Stop recording on mute (turn off mic indicator) #55
Conversation
This file contains hidden or bidirectional Unicode text that may be interpreted or compiled differently than what appears below. To review, open the file in an editor that reveals hidden Unicode characters.
Learn more about bidirectional Unicode characters
cloudwebrtc
pushed a commit
that referenced
this pull request
Jan 18, 2023
* initial impl * more comments * more comment * adjust indent * comments
cloudwebrtc
pushed a commit
that referenced
this pull request
Jun 6, 2023
* initial impl * more comments * more comment * adjust indent * comments
cloudwebrtc
added a commit
that referenced
this pull request
Jun 12, 2023
allow listen-only mode in AudioUnit, adjust when category changes release mic when category changes (#5) Change defaults to iOS defaults (#7) Sync audio session config (#8) * use `AVAudioSession` defaults * remove isRecordingEnabled feat: support bypass voice processing for iOS. (#15) Remove MacBookPro audio pan right code (#22) fix: Fix can't open mic alone when built-in AEC is enabled. (#29) feat: add audio device changes detect for windows. (#41) * feat: add audio device changes detect for windows. * Update audio_device_core_win.cc fix iOS/macOS/Android compile. fix Linux compile (#47) AudioUnit: Don't rely on category switch for mic indicator to turn off (#52) * progress * tweak * clean * simplify audio unit restart call to SetupAudioBuffersForActiveAudioSession() might not be needed since sample rate won't change during restart. This might help reduce the unwanted noise when restarting audio unit. * clean Stop recording on mute (turn off mic indicator) (#55) * initial impl * more comments * more comment * adjust indent * comments Cherry pick audio selection from m97 release (#35) * [Mac] Allow audio device selection (#21) * first attempt * remove unused dep * init playout / recording * use AudioDeviceID as guid * switch device method * equality * default device * `isDefault` property * dont format default device name * type param * bypass * refactor * fix * append Audio to thread labels * ref * lk headers * low level apis * fix thread checks Some methods of ADM needs to be run on worker thread, otherwise RTC's thread check will fail. * switch to default device when removed * close mixerManager if didn't switch to default device * default audio device switched * expose devices update handler * fix ios compile * fix bug: don't always recreate RTCAudioDeviceModule * handle guid. Co-authored-by: Hiroshi Horie <[email protected]>
cloudwebrtc
added a commit
that referenced
this pull request
Jun 12, 2023
allow listen-only mode in AudioUnit, adjust when category changes (#2) release mic when category changes (#5) Change defaults to iOS defaults (#7) Sync audio session config (#8) feat: support bypass voice processing for iOS. (#15) Remove MacBookPro audio pan right code (#22) fix: Fix can't open mic alone when built-in AEC is enabled. (#29) feat: add audio device changes detect for windows. (#41) fix Linux compile (#47) AudioUnit: Don't rely on category switch for mic indicator to turn off (#52) Stop recording on mute (turn off mic indicator) (#55) Cherry pick audio selection from m97 release (#35) [Mac] Allow audio device selection (#21) Co-authored-by: Hiroshi Horie <[email protected]> Co-authored-by: David Zhao <[email protected]>
cloudwebrtc
added a commit
that referenced
this pull request
Jun 12, 2023
allow listen-only mode in AudioUnit, adjust when category changes (#2) release mic when category changes (#5) Change defaults to iOS defaults (#7) Sync audio session config (#8) feat: support bypass voice processing for iOS. (#15) Remove MacBookPro audio pan right code (#22) fix: Fix can't open mic alone when built-in AEC is enabled. (#29) feat: add audio device changes detect for windows. (#41) fix Linux compile (#47) AudioUnit: Don't rely on category switch for mic indicator to turn off (#52) Stop recording on mute (turn off mic indicator) (#55) Cherry pick audio selection from m97 release (#35) [Mac] Allow audio device selection (#21) Co-authored-by: Hiroshi Horie <[email protected]> Co-authored-by: David Zhao <[email protected]>
Merged
cloudwebrtc
added a commit
that referenced
this pull request
Jun 12, 2023
allow listen-only mode in AudioUnit, adjust when category changes (#2) release mic when category changes (#5) Change defaults to iOS defaults (#7) Sync audio session config (#8) feat: support bypass voice processing for iOS. (#15) Remove MacBookPro audio pan right code (#22) fix: Fix can't open mic alone when built-in AEC is enabled. (#29) feat: add audio device changes detect for windows. (#41) fix Linux compile (#47) AudioUnit: Don't rely on category switch for mic indicator to turn off (#52) Stop recording on mute (turn off mic indicator) (#55) Cherry pick audio selection from m97 release (#35) [Mac] Allow audio device selection (#21) Co-authored-by: Hiroshi Horie <[email protected]> Co-authored-by: David Zhao <[email protected]>
cloudwebrtc
added a commit
that referenced
this pull request
Jul 12, 2023
