M64
WebRTC M64 branch (cut at r20918)
WebRTC M64, currently available in Chrome's beta channel and as native libraries for Android and iOS, contains over 10 new features and over 40 bug fixes, enhancements and stability/performance improvements. As with previous releases, we encourage all developers to run versions of Chrome on the Canary, Dev, and Beta channels frequently and quickly report any issues found. Please take a look at this page, for some pointers on how to file a good bug report. The help we have received has been invaluable!
The Chrome release schedule can be found here.
Since Chrome M64, getUserMedia() returns error values in accordance with the Media Capture and Streams spec. More details can be found here.
Significant portions of the RTP Media API are shipping for spec-compliant ways of handling the streams and tracks of an RTCPeerConnection. The new APIs, while not fully featured yet (more details here), allow applications to move away from legacy addStream()/removeStream()/getLocalStreams().
Added to RTCPeerConnection: addTrack(), removeTrack(), getSenders() and ontrack.
New interface: RTCRtpSender (track attribute only).
New event: RTCTrackEvent (full support minus transceiver attribute).
As of commit pos 20913, api:video_frame_api_i420 in the C++ API has been split out of api:video_frame_api. Depend on the former if you include i420_buffer.h.
When send-side bandwidth estimation is enabled for a screen content stream, WebRTC will now better handle the fact that the encoder may not always make full use of the available bandwidth. In conjunction, the aggressiveness of the paced sender has been reduced. This may increase median latency, but should result in a significant reduction of large latency events.
No action (apart from negotiating send-side BWE) is required to use this feature.
Platform | Issue | Description | Component |
Chrome | GetHistogramName() implementation in WebRTC can be removed | Cleanup | |
Chrome | Replace deprecated FFmpeg APIs | Video | |
Chrome | Re-deprecate ffmpeg avcodec_decode_audio4 and avcodec_decode_video2 | Blink>WebRTC, Internals>Media>FFmpeg | |
Chrome | Remove unused WebRTC event logging API | Blink>WebRTC>Tools |
Type | Issue | Description | Component |
Feature | APM quality assessment toolbox | Audio | |
Feature | Add relevant metrics for AEC3 | Audio | |
Feature | Add new CreateAudioDeviceWithDataObserver API without id to audio_device_data_observer.h | Audio | |
Feature | Unify logging mechanisms. | Audio, Video | |
Feature | BitrateAllocation should differentiate between 0 and an unsignaled bitrate. | BWE | |
Feature | Reduce number of mechanisms to select the TaskQueue implementation to one | Internals | |
Feature | Implement OpenSSLCertificate::GetChain | Network>DTLS | |
Feature | Add interface that allows customizing stun messages sent by TurnPort | Network>ICE | |
Feature | Application-specific Extension in RtpPacketReceived | Network>RTP | |
Feature | Report 95th percentile of interframe delay to UMA | Stats, Video | |
Feature | Make getStats report aggregated metrics over shorter interval on receive side. | Stats, Video | |
Feature | Unflag addTrack, removeTrack, ontrack and RTCRtpSender with track | Blink>WebRTC>PeerConnection | |
Feature | Mojofication of audio output stream control IPC. | Internals>Media>Audio | |
Bug | The analog gain controller does not sufficiently reduce the gain for clipped echo | Audio | |
Bug | AEC3 sometimes fails to detect a valid linear filter estimate in reverberant environments | Audio | |
Bug | AEC3 fails to detect the echo path delay in reverberant environments with moderate echo return loss | Audio | |
Bug | AEC3 is not sufficiently transparent for headsets and setup with echo paths with low gain | Audio | |
Bug | AEC3 sometimes fails to cancel echoes for highly saturated echoe | Audio | |
Bug | Too much echo suppression initially in the calls for AEC3 | Audio | |
Bug | BWE stuck after setting a high start bitrate after the call has started. | BWE | |
Bug | SRTP failure when using multiple PeerConnectionFactory's | Network>RTP | |
Bug | Intermittent TrackStartError when opening camera | Blink>GetUserMedia>Webcam | |
Bug | Too much echo suppression initially in the calls for AEC3 | Blink>WebRTC>Audio | |
Bug | The echo canceller 3 sometimes changes delay estimates when it should not | Blink>WebRTC>Audio | |
Bug | AEC3 sometimes fails to detect a valid linear filter estimate in reverberant environments | Blink>WebRTC>Audio | |
Bug | The echo canceller 3 sometimes changes delay estimates when it should not | Blink>WebRTC>Audio | |
Bug | Deadlock in AudioDeviceLinuxALSA? | Audio | |
Bug | FEC Packets can cause NetEq to incorrectly detect a frame length change | Audio | |
