Sync with livekit's m125 (#42)

* Update to m125. (#119)

Use M125 as the latest version and migrate historical patches to m125

Patches Group:

## 1. Update README.md
webrtc-sdk/webrtc@b6c65fc
* Add Apache-2.0 license and some note to README.md. (#9)
* Updated readme detailing changes from original (#42)
* Adding membrane framework (#51)
* Updated readme (#83)

## 2. Audio Device Optimization
webrtc-sdk/webrtc@7454824
* allow listen-only mode in AudioUnit, adjust when category changes
(webrtc-sdk/webrtc#2)
* release mic when category changes
(webrtc-sdk/webrtc#5)
* Change defaults to iOS defaults
(webrtc-sdk/webrtc#7)
* Sync audio session config
(webrtc-sdk/webrtc#8)
* feat: support bypass voice processing for iOS.
(webrtc-sdk/webrtc#15)
* Remove MacBookPro audio pan right code
(webrtc-sdk/webrtc#22)
* fix: Fix can't open mic alone when built-in AEC is enabled.
(webrtc-sdk/webrtc#29)
* feat: add audio device changes detect for windows.
(webrtc-sdk/webrtc#41)
* fix Linux compile (webrtc-sdk/webrtc#47)
* AudioUnit: Don't rely on category switch for mic indicator to turn off
(webrtc-sdk/webrtc#52)
* Stop recording on mute (turn off mic indicator)
(webrtc-sdk/webrtc#55)
* Cherry pick audio selection from m97 release
(webrtc-sdk/webrtc#35)
* [Mac] Allow audio device selection
(webrtc-sdk/webrtc#21)
* RTCAudioDeviceModule.outputDevice / inputDevice getter and setter
(webrtc-sdk/webrtc#80)
* Allow custom audio processing by exposing AudioProcessingModule
(webrtc-sdk/webrtc#85)
* Expose audio sample buffers for Android
(webrtc-sdk/webrtc#89)
* feat: add external audio processor for android.
(webrtc-sdk/webrtc#103)
* android: make audio output attributes modifiable
(webrtc-sdk/webrtc#118)
* Fix external audio processor sample rate calculation
(webrtc-sdk/webrtc#108)
* Expose remote audio sample buffers on RTCAudioTrack
(webrtc-sdk/webrtc#84)
* Fix memory leak when creating audio CMSampleBuffer
webrtc-sdk/webrtc#86

## 3. Simulcast/SVC support for iOS/Android.
webrtc-sdk/webrtc@b0b9fe9

- Simulcast support for iOS SDK (#4)
- Support for simulcast in Android SDK (#3)
- include simulcast headers for mac also (#10)
- Fix simulcast using hardware encoder on Android (#48)
- Add scalabilityMode support for AV1/VP9. (#90)

## 4. Android improvements.
webrtc-sdk/webrtc@9aaaab5
- Start/Stop receiving stream method for VideoTrack (#25)
- Properly remove observer upon deconstruction (#26)
- feat: Expose setCodecPreferences/getCapabilities for android. (#61)
- fix: add WrappedVideoDecoderFactory.java. (#74)

## 5. Darwin improvements
webrtc-sdk/webrtc@a13ea17
- [Mac/iOS] feat: Add RTCYUVHelper for darwin. (#28)
- Cross-platform `RTCMTLVideoView` for both iOS / macOS (#40)
- rotationOverride should not be assign (#44)
- [ObjC] Expose properties / methods required for AV1 codec support
(#60)
- Workaround: Render PixelBuffer in RTCMTLVideoView (#58)
- Improve iOS/macOS H264 encoder (#70)
- fix: fix video encoder not resuming correctly upon foregrounding
(#75).
- add PrivacyInfo.xcprivacy to darwin frameworks. (#112)
- Add NSPrivacyCollectedDataTypes key to xcprivacy file (#114)
- Thread-safe `RTCInitFieldTrialDictionary` (#116)
- Set RTCCameraVideoCapturer initial zoom factor (#121)
- Unlock configuration before starting capture session (#122)

## 6. Desktop Capture for macOS.
webrtc-sdk/webrtc@841d78f
- [Mac] feat: Support screen capture for macOS. (#24) (#36)
- fix: Get thumbnails asynchronously. (#37)
- fix: Use CVPixelBuffer to build DesktopCapture Frame, fix the crash
caused by non-CVPixelBuffer frame in RTCVideoEncoderH264 that cannot be
cropped. (#63)
- Fix the crash when setting the fps of the virtual camera. (#62)

