_TYPE(RTCCVPixelBuffer) alloc] initWithPixelBuffer:pixelBufferRef
+                                                      adaptedWidth:370
+                                                     adaptedHeight:640
+                                                         cropWidth:370
+                                                        cropHeight:640
+                                                             cropX:10
+                                                             cropY:0];
+  RTC_OBJC_TYPE(RTCVideoFrame) *frame =
+      [[RTC_OBJC_TYPE(RTCVideoFrame) alloc] initWithBuffer:buffer
+                                                  rotation:RTCVideoRotation_0
+                                               timeStampNs:0];
+
+  XCTestExpectation *callbackExpectation = [self expectationWithDescription:@"videoSinkCallback"];
+  ObjCCallbackVideoSink callback_video_sink(^void(RTC_OBJC_TYPE(RTCVideoFrame) * outputFrame) {
+    XCTAssertEqual(outputFrame.width, 360);
+    XCTAssertEqual(outputFrame.height, 640);
+
+    RTC_OBJC_TYPE(RTCCVPixelBuffer) *outputBuffer = outputFrame.buffer;
+    XCTAssertEqual(outputBuffer.cropX, 14);
+    XCTAssertEqual(outputBuffer.cropY, 0);
+    XCTAssertEqual(outputBuffer.cropWidth, 360);
+    XCTAssertEqual(outputBuffer.cropHeight, 640);
+    XCTAssertEqual(buffer.pixelBuffer, outputBuffer.pixelBuffer);
+
+    [callbackExpectation fulfill];
+  });
+
+  const rtc::VideoSinkWants video_sink_wants;
+  rtc::VideoSourceInterface<webrtc::VideoFrame> *video_source_interface = _video_source.get();
+  video_source_interface->AddOrUpdateSink(&callback_video_sink, video_sink_wants);
+
+  _video_source->OnOutputFormatRequest(640, 360, 30);
+  _video_source->OnCapturedFrame(frame);
+
+  [self waitForExpectations:@[ callbackExpectation ] timeout:10.0];
+  CVBufferRelease(pixelBufferRef);
+}
+
+- (void)testOnCapturedFrameSmallerPreCroppedCVPixelBufferNeedsCropping {
+  CVPixelBufferRef pixelBufferRef = NULL;
+  CVPixelBufferCreate(
+      NULL, 380, 640, kCVPixelFormatType_420YpCbCr8BiPlanarVideoRange, NULL, &pixelBufferRef);
+
+  RTC_OBJC_TYPE(RTCCVPixelBuffer) *buffer =
+      [[RTC_OBJC_TYPE(RTCCVPixelBuffer) alloc] initWithPixelBuffer:pixelBufferRef
+                                                      adaptedWidth:300
+                                                     adaptedHeight:640
+                                                         cropWidth:300
+                                                        cropHeight:640
+                                                             cropX:40
+                                                             cropY:0];
+  RTC_OBJC_TYPE(RTCVideoFrame) *frame =
+      [[RTC_OBJC_TYPE(RTCVideoFrame) alloc] initWithBuffer:buffer
+                                                  rotation:RTCVideoRotation_0
+                                               timeStampNs:0];
+
+  XCTestExpectation *callbackExpectation = [self expectationWithDescription:@"videoSinkCallback"];
+  ObjCCallbackVideoSink callback_video_sink(^void(RTC_OBJC_TYPE(RTCVideoFrame) * outputFrame) {
+    XCTAssertEqual(outputFrame.width, 300);
+    XCTAssertEqual(outputFrame.height, 534);
+
+    RTC_OBJC_TYPE(RTCCVPixelBuffer) *outputBuffer = outputFrame.buffer;
+    XCTAssertEqual(outputBuffer.cropX, 40);
+    XCTAssertEqual(outputBuffer.cropY, 52);
+    XCTAssertEqual(outputBuffer.cropWidth, 300);
+    XCTAssertEqual(outputBuffer.cropHeight, 534);
+    XCTAssertEqual(buffer.pixelBuffer, outputBuffer.pixelBuffer);
+
+    [callbackExpectation fulfill];
+  });
+
+  const rtc::VideoSinkWants video_sink_wants;
+  rtc::VideoSourceInterface<webrtc::VideoFrame> *video_source_interface = _video_source.get();
+  video_source_interface->AddOrUpdateSink(&callback_video_sink, video_sink_wants);
+
+  _video_source->OnOutputFormatRequest(640, 360, 30);
+  _video_source->OnCapturedFrame(frame);
+
+  [self waitForExpectations:@[ callbackExpectation ] timeout:10.0];
+  CVBufferRelease(pixelBufferRef);
+}
+
+- (void)testOnCapturedFrameI420BufferNeedsAdaptation {
+  rtc::scoped_refptr<webrtc::I420Buffer> i420Buffer = CreateI420Gradient(720, 1280);
+  RTC_OBJC_TYPE(RTCI420Buffer) *buffer =
+      [[RTC_OBJC_TYPE(RTCI420Buffer) alloc] initWithFrameBuffer:i420Buffer];
+  RTC_OBJC_TYPE(RTCVideoFrame) *frame =
+      [[RTC_OBJC_TYPE(RTCVideoFrame) alloc] initWithBuffer:buffer
+                                                  rotation:RTCVideoRotation_0
+                                               timeStampNs:0];
+
+  XCTestExpectation *callbackExpectation = [self expectationWithDescription:@"videoSinkCallback"];
+  ObjCCallbackVideoSink callback_video_sink(^void(RTC_OBJC_TYPE(RTCVideoFrame) * outputFrame) {
+    XCTAssertEqual(outputFrame.width, 360);
+    XCTAssertEqual(outputFrame.height, 640);
+
+    RTC_OBJC_TYPE(RTCI420Buffer) *outputBuffer = (RTC_OBJC_TYPE(RTCI420Buffer) *)outputFrame.buffer;
+
+    double psnr = I420PSNR(*[buffer nativeI420Buffer], *[outputBuffer nativeI420Buffer]);
+    XCTAssertEqual(psnr, webrtc::kPerfectPSNR);
+
+    [callbackExpectation fulfill];
+  });
+
+  const rtc::VideoSinkWants video_sink_wants;
+  rtc::VideoSourceInterface<webrtc::VideoFrame> *video_source_interface = _video_source.get();
+  video_source_interface->AddOrUpdateSink(&callback_video_sink, video_sink_wants);
+
+  _video_source->OnOutputFormatRequest(640, 360, 30);
+  _video_source->OnCapturedFrame(frame);
+
+  [self waitForExpectations:@[ callbackExpectation ] timeout:10.0];
+}
+
+- (void)testOnCapturedFrameI420BufferNeedsCropping {
+  rtc::scoped_refptr<webrtc::I420Buffer> i420Buffer = CreateI420Gradient(380, 640);
+  RTC_OBJC_TYPE(RTCI420Buffer) *buffer =
+      [[RTC_OBJC_TYPE(RTCI420Buffer) alloc] initWithFrameBuffer:i420Buffer];
+  RTC_OBJC_TYPE(RTCVideoFrame) *frame =
+      [[RTC_OBJC_TYPE(RTCVideoFrame) alloc] initWithBuffer:buffer
+                                                  rotation:RTCVideoRotation_0
+                                               timeStampNs:0];
+
+  XCTestExpectation *callbackExpectation = [self expectationWithDescription:@"videoSinkCallback"];
+  ObjCCallbackVideoSink callback_video_sink(^void(RTC_OBJC_TYPE(RTCVideoFrame) * outputFrame) {
+    XCTAssertEqual(outputFrame.width, 360);
+    XCTAssertEqual(outputFrame.height, 640);
+
+    RTC_OBJC_TYPE(RTCI420Buffer) *outputBuffer = (RTC_OBJC_TYPE(RTCI420Buffer) *)outputFrame.buffer;
+
+    double psnr = I420PSNR(*[buffer nativeI420Buffer], *[outputBuffer nativeI420Buffer]);
+    XCTAssertGreaterThanOrEqual(psnr, 40);
+
+    [callbackExpectation fulfill];
+  });
+
+  const rtc::VideoSinkWants video_sink_wants;
+  rtc::VideoSourceInterface<webrtc::VideoFrame> *video_source_interface = _video_source.get();
+  video_source_interface->AddOrUpdateSink(&callback_video_sink, video_sink_wants);
+
+  _video_source->OnOutputFormatRequest(640, 360, 30);
+  _video_source->OnCapturedFrame(frame);
+
+  [self waitForExpectations:@[ callbackExpectation ] timeout:10.0];
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/unittests/RTCAudioDeviceModule_xctest.mm b/third_party/libwebrtc/sdk/objc/unittests/RTCAudioDeviceModule_xctest.mm
new file mode 100644
index 0000000000..f8ce844652
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/unittests/RTCAudioDeviceModule_xctest.mm
@@ -0,0 +1,593 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <XCTest/XCTest.h>
+
+#if defined(WEBRTC_IOS)
+#import "sdk/objc/native/api/audio_device_module.h"
+#endif
+
+#include "api/scoped_refptr.h"
+
+typedef int32_t(^NeedMorePlayDataBlock)(const size_t nSamples,
+                                        const size_t nBytesPerSample,
+                                        const size_t nChannels,
+                                        const uint32_t samplesPerSec,
+                                        void* audioSamples,
+                                        size_t& nSamplesOut,
+                                        int64_t* elapsed_time_ms,
+                                        int64_t* ntp_time_ms);
+
+typedef int32_t(^RecordedDataIsAvailableBlock)(const void* audioSamples,
+                                               const size_t nSamples,
+                                               const size_t nBytesPerSample,
+                                               const size_t nChannels,
+                                               const uint32_t samplesPerSec,
+                                               const uint32_t totalDelayMS,
+                                               const int32_t clockDrift,
+                                               const uint32_t currentMicLevel,
+                                               const bool keyPressed,
+                                               uint32_t& newMicLevel);
+
+
+// This class implements the AudioTransport API and forwards all methods to the appropriate blocks.
+class MockAudioTransport : public webrtc::AudioTransport {
+public:
+  MockAudioTransport() {}
+  ~MockAudioTransport() override {}
+
+  void expectNeedMorePlayData(NeedMorePlayDataBlock block) {
+    needMorePlayDataBlock = block;
+  }
+
+  void expectRecordedDataIsAvailable(RecordedDataIsAvailableBlock block) {
+    recordedDataIsAvailableBlock = block;
+  }
+
+  int32_t NeedMorePlayData(const size_t nSamples,
+                           const size_t nBytesPerSample,
+                           const size_t nChannels,
+                           const uint32_t samplesPerSec,
+                           void* audioSamples,
+                           size_t& nSamplesOut,
+                           int64_t* elapsed_time_ms,
+                           int64_t* ntp_time_ms) override {
+    return needMorePlayDataBlock(nSamples,
+                                 nBytesPerSample,
+                                 nChannels,
+                                 samplesPerSec,
+                                 audioSamples,
+                                 nSamplesOut,
+                                 elapsed_time_ms,
+                                 ntp_time_ms);
+  }
+
+  int32_t RecordedDataIsAvailable(const void* audioSamples,
+                                  const size_t nSamples,
+                                  const size_t nBytesPerSample,
+                                  const size_t nChannels,
+                                  const uint32_t samplesPerSec,
+                                  const uint32_t totalDelayMS,
+                                  const int32_t clockDrift,
+                                  const uint32_t currentMicLevel,
+                                  const bool keyPressed,
+                                  uint32_t& newMicLevel) override {
+    return recordedDataIsAvailableBlock(audioSamples,
+                                        nSamples,
+                                        nBytesPerSample,
+                                        nChannels,
+                                        samplesPerSec,
+                                        totalDelayMS,
+                                        clockDrift,
+                                        currentMicLevel,
+                                        keyPressed,
+                                        newMicLevel);
+  }
+
+  void PullRenderData(int bits_per_sample,
+                      int sample_rate,
+                      size_t number_of_channels,
+                      size_t number_of_frames,
+                      void* audio_data,
+                      int64_t* elapsed_time_ms,
+                      int64_t* ntp_time_ms) override {}
+
+ private:
+  NeedMorePlayDataBlock needMorePlayDataBlock;
+  RecordedDataIsAvailableBlock recordedDataIsAvailableBlock;
+};
+
+// Number of callbacks (input or output) the tests waits for before we set
+// an event indicating that the test was OK.
+static const NSUInteger kNumCallbacks = 10;
+// Max amount of time we wait for an event to be set while counting callbacks.
+static const NSTimeInterval kTestTimeOutInSec = 20.0;
+// Number of bits per PCM audio sample.
+static const NSUInteger kBitsPerSample = 16;
+// Number of bytes per PCM audio sample.
+static const NSUInteger kBytesPerSample = kBitsPerSample / 8;
+// Average number of audio callbacks per second assuming 10ms packet size.
+static const NSUInteger kNumCallbacksPerSecond = 100;
+// Play out a test file during this time (unit is in seconds).
+static const NSUInteger kFilePlayTimeInSec = 15;
+// Run the full-duplex test during this time (unit is in seconds).
+// Note that first `kNumIgnoreFirstCallbacks` are ignored.
+static const NSUInteger kFullDuplexTimeInSec = 10;
+// Wait for the callback sequence to stabilize by ignoring this amount of the
+// initial callbacks (avoids initial FIFO access).
+// Only used in the RunPlayoutAndRecordingInFullDuplex test.
+static const NSUInteger kNumIgnoreFirstCallbacks = 50;
+
+@interface RTCAudioDeviceModuleTests : XCTestCase {
+  rtc::scoped_refptr<webrtc::AudioDeviceModule> audioDeviceModule;
+  MockAudioTransport mock;
+}
+
+@property(nonatomic, assign) webrtc::AudioParameters playoutParameters;
+@property(nonatomic, assign) webrtc::AudioParameters recordParameters;
+
+@end
+
+@implementation RTCAudioDeviceModuleTests
+
+@synthesize playoutParameters;
+@synthesize recordParameters;
+
+- (void)setUp {
+  [super setUp];
+  audioDeviceModule = webrtc::CreateAudioDeviceModule();
+  XCTAssertEqual(0, audioDeviceModule->Init());
+  XCTAssertEqual(0, audioDeviceModule->GetPlayoutAudioParameters(&playoutParameters));
+  XCTAssertEqual(0, audioDeviceModule->GetRecordAudioParameters(&recordParameters));
+}
+
+- (void)tearDown {
+  XCTAssertEqual(0, audioDeviceModule->Terminate());
+  audioDeviceModule = nullptr;
+  [super tearDown];
+}
+
+- (void)startPlayout {
+  XCTAssertFalse(audioDeviceModule->Playing());
+  XCTAssertEqual(0, audioDeviceModule->InitPlayout());
+  XCTAssertTrue(audioDeviceModule->PlayoutIsInitialized());
+  XCTAssertEqual(0, audioDeviceModule->StartPlayout());
+  XCTAssertTrue(audioDeviceModule->Playing());
+}
+
+- (void)stopPlayout {
+  XCTAssertEqual(0, audioDeviceModule->StopPlayout());
+  XCTAssertFalse(audioDeviceModule->Playing());
+}
+
+- (void)startRecording{
+  XCTAssertFalse(audioDeviceModule->Recording());
+  XCTAssertEqual(0, audioDeviceModule->InitRecording());
+  XCTAssertTrue(audioDeviceModule->RecordingIsInitialized());
+  XCTAssertEqual(0, audioDeviceModule->StartRecording());
+  XCTAssertTrue(audioDeviceModule->Recording());
+}
+
+- (void)stopRecording{
+  XCTAssertEqual(0, audioDeviceModule->StopRecording());
+  XCTAssertFalse(audioDeviceModule->Recording());
+}
+
+- (NSURL*)fileURLForSampleRate:(int)sampleRate {
+  XCTAssertTrue(sampleRate == 48000 || sampleRate == 44100 || sampleRate == 16000);
+  NSString *filename = [NSString stringWithFormat:@"audio_short%d", sampleRate / 1000];
+  NSURL *url = [[NSBundle mainBundle] URLForResource:filename withExtension:@"pcm"];
+  XCTAssertNotNil(url);
+
+  return url;
+}
+
+#pragma mark - Tests
+
+- (void)testConstructDestruct {
+  // Using the test fixture to create and destruct the audio device module.
+}
+
+- (void)testInitTerminate {
+  // Initialization is part of the test fixture.
+  XCTAssertTrue(audioDeviceModule->Initialized());
+  XCTAssertEqual(0, audioDeviceModule->Terminate());
+  XCTAssertFalse(audioDeviceModule->Initialized());
+}
+
+// Tests that playout can be initiated, started and stopped. No audio callback
+// is registered in this test.
+- (void)testStartStopPlayout {
+  [self startPlayout];
+  [self stopPlayout];
+  [self startPlayout];
+  [self stopPlayout];
+}
+
+// Tests that recording can be initiated, started and stopped. No audio callback
+// is registered in this test.
+- (void)testStartStopRecording {
+  [self startRecording];
+  [self stopRecording];
+  [self startRecording];
+  [self stopRecording];
+}
+// Verify that calling StopPlayout() will leave us in an uninitialized state
+// which will require a new call to InitPlayout(). This test does not call
+// StartPlayout() while being uninitialized since doing so will hit a
+// RTC_DCHECK.
+- (void)testStopPlayoutRequiresInitToRestart {
+  XCTAssertEqual(0, audioDeviceModule->InitPlayout());
+  XCTAssertEqual(0, audioDeviceModule->StartPlayout());
+  XCTAssertEqual(0, audioDeviceModule->StopPlayout());
+  XCTAssertFalse(audioDeviceModule->PlayoutIsInitialized());
+}
+
+// Verify that we can create two ADMs and start playing on the second ADM.
+// Only the first active instance shall activate an audio session and the
+// last active instance shall deactivate the audio session. The test does not
+// explicitly verify correct audio session calls but instead focuses on
+// ensuring that audio starts for both ADMs.
+- (void)testStartPlayoutOnTwoInstances {
+  // Create and initialize a second/extra ADM instance. The default ADM is
+  // created by the test harness.
+  rtc::scoped_refptr<webrtc::AudioDeviceModule> secondAudioDeviceModule =
+      webrtc::CreateAudioDeviceModule();
+  XCTAssertNotEqual(secondAudioDeviceModule.get(), nullptr);
+  XCTAssertEqual(0, secondAudioDeviceModule->Init());
+
+  // Start playout for the default ADM but don't wait here. Instead use the
+  // upcoming second stream for that. We set the same expectation on number
+  // of callbacks as for the second stream.
+  mock.expectNeedMorePlayData(^int32_t(const size_t nSamples,
+                                       const size_t nBytesPerSample,
+                                       const size_t nChannels,
+                                       const uint32_t samplesPerSec,
+                                       void *audioSamples,
+                                       size_t &nSamplesOut,
+                                       int64_t *elapsed_time_ms,
+                                       int64_t *ntp_time_ms) {
+    nSamplesOut = nSamples;
+    XCTAssertEqual(nSamples, self.playoutParameters.frames_per_10ms_buffer());
+    XCTAssertEqual(nBytesPerSample, kBytesPerSample);
+    XCTAssertEqual(nChannels, self.playoutParameters.channels());
+    XCTAssertEqual((int)samplesPerSec, self.playoutParameters.sample_rate());
+    XCTAssertNotEqual((void*)NULL, audioSamples);
+
+    return 0;
+  });
+
+  XCTAssertEqual(0, audioDeviceModule->RegisterAudioCallback(&mock));
+  [self startPlayout];
+
+  // Initialize playout for the second ADM. If all is OK, the second ADM shall
+  // reuse the audio session activated when the first ADM started playing.
+  // This call will also ensure that we avoid a problem related to initializing
+  // two different audio unit instances back to back (see webrtc:5166 for
+  // details).
+  XCTAssertEqual(0, secondAudioDeviceModule->InitPlayout());
+  XCTAssertTrue(secondAudioDeviceModule->PlayoutIsInitialized());
+
+  // Start playout for the second ADM and verify that it starts as intended.
+  // Passing this test ensures that initialization of the second audio unit
+  // has been done successfully and that there is no conflict with the already
+  // playing first ADM.
+  XCTestExpectation *playoutExpectation = [self expectationWithDescription:@"NeedMorePlayoutData"];
+  __block int num_callbacks = 0;
+
+  MockAudioTransport mock2;
+  mock2.expectNeedMorePlayData(^int32_t(const size_t nSamples,
+                                        const size_t nBytesPerSample,
+                                        const size_t nChannels,
+                                        const uint32_t samplesPerSec,
+                                        void *audioSamples,
+                                        size_t &nSamplesOut,
+                                        int64_t *elapsed_time_ms,
+                                        int64_t *ntp_time_ms) {
+    nSamplesOut = nSamples;
+    XCTAssertEqual(nSamples, self.playoutParameters.frames_per_10ms_buffer());
+    XCTAssertEqual(nBytesPerSample, kBytesPerSample);
+    XCTAssertEqual(nChannels, self.playoutParameters.channels());
+    XCTAssertEqual((int)samplesPerSec, self.playoutParameters.sample_rate());
+    XCTAssertNotEqual((void*)NULL, audioSamples);
+    if (++num_callbacks == kNumCallbacks) {
+      [playoutExpectation fulfill];
+    }
+
+    return 0;
+  });
+
+  XCTAssertEqual(0, secondAudioDeviceModule->RegisterAudioCallback(&mock2));
+  XCTAssertEqual(0, secondAudioDeviceModule->StartPlayout());
+  XCTAssertTrue(secondAudioDeviceModule->Playing());
+  [self waitForExpectationsWithTimeout:kTestTimeOutInSec handler:nil];
+  [self stopPlayout];
+  XCTAssertEqual(0, secondAudioDeviceModule->StopPlayout());
+  XCTAssertFalse(secondAudioDeviceModule->Playing());
+  XCTAssertFalse(secondAudioDeviceModule->PlayoutIsInitialized());
+
+  XCTAssertEqual(0, secondAudioDeviceModule->Terminate());
+}
+
+// Start playout and verify that the native audio layer starts asking for real
+// audio samples to play out using the NeedMorePlayData callback.
+- (void)testStartPlayoutVerifyCallbacks {
+
+  XCTestExpectation *playoutExpectation = [self expectationWithDescription:@"NeedMorePlayoutData"];
+  __block int num_callbacks = 0;
+  mock.expectNeedMorePlayData(^int32_t(const size_t nSamples,
+                                       const size_t nBytesPerSample,
+                                       const size_t nChannels,
+                                       const uint32_t samplesPerSec,
+                                       void *audioSamples,
+                                       size_t &nSamplesOut,
+                                       int64_t *elapsed_time_ms,
+                                       int64_t *ntp_time_ms) {
+    nSamplesOut = nSamples;
+    XCTAssertEqual(nSamples, self.playoutParameters.frames_per_10ms_buffer());
+    XCTAssertEqual(nBytesPerSample, kBytesPerSample);
+    XCTAssertEqual(nChannels, self.playoutParameters.channels());
+    XCTAssertEqual((int)samplesPerSec, self.playoutParameters.sample_rate());
+    XCTAssertNotEqual((void*)NULL, audioSamples);
+    if (++num_callbacks == kNumCallbacks) {
+      [playoutExpectation fulfill];
+    }
+    return 0;
+  });
+
+  XCTAssertEqual(0, audioDeviceModule->RegisterAudioCallback(&mock));
+
+  [self startPlayout];
+  [self waitForExpectationsWithTimeout:kTestTimeOutInSec handler:nil];
+  [self stopPlayout];
+}
+
+// Start recording and verify that the native audio layer starts feeding real
+// audio samples via the RecordedDataIsAvailable callback.
+- (void)testStartRecordingVerifyCallbacks {
+  XCTestExpectation *recordExpectation =
+  [self expectationWithDescription:@"RecordedDataIsAvailable"];
+  __block int num_callbacks = 0;
+
+  mock.expectRecordedDataIsAvailable(^(const void* audioSamples,
+                                       const size_t nSamples,
+                                       const size_t nBytesPerSample,
+                                       const size_t nChannels,
+                                       const uint32_t samplesPerSec,
+                                       const uint32_t totalDelayMS,
+                                       const int32_t clockDrift,
+                                       const uint32_t currentMicLevel,
+                                       const bool keyPressed,
+                                       uint32_t& newMicLevel) {
+    XCTAssertNotEqual((void*)NULL, audioSamples);
+    XCTAssertEqual(nSamples, self.recordParameters.frames_per_10ms_buffer());
+    XCTAssertEqual(nBytesPerSample, kBytesPerSample);
+    XCTAssertEqual(nChannels, self.recordParameters.channels());
+    XCTAssertEqual((int)samplesPerSec, self.recordParameters.sample_rate());
+    XCTAssertEqual(0, clockDrift);
+    XCTAssertEqual(0u, currentMicLevel);
+    XCTAssertFalse(keyPressed);
+    if (++num_callbacks == kNumCallbacks) {
+      [recordExpectation fulfill];
+    }
+
+    return 0;
+  });
+
+  XCTAssertEqual(0, audioDeviceModule->RegisterAudioCallback(&mock));
+  [self startRecording];
+  [self waitForExpectationsWithTimeout:kTestTimeOutInSec handler:nil];
+  [self stopRecording];
+}
+
+// Start playout and recording (full-duplex audio) and verify that audio is
+// active in both directions.
+- (void)testStartPlayoutAndRecordingVerifyCallbacks {
+  XCTestExpectation *playoutExpectation = [self expectationWithDescription:@"NeedMorePlayoutData"];
+  __block NSUInteger callbackCount = 0;
+
+  XCTestExpectation *recordExpectation =
+  [self expectationWithDescription:@"RecordedDataIsAvailable"];
+  recordExpectation.expectedFulfillmentCount = kNumCallbacks;
+
+  mock.expectNeedMorePlayData(^int32_t(const size_t nSamples,
+                                       const size_t nBytesPerSample,
+                                       const size_t nChannels,
+                                       const uint32_t samplesPerSec,
+                                       void *audioSamples,
+                                       size_t &nSamplesOut,
+                                       int64_t *elapsed_time_ms,
+                                       int64_t *ntp_time_ms) {
+    nSamplesOut = nSamples;
+    XCTAssertEqual(nSamples, self.playoutParameters.frames_per_10ms_buffer());
+    XCTAssertEqual(nBytesPerSample, kBytesPerSample);
+    XCTAssertEqual(nChannels, self.playoutParameters.channels());
+    XCTAssertEqual((int)samplesPerSec, self.playoutParameters.sample_rate());
+    XCTAssertNotEqual((void*)NULL, audioSamples);
+    if (callbackCount++ >= kNumCallbacks) {
+      [playoutExpectation fulfill];
+    }
+
+    return 0;
+  });
+
+  mock.expectRecordedDataIsAvailable(^(const void* audioSamples,
+                                       const size_t nSamples,
+                                       const size_t nBytesPerSample,
+                                       const size_t nChannels,
+                                       const uint32_t samplesPerSec,
+                                       const uint32_t totalDelayMS,
+                                       const int32_t clockDrift,
+                                       const uint32_t currentMicLevel,
+                                       const bool keyPressed,
+                                       uint32_t& newMicLevel) {
+    XCTAssertNotEqual((void*)NULL, audioSamples);
+    XCTAssertEqual(nSamples, self.recordParameters.frames_per_10ms_buffer());
+    XCTAssertEqual(nBytesPerSample, kBytesPerSample);
+    XCTAssertEqual(nChannels, self.recordParameters.channels());
+    XCTAssertEqual((int)samplesPerSec, self.recordParameters.sample_rate());
+    XCTAssertEqual(0, clockDrift);
+    XCTAssertEqual(0u, currentMicLevel);
+    XCTAssertFalse(keyPressed);
+    [recordExpectation fulfill];
+
+    return 0;
+  });
+
+  XCTAssertEqual(0, audioDeviceModule->RegisterAudioCallback(&mock));
+  [self startPlayout];
+  [self startRecording];
+  [self waitForExpectationsWithTimeout:kTestTimeOutInSec handler:nil];
+  [self stopRecording];
+  [self stopPlayout];
+}
+
+// Start playout and read audio from an external PCM file when the audio layer
+// asks for data to play out. Real audio is played out in this test but it does
+// not contain any explicit verification that the audio quality is perfect.
+- (void)testRunPlayoutWithFileAsSource {
+  XCTAssertEqual(1u, playoutParameters.channels());
+
+  // Using XCTestExpectation to count callbacks is very slow.
+  XCTestExpectation *playoutExpectation = [self expectationWithDescription:@"NeedMorePlayoutData"];
+  const int expectedCallbackCount = kFilePlayTimeInSec * kNumCallbacksPerSecond;
+  __block int callbackCount = 0;
+
+  NSURL *fileURL = [self fileURLForSampleRate:playoutParameters.sample_rate()];
+  NSInputStream *inputStream = [[NSInputStream alloc] initWithURL:fileURL];
+
+  mock.expectNeedMorePlayData(^int32_t(const size_t nSamples,
+                                       const size_t nBytesPerSample,
+                                       const size_t nChannels,
+                                       const uint32_t samplesPerSec,
+                                       void *audioSamples,
+                                       size_t &nSamplesOut,
+                                       int64_t *elapsed_time_ms,
+                                       int64_t *ntp_time_ms) {
+    [inputStream read:(uint8_t *)audioSamples maxLength:nSamples*nBytesPerSample*nChannels];
+    nSamplesOut = nSamples;
+    if (callbackCount++ == expectedCallbackCount) {
+      [playoutExpectation fulfill];
+    }
+
+    return 0;
+  });
+
+  XCTAssertEqual(0, audioDeviceModule->RegisterAudioCallback(&mock));
+  [self startPlayout];
+  NSTimeInterval waitTimeout = kFilePlayTimeInSec * 2.0;
+  [self waitForExpectationsWithTimeout:waitTimeout handler:nil];
+  [self stopPlayout];
+}
+
+- (void)testDevices {
+  // Device enumeration is not supported. Verify fixed values only.
+  XCTAssertEqual(1, audioDeviceModule->PlayoutDevices());
+  XCTAssertEqual(1, audioDeviceModule->RecordingDevices());
+}
+
+// Start playout and recording and store recorded data in an intermediate FIFO
+// buffer from which the playout side then reads its samples in the same order
+// as they were stored. Under ideal circumstances, a callback sequence would
+// look like: ...+-+-+-+-+-+-+-..., where '+' means 'packet recorded' and '-'
+// means 'packet played'. Under such conditions, the FIFO would only contain
+// one packet on average. However, under more realistic conditions, the size
+// of the FIFO will vary more due to an unbalance between the two sides.
+// This test tries to verify that the device maintains a balanced callback-
+// sequence by running in loopback for ten seconds while measuring the size
+// (max and average) of the FIFO. The size of the FIFO is increased by the
+// recording side and decreased by the playout side.
+// TODO(henrika): tune the final test parameters after running tests on several
+// different devices.
