Adding upstream version 110.0.1.upstream/110.0.1upstream

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

4 files changed, 385 insertions, 0 deletions

diff --git a/third_party/libwebrtc/modules/async_audio_processing/BUILD.gn b/third_party/libwebrtc/modules/async_audio_processing/BUILD.gn
new file mode 100644
index 0000000000..7a7ca20df1
--- /dev/null
+++ b/third_party/libwebrtc/modules/async_audio_processing/BUILD.gn
@@ -0,0 +1,43 @@
+# 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.
+
+import("../../webrtc.gni")
+
+rtc_library("async_audio_processing") {
+  sources = [
+    "async_audio_processing.cc",
+    "async_audio_processing.h",
+  ]
+
+  public = [ "async_audio_processing.h" ]
+
+  deps = [
+    "../../api:scoped_refptr",
+    "../../api:sequence_checker",
+    "../../api/audio:audio_frame_api",
+    "../../api/audio:audio_frame_processor",
+    "../../api/task_queue:task_queue",
+    "../../rtc_base:checks",
+    "../../rtc_base:refcount",
+    "../../rtc_base:rtc_task_queue",
+  ]
+}
+
+if (rtc_include_tests) {
+  rtc_library("async_audio_processing_test") {
+    testonly = true
+
+    sources = []
+
+    deps = [
+      ":async_audio_processing",
+      "../../api/audio:audio_frame_api",
+      "../../rtc_base:checks",
+    ]
+  }
+}
diff --git a/third_party/libwebrtc/modules/async_audio_processing/async_audio_processing.cc b/third_party/libwebrtc/modules/async_audio_processing/async_audio_processing.cc
new file mode 100644
index 0000000000..9452f3bcf9
--- /dev/null
+++ b/third_party/libwebrtc/modules/async_audio_processing/async_audio_processing.cc
@@ -0,0 +1,61 @@
+
+/*
+ *  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.
+ */
+#include "modules/async_audio_processing/async_audio_processing.h"
+
+#include <utility>
+
+#include "api/audio/audio_frame.h"
+#include "api/task_queue/task_queue_factory.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+AsyncAudioProcessing::Factory::~Factory() = default;
+AsyncAudioProcessing::Factory::Factory(AudioFrameProcessor& frame_processor,
+                                       TaskQueueFactory& task_queue_factory)
+    : frame_processor_(frame_processor),
+      task_queue_factory_(task_queue_factory) {}
+
+std::unique_ptr<AsyncAudioProcessing>
+AsyncAudioProcessing::Factory::CreateAsyncAudioProcessing(
+    AudioFrameProcessor::OnAudioFrameCallback on_frame_processed_callback) {
+  return std::make_unique<AsyncAudioProcessing>(
+      frame_processor_, task_queue_factory_,
+      std::move(on_frame_processed_callback));
+}
+
+AsyncAudioProcessing::~AsyncAudioProcessing() {
+  frame_processor_.SetSink(nullptr);
+}
+
+AsyncAudioProcessing::AsyncAudioProcessing(
+    AudioFrameProcessor& frame_processor,
+    TaskQueueFactory& task_queue_factory,
+    AudioFrameProcessor::OnAudioFrameCallback on_frame_processed_callback)
+    : on_frame_processed_callback_(std::move(on_frame_processed_callback)),
+      frame_processor_(frame_processor),
+      task_queue_(task_queue_factory.CreateTaskQueue(
+          "AsyncAudioProcessing",
+          TaskQueueFactory::Priority::NORMAL)) {
+  frame_processor_.SetSink([this](std::unique_ptr<AudioFrame> frame) {
+    task_queue_.PostTask([this, frame = std::move(frame)]() mutable {
+      on_frame_processed_callback_(std::move(frame));
+    });
+  });
+}
+
+void AsyncAudioProcessing::Process(std::unique_ptr<AudioFrame> frame) {
+  task_queue_.PostTask([this, frame = std::move(frame)]() mutable {
+    frame_processor_.Process(std::move(frame));
+  });
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/async_audio_processing/async_audio_processing.h b/third_party/libwebrtc/modules/async_audio_processing/async_audio_processing.h
new file mode 100644
index 0000000000..bbd0f69b1b
--- /dev/null
+++ b/third_party/libwebrtc/modules/async_audio_processing/async_audio_processing.h
@@ -0,0 +1,76 @@
+/*
+ *  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.
+ */
+
+#ifndef MODULES_ASYNC_AUDIO_PROCESSING_ASYNC_AUDIO_PROCESSING_H_
+#define MODULES_ASYNC_AUDIO_PROCESSING_ASYNC_AUDIO_PROCESSING_H_
+
+#include <memory>
+
+#include "api/audio/audio_frame_processor.h"
