Adding upstream version 124.0.1.upstream/124.0.1

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

7 files changed, 800 insertions, 0 deletions

diff --git a/third_party/libwebrtc/video/render/BUILD.gn b/third_party/libwebrtc/video/render/BUILD.gn
new file mode 100644
index 0000000000..ff721dc61c
--- /dev/null
+++ b/third_party/libwebrtc/video/render/BUILD.gn
@@ -0,0 +1,51 @@
+# 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("../../webrtc.gni")
+
+rtc_library("incoming_video_stream") {
+  visibility = [ "*" ]
+
+  sources = [
+    "incoming_video_stream.cc",
+    "incoming_video_stream.h",
+  ]
+
+  deps = [
+    ":video_render_frames",
+    "../../api:sequence_checker",
+    "../../api/task_queue:task_queue",
+    "../../api/units:time_delta",
+    "../../api/video:video_frame",
+    "../../rtc_base:checks",
+    "../../rtc_base:event_tracer",
+    "../../rtc_base:macromagic",
+    "../../rtc_base:race_checker",
+    "../../rtc_base:rtc_task_queue",
+  ]
+
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
+}
+
+rtc_library("video_render_frames") {
+  visibility = [ ":*" ]  # Private.
+
+  sources = [
+    "video_render_frames.cc",
+    "video_render_frames.h",
+  ]
+
+  deps = [
+    "../../api/video:video_frame",
+    "../../rtc_base:checks",
+    "../../rtc_base:logging",
+    "../../rtc_base:timeutils",
+    "../../system_wrappers:metrics",
+  ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
+}
diff --git a/third_party/libwebrtc/video/render/incoming_video_stream.cc b/third_party/libwebrtc/video/render/incoming_video_stream.cc
new file mode 100644
index 0000000000..e740c47bd0
--- /dev/null
+++ b/third_party/libwebrtc/video/render/incoming_video_stream.cc
@@ -0,0 +1,66 @@
+/*
+ *  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 "video/render/incoming_video_stream.h"
+
+#include <memory>
+#include <utility>
+
+#include "absl/types/optional.h"
+#include "api/units/time_delta.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/trace_event.h"
+#include "video/render/video_render_frames.h"
+
+namespace webrtc {
+
+IncomingVideoStream::IncomingVideoStream(
+    TaskQueueFactory* task_queue_factory,
+    int32_t delay_ms,
+    rtc::VideoSinkInterface<VideoFrame>* callback)
+    : render_buffers_(delay_ms),
+      callback_(callback),
+      incoming_render_queue_(task_queue_factory->CreateTaskQueue(
+          "IncomingVideoStream",
+          TaskQueueFactory::Priority::HIGH)) {}
+
+IncomingVideoStream::~IncomingVideoStream() {
+  RTC_DCHECK(main_thread_checker_.IsCurrent());
+}
+
+void IncomingVideoStream::OnFrame(const VideoFrame& video_frame) {
+  TRACE_EVENT0("webrtc", "IncomingVideoStream::OnFrame");
+  RTC_CHECK_RUNS_SERIALIZED(&decoder_race_checker_);
+  RTC_DCHECK(!incoming_render_queue_.IsCurrent());
+  // TODO(srte): Using video_frame = std::move(video_frame) would move the frame
+  // into the lambda instead of copying it, but it doesn't work unless we change
+  // OnFrame to take its frame argument by value instead of const reference.