+- (void)testRunPlayoutAndRecordingInFullDuplex {
+  XCTAssertEqual(recordParameters.channels(), playoutParameters.channels());
+  XCTAssertEqual(recordParameters.sample_rate(), playoutParameters.sample_rate());
+
+  XCTestExpectation *playoutExpectation = [self expectationWithDescription:@"NeedMorePlayoutData"];
+  __block NSUInteger playoutCallbacks = 0;
+  NSUInteger expectedPlayoutCallbacks = kFullDuplexTimeInSec * kNumCallbacksPerSecond;
+
+  // FIFO queue and measurements
+  NSMutableArray *fifoBuffer = [NSMutableArray arrayWithCapacity:20];
+  __block NSUInteger fifoMaxSize = 0;
+  __block NSUInteger fifoTotalWrittenElements = 0;
+  __block NSUInteger fifoWriteCount = 0;
+
+  mock.expectRecordedDataIsAvailable(^(const void* audioSamples,
+                                       const size_t nSamples,
+                                       const size_t nBytesPerSample,
+                                       const size_t nChannels,
+                                       const uint32_t samplesPerSec,
+                                       const uint32_t totalDelayMS,
+                                       const int32_t clockDrift,
+                                       const uint32_t currentMicLevel,
+                                       const bool keyPressed,
+                                       uint32_t& newMicLevel) {
+    if (fifoWriteCount++ < kNumIgnoreFirstCallbacks) {
+      return 0;
+    }
+
+    NSData *data = [NSData dataWithBytes:audioSamples length:nSamples*nBytesPerSample*nChannels];
+    @synchronized(fifoBuffer) {
+      [fifoBuffer addObject:data];
+      fifoMaxSize = MAX(fifoMaxSize, fifoBuffer.count);
+      fifoTotalWrittenElements += fifoBuffer.count;
+    }
+
+    return 0;
+  });
+
+  mock.expectNeedMorePlayData(^int32_t(const size_t nSamples,
+                                       const size_t nBytesPerSample,
+                                       const size_t nChannels,
+                                       const uint32_t samplesPerSec,
+                                       void *audioSamples,
+                                       size_t &nSamplesOut,
+                                       int64_t *elapsed_time_ms,
+                                       int64_t *ntp_time_ms) {
+    nSamplesOut = nSamples;
+    NSData *data;
+    @synchronized(fifoBuffer) {
+      data = fifoBuffer.firstObject;
+      if (data) {
+        [fifoBuffer removeObjectAtIndex:0];
+      }
+    }
+
+    if (data) {
+      memcpy(audioSamples, (char*) data.bytes, data.length);
+    } else {
+      memset(audioSamples, 0, nSamples*nBytesPerSample*nChannels);
+    }
+
+    if (playoutCallbacks++ == expectedPlayoutCallbacks) {
+      [playoutExpectation fulfill];
+    }
+    return 0;
+  });
+
+  XCTAssertEqual(0, audioDeviceModule->RegisterAudioCallback(&mock));
+  [self startRecording];
+  [self startPlayout];
+  NSTimeInterval waitTimeout = kFullDuplexTimeInSec * 2.0;
+  [self waitForExpectationsWithTimeout:waitTimeout handler:nil];
+
+  size_t fifoAverageSize =
+      (fifoTotalWrittenElements == 0)
+        ? 0.0
+        : 0.5 + (double)fifoTotalWrittenElements / (fifoWriteCount - kNumIgnoreFirstCallbacks);
+
+  [self stopPlayout];
+  [self stopRecording];
+  XCTAssertLessThan(fifoAverageSize, 10u);
+  XCTAssertLessThan(fifoMaxSize, 20u);
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/unittests/RTCAudioDevice_xctest.mm b/third_party/libwebrtc/sdk/objc/unittests/RTCAudioDevice_xctest.mm
new file mode 100644
index 0000000000..e01fdbd6e3
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/unittests/RTCAudioDevice_xctest.mm
@@ -0,0 +1,115 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <XCTest/XCTest.h>
+
+#include "api/task_queue/default_task_queue_factory.h"
+
+#import "sdk/objc/components/audio/RTCAudioSession+Private.h"
+#import "sdk/objc/native/api/audio_device_module.h"
+#import "sdk/objc/native/src/audio/audio_device_ios.h"
+
+@interface RTCAudioDeviceTests : XCTestCase {
+  rtc::scoped_refptr<webrtc::AudioDeviceModule> _audioDeviceModule;
+  std::unique_ptr<webrtc::ios_adm::AudioDeviceIOS> _audio_device;
+}
+
+@property(nonatomic) RTC_OBJC_TYPE(RTCAudioSession) * audioSession;
+
+@end
+
+@implementation RTCAudioDeviceTests
+
+@synthesize audioSession = _audioSession;
+
+- (void)setUp {
+  [super setUp];
+
+  _audioDeviceModule = webrtc::CreateAudioDeviceModule();
+  _audio_device.reset(new webrtc::ios_adm::AudioDeviceIOS(/*bypass_voice_processing=*/false));
+  self.audioSession = [RTC_OBJC_TYPE(RTCAudioSession) sharedInstance];
+
+  NSError *error = nil;
+  [self.audioSession lockForConfiguration];
+  [self.audioSession setCategory:AVAudioSessionCategoryPlayAndRecord withOptions:0 error:&error];
+  XCTAssertNil(error);
+
+  [self.audioSession setMode:AVAudioSessionModeVoiceChat error:&error];
+  XCTAssertNil(error);
+
+  [self.audioSession setActive:YES error:&error];
+  XCTAssertNil(error);
+
+  [self.audioSession unlockForConfiguration];
+}
+
+- (void)tearDown {
+  _audio_device->Terminate();
+  _audio_device.reset(nullptr);
+  _audioDeviceModule = nullptr;
+  [self.audioSession notifyDidEndInterruptionWithShouldResumeSession:NO];
+
+  [super tearDown];
+}
+
+// Verifies that the AudioDeviceIOS is_interrupted_ flag is reset correctly
+// after an iOS AVAudioSessionInterruptionTypeEnded notification event.
+// AudioDeviceIOS listens to RTC_OBJC_TYPE(RTCAudioSession) interrupted notifications by:
+// - In AudioDeviceIOS.InitPlayOrRecord registers its audio_session_observer_
+//   callback with RTC_OBJC_TYPE(RTCAudioSession)'s delegate list.
+// - When RTC_OBJC_TYPE(RTCAudioSession) receives an iOS audio interrupted notification, it
+//   passes the notification to callbacks in its delegate list which sets
+//   AudioDeviceIOS's is_interrupted_ flag to true.
+// - When AudioDeviceIOS.ShutdownPlayOrRecord is called, its
+//   audio_session_observer_ callback is removed from RTCAudioSessions's
+//   delegate list.
+//   So if RTC_OBJC_TYPE(RTCAudioSession) receives an iOS end audio interruption notification,
+//   AudioDeviceIOS is not notified as its callback is not in RTC_OBJC_TYPE(RTCAudioSession)'s
+//   delegate list. This causes AudioDeviceIOS's is_interrupted_ flag to be in
+//   the wrong (true) state and the audio session will ignore audio changes.
+// As RTC_OBJC_TYPE(RTCAudioSession) keeps its own interrupted state, the fix is to initialize
+// AudioDeviceIOS's is_interrupted_ flag to RTC_OBJC_TYPE(RTCAudioSession)'s isInterrupted
+// flag in AudioDeviceIOS.InitPlayOrRecord.
+- (void)testInterruptedAudioSession {
+  XCTAssertTrue(self.audioSession.isActive);
+  XCTAssertTrue([self.audioSession.category isEqual:AVAudioSessionCategoryPlayAndRecord] ||
+                [self.audioSession.category isEqual:AVAudioSessionCategoryPlayback]);
+  XCTAssertEqual(AVAudioSessionModeVoiceChat, self.audioSession.mode);
+
+  std::unique_ptr<webrtc::TaskQueueFactory> task_queue_factory =
+      webrtc::CreateDefaultTaskQueueFactory();
+  std::unique_ptr<webrtc::AudioDeviceBuffer> audio_buffer;
+  audio_buffer.reset(new webrtc::AudioDeviceBuffer(task_queue_factory.get()));
+  _audio_device->AttachAudioBuffer(audio_buffer.get());
+  XCTAssertEqual(webrtc::AudioDeviceGeneric::InitStatus::OK, _audio_device->Init());
+  XCTAssertEqual(0, _audio_device->InitPlayout());
+  XCTAssertEqual(0, _audio_device->StartPlayout());
+
+  // Force interruption.
+  [self.audioSession notifyDidBeginInterruption];
+
+  // Wait for notification to propagate.
+  rtc::ThreadManager::ProcessAllMessageQueuesForTesting();
+  XCTAssertTrue(_audio_device->IsInterrupted());
+
+  // Force it for testing.
+  _audio_device->StopPlayout();
+
+  [self.audioSession notifyDidEndInterruptionWithShouldResumeSession:YES];
+  // Wait for notification to propagate.
+  rtc::ThreadManager::ProcessAllMessageQueuesForTesting();
+  XCTAssertTrue(_audio_device->IsInterrupted());
+
+  _audio_device->Init();
+  _audio_device->InitPlayout();
+  XCTAssertFalse(_audio_device->IsInterrupted());
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/unittests/RTCAudioSessionTest.mm b/third_party/libwebrtc/sdk/objc/unittests/RTCAudioSessionTest.mm
new file mode 100644
index 0000000000..2a00909abe
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/unittests/RTCAudioSessionTest.mm
@@ -0,0 +1,324 @@
+/*
+ *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+#import <OCMock/OCMock.h>
+#import <XCTest/XCTest.h>
+
+#include <vector>
+
+#include "rtc_base/event.h"
+#include "rtc_base/gunit.h"
+
+#import "components/audio/RTCAudioSession+Private.h"
+
+#import "components/audio/RTCAudioSession.h"
+#import "components/audio/RTCAudioSessionConfiguration.h"
+
+@interface RTC_OBJC_TYPE (RTCAudioSession)
+(UnitTesting)
+
+    @property(nonatomic,
+              readonly) std::vector<__weak id<RTC_OBJC_TYPE(RTCAudioSessionDelegate)> > delegates;
+
+- (instancetype)initWithAudioSession:(id)audioSession;
+
+@end
+
+@interface MockAVAudioSession : NSObject
+
+@property (nonatomic, readwrite, assign) float outputVolume;
+
+@end
+
+@implementation MockAVAudioSession
+@synthesize outputVolume = _outputVolume;
+@end
+
+@interface RTCAudioSessionTestDelegate : NSObject <RTC_OBJC_TYPE (RTCAudioSessionDelegate)>
+
+@property (nonatomic, readonly) float outputVolume;
+
+@end
+
+@implementation RTCAudioSessionTestDelegate
+
+@synthesize outputVolume = _outputVolume;
+
+- (instancetype)init {
+  if (self = [super init]) {
+    _outputVolume = -1;
+  }
+  return self;
+}
+
+- (void)audioSessionDidBeginInterruption:(RTC_OBJC_TYPE(RTCAudioSession) *)session {
+}
+
+- (void)audioSessionDidEndInterruption:(RTC_OBJC_TYPE(RTCAudioSession) *)session
+                   shouldResumeSession:(BOOL)shouldResumeSession {
+}
+
+- (void)audioSessionDidChangeRoute:(RTC_OBJC_TYPE(RTCAudioSession) *)session
+                            reason:(AVAudioSessionRouteChangeReason)reason
+                     previousRoute:(AVAudioSessionRouteDescription *)previousRoute {
+}
+
+- (void)audioSessionMediaServerTerminated:(RTC_OBJC_TYPE(RTCAudioSession) *)session {
+}
+
+- (void)audioSessionMediaServerReset:(RTC_OBJC_TYPE(RTCAudioSession) *)session {
+}
+
+- (void)audioSessionShouldConfigure:(RTC_OBJC_TYPE(RTCAudioSession) *)session {
+}
+
+- (void)audioSessionShouldUnconfigure:(RTC_OBJC_TYPE(RTCAudioSession) *)session {
+}
+
+- (void)audioSession:(RTC_OBJC_TYPE(RTCAudioSession) *)audioSession
+    didChangeOutputVolume:(float)outputVolume {
+  _outputVolume = outputVolume;
+}
+
+@end
+
+// A delegate that adds itself to the audio session on init and removes itself
+// in its dealloc.
+@interface RTCTestRemoveOnDeallocDelegate : RTCAudioSessionTestDelegate
+@end
+
+@implementation RTCTestRemoveOnDeallocDelegate
+
+- (instancetype)init {
+  if (self = [super init]) {
+    RTC_OBJC_TYPE(RTCAudioSession) *session = [RTC_OBJC_TYPE(RTCAudioSession) sharedInstance];
+    [session addDelegate:self];
+  }
+  return self;
+}
+
+- (void)dealloc {
+  RTC_OBJC_TYPE(RTCAudioSession) *session = [RTC_OBJC_TYPE(RTCAudioSession) sharedInstance];
+  [session removeDelegate:self];
+}
+
+@end
+
+@interface RTCAudioSessionTest : XCTestCase
+
+@end
+
+@implementation RTCAudioSessionTest
+
+- (void)testAddAndRemoveDelegates {
+  RTC_OBJC_TYPE(RTCAudioSession) *session = [RTC_OBJC_TYPE(RTCAudioSession) sharedInstance];
+  NSMutableArray *delegates = [NSMutableArray array];
+  const size_t count = 5;
+  for (size_t i = 0; i < count; ++i) {
+    RTCAudioSessionTestDelegate *delegate =
+        [[RTCAudioSessionTestDelegate alloc] init];
+    [session addDelegate:delegate];
+    [delegates addObject:delegate];
+    EXPECT_EQ(i + 1, session.delegates.size());
+  }
+  [delegates enumerateObjectsUsingBlock:^(RTCAudioSessionTestDelegate *obj,
+                                          NSUInteger idx,
+                                          BOOL *stop) {
+    [session removeDelegate:obj];
+  }];
+  EXPECT_EQ(0u, session.delegates.size());
+}
+
+- (void)testPushDelegate {
+  RTC_OBJC_TYPE(RTCAudioSession) *session = [RTC_OBJC_TYPE(RTCAudioSession) sharedInstance];
+  NSMutableArray *delegates = [NSMutableArray array];
+  const size_t count = 2;
+  for (size_t i = 0; i < count; ++i) {
+    RTCAudioSessionTestDelegate *delegate =
+        [[RTCAudioSessionTestDelegate alloc] init];
+    [session addDelegate:delegate];
+    [delegates addObject:delegate];
+  }
+  // Test that it gets added to the front of the list.
+  RTCAudioSessionTestDelegate *pushedDelegate =
+      [[RTCAudioSessionTestDelegate alloc] init];
+  [session pushDelegate:pushedDelegate];
+  EXPECT_TRUE(pushedDelegate == session.delegates[0]);
+
+  // Test that it stays at the front of the list.
+  for (size_t i = 0; i < count; ++i) {
+    RTCAudioSessionTestDelegate *delegate =
+        [[RTCAudioSessionTestDelegate alloc] init];
+    [session addDelegate:delegate];
+    [delegates addObject:delegate];
+  }
+  EXPECT_TRUE(pushedDelegate == session.delegates[0]);
+
+  // Test that the next one goes to the front too.
+  pushedDelegate = [[RTCAudioSessionTestDelegate alloc] init];
+  [session pushDelegate:pushedDelegate];
+  EXPECT_TRUE(pushedDelegate == session.delegates[0]);
+}
+
+// Tests that delegates added to the audio session properly zero out. This is
+// checking an implementation detail (that vectors of __weak work as expected).
+- (void)testZeroingWeakDelegate {
+  RTC_OBJC_TYPE(RTCAudioSession) *session = [RTC_OBJC_TYPE(RTCAudioSession) sharedInstance];
+  @autoreleasepool {
+    // Add a delegate to the session. There should be one delegate at this
+    // point.
+    RTCAudioSessionTestDelegate *delegate =
+        [[RTCAudioSessionTestDelegate alloc] init];
+    [session addDelegate:delegate];
+    EXPECT_EQ(1u, session.delegates.size());
+    EXPECT_TRUE(session.delegates[0]);
+  }
+  // The previously created delegate should've de-alloced, leaving a nil ptr.
+  EXPECT_FALSE(session.delegates[0]);
+  RTCAudioSessionTestDelegate *delegate =
+      [[RTCAudioSessionTestDelegate alloc] init];
+  [session addDelegate:delegate];
+  // On adding a new delegate, nil ptrs should've been cleared.
+  EXPECT_EQ(1u, session.delegates.size());
+  EXPECT_TRUE(session.delegates[0]);
+}
+
+// Tests that we don't crash when removing delegates in dealloc.
+// Added as a regression test.
+- (void)testRemoveDelegateOnDealloc {
+  @autoreleasepool {
+    RTCTestRemoveOnDeallocDelegate *delegate =
+        [[RTCTestRemoveOnDeallocDelegate alloc] init];
+    EXPECT_TRUE(delegate);
+  }
+  RTC_OBJC_TYPE(RTCAudioSession) *session = [RTC_OBJC_TYPE(RTCAudioSession) sharedInstance];
+  EXPECT_EQ(0u, session.delegates.size());
+}
+
+- (void)testAudioSessionActivation {
+  RTC_OBJC_TYPE(RTCAudioSession) *audioSession = [RTC_OBJC_TYPE(RTCAudioSession) sharedInstance];
+  EXPECT_EQ(0, audioSession.activationCount);
+  [audioSession audioSessionDidActivate:[AVAudioSession sharedInstance]];
+  EXPECT_EQ(1, audioSession.activationCount);
+  [audioSession audioSessionDidDeactivate:[AVAudioSession sharedInstance]];
+  EXPECT_EQ(0, audioSession.activationCount);
+}
+
+// Hack - fixes OCMVerify link error
+// Link error is: Undefined symbols for architecture i386:
+// "OCMMakeLocation(objc_object*, char const*, int)", referenced from:
+// -[RTCAudioSessionTest testConfigureWebRTCSession] in RTCAudioSessionTest.o
+// ld: symbol(s) not found for architecture i386
+// REASON: https://github.com/erikdoe/ocmock/issues/238
+OCMLocation *OCMMakeLocation(id testCase, const char *fileCString, int line){
+  return [OCMLocation locationWithTestCase:testCase
+                                      file:[NSString stringWithUTF8String:fileCString]
+                                      line:line];
+}
+
+- (void)testConfigureWebRTCSession {
+  NSError *error = nil;
+
+  void (^setActiveBlock)(NSInvocation *invocation) = ^(NSInvocation *invocation) {
+    __autoreleasing NSError **retError;
+    [invocation getArgument:&retError atIndex:4];
+    *retError = [NSError errorWithDomain:@"AVAudioSession"
+                                    code:AVAudioSessionErrorCodeCannotInterruptOthers
+                                userInfo:nil];
+    BOOL failure = NO;
+    [invocation setReturnValue:&failure];
+  };
+
+  id mockAVAudioSession = OCMPartialMock([AVAudioSession sharedInstance]);
+  OCMStub([[mockAVAudioSession ignoringNonObjectArgs] setActive:YES
+                                                    withOptions:0
+                                                          error:([OCMArg anyObjectRef])])
+      .andDo(setActiveBlock);
+
+  id mockAudioSession = OCMPartialMock([RTC_OBJC_TYPE(RTCAudioSession) sharedInstance]);
+  OCMStub([mockAudioSession session]).andReturn(mockAVAudioSession);
+
+  RTC_OBJC_TYPE(RTCAudioSession) *audioSession = mockAudioSession;
+  EXPECT_EQ(0, audioSession.activationCount);
+  [audioSession lockForConfiguration];
+  // configureWebRTCSession is forced to fail in the above mock interface,
+  // so activationCount should remain 0
+  OCMExpect([[mockAVAudioSession ignoringNonObjectArgs] setActive:YES
+                                                      withOptions:0
+                                                            error:([OCMArg anyObjectRef])])
+      .andDo(setActiveBlock);
+  OCMExpect([mockAudioSession session]).andReturn(mockAVAudioSession);
+  EXPECT_FALSE([audioSession configureWebRTCSession:&error]);
+  EXPECT_EQ(0, audioSession.activationCount);
+
+  id session = audioSession.session;
+  EXPECT_EQ(session, mockAVAudioSession);
+  EXPECT_EQ(NO, [mockAVAudioSession setActive:YES withOptions:0 error:&error]);
+  [audioSession unlockForConfiguration];
+
+  OCMVerify([mockAudioSession session]);
+  OCMVerify([[mockAVAudioSession ignoringNonObjectArgs] setActive:YES withOptions:0 error:&error]);
+  OCMVerify([[mockAVAudioSession ignoringNonObjectArgs] setActive:NO withOptions:0 error:&error]);
+
+  [mockAVAudioSession stopMocking];
+  [mockAudioSession stopMocking];
+}
+
+- (void)testConfigureWebRTCSessionWithoutLocking {
+  NSError *error = nil;
+
+  id mockAVAudioSession = OCMPartialMock([AVAudioSession sharedInstance]);
+  id mockAudioSession = OCMPartialMock([RTC_OBJC_TYPE(RTCAudioSession) sharedInstance]);
+  OCMStub([mockAudioSession session]).andReturn(mockAVAudioSession);
+
+  RTC_OBJC_TYPE(RTCAudioSession) *audioSession = mockAudioSession;
+
+  std::unique_ptr<rtc::Thread> thread = rtc::Thread::Create();
+  EXPECT_TRUE(thread);
+  EXPECT_TRUE(thread->Start());
+
+  rtc::Event waitLock;
+  rtc::Event waitCleanup;
+  constexpr webrtc::TimeDelta timeout = webrtc::TimeDelta::Seconds(5);
+  thread->PostTask([audioSession, &waitLock, &waitCleanup, timeout] {
+    [audioSession lockForConfiguration];
+    waitLock.Set();
+    waitCleanup.Wait(timeout);
+    [audioSession unlockForConfiguration];
+  });
+
+  waitLock.Wait(timeout);
+  [audioSession setCategory:AVAudioSessionCategoryPlayAndRecord withOptions:0 error:&error];
+  EXPECT_TRUE(error != nil);
+  EXPECT_EQ(error.domain, kRTCAudioSessionErrorDomain);
+  EXPECT_EQ(error.code, kRTCAudioSessionErrorLockRequired);
+  waitCleanup.Set();
+  thread->Stop();
+
+  [mockAVAudioSession stopMocking];
+  [mockAudioSession stopMocking];
+}
+
+- (void)testAudioVolumeDidNotify {
+  MockAVAudioSession *mockAVAudioSession = [[MockAVAudioSession alloc] init];
+  RTC_OBJC_TYPE(RTCAudioSession) *session =
+      [[RTC_OBJC_TYPE(RTCAudioSession) alloc] initWithAudioSession:mockAVAudioSession];
+  RTCAudioSessionTestDelegate *delegate =
+      [[RTCAudioSessionTestDelegate alloc] init];
+  [session addDelegate:delegate];
+
+  float expectedVolume = 0.75;
+  mockAVAudioSession.outputVolume = expectedVolume;
+
+  EXPECT_EQ(expectedVolume, delegate.outputVolume);
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/unittests/RTCCVPixelBuffer_xctest.mm b/third_party/libwebrtc/sdk/objc/unittests/RTCCVPixelBuffer_xctest.mm
new file mode 100644
index 0000000000..cf759c5243
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/unittests/RTCCVPixelBuffer_xctest.mm
@@ -0,0 +1,461 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+#import <XCTest/XCTest.h>
+
+#import "components/video_frame_buffer/RTCCVPixelBuffer.h"
+
+#import "api/video_frame_buffer/RTCNativeI420Buffer+Private.h"
+#import "base/RTCVideoFrame.h"
+#import "base/RTCVideoFrameBuffer.h"
+#import "frame_buffer_helpers.h"
+
+#include "common_video/libyuv/include/webrtc_libyuv.h"
+#include "third_party/libyuv/include/libyuv.h"
+
+namespace {
+
+struct ToI420WithCropAndScaleSetting {
+  int inputWidth;
+  int inputHeight;
+  int offsetX;
+  int offsetY;
+  int cropWidth;
+  int cropHeight;
+  int scaleWidth;
+  int scaleHeight;
+};
+
+constexpr const ToI420WithCropAndScaleSetting kToI420WithCropAndScaleSettings[] = {
+    ToI420WithCropAndScaleSetting{
+        .inputWidth = 640,
+        .inputHeight = 360,
+        .offsetX = 0,
+        .offsetY = 0,
+        .cropWidth = 640,
+        .cropHeight = 360,
+        .scaleWidth = 320,
+        .scaleHeight = 180,
+    },
+    ToI420WithCropAndScaleSetting{
+        .inputWidth = 640,
+        .inputHeight = 360,
+        .offsetX = 160,
+        .offsetY = 90,
+        .cropWidth = 160,
+        .cropHeight = 90,
+        .scaleWidth = 320,
+        .scaleHeight = 180,
+    },
+};
+
+}  // namespace
+
+@interface RTCCVPixelBufferTests : XCTestCase
+@end
+
+@implementation RTCCVPixelBufferTests {
+}
+
+- (void)testRequiresCroppingNoCrop {
+  CVPixelBufferRef pixelBufferRef = NULL;
+  CVPixelBufferCreate(
+      NULL, 720, 1280, kCVPixelFormatType_420YpCbCr8BiPlanarVideoRange, NULL, &pixelBufferRef);
+  RTC_OBJC_TYPE(RTCCVPixelBuffer) *buffer =
+      [[RTC_OBJC_TYPE(RTCCVPixelBuffer) alloc] initWithPixelBuffer:pixelBufferRef];
+
+  XCTAssertFalse([buffer requiresCropping]);
+
+  CVBufferRelease(pixelBufferRef);
+}
+
+- (void)testRequiresCroppingWithCrop {
+  CVPixelBufferRef pixelBufferRef = NULL;
+  CVPixelBufferCreate(
+      NULL, 720, 1280, kCVPixelFormatType_420YpCbCr8BiPlanarVideoRange, NULL, &pixelBufferRef);
+  RTC_OBJC_TYPE(RTCCVPixelBuffer) *croppedBuffer =
+      [[RTC_OBJC_TYPE(RTCCVPixelBuffer) alloc] initWithPixelBuffer:pixelBufferRef
+                                                      adaptedWidth:720
+                                                     adaptedHeight:1280
+                                                         cropWidth:360
+                                                        cropHeight:640
+                                                             cropX:100
+                                                             cropY:100];
+
+  XCTAssertTrue([croppedBuffer requiresCropping]);
+
+  CVBufferRelease(pixelBufferRef);
+}
+
+- (void)testRequiresScalingNoScale {
+  CVPixelBufferRef pixelBufferRef = NULL;
+  CVPixelBufferCreate(
+      NULL, 720, 1280, kCVPixelFormatType_420YpCbCr8BiPlanarVideoRange, NULL, &pixelBufferRef);
+
+  RTC_OBJC_TYPE(RTCCVPixelBuffer) *buffer =
+      [[RTC_OBJC_TYPE(RTCCVPixelBuffer) alloc] initWithPixelBuffer:pixelBufferRef];
+  XCTAssertFalse([buffer requiresScalingToWidth:720 height:1280]);
+
+  CVBufferRelease(pixelBufferRef);
+}
+
+- (void)testRequiresScalingWithScale {
+  CVPixelBufferRef pixelBufferRef = NULL;
+  CVPixelBufferCreate(
+      NULL, 720, 1280, kCVPixelFormatType_420YpCbCr8BiPlanarVideoRange, NULL, &pixelBufferRef);
+
+  RTC_OBJC_TYPE(RTCCVPixelBuffer) *buffer =
+      [[RTC_OBJC_TYPE(RTCCVPixelBuffer) alloc] initWithPixelBuffer:pixelBufferRef];
+  XCTAssertTrue([buffer requiresScalingToWidth:360 height:640]);
+
+  CVBufferRelease(pixelBufferRef);
+}
+
+- (void)testRequiresScalingWithScaleAndMatchingCrop {
+  CVPixelBufferRef pixelBufferRef = NULL;
+  CVPixelBufferCreate(
+      NULL, 720, 1280, kCVPixelFormatType_420YpCbCr8BiPlanarVideoRange, NULL, &pixelBufferRef);
+
+  RTC_OBJC_TYPE(RTCCVPixelBuffer) *buffer =
+      [[RTC_OBJC_TYPE(RTCCVPixelBuffer) alloc] initWithPixelBuffer:pixelBufferRef
+                                                      adaptedWidth:720
+                                                     adaptedHeight:1280
+                                                         cropWidth:360
+                                                        cropHeight:640
+                                                             cropX:100
+                                                             cropY:100];
+  XCTAssertFalse([buffer requiresScalingToWidth:360 height:640]);
+
+  CVBufferRelease(pixelBufferRef);
+}
+
+- (void)testBufferSize_NV12 {
+  CVPixelBufferRef pixelBufferRef = NULL;
+  CVPixelBufferCreate(
+      NULL, 720, 1280, kCVPixelFormatType_420YpCbCr8BiPlanarVideoRange, NULL, &pixelBufferRef);
+
+  RTC_OBJC_TYPE(RTCCVPixelBuffer) *buffer =
+      [[RTC_OBJC_TYPE(RTCCVPixelBuffer) alloc] initWithPixelBuffer:pixelBufferRef];
+  XCTAssertEqual([buffer bufferSizeForCroppingAndScalingToWidth:360 height:640], 576000);
+
+  CVBufferRelease(pixelBufferRef);
+}
+
+- (void)testBufferSize_RGB {
+  CVPixelBufferRef pixelBufferRef = NULL;
+  CVPixelBufferCreate(NULL, 720, 1280, kCVPixelFormatType_32BGRA, NULL, &pixelBufferRef);
+
+  RTC_OBJC_TYPE(RTCCVPixelBuffer) *buffer =
+      [[RTC_OBJC_TYPE(RTCCVPixelBuffer) alloc] initWithPixelBuffer:pixelBufferRef];
+  XCTAssertEqual([buffer bufferSizeForCroppingAndScalingToWidth:360 height:640], 0);
+
+  CVBufferRelease(pixelBufferRef);
+}
+
+- (void)testCropAndScale_NV12 {
+  [self cropAndScaleTestWithNV12];
+}
+
+- (void)testCropAndScaleNoOp_NV12 {
+  [self cropAndScaleTestWithNV12InputFormat:kCVPixelFormatType_420YpCbCr8BiPlanarVideoRange
+                               outputFormat:kCVPixelFormatType_420YpCbCr8BiPlanarVideoRange
+                                 outputSize:CGSizeMake(720, 1280)];
+}
+
+- (void)testCropAndScale_NV12FullToVideo {
+  [self cropAndScaleTestWithNV12InputFormat:kCVPixelFormatType_420YpCbCr8BiPlanarFullRange
+                               outputFormat:kCVPixelFormatType_420YpCbCr8BiPlanarVideoRange];
+}
+
+- (void)testCropAndScaleZeroSizeFrame_NV12 {
+  [self cropAndScaleTestWithNV12InputFormat:kCVPixelFormatType_420YpCbCr8BiPlanarVideoRange
+                               outputFormat:kCVPixelFormatType_420YpCbCr8BiPlanarVideoRange
+                                 outputSize:CGSizeMake(0, 0)];
+}
+
+- (void)testCropAndScaleToSmallFormat_NV12 {
+  [self cropAndScaleTestWithNV12InputFormat:kCVPixelFormatType_420YpCbCr8BiPlanarVideoRange
+                               outputFormat:kCVPixelFormatType_420YpCbCr8BiPlanarVideoRange
+                                 outputSize:CGSizeMake(148, 320)];
+}
+
+- (void)testCropAndScaleToOddFormat_NV12 {
+  [self cropAndScaleTestWithNV12InputFormat:kCVPixelFormatType_420YpCbCr8BiPlanarVideoRange
+                               outputFormat:kCVPixelFormatType_420YpCbCr8BiPlanarVideoRange
+                                 outputSize:CGSizeMake(361, 640)];
+}
+
+- (void)testCropAndScale_32BGRA {
+  [self cropAndScaleTestWithRGBPixelFormat:kCVPixelFormatType_32BGRA];
+}
+
+- (void)testCropAndScale_32ARGB {
+  [self cropAndScaleTestWithRGBPixelFormat:kCVPixelFormatType_32ARGB];
+}
+
+- (void)testCropAndScaleWithSmallCropInfo_32ARGB {
+  [self cropAndScaleTestWithRGBPixelFormat:kCVPixelFormatType_32ARGB cropX:2 cropY:3];
+}
+
+- (void)testCropAndScaleWithLargeCropInfo_32ARGB {
+  [self cropAndScaleTestWithRGBPixelFormat:kCVPixelFormatType_32ARGB cropX:200 cropY:300];
+}
+
+- (void)testToI420_NV12 {
+  [self toI420WithPixelFormat:kCVPixelFormatType_420YpCbCr8BiPlanarVideoRange];
+}
+
+- (void)testToI420_32BGRA {
+  [self toI420WithPixelFormat:kCVPixelFormatType_32BGRA];
+}
+
+- (void)testToI420_32ARGB {
+  [self toI420WithPixelFormat:kCVPixelFormatType_32ARGB];
+}
+
+- (void)testToI420WithCropAndScale_NV12 {
+  for (const auto &setting : kToI420WithCropAndScaleSettings) {
+    [self toI420WithCropAndScaleWithPixelFormat:kCVPixelFormatType_420YpCbCr8BiPlanarVideoRange
+                                        setting:setting];
+  }
+}
+
+- (void)testToI420WithCropAndScale_32BGRA {
+  for (const auto &setting : kToI420WithCropAndScaleSettings) {
+    [self toI420WithCropAndScaleWithPixelFormat:kCVPixelFormatType_32BGRA setting:setting];
+  }
+}
+
+- (void)testToI420WithCropAndScale_32ARGB {
+  for (const auto &setting : kToI420WithCropAndScaleSettings) {
+    [self toI420WithCropAndScaleWithPixelFormat:kCVPixelFormatType_32ARGB setting:setting];
+  }
+}
+
+- (void)testScaleBufferTest {
+  CVPixelBufferRef pixelBufferRef = NULL;
+  CVPixelBufferCreate(
+      NULL, 1920, 1080, kCVPixelFormatType_420YpCbCr8BiPlanarVideoRange, NULL, &pixelBufferRef);
+
+  rtc::scoped_refptr<webrtc::I420Buffer> i420Buffer = CreateI420Gradient(1920, 1080);
+  CopyI420BufferToCVPixelBuffer(i420Buffer, pixelBufferRef);
+
+  RTC_OBJC_TYPE(RTCCVPixelBuffer) *buffer =
+      [[RTC_OBJC_TYPE(RTCCVPixelBuffer) alloc] initWithPixelBuffer:pixelBufferRef];
+
+  XCTAssertEqual(buffer.width, 1920);
+  XCTAssertEqual(buffer.height, 1080);
+  XCTAssertEqual(buffer.cropX, 0);
+  XCTAssertEqual(buffer.cropY, 0);
+  XCTAssertEqual(buffer.cropWidth, 1920);
+  XCTAssertEqual(buffer.cropHeight, 1080);
+
+  RTC_OBJC_TYPE(RTCCVPixelBuffer) *buffer2 =
+      (RTC_OBJC_TYPE(RTCCVPixelBuffer) *)[buffer cropAndScaleWith:320
+                                                          offsetY:180
+                                                        cropWidth:1280
+                                                       cropHeight:720
+                                                       scaleWidth:960
+                                                      scaleHeight:540];
+
+  XCTAssertEqual(buffer2.width, 960);
+  XCTAssertEqual(buffer2.height, 540);
+  XCTAssertEqual(buffer2.cropX, 320);
+  XCTAssertEqual(buffer2.cropY, 180);
+  XCTAssertEqual(buffer2.cropWidth, 1280);
+  XCTAssertEqual(buffer2.cropHeight, 720);
+
+  RTC_OBJC_TYPE(RTCCVPixelBuffer) *buffer3 =
+      (RTC_OBJC_TYPE(RTCCVPixelBuffer) *)[buffer2 cropAndScaleWith:240
+                                                           offsetY:135
+                                                         cropWidth:480
+                                                        cropHeight:270
+                                                        scaleWidth:320
+                                                       scaleHeight:180];
+
+  XCTAssertEqual(buffer3.width, 320);
+  XCTAssertEqual(buffer3.height, 180);
+  XCTAssertEqual(buffer3.cropX, 640);
+  XCTAssertEqual(buffer3.cropY, 360);
+  XCTAssertEqual(buffer3.cropWidth, 640);
+  XCTAssertEqual(buffer3.cropHeight, 360);
+
+  CVBufferRelease(pixelBufferRef);
+}
+
+#pragma mark - Shared test code
+
+- (void)cropAndScaleTestWithNV12 {
+  [self cropAndScaleTestWithNV12InputFormat:kCVPixelFormatType_420YpCbCr8BiPlanarVideoRange
+                               outputFormat:kCVPixelFormatType_420YpCbCr8BiPlanarVideoRange];
+}
+
+- (void)cropAndScaleTestWithNV12InputFormat:(OSType)inputFormat outputFormat:(OSType)outputFormat {
+  [self cropAndScaleTestWithNV12InputFormat:(OSType)inputFormat
+                               outputFormat:(OSType)outputFormat
+                                 outputSize:CGSizeMake(360, 640)];
+}
+
+- (void)cropAndScaleTestWithNV12InputFormat:(OSType)inputFormat
+                               outputFormat:(OSType)outputFormat
+                                 outputSize:(CGSize)outputSize {
+  CVPixelBufferRef pixelBufferRef = NULL;
+  CVPixelBufferCreate(NULL, 720, 1280, inputFormat, NULL, &pixelBufferRef);
+
+  rtc::scoped_refptr<webrtc::I420Buffer> i420Buffer = CreateI420Gradient(720, 1280);
+  CopyI420BufferToCVPixelBuffer(i420Buffer, pixelBufferRef);
+
+  RTC_OBJC_TYPE(RTCCVPixelBuffer) *buffer =
+      [[RTC_OBJC_TYPE(RTCCVPixelBuffer) alloc] initWithPixelBuffer:pixelBufferRef];
+  XCTAssertEqual(buffer.width, 720);
+  XCTAssertEqual(buffer.height, 1280);
+
+  CVPixelBufferRef outputPixelBufferRef = NULL;
+  CVPixelBufferCreate(
+      NULL, outputSize.width, outputSize.height, outputFormat, NULL, &outputPixelBufferRef);
+
+  std::vector<uint8_t> frameScaleBuffer;
+  if ([buffer requiresScalingToWidth:outputSize.width height:outputSize.height]) {
+    int size =
+        [buffer bufferSizeForCroppingAndScalingToWidth:outputSize.width height:outputSize.height];
+    frameScaleBuffer.resize(size);
+  } else {
+    frameScaleBuffer.clear();
+  }
+  frameScaleBuffer.shrink_to_fit();
+
+  [buffer cropAndScaleTo:outputPixelBufferRef withTempBuffer:frameScaleBuffer.data()];
+
+  RTC_OBJC_TYPE(RTCCVPixelBuffer) *scaledBuffer =
+      [[RTC_OBJC_TYPE(RTCCVPixelBuffer) alloc] initWithPixelBuffer:outputPixelBufferRef];
+  XCTAssertEqual(scaledBuffer.width, outputSize.width);
+  XCTAssertEqual(scaledBuffer.height, outputSize.height);
+
+  if (outputSize.width > 0 && outputSize.height > 0) {
+    RTC_OBJC_TYPE(RTCI420Buffer) *originalBufferI420 = [buffer toI420];
+    RTC_OBJC_TYPE(RTCI420Buffer) *scaledBufferI420 = [scaledBuffer toI420];
+    double psnr =
+        I420PSNR(*[originalBufferI420 nativeI420Buffer], *[scaledBufferI420 nativeI420Buffer]);
+    XCTAssertEqual(psnr, webrtc::kPerfectPSNR);
+  }
+
+  CVBufferRelease(pixelBufferRef);
+}
+
+- (void)cropAndScaleTestWithRGBPixelFormat:(OSType)pixelFormat {
+  [self cropAndScaleTestWithRGBPixelFormat:pixelFormat cropX:0 cropY:0];
+}
+
+- (void)cropAndScaleTestWithRGBPixelFormat:(OSType)pixelFormat cropX:(int)cropX cropY:(int)cropY {
+  CVPixelBufferRef pixelBufferRef = NULL;
+  CVPixelBufferCreate(NULL, 720, 1280, pixelFormat, NULL, &pixelBufferRef);
+
+  DrawGradientInRGBPixelBuffer(pixelBufferRef);
+
+  RTC_OBJC_TYPE(RTCCVPixelBuffer) *buffer = [[RTC_OBJC_TYPE(RTCCVPixelBuffer) alloc]
+      initWithPixelBuffer:pixelBufferRef
+             adaptedWidth:CVPixelBufferGetWidth(pixelBufferRef)
+            adaptedHeight:CVPixelBufferGetHeight(pixelBufferRef)
+                cropWidth:CVPixelBufferGetWidth(pixelBufferRef) - cropX
+               cropHeight:CVPixelBufferGetHeight(pixelBufferRef) - cropY
+                    cropX:cropX
+                    cropY:cropY];
+
+  XCTAssertEqual(buffer.width, 720);
+  XCTAssertEqual(buffer.height, 1280);
+
+  CVPixelBufferRef outputPixelBufferRef = NULL;
+  CVPixelBufferCreate(NULL, 360, 640, pixelFormat, NULL, &outputPixelBufferRef);
+  [buffer cropAndScaleTo:outputPixelBufferRef withTempBuffer:NULL];
+
+  RTC_OBJC_TYPE(RTCCVPixelBuffer) *scaledBuffer =
+      [[RTC_OBJC_TYPE(RTCCVPixelBuffer) alloc] initWithPixelBuffer:outputPixelBufferRef];
+  XCTAssertEqual(scaledBuffer.width, 360);
+  XCTAssertEqual(scaledBuffer.height, 640);
+
+  RTC_OBJC_TYPE(RTCI420Buffer) *originalBufferI420 = [buffer toI420];
+  RTC_OBJC_TYPE(RTCI420Buffer) *scaledBufferI420 = [scaledBuffer toI420];
+  double psnr =
+      I420PSNR(*[originalBufferI420 nativeI420Buffer], *[scaledBufferI420 nativeI420Buffer]);
+  XCTAssertEqual(psnr, webrtc::kPerfectPSNR);
+
+  CVBufferRelease(pixelBufferRef);
+}
+
+- (void)toI420WithPixelFormat:(OSType)pixelFormat {
+  rtc::scoped_refptr<webrtc::I420Buffer> i420Buffer = CreateI420Gradient(360, 640);
+
+  CVPixelBufferRef pixelBufferRef = NULL;
+  CVPixelBufferCreate(NULL, 360, 640, pixelFormat, NULL, &pixelBufferRef);
+
+  CopyI420BufferToCVPixelBuffer(i420Buffer, pixelBufferRef);
+
+  RTC_OBJC_TYPE(RTCCVPixelBuffer) *buffer =
+      [[RTC_OBJC_TYPE(RTCCVPixelBuffer) alloc] initWithPixelBuffer:pixelBufferRef];
+  RTC_OBJC_TYPE(RTCI420Buffer) *fromCVPixelBuffer = [buffer toI420];
+
+  double psnr = I420PSNR(*i420Buffer, *[fromCVPixelBuffer nativeI420Buffer]);
+  double target = webrtc::kPerfectPSNR;
+  if (pixelFormat != kCVPixelFormatType_420YpCbCr8BiPlanarVideoRange) {
+    // libyuv's I420ToRGB functions seem to lose some quality.
