Update to m137

.gitignore

@@ -73,3 +73,9 @@
 /xcodebuild
 /.vscode
 !webrtc/*
+/tmp.patch
+/out-release
+/out-debug
+/node_modules
+/libwebrtc
+/args.txt

DEPS

@@ -75,7 +75,7 @@ deps = {
   'src/base':
     'https://chromium.googlesource.com/chromium/src/base@86c814633cf284bc8057a539bc722e2a672afe2f',
   'src/build':
-    'https://chromium.googlesource.com/chromium/src/build@88030b320338e0706b6b93336c4b35e6bbaf467e',
+    'https://github.com/webrtc-sdk/build@9af2ddd8e5ad6278165cadfa554bea6f25081dd2',
   'src/buildtools':
     'https://chromium.googlesource.com/chromium/src/buildtools@0f32cb9025766951122d4ed19aba87a94ded3f43',
   # Gradle 6.6.1. Used for testing Android Studio project generation for WebRTC.

NOTICE

@@ -0,0 +1,26 @@
+###################################################################################
+
+The following modifications follow Apache License 2.0 from shiguredo.
+
+https://github.com/webrtc-sdk/webrtc/commit/dfec53e93a0a1cb93f444caf50f844ec0068c7b7
+https://github.com/webrtc-sdk/webrtc/commit/403b4678543c5d4ac77bd1ea5753c02637b3bb89
+https://github.com/webrtc-sdk/webrtc/commit/77d5d685a90fb4bded17835ae72ec6671b26d696
+
+Apache License 2.0
+
+Copyright 2019-2021, Wandbox LLC (Original Author)
+Copyright 2019-2021, Shiguredo Inc.
+
+Licensed under the Apache License, Version 2.0 (the "License");
+you may not use this file except in compliance with the License.
+You may obtain a copy of the License at
+
+    http://www.apache.org/licenses/LICENSE-2.0
+
+Unless required by applicable law or agreed to in writing, software
+distributed under the License is distributed on an "AS IS" BASIS,
+WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+See the License for the specific language governing permissions and
+limitations under the License.
+
+#####################################################################################

README.md

@@ -1,32 +1,47 @@
-**WebRTC is a free, open software project** that provides browsers and mobile
-applications with Real-Time Communications (RTC) capabilities via simple APIs.
-The WebRTC components have been optimized to best serve this purpose.
+# WebRTC-SDK
 
-**Our mission:** To enable rich, high-quality RTC applications to be
-developed for the browser, mobile platforms, and IoT devices, and allow them
-all to communicate via a common set of protocols.
+This repository contains a fork of WebRTC from Google with various improvements.
 
-The WebRTC initiative is a project supported by Google, Mozilla and Opera,
-amongst others.
+## Main changes
 
-### Development
+### All
 
-See [here][native-dev] for instructions on how to get started
-developing with the native code.
+- Dynamically acquire decoder to mitigate decoder limitations [#25](https://github.com/webrtc-sdk/webrtc/pull/25)
+- Support for video simulcast with hardware & software encoders [patch](https://github.com/webrtc-sdk/webrtc/commit/ee030264e2274a2c90548a99b448782049e48fb4)
+- Frame cryptor support (for end-to-end encryption) [patch](https://github.com/webrtc-sdk/webrtc/commit/3a2c008529a15fecde5f979a6ebb75c05463d45e)
 
-[Authoritative list](native-api.md) of directories that contain the
-native API header files.
+### Android
 
-### More info
+- WrappedVideoDecoderFactory [#74](https://github.com/webrtc-sdk/webrtc/pull/74)
 
- * Official web site: http://www.webrtc.org
- * Master source code repo: https://webrtc.googlesource.com/src
- * Samples and reference apps: https://github.com/webrtc
- * Mailing list: http://groups.google.com/group/discuss-webrtc
- * Continuous build: https://ci.chromium.org/p/webrtc/g/ci/console
- * [Coding style guide](g3doc/style-guide.md)
- * [Code of conduct](CODE_OF_CONDUCT.md)
- * [Reporting bugs](docs/bug-reporting.md)
- * [Documentation](g3doc/sitemap.md)
+### iOS / Mac
 
