M62
WebRTC M62 branch (cut at r19592)
WebRTC M62, currently available in Chrome's beta channel and as native libraries for Android and iOS, contains over 10 new features and over 20 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.
We have launched prebuilt libraries for mobile development on Android and iOS. The easiest way to get started is using the libraries available at JCenter and cocoapods.org. These libraries are compiled from the tip-of-tree and are meant for development purposes only. The versioning system used is 1.1.cr-commit-position, where cr-commit-position can be used to identify the exact WebRTC revision the .aar/pod was built from. We intent to release the libraries on a weekly basis.
PeerConnection::SetBitrate limits the minimum and maximum bandwidth allocated for all RTP streams sent by a PeerConnection. Other limitations might affect these parameters and are respected (for example “b=AS” in SDP). ObjC and Java bindings are also provided. See webrtc:7395 for details.
The deprecated interface RTCAVFoundationVideoSource is now removed (in iOS). If you have an application that still uses RTCAVFoundationVideoSource, you are strongly encouraged to migrate ASAP. More details in the original PSA.
Platform | Issue | Description | Component |
Chrome | Remove deprecated code from DirectTransport | Network |
Type | Issue | Description | Component |
Feature | Merge to M62: Additional metrics reporting for WebRTC experiments. | Blink>WebRTC>Video | |
Feature | Implement RTP keep-alive | ORTC, Video | |
Feature | APM quality assessment toolbox | Audio | |
Feature | Add audio call duration UMA metric based on packet traffic | Audio, Stats | |
Feature | Add a bursty frame generator that doesn't rely on files | Video | |
Feature | Report end-to-end delay in UMA separately for screenshare and realtime video. | Stats, Video | |
Feature | Implement RTCMediaStreamTrackStats.totalSamplesReceived and concealedSamples, used to calculate "expand rate". | Stats, Audio | |
Bug | Slow recovery in AEC3 for some variants of capture buffer issues | Audio | |
Bug | Internal corruption to incoming VP8 stream | Video | |
Bug | Thread-checkers fail on UT | Audio | |
Bug | Packet discard rate unimplemented in NetEq. | Audio | |
Bug | "Enable diagnostic packet and event recording" checkbox state forgotten | Blink>WebRTC>Tools | |
Bug | REMB packets have audio SSRC | BWE | |
Bug | STUN / TURN Candidates not gathered when re-creating PeerConnection | Network>ICE | |
Bug | RTCP receiver should verify sender ssrc when parsing target bitrate | Network>RTP | |
Bug | OnNegotiationNeeded not called when track is added to stream | PeerConnection | |
Bug | Turn off error resilience when running without temporal and spatial layers for VP9. | Video | |
Bug | RtpVideoStreamReceiver::receive_cs_ suspected unnecessary and replaceable by ThreadChecker | Video | |
Bug | Remove or throttle logging from FrameBuffer2 | Video | |
Bug | VP9 SVC stream freeze. | Video | |
Bug | Don't clear newer packets from video_coding::PacketBuffer when calling ClearTo | Video |
Type | Issue | Description | Component |
Feature | Consider exposing EglBase14 and EglBase10 as part of the Android SDK API | Mobile (Android) | |
Feature | Support more formats in RTCVideoFrame | Mobile (iOS) | |
Feature | Injectable Obj-C video codecs | Mobile (iOS) | |
Feature | Refactor iOS video source API | Mobile, Video (iOS) | |
Feature | Expose RTCAudioSession.h in iOS SDK/Framework build to allow using manual audio | Mobile (iOS) | |
Feature | Add API for setting min/max BWE to PeerConnection and/or RtpTransportController | PeerConnection (Android, iOS) | |
Feature | Unity native plugin example | SampleApps (WIndows, Android) | |
Bug | rtc::CurrentThreadId wrong on Android | Mobile (Android) | |
Bug | SurfaceViewRenderer does not respond to changes of scaling type | Mobile (Android) | |
Bug | PeerConnectionFactory crashes if it is reinitialized without hardware codec support | Mobile (Android) | |
Bug | HardwareVideoDecoder stops working after a resolution change | Mobile (Android) | |
Bug | ObjcVideoTrackSource::OnCapturedFrame ignores cropX & cropY | Mobile (iOS) | |
Bug | Receiving video is momentarily shown upside down | Mobile (iOS) | |
Bug | H264 vt compression session is created with hardware accl disabled always on mac osx. | Mobile (iOS) | |
Bug | Setting field trial flags in AppRTCMobile not working | Mobile (iOS) | |
Bug | iOS AppRTCDemo doesn't call PeerConnection close() | Mobile (iOS) | |
Bug | iPhone fails to send and receive video on T-Mobile network (due to limit of 5 IPv6 interfaces) | Mobile, Network (iOS) | |
Bug | Issue on exiting the video call on Android | Mobile, PeerConnection (Android) | |
Bug | When network interface has multiple IPv6 addresses, and OS selects one WebRTC doesn't expect, the port is discarded | Network>ICE (Android) | |
Bug | SDP parsing will fail on AppRTCMobile when preferred codec is the last in the list. | SampleApps (iOS) | |
Bug | UIView.layer accessed off main thread. | Mobile (iOS) |
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