## 7. Frame Cryptor Support.
webrtc-sdk/webrtc@fc08745
- feat: Frame Cryptor (aes gcm/cbc). (#54)
- feat: key ratchet/derive. (#66)
- fix: skip invalid key when decryption failed. (#81)
- Improve e2ee, add setSharedKey to KeyProvider. (#88)
- add failure tolerance for framecryptor. (#91)
- fix h264 freeze. (#93)
- Fix/send frame cryptor events from signaling thread (#95)
- more improvements for E2EE. (#96)
- remove too verbose logs (#107)
- Add key ring size to keyProviderOptions. (#109)

## 8. Other improvements.
webrtc-sdk/webrtc@eed6c8a
- Added yuv_helper (#57)
- ABGRToI420, ARGBToI420 & ARGBToRGB24 (#65)
- more yuv wrappers (#87)
- Fix naming for yuv helper (#113)
- Fix missing `RTC_OBJC_TYPE` macros (#100)

---------

Co-authored-by: Hiroshi Horie <[email protected]>
Co-authored-by: David Zhao <[email protected]>
Co-authored-by: davidliu <[email protected]>
Co-authored-by: Angelika Serwa <[email protected]>
Co-authored-by: Théo Monnom <[email protected]>
# Conflicts:
#	README.md
#	media/engine/webrtc_video_engine.cc
#	media/engine/webrtc_video_engine.h
#	modules/audio_device/audio_device_impl.cc
#	sdk/BUILD.gn
#	sdk/android/BUILD.gn
#	sdk/android/api/org/webrtc/RtpParameters.java
#	sdk/android/api/org/webrtc/SimulcastVideoEncoder.java
#	sdk/android/api/org/webrtc/SimulcastVideoEncoderFactory.java
#	sdk/android/api/org/webrtc/VideoCodecInfo.java
#	sdk/android/src/jni/pc/rtp_parameters.cc
#	sdk/android/src/jni/simulcast_video_encoder.cc
#	sdk/android/src/jni/simulcast_video_encoder.h
#	sdk/android/src/jni/video_codec_info.cc
#	sdk/objc/api/peerconnection/RTCAudioDeviceModule+Private.h
#	sdk/objc/api/peerconnection/RTCAudioDeviceModule.h
#	sdk/objc/api/peerconnection/RTCAudioDeviceModule.mm
#	sdk/objc/api/peerconnection/RTCAudioTrack.mm
#	sdk/objc/api/peerconnection/RTCIODevice+Private.h
#	sdk/objc/api/peerconnection/RTCIODevice.mm
#	sdk/objc/api/peerconnection/RTCPeerConnectionFactory.h
#	sdk/objc/api/peerconnection/RTCPeerConnectionFactory.mm
#	sdk/objc/api/video_codec/RTCVideoEncoderSimulcast.h
#	sdk/objc/api/video_codec/RTCVideoEncoderSimulcast.mm
#	sdk/objc/base/RTCAudioRenderer.h
#	sdk/objc/components/video_codec/RTCVideoEncoderFactorySimulcast.h
#	sdk/objc/components/video_codec/RTCVideoEncoderFactorySimulcast.mm

* fix: duplicate simulcast entries

* remove duplicate declaration

* remove duplicate audioDeviceModule

* fix: removed livekit's external audio processor

* fix: add back simulcast factories

* Fix missing RTC_OBJC_TYPE macros

* Fix missing headers and Metal linking

# Conflicts:
#	sdk/BUILD.gn

* Fix Mac Catalyst `RTCCameraVideoCapturer` rotation (#126)

* Fix set frame transformer (#125)

* Fix webrtc_voice_engine not notifying mute change (#128)

Looks like this line was missed during the m125 update.

webrtc-sdk/webrtc@272127d#diff-56f5e0c459b287281ef3b0431d3f4129e8e4be4c6955d845bcb22210f08b7ba5R2289

Adding it back in so that mic is properly released when muted.
# Conflicts:
#	media/engine/webrtc_voice_engine.cc

* android: Allow for skipping checking the audio playstate if needed (#129)

Pausing/stopping the audio track can lead to a race condition against
the AudioTrackThread due to this assert. Normally this is fine since
directly pausing/stopping isn't possible, but user is using reflection
to workaround another audio issue (muted participants still have a
sending audio stream which keeps the audio alive, affecting global sound
if in the background).

Not a full fix, as would like to manually control the audio track
directly (needs a bigger fix to handle proper synchronization before
allowing public access), but this will work through reflection (user
takes responsibility for usage).

* Allow to pass in capture session to RTCCameraVideoCapturer (#132)

Expose initializers to pass in capture session to RTCCameraVideoCapturer
so we can use AVCaptureMultiCamSession etc to capture front and back
simultaneously for iOS.

* Fix NetworkMonitor race condition when dispatching native observers (#135)

There is a race condition in NetworkMonitor where native observers may
be removed concurrently with a notification being dispatched, leading to
a dangling pointer dereference (trying to dispatch an observer that was
already removed and destroyed), and from there a crash with access
violation.