allow listen-only mode in AudioUnit, adjust when category changes (#2) release mic when category changes (#5) Change defaults to iOS defaults (#7) Sync audio session config (#8) feat: support bypass voice processing for iOS. (#15) Remove MacBookPro audio pan right code (#22) fix: Fix can't open mic alone when built-in AEC is enabled. (#29) feat: add audio device changes detect for windows. (#41) fix Linux compile (#47) AudioUnit: Don't rely on category switch for mic indicator to turn off (#52) Stop recording on mute (turn off mic indicator) (#55) Cherry pick audio selection from m97 release (#35) [Mac] Allow audio device selection (#21) RTCAudioDeviceModule.outputDevice / inputDevice getter and setter (#80) Co-authored-by: Hiroshi Horie <[email protected]> Co-authored-by: David Zhao <[email protected]>
cloudwebrtc
added a commit
that referenced
this pull request
Jul 13, 2023
allow listen-only mode in AudioUnit, adjust when category changes (#2) release mic when category changes (#5) Change defaults to iOS defaults (#7) Sync audio session config (#8) feat: support bypass voice processing for iOS. (#15) Remove MacBookPro audio pan right code (#22) fix: Fix can't open mic alone when built-in AEC is enabled. (#29) feat: add audio device changes detect for windows. (#41) fix Linux compile (#47) AudioUnit: Don't rely on category switch for mic indicator to turn off (#52) Stop recording on mute (turn off mic indicator) (#55) Cherry pick audio selection from m97 release (#35) [Mac] Allow audio device selection (#21) RTCAudioDeviceModule.outputDevice / inputDevice getter and setter (#80) Co-authored-by: Hiroshi Horie <[email protected]> Co-authored-by: David Zhao <[email protected]>
cloudwebrtc
added a commit
that referenced
this pull request
May 20, 2024
allow listen-only mode in AudioUnit, adjust when category changes (#2) release mic when category changes (#5) Change defaults to iOS defaults (#7) Sync audio session config (#8) feat: support bypass voice processing for iOS. (#15) Remove MacBookPro audio pan right code (#22) fix: Fix can't open mic alone when built-in AEC is enabled. (#29) feat: add audio device changes detect for windows. (#41) fix Linux compile (#47) AudioUnit: Don't rely on category switch for mic indicator to turn off (#52) Stop recording on mute (turn off mic indicator) (#55) Cherry pick audio selection from m97 release (#35) [Mac] Allow audio device selection (#21) RTCAudioDeviceModule.outputDevice / inputDevice getter and setter (#80) Co-authored-by: Hiroshi Horie <[email protected]> Co-authored-by: David Zhao <[email protected]>
cloudwebrtc
added a commit
that referenced
this pull request
May 20, 2024
allow listen-only mode in AudioUnit, adjust when category changes (#2) release mic when category changes (#5) Change defaults to iOS defaults (#7) Sync audio session config (#8) feat: support bypass voice processing for iOS. (#15) Remove MacBookPro audio pan right code (#22) fix: Fix can't open mic alone when built-in AEC is enabled. (#29) feat: add audio device changes detect for windows. (#41) fix Linux compile (#47) AudioUnit: Don't rely on category switch for mic indicator to turn off (#52) Stop recording on mute (turn off mic indicator) (#55) Cherry pick audio selection from m97 release (#35) [Mac] Allow audio device selection (#21) RTCAudioDeviceModule.outputDevice / inputDevice getter and setter (#80) Co-authored-by: Hiroshi Horie <[email protected]> Co-authored-by: David Zhao <[email protected]>
cloudwebrtc
added a commit
that referenced
this pull request
May 21, 2024
allow listen-only mode in AudioUnit, adjust when category changes (#2) release mic when category changes (#5) Change defaults to iOS defaults (#7) Sync audio session config (#8) feat: support bypass voice processing for iOS. (#15) Remove MacBookPro audio pan right code (#22) fix: Fix can't open mic alone when built-in AEC is enabled. (#29) feat: add audio device changes detect for windows. (#41) fix Linux compile (#47) AudioUnit: Don't rely on category switch for mic indicator to turn off (#52) Stop recording on mute (turn off mic indicator) (#55) Cherry pick audio selection from m97 release (#35) [Mac] Allow audio device selection (#21) RTCAudioDeviceModule.outputDevice / inputDevice getter and setter (#80) Allow custom audio processing by exposing AudioProcessingModule (#85) Expose audio sample buffers for Android (#89) feat: add external audio processor for android. (#103) android: make audio output attributes modifiable (#118) Co-authored-by: Hiroshi Horie <[email protected]> Co-authored-by: David Zhao <[email protected]> Co-authored-by: davidliu <[email protected]>
cloudwebrtc
added a commit
that referenced
this pull request
May 21, 2024
allow listen-only mode in AudioUnit, adjust when category changes (#2) release mic when category changes (#5) Change defaults to iOS defaults (#7) Sync audio session config (#8) feat: support bypass voice processing for iOS. (#15) Remove MacBookPro audio pan right code (#22) fix: Fix can't open mic alone when built-in AEC is enabled. (#29) feat: add audio device changes detect for windows. (#41) fix Linux compile (#47) AudioUnit: Don't rely on category switch for mic indicator to turn off (#52) Stop recording on mute (turn off mic indicator) (#55) Cherry pick audio selection from m97 release (#35) [Mac] Allow audio device selection (#21) RTCAudioDeviceModule.outputDevice / inputDevice getter and setter (#80) Allow custom audio processing by exposing AudioProcessingModule (#85) Expose audio sample buffers for Android (#89) feat: add external audio processor for android. (#103) android: make audio output attributes modifiable (#118) Fix external audio processor sample rate calculation (#108) Expose remote audio sample buffers on RTCAudioTrack (#84) Fix memory leak when creating audio CMSampleBuffer #86 Co-authored-by: Hiroshi Horie <[email protected]> Co-authored-by: David Zhao <[email protected]> Co-authored-by: davidliu <[email protected]>