+  incoming_render_queue_.PostTask([this, video_frame = video_frame]() mutable {
+    RTC_DCHECK_RUN_ON(&incoming_render_queue_);
+    if (render_buffers_.AddFrame(std::move(video_frame)) == 1)
+      Dequeue();
+  });
+}
+
+void IncomingVideoStream::Dequeue() {
+  TRACE_EVENT0("webrtc", "IncomingVideoStream::Dequeue");
+  RTC_DCHECK_RUN_ON(&incoming_render_queue_);
+  absl::optional<VideoFrame> frame_to_render = render_buffers_.FrameToRender();
+  if (frame_to_render)
+    callback_->OnFrame(*frame_to_render);
+
+  if (render_buffers_.HasPendingFrames()) {
+    uint32_t wait_time = render_buffers_.TimeToNextFrameRelease();
+    incoming_render_queue_.PostDelayedHighPrecisionTask(
+        [this]() { Dequeue(); }, TimeDelta::Millis(wait_time));
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/video/render/incoming_video_stream.h b/third_party/libwebrtc/video/render/incoming_video_stream.h
new file mode 100644
index 0000000000..4873ae7dcb
--- /dev/null
+++ b/third_party/libwebrtc/video/render/incoming_video_stream.h
@@ -0,0 +1,48 @@
+/*
+ *  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 VIDEO_RENDER_INCOMING_VIDEO_STREAM_H_
+#define VIDEO_RENDER_INCOMING_VIDEO_STREAM_H_
+
+#include <stdint.h>
+
+#include "api/sequence_checker.h"
+#include "api/task_queue/task_queue_factory.h"
+#include "api/video/video_frame.h"
+#include "api/video/video_sink_interface.h"
+#include "rtc_base/race_checker.h"
+#include "rtc_base/task_queue.h"
+#include "rtc_base/thread_annotations.h"
+#include "video/render/video_render_frames.h"
+
+namespace webrtc {
+
+class IncomingVideoStream : public rtc::VideoSinkInterface<VideoFrame> {
+ public:
+  IncomingVideoStream(TaskQueueFactory* task_queue_factory,
+                      int32_t delay_ms,
+                      rtc::VideoSinkInterface<VideoFrame>* callback);
+  ~IncomingVideoStream() override;
+
+ private:
+  void OnFrame(const VideoFrame& video_frame) override;
+  void Dequeue();
+
+  SequenceChecker main_thread_checker_;
+  rtc::RaceChecker decoder_race_checker_;
+
+  VideoRenderFrames render_buffers_ RTC_GUARDED_BY(&incoming_render_queue_);
+  rtc::VideoSinkInterface<VideoFrame>* const callback_;
+  rtc::TaskQueue incoming_render_queue_;
+};
+
+}  // namespace webrtc
+
+#endif  // VIDEO_RENDER_INCOMING_VIDEO_STREAM_H_
diff --git a/third_party/libwebrtc/video/render/incoming_video_stream_gn/moz.build b/third_party/libwebrtc/video/render/incoming_video_stream_gn/moz.build
new file mode 100644
index 0000000000..ad1d7fbfb3
--- /dev/null
+++ b/third_party/libwebrtc/video/render/incoming_video_stream_gn/moz.build
@@ -0,0 +1,232 @@
+# 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.                                       ###
+
+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_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/video/render/incoming_video_stream.cc"
+]
+
+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"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_ENABLE_LIBEVENT"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    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
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_ENABLE_LIBEVENT"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_ENABLE_LIBEVENT"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["RTC_ENABLE_WIN_WGC"] = True
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "crypt32",
+        "iphlpapi",
+        "secur32",
+        "winmm"
+    ]
+
+if CONFIG["TARGET_CPU"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["TARGET_CPU"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["TARGET_CPU"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["TARGET_CPU"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["TARGET_CPU"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["TARGET_CPU"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "arm":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "x86":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "x86_64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("incoming_video_stream_gn")
diff --git a/third_party/libwebrtc/video/render/video_render_frames.cc b/third_party/libwebrtc/video/render/video_render_frames.cc
new file mode 100644
index 0000000000..ea1362abbb
--- /dev/null
+++ b/third_party/libwebrtc/video/render/video_render_frames.cc
@@ -0,0 +1,116 @@
+/*
+ *  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 "video/render/video_render_frames.h"
+
+#include <type_traits>
+#include <utility>
+
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/time_utils.h"
+#include "system_wrappers/include/metrics.h"
+
+namespace webrtc {
+namespace {
+// Don't render frames with timestamp older than 500ms from now.
+const int kOldRenderTimestampMS = 500;
+// Don't render frames with timestamp more than 10s into the future.
