feat(m144 upgrade): patch from m137_release (#173)#217
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cloudwebrtc wants to merge 49 commits intom144_releasefrom
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feat(m144 upgrade): patch from m137_release (#173)#217cloudwebrtc wants to merge 49 commits intom144_releasefrom
cloudwebrtc wants to merge 49 commits intom144_releasefrom
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cloudwebrtc
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fixed VP9/AV1 encoder unusable. We have to apply this patch and publish new binary packages. Otherwise we can't use AV1/VP9 codec in Apple platform.
Co-authored-by: Hiroshi Horie <548776+hiroshihorie@users.noreply.github.com> Co-authored-by: Harsh Shandilya <122249239+harshs-dyte@users.noreply.github.com>
Remote audio goes through AudioProcessingModule now so it's possible to get the buffer.
Avoids crash at transceiver deinit that channel is not cleaned up. --------- Co-authored-by: Błażej Pankowski <86720177+pblazej@users.noreply.github.com>
TODO: - [x] Core Implementation - [x] Android JNI Wrapper - [x] Objective-C Wrapper --------- Co-authored-by: davidliu <davidliu@deviange.net>
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Adds a flag to override engine start/stop behavior, useful in scenarios such as CallKit, where the timing for starting AVAudioSession and AVAudioEngine is restricted.
Fixes crash when toggling audio rendering mode at run-time.
Check recording perm & AVAudioSession.category when enabling input for device rendering mode to avoid crashes.
… setup to prevent WebRTC deadlocks (#197) ## Description This PR makes two small but important changes to improve thread safety and reduce deadlock risk in the iOS bindings: ### 1. RTCAudioTrack.mm - Remove the explicit [self removeAllRenderers]. - Instead, iterate _adapters and remove each sink from the native audio track without switching threads, aligning the behavior with RTCVideoTrack.mm. - This simplifies teardown and avoids extra synchronization points. ### 2.RTCFrameCryptor.mm - Remove factory.signalingThread->BlockingCall around nativeRtpReceiver->SetDepacketizerToDecoderFrameTransformer(...). - The native call path already runs on the worker thread, so performing a BlockingCall from the signaling thread introduces an unnecessary signaling → worker hop that can increase deadlock risk in tight teardown/configuration sequences. ## Problem Solved - Redundant signaling → worker thread switching during audio renderer cleanup and frame-cryptor configuration could contribute to deadlocks under timing-sensitive conditions (e.g., track teardown overlapping with transformer setup). - By eliminating that extra hop, we reduce contention between signaling and worker threads without changing observable behavior. ## Impact - Lower likelihood of hangs/deadlocks on teardown/configuration paths. - Implementation now matches RTCVideoTrack.mm for audio cleanup. - No public API changes. --------- Co-authored-by: Hiroshi Horie <548776+hiroshihorie@users.noreply.github.com>
…oid (#205) These default values got changed from true to false in #178. On Android, the way the audio options are set is fairly janky, and actually relied on these initially defaulting to true. I'm not sure if other non iOS platforms rely on these initial default values as well, but that's something to be aware of for those platforms.
Alternative of #208
Set automaticallyConfiguresApplicationAudioSession to false just in case.
* Use enums for ducking level * Change default ducking level settings
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pblazej
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Feb 12, 2026
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| static webrtc::Environment CreateDefaultEnvironment() { | ||
| return webrtc::CreateEnvironment( | ||
| std::make_unique<webrtc::FieldTrials>("WebRTC-IceHandshakeDtls/Enabled/")); |
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I think there's no way to pass it on Apple vs Kotlin RTCModule.kt via PeerConnectionFactory.Builder.setFieldTrials(), maybe we should invent one to avoid hardcoding?
cc @davidliu
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🍏 AV1 encoder is failing: The hypothesis is dimensions mismatch, but I could not reproduce that with VP (same cases): Logs: Upstream commit: |
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