Merged
cloudwebrtc
added a commit
that referenced
this pull request
Jun 12, 2024
Use M125 as the latest version and migrate historical patches to m125 Patches Group: ## 1. Update README.md 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 7454824 * allow listen-only mode in AudioUnit, adjust when category changes (#2) * release mic when category changes (#5) * Change defaults to iOS defaults (#7) * Sync audio session config (#8) * feat: support bypass voice processing for iOS. (#15) * Remove MacBookPro audio pan right code (#22) * fix: Fix can't open mic alone when built-in AEC is enabled. (#29) * feat: add audio device changes detect for windows. (#41) * fix Linux compile (#47) * AudioUnit: Don't rely on category switch for mic indicator to turn off (#52) * Stop recording on mute (turn off mic indicator) (#55) * Cherry pick audio selection from m97 release (#35) * [Mac] Allow audio device selection (#21) * RTCAudioDeviceModule.outputDevice / inputDevice getter and setter (#80) * Allow custom audio processing by exposing AudioProcessingModule (#85) * Expose audio sample buffers for Android (#89) * feat: add external audio processor for android. (#103) * android: make audio output attributes modifiable (#118) * Fix external audio processor sample rate calculation (#108) * Expose remote audio sample buffers on RTCAudioTrack (#84) * Fix memory leak when creating audio CMSampleBuffer #86 ## 3. Simulcast/SVC support for iOS/Android. 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. 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 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. 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. 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. 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]>
npazkevich
pushed a commit
to npazkevich/webrtc
that referenced
this pull request
Jun 24, 2024
allow listen-only mode in AudioUnit, adjust when category changes (webrtc-sdk#2) release mic when category changes (webrtc-sdk#5) Change defaults to iOS defaults (webrtc-sdk#7) Sync audio session config (webrtc-sdk#8) feat: support bypass voice processing for iOS. (webrtc-sdk#15) Remove MacBookPro audio pan right code (webrtc-sdk#22) fix: Fix can't open mic alone when built-in AEC is enabled. (webrtc-sdk#29) feat: add audio device changes detect for windows. (webrtc-sdk#41) fix Linux compile (webrtc-sdk#47) AudioUnit: Don't rely on category switch for mic indicator to turn off (webrtc-sdk#52) Stop recording on mute (turn off mic indicator) (webrtc-sdk#55) Cherry pick audio selection from m97 release (webrtc-sdk#35) [Mac] Allow audio device selection (webrtc-sdk#21) RTCAudioDeviceModule.outputDevice / inputDevice getter and setter (webrtc-sdk#80) Co-authored-by: Hiroshi Horie <[email protected]> Co-authored-by: David Zhao <[email protected]>
hiroshihorie
added a commit
that referenced
this pull request
Oct 14, 2024
This reverts commit c0209ef.
santhoshvai
pushed a commit
to GetStream/webrtc
that referenced
this pull request
Nov 20, 2024
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
to GetStream/webrtc
that referenced
this pull request
Nov 22, 2024
* 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]>
hiroshihorie
added a commit
that referenced
this pull request
Dec 9, 2024
commit 102940838f859941a41b2f3062d4eb8a61be6414 Author: Hiroshi Horie <[email protected]> Date: Thu Dec 5 15:27:34 2024 +0700 audio engine 1 commit d31187b Author: Hiroshi Horie <[email protected]> Date: Thu Dec 5 15:26:38 2024 +0700 Connect voice engine mute to adm commit 3df68d4 Author: Hiroshi Horie <[email protected]> Date: Fri Oct 11 20:41:44 2024 +0900 Revert "Stop recording on mute (turn off mic indicator) (#55)" This reverts commit c0209ef. commit b99fd2c Author: Michael Sloan <[email protected]> Date: Mon Dec 2 18:59:22 2024 -0700 Use `rtc::ToString` instead of `std::to_string` in `SocketAddress::PortAsString()` (#156) Justification for this change is that `std::to_string` should be avoided as it uses the user's locale and calls to it get serialized, which is bad for concurrency. My actual motivation for this is quite bizarre. Before this change, with Zed's use of the LiveKit Rust SDK, I was getting connection strings that were not valid utf-8, instead having a port of `3\u0000\u0000\u001c\u0000`. I have not figured out how that could happen or why this change fixes it. commit 543121b Author: davidliu <[email protected]> Date: Wed Oct 30 20:33:46 2024 -0700 Custom audio input for Android (#154)
hiroshihorie
added a commit
that referenced
this pull request
Dec 10, 2024
This reverts commit c0209ef.