+const int kFutureRenderTimestampMS = 10000;
+
+const uint32_t kEventMaxWaitTimeMs = 200;
+const uint32_t kMinRenderDelayMs = 10;
+const uint32_t kMaxRenderDelayMs = 500;
+const size_t kMaxIncomingFramesBeforeLogged = 100;
+
+uint32_t EnsureValidRenderDelay(uint32_t render_delay) {
+  return (render_delay < kMinRenderDelayMs || render_delay > kMaxRenderDelayMs)
+             ? kMinRenderDelayMs
+             : render_delay;
+}
+}  // namespace
+
+VideoRenderFrames::VideoRenderFrames(uint32_t render_delay_ms)
+    : render_delay_ms_(EnsureValidRenderDelay(render_delay_ms)) {}
+
+VideoRenderFrames::~VideoRenderFrames() {
+  frames_dropped_ += incoming_frames_.size();
+  RTC_HISTOGRAM_COUNTS_1000("WebRTC.Video.DroppedFrames.RenderQueue",
+                            frames_dropped_);
+  RTC_LOG(LS_INFO) << "WebRTC.Video.DroppedFrames.RenderQueue "
+                   << frames_dropped_;
+}
+
+int32_t VideoRenderFrames::AddFrame(VideoFrame&& new_frame) {
+  const int64_t time_now = rtc::TimeMillis();
+
+  // Drop old frames only when there are other frames in the queue, otherwise, a
+  // really slow system never renders any frames.
+  if (!incoming_frames_.empty() &&
+      new_frame.render_time_ms() + kOldRenderTimestampMS < time_now) {
+    RTC_LOG(LS_WARNING) << "Too old frame, timestamp=" << new_frame.timestamp();
+    ++frames_dropped_;
+    return -1;
+  }
+
+  if (new_frame.render_time_ms() > time_now + kFutureRenderTimestampMS) {
+    RTC_LOG(LS_WARNING) << "Frame too long into the future, timestamp="
+                        << new_frame.timestamp();
+    ++frames_dropped_;
+    return -1;
+  }
+
+  if (new_frame.render_time_ms() < last_render_time_ms_) {
+    RTC_LOG(LS_WARNING) << "Frame scheduled out of order, render_time="
+                        << new_frame.render_time_ms()
+                        << ", latest=" << last_render_time_ms_;
+    // For more details, see bug:
+    // https://bugs.chromium.org/p/webrtc/issues/detail?id=7253
+    ++frames_dropped_;
+    return -1;
+  }
+
+  last_render_time_ms_ = new_frame.render_time_ms();
+  incoming_frames_.emplace_back(std::move(new_frame));
+
+  if (incoming_frames_.size() > kMaxIncomingFramesBeforeLogged) {
+    RTC_LOG(LS_WARNING) << "Stored incoming frames: "
+                        << incoming_frames_.size();
+  }
+  return static_cast<int32_t>(incoming_frames_.size());
+}
+
+absl::optional<VideoFrame> VideoRenderFrames::FrameToRender() {
+  absl::optional<VideoFrame> render_frame;
+  // Get the newest frame that can be released for rendering.
+  while (!incoming_frames_.empty() && TimeToNextFrameRelease() <= 0) {
+    if (render_frame) {
+      ++frames_dropped_;
+    }
+    render_frame = std::move(incoming_frames_.front());
+    incoming_frames_.pop_front();
+  }
+  return render_frame;
+}
+
+uint32_t VideoRenderFrames::TimeToNextFrameRelease() {
+  if (incoming_frames_.empty()) {
+    return kEventMaxWaitTimeMs;
+  }
+  const int64_t time_to_release = incoming_frames_.front().render_time_ms() -
+                                  render_delay_ms_ - rtc::TimeMillis();
+  return time_to_release < 0 ? 0u : static_cast<uint32_t>(time_to_release);
+}
+
+bool VideoRenderFrames::HasPendingFrames() const {
+  return !incoming_frames_.empty();
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/video/render/video_render_frames.h b/third_party/libwebrtc/video/render/video_render_frames.h
new file mode 100644
index 0000000000..7f48eae496
--- /dev/null
+++ b/third_party/libwebrtc/video/render/video_render_frames.h
@@ -0,0 +1,55 @@
+/*
+ *  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 VIDEO_RENDER_VIDEO_RENDER_FRAMES_H_
+#define VIDEO_RENDER_VIDEO_RENDER_FRAMES_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <list>
+
+#include "absl/types/optional.h"
+#include "api/video/video_frame.h"
+
+namespace webrtc {
+
+// Class definitions
+class VideoRenderFrames {
+ public:
+  explicit VideoRenderFrames(uint32_t render_delay_ms);
+  VideoRenderFrames(const VideoRenderFrames&) = delete;
+  ~VideoRenderFrames();
+
+  // Add a frame to the render queue
+  int32_t AddFrame(VideoFrame&& new_frame);
+
+  // Get a frame for rendering, or false if it's not time to render.