+    target = 19.0;
+  }
+  XCTAssertGreaterThanOrEqual(psnr, target);
+
+  CVBufferRelease(pixelBufferRef);
+}
+
+- (void)toI420WithCropAndScaleWithPixelFormat:(OSType)pixelFormat
+                                      setting:(const ToI420WithCropAndScaleSetting &)setting {
+  rtc::scoped_refptr<webrtc::I420Buffer> i420Buffer =
+      CreateI420Gradient(setting.inputWidth, setting.inputHeight);
+
+  CVPixelBufferRef pixelBufferRef = NULL;
+  CVPixelBufferCreate(
+      NULL, setting.inputWidth, setting.inputHeight, pixelFormat, NULL, &pixelBufferRef);
+
+  CopyI420BufferToCVPixelBuffer(i420Buffer, pixelBufferRef);
+
+  RTC_OBJC_TYPE(RTCI420Buffer) *objcI420Buffer =
+      [[RTC_OBJC_TYPE(RTCI420Buffer) alloc] initWithFrameBuffer:i420Buffer];
+  RTC_OBJC_TYPE(RTCI420Buffer) *scaledObjcI420Buffer =
+      (RTC_OBJC_TYPE(RTCI420Buffer) *)[objcI420Buffer cropAndScaleWith:setting.offsetX
+                                                               offsetY:setting.offsetY
+                                                             cropWidth:setting.cropWidth
+                                                            cropHeight:setting.cropHeight
+                                                            scaleWidth:setting.scaleWidth
+                                                           scaleHeight:setting.scaleHeight];
+  RTC_OBJC_TYPE(RTCCVPixelBuffer) *buffer =
+      [[RTC_OBJC_TYPE(RTCCVPixelBuffer) alloc] initWithPixelBuffer:pixelBufferRef];
+  id<RTC_OBJC_TYPE(RTCVideoFrameBuffer)> scaledBuffer =
+      [buffer cropAndScaleWith:setting.offsetX
+                       offsetY:setting.offsetY
+                     cropWidth:setting.cropWidth
+                    cropHeight:setting.cropHeight
+                    scaleWidth:setting.scaleWidth
+                   scaleHeight:setting.scaleHeight];
+  XCTAssertTrue([scaledBuffer isKindOfClass:[RTC_OBJC_TYPE(RTCCVPixelBuffer) class]]);
+
+  RTC_OBJC_TYPE(RTCI420Buffer) *fromCVPixelBuffer = [scaledBuffer toI420];
+
+  double psnr =
+      I420PSNR(*[scaledObjcI420Buffer nativeI420Buffer], *[fromCVPixelBuffer nativeI420Buffer]);
+  double target = webrtc::kPerfectPSNR;
+  if (pixelFormat != kCVPixelFormatType_420YpCbCr8BiPlanarVideoRange) {
+    // libyuv's I420ToRGB functions seem to lose some quality.
+    target = 19.0;
+  }
+  XCTAssertGreaterThanOrEqual(psnr, target);
+
+  CVBufferRelease(pixelBufferRef);
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/unittests/RTCCallbackLogger_xctest.m b/third_party/libwebrtc/sdk/objc/unittests/RTCCallbackLogger_xctest.m
new file mode 100644
index 0000000000..1b6fb1c07b
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/unittests/RTCCallbackLogger_xctest.m
@@ -0,0 +1,244 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "api/logging/RTCCallbackLogger.h"
+
+#import <XCTest/XCTest.h>
+
+@interface RTCCallbackLoggerTests : XCTestCase
+
+@property(nonatomic, strong) RTC_OBJC_TYPE(RTCCallbackLogger) * logger;
+
+@end
+
+@implementation RTCCallbackLoggerTests
+
+@synthesize logger;
+
+- (void)setUp {
+  self.logger = [[RTC_OBJC_TYPE(RTCCallbackLogger) alloc] init];
+}
+
+- (void)tearDown {
+  self.logger = nil;
+}
+
+- (void)testDefaultSeverityLevel {
+  XCTAssertEqual(self.logger.severity, RTCLoggingSeverityInfo);
+}
+
+- (void)testCallbackGetsCalledForAppropriateLevel {
+  self.logger.severity = RTCLoggingSeverityWarning;
+
+  XCTestExpectation *callbackExpectation = [self expectationWithDescription:@"callbackWarning"];
+
+  [self.logger start:^(NSString *message) {
+    XCTAssertTrue([message hasSuffix:@"Horrible error\n"]);
+    [callbackExpectation fulfill];
+  }];
+
+  RTCLogError("Horrible error");
+
+  [self waitForExpectations:@[ callbackExpectation ] timeout:10.0];
+}
+
+- (void)testCallbackWithSeverityGetsCalledForAppropriateLevel {
+  self.logger.severity = RTCLoggingSeverityWarning;
+
+  XCTestExpectation *callbackExpectation = [self expectationWithDescription:@"callbackWarning"];
+
+  [self.logger
+      startWithMessageAndSeverityHandler:^(NSString *message, RTCLoggingSeverity severity) {
+        XCTAssertTrue([message hasSuffix:@"Horrible error\n"]);
+        XCTAssertEqual(severity, RTCLoggingSeverityError);
+        [callbackExpectation fulfill];
+      }];
+
+  RTCLogError("Horrible error");
+
+  [self waitForExpectations:@[ callbackExpectation ] timeout:10.0];
+}
+
+- (void)testCallbackDoesNotGetCalledForOtherLevels {
+  self.logger.severity = RTCLoggingSeverityError;
+
+  XCTestExpectation *callbackExpectation = [self expectationWithDescription:@"callbackError"];
+
+  [self.logger start:^(NSString *message) {
+    XCTAssertTrue([message hasSuffix:@"Horrible error\n"]);
+    [callbackExpectation fulfill];
+  }];
+
+  RTCLogInfo("Just some info");
+  RTCLogWarning("Warning warning");
+  RTCLogError("Horrible error");
+
+  [self waitForExpectations:@[ callbackExpectation ] timeout:10.0];
+}
+
+- (void)testCallbackWithSeverityDoesNotGetCalledForOtherLevels {
+  self.logger.severity = RTCLoggingSeverityError;
+
+  XCTestExpectation *callbackExpectation = [self expectationWithDescription:@"callbackError"];
+
+  [self.logger
+      startWithMessageAndSeverityHandler:^(NSString *message, RTCLoggingSeverity severity) {
+        XCTAssertTrue([message hasSuffix:@"Horrible error\n"]);
+        XCTAssertEqual(severity, RTCLoggingSeverityError);
+        [callbackExpectation fulfill];
+      }];
+
+  RTCLogInfo("Just some info");
+  RTCLogWarning("Warning warning");
+  RTCLogError("Horrible error");
+
+  [self waitForExpectations:@[ callbackExpectation ] timeout:10.0];
+}
+
+- (void)testCallbackDoesNotgetCalledForSeverityNone {
+  self.logger.severity = RTCLoggingSeverityNone;
+
+  XCTestExpectation *callbackExpectation = [self expectationWithDescription:@"unexpectedCallback"];
+
+  [self.logger start:^(NSString *message) {
+    [callbackExpectation fulfill];
+    XCTAssertTrue(false);
+  }];
+
+  RTCLogInfo("Just some info");
+  RTCLogWarning("Warning warning");
+  RTCLogError("Horrible error");
+
+  XCTWaiter *waiter = [[XCTWaiter alloc] init];
+  XCTWaiterResult result = [waiter waitForExpectations:@[ callbackExpectation ] timeout:1.0];
+  XCTAssertEqual(result, XCTWaiterResultTimedOut);
+}
+
+- (void)testCallbackWithSeverityDoesNotgetCalledForSeverityNone {
+  self.logger.severity = RTCLoggingSeverityNone;
+
+  XCTestExpectation *callbackExpectation = [self expectationWithDescription:@"unexpectedCallback"];
+
+  [self.logger
+      startWithMessageAndSeverityHandler:^(NSString *message, RTCLoggingSeverity severity) {
+        [callbackExpectation fulfill];
+        XCTAssertTrue(false);
+      }];
+
+  RTCLogInfo("Just some info");
+  RTCLogWarning("Warning warning");
+  RTCLogError("Horrible error");
+
+  XCTWaiter *waiter = [[XCTWaiter alloc] init];
+  XCTWaiterResult result = [waiter waitForExpectations:@[ callbackExpectation ] timeout:1.0];
+  XCTAssertEqual(result, XCTWaiterResultTimedOut);
+}
+
+- (void)testStartingWithNilCallbackDoesNotCrash {
+  [self.logger start:nil];
+
+  RTCLogError("Horrible error");
+}
+
+- (void)testStartingWithNilCallbackWithSeverityDoesNotCrash {
+  [self.logger startWithMessageAndSeverityHandler:nil];
+
+  RTCLogError("Horrible error");
+}
+
+- (void)testStopCallbackLogger {
+  XCTestExpectation *callbackExpectation = [self expectationWithDescription:@"stopped"];
+
+  [self.logger start:^(NSString *message) {
+    [callbackExpectation fulfill];
+  }];
+
+  [self.logger stop];
+
+  RTCLogInfo("Just some info");
+
+  XCTWaiter *waiter = [[XCTWaiter alloc] init];
+  XCTWaiterResult result = [waiter waitForExpectations:@[ callbackExpectation ] timeout:1.0];
+  XCTAssertEqual(result, XCTWaiterResultTimedOut);
+}
+
+- (void)testStopCallbackWithSeverityLogger {
+  XCTestExpectation *callbackExpectation = [self expectationWithDescription:@"stopped"];
+
+  [self.logger
+      startWithMessageAndSeverityHandler:^(NSString *message, RTCLoggingSeverity loggingServerity) {
+        [callbackExpectation fulfill];
+      }];
+
+  [self.logger stop];
+
+  RTCLogInfo("Just some info");
+
+  XCTWaiter *waiter = [[XCTWaiter alloc] init];
+  XCTWaiterResult result = [waiter waitForExpectations:@[ callbackExpectation ] timeout:1.0];
+  XCTAssertEqual(result, XCTWaiterResultTimedOut);
+}
+
+- (void)testDestroyingCallbackLogger {
+  XCTestExpectation *callbackExpectation = [self expectationWithDescription:@"destroyed"];
+
+  [self.logger start:^(NSString *message) {
+    [callbackExpectation fulfill];
+  }];
+
+  self.logger = nil;
+
+  RTCLogInfo("Just some info");
+
+  XCTWaiter *waiter = [[XCTWaiter alloc] init];
+  XCTWaiterResult result = [waiter waitForExpectations:@[ callbackExpectation ] timeout:1.0];
+  XCTAssertEqual(result, XCTWaiterResultTimedOut);
+}
+
+- (void)testDestroyingCallbackWithSeverityLogger {
+  XCTestExpectation *callbackExpectation = [self expectationWithDescription:@"destroyed"];
+
+  [self.logger
+      startWithMessageAndSeverityHandler:^(NSString *message, RTCLoggingSeverity loggingServerity) {
+        [callbackExpectation fulfill];
+      }];
+
+  self.logger = nil;
+
+  RTCLogInfo("Just some info");
+
+  XCTWaiter *waiter = [[XCTWaiter alloc] init];
+  XCTWaiterResult result = [waiter waitForExpectations:@[ callbackExpectation ] timeout:1.0];
+  XCTAssertEqual(result, XCTWaiterResultTimedOut);
+}
+
+- (void)testCallbackWithSeverityLoggerCannotStartTwice {
+  self.logger.severity = RTCLoggingSeverityWarning;
+
+  XCTestExpectation *callbackExpectation = [self expectationWithDescription:@"callbackWarning"];
+
+  [self.logger
+      startWithMessageAndSeverityHandler:^(NSString *message, RTCLoggingSeverity loggingServerity) {
+        XCTAssertTrue([message hasSuffix:@"Horrible error\n"]);
+        XCTAssertEqual(loggingServerity, RTCLoggingSeverityError);
+        [callbackExpectation fulfill];
+      }];
+
+  [self.logger start:^(NSString *message) {
+    [callbackExpectation fulfill];
+    XCTAssertTrue(false);
+  }];
+
+  RTCLogError("Horrible error");
+
+  [self waitForExpectations:@[ callbackExpectation ] timeout:10.0];
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/unittests/RTCCameraVideoCapturerTests.mm b/third_party/libwebrtc/sdk/objc/unittests/RTCCameraVideoCapturerTests.mm
new file mode 100644
index 0000000000..7919878260
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/unittests/RTCCameraVideoCapturerTests.mm
@@ -0,0 +1,570 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <OCMock/OCMock.h>
+#import <XCTest/XCTest.h>
+
+#if TARGET_OS_IPHONE
+#import <UIKit/UIKit.h>
+#endif
+
+#import "base/RTCVideoFrame.h"
+#import "components/capturer/RTCCameraVideoCapturer.h"
+#import "helpers/AVCaptureSession+DevicePosition.h"
+#import "helpers/RTCDispatcher.h"
+#import "helpers/scoped_cftyperef.h"
+
+#define WAIT(timeoutMs)                                                        \
+  do {                                                                         \
+    id expectation = [[XCTestExpectation alloc] initWithDescription:@"Dummy"]; \
+    XCTWaiterResult res = [XCTWaiter waitForExpectations:@[ expectation ]      \
+                                                 timeout:timeoutMs / 1000.0];  \
+    XCTAssertEqual(XCTWaiterResultTimedOut, res);                              \
+  } while (false);
+
+#if TARGET_OS_IPHONE
+// Helper method.
+CMSampleBufferRef createTestSampleBufferRef() {
+
+  // This image is already in the testing bundle.
+  UIImage *image = [UIImage imageNamed:@"Default.png"];
+  CGSize size = image.size;
+  CGImageRef imageRef = [image CGImage];
+
+  CVPixelBufferRef pixelBuffer = nullptr;
+  CVPixelBufferCreate(kCFAllocatorDefault, size.width, size.height, kCVPixelFormatType_32ARGB, nil,
+                      &pixelBuffer);
+
+  CGColorSpaceRef rgbColorSpace = CGColorSpaceCreateDeviceRGB();
+  // We don't care about bitsPerComponent and bytesPerRow so arbitrary value of 8 for both.
+  CGContextRef context = CGBitmapContextCreate(nil, size.width, size.height, 8, 8 * size.width,
+                                               rgbColorSpace, kCGImageAlphaPremultipliedFirst);
+
+  CGContextDrawImage(
+      context, CGRectMake(0, 0, CGImageGetWidth(imageRef), CGImageGetHeight(imageRef)), imageRef);
+
+  CGColorSpaceRelease(rgbColorSpace);
+  CGContextRelease(context);
+
+  // We don't really care about the timing.
+  CMSampleTimingInfo timing = {kCMTimeInvalid, kCMTimeInvalid, kCMTimeInvalid};
+  CMVideoFormatDescriptionRef description = nullptr;
+  CMVideoFormatDescriptionCreateForImageBuffer(NULL, pixelBuffer, &description);
+
+  CMSampleBufferRef sampleBuffer = nullptr;
+  CMSampleBufferCreateForImageBuffer(kCFAllocatorDefault, pixelBuffer, YES, NULL, NULL, description,
+                                     &timing, &sampleBuffer);
+  CFRelease(pixelBuffer);
+
+  return sampleBuffer;
+
+}
+#endif
+@interface RTC_OBJC_TYPE (RTCCameraVideoCapturer)
+(Tests)<AVCaptureVideoDataOutputSampleBufferDelegate> -
+    (instancetype)initWithDelegate
+    : (__weak id<RTC_OBJC_TYPE(RTCVideoCapturerDelegate)>)delegate captureSession
+    : (AVCaptureSession *)captureSession;
+@end
+
+@interface RTCCameraVideoCapturerTests : XCTestCase
+@property(nonatomic, strong) id delegateMock;
+@property(nonatomic, strong) id deviceMock;
+@property(nonatomic, strong) id captureConnectionMock;
+@property(nonatomic, strong) RTC_OBJC_TYPE(RTCCameraVideoCapturer) * capturer;
+@end
+
+@implementation RTCCameraVideoCapturerTests
+@synthesize delegateMock = _delegateMock;
+@synthesize deviceMock = _deviceMock;
+@synthesize captureConnectionMock = _captureConnectionMock;
+@synthesize capturer = _capturer;
+
+- (void)setUp {
+  self.delegateMock = OCMProtocolMock(@protocol(RTC_OBJC_TYPE(RTCVideoCapturerDelegate)));
+  self.captureConnectionMock = OCMClassMock([AVCaptureConnection class]);
+  self.capturer =
+      [[RTC_OBJC_TYPE(RTCCameraVideoCapturer) alloc] initWithDelegate:self.delegateMock];
+  self.deviceMock = [RTCCameraVideoCapturerTests createDeviceMock];
+}
+
+- (void)tearDown {
+  [self.delegateMock stopMocking];
+  [self.deviceMock stopMocking];
+  self.delegateMock = nil;
+  self.deviceMock = nil;
+  self.capturer = nil;
+}
+
+#pragma mark - utils
+
++ (id)createDeviceMock {
+  return OCMClassMock([AVCaptureDevice class]);
+}
+
+#pragma mark - test cases
+
+- (void)testSetupSession {
+  AVCaptureSession *session = self.capturer.captureSession;
+  XCTAssertTrue(session != nil);
+
+#if TARGET_OS_IPHONE
+  XCTAssertEqual(session.sessionPreset, AVCaptureSessionPresetInputPriority);
+  XCTAssertEqual(session.usesApplicationAudioSession, NO);
+#endif
+  XCTAssertEqual(session.outputs.count, 1u);
+}
+
+- (void)testSetupSessionOutput {
+  AVCaptureVideoDataOutput *videoOutput = self.capturer.captureSession.outputs[0];
+  XCTAssertEqual(videoOutput.alwaysDiscardsLateVideoFrames, NO);
+  XCTAssertEqual(videoOutput.sampleBufferDelegate, self.capturer);
+}
+
+- (void)testSupportedFormatsForDevice {
+  // given
+  id validFormat1 = OCMClassMock([AVCaptureDeviceFormat class]);
+  CMVideoFormatDescriptionRef format;
+
+  // We don't care about width and heigth so arbitrary 123 and 456 values.
+  int width = 123;
+  int height = 456;
+  CMVideoFormatDescriptionCreate(nil, kCVPixelFormatType_420YpCbCr8PlanarFullRange, width, height,
+                                 nil, &format);
+  OCMStub([validFormat1 formatDescription]).andReturn(format);
+
+  id validFormat2 = OCMClassMock([AVCaptureDeviceFormat class]);
+  CMVideoFormatDescriptionCreate(nil, kCVPixelFormatType_420YpCbCr8BiPlanarVideoRange, width,
+                                 height, nil, &format);
+  OCMStub([validFormat2 formatDescription]).andReturn(format);
+
+  id invalidFormat = OCMClassMock([AVCaptureDeviceFormat class]);
+  CMVideoFormatDescriptionCreate(nil, kCVPixelFormatType_422YpCbCr8_yuvs, width, height, nil,
+                                 &format);
+  OCMStub([invalidFormat formatDescription]).andReturn(format);
+
+  NSArray *formats = @[ validFormat1, validFormat2, invalidFormat ];
+  OCMStub([self.deviceMock formats]).andReturn(formats);
+
+  // when
+  NSArray *supportedFormats =
+      [RTC_OBJC_TYPE(RTCCameraVideoCapturer) supportedFormatsForDevice:self.deviceMock];
+
+  // then
+  XCTAssertEqual(supportedFormats.count, 3u);
+  XCTAssertTrue([supportedFormats containsObject:validFormat1]);
+  XCTAssertTrue([supportedFormats containsObject:validFormat2]);
+  XCTAssertTrue([supportedFormats containsObject:invalidFormat]);
+
+  // cleanup
+  [validFormat1 stopMocking];
+  [validFormat2 stopMocking];
+  [invalidFormat stopMocking];
+  validFormat1 = nil;
+  validFormat2 = nil;
+  invalidFormat = nil;
+}
+
+- (void)testDelegateCallbackNotCalledWhenInvalidBuffer {
+  // given
+  CMSampleBufferRef sampleBuffer = nullptr;
+  [[self.delegateMock reject] capturer:[OCMArg any] didCaptureVideoFrame:[OCMArg any]];
+
+  // when
+  [self.capturer captureOutput:self.capturer.captureSession.outputs[0]
+         didOutputSampleBuffer:sampleBuffer
+                fromConnection:self.captureConnectionMock];
+
+  // then
+  [self.delegateMock verify];
+}
+
+#if 0
+// See crbug.com/1404878 - XCTExpectFailure and XCTSkip are considered failures
+
+- (void)testDelegateCallbackWithValidBufferAndOrientationUpdate {
+#if TARGET_OS_IPHONE
+  XCTExpectFailure(@"Setting orientation on UIDevice is not supported");
+  [UIDevice.currentDevice setValue:@(UIDeviceOrientationPortraitUpsideDown) forKey:@"orientation"];
+  CMSampleBufferRef sampleBuffer = createTestSampleBufferRef();
+
+  // then
+  [[self.delegateMock expect] capturer:self.capturer
+                  didCaptureVideoFrame:[OCMArg checkWithBlock:^BOOL(RTC_OBJC_TYPE(RTCVideoFrame) *
+                                                                    expectedFrame) {
+                    XCTAssertEqual(expectedFrame.rotation, RTCVideoRotation_270);
+                    return YES;
+                  }]];
+
+  // when
+  NSNotificationCenter *center = [NSNotificationCenter defaultCenter];
+  [center postNotificationName:UIDeviceOrientationDidChangeNotification object:nil];
+
+  // We need to wait for the dispatch to finish.
+  WAIT(1000);
+
+  [self.capturer captureOutput:self.capturer.captureSession.outputs[0]
+         didOutputSampleBuffer:sampleBuffer
+                fromConnection:self.captureConnectionMock];
+
+  [self.delegateMock verify];
+  CFRelease(sampleBuffer);
+#endif
+}
+
+// The XCTest framework considers functions that don't take arguments tests. This is a helper.
+- (void)testRotationCamera:(AVCaptureDevicePosition)camera
+           withOrientation:(UIDeviceOrientation)deviceOrientation {
+#if TARGET_OS_IPHONE
+  // Mock the AVCaptureConnection as we will get the camera position from the connection's
+  // input ports.
+  AVCaptureDeviceInput *inputPortMock = OCMClassMock([AVCaptureDeviceInput class]);
+  AVCaptureInputPort *captureInputPort = OCMClassMock([AVCaptureInputPort class]);
+  NSArray *inputPortsArrayMock = @[captureInputPort];
+  AVCaptureDevice *captureDeviceMock = OCMClassMock([AVCaptureDevice class]);
+  OCMStub(((AVCaptureConnection *)self.captureConnectionMock).inputPorts).
+      andReturn(inputPortsArrayMock);
+  OCMStub(captureInputPort.input).andReturn(inputPortMock);
+  OCMStub(inputPortMock.device).andReturn(captureDeviceMock);
+  OCMStub(captureDeviceMock.position).andReturn(camera);
+
+  XCTExpectFailure(@"Setting orientation on UIDevice is not supported");
+  [UIDevice.currentDevice setValue:@(deviceOrientation) forKey:@"orientation"];
+
+  CMSampleBufferRef sampleBuffer = createTestSampleBufferRef();
+
+  [[self.delegateMock expect] capturer:self.capturer
+                  didCaptureVideoFrame:[OCMArg checkWithBlock:^BOOL(RTC_OBJC_TYPE(RTCVideoFrame) *
+                                                                    expectedFrame) {
+                    if (camera == AVCaptureDevicePositionFront) {
+                      if (deviceOrientation == UIDeviceOrientationLandscapeLeft) {
+                        XCTAssertEqual(expectedFrame.rotation, RTCVideoRotation_180);
+                      } else if (deviceOrientation == UIDeviceOrientationLandscapeRight) {
+                        XCTAssertEqual(expectedFrame.rotation, RTCVideoRotation_0);
+                      }
+                    } else if (camera == AVCaptureDevicePositionBack) {
+                      if (deviceOrientation == UIDeviceOrientationLandscapeLeft) {
+                        XCTAssertEqual(expectedFrame.rotation, RTCVideoRotation_0);
+                      } else if (deviceOrientation == UIDeviceOrientationLandscapeRight) {
+                        XCTAssertEqual(expectedFrame.rotation, RTCVideoRotation_180);
+                      }
+                    }
+                    return YES;
+                  }]];
+
+  NSNotificationCenter *center = [NSNotificationCenter defaultCenter];
+  [center postNotificationName:UIDeviceOrientationDidChangeNotification object:nil];
+
+  // We need to wait for the dispatch to finish.