By ensuring dispatching to native observers is done within the
synchronization lock that guards additions/removals of native observers
protects against this race condition. Since native observers callbacks
are posted to the networking thread in the C++ side anyway, there should
be no risk of deadlock/starvation due to long-running observers.

Bug: webrtc:15837
Change-Id: Id2b788f102dbd25de76ceed434c4cd68aa9a569e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/338643
Reviewed-by: Taylor Brandstetter <[email protected]>
Commit-Queue: Harald Alvestrand <[email protected]>
Reviewed-by: Harald Alvestrand <[email protected]>
Cr-Commit-Position: refs/heads/main@{#42256}

Co-authored-by: Guy Hershenbaum <[email protected]>

* Support for Vision Pro (#131)

TODO:
- [x]  fix compile for RTCCameraVideoCapturer
- [ ]  fix RTCMTLRenderer ?

---------

Co-authored-by: Hiroshi Horie <[email protected]>

* Multicam support (#137)

TODO: 
- [x] Return `.systemPreferredCamera` for devices (visionOS only).
- [x] Use `AVCaptureMultiCamSession` only if `isMultiCamSupported` is
true.
- [x] Silence statusBarOrientation warning.

---------

Co-authored-by: [email protected] <[email protected]>

* tvOS support (#139)

17.0+ only atm

---------

Co-authored-by: cloudwebrtc <[email protected]>

* Add isDisposed to MediaStreamTrack (#140)

* chore: handle invalid cipher from key size. (#142)

* Allow software AEC for Simulator (#143)

~Allow to use "googEchoCancellation" constraint for software AEC.
For devices "googEchoCancellation" should be false to use
VoiceProcessingIO.~

* Fix AudioRenderer crash & expose AVAudioPCMBuffer (#144)

* fix: Fix bug for bypass voice processing. (#147)

* chore: remove aes cbc for framecryptor. (#145)

* Change audio renderer output format (#149)

Instead of converting to Float, output original Int data without
conversion.
Output the raw format and convert when required.

* Fixed issue with missing network interfaces on iOS (#151)

Related issue: webrtc-sdk/webrtc#148
Cherry-pick :
https://webrtc.googlesource.com/src/+/fea60ef8e72fb17b4f8a5363aff7e63ab8027b4f

Fixed issue with network interfaces due to a missing return value in the
"nw_path_enumerate_interfaces(...)" block. Exposed in iOS 18,
RTCNetworkMonitor::initWithObserver will only enumerate the first
interface, instead of all device interfaces

Bug: webrtc:359245764
Change-Id: Ifb9f28c33306c0096476a4afb0cdb4d734e87b2c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/359541
Auto-Submit: Corby <[email protected]>
Commit-Queue: Jonas Oreland <[email protected]>
Reviewed-by: Kári Helgason <[email protected]>
Reviewed-by: Jonas Oreland <[email protected]>
Cr-Commit-Position: refs/heads/main@{#42818}

Co-authored-by: Corby Hoback <[email protected]>

* Custom audio input for Android (#154)

# Conflicts:
#	sdk/android/api/org/webrtc/audio/JavaAudioDeviceModule.java
#	sdk/android/src/java/org/webrtc/audio/WebRtcAudioRecord.java

---------

Co-authored-by: CloudWebRTC <[email protected]>
Co-authored-by: Hiroshi Horie <[email protected]>
Co-authored-by: davidliu <[email protected]>
Co-authored-by: Guy Hershenbaum <[email protected]>
Co-authored-by: Corby Hoback <[email protected]>

Author: 6 people

.gitignore

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

AUTHORS

@@ -33,6 +33,7 @@ Christophe Dumez <[email protected]>
 Chris Tserng <[email protected]>
 Cody Barnes <[email protected]>
 Colin Plumb
+Corby Hoback <[email protected]>
 Cyril Lashkevich <[email protected]>
 CZ Theng <[email protected]>
 Danail Kirov <[email protected]>

DEPS

@@ -54,7 +54,7 @@ deps = {
   'src/base':
     'https://chromium.googlesource.com/chromium/src/base@738cf0c976fd3d07c5f1853f050594c5295300d8',
   'src/build':
-    'https://chromium.googlesource.com/chromium/src/build@cab574b350bc82dc3e7a1f634fedeb3079bf9e9d',
+    'https://github.com/webrtc-sdk/build@6978bac6466311e4bee4c7a9fd395faa939e0fcd',
   'src/buildtools':
     'https://chromium.googlesource.com/chromium/src/buildtools@5eb927f0a922dfacf10cfa84ee76f39dcf2a7311',
   # 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.
+
+#####################################################################################

api/BUILD.gn

@@ -368,6 +368,7 @@ 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 = [