hiroshihorie
added a commit
that referenced
this pull request
Jan 29, 2025
commit 98dc0ac Author: Hiroshi Horie <[email protected]> Date: Thu Jan 23 03:27:20 2025 +0900 Rendering fix commit 345f8b7 Author: Hiroshi Horie <[email protected]> Date: Thu Jan 23 00:41:35 2025 +0900 Manual rendering commit be003d5 Author: Hiroshi Horie <[email protected]> Date: Wed Jan 22 12:03:18 2025 +0900 RTCAudioDeviceModuleDelegate commit 2babb14 Author: Hiroshi Horie <[email protected]> Date: Fri Jan 17 16:18:39 2025 +0900 Squashed recent improvements Pre initialize mode Pre initialize logic Persistent Checks Fix buffer logic Patch default input_mute state Buffer checks Start buffer on enable Delay estimate 0 Stop engine on interrupt Pass should_resume Silence warning Correct session config Fix state Start logic Misc Rem ses Rem ses2 State helper Minor patch Simplify Change stop create order Working state Ref State helpers commit 235da97 Author: Hiroshi Horie <[email protected]> Date: Tue Jan 14 15:52:32 2025 +0900 Squashed recent progress commit 6ba820c Author: Hiroshi Horie <[email protected]> Date: Tue Dec 31 01:57:09 2024 +0900 Squashed recent progress Fix adm selection Fixes Revert adm selection in audio_device_impl Rename IsManualRenderingMode Simplify pcm buffer delegate Fixes Fixes Ducking config Strip manual rendering logic Runtime-ducking config Fix compile Fix start recording Connect output Buffer logic Enable output when input is enabled commit 49ca1ee Author: Hiroshi Horie <[email protected]> Date: Sun Dec 22 04:32:07 2024 +0700 Check AGC commit 5bbeb48 Author: Hiroshi Horie <[email protected]> Date: Sat Dec 21 14:15:31 2024 +0700 Debug print audio graph commit 631126f Author: Hiroshi Horie <[email protected]> Date: Fri Dec 20 23:24:42 2024 +0700 Fix macOS vp commit e07b814 Author: Hiroshi Horie <[email protected]> Date: Tue Dec 17 12:28:55 2024 +0700 Clean up imports commit 1bdb158 Author: Hiroshi Horie <[email protected]> Date: Wed Dec 11 12:35:12 2024 +0700 Muted talker detection commit 0324b22 Author: Hiroshi Horie <[email protected]> Date: Mon Dec 16 21:11:45 2024 +0700 Rename AudioDeviceSink commit ed22ffb Author: Hiroshi Horie <[email protected]> Date: Tue Dec 17 01:11:23 2024 +0700 Move to private method commit db00fe4 Author: Hiroshi Horie <[email protected]> Date: Wed Dec 11 00:16:55 2024 +0700 Other audio ducking commit a7282bd Author: Hiroshi Horie <[email protected]> Date: Thu Dec 5 15:27:34 2024 +0700 AudioEngine commit d31187b Author: Hiroshi Horie <[email protected]> Date: Thu Dec 5 15:26:38 2024 +0700 Connect voice engine mute to adm commit 3df68d4 Author: Hiroshi Horie <[email protected]> Date: Fri Oct 11 20:41:44 2024 +0900 Revert "Stop recording on mute (turn off mic indicator) (#55)" This reverts commit c0209ef.
pblazej
pushed a commit
that referenced
this pull request
Jun 12, 2025
allow listen-only mode in AudioUnit, adjust when category changes (#2) release mic when category changes (#5) Change defaults to iOS defaults (#7) Sync audio session config (#8) feat: support bypass voice processing for iOS. (#15) Remove MacBookPro audio pan right code (#22) fix: Fix can't open mic alone when built-in AEC is enabled. (#29) feat: add audio device changes detect for windows. (#41) fix Linux compile (#47) AudioUnit: Don't rely on category switch for mic indicator to turn off (#52) Stop recording on mute (turn off mic indicator) (#55) Cherry pick audio selection from m97 release (#35) [Mac] Allow audio device selection (#21) RTCAudioDeviceModule.outputDevice / inputDevice getter and setter (#80) Allow custom audio processing by exposing AudioProcessingModule (#85) Expose audio sample buffers for Android (#89) feat: add external audio processor for android. (#103) android: make audio output attributes modifiable (#118) Fix external audio processor sample rate calculation (#108) Expose remote audio sample buffers on RTCAudioTrack (#84) Fix memory leak when creating audio CMSampleBuffer #86 Co-authored-by: Hiroshi Horie <[email protected]> Co-authored-by: David Zhao <[email protected]> Co-authored-by: davidliu <[email protected]> (cherry picked from commit 7454824) # Conflicts: # audio/audio_state.cc # call/audio_state.h # media/engine/webrtc_voice_engine.h # modules/audio_device/audio_device_impl.cc # modules/audio_device/include/audio_device.h # modules/audio_device/mac/audio_device_mac.cc # modules/audio_device/mac/audio_device_mac.h # sdk/objc/api/peerconnection/RTCAudioTrack+Private.h # sdk/objc/api/peerconnection/RTCPeerConnectionFactory+Native.h # sdk/objc/api/peerconnection/RTCPeerConnectionFactory.mm # sdk/objc/api/peerconnection/RTCPeerConnectionFactoryBuilder.mm # sdk/objc/components/audio/RTCAudioSessionConfiguration.m # sdk/objc/native/src/audio/audio_device_ios.mm # sdk/objc/native/src/audio/audio_device_module_ios.mm # sdk/objc/native/src/audio/voice_processing_audio_unit.mm