+  WAIT(1000);
+
+  [self.capturer captureOutput:self.capturer.captureSession.outputs[0]
+         didOutputSampleBuffer:sampleBuffer
+                fromConnection:self.captureConnectionMock];
+
+  [self.delegateMock verify];
+
+  CFRelease(sampleBuffer);
+#endif
+}
+
+- (void)testRotationCameraBackLandscapeLeft {
+  [self testRotationCamera:AVCaptureDevicePositionBack
+           withOrientation:UIDeviceOrientationLandscapeLeft];
+}
+
+- (void)testRotationCameraFrontLandscapeLeft {
+  [self testRotationCamera:AVCaptureDevicePositionFront
+           withOrientation:UIDeviceOrientationLandscapeLeft];
+}
+
+- (void)testRotationCameraBackLandscapeRight {
+  [self testRotationCamera:AVCaptureDevicePositionBack
+           withOrientation:UIDeviceOrientationLandscapeRight];
+}
+
+- (void)testRotationCameraFrontLandscapeRight {
+  [self testRotationCamera:AVCaptureDevicePositionFront
+           withOrientation:UIDeviceOrientationLandscapeRight];
+}
+
+#endif
+
+- (void)setExif:(CMSampleBufferRef)sampleBuffer {
+  rtc::ScopedCFTypeRef<CFMutableDictionaryRef> exif(CFDictionaryCreateMutable(
+      kCFAllocatorDefault, 0, &kCFTypeDictionaryKeyCallBacks, &kCFTypeDictionaryValueCallBacks));
+  CFDictionarySetValue(exif.get(), CFSTR("LensModel"), CFSTR("iPhone SE back camera 4.15mm f/2.2"));
+  CMSetAttachment(sampleBuffer, CFSTR("{Exif}"), exif.get(), kCMAttachmentMode_ShouldPropagate);
+}
+
+#if 0
+// See crbug.com/1404878 - XCTExpectFailure and XCTSkip are considered failures
+
+- (void)testRotationFrame {
+#if TARGET_OS_IPHONE
+  // Mock the AVCaptureConnection as we will get the camera position from the connection's
+  // input ports.
+  AVCaptureDeviceInput *inputPortMock = OCMClassMock([AVCaptureDeviceInput class]);
+  AVCaptureInputPort *captureInputPort = OCMClassMock([AVCaptureInputPort class]);
+  NSArray *inputPortsArrayMock = @[captureInputPort];
+  AVCaptureDevice *captureDeviceMock = OCMClassMock([AVCaptureDevice class]);
+  OCMStub(((AVCaptureConnection *)self.captureConnectionMock).inputPorts).
+      andReturn(inputPortsArrayMock);
+  OCMStub(captureInputPort.input).andReturn(inputPortMock);
+  OCMStub(inputPortMock.device).andReturn(captureDeviceMock);
+  OCMStub(captureDeviceMock.position).andReturn(AVCaptureDevicePositionFront);
+
+  XCTExpectFailure(@"Setting orientation on UIDevice is not supported");
+  [UIDevice.currentDevice setValue:@(UIDeviceOrientationLandscapeLeft) forKey:@"orientation"];
+
+  CMSampleBufferRef sampleBuffer = createTestSampleBufferRef();
+
+  [[self.delegateMock expect] capturer:self.capturer
+                  didCaptureVideoFrame:[OCMArg checkWithBlock:^BOOL(RTC_OBJC_TYPE(RTCVideoFrame) *
+                                                                    expectedFrame) {
+                    // Front camera and landscape left should return 180. But the frame's exif
+                    // we add below says its from the back camera, so rotation should be 0.
+                    XCTAssertEqual(expectedFrame.rotation, RTCVideoRotation_0);
+                    return YES;
+                  }]];
+
+  NSNotificationCenter *center = [NSNotificationCenter defaultCenter];
+  [center postNotificationName:UIDeviceOrientationDidChangeNotification object:nil];
+
+  // We need to wait for the dispatch to finish.
+  WAIT(1000);
+
+  [self setExif:sampleBuffer];
+
+  [self.capturer captureOutput:self.capturer.captureSession.outputs[0]
+         didOutputSampleBuffer:sampleBuffer
+                fromConnection:self.captureConnectionMock];
+
+  [self.delegateMock verify];
+  CFRelease(sampleBuffer);
+#endif
+}
+
+#endif
+
+- (void)testImageExif {
+#if TARGET_OS_IPHONE
+  CMSampleBufferRef sampleBuffer = createTestSampleBufferRef();
+  [self setExif:sampleBuffer];
+
+  AVCaptureDevicePosition cameraPosition = [AVCaptureSession
+                                            devicePositionForSampleBuffer:sampleBuffer];
+  XCTAssertEqual(cameraPosition, AVCaptureDevicePositionBack);
+#endif
+}
+
+@end
+
+@interface RTCCameraVideoCapturerTestsWithMockedCaptureSession : XCTestCase
+@property(nonatomic, strong) id delegateMock;
+@property(nonatomic, strong) id deviceMock;
+@property(nonatomic, strong) id captureSessionMock;
+@property(nonatomic, strong) RTC_OBJC_TYPE(RTCCameraVideoCapturer) * capturer;
+@end
+
+@implementation RTCCameraVideoCapturerTestsWithMockedCaptureSession
+@synthesize delegateMock = _delegateMock;
+@synthesize deviceMock = _deviceMock;
+@synthesize captureSessionMock = _captureSessionMock;
+@synthesize capturer = _capturer;
+
+- (void)setUp {
+  self.captureSessionMock = OCMStrictClassMock([AVCaptureSession class]);
+  OCMStub([self.captureSessionMock setSessionPreset:[OCMArg any]]);
+  OCMStub([self.captureSessionMock setUsesApplicationAudioSession:NO]);
+  OCMStub([self.captureSessionMock canAddOutput:[OCMArg any]]).andReturn(YES);
+  OCMStub([self.captureSessionMock addOutput:[OCMArg any]]);
+  OCMStub([self.captureSessionMock beginConfiguration]);
+  OCMStub([self.captureSessionMock commitConfiguration]);
+  self.delegateMock = OCMProtocolMock(@protocol(RTC_OBJC_TYPE(RTCVideoCapturerDelegate)));
+  self.capturer =
+      [[RTC_OBJC_TYPE(RTCCameraVideoCapturer) alloc] initWithDelegate:self.delegateMock
+                                                       captureSession:self.captureSessionMock];
+  self.deviceMock = [RTCCameraVideoCapturerTests createDeviceMock];
+}
+
+- (void)tearDown {
+  [self.delegateMock stopMocking];
+  [self.deviceMock stopMocking];
+  self.delegateMock = nil;
+  self.deviceMock = nil;
+  self.capturer = nil;
+  self.captureSessionMock = nil;
+}
+
+#pragma mark - test cases
+
+- (void)testStartingAndStoppingCapture {
+  id expectedDeviceInputMock = OCMClassMock([AVCaptureDeviceInput class]);
+  id captureDeviceInputMock = OCMClassMock([AVCaptureDeviceInput class]);
+  OCMStub([captureDeviceInputMock deviceInputWithDevice:self.deviceMock error:[OCMArg setTo:nil]])
+      .andReturn(expectedDeviceInputMock);
+
+  OCMStub([self.deviceMock lockForConfiguration:[OCMArg setTo:nil]]).andReturn(YES);
+  OCMStub([self.deviceMock unlockForConfiguration]);
+  OCMStub([_captureSessionMock canAddInput:expectedDeviceInputMock]).andReturn(YES);
+  OCMStub([_captureSessionMock inputs]).andReturn(@[ expectedDeviceInputMock ]);
+  OCMStub([_captureSessionMock removeInput:expectedDeviceInputMock]);
+
+  // Set expectation that the capture session should be started with correct device.
+  OCMExpect([_captureSessionMock addInput:expectedDeviceInputMock]);
+  OCMExpect([_captureSessionMock startRunning]);
+  OCMExpect([_captureSessionMock stopRunning]);
+
+  id format = OCMClassMock([AVCaptureDeviceFormat class]);
+  [self.capturer startCaptureWithDevice:self.deviceMock format:format fps:30];
+  [self.capturer stopCapture];
+
+  // Start capture code is dispatched async.
+  OCMVerifyAllWithDelay(_captureSessionMock, 15);
+}
+
+- (void)testStartCaptureFailingToLockForConfiguration {
+  // The captureSessionMock is a strict mock, so this test will crash if the startCapture
+  // method does not return when failing to lock for configuration.
+  OCMExpect([self.deviceMock lockForConfiguration:[OCMArg setTo:nil]]).andReturn(NO);
+
+  id format = OCMClassMock([AVCaptureDeviceFormat class]);
+  [self.capturer startCaptureWithDevice:self.deviceMock format:format fps:30];
+
+  // Start capture code is dispatched async.
+  OCMVerifyAllWithDelay(self.deviceMock, 15);
+}
+
+- (void)testStartingAndStoppingCaptureWithCallbacks {
+  id expectedDeviceInputMock = OCMClassMock([AVCaptureDeviceInput class]);
+  id captureDeviceInputMock = OCMClassMock([AVCaptureDeviceInput class]);
+  OCMStub([captureDeviceInputMock deviceInputWithDevice:self.deviceMock error:[OCMArg setTo:nil]])
+      .andReturn(expectedDeviceInputMock);
+
+  OCMStub([self.deviceMock lockForConfiguration:[OCMArg setTo:nil]]).andReturn(YES);
+  OCMStub([self.deviceMock unlockForConfiguration]);
+  OCMStub([_captureSessionMock canAddInput:expectedDeviceInputMock]).andReturn(YES);
+  OCMStub([_captureSessionMock inputs]).andReturn(@[ expectedDeviceInputMock ]);
+  OCMStub([_captureSessionMock removeInput:expectedDeviceInputMock]);
+
+  // Set expectation that the capture session should be started with correct device.
+  OCMExpect([_captureSessionMock addInput:expectedDeviceInputMock]);
+  OCMExpect([_captureSessionMock startRunning]);
+  OCMExpect([_captureSessionMock stopRunning]);
+
+  dispatch_semaphore_t completedStopSemaphore = dispatch_semaphore_create(0);
+
+  __block BOOL completedStart = NO;
+  id format = OCMClassMock([AVCaptureDeviceFormat class]);
+  [self.capturer startCaptureWithDevice:self.deviceMock
+                                 format:format
+                                    fps:30
+                      completionHandler:^(NSError *error) {
+                        XCTAssertEqual(error, nil);
+                        completedStart = YES;
+                      }];
+
+  __block BOOL completedStop = NO;
+  [self.capturer stopCaptureWithCompletionHandler:^{
+    completedStop = YES;
+    dispatch_semaphore_signal(completedStopSemaphore);
+  }];
+
+  dispatch_semaphore_wait(completedStopSemaphore,
+                          dispatch_time(DISPATCH_TIME_NOW, 15.0 * NSEC_PER_SEC));
+  OCMVerifyAllWithDelay(_captureSessionMock, 15);
+  XCTAssertTrue(completedStart);
+  XCTAssertTrue(completedStop);
+}
+
+- (void)testStartCaptureFailingToLockForConfigurationWithCallback {
+  id expectedDeviceInputMock = OCMClassMock([AVCaptureDeviceInput class]);
+  id captureDeviceInputMock = OCMClassMock([AVCaptureDeviceInput class]);
+  OCMStub([captureDeviceInputMock deviceInputWithDevice:self.deviceMock error:[OCMArg setTo:nil]])
+      .andReturn(expectedDeviceInputMock);
+
+  id errorMock = OCMClassMock([NSError class]);
+
+  OCMStub([self.deviceMock lockForConfiguration:[OCMArg setTo:errorMock]]).andReturn(NO);
+  OCMStub([_captureSessionMock canAddInput:expectedDeviceInputMock]).andReturn(YES);
+  OCMStub([self.deviceMock unlockForConfiguration]);
+
+  OCMExpect([_captureSessionMock addInput:expectedDeviceInputMock]);
+
+  dispatch_semaphore_t completedStartSemaphore = dispatch_semaphore_create(0);
+  __block NSError *callbackError = nil;
+
+  id format = OCMClassMock([AVCaptureDeviceFormat class]);
+  [self.capturer startCaptureWithDevice:self.deviceMock
+                                 format:format
+                                    fps:30
+                      completionHandler:^(NSError *error) {
+                        callbackError = error;
+                        dispatch_semaphore_signal(completedStartSemaphore);
+                      }];
+
+  long ret = dispatch_semaphore_wait(completedStartSemaphore,
+                                     dispatch_time(DISPATCH_TIME_NOW, 15.0 * NSEC_PER_SEC));
+  XCTAssertEqual(ret, 0);
+  XCTAssertEqual(callbackError, errorMock);
+}
+
+// TODO(crbug.com/webrtc/14829): Test is disabled on iOS < 16 and broken on iOS 16.
+- (void)DISABLED_testStartCaptureSetsOutputDimensionsInvalidPixelFormat {
+  id expectedDeviceInputMock = OCMClassMock([AVCaptureDeviceInput class]);
+  id captureDeviceInputMock = OCMClassMock([AVCaptureDeviceInput class]);
+  OCMStub([captureDeviceInputMock deviceInputWithDevice:_deviceMock error:[OCMArg setTo:nil]])
+      .andReturn(expectedDeviceInputMock);
+
+  OCMStub([_deviceMock lockForConfiguration:[OCMArg setTo:nil]]).andReturn(YES);
+  OCMStub([_deviceMock unlockForConfiguration]);
+  OCMStub([_captureSessionMock canAddInput:expectedDeviceInputMock]).andReturn(YES);
+  OCMStub([_captureSessionMock addInput:expectedDeviceInputMock]);
+  OCMStub([_captureSessionMock inputs]).andReturn(@[ expectedDeviceInputMock ]);
+  OCMStub([_captureSessionMock removeInput:expectedDeviceInputMock]);
+  OCMStub([_captureSessionMock startRunning]);
+  OCMStub([_captureSessionMock stopRunning]);
+
+  id deviceFormatMock = OCMClassMock([AVCaptureDeviceFormat class]);
+  CMVideoFormatDescriptionRef formatDescription;
+
+  int width = 110;
+  int height = 220;
+  FourCharCode pixelFormat = 0x18000000;
+  CMVideoFormatDescriptionCreate(nil, pixelFormat, width, height, nil, &formatDescription);
+  OCMStub([deviceFormatMock formatDescription]).andReturn(formatDescription);
+
+  [_capturer startCaptureWithDevice:_deviceMock format:deviceFormatMock fps:30];
+
+  XCTestExpectation *expectation = [self expectationWithDescription:@"StopCompletion"];
+  [_capturer stopCaptureWithCompletionHandler:^(void) {
+    [expectation fulfill];
+  }];
+
+  [self waitForExpectationsWithTimeout:15 handler:nil];
+
+  OCMVerify([_captureSessionMock
+      addOutput:[OCMArg checkWithBlock:^BOOL(AVCaptureVideoDataOutput *output) {
+        if (@available(iOS 16, *)) {
+          XCTAssertEqual(width, [output.videoSettings[(id)kCVPixelBufferWidthKey] intValue]);
+          XCTAssertEqual(height, [output.videoSettings[(id)kCVPixelBufferHeightKey] intValue]);
+        } else {
+          XCTAssertEqual(0, [output.videoSettings[(id)kCVPixelBufferWidthKey] intValue]);
+          XCTAssertEqual(0, [output.videoSettings[(id)kCVPixelBufferHeightKey] intValue]);
+        }
+        XCTAssertEqual(
+            (FourCharCode)kCVPixelFormatType_420YpCbCr8BiPlanarVideoRange,
+            [output.videoSettings[(id)kCVPixelBufferPixelFormatTypeKey] unsignedIntValue]);
+        return YES;
+      }]]);
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/unittests/RTCCertificateTest.mm b/third_party/libwebrtc/sdk/objc/unittests/RTCCertificateTest.mm
new file mode 100644
index 0000000000..bc1347336c
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/unittests/RTCCertificateTest.mm
@@ -0,0 +1,73 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+#import <XCTest/XCTest.h>
+
+#include <vector>
+
+#include "rtc_base/gunit.h"
+
+#import "api/peerconnection/RTCConfiguration+Private.h"
+#import "api/peerconnection/RTCConfiguration.h"
+#import "api/peerconnection/RTCIceServer.h"
+#import "api/peerconnection/RTCMediaConstraints.h"
+#import "api/peerconnection/RTCPeerConnection.h"
+#import "api/peerconnection/RTCPeerConnectionFactory.h"
+#import "helpers/NSString+StdString.h"
+
+@interface RTCCertificateTest : XCTestCase
+@end
+
+@implementation RTCCertificateTest
+
+- (void)testCertificateIsUsedInConfig {
+  RTC_OBJC_TYPE(RTCConfiguration) *originalConfig = [[RTC_OBJC_TYPE(RTCConfiguration) alloc] init];
+
+  NSArray *urlStrings = @[ @"stun:stun1.example.net" ];
+  RTC_OBJC_TYPE(RTCIceServer) *server =
+      [[RTC_OBJC_TYPE(RTCIceServer) alloc] initWithURLStrings:urlStrings];
+  originalConfig.iceServers = @[ server ];
+
+  // Generate a new certificate.
+  RTC_OBJC_TYPE(RTCCertificate) *originalCertificate = [RTC_OBJC_TYPE(RTCCertificate)
+      generateCertificateWithParams:@{@"expires" : @100000, @"name" : @"RSASSA-PKCS1-v1_5"}];
+
+  // Store certificate in configuration.
+  originalConfig.certificate = originalCertificate;
+
+  RTC_OBJC_TYPE(RTCMediaConstraints) *contraints =
+      [[RTC_OBJC_TYPE(RTCMediaConstraints) alloc] initWithMandatoryConstraints:@{}
+                                                           optionalConstraints:nil];
+  RTC_OBJC_TYPE(RTCPeerConnectionFactory) *factory =
+      [[RTC_OBJC_TYPE(RTCPeerConnectionFactory) alloc] init];
+
+  // Create PeerConnection with this certificate.
+  RTC_OBJC_TYPE(RTCPeerConnection) *peerConnection =
+      [factory peerConnectionWithConfiguration:originalConfig constraints:contraints delegate:nil];
+
+  // Retrieve certificate from the configuration.
+  RTC_OBJC_TYPE(RTCConfiguration) *retrievedConfig = peerConnection.configuration;
+
+  // Extract PEM strings from original certificate.
+  std::string originalPrivateKeyField = [[originalCertificate private_key] UTF8String];
+  std::string originalCertificateField = [[originalCertificate certificate] UTF8String];
+
+  // Extract PEM strings from certificate retrieved from configuration.
+  RTC_OBJC_TYPE(RTCCertificate) *retrievedCertificate = retrievedConfig.certificate;
+  std::string retrievedPrivateKeyField = [[retrievedCertificate private_key] UTF8String];
+  std::string retrievedCertificateField = [[retrievedCertificate certificate] UTF8String];
+
+  // Check that the original certificate and retrieved certificate match.
+  EXPECT_EQ(originalPrivateKeyField, retrievedPrivateKeyField);
+  EXPECT_EQ(retrievedCertificateField, retrievedCertificateField);
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/unittests/RTCConfigurationTest.mm b/third_party/libwebrtc/sdk/objc/unittests/RTCConfigurationTest.mm
new file mode 100644
index 0000000000..18cc97191e
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/unittests/RTCConfigurationTest.mm
@@ -0,0 +1,162 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+#import <XCTest/XCTest.h>
+
+#include <vector>
+
+#include "rtc_base/gunit.h"
+
+#import "api/peerconnection/RTCConfiguration+Private.h"
+#import "api/peerconnection/RTCConfiguration.h"
+#import "api/peerconnection/RTCIceServer.h"
+#import "helpers/NSString+StdString.h"
+
+@interface RTCConfigurationTest : XCTestCase
+@end
+
+@implementation RTCConfigurationTest
+
+- (void)testConversionToNativeConfiguration {
+  NSArray *urlStrings = @[ @"stun:stun1.example.net" ];
+  RTC_OBJC_TYPE(RTCIceServer) *server =
+      [[RTC_OBJC_TYPE(RTCIceServer) alloc] initWithURLStrings:urlStrings];
+
+  RTC_OBJC_TYPE(RTCConfiguration) *config = [[RTC_OBJC_TYPE(RTCConfiguration) alloc] init];
+  config.iceServers = @[ server ];
+  config.iceTransportPolicy = RTCIceTransportPolicyRelay;
+  config.bundlePolicy = RTCBundlePolicyMaxBundle;
+  config.rtcpMuxPolicy = RTCRtcpMuxPolicyNegotiate;
+  config.tcpCandidatePolicy = RTCTcpCandidatePolicyDisabled;
+  config.candidateNetworkPolicy = RTCCandidateNetworkPolicyLowCost;
+  const int maxPackets = 60;
+  const int timeout = 1;
+  const int interval = 2;
+  config.audioJitterBufferMaxPackets = maxPackets;
+  config.audioJitterBufferFastAccelerate = YES;
+  config.iceConnectionReceivingTimeout = timeout;
+  config.iceBackupCandidatePairPingInterval = interval;
+  config.continualGatheringPolicy =
+      RTCContinualGatheringPolicyGatherContinually;
+  config.shouldPruneTurnPorts = YES;
+  config.cryptoOptions =
+      [[RTC_OBJC_TYPE(RTCCryptoOptions) alloc] initWithSrtpEnableGcmCryptoSuites:YES
+                                             srtpEnableAes128Sha1_32CryptoCipher:YES
+                                          srtpEnableEncryptedRtpHeaderExtensions:YES
+                                                    sframeRequireFrameEncryption:YES];
+  config.rtcpAudioReportIntervalMs = 2500;
+  config.rtcpVideoReportIntervalMs = 3750;
+
+  std::unique_ptr<webrtc::PeerConnectionInterface::RTCConfiguration>
+      nativeConfig([config createNativeConfiguration]);
+  EXPECT_TRUE(nativeConfig.get());
+  EXPECT_EQ(1u, nativeConfig->servers.size());
+  webrtc::PeerConnectionInterface::IceServer nativeServer =
+      nativeConfig->servers.front();
+  EXPECT_EQ(1u, nativeServer.urls.size());
+  EXPECT_EQ("stun:stun1.example.net", nativeServer.urls.front());
+
+  EXPECT_EQ(webrtc::PeerConnectionInterface::kRelay, nativeConfig->type);
+  EXPECT_EQ(webrtc::PeerConnectionInterface::kBundlePolicyMaxBundle,
+            nativeConfig->bundle_policy);
+  EXPECT_EQ(webrtc::PeerConnectionInterface::kRtcpMuxPolicyNegotiate,
+            nativeConfig->rtcp_mux_policy);
+  EXPECT_EQ(webrtc::PeerConnectionInterface::kTcpCandidatePolicyDisabled,
+            nativeConfig->tcp_candidate_policy);
+  EXPECT_EQ(webrtc::PeerConnectionInterface::kCandidateNetworkPolicyLowCost,
+            nativeConfig->candidate_network_policy);
+  EXPECT_EQ(maxPackets, nativeConfig->audio_jitter_buffer_max_packets);
+  EXPECT_EQ(true, nativeConfig->audio_jitter_buffer_fast_accelerate);
+  EXPECT_EQ(timeout, nativeConfig->ice_connection_receiving_timeout);
+  EXPECT_EQ(interval, nativeConfig->ice_backup_candidate_pair_ping_interval);
+  EXPECT_EQ(webrtc::PeerConnectionInterface::GATHER_CONTINUALLY,
+            nativeConfig->continual_gathering_policy);
+  EXPECT_EQ(true, nativeConfig->prune_turn_ports);
+  EXPECT_EQ(true, nativeConfig->crypto_options->srtp.enable_gcm_crypto_suites);
+  EXPECT_EQ(true, nativeConfig->crypto_options->srtp.enable_aes128_sha1_32_crypto_cipher);
+  EXPECT_EQ(true, nativeConfig->crypto_options->srtp.enable_encrypted_rtp_header_extensions);
+  EXPECT_EQ(true, nativeConfig->crypto_options->sframe.require_frame_encryption);
+  EXPECT_EQ(2500, nativeConfig->audio_rtcp_report_interval_ms());
+  EXPECT_EQ(3750, nativeConfig->video_rtcp_report_interval_ms());
+}
+
+- (void)testNativeConversionToConfiguration {
+  NSArray *urlStrings = @[ @"stun:stun1.example.net" ];
+  RTC_OBJC_TYPE(RTCIceServer) *server =
+      [[RTC_OBJC_TYPE(RTCIceServer) alloc] initWithURLStrings:urlStrings];
+
+  RTC_OBJC_TYPE(RTCConfiguration) *config = [[RTC_OBJC_TYPE(RTCConfiguration) alloc] init];
+  config.iceServers = @[ server ];
+  config.iceTransportPolicy = RTCIceTransportPolicyRelay;
+  config.bundlePolicy = RTCBundlePolicyMaxBundle;
+  config.rtcpMuxPolicy = RTCRtcpMuxPolicyNegotiate;
+  config.tcpCandidatePolicy = RTCTcpCandidatePolicyDisabled;
+  config.candidateNetworkPolicy = RTCCandidateNetworkPolicyLowCost;
+  const int maxPackets = 60;
+  const int timeout = 1;
+  const int interval = 2;
+  config.audioJitterBufferMaxPackets = maxPackets;
+  config.audioJitterBufferFastAccelerate = YES;
+  config.iceConnectionReceivingTimeout = timeout;
+  config.iceBackupCandidatePairPingInterval = interval;
+  config.continualGatheringPolicy =
+      RTCContinualGatheringPolicyGatherContinually;
+  config.shouldPruneTurnPorts = YES;
+  config.cryptoOptions =
+      [[RTC_OBJC_TYPE(RTCCryptoOptions) alloc] initWithSrtpEnableGcmCryptoSuites:YES
+                                             srtpEnableAes128Sha1_32CryptoCipher:NO
+                                          srtpEnableEncryptedRtpHeaderExtensions:NO
+                                                    sframeRequireFrameEncryption:NO];
+  config.rtcpAudioReportIntervalMs = 1500;
+  config.rtcpVideoReportIntervalMs = 2150;
+
+  webrtc::PeerConnectionInterface::RTCConfiguration *nativeConfig =
+      [config createNativeConfiguration];
+  RTC_OBJC_TYPE(RTCConfiguration) *newConfig =
+      [[RTC_OBJC_TYPE(RTCConfiguration) alloc] initWithNativeConfiguration:*nativeConfig];
+  EXPECT_EQ([config.iceServers count], newConfig.iceServers.count);
+  RTC_OBJC_TYPE(RTCIceServer) *newServer = newConfig.iceServers[0];
+  RTC_OBJC_TYPE(RTCIceServer) *origServer = config.iceServers[0];
+  EXPECT_EQ(origServer.urlStrings.count, server.urlStrings.count);
+  std::string origUrl = origServer.urlStrings.firstObject.UTF8String;
+  std::string url = newServer.urlStrings.firstObject.UTF8String;
+  EXPECT_EQ(origUrl, url);
+
+  EXPECT_EQ(config.iceTransportPolicy, newConfig.iceTransportPolicy);
+  EXPECT_EQ(config.bundlePolicy, newConfig.bundlePolicy);
+  EXPECT_EQ(config.rtcpMuxPolicy, newConfig.rtcpMuxPolicy);
+  EXPECT_EQ(config.tcpCandidatePolicy, newConfig.tcpCandidatePolicy);
+  EXPECT_EQ(config.candidateNetworkPolicy, newConfig.candidateNetworkPolicy);
+  EXPECT_EQ(config.audioJitterBufferMaxPackets, newConfig.audioJitterBufferMaxPackets);
+  EXPECT_EQ(config.audioJitterBufferFastAccelerate, newConfig.audioJitterBufferFastAccelerate);
+  EXPECT_EQ(config.iceConnectionReceivingTimeout, newConfig.iceConnectionReceivingTimeout);
+  EXPECT_EQ(config.iceBackupCandidatePairPingInterval,
+            newConfig.iceBackupCandidatePairPingInterval);
+  EXPECT_EQ(config.continualGatheringPolicy, newConfig.continualGatheringPolicy);
+  EXPECT_EQ(config.shouldPruneTurnPorts, newConfig.shouldPruneTurnPorts);
+  EXPECT_EQ(config.cryptoOptions.srtpEnableGcmCryptoSuites,
+            newConfig.cryptoOptions.srtpEnableGcmCryptoSuites);
+  EXPECT_EQ(config.cryptoOptions.srtpEnableAes128Sha1_32CryptoCipher,
+            newConfig.cryptoOptions.srtpEnableAes128Sha1_32CryptoCipher);
+  EXPECT_EQ(config.cryptoOptions.srtpEnableEncryptedRtpHeaderExtensions,
+            newConfig.cryptoOptions.srtpEnableEncryptedRtpHeaderExtensions);
+  EXPECT_EQ(config.cryptoOptions.sframeRequireFrameEncryption,
+            newConfig.cryptoOptions.sframeRequireFrameEncryption);
+  EXPECT_EQ(config.rtcpAudioReportIntervalMs, newConfig.rtcpAudioReportIntervalMs);
+  EXPECT_EQ(config.rtcpVideoReportIntervalMs, newConfig.rtcpVideoReportIntervalMs);
+}
+
+- (void)testDefaultValues {
+  RTC_OBJC_TYPE(RTCConfiguration) *config = [[RTC_OBJC_TYPE(RTCConfiguration) alloc] init];
+  EXPECT_EQ(config.cryptoOptions, nil);
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/unittests/RTCDataChannelConfigurationTest.mm b/third_party/libwebrtc/sdk/objc/unittests/RTCDataChannelConfigurationTest.mm
new file mode 100644
index 0000000000..ccebd74198
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/unittests/RTCDataChannelConfigurationTest.mm
@@ -0,0 +1,51 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+#import <XCTest/XCTest.h>
+
+#include "rtc_base/gunit.h"
+
+#import "api/peerconnection/RTCDataChannelConfiguration+Private.h"
+#import "api/peerconnection/RTCDataChannelConfiguration.h"
+#import "helpers/NSString+StdString.h"
+
+@interface RTCDataChannelConfigurationTest : XCTestCase
+@end
+
+@implementation RTCDataChannelConfigurationTest
+
+- (void)testConversionToNativeDataChannelInit {
+  BOOL isOrdered = NO;
+  int maxPacketLifeTime = 5;
+  int maxRetransmits = 4;
+  BOOL isNegotiated = YES;
+  int channelId = 4;
+  NSString *protocol = @"protocol";
+
+  RTC_OBJC_TYPE(RTCDataChannelConfiguration) *dataChannelConfig =
+      [[RTC_OBJC_TYPE(RTCDataChannelConfiguration) alloc] init];
+  dataChannelConfig.isOrdered = isOrdered;
+  dataChannelConfig.maxPacketLifeTime = maxPacketLifeTime;
+  dataChannelConfig.maxRetransmits = maxRetransmits;
+  dataChannelConfig.isNegotiated = isNegotiated;
+  dataChannelConfig.channelId = channelId;
+  dataChannelConfig.protocol = protocol;
+
+  webrtc::DataChannelInit nativeInit = dataChannelConfig.nativeDataChannelInit;
+  EXPECT_EQ(isOrdered, nativeInit.ordered);
+  EXPECT_EQ(maxPacketLifeTime, nativeInit.maxRetransmitTime);
+  EXPECT_EQ(maxRetransmits, nativeInit.maxRetransmits);
+  EXPECT_EQ(isNegotiated, nativeInit.negotiated);
+  EXPECT_EQ(channelId, nativeInit.id);
+  EXPECT_EQ(protocol.stdString, nativeInit.protocol);
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/unittests/RTCDoNotPutCPlusPlusInFrameworkHeaders_xctest.m b/third_party/libwebrtc/sdk/objc/unittests/RTCDoNotPutCPlusPlusInFrameworkHeaders_xctest.m
new file mode 100644
index 0000000000..02bef9bfb7
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/unittests/RTCDoNotPutCPlusPlusInFrameworkHeaders_xctest.m
@@ -0,0 +1,30 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <XCTest/XCTest.h>
+
+#import <Foundation/Foundation.h>
+
+#import <WebRTC/WebRTC.h>
+
+@interface RTCDoNotPutCPlusPlusInFrameworkHeaders : XCTestCase
+@end
+
+@implementation RTCDoNotPutCPlusPlusInFrameworkHeaders
+
+- (void)testNoCPlusPlusInFrameworkHeaders {
+  NSString *fullPath = [NSString stringWithFormat:@"%s", __FILE__];
+  NSString *extension = fullPath.pathExtension;
+
+  XCTAssertEqualObjects(
+      @"m", extension, @"Do not rename %@. It should end with .m.", fullPath.lastPathComponent);
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/unittests/RTCEncodedImage_xctest.mm b/third_party/libwebrtc/sdk/objc/unittests/RTCEncodedImage_xctest.mm
new file mode 100644
index 0000000000..84804fee87
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/unittests/RTCEncodedImage_xctest.mm
@@ -0,0 +1,55 @@
+/*
+ *  Copyright 2020 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "api/peerconnection/RTCEncodedImage+Private.h"
+
+#import <XCTest/XCTest.h>
+
+@interface RTCEncodedImageTests : XCTestCase
+@end
+
+@implementation RTCEncodedImageTests
+
+- (void)testInitializedWithNativeEncodedImage {
+  const auto encoded_data = webrtc::EncodedImageBuffer::Create();
+  webrtc::EncodedImage encoded_image;
+  encoded_image.SetEncodedData(encoded_data);
+
+  RTC_OBJC_TYPE(RTCEncodedImage) *encodedImage =
+      [[RTC_OBJC_TYPE(RTCEncodedImage) alloc] initWithNativeEncodedImage:encoded_image];
+
+  XCTAssertEqual([encodedImage nativeEncodedImage].GetEncodedData(), encoded_data);
+}
+
+- (void)testInitWithNSData {
+  NSData *bufferData = [NSData data];
+  RTC_OBJC_TYPE(RTCEncodedImage) *encodedImage = [[RTC_OBJC_TYPE(RTCEncodedImage) alloc] init];
+  encodedImage.buffer = bufferData;
+
+  webrtc::EncodedImage result_encoded_image = [encodedImage nativeEncodedImage];
+  XCTAssertTrue(result_encoded_image.GetEncodedData() != nullptr);
+  XCTAssertEqual(result_encoded_image.GetEncodedData()->data(), bufferData.bytes);
+}
+
+- (void)testRetainsNativeEncodedImage {
+  RTC_OBJC_TYPE(RTCEncodedImage) * encodedImage;
+  {
+    const auto encoded_data = webrtc::EncodedImageBuffer::Create();
+    webrtc::EncodedImage encoded_image;
+    encoded_image.SetEncodedData(encoded_data);
+    encodedImage =
+        [[RTC_OBJC_TYPE(RTCEncodedImage) alloc] initWithNativeEncodedImage:encoded_image];
+  }
+  webrtc::EncodedImage result_encoded_image = [encodedImage nativeEncodedImage];
+  XCTAssertTrue(result_encoded_image.GetEncodedData() != nullptr);
+  XCTAssertTrue(result_encoded_image.GetEncodedData()->data() != nullptr);
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/unittests/RTCFileVideoCapturer_xctest.mm b/third_party/libwebrtc/sdk/objc/unittests/RTCFileVideoCapturer_xctest.mm
new file mode 100644
index 0000000000..2407c88c1a
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/unittests/RTCFileVideoCapturer_xctest.mm
@@ -0,0 +1,114 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "components/capturer/RTCFileVideoCapturer.h"
+
+#import <XCTest/XCTest.h>
+
+#include "rtc_base/gunit.h"
+
+NSString *const kTestFileName = @"foreman.mp4";
+static const int kTestTimeoutMs = 5 * 1000;  // 5secs.