+  absl::optional<VideoFrame> FrameToRender();
+
+  // Returns the number of ms to next frame to render
+  uint32_t TimeToNextFrameRelease();
+
+  bool HasPendingFrames() const;
+
+ private:
+  // Sorted list with framed to be rendered, oldest first.
+  std::list<VideoFrame> incoming_frames_;
+
+  // Estimated delay from a frame is released until it's rendered.
+  const uint32_t render_delay_ms_;
+
+  int64_t last_render_time_ms_ = 0;
+  size_t frames_dropped_ = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // VIDEO_RENDER_VIDEO_RENDER_FRAMES_H_
diff --git a/third_party/libwebrtc/video/render/video_render_frames_gn/moz.build b/third_party/libwebrtc/video/render/video_render_frames_gn/moz.build
new file mode 100644
index 0000000000..d5de8cd178
--- /dev/null
+++ b/third_party/libwebrtc/video/render/video_render_frames_gn/moz.build
@@ -0,0 +1,232 @@
+# 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.                                       ###
+
+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_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "!/third_party/libwebrtc/gen",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/video/render/video_render_frames.cc"
+]
+
+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"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["ANDROID"] = True
+    DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+    DEFINES["HAVE_SYS_UIO_H"] = True
+    DEFINES["WEBRTC_ANDROID"] = True
+    DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+    DEFINES["WEBRTC_ENABLE_LIBEVENT"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_GNU_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+    OS_LIBS += [
+        "log"
+    ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["WEBRTC_MAC"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    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
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_NSS_CERTS"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_UDEV"] = True
+    DEFINES["WEBRTC_ENABLE_LIBEVENT"] = True
+    DEFINES["WEBRTC_LINUX"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["USE_GLIB"] = "1"
+    DEFINES["USE_OZONE"] = "1"
+    DEFINES["USE_X11"] = "1"
+    DEFINES["WEBRTC_BSD"] = True
+    DEFINES["WEBRTC_ENABLE_LIBEVENT"] = True
+    DEFINES["WEBRTC_POSIX"] = True
+    DEFINES["_FILE_OFFSET_BITS"] = "64"
+    DEFINES["_LARGEFILE64_SOURCE"] = True
+    DEFINES["_LARGEFILE_SOURCE"] = True
+    DEFINES["__STDC_CONSTANT_MACROS"] = True
+    DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+    DEFINES["NOMINMAX"] = True
+    DEFINES["NTDDI_VERSION"] = "0x0A000000"
+    DEFINES["PSAPI_VERSION"] = "2"
+    DEFINES["RTC_ENABLE_WIN_WGC"] = True
+    DEFINES["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_WIN"] = True
+    DEFINES["WIN32"] = True
+    DEFINES["WIN32_LEAN_AND_MEAN"] = True
+    DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+    DEFINES["WINVER"] = "0x0A00"
+    DEFINES["_ATL_NO_OPENGL"] = True
+    DEFINES["_CRT_RAND_S"] = True
+    DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+    DEFINES["_HAS_EXCEPTIONS"] = "0"
+    DEFINES["_HAS_NODISCARD"] = True
+    DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+    DEFINES["_SECURE_ATL"] = True
+    DEFINES["_UNICODE"] = True
+    DEFINES["_WIN32_WINNT"] = "0x0A00"
+    DEFINES["_WINDOWS"] = True
+    DEFINES["__STD_C"] = True
+
+    OS_LIBS += [
+        "crypt32",
+        "iphlpapi",
+        "secur32",
+        "winmm"
+    ]
+
+if CONFIG["TARGET_CPU"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["TARGET_CPU"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["TARGET_CPU"] == "mips32":
+
+    DEFINES["MIPS32_LE"] = True
+    DEFINES["MIPS_FPU_LE"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["TARGET_CPU"] == "mips64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["TARGET_CPU"] == "x86":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["TARGET_CPU"] == "x86_64":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+    DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+    DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["USE_X11"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "arm":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["OS_TARGET"] == "Android" and CONFIG["TARGET_CPU"] == "x86":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "aarch64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "arm":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "x86":
+
+    CXXFLAGS += [
+        "-msse2"
+    ]
+
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["OS_TARGET"] == "Linux" and CONFIG["TARGET_CPU"] == "x86_64":
+
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("video_render_frames_gn")