hiroshihorie
added a commit
that referenced
this pull request
Jul 10, 2025
commit 47b4b71 Author: Hiroshi Horie <[email protected]> Date: Thu Jul 10 17:01:34 2025 +0900 Simplify AudioCustomProcessingAdapter commit 09ec011 Author: Hiroshi Horie <[email protected]> Date: Thu Jul 10 15:26:06 2025 +0900 Fix typo commit 70272d0 Author: Hiroshi Horie <[email protected]> Date: Tue Jul 1 16:41:23 2025 +0900 Revert LK prefix commit 81a78b6 Merge: 316d768 bfcfa65 Author: Hiroshi Horie <[email protected]> Date: Sun Jun 29 19:57:16 2025 +0900 Merge branch 'm125_release' into hiroshi/livekit-m125-adm-audioengine commit bfcfa65 Author: Hiroshi Horie <[email protected]> Date: Sun Jun 29 19:55:16 2025 +0900 Fix audio frame sample rate (#175) Fix audio frame sample rate when no senders are attached commit 0df968e Author: Harsh Shandilya <[email protected]> Date: Wed Jun 25 22:49:45 2025 +0530 Fix missing `RTC_OBJC_TYPE` macro (#174) Specifying the `rtc_objc_prefix` flag fails the build for these symbols since the macro wasn't applied to them. commit 316d768 Author: Hiroshi Horie <[email protected]> Date: Tue Jun 24 16:53:10 2025 +0900 Change workaround sleep time to 0.1 sec commit daeb79d Author: Hiroshi Horie <[email protected]> Date: Sat May 24 01:05:49 2025 +0900 Audio device symbols commit 6e5f1c6 Author: Hiroshi Horie <[email protected]> Date: Sat May 24 00:50:03 2025 +0900 Fix logging commit 6a49562 Merge: 212f078 ed96590 Author: Hiroshi Horie <[email protected]> Date: Tue Jun 24 15:35:31 2025 +0900 Merge branch 'm125_release' into hiroshi/livekit-m125-adm-audioengine commit 212f078 Merge: a58a087 7ec4c03 Author: Hiroshi Horie <[email protected]> Date: Thu May 22 10:56:54 2025 +0900 Merge branch 'm125_release' into hiroshi/livekit-m125-adm-audioengine commit a58a087 Author: Hiroshi Horie <[email protected]> Date: Thu May 15 15:36:56 2025 +0900 macOS device patch 1 Format Log device name Default patch 1 Engine start bug workaround Change sleep time Restart engine only if stopped Patch Default device update count Recreate on device change commit a1bb19a Author: Hiroshi Horie <[email protected]> Date: Sat May 3 15:13:47 2025 +0900 Move back device config timing commit 6566dd4 Author: Hiroshi Horie <[email protected]> Date: Sat May 3 00:11:51 2025 +0900 Rollback logic commit fa864f3 Author: Hiroshi Horie <[email protected]> Date: Fri May 2 23:37:38 2025 +0900 Only update state if apply succeeds commit 9f27f86 Author: Hiroshi Horie <[email protected]> Date: Fri May 2 23:37:01 2025 +0900 Move set device logic earlier commit 21ee7f0 Author: Hiroshi Horie <[email protected]> Date: Thu Apr 10 23:56:28 2025 +0800 input mixer mute mode commit e114436 Author: Hiroshi Horie <[email protected]> Date: Thu Apr 3 18:25:09 2025 +0800 Unmute on stop recording commit a940515 Author: Hiroshi Horie <[email protected]> Date: Sat Mar 29 00:19:40 2025 +0800 Inline input output node check commit 07ae103 Author: Hiroshi Horie <[email protected]> Date: Thu Mar 27 17:48:32 2025 +0800 Specify adm type on pc init commit adfee74 Author: Hiroshi Horie <[email protected]> Date: Thu Mar 27 18:01:05 2025 +0800 Simplify pc factory init p3 commit 68ef845 Author: Hiroshi Horie <[email protected]> Date: Wed Mar 26 02:41:46 2025 +0800 Don't mute when removing audio stream commit 081cc0f Author: Hiroshi Horie <[email protected]> Date: Thu Apr 3 15:19:46 2025 +0800 Revert "Engine state transition" This reverts commit 821c91a. commit e6614c9 Author: Hiroshi Horie <[email protected]> Date: Thu Mar 20 18:06:37 2025 +0800 Fix state check commit 584715e Merge: 3c0a82e 1d5d3b8 Author: Hiroshi Horie <[email protected]> Date: Thu Mar 20 16:24:33 2025 +0800 Merge branch 'm125_release' into