+
+@interface MockCapturerDelegate : NSObject <RTC_OBJC_TYPE (RTCVideoCapturerDelegate)>
+
+@property(nonatomic, assign) NSInteger capturedFramesCount;
+
+@end
+
+@implementation MockCapturerDelegate
+@synthesize capturedFramesCount = _capturedFramesCount;
+
+- (void)capturer:(RTC_OBJC_TYPE(RTCVideoCapturer) *)capturer
+    didCaptureVideoFrame:(RTC_OBJC_TYPE(RTCVideoFrame) *)frame {
+  self.capturedFramesCount++;
+}
+
+@end
+
+NS_CLASS_AVAILABLE_IOS(10)
+@interface RTCFileVideoCapturerTests : XCTestCase
+
+@property(nonatomic, strong) RTC_OBJC_TYPE(RTCFileVideoCapturer) * capturer;
+@property(nonatomic, strong) MockCapturerDelegate *mockDelegate;
+
+@end
+
+@implementation RTCFileVideoCapturerTests
+@synthesize capturer = _capturer;
+@synthesize mockDelegate = _mockDelegate;
+
+- (void)setUp {
+  self.mockDelegate = [[MockCapturerDelegate alloc] init];
+  self.capturer = [[RTC_OBJC_TYPE(RTCFileVideoCapturer) alloc] initWithDelegate:self.mockDelegate];
+}
+
+- (void)tearDown {
+  self.capturer = nil;
+  self.mockDelegate = nil;
+}
+
+- (void)testCaptureWhenFileNotInBundle {
+  __block BOOL errorOccured = NO;
+
+  RTCFileVideoCapturerErrorBlock errorBlock = ^void(NSError *error) {
+    errorOccured = YES;
+  };
+
+  [self.capturer startCapturingFromFileNamed:@"not_in_bundle.mov" onError:errorBlock];
+  ASSERT_TRUE_WAIT(errorOccured, kTestTimeoutMs);
+}
+
+- (void)testSecondStartCaptureCallFails {
+  __block BOOL secondError = NO;
+
+  RTCFileVideoCapturerErrorBlock firstErrorBlock = ^void(NSError *error) {
+    // This block should never be called.
+    NSLog(@"Error: %@", [error userInfo]);
+    ASSERT_TRUE(false);
+  };
+
+  RTCFileVideoCapturerErrorBlock secondErrorBlock = ^void(NSError *error) {
+    secondError = YES;
+  };
+
+  [self.capturer startCapturingFromFileNamed:kTestFileName onError:firstErrorBlock];
+  [self.capturer startCapturingFromFileNamed:kTestFileName onError:secondErrorBlock];
+
+  ASSERT_TRUE_WAIT(secondError, kTestTimeoutMs);
+}
+
+- (void)testStartStopCapturer {
+#if defined(__IPHONE_11_0) && (__IPHONE_OS_VERSION_MAX_ALLOWED >= __IPHONE_11_0)
+  if (@available(iOS 10, *)) {
+    [self.capturer startCapturingFromFileNamed:kTestFileName onError:nil];
+
+    __block BOOL done = NO;
+    __block NSInteger capturedFrames = -1;
+    NSInteger capturedFramesAfterStop = -1;
+
+    // We're dispatching the `stopCapture` with delay to ensure the capturer has
+    // had the chance to capture several frames.
+    dispatch_time_t captureDelay = dispatch_time(DISPATCH_TIME_NOW, 2 * NSEC_PER_SEC);  // 2secs.
+    dispatch_after(captureDelay, dispatch_get_global_queue(DISPATCH_QUEUE_PRIORITY_DEFAULT, 0), ^{
+      capturedFrames = self.mockDelegate.capturedFramesCount;
+      [self.capturer stopCapture];
+      done = YES;
+    });
+    WAIT(done, kTestTimeoutMs);
+
+    capturedFramesAfterStop = self.mockDelegate.capturedFramesCount;
+    ASSERT_TRUE(capturedFrames != -1);
+    ASSERT_EQ(capturedFrames, capturedFramesAfterStop);
+  }
+#endif
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/unittests/RTCH264ProfileLevelId_xctest.m b/third_party/libwebrtc/sdk/objc/unittests/RTCH264ProfileLevelId_xctest.m
new file mode 100644
index 0000000000..ec9dc41796
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/unittests/RTCH264ProfileLevelId_xctest.m
@@ -0,0 +1,48 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "components/video_codec/RTCH264ProfileLevelId.h"
+
+#import <XCTest/XCTest.h>
+
+@interface RTCH264ProfileLevelIdTests : XCTestCase
+
+@end
+
+static NSString *level31ConstrainedHigh = @"640c1f";
+static NSString *level31ConstrainedBaseline = @"42e01f";
+
+@implementation RTCH264ProfileLevelIdTests
+
+- (void)testInitWithString {
+  RTC_OBJC_TYPE(RTCH264ProfileLevelId) *profileLevelId =
+      [[RTC_OBJC_TYPE(RTCH264ProfileLevelId) alloc] initWithHexString:level31ConstrainedHigh];
+  XCTAssertEqual(profileLevelId.profile, RTCH264ProfileConstrainedHigh);
+  XCTAssertEqual(profileLevelId.level, RTCH264Level3_1);
+
+  profileLevelId =
+      [[RTC_OBJC_TYPE(RTCH264ProfileLevelId) alloc] initWithHexString:level31ConstrainedBaseline];
+  XCTAssertEqual(profileLevelId.profile, RTCH264ProfileConstrainedBaseline);
+  XCTAssertEqual(profileLevelId.level, RTCH264Level3_1);
+}
+
+- (void)testInitWithProfileAndLevel {
+  RTC_OBJC_TYPE(RTCH264ProfileLevelId) *profileLevelId =
+      [[RTC_OBJC_TYPE(RTCH264ProfileLevelId) alloc] initWithProfile:RTCH264ProfileConstrainedHigh
+                                                              level:RTCH264Level3_1];
+  XCTAssertEqualObjects(profileLevelId.hexString, level31ConstrainedHigh);
+
+  profileLevelId = [[RTC_OBJC_TYPE(RTCH264ProfileLevelId) alloc]
+      initWithProfile:RTCH264ProfileConstrainedBaseline
+                level:RTCH264Level3_1];
+  XCTAssertEqualObjects(profileLevelId.hexString, level31ConstrainedBaseline);
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/unittests/RTCIceCandidateTest.mm b/third_party/libwebrtc/sdk/objc/unittests/RTCIceCandidateTest.mm
new file mode 100644
index 0000000000..576411985d
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/unittests/RTCIceCandidateTest.mm
@@ -0,0 +1,60 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+#import <XCTest/XCTest.h>
+
+#include <memory>
+
+#include "rtc_base/gunit.h"
+
+#import "api/peerconnection/RTCIceCandidate+Private.h"
+#import "api/peerconnection/RTCIceCandidate.h"
+#import "helpers/NSString+StdString.h"
+
+@interface RTCIceCandidateTest : XCTestCase
+@end
+
+@implementation RTCIceCandidateTest
+
+- (void)testCandidate {
+  NSString *sdp = @"candidate:4025901590 1 udp 2122265343 "
+                   "fdff:2642:12a6:fe38:c001:beda:fcf9:51aa "
+                   "59052 typ host generation 0";
+
+  RTC_OBJC_TYPE(RTCIceCandidate) *candidate =
+      [[RTC_OBJC_TYPE(RTCIceCandidate) alloc] initWithSdp:sdp sdpMLineIndex:0 sdpMid:@"audio"];
+
+  std::unique_ptr<webrtc::IceCandidateInterface> nativeCandidate =
+      candidate.nativeCandidate;
+  EXPECT_EQ("audio", nativeCandidate->sdp_mid());
+  EXPECT_EQ(0, nativeCandidate->sdp_mline_index());
+
+  std::string sdpString;
+  nativeCandidate->ToString(&sdpString);
+  EXPECT_EQ(sdp.stdString, sdpString);
+}
+
+- (void)testInitFromNativeCandidate {
+  std::string sdp("candidate:4025901590 1 udp 2122265343 "
+                  "fdff:2642:12a6:fe38:c001:beda:fcf9:51aa "
+                  "59052 typ host generation 0");
+  webrtc::IceCandidateInterface *nativeCandidate =
+      webrtc::CreateIceCandidate("audio", 0, sdp, nullptr);
+
+  RTC_OBJC_TYPE(RTCIceCandidate) *iceCandidate =
+      [[RTC_OBJC_TYPE(RTCIceCandidate) alloc] initWithNativeCandidate:nativeCandidate];
+  EXPECT_TRUE([@"audio" isEqualToString:iceCandidate.sdpMid]);
+  EXPECT_EQ(0, iceCandidate.sdpMLineIndex);
+
+  EXPECT_EQ(sdp, iceCandidate.sdp.stdString);
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/unittests/RTCIceServerTest.mm b/third_party/libwebrtc/sdk/objc/unittests/RTCIceServerTest.mm
new file mode 100644
index 0000000000..772653c4dc
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/unittests/RTCIceServerTest.mm
@@ -0,0 +1,136 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+#import <XCTest/XCTest.h>
+
+#include <vector>
+
+#include "rtc_base/gunit.h"
+
+#import "api/peerconnection/RTCIceServer+Private.h"
+#import "api/peerconnection/RTCIceServer.h"
+#import "helpers/NSString+StdString.h"
+
+@interface RTCIceServerTest : XCTestCase
+@end
+
+@implementation RTCIceServerTest
+
+- (void)testOneURLServer {
+  RTC_OBJC_TYPE(RTCIceServer) *server =
+      [[RTC_OBJC_TYPE(RTCIceServer) alloc] initWithURLStrings:@[ @"stun:stun1.example.net" ]];
+
+  webrtc::PeerConnectionInterface::IceServer iceStruct = server.nativeServer;
+  EXPECT_EQ(1u, iceStruct.urls.size());
+  EXPECT_EQ("stun:stun1.example.net", iceStruct.urls.front());
+  EXPECT_EQ("", iceStruct.username);
+  EXPECT_EQ("", iceStruct.password);
+}
+
+- (void)testTwoURLServer {
+  RTC_OBJC_TYPE(RTCIceServer) *server = [[RTC_OBJC_TYPE(RTCIceServer) alloc]
+      initWithURLStrings:@[ @"turn1:turn1.example.net", @"turn2:turn2.example.net" ]];
+
+  webrtc::PeerConnectionInterface::IceServer iceStruct = server.nativeServer;
+  EXPECT_EQ(2u, iceStruct.urls.size());
+  EXPECT_EQ("turn1:turn1.example.net", iceStruct.urls.front());
+  EXPECT_EQ("turn2:turn2.example.net", iceStruct.urls.back());
+  EXPECT_EQ("", iceStruct.username);
+  EXPECT_EQ("", iceStruct.password);
+}
+
+- (void)testPasswordCredential {
+  RTC_OBJC_TYPE(RTCIceServer) *server =
+      [[RTC_OBJC_TYPE(RTCIceServer) alloc] initWithURLStrings:@[ @"turn1:turn1.example.net" ]
+                                                     username:@"username"
+                                                   credential:@"credential"];
+  webrtc::PeerConnectionInterface::IceServer iceStruct = server.nativeServer;
+  EXPECT_EQ(1u, iceStruct.urls.size());
+  EXPECT_EQ("turn1:turn1.example.net", iceStruct.urls.front());
+  EXPECT_EQ("username", iceStruct.username);
+  EXPECT_EQ("credential", iceStruct.password);
+}
+
+- (void)testHostname {
+  RTC_OBJC_TYPE(RTCIceServer) *server =
+      [[RTC_OBJC_TYPE(RTCIceServer) alloc] initWithURLStrings:@[ @"turn1:turn1.example.net" ]
+                                                     username:@"username"
+                                                   credential:@"credential"
+                                                tlsCertPolicy:RTCTlsCertPolicySecure
+                                                     hostname:@"hostname"];
+  webrtc::PeerConnectionInterface::IceServer iceStruct = server.nativeServer;
+  EXPECT_EQ(1u, iceStruct.urls.size());
+  EXPECT_EQ("turn1:turn1.example.net", iceStruct.urls.front());
+  EXPECT_EQ("username", iceStruct.username);
+  EXPECT_EQ("credential", iceStruct.password);
+  EXPECT_EQ("hostname", iceStruct.hostname);
+}
+
+- (void)testTlsAlpnProtocols {
+  RTC_OBJC_TYPE(RTCIceServer) *server =
+      [[RTC_OBJC_TYPE(RTCIceServer) alloc] initWithURLStrings:@[ @"turn1:turn1.example.net" ]
+                                                     username:@"username"
+                                                   credential:@"credential"
+                                                tlsCertPolicy:RTCTlsCertPolicySecure
+                                                     hostname:@"hostname"
+                                             tlsAlpnProtocols:@[ @"proto1", @"proto2" ]];
+  webrtc::PeerConnectionInterface::IceServer iceStruct = server.nativeServer;
+  EXPECT_EQ(1u, iceStruct.urls.size());
+  EXPECT_EQ("turn1:turn1.example.net", iceStruct.urls.front());
+  EXPECT_EQ("username", iceStruct.username);
+  EXPECT_EQ("credential", iceStruct.password);
+  EXPECT_EQ("hostname", iceStruct.hostname);
+  EXPECT_EQ(2u, iceStruct.tls_alpn_protocols.size());
+}
+
+- (void)testTlsEllipticCurves {
+  RTC_OBJC_TYPE(RTCIceServer) *server =
+      [[RTC_OBJC_TYPE(RTCIceServer) alloc] initWithURLStrings:@[ @"turn1:turn1.example.net" ]
+                                                     username:@"username"
+                                                   credential:@"credential"
+                                                tlsCertPolicy:RTCTlsCertPolicySecure
+                                                     hostname:@"hostname"
+                                             tlsAlpnProtocols:@[ @"proto1", @"proto2" ]
+                                            tlsEllipticCurves:@[ @"curve1", @"curve2" ]];
+  webrtc::PeerConnectionInterface::IceServer iceStruct = server.nativeServer;
+  EXPECT_EQ(1u, iceStruct.urls.size());
+  EXPECT_EQ("turn1:turn1.example.net", iceStruct.urls.front());
+  EXPECT_EQ("username", iceStruct.username);
+  EXPECT_EQ("credential", iceStruct.password);
+  EXPECT_EQ("hostname", iceStruct.hostname);
+  EXPECT_EQ(2u, iceStruct.tls_alpn_protocols.size());
+  EXPECT_EQ(2u, iceStruct.tls_elliptic_curves.size());
+}
+
+- (void)testInitFromNativeServer {
+  webrtc::PeerConnectionInterface::IceServer nativeServer;
+  nativeServer.username = "username";
+  nativeServer.password = "password";
+  nativeServer.urls.push_back("stun:stun.example.net");
+  nativeServer.hostname = "hostname";
+  nativeServer.tls_alpn_protocols.push_back("proto1");
+  nativeServer.tls_alpn_protocols.push_back("proto2");
+  nativeServer.tls_elliptic_curves.push_back("curve1");
+  nativeServer.tls_elliptic_curves.push_back("curve2");
+
+  RTC_OBJC_TYPE(RTCIceServer) *iceServer =
+      [[RTC_OBJC_TYPE(RTCIceServer) alloc] initWithNativeServer:nativeServer];
+  EXPECT_EQ(1u, iceServer.urlStrings.count);
+  EXPECT_EQ("stun:stun.example.net",
+      [NSString stdStringForString:iceServer.urlStrings.firstObject]);
+  EXPECT_EQ("username", [NSString stdStringForString:iceServer.username]);
+  EXPECT_EQ("password", [NSString stdStringForString:iceServer.credential]);
+  EXPECT_EQ("hostname", [NSString stdStringForString:iceServer.hostname]);
+  EXPECT_EQ(2u, iceServer.tlsAlpnProtocols.count);
+  EXPECT_EQ(2u, iceServer.tlsEllipticCurves.count);
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/unittests/RTCMTLVideoView_xctest.m b/third_party/libwebrtc/sdk/objc/unittests/RTCMTLVideoView_xctest.m
new file mode 100644
index 0000000000..f152eeec91
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/unittests/RTCMTLVideoView_xctest.m
@@ -0,0 +1,298 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <XCTest/XCTest.h>
+
+#import <Foundation/Foundation.h>
+#import <MetalKit/MetalKit.h>
+#import <OCMock/OCMock.h>
+
+#import "components/renderer/metal/RTCMTLVideoView.h"
+
+#import "api/video_frame_buffer/RTCNativeI420Buffer.h"
+#import "base/RTCVideoFrameBuffer.h"
+#import "components/renderer/metal/RTCMTLNV12Renderer.h"
+#import "components/video_frame_buffer/RTCCVPixelBuffer.h"
+
+static size_t kBufferWidth = 200;
+static size_t kBufferHeight = 200;
+
+// Extension of RTC_OBJC_TYPE(RTCMTLVideoView) for testing purposes.
+@interface RTC_OBJC_TYPE (RTCMTLVideoView)
+(Testing)
+
+    @property(nonatomic, readonly) MTKView *metalView;
+
++ (BOOL)isMetalAvailable;
++ (UIView *)createMetalView:(CGRect)frame;
++ (id<RTCMTLRenderer>)createNV12Renderer;
++ (id<RTCMTLRenderer>)createI420Renderer;
+- (void)drawInMTKView:(id)view;
+@end
+
+@interface RTCMTLVideoViewTests : XCTestCase
+@property(nonatomic, strong) id classMock;
+@property(nonatomic, strong) id rendererNV12Mock;
+@property(nonatomic, strong) id rendererI420Mock;
+@property(nonatomic, strong) id frameMock;
+@end
+
+@implementation RTCMTLVideoViewTests
+
+@synthesize classMock = _classMock;
+@synthesize rendererNV12Mock = _rendererNV12Mock;
+@synthesize rendererI420Mock = _rendererI420Mock;
+@synthesize frameMock = _frameMock;
+
+- (void)setUp {
+  self.classMock = OCMClassMock([RTC_OBJC_TYPE(RTCMTLVideoView) class]);
+  [self startMockingNilView];
+}
+
+- (void)tearDown {
+  [self.classMock stopMocking];
+  [self.rendererI420Mock stopMocking];
+  [self.rendererNV12Mock stopMocking];
+  [self.frameMock stopMocking];
+  self.classMock = nil;
+  self.rendererI420Mock = nil;
+  self.rendererNV12Mock = nil;
+  self.frameMock = nil;
+}
+
+- (id)frameMockWithCVPixelBuffer:(BOOL)hasCVPixelBuffer {
+  id frameMock = OCMClassMock([RTC_OBJC_TYPE(RTCVideoFrame) class]);
+  if (hasCVPixelBuffer) {
+    CVPixelBufferRef pixelBufferRef;
+    CVPixelBufferCreate(kCFAllocatorDefault,
+                        kBufferWidth,
+                        kBufferHeight,
+                        kCVPixelFormatType_420YpCbCr8Planar,
+                        nil,
+                        &pixelBufferRef);
+    OCMStub([frameMock buffer])
+        .andReturn([[RTC_OBJC_TYPE(RTCCVPixelBuffer) alloc] initWithPixelBuffer:pixelBufferRef]);
+  } else {
+    OCMStub([frameMock buffer])
+        .andReturn([[RTC_OBJC_TYPE(RTCI420Buffer) alloc] initWithWidth:kBufferWidth
+                                                                height:kBufferHeight]);
+  }
+  OCMStub([((RTC_OBJC_TYPE(RTCVideoFrame) *)frameMock) width]).andReturn(kBufferWidth);
+  OCMStub([((RTC_OBJC_TYPE(RTCVideoFrame) *)frameMock) height]).andReturn(kBufferHeight);
+  OCMStub([frameMock timeStampNs]).andReturn(arc4random_uniform(INT_MAX));
+  return frameMock;
+}
+
+- (id)rendererMockWithSuccessfulSetup:(BOOL)success {
+  id rendererMock = OCMClassMock([RTCMTLRenderer class]);
+  OCMStub([rendererMock addRenderingDestination:[OCMArg any]]).andReturn(success);
+  return rendererMock;
+}
+
+- (void)startMockingNilView {
+  // Use OCMock 2 syntax here until OCMock is upgraded to 3.4
+  [[[self.classMock stub] andReturn:nil] createMetalView:CGRectZero];
+}
+
+#pragma mark - Test cases
+
+- (void)testInitAssertsIfMetalUnavailabe {
+  // given
+  OCMStub([self.classMock isMetalAvailable]).andReturn(NO);
+
+  // when
+  BOOL asserts = NO;
+  @try {
+    RTC_OBJC_TYPE(RTCMTLVideoView) *realView =
+        [[RTC_OBJC_TYPE(RTCMTLVideoView) alloc] initWithFrame:CGRectZero];
+    (void)realView;
+  } @catch (NSException *ex) {
+    asserts = YES;
+  }
+
+  XCTAssertTrue(asserts);
+}
+
+- (void)testRTCVideoRenderNilFrameCallback {
+  // given
+  OCMStub([self.classMock isMetalAvailable]).andReturn(YES);
+
+  RTC_OBJC_TYPE(RTCMTLVideoView) *realView =
+      [[RTC_OBJC_TYPE(RTCMTLVideoView) alloc] initWithFrame:CGRectMake(0, 0, 640, 480)];
+  self.frameMock = OCMClassMock([RTC_OBJC_TYPE(RTCVideoFrame) class]);
+
+  [[self.frameMock reject] buffer];
+  [[self.classMock reject] createNV12Renderer];
+  [[self.classMock reject] createI420Renderer];
+
+  // when
+  [realView renderFrame:nil];
+  [realView drawInMTKView:realView.metalView];
+
+  // then
+  [self.frameMock verify];
+  [self.classMock verify];
+}
+
+- (void)testRTCVideoRenderFrameCallbackI420 {
+  // given
+  OCMStub([self.classMock isMetalAvailable]).andReturn(YES);
+  self.rendererI420Mock = [self rendererMockWithSuccessfulSetup:YES];
+  self.frameMock = [self frameMockWithCVPixelBuffer:NO];
+
+  OCMExpect([self.rendererI420Mock drawFrame:self.frameMock]);
+  OCMExpect([self.classMock createI420Renderer]).andReturn(self.rendererI420Mock);
+  [[self.classMock reject] createNV12Renderer];
+
+  RTC_OBJC_TYPE(RTCMTLVideoView) *realView =
+      [[RTC_OBJC_TYPE(RTCMTLVideoView) alloc] initWithFrame:CGRectMake(0, 0, 640, 480)];
+
+  // when
+  [realView renderFrame:self.frameMock];
+  [realView drawInMTKView:realView.metalView];
+
+  // then
+  [self.rendererI420Mock verify];
+  [self.classMock verify];
+}
+
+- (void)testRTCVideoRenderFrameCallbackNV12 {
+  // given
+  OCMStub([self.classMock isMetalAvailable]).andReturn(YES);
+  self.rendererNV12Mock = [self rendererMockWithSuccessfulSetup:YES];
+  self.frameMock = [self frameMockWithCVPixelBuffer:YES];
+
+  OCMExpect([self.rendererNV12Mock drawFrame:self.frameMock]);
+  OCMExpect([self.classMock createNV12Renderer]).andReturn(self.rendererNV12Mock);
+  [[self.classMock reject] createI420Renderer];
+
+  RTC_OBJC_TYPE(RTCMTLVideoView) *realView =
+      [[RTC_OBJC_TYPE(RTCMTLVideoView) alloc] initWithFrame:CGRectMake(0, 0, 640, 480)];
+
+  // when
+  [realView renderFrame:self.frameMock];
+  [realView drawInMTKView:realView.metalView];
+
+  // then
+  [self.rendererNV12Mock verify];
+  [self.classMock verify];
+}
+
+- (void)testRTCVideoRenderWorksAfterReconstruction {
+  OCMStub([self.classMock isMetalAvailable]).andReturn(YES);
+  self.rendererNV12Mock = [self rendererMockWithSuccessfulSetup:YES];
+  self.frameMock = [self frameMockWithCVPixelBuffer:YES];
+
+  OCMExpect([self.rendererNV12Mock drawFrame:self.frameMock]);
+  OCMExpect([self.classMock createNV12Renderer]).andReturn(self.rendererNV12Mock);
+  [[self.classMock reject] createI420Renderer];
+
+  RTC_OBJC_TYPE(RTCMTLVideoView) *realView =
+      [[RTC_OBJC_TYPE(RTCMTLVideoView) alloc] initWithFrame:CGRectMake(0, 0, 640, 480)];
+
+  [realView renderFrame:self.frameMock];
+  [realView drawInMTKView:realView.metalView];
+  [self.rendererNV12Mock verify];
+  [self.classMock verify];
+
+  // Recreate view.
+  realView = [[RTC_OBJC_TYPE(RTCMTLVideoView) alloc] initWithFrame:CGRectMake(0, 0, 640, 480)];
+  OCMExpect([self.rendererNV12Mock drawFrame:self.frameMock]);
+  // View hould reinit renderer.
+  OCMExpect([self.classMock createNV12Renderer]).andReturn(self.rendererNV12Mock);
+
+  [realView renderFrame:self.frameMock];
+  [realView drawInMTKView:realView.metalView];
+  [self.rendererNV12Mock verify];
+  [self.classMock verify];
+}
+
+- (void)testDontRedrawOldFrame {
+  OCMStub([self.classMock isMetalAvailable]).andReturn(YES);
+  self.rendererNV12Mock = [self rendererMockWithSuccessfulSetup:YES];
+  self.frameMock = [self frameMockWithCVPixelBuffer:YES];
+
+  OCMExpect([self.rendererNV12Mock drawFrame:self.frameMock]);
+  OCMExpect([self.classMock createNV12Renderer]).andReturn(self.rendererNV12Mock);
+  [[self.classMock reject] createI420Renderer];
+
+  RTC_OBJC_TYPE(RTCMTLVideoView) *realView =
+      [[RTC_OBJC_TYPE(RTCMTLVideoView) alloc] initWithFrame:CGRectMake(0, 0, 640, 480)];
+  [realView renderFrame:self.frameMock];
+  [realView drawInMTKView:realView.metalView];
+
+  [self.rendererNV12Mock verify];
+  [self.classMock verify];
+
+  [[self.rendererNV12Mock reject] drawFrame:[OCMArg any]];
+
+  [realView renderFrame:self.frameMock];
+  [realView drawInMTKView:realView.metalView];
+
+  [self.rendererNV12Mock verify];
+}
+
+- (void)testDoDrawNewFrame {
+  OCMStub([self.classMock isMetalAvailable]).andReturn(YES);
+  self.rendererNV12Mock = [self rendererMockWithSuccessfulSetup:YES];
+  self.frameMock = [self frameMockWithCVPixelBuffer:YES];
+
+  OCMExpect([self.rendererNV12Mock drawFrame:self.frameMock]);
+  OCMExpect([self.classMock createNV12Renderer]).andReturn(self.rendererNV12Mock);
+  [[self.classMock reject] createI420Renderer];
+
+  RTC_OBJC_TYPE(RTCMTLVideoView) *realView =
+      [[RTC_OBJC_TYPE(RTCMTLVideoView) alloc] initWithFrame:CGRectMake(0, 0, 640, 480)];
+  [realView renderFrame:self.frameMock];
+  [realView drawInMTKView:realView.metalView];
+
+  [self.rendererNV12Mock verify];
+  [self.classMock verify];
+
+  // Get new frame.
+  self.frameMock = [self frameMockWithCVPixelBuffer:YES];
+  OCMExpect([self.rendererNV12Mock drawFrame:self.frameMock]);
+
+  [realView renderFrame:self.frameMock];
+  [realView drawInMTKView:realView.metalView];
+
+  [self.rendererNV12Mock verify];
+}
+
+- (void)testReportsSizeChangesToDelegate {
+  OCMStub([self.classMock isMetalAvailable]).andReturn(YES);
+
+  id delegateMock = OCMProtocolMock(@protocol(RTC_OBJC_TYPE(RTCVideoViewDelegate)));
+  CGSize size = CGSizeMake(640, 480);
+  OCMExpect([delegateMock videoView:[OCMArg any] didChangeVideoSize:size]);
+
+  RTC_OBJC_TYPE(RTCMTLVideoView) *realView =
+      [[RTC_OBJC_TYPE(RTCMTLVideoView) alloc] initWithFrame:CGRectMake(0, 0, 640, 480)];
+  realView.delegate = delegateMock;
+  [realView setSize:size];
+
+  // Delegate method is invoked with a dispatch_async.
