Sync with livekit's m125 #42
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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 <548776+hiroshihorie@users.noreply.github.com> Co-authored-by: David Zhao <dz@livekit.io> Co-authored-by: davidliu <davidliu@deviange.net> Co-authored-by: Angelika Serwa <angelika.serwa@gmail.com> Co-authored-by: Théo Monnom <theo.monnom@outlook.com> # 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
reviewed
Nov 21, 2024
# Conflicts: # sdk/BUILD.gn
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
…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).
Expose initializers to pass in capture session to RTCCameraVideoCapturer so we can use AVCaptureMultiCamSession etc to capture front and back simultaneously for iOS.
…#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 <deadbeef@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42256} Co-authored-by: Guy Hershenbaum <hershi@meta.com>
TODO: - [x] fix compile for RTCCameraVideoCapturer - [ ] fix RTCMTLRenderer ? --------- Co-authored-by: Hiroshi Horie <548776+hiroshihorie@users.noreply.github.com>
TODO: - [x] Return `.systemPreferredCamera` for devices (visionOS only). - [x] Use `AVCaptureMultiCamSession` only if `isMultiCamSupported` is true. - [x] Silence statusBarOrientation warning. --------- Co-authored-by: duanweiwei1982@gmail.com <duanweiwei1982@gmail.com>
17.0+ only atm --------- Co-authored-by: cloudwebrtc <duanweiwei1982@gmail.com>
~Allow to use "googEchoCancellation" constraint for software AEC. For devices "googEchoCancellation" should be false to use VoiceProcessingIO.~
Instead of converting to Float, output original Int data without conversion. Output the raw format and convert when required.
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 <corby.hoback@gmail.com> Commit-Queue: Jonas Oreland <jonaso@webrtc.org> Reviewed-by: Kári Helgason <kthelgason@webrtc.org> Reviewed-by: Jonas Oreland <jonaso@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42818} Co-authored-by: Corby Hoback <corby.hoback@gmail.com>
# Conflicts: # sdk/android/api/org/webrtc/audio/JavaAudioDeviceModule.java # sdk/android/src/java/org/webrtc/audio/WebRtcAudioRecord.java
kanat
approved these changes
Nov 22, 2024
Merged
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cherry pick of webrtc-sdk/webrtc#119
and
cherry pick of these commits: webrtc-sdk/webrtc@46226b5...543121b
iOS
The iOS deployment target is now 13.0 - https://github.com/webrtc-sdk/Specs/blame/bb9b52fda9582a3a36e07879e7396100864e5044/WebRTC-SDK.podspec#L15
Building should be done using
./tools_webrtc/ios/build_ios_libs.py --deployment-target 13.0Android
Jdk 17 is required now.