hiroshi/livekit-m125-adm-audioengine commit 3c0a82e Merge: 821c91a 762d567 Author: Hiroshi Horie <[email protected]> Date: Thu Mar 20 16:21:45 2025 +0800 Merge branch 'hiroshi/livekit-m125-adm-audioengine-fix-stop-crash' into hiroshi/livekit-m125-adm-audioengine commit 762d567 Author: Hiroshi Horie <[email protected]> Date: Thu Mar 20 16:19:44 2025 +0800 setMicrophoneMuted commit 1d91e93 Author: Hiroshi Horie <[email protected]> Date: Thu Mar 13 01:16:12 2025 +0900 Metal renderer scale patch commit 9dbf95c Author: Hiroshi Horie <[email protected]> Date: Tue Mar 11 23:39:42 2025 +0900 Safe node detach commit 821c91a Author: Hiroshi Horie <[email protected]> Date: Sun Mar 9 00:49:56 2025 +0900 Engine state transition Squashed: Rename AudioEngineState Refactor engine observer Refactor EngineStateTransition Move EngineStateTransition Refactor EngineState State as class commit 0b84894 Author: Hiroshi Horie <[email protected]> Date: Fri Mar 7 19:33:08 2025 +0900 adm return error code commit b1074f3 Author: Hiroshi Horie <[email protected]> Date: Thu Mar 6 05:57:11 2025 +0900 propagate error codes commit 01f1554 Author: Hiroshi Horie <[email protected]> Date: Thu Mar 6 05:18:29 2025 +0900 Fix buffer state check commit 4a5ac4c Author: Hiroshi Horie <[email protected]> Date: Wed Mar 5 18:29:27 2025 +0900 Fix device format error crash & return error Squashed: Buffer checks only when modify state success Create converter only when required Catch output device errors also Return error if input not available Define error codes ApplyEngineState ModifyEngineState commit a890367 Author: Hiroshi Horie <[email protected]> Date: Tue Feb 18 19:33:38 2025 +0900 set vp enabled property commit e2a567b Author: Hiroshi Horie <[email protected]> Date: Mon Feb 17 18:07:46 2025 +0900 Explicit Int16 conversion commit fa4cddc Author: Hiroshi Horie <[email protected]> Date: Sat Feb 15 03:36:28 2025 +0900 set audio device buffer from rtc format commit 6732a8c Author: Hiroshi Horie <[email protected]> Date: Fri Feb 14 16:59:58 2025 +0900 Fix input mixer connection count commit 37f9711 Author: Hiroshi Horie <[email protected]> Date: Mon Feb 10 20:59:53 2025 +0900 Initial unmute for restart mute mode commit 24117ac Author: Hiroshi Horie <[email protected]> Date: Mon Feb 10 10:22:29 2025 +0900 Refactor commit cbccb5d Author: Hiroshi Horie <[email protected]> Date: Sun Feb 9 17:54:35 2025 +0900 is microphone muted property commit d8903c8 Author: Hiroshi Horie <[email protected]> Date: Sun Feb 9 13:38:17 2025 +0900 is engine running property commit 712009d Author: Hiroshi Horie <[email protected]> Date: Sun Feb 9 00:27:48 2025 +0900 Fix state getters commit 4f65581 Author: Hiroshi Horie <[email protected]> Date: Mon Feb 10 21:15:38 2025 +0900 Simplify logic commit 40e7d20 Author: Hiroshi Horie <[email protected]> Date: Sat Feb 8 20:21:43 2025 +0900 Mute mode commit 2fd53d6 Author: Hiroshi Horie <[email protected]> Date: Sat Feb 8 00:31:41 2025 +0900 Fix audio frame sample rate when no senders commit 1b4bcce Author: Hiroshi Horie <[email protected]> Date: Fri Feb 7 22:03:45 2025 +0900 apm mute / unmute commit 0b96f9a Author: Hiroshi Horie <[email protected]> Date: Mon Feb 3 20:12:00 2025 +0900 Update io node connection methods commit 1460fb4 Author: Hiroshi Horie <[email protected]> Date: Fri Jan 31 02:29:52 2025 +0900 Catch exception at start commit fbb0ed7 Author: Hiroshi Horie <[email protected]> Date: Thu Jan 30 12:17:44 2025 +0900 Fix engine state commit 1c01168 Author: Hiroshi Horie <[email protected]> Date: Thu Jan 30 04:12:13 2025 +0900 Disable apm option manipulation for ios mac commit 2ba3fd6 Author: Hiroshi Horie <[email protected]> Date: Wed Jan 29 18:37:01 2025 +0900 macos device logic commit 7f0aadb Author: Hiroshi Horie <[email protected]> Date: Tue Jan 28 15:52:17 2025 +0900 Start fail workaround commit 151ac87 Merge: 83551f3 0397078 Author: Hiroshi Horie <[email protected]> Date: Tue Jan 28 11:42:46 2025 +0900 Merge branch 'hiroshi/livekit-m125' into hiroshi/livekit-m125-adm-audioengine commit 0397078 Merge: bbe4412 844bafa Author: Hiroshi Horie <[email protected]> Date: Tue Jan 28 11:41:47 2025 +0900 Merge branch 'm125_release' into hiroshi/livekit-m125 commit 83551f3 Author: Hiroshi Horie <[email protected]> Date: Tue Jan 28 11:36:18 2025 +0900 Update