+  OCMVerifyAllWithDelay(delegateMock, 1);
+}
+
+- (void)testSetContentMode {
+  OCMStub([self.classMock isMetalAvailable]).andReturn(YES);
+  id metalKitView = OCMClassMock([MTKView class]);
+  [[[[self.classMock stub] ignoringNonObjectArgs] andReturn:metalKitView]
+      createMetalView:CGRectZero];
+  OCMExpect([metalKitView setContentMode:UIViewContentModeScaleAspectFill]);
+
+  RTC_OBJC_TYPE(RTCMTLVideoView) *realView = [[RTC_OBJC_TYPE(RTCMTLVideoView) alloc] init];
+  [realView setVideoContentMode:UIViewContentModeScaleAspectFill];
+
+  OCMVerify(metalKitView);
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/unittests/RTCMediaConstraintsTest.mm b/third_party/libwebrtc/sdk/objc/unittests/RTCMediaConstraintsTest.mm
new file mode 100644
index 0000000000..6ed7859ba1
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/unittests/RTCMediaConstraintsTest.mm
@@ -0,0 +1,58 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+#import <XCTest/XCTest.h>
+
+#include <memory>
+
+#include "rtc_base/gunit.h"
+
+#import "api/peerconnection/RTCMediaConstraints+Private.h"
+#import "api/peerconnection/RTCMediaConstraints.h"
+#import "helpers/NSString+StdString.h"
+
+@interface RTCMediaConstraintsTests : XCTestCase
+@end
+
+@implementation RTCMediaConstraintsTests
+
+- (void)testMediaConstraints {
+  NSDictionary *mandatory = @{@"key1": @"value1", @"key2": @"value2"};
+  NSDictionary *optional = @{@"key3": @"value3", @"key4": @"value4"};
+
+  RTC_OBJC_TYPE(RTCMediaConstraints) *constraints =
+      [[RTC_OBJC_TYPE(RTCMediaConstraints) alloc] initWithMandatoryConstraints:mandatory
+                                                           optionalConstraints:optional];
+  std::unique_ptr<webrtc::MediaConstraints> nativeConstraints =
+      [constraints nativeConstraints];
+
+  webrtc::MediaConstraints::Constraints nativeMandatory = nativeConstraints->GetMandatory();
+  [self expectConstraints:mandatory inNativeConstraints:nativeMandatory];
+
+  webrtc::MediaConstraints::Constraints nativeOptional = nativeConstraints->GetOptional();
+  [self expectConstraints:optional inNativeConstraints:nativeOptional];
+}
+
+- (void)expectConstraints:(NSDictionary *)constraints
+      inNativeConstraints:(webrtc::MediaConstraints::Constraints)nativeConstraints {
+  EXPECT_EQ(constraints.count, nativeConstraints.size());
+
+  for (NSString *key in constraints) {
+    NSString *value = [constraints objectForKey:key];
+
+    std::string nativeValue;
+    bool found = nativeConstraints.FindFirst(key.stdString, &nativeValue);
+    EXPECT_TRUE(found);
+    EXPECT_EQ(value.stdString, nativeValue);
+  }
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/unittests/RTCNV12TextureCache_xctest.m b/third_party/libwebrtc/sdk/objc/unittests/RTCNV12TextureCache_xctest.m
new file mode 100644
index 0000000000..7bdc538f67
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/unittests/RTCNV12TextureCache_xctest.m
@@ -0,0 +1,55 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <CoreVideo/CoreVideo.h>
+#import <Foundation/Foundation.h>
+#import <GLKit/GLKit.h>
+#import <XCTest/XCTest.h>
+
+#import "base/RTCVideoFrame.h"
+#import "base/RTCVideoFrameBuffer.h"
+#import "components/renderer/opengl/RTCNV12TextureCache.h"
+#import "components/video_frame_buffer/RTCCVPixelBuffer.h"
+
+@interface RTCNV12TextureCacheTests : XCTestCase
+@end
+
+@implementation RTCNV12TextureCacheTests {
+  EAGLContext *_glContext;
+  RTCNV12TextureCache *_nv12TextureCache;
+}
+
+- (void)setUp {
+  [super setUp];
+  _glContext = [[EAGLContext alloc] initWithAPI:kEAGLRenderingAPIOpenGLES3];
+  if (!_glContext) {
+    _glContext = [[EAGLContext alloc] initWithAPI:kEAGLRenderingAPIOpenGLES2];
+  }
+  _nv12TextureCache = [[RTCNV12TextureCache alloc] initWithContext:_glContext];
+}
+
+- (void)tearDown {
+  _nv12TextureCache = nil;
+  _glContext = nil;
+  [super tearDown];
+}
+
+- (void)testNV12TextureCacheDoesNotCrashOnEmptyFrame {
+  CVPixelBufferRef nullPixelBuffer = NULL;
+  RTC_OBJC_TYPE(RTCCVPixelBuffer) *badFrameBuffer =
+      [[RTC_OBJC_TYPE(RTCCVPixelBuffer) alloc] initWithPixelBuffer:nullPixelBuffer];
+  RTC_OBJC_TYPE(RTCVideoFrame) *badFrame =
+      [[RTC_OBJC_TYPE(RTCVideoFrame) alloc] initWithBuffer:badFrameBuffer
+                                                  rotation:RTCVideoRotation_0
+                                               timeStampNs:0];
+  [_nv12TextureCache uploadFrameToTextures:badFrame];
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/unittests/RTCPeerConnectionFactoryBuilderTest.mm b/third_party/libwebrtc/sdk/objc/unittests/RTCPeerConnectionFactoryBuilderTest.mm
new file mode 100644
index 0000000000..5ba5a52a53
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/unittests/RTCPeerConnectionFactoryBuilderTest.mm
@@ -0,0 +1,72 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+#import <XCTest/XCTest.h>
+#ifdef __cplusplus
+extern "C" {
+#endif
+#import <OCMock/OCMock.h>
+#ifdef __cplusplus
+}
+#endif
+#import "api/peerconnection/RTCPeerConnectionFactory+Native.h"
+#import "api/peerconnection/RTCPeerConnectionFactoryBuilder+DefaultComponents.h"
+#import "api/peerconnection/RTCPeerConnectionFactoryBuilder.h"
+
+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
+#include "api/audio_codecs/builtin_audio_encoder_factory.h"
+#include "api/video_codecs/video_decoder_factory.h"
+#include "api/video_codecs/video_encoder_factory.h"
+#include "modules/audio_device/include/audio_device.h"
+#include "modules/audio_processing/include/audio_processing.h"
+
+#include "rtc_base/gunit.h"
+#include "rtc_base/system/unused.h"
+
+@interface RTCPeerConnectionFactoryBuilderTests : XCTestCase
+@end
+
+@implementation RTCPeerConnectionFactoryBuilderTests
+
+- (void)testBuilder {
+  id factoryMock = OCMStrictClassMock([RTC_OBJC_TYPE(RTCPeerConnectionFactory) class]);
+  OCMExpect([factoryMock alloc]).andReturn(factoryMock);
+  RTC_UNUSED([[[[factoryMock expect] andReturn:factoryMock] ignoringNonObjectArgs]
+      initWithNativeAudioEncoderFactory:nullptr
+              nativeAudioDecoderFactory:nullptr
+              nativeVideoEncoderFactory:nullptr
+              nativeVideoDecoderFactory:nullptr
+                      audioDeviceModule:nullptr
+                  audioProcessingModule:nullptr]);
+  RTCPeerConnectionFactoryBuilder* builder = [[RTCPeerConnectionFactoryBuilder alloc] init];
+  RTC_OBJC_TYPE(RTCPeerConnectionFactory)* peerConnectionFactory =
+      [builder createPeerConnectionFactory];
+  EXPECT_TRUE(peerConnectionFactory != nil);
+  OCMVerifyAll(factoryMock);
+}
+
+- (void)testDefaultComponentsBuilder {
+  id factoryMock = OCMStrictClassMock([RTC_OBJC_TYPE(RTCPeerConnectionFactory) class]);
+  OCMExpect([factoryMock alloc]).andReturn(factoryMock);
+  RTC_UNUSED([[[[factoryMock expect] andReturn:factoryMock] ignoringNonObjectArgs]
+      initWithNativeAudioEncoderFactory:nullptr
+              nativeAudioDecoderFactory:nullptr
+              nativeVideoEncoderFactory:nullptr
+              nativeVideoDecoderFactory:nullptr
+                      audioDeviceModule:nullptr
+                  audioProcessingModule:nullptr]);
+  RTCPeerConnectionFactoryBuilder* builder = [RTCPeerConnectionFactoryBuilder defaultBuilder];
+  RTC_OBJC_TYPE(RTCPeerConnectionFactory)* peerConnectionFactory =
+      [builder createPeerConnectionFactory];
+  EXPECT_TRUE(peerConnectionFactory != nil);
+  OCMVerifyAll(factoryMock);
+}
+@end
diff --git a/third_party/libwebrtc/sdk/objc/unittests/RTCPeerConnectionFactory_xctest.m b/third_party/libwebrtc/sdk/objc/unittests/RTCPeerConnectionFactory_xctest.m
new file mode 100644
index 0000000000..28743a6fc1
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/unittests/RTCPeerConnectionFactory_xctest.m
@@ -0,0 +1,380 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import "api/peerconnection/RTCAudioSource.h"
+#import "api/peerconnection/RTCConfiguration.h"
+#import "api/peerconnection/RTCDataChannel.h"
+#import "api/peerconnection/RTCDataChannelConfiguration.h"
+#import "api/peerconnection/RTCMediaConstraints.h"
+#import "api/peerconnection/RTCMediaStreamTrack.h"
+#import "api/peerconnection/RTCPeerConnection.h"
+#import "api/peerconnection/RTCPeerConnectionFactory.h"
+#import "api/peerconnection/RTCRtpReceiver.h"
+#import "api/peerconnection/RTCRtpSender.h"
+#import "api/peerconnection/RTCRtpTransceiver.h"
+#import "api/peerconnection/RTCSessionDescription.h"
+#import "api/peerconnection/RTCVideoSource.h"
+#import "rtc_base/system/unused.h"
+
+#import <XCTest/XCTest.h>
+
+@interface RTCPeerConnectionFactoryTests : XCTestCase
+@end
+
+@implementation RTCPeerConnectionFactoryTests
+
+- (void)testPeerConnectionLifetime {
+  @autoreleasepool {
+    RTC_OBJC_TYPE(RTCConfiguration) *config = [[RTC_OBJC_TYPE(RTCConfiguration) alloc] init];
+
+    RTC_OBJC_TYPE(RTCMediaConstraints) *constraints =
+        [[RTC_OBJC_TYPE(RTCMediaConstraints) alloc] initWithMandatoryConstraints:@{}
+                                                             optionalConstraints:nil];
+
+    RTC_OBJC_TYPE(RTCPeerConnectionFactory) * factory;
+    RTC_OBJC_TYPE(RTCPeerConnection) * peerConnection;
+
+    @autoreleasepool {
+      factory = [[RTC_OBJC_TYPE(RTCPeerConnectionFactory) alloc] init];
+      peerConnection =
+          [factory peerConnectionWithConfiguration:config constraints:constraints delegate:nil];
+      [peerConnection close];
+      factory = nil;
+    }
+    peerConnection = nil;
+  }
+
+  XCTAssertTrue(true, @"Expect test does not crash");
+}
+
+- (void)testMediaStreamLifetime {
+  @autoreleasepool {
+    RTC_OBJC_TYPE(RTCPeerConnectionFactory) * factory;
+    RTC_OBJC_TYPE(RTCMediaStream) * mediaStream;
+
+    @autoreleasepool {
+      factory = [[RTC_OBJC_TYPE(RTCPeerConnectionFactory) alloc] init];
+      mediaStream = [factory mediaStreamWithStreamId:@"mediaStream"];
+      factory = nil;
+    }
+    mediaStream = nil;
+    RTC_UNUSED(mediaStream);
+  }
+
+  XCTAssertTrue(true, "Expect test does not crash");
+}
+
+- (void)testDataChannelLifetime {
+  @autoreleasepool {
+    RTC_OBJC_TYPE(RTCConfiguration) *config = [[RTC_OBJC_TYPE(RTCConfiguration) alloc] init];
+    RTC_OBJC_TYPE(RTCMediaConstraints) *constraints =
+        [[RTC_OBJC_TYPE(RTCMediaConstraints) alloc] initWithMandatoryConstraints:@{}
+                                                             optionalConstraints:nil];
+    RTC_OBJC_TYPE(RTCDataChannelConfiguration) *dataChannelConfig =
+        [[RTC_OBJC_TYPE(RTCDataChannelConfiguration) alloc] init];
+
+    RTC_OBJC_TYPE(RTCPeerConnectionFactory) * factory;
+    RTC_OBJC_TYPE(RTCPeerConnection) * peerConnection;
+    RTC_OBJC_TYPE(RTCDataChannel) * dataChannel;
+
+    @autoreleasepool {
+      factory = [[RTC_OBJC_TYPE(RTCPeerConnectionFactory) alloc] init];
+      peerConnection =
+          [factory peerConnectionWithConfiguration:config constraints:constraints delegate:nil];
+      dataChannel =
+          [peerConnection dataChannelForLabel:@"test_channel" configuration:dataChannelConfig];
+      XCTAssertNotNil(dataChannel);
+      [peerConnection close];
+      peerConnection = nil;
+      factory = nil;
+    }
+    dataChannel = nil;
+  }
+
+  XCTAssertTrue(true, "Expect test does not crash");
+}
+
+- (void)testRTCRtpTransceiverLifetime {
+  @autoreleasepool {
+    RTC_OBJC_TYPE(RTCConfiguration) *config = [[RTC_OBJC_TYPE(RTCConfiguration) alloc] init];
+    config.sdpSemantics = RTCSdpSemanticsUnifiedPlan;
+    RTC_OBJC_TYPE(RTCMediaConstraints) *contraints =
+        [[RTC_OBJC_TYPE(RTCMediaConstraints) alloc] initWithMandatoryConstraints:@{}
+                                                             optionalConstraints:nil];
+    RTC_OBJC_TYPE(RTCRtpTransceiverInit) *init =
+        [[RTC_OBJC_TYPE(RTCRtpTransceiverInit) alloc] init];
+
+    RTC_OBJC_TYPE(RTCPeerConnectionFactory) * factory;
+    RTC_OBJC_TYPE(RTCPeerConnection) * peerConnection;
+    RTC_OBJC_TYPE(RTCRtpTransceiver) * tranceiver;
+
+    @autoreleasepool {
+      factory = [[RTC_OBJC_TYPE(RTCPeerConnectionFactory) alloc] init];
+      peerConnection =
+          [factory peerConnectionWithConfiguration:config constraints:contraints delegate:nil];
+      tranceiver = [peerConnection addTransceiverOfType:RTCRtpMediaTypeAudio init:init];
+      XCTAssertNotNil(tranceiver);
+      [peerConnection close];
+      peerConnection = nil;
+      factory = nil;
+    }
+    tranceiver = nil;
+  }
+
+  XCTAssertTrue(true, "Expect test does not crash");
+}
+
+- (void)testRTCRtpSenderLifetime {
+  @autoreleasepool {
+    RTC_OBJC_TYPE(RTCConfiguration) *config = [[RTC_OBJC_TYPE(RTCConfiguration) alloc] init];
+    config.sdpSemantics = RTCSdpSemanticsPlanB;
+    RTC_OBJC_TYPE(RTCMediaConstraints) *constraints =
+        [[RTC_OBJC_TYPE(RTCMediaConstraints) alloc] initWithMandatoryConstraints:@{}
+                                                             optionalConstraints:nil];
+
+    RTC_OBJC_TYPE(RTCPeerConnectionFactory) * factory;
+    RTC_OBJC_TYPE(RTCPeerConnection) * peerConnection;
+    RTC_OBJC_TYPE(RTCRtpSender) * sender;
+
+    @autoreleasepool {
+      factory = [[RTC_OBJC_TYPE(RTCPeerConnectionFactory) alloc] init];
+      peerConnection =
+          [factory peerConnectionWithConfiguration:config constraints:constraints delegate:nil];
+      sender = [peerConnection senderWithKind:kRTCMediaStreamTrackKindVideo streamId:@"stream"];
+      XCTAssertNotNil(sender);
+      [peerConnection close];
+      peerConnection = nil;
+      factory = nil;
+    }
+    sender = nil;
+  }
+
+  XCTAssertTrue(true, "Expect test does not crash");
+}
+
+- (void)testRTCRtpReceiverLifetime {
+  @autoreleasepool {
+    RTC_OBJC_TYPE(RTCConfiguration) *config = [[RTC_OBJC_TYPE(RTCConfiguration) alloc] init];
+    config.sdpSemantics = RTCSdpSemanticsPlanB;
+    RTC_OBJC_TYPE(RTCMediaConstraints) *constraints =
+        [[RTC_OBJC_TYPE(RTCMediaConstraints) alloc] initWithMandatoryConstraints:@{}
+                                                             optionalConstraints:nil];
+
+    RTC_OBJC_TYPE(RTCPeerConnectionFactory) * factory;
+    RTC_OBJC_TYPE(RTCPeerConnection) * pc1;
+    RTC_OBJC_TYPE(RTCPeerConnection) * pc2;
+
+    NSArray<RTC_OBJC_TYPE(RTCRtpReceiver) *> *receivers1;
+    NSArray<RTC_OBJC_TYPE(RTCRtpReceiver) *> *receivers2;
+
+    @autoreleasepool {
+      factory = [[RTC_OBJC_TYPE(RTCPeerConnectionFactory) alloc] init];
+      pc1 = [factory peerConnectionWithConfiguration:config constraints:constraints delegate:nil];
+      [pc1 senderWithKind:kRTCMediaStreamTrackKindAudio streamId:@"stream"];
+
+      pc2 = [factory peerConnectionWithConfiguration:config constraints:constraints delegate:nil];
+      [pc2 senderWithKind:kRTCMediaStreamTrackKindAudio streamId:@"stream"];
+
+      NSTimeInterval negotiationTimeout = 15;
+      XCTAssertTrue([self negotiatePeerConnection:pc1
+                               withPeerConnection:pc2
+                               negotiationTimeout:negotiationTimeout]);
+
+      XCTAssertEqual(pc1.signalingState, RTCSignalingStateStable);
+      XCTAssertEqual(pc2.signalingState, RTCSignalingStateStable);
+
+      receivers1 = pc1.receivers;
+      receivers2 = pc2.receivers;
+      XCTAssertTrue(receivers1.count > 0);
+      XCTAssertTrue(receivers2.count > 0);
+      [pc1 close];
+      [pc2 close];
+      pc1 = nil;
+      pc2 = nil;
+      factory = nil;
+    }
+    receivers1 = nil;
+    receivers2 = nil;
+  }
+
+  XCTAssertTrue(true, "Expect test does not crash");
+}
+
+- (void)testAudioSourceLifetime {
+  @autoreleasepool {
+    RTC_OBJC_TYPE(RTCPeerConnectionFactory) * factory;
+    RTC_OBJC_TYPE(RTCAudioSource) * audioSource;
+
+    @autoreleasepool {
+      factory = [[RTC_OBJC_TYPE(RTCPeerConnectionFactory) alloc] init];
+      audioSource = [factory audioSourceWithConstraints:nil];
+      XCTAssertNotNil(audioSource);
+      factory = nil;
+    }
+    audioSource = nil;
+  }
+
+  XCTAssertTrue(true, "Expect test does not crash");
+}
+
+- (void)testVideoSourceLifetime {
+  @autoreleasepool {
+    RTC_OBJC_TYPE(RTCPeerConnectionFactory) * factory;
+    RTC_OBJC_TYPE(RTCVideoSource) * videoSource;
+
+    @autoreleasepool {
+      factory = [[RTC_OBJC_TYPE(RTCPeerConnectionFactory) alloc] init];
+      videoSource = [factory videoSource];
+      XCTAssertNotNil(videoSource);
+      factory = nil;
+    }
+    videoSource = nil;
+  }
+
+  XCTAssertTrue(true, "Expect test does not crash");
+}
+
+- (void)testAudioTrackLifetime {
+  @autoreleasepool {
+    RTC_OBJC_TYPE(RTCPeerConnectionFactory) * factory;
+    RTC_OBJC_TYPE(RTCAudioTrack) * audioTrack;
+
+    @autoreleasepool {
+      factory = [[RTC_OBJC_TYPE(RTCPeerConnectionFactory) alloc] init];
+      audioTrack = [factory audioTrackWithTrackId:@"audioTrack"];
+      XCTAssertNotNil(audioTrack);
+      factory = nil;
+    }
+    audioTrack = nil;
+  }
+
+  XCTAssertTrue(true, "Expect test does not crash");
+}
+
+- (void)testVideoTrackLifetime {
+  @autoreleasepool {
+    RTC_OBJC_TYPE(RTCPeerConnectionFactory) * factory;
+    RTC_OBJC_TYPE(RTCVideoTrack) * videoTrack;
+
+    @autoreleasepool {
+      factory = [[RTC_OBJC_TYPE(RTCPeerConnectionFactory) alloc] init];
+      videoTrack = [factory videoTrackWithSource:[factory videoSource] trackId:@"videoTrack"];
+      XCTAssertNotNil(videoTrack);
+      factory = nil;
+    }
+    videoTrack = nil;
+  }
+
+  XCTAssertTrue(true, "Expect test does not crash");
+}
+
+- (void)testRollback {
+  @autoreleasepool {
+    RTC_OBJC_TYPE(RTCConfiguration) *config = [[RTC_OBJC_TYPE(RTCConfiguration) alloc] init];
+    config.sdpSemantics = RTCSdpSemanticsUnifiedPlan;
+    RTC_OBJC_TYPE(RTCMediaConstraints) *constraints =
+        [[RTC_OBJC_TYPE(RTCMediaConstraints) alloc] initWithMandatoryConstraints:@{
+          kRTCMediaConstraintsOfferToReceiveAudio : kRTCMediaConstraintsValueTrue
+        }
+                                                             optionalConstraints:nil];
+
+    __block RTC_OBJC_TYPE(RTCPeerConnectionFactory) * factory;
+    __block RTC_OBJC_TYPE(RTCPeerConnection) * pc1;
+    RTCSessionDescription *rollback = [[RTCSessionDescription alloc] initWithType:RTCSdpTypeRollback
+                                                                              sdp:@""];
+
+    @autoreleasepool {
+      factory = [[RTC_OBJC_TYPE(RTCPeerConnectionFactory) alloc] init];
+      pc1 = [factory peerConnectionWithConfiguration:config constraints:constraints delegate:nil];
+      dispatch_semaphore_t negotiatedSem = dispatch_semaphore_create(0);
+      [pc1 offerForConstraints:constraints
+             completionHandler:^(RTC_OBJC_TYPE(RTCSessionDescription) * offer, NSError * error) {
+               XCTAssertNil(error);
+               XCTAssertNotNil(offer);
+
+               __weak RTC_OBJC_TYPE(RTCPeerConnection) *weakPC1 = pc1;
+               [pc1 setLocalDescription:offer
+                      completionHandler:^(NSError *error) {
+                        XCTAssertNil(error);
+                        [weakPC1 setLocalDescription:rollback
+                                   completionHandler:^(NSError *error) {
+                                     XCTAssertNil(error);
+                                   }];
+                      }];
+               NSTimeInterval negotiationTimeout = 15;
+               dispatch_semaphore_wait(
+                   negotiatedSem,
+                   dispatch_time(DISPATCH_TIME_NOW, (int64_t)(negotiationTimeout * NSEC_PER_SEC)));
+
+               XCTAssertEqual(pc1.signalingState, RTCSignalingStateStable);
+
+               [pc1 close];
+               pc1 = nil;
+               factory = nil;
+             }];
+    }
+
+    XCTAssertTrue(true, "Expect test does not crash");
+  }
+}
+
+- (bool)negotiatePeerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)pc1
+             withPeerConnection:(RTC_OBJC_TYPE(RTCPeerConnection) *)pc2
+             negotiationTimeout:(NSTimeInterval)timeout {
+  __weak RTC_OBJC_TYPE(RTCPeerConnection) *weakPC1 = pc1;
+  __weak RTC_OBJC_TYPE(RTCPeerConnection) *weakPC2 = pc2;
+  RTC_OBJC_TYPE(RTCMediaConstraints) *sdpConstraints =
+      [[RTC_OBJC_TYPE(RTCMediaConstraints) alloc] initWithMandatoryConstraints:@{
+        kRTCMediaConstraintsOfferToReceiveAudio : kRTCMediaConstraintsValueTrue
+      }
+                                                           optionalConstraints:nil];
+
+  dispatch_semaphore_t negotiatedSem = dispatch_semaphore_create(0);
+  [weakPC1 offerForConstraints:sdpConstraints
+             completionHandler:^(RTC_OBJC_TYPE(RTCSessionDescription) * offer, NSError * error) {
+               XCTAssertNil(error);
+               XCTAssertNotNil(offer);
+               [weakPC1
+                   setLocalDescription:offer
+                     completionHandler:^(NSError *error) {
+                       XCTAssertNil(error);
+                       [weakPC2
+                           setRemoteDescription:offer
+                              completionHandler:^(NSError *error) {
+                                XCTAssertNil(error);
+                                [weakPC2
+                                    answerForConstraints:sdpConstraints
+                                       completionHandler:^(
+                                           RTC_OBJC_TYPE(RTCSessionDescription) * answer,
+                                           NSError * error) {
+                                         XCTAssertNil(error);
+                                         XCTAssertNotNil(answer);
+                                         [weakPC2
+                                             setLocalDescription:answer
+                                               completionHandler:^(NSError *error) {
+                                                 XCTAssertNil(error);
+                                                 [weakPC1
+                                                     setRemoteDescription:answer
+                                                        completionHandler:^(NSError *error) {
+                                                          XCTAssertNil(error);
+                                                          dispatch_semaphore_signal(negotiatedSem);
+                                                        }];
+                                               }];
+                                       }];
+                              }];
+                     }];
+             }];
+
+  return 0 ==
+      dispatch_semaphore_wait(negotiatedSem,
+                              dispatch_time(DISPATCH_TIME_NOW, (int64_t)(timeout * NSEC_PER_SEC)));
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/unittests/RTCPeerConnectionTest.mm b/third_party/libwebrtc/sdk/objc/unittests/RTCPeerConnectionTest.mm
new file mode 100644
index 0000000000..9ca8403559
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/unittests/RTCPeerConnectionTest.mm
@@ -0,0 +1,204 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+#import <XCTest/XCTest.h>
+
+#include <memory>
+#include <vector>
+
+#include "rtc_base/gunit.h"
+
+#import "api/peerconnection/RTCConfiguration+Private.h"
+#import "api/peerconnection/RTCConfiguration.h"
+#import "api/peerconnection/RTCCryptoOptions.h"
+#import "api/peerconnection/RTCIceCandidate.h"
+#import "api/peerconnection/RTCIceServer.h"
+#import "api/peerconnection/RTCMediaConstraints.h"
+#import "api/peerconnection/RTCPeerConnection.h"
+#import "api/peerconnection/RTCPeerConnectionFactory+Native.h"
+#import "api/peerconnection/RTCPeerConnectionFactory.h"
+#import "api/peerconnection/RTCSessionDescription.h"
+#import "helpers/NSString+StdString.h"
+
+@interface RTCPeerConnectionTests : XCTestCase
+@end
+
+@implementation RTCPeerConnectionTests
+
+- (void)testConfigurationGetter {
+  NSArray *urlStrings = @[ @"stun:stun1.example.net" ];
+  RTC_OBJC_TYPE(RTCIceServer) *server =
+      [[RTC_OBJC_TYPE(RTCIceServer) alloc] initWithURLStrings:urlStrings];
+
+  RTC_OBJC_TYPE(RTCConfiguration) *config = [[RTC_OBJC_TYPE(RTCConfiguration) alloc] init];
+  config.sdpSemantics = RTCSdpSemanticsUnifiedPlan;
+  config.iceServers = @[ server ];
+  config.iceTransportPolicy = RTCIceTransportPolicyRelay;
+  config.bundlePolicy = RTCBundlePolicyMaxBundle;
+  config.rtcpMuxPolicy = RTCRtcpMuxPolicyNegotiate;
+  config.tcpCandidatePolicy = RTCTcpCandidatePolicyDisabled;
+  config.candidateNetworkPolicy = RTCCandidateNetworkPolicyLowCost;
+  const int maxPackets = 60;
+  const int timeout = 1500;
+  const int interval = 2000;
+  config.audioJitterBufferMaxPackets = maxPackets;
+  config.audioJitterBufferFastAccelerate = YES;
+  config.iceConnectionReceivingTimeout = timeout;
+  config.iceBackupCandidatePairPingInterval = interval;
+  config.continualGatheringPolicy =
+      RTCContinualGatheringPolicyGatherContinually;
+  config.shouldPruneTurnPorts = YES;
+  config.activeResetSrtpParams = YES;
+  config.cryptoOptions =
+      [[RTC_OBJC_TYPE(RTCCryptoOptions) alloc] initWithSrtpEnableGcmCryptoSuites:YES
+                                             srtpEnableAes128Sha1_32CryptoCipher:YES
+                                          srtpEnableEncryptedRtpHeaderExtensions:NO
+                                                    sframeRequireFrameEncryption:NO];
+
+  RTC_OBJC_TYPE(RTCMediaConstraints) *contraints =
+      [[RTC_OBJC_TYPE(RTCMediaConstraints) alloc] initWithMandatoryConstraints:@{}
+                                                           optionalConstraints:nil];
+  RTC_OBJC_TYPE(RTCPeerConnectionFactory) *factory =
+      [[RTC_OBJC_TYPE(RTCPeerConnectionFactory) alloc] init];
+
+  RTC_OBJC_TYPE(RTCConfiguration) * newConfig;
+  @autoreleasepool {
+    RTC_OBJC_TYPE(RTCPeerConnection) *peerConnection =
+        [factory peerConnectionWithConfiguration:config constraints:contraints delegate:nil];
+    newConfig = peerConnection.configuration;
+
+    EXPECT_TRUE([peerConnection setBweMinBitrateBps:[NSNumber numberWithInt:100000]
+                                  currentBitrateBps:[NSNumber numberWithInt:5000000]
+                                      maxBitrateBps:[NSNumber numberWithInt:500000000]]);
+    EXPECT_FALSE([peerConnection setBweMinBitrateBps:[NSNumber numberWithInt:2]
+                                   currentBitrateBps:[NSNumber numberWithInt:1]
+                                       maxBitrateBps:nil]);
+  }
+
+  EXPECT_EQ([config.iceServers count], [newConfig.iceServers count]);
+  RTC_OBJC_TYPE(RTCIceServer) *newServer = newConfig.iceServers[0];
+  RTC_OBJC_TYPE(RTCIceServer) *origServer = config.iceServers[0];
+  std::string origUrl = origServer.urlStrings.firstObject.UTF8String;
+  std::string url = newServer.urlStrings.firstObject.UTF8String;
+  EXPECT_EQ(origUrl, url);
+
+  EXPECT_EQ(config.iceTransportPolicy, newConfig.iceTransportPolicy);
+  EXPECT_EQ(config.bundlePolicy, newConfig.bundlePolicy);
+  EXPECT_EQ(config.rtcpMuxPolicy, newConfig.rtcpMuxPolicy);
+  EXPECT_EQ(config.tcpCandidatePolicy, newConfig.tcpCandidatePolicy);
+  EXPECT_EQ(config.candidateNetworkPolicy, newConfig.candidateNetworkPolicy);
+  EXPECT_EQ(config.audioJitterBufferMaxPackets, newConfig.audioJitterBufferMaxPackets);
+  EXPECT_EQ(config.audioJitterBufferFastAccelerate, newConfig.audioJitterBufferFastAccelerate);
+  EXPECT_EQ(config.iceConnectionReceivingTimeout, newConfig.iceConnectionReceivingTimeout);
+  EXPECT_EQ(config.iceBackupCandidatePairPingInterval,
+            newConfig.iceBackupCandidatePairPingInterval);
+  EXPECT_EQ(config.continualGatheringPolicy, newConfig.continualGatheringPolicy);
+  EXPECT_EQ(config.shouldPruneTurnPorts, newConfig.shouldPruneTurnPorts);
+  EXPECT_EQ(config.activeResetSrtpParams, newConfig.activeResetSrtpParams);
+  EXPECT_EQ(config.cryptoOptions.srtpEnableGcmCryptoSuites,
+            newConfig.cryptoOptions.srtpEnableGcmCryptoSuites);
+  EXPECT_EQ(config.cryptoOptions.srtpEnableAes128Sha1_32CryptoCipher,
+            newConfig.cryptoOptions.srtpEnableAes128Sha1_32CryptoCipher);
+  EXPECT_EQ(config.cryptoOptions.srtpEnableEncryptedRtpHeaderExtensions,
+            newConfig.cryptoOptions.srtpEnableEncryptedRtpHeaderExtensions);
+  EXPECT_EQ(config.cryptoOptions.sframeRequireFrameEncryption,
+            newConfig.cryptoOptions.sframeRequireFrameEncryption);
+}
+
+- (void)testWithDependencies {
+  NSArray *urlStrings = @[ @"stun:stun1.example.net" ];
+  RTC_OBJC_TYPE(RTCIceServer) *server =
+      [[RTC_OBJC_TYPE(RTCIceServer) alloc] initWithURLStrings:urlStrings];
+
+  RTC_OBJC_TYPE(RTCConfiguration) *config = [[RTC_OBJC_TYPE(RTCConfiguration) alloc] init];
+  config.sdpSemantics = RTCSdpSemanticsUnifiedPlan;
+  config.iceServers = @[ server ];
+  RTC_OBJC_TYPE(RTCMediaConstraints) *contraints =
+      [[RTC_OBJC_TYPE(RTCMediaConstraints) alloc] initWithMandatoryConstraints:@{}
+                                                           optionalConstraints:nil];
+  RTC_OBJC_TYPE(RTCPeerConnectionFactory) *factory =
+      [[RTC_OBJC_TYPE(RTCPeerConnectionFactory) alloc] init];
+
+  std::unique_ptr<webrtc::PeerConnectionDependencies> pc_dependencies =
+      std::make_unique<webrtc::PeerConnectionDependencies>(nullptr);
+  @autoreleasepool {
+    RTC_OBJC_TYPE(RTCPeerConnection) *peerConnection =
+        [factory peerConnectionWithDependencies:config
+                                    constraints:contraints
+                                   dependencies:std::move(pc_dependencies)
+                                       delegate:nil];
+    ASSERT_NE(peerConnection, nil);
+  }
+}
+
+- (void)testWithInvalidSDP {
+  RTC_OBJC_TYPE(RTCPeerConnectionFactory) *factory =
+      [[RTC_OBJC_TYPE(RTCPeerConnectionFactory) alloc] init];
+
+  RTC_OBJC_TYPE(RTCConfiguration) *config = [[RTC_OBJC_TYPE(RTCConfiguration) alloc] init];
+  config.sdpSemantics = RTCSdpSemanticsUnifiedPlan;
+  RTC_OBJC_TYPE(RTCMediaConstraints) *contraints =
+      [[RTC_OBJC_TYPE(RTCMediaConstraints) alloc] initWithMandatoryConstraints:@{}
+                                                           optionalConstraints:nil];
+  RTC_OBJC_TYPE(RTCPeerConnection) *peerConnection =
+      [factory peerConnectionWithConfiguration:config constraints:contraints delegate:nil];
+
+  dispatch_semaphore_t negotiatedSem = dispatch_semaphore_create(0);
+  [peerConnection setRemoteDescription:[[RTC_OBJC_TYPE(RTCSessionDescription) alloc]
+                                           initWithType:RTCSdpTypeOffer
+                                                    sdp:@"invalid"]
+                     completionHandler:^(NSError *error) {
+                       ASSERT_NE(error, nil);
+                       if (error != nil) {
+                         dispatch_semaphore_signal(negotiatedSem);
+                       }
+                     }];
+
+  NSTimeInterval timeout = 5;
+  ASSERT_EQ(
+      0,
+      dispatch_semaphore_wait(negotiatedSem,
+                              dispatch_time(DISPATCH_TIME_NOW, (int64_t)(timeout * NSEC_PER_SEC))));
+  [peerConnection close];
+}
+
+- (void)testWithInvalidIceCandidate {
+  RTC_OBJC_TYPE(RTCPeerConnectionFactory) *factory =
+      [[RTC_OBJC_TYPE(RTCPeerConnectionFactory) alloc] init];
+
+  RTC_OBJC_TYPE(RTCConfiguration) *config = [[RTC_OBJC_TYPE(RTCConfiguration) alloc] init];
+  config.sdpSemantics = RTCSdpSemanticsUnifiedPlan;
+  RTC_OBJC_TYPE(RTCMediaConstraints) *contraints =
+      [[RTC_OBJC_TYPE(RTCMediaConstraints) alloc] initWithMandatoryConstraints:@{}
+                                                           optionalConstraints:nil];
+  RTC_OBJC_TYPE(RTCPeerConnection) *peerConnection =
+      [factory peerConnectionWithConfiguration:config constraints:contraints delegate:nil];
+
+  dispatch_semaphore_t negotiatedSem = dispatch_semaphore_create(0);
+  [peerConnection addIceCandidate:[[RTC_OBJC_TYPE(RTCIceCandidate) alloc] initWithSdp:@"invalid"
+                                                                        sdpMLineIndex:-1
+                                                                               sdpMid:nil]
+                completionHandler:^(NSError *error) {
+                  ASSERT_NE(error, nil);
+                  if (error != nil) {
+                    dispatch_semaphore_signal(negotiatedSem);
+                  }
+                }];
+
+  NSTimeInterval timeout = 5;
+  ASSERT_EQ(
+      0,
+      dispatch_semaphore_wait(negotiatedSem,
+                              dispatch_time(DISPATCH_TIME_NOW, (int64_t)(timeout * NSEC_PER_SEC))));
+  [peerConnection close];
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/unittests/RTCSessionDescriptionTest.mm b/third_party/libwebrtc/sdk/objc/unittests/RTCSessionDescriptionTest.mm
new file mode 100644
index 0000000000..70c82f78ce
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/unittests/RTCSessionDescriptionTest.mm
@@ -0,0 +1,122 @@
+/*
+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+#import <XCTest/XCTest.h>
+
+#include "rtc_base/gunit.h"
+
+#import "api/peerconnection/RTCSessionDescription+Private.h"
+#import "api/peerconnection/RTCSessionDescription.h"
+#import "helpers/NSString+StdString.h"
+
+@interface RTCSessionDescriptionTests : XCTestCase
+@end
+
+@implementation RTCSessionDescriptionTests
+
+/**
+ * Test conversion of an Objective-C RTC_OBJC_TYPE(RTCSessionDescription) to a native
+ * SessionDescriptionInterface (based on the types and SDP strings being equal).
