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(Java) Add support for resolution alignment during encoding #25

Merged
merged 10 commits into from
Mar 26, 2024
Merged

(Java) Add support for resolution alignment during encoding #25

merged 10 commits into from
Mar 26, 2024

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kanat
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@kanat kanat commented Mar 26, 2024 •
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This PR converts Kotlin classes to Java classes, cause Java is compatible with WebRTC build system. Additionally new classes were added to BUILD.gn

Original PR:

Original Description

This PR adds two new encoder factories:

  • SimulcastAlignedVideoEncoderFactory
  • DefaultAlignedVideoEncoderFactory

These should be generally always preferred to be used instead of the default SimulcastVideoEncoderFactory and DefaultVideoEncoderFactory. The difference is that the new factories allow to specify a resolution alignment requirement (resolutionAdjustmentin the constructor). HW encoders for VP8 (and H264) require a 16x16 aligned resolutions.

This code is based on the original source from:
https://github.com/shiguredo/sora-android-sdk/blob/3cc88e806ab2f2327bf3042072e98d6da9df4408/sora-android-sdk/src/main/kotlin/jp/shiguredo/sora/sdk/codec/HardwareVideoEncoderWrapperFactory.kt
Credit and thanks go to Shiguredo, Inc.

This is especially a problem while simulcasting - webrtc computes the resolutions based on the scaleResolutionDownBy defined in RtpParameters.Encodings and these resolutions are not 16x16 and this leads to various hard to debug problems with the encoder. WebRTC allows you to override the getEncoderInfo() in HardwareVideoEncoder.java and define the alignment but in most cases it will not crop the resolution and instead it will just adjust the scaleResolutionDownBy you defined. This behaviour is defined here. So from 1, 2, 4 (full, half, quarter resolution) it can create 1, 1, 1 to match the alignment and this is not a correct approach. In our tests the encoder would not work correctly and would drop frames in simulcast scenarios (1280x720, 3 simulcast layers - 1, 2, 4. This breaks if you are stream with phone oriented to left-orientation on Pixel devices).

Relevant issues:

@kanat kanat mentioned this pull request Mar 26, 2024
Merged
@kanat kanat force-pushed the resolution_alignment_java branch from 14f694b to 20f3203 Compare March 26, 2024 04:18
@kanat kanat merged commit c3b9942 into patch/m118 Mar 26, 2024
@kanat kanat deleted the resolution_alignment_java branch March 26, 2024 20:35
kanat added a commit that referenced this pull request Mar 26, 2024
@kanat
Android improvements. (#20)
e9891d8 
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)

Exposing Adapter types in PeerConnectionFactory (#78)

Co-authored-by: davidliu <[email protected]>
Co-authored-by: Mohamed Risaldar UT <[email protected]>
(cherry picked from commit e91f003)

# Conflicts:
#	media/base/media_channel.h
#	media/engine/webrtc_video_engine.cc
#	media/engine/webrtc_video_engine.h
santhoshvai pushed a commit that referenced this pull request Nov 20, 2024
@kanat @santhoshvai
Android improvements. (#20)
2e753c1 
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)

Exposing Adapter types in PeerConnectionFactory (#78)

Co-authored-by: davidliu <[email protected]>
Co-authored-by: Mohamed Risaldar UT <[email protected]>
(cherry picked from commit e91f003)

# Conflicts:
#	media/base/media_channel.h
#	media/engine/webrtc_video_engine.cc
#	media/engine/webrtc_video_engine.h
santhoshvai pushed a commit that referenced this pull request Nov 20, 2024
@kanat @santhoshvai
(Java) Add support for resolution alignment during encoding (#25)
e4cbe5a 
* (Java) Add support for resolution alignment during encoding

* rename native to delegate

* extract HardwareVideoEncoderWrapperFactory

* change class order

* add missing import

* compile fixes

* fix logging

* fix unreachable return statement

* fix dependencies

* move ResolutionAdjustment to video_java
santhoshvai pushed a commit that referenced this pull request Nov 20, 2024
@cloudwebrtc @hiroshihorie
@davidzhao @davidliu @graszka22 @theomonnom
Update to m125. (#119)
46226b5 
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]>
santhoshvai pushed a commit that referenced this pull request Nov 20, 2024
@cloudwebrtc @santhoshvai
Update to m125. (#119)
61fc0e6 
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
kanat pushed a commit that referenced this pull request Nov 22, 2024
@santhoshvai @cloudwebrtc
@hiroshihorie @davidliu
Sync with livekit's m125 (#42)
25e8120 
* 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]>
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@kanat