Commit
This commit does not belong to any branch on this repository, and may belong to a fork outside of the repository.
* 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]>
- Loading branch information
6 people
authored
Nov 22, 2024
1 parent
52c1e68
commit 25e8120