+ */
+- (void)testSessionDescriptionConversion {
+  RTC_OBJC_TYPE(RTCSessionDescription) *description =
+      [[RTC_OBJC_TYPE(RTCSessionDescription) alloc] initWithType:RTCSdpTypeAnswer sdp:[self sdp]];
+
+  std::unique_ptr<webrtc::SessionDescriptionInterface> nativeDescription =
+      description.nativeDescription;
+
+  EXPECT_EQ(RTCSdpTypeAnswer,
+            [RTC_OBJC_TYPE(RTCSessionDescription) typeForStdString:nativeDescription->type()]);
+
+  std::string sdp;
+  nativeDescription->ToString(&sdp);
+  EXPECT_EQ([self sdp].stdString, sdp);
+}
+
+- (void)testInitFromNativeSessionDescription {
+  webrtc::SessionDescriptionInterface *nativeDescription;
+
+  nativeDescription = webrtc::CreateSessionDescription(
+      webrtc::SessionDescriptionInterface::kAnswer,
+      [self sdp].stdString,
+      nullptr);
+
+  RTC_OBJC_TYPE(RTCSessionDescription) *description =
+      [[RTC_OBJC_TYPE(RTCSessionDescription) alloc] initWithNativeDescription:nativeDescription];
+  EXPECT_EQ(webrtc::SessionDescriptionInterface::kAnswer,
+            [RTC_OBJC_TYPE(RTCSessionDescription) stdStringForType:description.type]);
+  EXPECT_TRUE([[self sdp] isEqualToString:description.sdp]);
+}
+
+- (NSString *)sdp {
+  return @"v=0\r\n"
+          "o=- 5319989746393411314 2 IN IP4 127.0.0.1\r\n"
+          "s=-\r\n"
+          "t=0 0\r\n"
+          "a=group:BUNDLE audio video\r\n"
+          "a=msid-semantic: WMS ARDAMS\r\n"
+          "m=audio 9 UDP/TLS/RTP/SAVPF 111 103 9 0 8 126\r\n"
+          "c=IN IP4 0.0.0.0\r\n"
+          "a=rtcp:9 IN IP4 0.0.0.0\r\n"
+          "a=ice-ufrag:f3o+0HG7l9nwIWFY\r\n"
+          "a=ice-pwd:VDctmJNCptR2TB7+meDpw7w5\r\n"
+          "a=fingerprint:sha-256 A9:D5:8D:A8:69:22:39:60:92:AD:94:1A:22:2D:5E:"
+          "A5:4A:A9:18:C2:35:5D:46:5E:59:BD:1C:AF:38:9F:E6:E1\r\n"
+          "a=setup:active\r\n"
+          "a=mid:audio\r\n"
+          "a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level\r\n"
+          "a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/"
+          "abs-send-time\r\n"
+          "a=sendrecv\r\n"
+          "a=rtcp-mux\r\n"
+          "a=rtpmap:111 opus/48000/2\r\n"
+          "a=fmtp:111 minptime=10;useinbandfec=1\r\n"
+          "a=rtpmap:103 ISAC/16000\r\n"
+          "a=rtpmap:9 G722/8000\r\n"
+          "a=rtpmap:0 PCMU/8000\r\n"
+          "a=rtpmap:8 PCMA/8000\r\n"
+          "a=rtpmap:126 telephone-event/8000\r\n"
+          "a=maxptime:60\r\n"
+          "a=ssrc:1504474588 cname:V+FdIC5AJpxLhdYQ\r\n"
+          "a=ssrc:1504474588 msid:ARDAMS ARDAMSa0\r\n"
+          "m=video 9 UDP/TLS/RTP/SAVPF 100 116 117 96\r\n"
+          "c=IN IP4 0.0.0.0\r\n"
+          "a=rtcp:9 IN IP4 0.0.0.0\r\n"
+          "a=ice-ufrag:f3o+0HG7l9nwIWFY\r\n"
+          "a=ice-pwd:VDctmJNCptR2TB7+meDpw7w5\r\n"
+          "a=fingerprint:sha-256 A9:D5:8D:A8:69:22:39:60:92:AD:94:1A:22:2D:5E:"
+          "A5:4A:A9:18:C2:35:5D:46:5E:59:BD:1C:AF:38:9F:E6:E1\r\n"
+          "a=setup:active\r\n"
+          "a=mid:video\r\n"
+          "a=extmap:2 urn:ietf:params:rtp-hdrext:toffset\r\n"
+          "a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/"
+          "abs-send-time\r\n"
+          "a=extmap:4 urn:3gpp:video-orientation\r\n"
+          "a=sendrecv\r\n"
+          "a=rtcp-mux\r\n"
+          "a=rtpmap:100 VP8/90000\r\n"
+          "a=rtcp-fb:100 ccm fir\r\n"
+          "a=rtcp-fb:100 goog-lntf\r\n"
+          "a=rtcp-fb:100 nack\r\n"
+          "a=rtcp-fb:100 nack pli\r\n"
+          "a=rtcp-fb:100 goog-remb\r\n"
+          "a=rtpmap:116 red/90000\r\n"
+          "a=rtpmap:117 ulpfec/90000\r\n"
+          "a=rtpmap:96 rtx/90000\r\n"
+          "a=fmtp:96 apt=100\r\n"
+          "a=ssrc-group:FID 498297514 1644357692\r\n"
+          "a=ssrc:498297514 cname:V+FdIC5AJpxLhdYQ\r\n"
+          "a=ssrc:498297514 msid:ARDAMS ARDAMSv0\r\n"
+          "a=ssrc:1644357692 cname:V+FdIC5AJpxLhdYQ\r\n"
+          "a=ssrc:1644357692 msid:ARDAMS ARDAMSv0\r\n";
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/unittests/RTCTracingTest.mm b/third_party/libwebrtc/sdk/objc/unittests/RTCTracingTest.mm
new file mode 100644
index 0000000000..ff93047bdf
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/unittests/RTCTracingTest.mm
@@ -0,0 +1,41 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+#import <XCTest/XCTest.h>
+
+#include <vector>
+
+#include "rtc_base/gunit.h"
+
+#import "api/peerconnection/RTCTracing.h"
+#import "helpers/NSString+StdString.h"
+
+@interface RTCTracingTests : XCTestCase
+@end
+
+@implementation RTCTracingTests
+
+- (NSString *)documentsFilePathForFileName:(NSString *)fileName {
+  NSParameterAssert(fileName.length);
+  NSArray *paths = NSSearchPathForDirectoriesInDomains(NSDocumentDirectory, NSUserDomainMask, YES);
+  NSString *documentsDirPath = paths.firstObject;
+  NSString *filePath =
+  [documentsDirPath stringByAppendingPathComponent:fileName];
+  return filePath;
+}
+
+- (void)testTracingTestNoInitialization {
+  NSString *filePath = [self documentsFilePathForFileName:@"webrtc-trace.txt"];
+  EXPECT_EQ(NO, RTCStartInternalCapture(filePath));
+  RTCStopInternalCapture();
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/unittests/audio_short16.pcm b/third_party/libwebrtc/sdk/objc/unittests/audio_short16.pcm
new file mode 100644
index 0000000000..15a0f1811c
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/unittests/audio_short16.pcm
diff --git a/third_party/libwebrtc/sdk/objc/unittests/audio_short44.pcm b/third_party/libwebrtc/sdk/objc/unittests/audio_short44.pcm
new file mode 100644
index 0000000000..011cdce959
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/unittests/audio_short44.pcm
diff --git a/third_party/libwebrtc/sdk/objc/unittests/audio_short48.pcm b/third_party/libwebrtc/sdk/objc/unittests/audio_short48.pcm
new file mode 100644
index 0000000000..06fd8261cd
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/unittests/audio_short48.pcm
diff --git a/third_party/libwebrtc/sdk/objc/unittests/avformatmappertests.mm b/third_party/libwebrtc/sdk/objc/unittests/avformatmappertests.mm
new file mode 100644
index 0000000000..35e95a8c22
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/unittests/avformatmappertests.mm
@@ -0,0 +1,243 @@
+/*
+ *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+#import <OCMock/OCMock.h>
+
+#include "rtc_base/gunit.h"
+
+// Width and height don't play any role so lets use predefined values throughout
+// the tests.
+static const int kFormatWidth = 789;
+static const int kFormatHeight = 987;
+
+// Hardcoded framrate to be used throughout the tests.
+static const int kFramerate = 30;
+
+// Same width and height is used so it's ok to expect same cricket::VideoFormat
+static cricket::VideoFormat expectedFormat =
+    cricket::VideoFormat(kFormatWidth,
+                         kFormatHeight,
+                         cricket::VideoFormat::FpsToInterval(kFramerate),
+                         cricket::FOURCC_NV12);
+
+// Mock class for AVCaptureDeviceFormat.
+// Custom implementation needed because OCMock cannot handle the
+// CMVideoDescriptionRef mocking.
+@interface AVCaptureDeviceFormatMock : NSObject
+
+@property (nonatomic, assign) CMVideoFormatDescriptionRef format;
+@property (nonatomic, strong) OCMockObject *rangeMock;
+
+- (instancetype)initWithMediaSubtype:(FourCharCode)subtype
+                              minFps:(float)minFps
+                              maxFps:(float)maxFps;
++ (instancetype)validFormat;
++ (instancetype)invalidFpsFormat;
++ (instancetype)invalidMediaSubtypeFormat;
+
+@end
+
+@implementation AVCaptureDeviceFormatMock
+
+@synthesize format = _format;
+@synthesize rangeMock = _rangeMock;
+
+- (instancetype)initWithMediaSubtype:(FourCharCode)subtype
+                              minFps:(float)minFps
+                              maxFps:(float)maxFps {
+  if (self = [super init]) {
+    CMVideoFormatDescriptionCreate(nil, subtype, kFormatWidth, kFormatHeight,
+                                   nil, &_format);
+    // We can use OCMock for the range.
+    _rangeMock = [OCMockObject mockForClass:[AVFrameRateRange class]];
+    [[[_rangeMock stub] andReturnValue:@(minFps)] minFrameRate];
+    [[[_rangeMock stub] andReturnValue:@(maxFps)] maxFrameRate];
+  }
+
+  return self;
+}
+
++ (instancetype)validFormat {
+  AVCaptureDeviceFormatMock *instance = [[AVCaptureDeviceFormatMock alloc]
+      initWithMediaSubtype:kCVPixelFormatType_420YpCbCr8BiPlanarVideoRange
+                    minFps:0.0
+                    maxFps:30.0];
+  return instance;
+}
+
++ (instancetype)invalidFpsFormat {
+  AVCaptureDeviceFormatMock *instance = [[AVCaptureDeviceFormatMock alloc]
+      initWithMediaSubtype:kCVPixelFormatType_420YpCbCr8BiPlanarVideoRange
+                    minFps:0.0
+                    maxFps:22.0];
+  return instance;
+}
+
++ (instancetype)invalidMediaSubtypeFormat {
+  AVCaptureDeviceFormatMock *instance = [[AVCaptureDeviceFormatMock alloc]
+      initWithMediaSubtype:kCVPixelFormatType_420YpCbCr8Planar
+                    minFps:0.0
+                    maxFps:60.0];
+  return instance;
+}
+
+- (void)dealloc {
+  if (_format != nil) {
+    CFRelease(_format);
+    _format = nil;
+  }
+}
+
+// Redefinition of AVCaptureDevice methods we want to mock.
+- (CMVideoFormatDescriptionRef)formatDescription {
+  return self.format;
+}
+
+- (NSArray *)videoSupportedFrameRateRanges {
+  return @[ self.rangeMock ];
+}
+
+@end
+
+TEST(AVFormatMapperTest, SuportedCricketFormatsWithInvalidFramerateFormats) {
+  // given
+  id mockDevice = OCMClassMock([AVCaptureDevice class]);
+
+  // Valid media subtype, invalid framerate
+  AVCaptureDeviceFormatMock* mock =
+      [AVCaptureDeviceFormatMock invalidFpsFormat];
+  OCMStub([mockDevice formats]).andReturn(@[ mock ]);
+
+  // when
+  std::set<cricket::VideoFormat> result =
+      webrtc::GetSupportedVideoFormatsForDevice(mockDevice);
+
+  // then
+  EXPECT_TRUE(result.empty());
+}
+
+TEST(AVFormatMapperTest, SuportedCricketFormatsWithInvalidFormats) {
+  // given
+  id mockDevice = OCMClassMock([AVCaptureDevice class]);
+
+  // Invalid media subtype, valid framerate
+  AVCaptureDeviceFormatMock* mock =
+      [AVCaptureDeviceFormatMock invalidMediaSubtypeFormat];
+  OCMStub([mockDevice formats]).andReturn(@[ mock ]);
+
+  // when
+  std::set<cricket::VideoFormat> result =
+      webrtc::GetSupportedVideoFormatsForDevice(mockDevice);
+
+  // then
+  EXPECT_TRUE(result.empty());
+}
+
+TEST(AVFormatMapperTest, SuportedCricketFormats) {
+  // given
+  id mockDevice = OCMClassMock([AVCaptureDevice class]);
+
+  // valid media subtype, valid framerate
+  AVCaptureDeviceFormatMock* mock = [AVCaptureDeviceFormatMock validFormat];
+  OCMStub([mockDevice formats]).andReturn(@[ mock ]);
+
+  // when
+  std::set<cricket::VideoFormat> result =
+      webrtc::GetSupportedVideoFormatsForDevice(mockDevice);
+
+  // then
+  EXPECT_EQ(1u, result.size());
+  // make sure the set has the expected format
+  EXPECT_EQ(expectedFormat, *result.begin());
+}
+
+TEST(AVFormatMapperTest, MediaSubtypePreference) {
+  // given
+  id mockDevice = OCMClassMock([AVCaptureDevice class]);
+
+  // valid media subtype, valid framerate
+  AVCaptureDeviceFormatMock* mockOne = [[AVCaptureDeviceFormatMock alloc]
+      initWithMediaSubtype:kCVPixelFormatType_420YpCbCr8BiPlanarFullRange
+                    minFps:0.0
+                    maxFps:30.0];
+  // valid media subtype, valid framerate.
+  // This media subtype should be the preffered one.
+  AVCaptureDeviceFormatMock* mockTwo = [[AVCaptureDeviceFormatMock alloc]
+      initWithMediaSubtype:kCVPixelFormatType_420YpCbCr8BiPlanarVideoRange
+                    minFps:0.0
+                    maxFps:30.0];
+  OCMStub([mockDevice lockForConfiguration:[OCMArg setTo:nil]]).andReturn(YES);
+  OCMStub([mockDevice unlockForConfiguration]);
+  NSArray* array = @[ mockOne, mockTwo ];
+  OCMStub([mockDevice formats]).andReturn(array);
+
+  // to verify
+  OCMExpect([mockDevice setActiveFormat:(AVCaptureDeviceFormat*)mockTwo]);
+  OCMExpect(
+      [mockDevice setActiveVideoMinFrameDuration:CMTimeMake(1, kFramerate)]);
+
+  // when
+  bool resultFormat =
+      webrtc::SetFormatForCaptureDevice(mockDevice, nil, expectedFormat);
+
+  // then
+  EXPECT_TRUE(resultFormat);
+  [mockDevice verify];
+}
+
+TEST(AVFormatMapperTest, SetFormatWhenDeviceCannotLock) {
+  // given
+  id mockDevice = OCMClassMock([AVCaptureDevice class]);
+  [[[mockDevice stub] andReturnValue:@(NO)]
+      lockForConfiguration:[OCMArg setTo:nil]];
+  [[[mockDevice stub] andReturn:@[]] formats];
+
+  // when
+  bool resultFormat = webrtc::SetFormatForCaptureDevice(mockDevice, nil,
+                                                        cricket::VideoFormat());
+
+  // then
+  EXPECT_FALSE(resultFormat);
+}
+
+TEST(AVFormatMapperTest, SetFormatWhenFormatIsIncompatible) {
+  // given
+  id mockDevice = OCMClassMock([AVCaptureDevice class]);
+  OCMStub([mockDevice formats]).andReturn(@[]);
+  OCMStub([mockDevice lockForConfiguration:[OCMArg setTo:nil]]).andReturn(YES);
+  NSException* testException =
+      [NSException exceptionWithName:@"Test exception"
+                              reason:@"Raised from unit tests"
+                            userInfo:nil];
+  OCMStub([mockDevice setActiveFormat:[OCMArg any]]).andThrow(testException);
+  OCMExpect([mockDevice unlockForConfiguration]);
+
+  // when
+  bool resultFormat = webrtc::SetFormatForCaptureDevice(mockDevice, nil,
+                                                        cricket::VideoFormat());
+
+  // then
+  EXPECT_FALSE(resultFormat);
+
+  // TODO(denicija): Remove try-catch when Chromium rolls this change:
+  // https://github.com/erikdoe/ocmock/commit/de1419415581dc307045e54bfe9c98c86efea96b
+  // Without it, stubbed exceptions are being re-raised on [mock verify].
+  // More information here:
+  //https://github.com/erikdoe/ocmock/issues/241
+  @try {
+    [mockDevice verify];
+  } @catch (NSException* exception) {
+    if ([exception.reason isEqual:testException.reason]) {
+      // Nothing dangerous here
+      EXPECT_TRUE([exception.reason isEqualToString:exception.reason]);
+    }
+  }
+}
diff --git a/third_party/libwebrtc/sdk/objc/unittests/foreman.mp4 b/third_party/libwebrtc/sdk/objc/unittests/foreman.mp4
new file mode 100644
index 0000000000..ccffbf4722
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/unittests/foreman.mp4
diff --git a/third_party/libwebrtc/sdk/objc/unittests/frame_buffer_helpers.h b/third_party/libwebrtc/sdk/objc/unittests/frame_buffer_helpers.h
new file mode 100644
index 0000000000..76c0d15c7e
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/unittests/frame_buffer_helpers.h
@@ -0,0 +1,22 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <AVFoundation/AVFoundation.h>
+
+#include "api/video/i420_buffer.h"
+
+void DrawGradientInRGBPixelBuffer(CVPixelBufferRef pixelBuffer);
+
+rtc::scoped_refptr<webrtc::I420Buffer> CreateI420Gradient(int width,
+                                                          int height);
+
+void CopyI420BufferToCVPixelBuffer(
+    rtc::scoped_refptr<webrtc::I420Buffer> i420Buffer,
+    CVPixelBufferRef pixelBuffer);
diff --git a/third_party/libwebrtc/sdk/objc/unittests/frame_buffer_helpers.mm b/third_party/libwebrtc/sdk/objc/unittests/frame_buffer_helpers.mm
new file mode 100644
index 0000000000..98b86c54c0
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/unittests/frame_buffer_helpers.mm
@@ -0,0 +1,126 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "sdk/objc/unittests/frame_buffer_helpers.h"
+
+#include "third_party/libyuv/include/libyuv.h"
+
+void DrawGradientInRGBPixelBuffer(CVPixelBufferRef pixelBuffer) {
+  CVPixelBufferLockBaseAddress(pixelBuffer, kCVPixelBufferLock_ReadOnly);
+  void* baseAddr = CVPixelBufferGetBaseAddress(pixelBuffer);
+  size_t width = CVPixelBufferGetWidth(pixelBuffer);
+  size_t height = CVPixelBufferGetHeight(pixelBuffer);
+  CGColorSpaceRef colorSpace = CGColorSpaceCreateDeviceRGB();
+  int byteOrder = CVPixelBufferGetPixelFormatType(pixelBuffer) == kCVPixelFormatType_32ARGB ?
+      kCGBitmapByteOrder32Little :
+      0;
+  CGContextRef cgContext = CGBitmapContextCreate(baseAddr,
+                                                 width,
+                                                 height,
+                                                 8,
+                                                 CVPixelBufferGetBytesPerRow(pixelBuffer),
+                                                 colorSpace,
+                                                 byteOrder | kCGImageAlphaNoneSkipLast);
+
+  // Create a gradient
+  CGFloat colors[] = {
+      1.0, 1.0, 1.0, 1.0, 1.0, 0.0, 0.0, 1.0, 0.0, 1.0, 0.0, 1.0, 0.0, 0.0, 1.0, 1.0,
+  };
+  CGGradientRef gradient = CGGradientCreateWithColorComponents(colorSpace, colors, NULL, 4);
+
+  CGContextDrawLinearGradient(
+      cgContext, gradient, CGPointMake(0, 0), CGPointMake(width, height), 0);
+  CGGradientRelease(gradient);
+
+  CGImageRef cgImage = CGBitmapContextCreateImage(cgContext);
+  CGContextRelease(cgContext);
+  CGImageRelease(cgImage);
+  CGColorSpaceRelease(colorSpace);
+
+  CVPixelBufferUnlockBaseAddress(pixelBuffer, kCVPixelBufferLock_ReadOnly);
+}
+
+rtc::scoped_refptr<webrtc::I420Buffer> CreateI420Gradient(int width, int height) {
+  rtc::scoped_refptr<webrtc::I420Buffer> buffer(webrtc::I420Buffer::Create(width, height));
+  // Initialize with gradient, Y = 128(x/w + y/h), U = 256 x/w, V = 256 y/h
+  for (int x = 0; x < width; x++) {
+    for (int y = 0; y < height; y++) {
+      buffer->MutableDataY()[x + y * width] = 128 * (x * height + y * width) / (width * height);
+    }
+  }
+  int chroma_width = buffer->ChromaWidth();
+  int chroma_height = buffer->ChromaHeight();
+  for (int x = 0; x < chroma_width; x++) {
+    for (int y = 0; y < chroma_height; y++) {
+      buffer->MutableDataU()[x + y * chroma_width] = 255 * x / (chroma_width - 1);
+      buffer->MutableDataV()[x + y * chroma_width] = 255 * y / (chroma_height - 1);
+    }
+  }
+  return buffer;
+}
+
+void CopyI420BufferToCVPixelBuffer(rtc::scoped_refptr<webrtc::I420Buffer> i420Buffer,
+                                   CVPixelBufferRef pixelBuffer) {
+  CVPixelBufferLockBaseAddress(pixelBuffer, 0);
+
+  const OSType pixelFormat = CVPixelBufferGetPixelFormatType(pixelBuffer);
+  if (pixelFormat == kCVPixelFormatType_420YpCbCr8BiPlanarVideoRange ||
+      pixelFormat == kCVPixelFormatType_420YpCbCr8BiPlanarFullRange) {
+    // NV12
+    uint8_t* dstY = static_cast<uint8_t*>(CVPixelBufferGetBaseAddressOfPlane(pixelBuffer, 0));
+    const int dstYStride = CVPixelBufferGetBytesPerRowOfPlane(pixelBuffer, 0);
+    uint8_t* dstUV = static_cast<uint8_t*>(CVPixelBufferGetBaseAddressOfPlane(pixelBuffer, 1));
+    const int dstUVStride = CVPixelBufferGetBytesPerRowOfPlane(pixelBuffer, 1);
+
+    libyuv::I420ToNV12(i420Buffer->DataY(),
+                       i420Buffer->StrideY(),
+                       i420Buffer->DataU(),
+                       i420Buffer->StrideU(),
+                       i420Buffer->DataV(),
+                       i420Buffer->StrideV(),
+                       dstY,
+                       dstYStride,
+                       dstUV,
+                       dstUVStride,
+                       i420Buffer->width(),
+                       i420Buffer->height());
+  } else {
+    uint8_t* dst = static_cast<uint8_t*>(CVPixelBufferGetBaseAddress(pixelBuffer));
+    const int bytesPerRow = CVPixelBufferGetBytesPerRow(pixelBuffer);
+
+    if (pixelFormat == kCVPixelFormatType_32BGRA) {
+      // Corresponds to libyuv::FOURCC_ARGB
+      libyuv::I420ToARGB(i420Buffer->DataY(),
+                         i420Buffer->StrideY(),
+                         i420Buffer->DataU(),
+                         i420Buffer->StrideU(),
+                         i420Buffer->DataV(),
+                         i420Buffer->StrideV(),
+                         dst,
+                         bytesPerRow,
+                         i420Buffer->width(),
+                         i420Buffer->height());
+    } else if (pixelFormat == kCVPixelFormatType_32ARGB) {
+      // Corresponds to libyuv::FOURCC_BGRA
+      libyuv::I420ToBGRA(i420Buffer->DataY(),
+                         i420Buffer->StrideY(),
+                         i420Buffer->DataU(),
+                         i420Buffer->StrideU(),
+                         i420Buffer->DataV(),
+                         i420Buffer->StrideV(),
+                         dst,
+                         bytesPerRow,
+                         i420Buffer->width(),
+                         i420Buffer->height());
+    }
+  }
+
+  CVPixelBufferUnlockBaseAddress(pixelBuffer, 0);
+}
diff --git a/third_party/libwebrtc/sdk/objc/unittests/main.mm b/third_party/libwebrtc/sdk/objc/unittests/main.mm
new file mode 100644
index 0000000000..9c513762c1
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/unittests/main.mm
@@ -0,0 +1,24 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+#import <UIKit/UIKit.h>
+#include "rtc_base/thread.h"
+#include "test/ios/coverage_util_ios.h"
+
+int main(int argc, char* argv[]) {
+  rtc::test::ConfigureCoverageReportPath();
+
+  rtc::AutoThread main_thread;
+
+  @autoreleasepool {
+    return UIApplicationMain(argc, argv, nil, nil);
+  }
+}
diff --git a/third_party/libwebrtc/sdk/objc/unittests/nalu_rewriter_xctest.mm b/third_party/libwebrtc/sdk/objc/unittests/nalu_rewriter_xctest.mm
new file mode 100644
index 0000000000..82da549bb6
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/unittests/nalu_rewriter_xctest.mm
@@ -0,0 +1,374 @@
+/*
+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "common_video/h264/h264_common.h"
+#include "components/video_codec/nalu_rewriter.h"
+#include "rtc_base/arraysize.h"
+#include "rtc_base/gunit.h"
+
+#import <XCTest/XCTest.h>
+
+#if TARGET_OS_IPHONE
+#import <AVFoundation/AVFoundation.h>
+#import <UIKit/UIKit.h>
+#endif
+
+@interface NaluRewriterTests : XCTestCase
+
+@end
+
+static const uint8_t NALU_TEST_DATA_0[] = {0xAA, 0xBB, 0xCC};
+static const uint8_t NALU_TEST_DATA_1[] = {0xDE, 0xAD, 0xBE, 0xEF};
+
+// clang-format off
+static const uint8_t SPS_PPS_BUFFER[] = {
+  // SPS nalu.
+  0x00, 0x00, 0x00, 0x01, 0x27, 0x42, 0x00, 0x1E, 0xAB, 0x40, 0xF0, 0x28,
+  0xD3, 0x70, 0x20, 0x20, 0x20, 0x20,
+  // PPS nalu.
+  0x00, 0x00, 0x00, 0x01, 0x28, 0xCE, 0x3C, 0x30};
+// clang-format on
+
+@implementation NaluRewriterTests
+
+- (void)testCreateVideoFormatDescription {
+  CMVideoFormatDescriptionRef description =
+      webrtc::CreateVideoFormatDescription(SPS_PPS_BUFFER, arraysize(SPS_PPS_BUFFER));
+  XCTAssertTrue(description);
+  if (description) {
+    CFRelease(description);
+    description = nullptr;
+  }
+
+  // clang-format off
+  const uint8_t sps_pps_not_at_start_buffer[] = {
+    // Add some non-SPS/PPS NALUs at the beginning
+    0x00, 0x00, 0x00, 0x01, 0x00, 0x00, 0x01, 0xFF, 0x00, 0x00, 0x00, 0x01,
+    0xAB, 0x33, 0x21,
+    // SPS nalu.
+    0x00, 0x00, 0x01, 0x27, 0x42, 0x00, 0x1E, 0xAB, 0x40, 0xF0, 0x28, 0xD3,
+    0x70, 0x20, 0x20, 0x20, 0x20,
+    // PPS nalu.