delegate nullable node for output commit 85a0628 Author: Hiroshi Horie <[email protected]> Date: Mon Jan 27 13:04:14 2025 +0900 Fix build commit d9c7165 Author: Hiroshi Horie <[email protected]> Date: Sun Jan 26 17:02:28 2025 +0900 Mac device commit 261126c Author: Hiroshi Horie <[email protected]> Date: Sun Jan 26 16:49:29 2025 +0900 AGC & AEC available commit 78c9425 Author: Hiroshi Horie <[email protected]> Date: Sun Jan 26 08:54:06 2025 +0900 Engine reconfigure commit 536e8ff Author: Hiroshi Horie <[email protected]> Date: Fri Jan 24 00:29:33 2025 +0900 Mac aec off by default AudioOptions initially false commit 7ece395 Author: Hiroshi Horie <[email protected]> Date: Sat Jan 25 07:06:43 2025 +0900 bypass & agc commit 591eb97 Author: Hiroshi Horie <[email protected]> Date: Fri Jan 24 05:06:52 2025 +0900 Re-wire manual audio input commit 917c720 Author: Hiroshi Horie <[email protected]> Date: Thu Jan 23 23:41:20 2025 +0900 Squashed commit of the following: commit 98dc0ac Author: Hiroshi Horie <[email protected]> Date: Thu Jan 23 03:27:20 2025 +0900 Rendering fix commit 345f8b7 Author: Hiroshi Horie <[email protected]> Date: Thu Jan 23 00:41:35 2025 +0900 Manual rendering commit be003d5 Author: Hiroshi Horie <[email protected]> Date: Wed Jan 22 12:03:18 2025 +0900 RTCAudioDeviceModuleDelegate commit 2babb14 Author: Hiroshi Horie <[email protected]> Date: Fri Jan 17 16:18:39 2025 +0900 Squashed recent improvements Pre initialize mode Pre initialize logic Persistent Checks Fix buffer logic Patch default input_mute state Buffer checks Start buffer on enable Delay estimate 0 Stop engine on interrupt Pass should_resume Silence warning Correct session config Fix state Start logic Misc Rem ses Rem ses2 State helper Minor patch Simplify Change stop create order Working state Ref State helpers commit 235da97 Author: Hiroshi Horie <[email protected]> Date: Tue Jan 14 15:52:32 2025 +0900 Squashed recent progress commit 6ba820c Author: Hiroshi Horie <[email protected]> Date: Tue Dec 31 01:57:09 2024 +0900 Squashed recent progress Fix adm selection Fixes Revert adm selection in audio_device_impl Rename IsManualRenderingMode Simplify pcm buffer delegate Fixes Fixes Ducking config Strip manual rendering logic Runtime-ducking config Fix compile Fix start recording Connect output Buffer logic Enable output when input is enabled commit 49ca1ee Author: Hiroshi Horie <[email protected]> Date: Sun Dec 22 04:32:07 2024 +0700 Check AGC commit 5bbeb48 Author: Hiroshi Horie <[email protected]> Date: Sat Dec 21 14:15:31 2024 +0700 Debug print audio graph commit 631126f Author: Hiroshi Horie <[email protected]> Date: Fri Dec 20 23:24:42 2024 +0700 Fix macOS vp commit e07b814 Author: Hiroshi Horie <[email protected]> Date: Tue Dec 17 12:28:55 2024 +0700 Clean up imports commit 1bdb158 Author: Hiroshi Horie <[email protected]> Date: Wed Dec 11 12:35:12 2024 +0700 Muted talker detection commit 0324b22 Author: Hiroshi Horie <[email protected]> Date: Mon Dec 16 21:11:45 2024 +0700 Rename AudioDeviceSink commit ed22ffb Author: Hiroshi Horie <[email protected]> Date: Tue Dec 17 01:11:23 2024 +0700 Move to private method commit db00fe4 Author: Hiroshi Horie <[email protected]> Date: Wed Dec 11 00:16:55 2024 +0700 Other audio ducking commit a7282bd Author: Hiroshi Horie <[email protected]> Date: Thu Dec 5 15:27:34 2024 +0700 AudioEngine commit d31187b Author: Hiroshi Horie <[email protected]> Date: Thu Dec 5 15:26:38 2024 +0700 Connect voice engine mute to adm commit 3df68d4 Author: Hiroshi Horie <[email protected]> Date: Fri Oct 11 20:41:44 2024 +0900 Revert "Stop recording on mute (turn off mic indicator) (#55)" This reverts commit c0209ef. commit bbe4412 Merge: 0aca080 f5243e3 Author: Hiroshi Horie <[email protected]> Date: Fri Jan 17 09:43:21 2025 +0900 Merge branch 'm125_release' into hiroshi/livekit-m125 commit 0aca080 Merge: d29d62c b99fd2c Author: Hiroshi Horie <[email protected]> Date: Mon Dec 9 17:12:33 2024 +0700 Merge branch 'm125_release' into hiroshi/livekit-m125 commit d29d62c Author: Hiroshi Horie <[email protected]> Date: Sat Oct 19 17:12:57 2024 +0900 Remove duplicate RTCCameraVideoCapturer init methods commit f50e159 Merge: 9742a13 cd6792e Author: Hiroshi Horie <[email protected]> Date: Sat Oct 19 16:57:24 2024 +0900 Merge branch 