update to m114

.gitignore

@@ -72,3 +72,9 @@
 /xcodebuild
 /.vscode
 !webrtc/*
+/tmp.patch
+/out-release
+/out-debug
+/node_modules
+/libwebrtc
+/args.txt

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 +1,49 @@
-**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.
+## 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 #26
 
-[Authoritative list](native-api.md) of directories that contain the
-native API header files.
+### Android
 
-### More info
+- Support for video simulcast #3
 
- * 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
+
+- Do not request microphone permissions for playback-only #2 #5
+- Improvements to AVAudioSession interactions #7 #8
+- Support for video simulcast #4
+- Support for voice processing bypass #15
+
+### Mac
+
+- Support for video simulcast #10
+- Remove hardcoded limitation of outputting to only right speaker on MBP #22
+- Screen capture support #24 #36 #37
+- Support for audio output device selection #35
+- Cross-platform RTCMTLVideoView #40
+
+### Windows
+
+- Fixed unable to acquire Mic when built-in AEC is enabled #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)
 
-[native-dev]: https://webrtc.googlesource.com/src/+/main/docs/native-code/index.md

api/BUILD.gn

@@ -330,6 +330,7 @@ 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",
 

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 = [