Bug | Frame length changes can cause increased target buffer level in NetEq | Audio | |
Bug | Windows audio device excessive logging | Audio | |
Bug | Remote ntp time estimation is way off sometimes. | Network>RTP | |
Bug | SetRemoteDescription: Unify callbacks into one place | PeerConnection | |
Bug | In SrtpTransport, send_rtcp_session_ should call SetSend rather than SetRecv | PeerConnection | |
Bug | Wrong codec parameters are inserted into SDP answer | PeerConnection | |
Bug | Add networkType to RTCIceCandidateStats | Stats | |
Bug | Sent framerate statistics could be incorrect. | Video | |
Bug | VideoDecoderSoftwareFallbackWrapper should handle any InitDecode() failure | Video | |
Bug | CPU adaptation doesn't work in chrome for screenshare. | Video | |
Bug | Injectable SW codecs does not have ulpfec, red, or flexfec | Video | |
Bug | Do not cache device ID salts | Blink>GetUserMedia | |
Bug | [Desktop Capture] wrong cursor position during Window sharing on Linux | Blink>GetUserMedia>Desktop | |
Bug | [Desktop Capture] Inaccurate mouse cursor position when sharing window on Mac with retina screen | Blink>GetUserMedia>Desktop | |
Bug | Add I420 format option to CopyOutputRequest | Blink>GetUserMedia>Tab | |
Bug | Add pcm/float32 support to MediaRecorder | Blink>MediaRecording | |
Bug | RTCPeerConnection: Support remote MediaStreamTrack being added and removed from MediaStream and pc/sender/receiver | Blink>WebRTC>Network | |
Bug | WebRTC MediaStreamTrack not created after renegotiation when remote peer adds track to existing stream | Blink>WebRTC>PeerConnection | |
Bug | RTCPeerConnection: RTP Media API UseCounters | Blink>WebRTC>PeerConnection | |
Bug | RTCPeerConnection, MediaStream and MediaStreamTrack events should fire in the right order | Blink>WebRTC>PeerConnection | |
Bug | Mac OS X Playout device init fails when using a multi-output device | Audio | |
Bug | Incorrect resampler mode in Resampler::Reset() which causes invalid pointer access | Audio | |
Bug | New RTC event logs can override existing files | Blink>WebRTC>Tools | |
Bug | Data race in media::AudioOutputDevice::Initialize | Internals>Media>Audio |
Type | Issue | Description | Component |
Bug | RTCCameraVideoCapturer does not respect format | Mobile (iOS) | |
Bug | WebRTC causes Android to route non-WebRTC audio in undesirable ways, even when nothing is playing | Audio, Mobile (Android), PeerConnection | |
Bug | Generated JNI code fails for nested enums | Mobile (Android) | |
Bug | Android: HardwareVideoEncoderFactory reports incorrect profile level | Mobile (Android) | |
Bug | Define Codec name NSString constants in a central place for iOS SDK. | Mobile (iOS) | |
Bug | Objc interface for peer connection factory does not allow external audio device module to be used. | PeerConnection (iOS) | |
Bug | iOS 11 UIDeviceOrientationFaceUp resets video orientation | Mobile (iOS), Video | |
Bug | No IceCandidates are created for Wifi P2p (aka Wifi-Direct) addresses | Network>ICE, Mobile (Android) | |
Bug | RTCEAGLVideoView crashing when created in background (iOS SDK) | Video, Mobile (iOS) | |
Bug | RTCAudioSession is not using a KVO context | Mobile (iOS) |
PSA: addTrack(), removeTrack(), ontrack and getSenders()
Significant portions of the RTP Media API are shipping for spec-compliant ways of handling the streams and tracks of an RTCPeerConnection. The new APIs, while not fully featured yet (more details here), allow applications to move away from legacy addStream()/removeStream()/getLocalStreams().
Added to RTCPeerConnection: addTrack(), removeTrack(), getSenders() and ontrack.
New interface: RTCRtpSender (track attribute only).
New event: RTCTrackEvent (full support minus transceiver attribute).
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Good. Maybe Chrome team did a lot of hard work. Passion.
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This mode is in use when the value of the OPTIONAL packetization-mode MIME parameter is equal to 0, the packetization-mode is not present, or no other packetization mode is signaled by external means. All receivers MUST support this mode.
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Bug
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We are currently researching some development options using WebRTC. Part of the solution we would like develop is the ability to shift audio and video out of sync, e.a. deliberately create ‘lipsync’ issues with the video presented.
We want modify the existing google AppRTC code so that video playback in the browser is 500 milliseconds or more behind the audio playback. This is to be achieved by modifying relevant timestamps only but not sure how to do that
can any one help us?
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