'm125_release' into livekit-prefixed-m125 commit 9742a13 Merge: 7c29b54 c38ce7f Author: Hiroshi Horie <[email protected]> Date: Sat Oct 19 16:25:03 2024 +0900 Merge branch 'm125_release' into livekit-prefixed-m125 commit 7c29b54 Merge: 6902a18 0ae5688 Author: Hiroshi Horie <[email protected]> Date: Fri Oct 18 04:34:17 2024 +0900 Merge branch 'm125_release' into livekit-prefixed-m125 commit 6902a18 Merge: 00fd89c 7662c43 Author: Hiroshi Horie <[email protected]> Date: Tue Sep 24 03:14:40 2024 +0900 Merge branch 'm125_release' into livekit-prefixed-m125 commit 00fd89c Merge: bd25079 3c17c96 Author: Hiroshi Horie <[email protected]> Date: Mon Sep 23 18:42:48 2024 +0900 Merge branch 'm125_release' into livekit-prefixed-m125 commit bd25079 Merge: f8f9dc1 cdc3bba Author: Hiroshi Horie <[email protected]> Date: Tue Sep 17 11:24:07 2024 +0900 Merge branch 'm125_release' into livekit-prefixed-m125 commit f8f9dc1 Merge: 67cf254 c852b0e Author: Hiroshi Horie <[email protected]> Date: Wed Aug 21 01:36:38 2024 +0900 Merge branch 'm125_release' into livekit-prefixed-m125 commit 67cf254 Author: Hiroshi Horie <[email protected]> Date: Wed Aug 14 01:48:35 2024 +0900 Prefix RTCDevice category commit d05816e Merge: 634b7d0 6bb47f5 Author: Hiroshi Horie <[email protected]> Date: Wed Aug 14 01:27:36 2024 +0900 Merge branch 'm125_release' into livekit-prefixed-m125 commit 634b7d0 Merge: 07d9a46 d1b814a Author: Hiroshi Horie <[email protected]> Date: Tue Jul 16 15:19:04 2024 +0800 Merge branch 'm125_release' into livekit-prefixed-m125 commit d1b814a Author: Hiroshi Horie <[email protected]> Date: Sun Jul 14 01:45:07 2024 +0900 Allow to pass in capture session to RTCCameraVideoCapturer commit 07d9a46 Merge: b6d07b8 7ddfc43 Author: Hiroshi Horie <[email protected]> Date: Tue Jul 9 15:06:55 2024 +0900 Merge branch 'm125_release' into livekit-prefixed-m125 commit b6d07b8 Merge: 8b1c7f3 432a28b Author: Hiroshi Horie <[email protected]> Date: Fri Jun 21 04:52:40 2024 +0900 Merge branch 'm125_release' into livekit-prefixed-m125 commit 8b1c7f3 Author: Hiroshi Horie <[email protected]> Date: Fri Jun 14 18:07:04 2024 +0900 LK prefixed framework commit aeef504 Author: Hiroshi Horie <[email protected]> Date: Fri Jun 14 16:53:16 2024 +0900 Network monitor always enabled commit fd6c13d Author: Hiroshi Horie <[email protected]> Date: Fri Jun 14 18:10:54 2024 +0900 Fix missing headers and Metal linking commit b1f993d Author: Hiroshi Horie <[email protected]> Date: Fri Jun 14 16:58:13 2024 +0900 Fix missing RTC_OBJC_TYPE macros
ipavlidakis
pushed a commit
to GetStream/webrtc
that referenced
this pull request
Jul 28, 2025
* 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]>
ipavlidakis
pushed a commit
to GetStream/webrtc
that referenced
this pull request
Jul 29, 2025
* 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]>
ipavlidakis
pushed a commit
to GetStream/webrtc
that referenced
this pull request
Sep 11, 2025
* 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]>
Sign up for free
to join this conversation on GitHub.
Already have an account?
Sign in to comment
Add this suggestion to a batch that can be applied as a single commit.
This suggestion is invalid because no changes were made to the code.
Suggestions cannot be applied while the pull request is closed.
Suggestions cannot be applied while viewing a subset of changes.
Only one suggestion per line can be applied in a batch.
Add this suggestion to a batch that can be applied as a single commit.
Applying suggestions on deleted lines is not supported.
You must change the existing code in this line in order to create a valid suggestion.
Outdated suggestions cannot be applied.
This suggestion has been applied or marked resolved.
Suggestions cannot be applied from pending reviews.
Suggestions cannot be applied on multi-line comments.
Suggestions cannot be applied while the pull request is queued to merge.
Suggestion cannot be applied right now. Please check back later.
Now, if all audio tracks are muted(disabled), recording will stop and mic indicator will turn off.
Unmuting(enabling) any audio track will restart audio recording.
This is possible because of #52
Works(tested) with iOS / macOS.
macOS:
https://user-images.githubusercontent.com/548776/204782648-db38668f-21d6-46ac-876a-728376b716f2.mov