+#include "rtc_base/ref_count.h"
+#include "rtc_base/task_queue.h"
+
+namespace webrtc {
+
+class AudioFrame;
+class TaskQueueFactory;
+
+// Helper class taking care of interactions with AudioFrameProcessor
+// in asynchronous manner. Offloads AudioFrameProcessor::Process calls
+// to a dedicated task queue. Makes sure that it's always safe for
+// AudioFrameProcessor to pass processed frames back to its sink.
+class AsyncAudioProcessing final {
+ public:
+  // Helper class passing AudioFrameProcessor and TaskQueueFactory into
+  // AsyncAudioProcessing constructor.
+  class Factory : public rtc::RefCountInterface {
+   public:
+    Factory(const Factory&) = delete;
+    Factory& operator=(const Factory&) = delete;
+
+    ~Factory();
+    Factory(AudioFrameProcessor& frame_processor,
+            TaskQueueFactory& task_queue_factory);
+
+    std::unique_ptr<AsyncAudioProcessing> CreateAsyncAudioProcessing(
+        AudioFrameProcessor::OnAudioFrameCallback on_frame_processed_callback);
+
+   private:
+    AudioFrameProcessor& frame_processor_;
+    TaskQueueFactory& task_queue_factory_;
+  };
+
+  AsyncAudioProcessing(const AsyncAudioProcessing&) = delete;
+  AsyncAudioProcessing& operator=(const AsyncAudioProcessing&) = delete;
+
+  ~AsyncAudioProcessing();
+
+  // Creates AsyncAudioProcessing which will pass audio frames to
+  // `frame_processor` on `task_queue_` and reply with processed frames passed
+  // into `on_frame_processed_callback`, which is posted back onto
+  // `task_queue_`. `task_queue_` is created using the provided
+  // `task_queue_factory`.
+  AsyncAudioProcessing(
+      AudioFrameProcessor& frame_processor,
+      TaskQueueFactory& task_queue_factory,
+      AudioFrameProcessor::OnAudioFrameCallback on_frame_processed_callback);
+
+  // Accepts `frame` for asynchronous processing. Thread-safe.
+  void Process(std::unique_ptr<AudioFrame> frame);
+
+ private:
+  AudioFrameProcessor::OnAudioFrameCallback on_frame_processed_callback_;
+  AudioFrameProcessor& frame_processor_;
+  rtc::TaskQueue task_queue_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_ASYNC_AUDIO_PROCESSING_ASYNC_AUDIO_PROCESSING_H_
diff --git a/third_party/libwebrtc/modules/async_audio_processing/async_audio_processing_gn/moz.build b/third_party/libwebrtc/modules/async_audio_processing/async_audio_processing_gn/moz.build
new file mode 100644
index 0000000000..24f8656db0
--- /dev/null
+++ b/third_party/libwebrtc/modules/async_audio_processing/async_audio_processing_gn/moz.build
@@ -0,0 +1,205 @@
+# 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"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+    "!/ipc/ipdl/_ipdlheaders",
+    "/ipc/chromium/src",
+    "/third_party/libwebrtc/",
+    "/third_party/libwebrtc/third_party/abseil-cpp/",
+    "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+    "/third_party/libwebrtc/modules/async_audio_processing/async_audio_processing.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_AVX2"] = 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_ENABLE_AVX2"] = True
+    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_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_AVX2"] = 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["UNICODE"] = True
+    DEFINES["USE_AURA"] = "1"
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+    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 += [
+        "winmm"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+    DEFINES["WEBRTC_ARCH_ARM64"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+    CXXFLAGS += [
+        "-mfpu=neon"
+    ]
+
+    DEFINES["WEBRTC_ARCH_ARM"] = True
+    DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+    DEFINES["WEBRTC_HAS_NEON"] = 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["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support",
+        "unwind"
+    ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+    OS_LIBS += [
+        "android_support"
+    ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+    DEFINES["WEBRTC_ENABLE_AVX2"] = True
+    DEFINES["_GNU_SOURCE"] = True
+
+Library("async_audio_processing_gn")