+    0x00, 0x00, 0x01, 0x28, 0xCE, 0x3C, 0x30};
+  // clang-format on
+  description = webrtc::CreateVideoFormatDescription(sps_pps_not_at_start_buffer,
+                                                     arraysize(sps_pps_not_at_start_buffer));
+
+  XCTAssertTrue(description);
+
+  if (description) {
+    CFRelease(description);
+    description = nullptr;
+  }
+
+  const uint8_t other_buffer[] = {0x00, 0x00, 0x00, 0x01, 0x28};
+  XCTAssertFalse(webrtc::CreateVideoFormatDescription(other_buffer, arraysize(other_buffer)));
+}
+
+- (void)testReadEmptyInput {
+  const uint8_t annex_b_test_data[] = {0x00};
+  webrtc::AnnexBBufferReader reader(annex_b_test_data, 0);
+  const uint8_t* nalu = nullptr;
+  size_t nalu_length = 0;
+  XCTAssertEqual(0u, reader.BytesRemaining());
+  XCTAssertFalse(reader.ReadNalu(&nalu, &nalu_length));
+  XCTAssertEqual(nullptr, nalu);
+  XCTAssertEqual(0u, nalu_length);
+}
+
+- (void)testReadSingleNalu {
+  const uint8_t annex_b_test_data[] = {0x00, 0x00, 0x00, 0x01, 0xAA};
+  webrtc::AnnexBBufferReader reader(annex_b_test_data, arraysize(annex_b_test_data));
+  const uint8_t* nalu = nullptr;
+  size_t nalu_length = 0;
+  XCTAssertEqual(arraysize(annex_b_test_data), reader.BytesRemaining());
+  XCTAssertTrue(reader.ReadNalu(&nalu, &nalu_length));
+  XCTAssertEqual(annex_b_test_data + 4, nalu);
+  XCTAssertEqual(1u, nalu_length);
+  XCTAssertEqual(0u, reader.BytesRemaining());
+  XCTAssertFalse(reader.ReadNalu(&nalu, &nalu_length));
+  XCTAssertEqual(nullptr, nalu);
+  XCTAssertEqual(0u, nalu_length);
+}
+
+- (void)testReadSingleNalu3ByteHeader {
+  const uint8_t annex_b_test_data[] = {0x00, 0x00, 0x01, 0xAA};
+  webrtc::AnnexBBufferReader reader(annex_b_test_data, arraysize(annex_b_test_data));
+  const uint8_t* nalu = nullptr;
+  size_t nalu_length = 0;
+  XCTAssertEqual(arraysize(annex_b_test_data), reader.BytesRemaining());
+  XCTAssertTrue(reader.ReadNalu(&nalu, &nalu_length));
+  XCTAssertEqual(annex_b_test_data + 3, nalu);
+  XCTAssertEqual(1u, nalu_length);
+  XCTAssertEqual(0u, reader.BytesRemaining());
+  XCTAssertFalse(reader.ReadNalu(&nalu, &nalu_length));
+  XCTAssertEqual(nullptr, nalu);
+  XCTAssertEqual(0u, nalu_length);
+}
+
+- (void)testReadMissingNalu {
+  // clang-format off
+  const uint8_t annex_b_test_data[] = {0x01,
+                                       0x00, 0x01,
+                                       0x00, 0x00, 0x00, 0xFF};
+  // clang-format on
+  webrtc::AnnexBBufferReader reader(annex_b_test_data, arraysize(annex_b_test_data));
+  const uint8_t* nalu = nullptr;
+  size_t nalu_length = 0;
+  XCTAssertEqual(0u, reader.BytesRemaining());
+  XCTAssertFalse(reader.ReadNalu(&nalu, &nalu_length));
+  XCTAssertEqual(nullptr, nalu);
+  XCTAssertEqual(0u, nalu_length);
+}
+
+- (void)testReadMultipleNalus {
+  // clang-format off
+  const uint8_t annex_b_test_data[] = {0x00, 0x00, 0x00, 0x01, 0xFF,
+                                       0x01,
+                                       0x00, 0x01,
+                                       0x00, 0x00, 0x00, 0xFF,
+                                       0x00, 0x00, 0x01, 0xAA, 0xBB};
+  // clang-format on
+  webrtc::AnnexBBufferReader reader(annex_b_test_data, arraysize(annex_b_test_data));
+  const uint8_t* nalu = nullptr;
+  size_t nalu_length = 0;
+  XCTAssertEqual(arraysize(annex_b_test_data), reader.BytesRemaining());
+  XCTAssertTrue(reader.ReadNalu(&nalu, &nalu_length));
+  XCTAssertEqual(annex_b_test_data + 4, nalu);
+  XCTAssertEqual(8u, nalu_length);
+  XCTAssertEqual(5u, reader.BytesRemaining());
+  XCTAssertTrue(reader.ReadNalu(&nalu, &nalu_length));
+  XCTAssertEqual(annex_b_test_data + 15, nalu);
+  XCTAssertEqual(2u, nalu_length);
+  XCTAssertEqual(0u, reader.BytesRemaining());
+  XCTAssertFalse(reader.ReadNalu(&nalu, &nalu_length));
+  XCTAssertEqual(nullptr, nalu);
+  XCTAssertEqual(0u, nalu_length);
+}
+
+- (void)testEmptyOutputBuffer {
+  const uint8_t expected_buffer[] = {0x00};
+  const size_t buffer_size = 1;
+  std::unique_ptr<uint8_t[]> buffer(new uint8_t[buffer_size]);
+  memset(buffer.get(), 0, buffer_size);
+  webrtc::AvccBufferWriter writer(buffer.get(), 0);
+  XCTAssertEqual(0u, writer.BytesRemaining());
+  XCTAssertFalse(writer.WriteNalu(NALU_TEST_DATA_0, arraysize(NALU_TEST_DATA_0)));
+  XCTAssertEqual(0, memcmp(expected_buffer, buffer.get(), arraysize(expected_buffer)));
+}
+
+- (void)testWriteSingleNalu {
+  const uint8_t expected_buffer[] = {
+      0x00, 0x00, 0x00, 0x03, 0xAA, 0xBB, 0xCC,
+  };
+  const size_t buffer_size = arraysize(NALU_TEST_DATA_0) + 4;
+  std::unique_ptr<uint8_t[]> buffer(new uint8_t[buffer_size]);
+  webrtc::AvccBufferWriter writer(buffer.get(), buffer_size);
+  XCTAssertEqual(buffer_size, writer.BytesRemaining());
+  XCTAssertTrue(writer.WriteNalu(NALU_TEST_DATA_0, arraysize(NALU_TEST_DATA_0)));
+  XCTAssertEqual(0u, writer.BytesRemaining());
+  XCTAssertFalse(writer.WriteNalu(NALU_TEST_DATA_1, arraysize(NALU_TEST_DATA_1)));
+  XCTAssertEqual(0, memcmp(expected_buffer, buffer.get(), arraysize(expected_buffer)));
+}
+
+- (void)testWriteMultipleNalus {
+  // clang-format off
+  const uint8_t expected_buffer[] = {
+    0x00, 0x00, 0x00, 0x03, 0xAA, 0xBB, 0xCC,
+    0x00, 0x00, 0x00, 0x04, 0xDE, 0xAD, 0xBE, 0xEF
+  };
+  // clang-format on
+  const size_t buffer_size = arraysize(NALU_TEST_DATA_0) + arraysize(NALU_TEST_DATA_1) + 8;
+  std::unique_ptr<uint8_t[]> buffer(new uint8_t[buffer_size]);
+  webrtc::AvccBufferWriter writer(buffer.get(), buffer_size);
+  XCTAssertEqual(buffer_size, writer.BytesRemaining());
+  XCTAssertTrue(writer.WriteNalu(NALU_TEST_DATA_0, arraysize(NALU_TEST_DATA_0)));
+  XCTAssertEqual(buffer_size - (arraysize(NALU_TEST_DATA_0) + 4), writer.BytesRemaining());
+  XCTAssertTrue(writer.WriteNalu(NALU_TEST_DATA_1, arraysize(NALU_TEST_DATA_1)));
+  XCTAssertEqual(0u, writer.BytesRemaining());
+  XCTAssertEqual(0, memcmp(expected_buffer, buffer.get(), arraysize(expected_buffer)));
+}
+
+- (void)testOverflow {
+  const uint8_t expected_buffer[] = {0x00, 0x00, 0x00};
+  const size_t buffer_size = arraysize(NALU_TEST_DATA_0);
+  std::unique_ptr<uint8_t[]> buffer(new uint8_t[buffer_size]);
+  memset(buffer.get(), 0, buffer_size);
+  webrtc::AvccBufferWriter writer(buffer.get(), buffer_size);
+  XCTAssertEqual(buffer_size, writer.BytesRemaining());
+  XCTAssertFalse(writer.WriteNalu(NALU_TEST_DATA_0, arraysize(NALU_TEST_DATA_0)));
+  XCTAssertEqual(buffer_size, writer.BytesRemaining());
+  XCTAssertEqual(0, memcmp(expected_buffer, buffer.get(), arraysize(expected_buffer)));
+}
+
+- (void)testH264AnnexBBufferToCMSampleBuffer {
+  // clang-format off
+  const uint8_t annex_b_test_data[] = {
+    0x00,
+    0x00, 0x00, 0x01,
+    0x01, 0x00, 0x00, 0xFF, // first chunk, 4 bytes
+    0x00, 0x00, 0x01,
+    0xAA, 0xFF, // second chunk, 2 bytes
+    0x00, 0x00, 0x01,
+    0xBB};  // third chunk, 1 byte, will not fit into output array
+
+  const uint8_t expected_cmsample_data[] = {
+    0x00, 0x00, 0x00, 0x04,
+    0x01, 0x00, 0x00, 0xFF, // first chunk, 4 bytes
+    0x00, 0x00, 0x00, 0x02,
+    0xAA, 0xFF}; // second chunk, 2 bytes
+  // clang-format on
+
+  CMMemoryPoolRef memory_pool = CMMemoryPoolCreate(nil);
+  CMSampleBufferRef out_sample_buffer = nil;
+  CMVideoFormatDescriptionRef description = [self createDescription];
+
+  Boolean result = webrtc::H264AnnexBBufferToCMSampleBuffer(annex_b_test_data,
+                                                            arraysize(annex_b_test_data),
+                                                            description,
+                                                            &out_sample_buffer,
+                                                            memory_pool);
+
+  XCTAssertTrue(result);
+
+  XCTAssertEqual(description, CMSampleBufferGetFormatDescription(out_sample_buffer));
+
+  char* data_ptr = nullptr;
+  CMBlockBufferRef block_buffer = CMSampleBufferGetDataBuffer(out_sample_buffer);
+  size_t block_buffer_size = CMBlockBufferGetDataLength(block_buffer);
+  CMBlockBufferGetDataPointer(block_buffer, 0, nullptr, nullptr, &data_ptr);
+  XCTAssertEqual(block_buffer_size, arraysize(annex_b_test_data));
+
+  int data_comparison_result =
+      memcmp(expected_cmsample_data, data_ptr, arraysize(expected_cmsample_data));
+
+  XCTAssertEqual(0, data_comparison_result);
+
+  if (description) {
+    CFRelease(description);
+    description = nullptr;
+  }
+
+  CMMemoryPoolInvalidate(memory_pool);
+  CFRelease(memory_pool);
+}
+
+- (void)testH264CMSampleBufferToAnnexBBuffer {
+  // clang-format off
+  const uint8_t cmsample_data[] = {
+    0x00, 0x00, 0x00, 0x04,
+    0x01, 0x00, 0x00, 0xFF, // first chunk, 4 bytes
+    0x00, 0x00, 0x00, 0x02,
+    0xAA, 0xFF}; // second chunk, 2 bytes
+
+  const uint8_t expected_annex_b_data[] = {
+    0x00, 0x00, 0x00, 0x01,
+    0x01, 0x00, 0x00, 0xFF, // first chunk, 4 bytes
+    0x00, 0x00, 0x00, 0x01,
+    0xAA, 0xFF}; // second chunk, 2 bytes
+  // clang-format on
+
+  rtc::Buffer annexb_buffer(arraysize(cmsample_data));
+  CMSampleBufferRef sample_buffer =
+      [self createCMSampleBufferRef:(void*)cmsample_data cmsampleSize:arraysize(cmsample_data)];
+
+  Boolean result = webrtc::H264CMSampleBufferToAnnexBBuffer(sample_buffer,
+                                                            /* is_keyframe */ false,
+                                                            &annexb_buffer);
+
+  XCTAssertTrue(result);
+
+  XCTAssertEqual(arraysize(expected_annex_b_data), annexb_buffer.size());
+
+  int data_comparison_result =
+      memcmp(expected_annex_b_data, annexb_buffer.data(), arraysize(expected_annex_b_data));
+
+  XCTAssertEqual(0, data_comparison_result);
+}
+
+- (void)testH264CMSampleBufferToAnnexBBufferWithKeyframe {
+  // clang-format off
+  const uint8_t cmsample_data[] = {
+    0x00, 0x00, 0x00, 0x04,
+    0x01, 0x00, 0x00, 0xFF, // first chunk, 4 bytes
+    0x00, 0x00, 0x00, 0x02,
+    0xAA, 0xFF}; // second chunk, 2 bytes
+
+  const uint8_t expected_annex_b_data[] = {
+    0x00, 0x00, 0x00, 0x01,
+    0x01, 0x00, 0x00, 0xFF, // first chunk, 4 bytes
+    0x00, 0x00, 0x00, 0x01,
+    0xAA, 0xFF}; // second chunk, 2 bytes
+  // clang-format on
+
+  rtc::Buffer annexb_buffer(arraysize(cmsample_data));
+  CMSampleBufferRef sample_buffer =
+      [self createCMSampleBufferRef:(void*)cmsample_data cmsampleSize:arraysize(cmsample_data)];
+
+  Boolean result = webrtc::H264CMSampleBufferToAnnexBBuffer(sample_buffer,
+                                                            /* is_keyframe */ true,
+                                                            &annexb_buffer);
+
+  XCTAssertTrue(result);
+
+  XCTAssertEqual(arraysize(SPS_PPS_BUFFER) + arraysize(expected_annex_b_data),
+                 annexb_buffer.size());
+
+  XCTAssertEqual(0, memcmp(SPS_PPS_BUFFER, annexb_buffer.data(), arraysize(SPS_PPS_BUFFER)));
+
+  XCTAssertEqual(0,
+                 memcmp(expected_annex_b_data,
+                        annexb_buffer.data() + arraysize(SPS_PPS_BUFFER),
+                        arraysize(expected_annex_b_data)));
+}
+
+- (CMVideoFormatDescriptionRef)createDescription {
+  CMVideoFormatDescriptionRef description =
+      webrtc::CreateVideoFormatDescription(SPS_PPS_BUFFER, arraysize(SPS_PPS_BUFFER));
+  XCTAssertTrue(description);
+  return description;
+}
+
+- (CMSampleBufferRef)createCMSampleBufferRef:(void*)cmsampleData cmsampleSize:(size_t)cmsampleSize {
+  CMSampleBufferRef sample_buffer = nil;
+  OSStatus status;
+
+  CMVideoFormatDescriptionRef description = [self createDescription];
+  CMBlockBufferRef block_buffer = nullptr;
+
+  status = CMBlockBufferCreateWithMemoryBlock(nullptr,
+                                              cmsampleData,
+                                              cmsampleSize,
+                                              nullptr,
+                                              nullptr,
+                                              0,
+                                              cmsampleSize,
+                                              kCMBlockBufferAssureMemoryNowFlag,
+                                              &block_buffer);
+  XCTAssertEqual(kCMBlockBufferNoErr, status);
+
+  status = CMSampleBufferCreate(nullptr,
+                                block_buffer,
+                                true,
+                                nullptr,
+                                nullptr,
+                                description,
+                                1,
+                                0,
+                                nullptr,
+                                0,
+                                nullptr,
+                                &sample_buffer);
+  XCTAssertEqual(noErr, status);
+
+  return sample_buffer;
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/unittests/objc_video_decoder_factory_tests.mm b/third_party/libwebrtc/sdk/objc/unittests/objc_video_decoder_factory_tests.mm
new file mode 100644
index 0000000000..f44d831d29
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/unittests/objc_video_decoder_factory_tests.mm
@@ -0,0 +1,107 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+#import <OCMock/OCMock.h>
+#import <XCTest/XCTest.h>
+
+#include "sdk/objc/native/src/objc_video_decoder_factory.h"
+
+#import "base/RTCMacros.h"
+#import "base/RTCVideoDecoder.h"
+#import "base/RTCVideoDecoderFactory.h"
+#include "media/base/codec.h"
+#include "modules/video_coding/include/video_codec_interface.h"
+#include "modules/video_coding/include/video_error_codes.h"
+#include "rtc_base/gunit.h"
+
+id<RTC_OBJC_TYPE(RTCVideoDecoderFactory)> CreateDecoderFactoryReturning(int return_code) {
+  id decoderMock = OCMProtocolMock(@protocol(RTC_OBJC_TYPE(RTCVideoDecoder)));
+  OCMStub([decoderMock startDecodeWithNumberOfCores:1]).andReturn(return_code);
+  OCMStub([decoderMock decode:[OCMArg any]
+                    missingFrames:NO
+                codecSpecificInfo:[OCMArg any]
+                     renderTimeMs:0])
+      .andReturn(return_code);
+  OCMStub([decoderMock releaseDecoder]).andReturn(return_code);
+
+  id decoderFactoryMock = OCMProtocolMock(@protocol(RTC_OBJC_TYPE(RTCVideoDecoderFactory)));
+  RTC_OBJC_TYPE(RTCVideoCodecInfo)* supported =
+      [[RTC_OBJC_TYPE(RTCVideoCodecInfo) alloc] initWithName:@"H264" parameters:nil];
+  OCMStub([decoderFactoryMock supportedCodecs]).andReturn(@[ supported ]);
+  OCMStub([decoderFactoryMock createDecoder:[OCMArg any]]).andReturn(decoderMock);
+  return decoderFactoryMock;
+}
+
+id<RTC_OBJC_TYPE(RTCVideoDecoderFactory)> CreateOKDecoderFactory() {
+  return CreateDecoderFactoryReturning(WEBRTC_VIDEO_CODEC_OK);
+}
+
+id<RTC_OBJC_TYPE(RTCVideoDecoderFactory)> CreateErrorDecoderFactory() {
+  return CreateDecoderFactoryReturning(WEBRTC_VIDEO_CODEC_ERROR);
+}
+
+std::unique_ptr<webrtc::VideoDecoder> GetObjCDecoder(
+    id<RTC_OBJC_TYPE(RTCVideoDecoderFactory)> factory) {
+  webrtc::ObjCVideoDecoderFactory decoder_factory(factory);
+  return decoder_factory.CreateVideoDecoder(webrtc::SdpVideoFormat(cricket::kH264CodecName));
+}
+
+#pragma mark -
+
+@interface ObjCVideoDecoderFactoryTests : XCTestCase
+@end
+
+@implementation ObjCVideoDecoderFactoryTests
+
+- (void)testConfigureReturnsTrueOnSuccess {
+  std::unique_ptr<webrtc::VideoDecoder> decoder = GetObjCDecoder(CreateOKDecoderFactory());
+
+  webrtc::VideoDecoder::Settings settings;
+  EXPECT_TRUE(decoder->Configure(settings));
+}
+
+- (void)testConfigureReturnsFalseOnFail {
+  std::unique_ptr<webrtc::VideoDecoder> decoder = GetObjCDecoder(CreateErrorDecoderFactory());
+
+  webrtc::VideoDecoder::Settings settings;
+  EXPECT_FALSE(decoder->Configure(settings));
+}
+
+- (void)testDecodeReturnsOKOnSuccess {
+  std::unique_ptr<webrtc::VideoDecoder> decoder = GetObjCDecoder(CreateOKDecoderFactory());
+
+  webrtc::EncodedImage encoded_image;
+  encoded_image.SetEncodedData(webrtc::EncodedImageBuffer::Create());
+
+  EXPECT_EQ(decoder->Decode(encoded_image, false, 0), WEBRTC_VIDEO_CODEC_OK);
+}
+
+- (void)testDecodeReturnsErrorOnFail {
+  std::unique_ptr<webrtc::VideoDecoder> decoder = GetObjCDecoder(CreateErrorDecoderFactory());
+
+  webrtc::EncodedImage encoded_image;
+  encoded_image.SetEncodedData(webrtc::EncodedImageBuffer::Create());
+
+  EXPECT_EQ(decoder->Decode(encoded_image, false, 0), WEBRTC_VIDEO_CODEC_ERROR);
+}
+
+- (void)testReleaseDecodeReturnsOKOnSuccess {
+  std::unique_ptr<webrtc::VideoDecoder> decoder = GetObjCDecoder(CreateOKDecoderFactory());
+
+  EXPECT_EQ(decoder->Release(), WEBRTC_VIDEO_CODEC_OK);
+}
+
+- (void)testReleaseDecodeReturnsErrorOnFail {
+  std::unique_ptr<webrtc::VideoDecoder> decoder = GetObjCDecoder(CreateErrorDecoderFactory());
+
+  EXPECT_EQ(decoder->Release(), WEBRTC_VIDEO_CODEC_ERROR);
+}
+@end
diff --git a/third_party/libwebrtc/sdk/objc/unittests/objc_video_encoder_factory_tests.mm b/third_party/libwebrtc/sdk/objc/unittests/objc_video_encoder_factory_tests.mm
new file mode 100644
index 0000000000..9a4fee2e95
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/unittests/objc_video_encoder_factory_tests.mm
@@ -0,0 +1,148 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <Foundation/Foundation.h>
+#import <OCMock/OCMock.h>
+#import <XCTest/XCTest.h>
+
+#include "sdk/objc/native/src/objc_video_encoder_factory.h"
+
+#include "api/video_codecs/sdp_video_format.h"
+#include "api/video_codecs/video_encoder.h"
+#import "base/RTCVideoEncoder.h"
+#import "base/RTCVideoEncoderFactory.h"
+#import "base/RTCVideoFrameBuffer.h"
+#import "components/video_frame_buffer/RTCCVPixelBuffer.h"
+#include "modules/video_coding/include/video_codec_interface.h"
+#include "modules/video_coding/include/video_error_codes.h"
+#include "rtc_base/gunit.h"
+#include "sdk/objc/native/src/objc_frame_buffer.h"
+
+id<RTC_OBJC_TYPE(RTCVideoEncoderFactory)> CreateEncoderFactoryReturning(int return_code) {
+  id encoderMock = OCMProtocolMock(@protocol(RTC_OBJC_TYPE(RTCVideoEncoder)));
+  OCMStub([encoderMock startEncodeWithSettings:[OCMArg any] numberOfCores:1])
+      .andReturn(return_code);
+  OCMStub([encoderMock encode:[OCMArg any] codecSpecificInfo:[OCMArg any] frameTypes:[OCMArg any]])
+      .andReturn(return_code);
+  OCMStub([encoderMock releaseEncoder]).andReturn(return_code);
+  OCMStub([encoderMock setBitrate:0 framerate:0]).andReturn(return_code);
+
+  id encoderFactoryMock = OCMProtocolMock(@protocol(RTC_OBJC_TYPE(RTCVideoEncoderFactory)));
+  RTC_OBJC_TYPE(RTCVideoCodecInfo)* supported =
+      [[RTC_OBJC_TYPE(RTCVideoCodecInfo) alloc] initWithName:@"H264" parameters:nil];
+  OCMStub([encoderFactoryMock supportedCodecs]).andReturn(@[ supported ]);
+  OCMStub([encoderFactoryMock implementations]).andReturn(@[ supported ]);
+  OCMStub([encoderFactoryMock createEncoder:[OCMArg any]]).andReturn(encoderMock);
+  return encoderFactoryMock;
+}
+
+id<RTC_OBJC_TYPE(RTCVideoEncoderFactory)> CreateOKEncoderFactory() {
+  return CreateEncoderFactoryReturning(WEBRTC_VIDEO_CODEC_OK);
+}
+
+id<RTC_OBJC_TYPE(RTCVideoEncoderFactory)> CreateErrorEncoderFactory() {
+  return CreateEncoderFactoryReturning(WEBRTC_VIDEO_CODEC_ERROR);
+}
+
+std::unique_ptr<webrtc::VideoEncoder> GetObjCEncoder(
+    id<RTC_OBJC_TYPE(RTCVideoEncoderFactory)> factory) {
+  webrtc::ObjCVideoEncoderFactory encoder_factory(factory);
+  webrtc::SdpVideoFormat format("H264");
+  return encoder_factory.CreateVideoEncoder(format);
+}
+
+#pragma mark -
+
+@interface ObjCVideoEncoderFactoryTests : XCTestCase
+@end
+
+@implementation ObjCVideoEncoderFactoryTests
+
+- (void)testInitEncodeReturnsOKOnSuccess {
+  std::unique_ptr<webrtc::VideoEncoder> encoder = GetObjCEncoder(CreateOKEncoderFactory());
+
+  auto* settings = new webrtc::VideoCodec();
+  const webrtc::VideoEncoder::Capabilities kCapabilities(false);
+  EXPECT_EQ(encoder->InitEncode(settings, webrtc::VideoEncoder::Settings(kCapabilities, 1, 0)),
+            WEBRTC_VIDEO_CODEC_OK);
+}
+
+- (void)testInitEncodeReturnsErrorOnFail {
+  std::unique_ptr<webrtc::VideoEncoder> encoder = GetObjCEncoder(CreateErrorEncoderFactory());
+
+  auto* settings = new webrtc::VideoCodec();
+  const webrtc::VideoEncoder::Capabilities kCapabilities(false);
+  EXPECT_EQ(encoder->InitEncode(settings, webrtc::VideoEncoder::Settings(kCapabilities, 1, 0)),
+            WEBRTC_VIDEO_CODEC_ERROR);
+}
+
+- (void)testEncodeReturnsOKOnSuccess {
+  std::unique_ptr<webrtc::VideoEncoder> encoder = GetObjCEncoder(CreateOKEncoderFactory());
+
+  CVPixelBufferRef pixel_buffer;
+  CVPixelBufferCreate(kCFAllocatorDefault, 640, 480, kCVPixelFormatType_32ARGB, nil, &pixel_buffer);
+  rtc::scoped_refptr<webrtc::VideoFrameBuffer> buffer =
+      rtc::make_ref_counted<webrtc::ObjCFrameBuffer>(
+          [[RTC_OBJC_TYPE(RTCCVPixelBuffer) alloc] initWithPixelBuffer:pixel_buffer]);
+  webrtc::VideoFrame frame = webrtc::VideoFrame::Builder()
+                                 .set_video_frame_buffer(buffer)
+                                 .set_rotation(webrtc::kVideoRotation_0)
+                                 .set_timestamp_us(0)
+                                 .build();
+  std::vector<webrtc::VideoFrameType> frame_types;
+
+  EXPECT_EQ(encoder->Encode(frame, &frame_types), WEBRTC_VIDEO_CODEC_OK);
+}
+
+- (void)testEncodeReturnsErrorOnFail {
+  std::unique_ptr<webrtc::VideoEncoder> encoder = GetObjCEncoder(CreateErrorEncoderFactory());
+
+  CVPixelBufferRef pixel_buffer;
+  CVPixelBufferCreate(kCFAllocatorDefault, 640, 480, kCVPixelFormatType_32ARGB, nil, &pixel_buffer);
+  rtc::scoped_refptr<webrtc::VideoFrameBuffer> buffer =
+      rtc::make_ref_counted<webrtc::ObjCFrameBuffer>(
+          [[RTC_OBJC_TYPE(RTCCVPixelBuffer) alloc] initWithPixelBuffer:pixel_buffer]);
+  webrtc::VideoFrame frame = webrtc::VideoFrame::Builder()
+                                 .set_video_frame_buffer(buffer)
+                                 .set_rotation(webrtc::kVideoRotation_0)
+                                 .set_timestamp_us(0)
+                                 .build();
+  std::vector<webrtc::VideoFrameType> frame_types;
+
+  EXPECT_EQ(encoder->Encode(frame, &frame_types), WEBRTC_VIDEO_CODEC_ERROR);
+}
+
+- (void)testReleaseEncodeReturnsOKOnSuccess {
+  std::unique_ptr<webrtc::VideoEncoder> encoder = GetObjCEncoder(CreateOKEncoderFactory());
+
+  EXPECT_EQ(encoder->Release(), WEBRTC_VIDEO_CODEC_OK);
+}
+
+- (void)testReleaseEncodeReturnsErrorOnFail {
+  std::unique_ptr<webrtc::VideoEncoder> encoder = GetObjCEncoder(CreateErrorEncoderFactory());
+
+  EXPECT_EQ(encoder->Release(), WEBRTC_VIDEO_CODEC_ERROR);
+}
+
+- (void)testGetSupportedFormats {
+  webrtc::ObjCVideoEncoderFactory encoder_factory(CreateOKEncoderFactory());
+  std::vector<webrtc::SdpVideoFormat> supportedFormats = encoder_factory.GetSupportedFormats();
+  EXPECT_EQ(supportedFormats.size(), 1u);
+  EXPECT_EQ(supportedFormats[0].name, "H264");
+}
+
+- (void)testGetImplementations {
+  webrtc::ObjCVideoEncoderFactory encoder_factory(CreateOKEncoderFactory());
+  std::vector<webrtc::SdpVideoFormat> supportedFormats = encoder_factory.GetImplementations();
+  EXPECT_EQ(supportedFormats.size(), 1u);
+  EXPECT_EQ(supportedFormats[0].name, "H264");
+}
+
+@end
diff --git a/third_party/libwebrtc/sdk/objc/unittests/scoped_cftyperef_tests.mm b/third_party/libwebrtc/sdk/objc/unittests/scoped_cftyperef_tests.mm
new file mode 100644
index 0000000000..a354410ede
--- /dev/null
+++ b/third_party/libwebrtc/sdk/objc/unittests/scoped_cftyperef_tests.mm
@@ -0,0 +1,111 @@
+/*
+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#import <XCTest/XCTest.h>
+
+#include "sdk/objc/helpers/scoped_cftyperef.h"
+
+namespace {
+struct TestType {
+  TestType() : has_value(true) {}
+  TestType(bool b) : has_value(b) {}
+  explicit operator bool() { return has_value; }
+  bool has_value;
+  int retain_count = 0;
+};
+
+typedef TestType* TestTypeRef;
+
+struct TestTypeTraits {
+  static TestTypeRef InvalidValue() { return TestTypeRef(false); }
+  static void Release(TestTypeRef t) { t->retain_count--; }
+  static TestTypeRef Retain(TestTypeRef t) {
+    t->retain_count++;
+    return t;
+  }
+};
+}  // namespace
+
+using ScopedTestType = rtc::internal::ScopedTypeRef<TestTypeRef, TestTypeTraits>;
+
+// In these tests we sometime introduce variables just to
+// observe side-effects. Ignore the compilers complaints.
+#pragma clang diagnostic push
+#pragma clang diagnostic ignored "-Wunused-variable"
+
+@interface ScopedTypeRefTests : XCTestCase
+@end
+
+@implementation ScopedTypeRefTests
+
+- (void)testShouldNotRetainByDefault {
+  TestType a;
+  ScopedTestType ref(&a);
+  XCTAssertEqual(0, a.retain_count);
+}
+
+- (void)testShouldRetainWithPolicy {
+  TestType a;
+  ScopedTestType ref(&a, rtc::RetainPolicy::RETAIN);
+  XCTAssertEqual(1, a.retain_count);
+}
+
+- (void)testShouldReleaseWhenLeavingScope {
+  TestType a;
+  XCTAssertEqual(0, a.retain_count);
+  {
+    ScopedTestType ref(&a, rtc::RetainPolicy::RETAIN);
+    XCTAssertEqual(1, a.retain_count);
+  }
+  XCTAssertEqual(0, a.retain_count);
+}
+
+- (void)testShouldBeCopyable {
+  TestType a;
+  XCTAssertEqual(0, a.retain_count);
+  {
+    ScopedTestType ref1(&a, rtc::RetainPolicy::RETAIN);
+    XCTAssertEqual(1, a.retain_count);
+    ScopedTestType ref2 = ref1;
+    XCTAssertEqual(2, a.retain_count);
+  }
+  XCTAssertEqual(0, a.retain_count);
+}
+
+- (void)testCanReleaseOwnership {
+  TestType a;
+  XCTAssertEqual(0, a.retain_count);
+  {
+    ScopedTestType ref(&a, rtc::RetainPolicy::RETAIN);
+    XCTAssertEqual(1, a.retain_count);
+    TestTypeRef b = ref.release();
+  }
+  XCTAssertEqual(1, a.retain_count);
+}
+
+- (void)testShouldBeTestableForTruthiness {
+  ScopedTestType ref;
+  XCTAssertFalse(ref);
+  TestType a;
+  ref = &a;
+  XCTAssertTrue(ref);
+  ref.release();
+  XCTAssertFalse(ref);
+}
+
+- (void)testShouldProvideAccessToWrappedType {
+  TestType a;
+  ScopedTestType ref(&a);
+  XCTAssertEqual(&(a.retain_count), &(ref->retain_count));
+}
+
+@end
+
+#pragma clang diagnostic pop
diff --git a/third_party/libwebrtc/sdk/videocapture_objc_gn/moz.build b/third_party/libwebrtc/sdk/videocapture_objc_gn/moz.build
new file mode 100644
index 0000000000..64a7dbcb63
--- /dev/null
+++ b/third_party/libwebrtc/sdk/videocapture_objc_gn/moz.build
@@ -0,0 +1,70 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+CMFLAGS += [
+    "-fobjc-arc"
+]
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MAC"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_POSIX"] = True
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+DEFINES["__STDC_CONSTANT_MACROS"] = True
+DEFINES["__STDC_FORMAT_MACROS"] = True
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/sdk/objc/",
+    "/third_party/libwebrtc/sdk/objc/base/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/sdk/objc/components/capturer/RTCCameraVideoCapturer.m",
+    "/third_party/libwebrtc/sdk/objc/components/capturer/RTCFileVideoCapturer.m"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+Library("videocapture_objc_gn")
diff --git a/third_party/libwebrtc/sdk/videoframebuffer_objc_gn/moz.build b/third_party/libwebrtc/sdk/videoframebuffer_objc_gn/moz.build
new file mode 100644
index 0000000000..dea03fa574
--- /dev/null
+++ b/third_party/libwebrtc/sdk/videoframebuffer_objc_gn/moz.build
@@ -0,0 +1,73 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+  ### This moz.build was AUTOMATICALLY GENERATED from a GN config,  ###
+  ### DO NOT edit it by hand.                                       ###
+
+CMMFLAGS += [
+    "-fobjc-arc"
+]
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MAC"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_POSIX"] = True
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+DEFINES["__STDC_CONSTANT_MACROS"] = True
+DEFINES["__STDC_FORMAT_MACROS"] = True
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/media/libyuv/",
+    "/media/libyuv/libyuv/include/",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/sdk/objc/",
+    "/third_party/libwebrtc/sdk/objc/base/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/sdk/objc/api/video_frame_buffer/RTCNativeI420Buffer.mm",
+    "/third_party/libwebrtc/sdk/objc/api/video_frame_buffer/RTCNativeMutableI420Buffer.mm",
+    "/third_party/libwebrtc/sdk/objc/components/video_frame_buffer/RTCCVPixelBuffer.mm"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+    DEFINES["NDEBUG"] = True
+    DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+    DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+Library("videoframebuffer_objc_gn")