-[native-dev]: https://webrtc.googlesource.com/src/+/main/docs/native-code/
+- Sane audio handling [patch](https://github.com/webrtc-sdk/webrtc/commit/272127d457ab48e36241e82549870405864851f6)
+  - Do not acquire microphone/permissions unless actively publishing audio
+  - Abililty to bypass voice processing on iOS
+  - Remove hardcoded limitation of outputting only to right speaker on MacBook Pro
+- Desktop capture for Mac [patch](https://github.com/webrtc-sdk/webrtc/commit/8e832d1163644ab504412c9b8f3ba8510d9890d6)
+
+### Windows
+
+- Fixed unable to acquire Mic when built-in AEC is enabled [#29](https://github.com/webrtc-sdk/webrtc/pull/29)
+
+## LICENSE
+
+- [Google WebRTC](https://chromium.googlesource.com/external/webrtc.git), is licensed under [BSD license](/LICENSE).
+
+- Contains patches from [shiguredo-webrtc-build](https://github.com/shiguredo-webrtc-build), licensed under [Apache 2.0](/NOTICE).
+
+- Contains changes from LiveKit, licensed under Apache 2.0.
+
+## Who is using this project
+
+- [flutter-webrtc](https://github.com/flutter-webrtc/flutter-webrtc)
+
+- [LiveKit](https://github.com/livekit)
+
+- [Membrane Framework](https://github.com/membraneframework/membrane_rtc_engine)
+
+- [Louper](https://louper.io)
+
+Are you using WebRTC SDK in your framework or app? Feel free to open a PR and add yourself!

api/BUILD.gn

@@ -382,6 +382,7 @@ rtc_library("libjingle_peerconnection_api") {
     "video:encoded_image",
     "video:video_bitrate_allocator_factory",
     "video:video_frame",
+    "video:yuv_helper",
     "video:video_rtp_headers",
     "video_codecs:video_codecs_api",
     "//third_party/abseil-cpp/absl/algorithm:container",

api/audio/audio_device.h

@@ -22,6 +22,15 @@ namespace webrtc {
 
 class AudioDeviceModuleForTest;
 
+// Sink for callbacks related to a audio device.
+class AudioDeviceSink {
+ public:
+  virtual ~AudioDeviceSink() = default;
+
+  // input/output devices updated or default device changed
+  virtual void OnDevicesUpdated() = 0;
+};
+
 class AudioDeviceModule : public webrtc::RefCountInterface {
  public:
   enum AudioLayer {
@@ -62,11 +71,11 @@ class AudioDeviceModule : public webrtc::RefCountInterface {
  public:
   // Creates a default ADM for usage in production code.
   static scoped_refptr<AudioDeviceModule> Create(
-      AudioLayer audio_layer, TaskQueueFactory* task_queue_factory);
+      AudioLayer audio_layer, TaskQueueFactory* task_queue_factory, bool bypass_voice_processing = false);
   // Creates an ADM with support for extra test methods. Don't use this factory
   // in production code.
   static scoped_refptr<AudioDeviceModuleForTest> CreateForTest(
-      AudioLayer audio_layer, TaskQueueFactory* task_queue_factory);
+      AudioLayer audio_layer, TaskQueueFactory* task_queue_factory, bool bypass_voice_processing = false);
 
   // Retrieve the currently utilized audio layer
   virtual int32_t ActiveAudioLayer(AudioLayer* audioLayer) const = 0;
@@ -176,6 +185,10 @@ class AudioDeviceModule : public webrtc::RefCountInterface {
   virtual int GetRecordAudioParameters(AudioParameters* params) const = 0;
 #endif  // WEBRTC_IOS
 
+  virtual int32_t SetAudioDeviceSink(AudioDeviceSink* sink) const { return -1; }
+  virtual int32_t GetPlayoutDevice() const { return -1; }
+  virtual int32_t GetRecordingDevice() const { return -1; }
+
  protected:
   ~AudioDeviceModule() override {}
 };

api/crypto/BUILD.gn

@@ -16,6 +16,24 @@ group("crypto") {
   ]
 }
 
+rtc_library("frame_crypto_transformer") {
+  visibility = [ "*" ]
+  sources = [
+    "frame_crypto_transformer.cc",
+    "frame_crypto_transformer.h",
+  ]
+
+  deps = [
+   "//api:frame_transformer_interface",
+  ]
+
+  if (rtc_build_ssl) {
+    deps += [ "//third_party/boringssl" ]
+  } else {
+    configs += [ ":external_ssl_library" ]
+  }
+}
+
 rtc_library("options") {
   visibility = [ "*" ]
   sources = [