PSA: WebRTC M60 Release Notes

[Anatoli Davidson]

M60

WebRTC M60 branch (cut at r18252)

Summary

WebRTC M60, 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.

Important PSAs

Video jitter buffer fixes

We've fixed two regressions related to the video jitter buffer, see issues 7518 and 7532 for details. These issues could lead to decoding artifacts or frozen video.

Issues with TCP + send-side bandwidth estimation fixed

In M57, a regression was introduced that affects send-side bandwidth estimation when used over TCP connections, resulting in a bitrate estimate lower than expected. In M60, we believe all the issues should be resolved and the fixes were also merged back to M59. See crbug 726962 for more details. Also, see this article for recommendations for working around the issue.

Features

Stopping audio track when audio input data is missing

A mechanism to detect missing input (mic) audio data has been implemented in Chrome. The audio track and its source is stopped and the ‘ended’ event is fired. See the details in the public PSA on the discuss-webrtc forum.

Updated VideoTrack labels for screen sharing

You can now find out if a screen share VideoTrack comes from Desktop, window or a tab by looking at the VideoTrack label. See crbug 724668 for details.

Exposed new, standards-compliant getStats implementation in Java API

You can now access the new stats implementation from the Java PeerConnection API. In addition to being valuable for obtaining standards-compliant or newly-implemented stats that wouldn’t be available otherwise, it makes some improvements over the previous version of the API, exposing values as their actual types (boolean, double, integer, etc.)

Deprecations

Platform	Issue	Description	Component
Chrome	7501	Delete MessageQueue::set_socketserver method and related locks.	Internals

Features and Bugfixes

Chrome

Type	Issue	Description	Component
Feature	722335	Fire "ended" event when mic is receiving no signal	Blink>WebRTC>Audio
Feature	7394	Opus cbr not supported	Audio
Feature	6533	Cannot change volume on RTCAudioTrack in WebRtc	Audio
Feature	725819	Remove Mac audio input restart mechanism	Blink>WebRTC>Audio
Feature	7433	Support dynamic size rtp header extension	Network>RTP
Feature	7516	Reduce the binary size of libjingle_peerconnection	PeerConnection
Feature	7669	Request keyframe if the first frame in the stream is not a keyframe.	Video
Feature	7389	Unity native plugin example	SampleApps
Feature	7404	Move aecdump file IO from real-time audio thread to low-prio task queue	Audio
Feature	724668	Label screenshare VideoTracks according to source	Blink>GetUserMedia>Desktop
Bug	5651	AudioEncoderOpus - unaddressable mem read	Audio
Bug	6444	Memcheck errors in Opus decoder	Audio
Bug	7610	NetEq target buffer level increases when audio probing packets are used	Audio
Bug	7631	Cache RTP state per SSRC throughout lifetime of Call	Audio, Network
Bug	7717	Send-side bandwidth estimation not working over TCP TURN connections	Network>ICE
Bug	704277	Tab capture always starts on max resolution and then scales down	Blink>GetUserMedia>Desktop
Bug	721387	Desktop picker not working well for multiple monitors on OSX	Blink>GetUserMedia>Desktop
Bug	710818	The WebRTC echo canceller 3 has poor transparency when the analog microphone gain is changed	Blink>WebRTC>Audio
Bug	712651	The WebRTC echo canceller 3 sometimes has poor transparency when there is weak, or no, echo	Blink>WebRTC>Audio
Bug	722343	The WebRTC echo canceller 3 setup does not work properly for stereo output	Blink>WebRTC>Audio
Bug	726962	Send-side bandwidth estimation not working over TCP connections	Blink>WebRTC>Network
Bug	715227	Refactor WebMediaPlayerMSCompositor::ReplaceCurrentFrameWithACopy() logic	Blink>WebRTC>Video
Bug	7509	SignalSentPacket not fired by TCPPort for outgoing TCP connections - breaks send-size bandwidth estimation for direct TCP connections (not through TURN server)	BWE, Network
Bug	7475	Maintain picture_id monotonicity between calls to Release and InitEncode, for all encoders	Video
Bug	7625	Padding packets are not properly handled by audio stream receiver.	Audio
Bug	7540	Race in voe::Channel destruction	Audio
Bug	6514	DelayBasedBwe should not use GetFeedbackInterval	Audio
Bug	7558	Make sure the result is floating point in InitLowFrequencyCorrectionRanges	Audio
Bug	7528	AEC3 sometimes attenuates low-powered regions of the near-end speech	Audio
Bug	7559	The echo canceller 3 introduces signal artefacts when there are buffering issues	Audio
Bug	7666	Modulus usage in the residual echo detector are causing the complexity to be higher than necessary.	Audio
Bug	7462	Invalid output buffer setting used within AudioCoder class	Audio
Bug	718082	Android screencapture: possible hangs in stopCapture()	Blink>GetUserMedia>Desktop
Bug	723889	Presenters mouse cursor is in the wrong place on viewers screen	Blink>GetUserMedia>Desktop
Bug	638664	MediaRecorder in Android not using Encode Accelerator for VP8/9	Blink>MediaRecording
Bug	711825	Avoid setting alpha mode for H264 webm container	Blink>MediaRecording
Bug	722720	Audio glitches on Posix machines when number of file descriptors are high	Blink>WebRTC>Audio
Bug	710760	ICE-Lite server re-offers with new ICE ufrag/pwd and Chrome generates wrong STUN requests with ICE-CONTROLLED	Blink>WebRTC>Network
Bug	711243	Chrome 57 does not send use_srtp extension in DTLS hello for webRTC datachannel	Blink>WebRTC>Network
Bug	712311	Using DXVA HW decoder for WebRTC fails	Blink>WebRTC>Video
Bug	705679	Rendering WebRTC simulcast from RTCVideoEncoder freezes between simulcast layers.	Blink>WebRTC>Video
Bug	7592	inet_pton_v6 doesn't handle invalid IPv6 addresses properly.	Network
Bug	7443	no ice-options:trickle in SDP	PeerConnection
Bug	7560	VideoReceiverStream::Stats render_frame_rate and decode_frame_rate may have stale values	Stats
Bug	7492	Scale counter and resolution downgrade stats may be incorrect.	Video
Bug	7535	We should not reinit the encoder when the frame rotation changes	Video
Bug	7590	Broken sender controlled receiver smoothing due to new jitter buffer in M58	Video
Bug	7518	Inserting same frame multiple times into FrameBuffer2 could cause a non-decodable stream.	Video
Bug	7520	Keyframe request spam.	Video
Bug	7532	Some H264 frames may cause infinite loop in Packet_buffer	Video
Bug	715893	The WebRTC echo canceller 3 sometimes attenuates low-powered portions of the near-end speech	Blink>WebRTC>Audio
Bug	5847	Unexpected error ERROR_CONTENT if audio codec is removed, then later added with a different payload type.	PeerConnection

Native Android/iOS

Type	Issue	Description	Component
Feature	7448	Make WebRtcAudioEffects reusable	Audio (Android)
Feature	7446	WebRTC should know when Apple actives the audio session through CallKit or other means	Audio, Mobile (iOS)
Feature	6871	Add Java shims to invoke the new GetStats API	Stats (Android)
Bug	7471	configureWebRTCSession/unconfigureWebRTCSession doesn't honor setActive contract	Mobile (iOS)
Bug	7636	AppRTCMobile cannot run without a TURN ICE server	Mobile (Android)
Bug	6749	RTCCameraPreviewView doesn't rotate context after changing device orientation	Video, Mobile (iOS)