PSA: WebRTC M63 Release Notes

[Anatoli Davidson]

M63

WebRTC M63 branch (cut at r20237)

Summary

WebRTC M63, currently available in Chrome's beta channel and as native libraries for Android and iOS, contains over 15 new features and over 50 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

Removing support for DTLS to SDES fallback

Applying a session description with both “a=fingerprint” (for DTLS) and “a=crypto” (for SDES) will now fail; the offerer must decide on one or the other. Support for SDES itself is not being removed yet. See this announcement for more details.

Workaround for Chrome Microphone Issues

A workaround for the macOS missing audio issue has landed (webrc:4799) in Chrome and is currently running under an experiment. Details of the issue have been submitted to Apple with the aim of fixing the underlying problems. The current workaround only applies to macOS 10.10 and newer.

Features

Injectable Audio Codecs

As of M63, webrtc::CreatePeerConnectionFactory() accepts audio encoder and decoder factory arguments; WebRTC will use the supplied factories when creating audio encoder and decoder objects. The point is that the factories determine the set of supported codecs, so it is now possible for applications to support codecs that aren’t included in the WebRTC code base, and to omit support for codecs that are included. See this announcement for details.

Build WebRTC with DataChannel only

webrtc::CreateModularPeerConnectionFactory() is available and it can be used to build WebRTC with DataChannel support only. It accepts MediaEngineInterface, CallFactoryInterface and RtcEventLogFactoryInterface as arguments. If the application knows it won't use certain modules, it can pass in null pointers for specific modules and omit the corresponding modules from its build. See an example of usage in webrtc:7613.

Deprecations

Legacy VoiceEngine

Most of the legacy VoiceEngine interfaces have been removed: VoENetwork, VoERTP_RTCP, VoECodec and VoEFile, as well as unused methods from VoEBase, and the VoiceEngineObserver callback interface. As a result, the internal classes voe::OutputMixer and AudioConferenceMixer were no longer required so they have been removed too.

Platform	Issue	Description	Component
Native	7306	Removed AudioDeviceObserver and make ADM not inherit from the Module interface.	Audio

Features and Bugfixes

Chrome

Type	Issue	Description	Component
Feature	8201	Add post processing interface to audio processing module	Audio
Feature	8246	Implement RTCMediaStreamTrackStats.concealmentEvents.	Audio
Feature	8281	Implement RTCMediaStreamTrackStats.jitterBufferDelay.	Audio
Feature	8127	Better slicing for calls for analyzing the benefits of audio network adaptor.	Audio
Feature	8250	AEC3 buffering issues due to clockdrift and audio pipeline glitches should be more visible	Audio
Feature	5118	Unify logging mechanisms.	Audio, Video
Feature	7613	Build WebRTC with DataChannel only	Build
Feature	8160	Allow external implementations of TaskQueues	Internals
Feature	8032	Slice receive stats on simulcast id and ALR experiment group	Stats, Video
Feature	8174	Report received content_type in old getstats from recieve statstics proxy.	Stats, Video
Feature	8178	Implement new SentToInputFpsRatioPercent send-side UMA metric.	Stats, Video
Feature	8326	It should be possible to change the complexity of the frames generated by the square generator.	Video
Feature	7874	Refactor FrameBuffer2 to use unwrapped picture ids.	Video
Bug	8223	Switching to conservative mode after timeout mode causes excessive ducking	Audio
Bug	8224	The size of the allowed jitter in EchoCanceller3 reduces the glitch recovery time	Audio
Bug	8320	AEC3 leaks echoes for speech with strong high-frequency content initially in the calls	Audio
Bug	8322	The AEC3 transparency to nearend is sometimes low	Audio
Bug	8358	On systems with a known echo path delay, the AEC3 would benefit from a specified initial delay value	Audio
Bug	8359	Suboptimal NLP gains for the nonlinear echo power estimate	Audio
Bug	8360	The amount of echo being masked by the nearend signal is sometimes underestimated.	Audio
Bug	8361	The AEC3 NLP gain is applied in a too abrupt manner	Audio
Bug	8378	The transparency in AEC3 is sometimes not ideal	Audio
Bug	8379	AEC3 fails to detect the echo path delay in reverberant environments with moderate echo return loss	Audio
Bug	8397	AEC3 sometimes fails to detect a valid linear filter estimate in reverberant environments	Audio
Bug	8398	Too much echo suppression initially in the calls for AEC3	Audio
Bug	769894	Chrome extension desktopCapture for tabs regression	Blink>GetUserMedia
Bug	761990	Suspend/resume can cause MediaStream track onended event to be fired.	Blink>GetUserMedia>Mic, Blink>WebRTC>Audio
Bug	763761	Switching to conservative mode after timeout mode causes excessive ducking	Blink>WebRTC>Audio
Bug	763775	The size of the allowed jitter in EchoCanceller3 reduces the glitch recovery time	Blink>WebRTC>Audio
Bug	765242	The echo canceller3 alignment functionality may get stuck in a noncausal state during capture data loss	Blink>WebRTC>Audio
Bug	8251	Video sometimes freezes	Network>RTP
Bug	5986	AudioEncoderOpus - invalid read	Audio
Bug	6861	WebRTC generates often this error "AcmReceiver::SetExtraDelay" failed: delay_ms=10000	Audio
Bug	7554	Expand rate may be wrong in some cases.	Audio
Bug	8041	WebRTC Mac app enable audio fail	Audio
Bug	8175	On some platforms, the allowed API call jitter in AEC3 is too low, resulting in excessive resets	Audio
Bug	770246	Improve reading SkImage pixels in canvas captureStream()	Blink>Canvas, Blink>MediaStream> CaptureFromElement
Bug	732224	WebRTC screen sharing issue	Blink>GetUserMedia> Desktop
Bug	741618	RTCPeerConnection.getRemoteStreams() should return all streams of all receivers	Blink>WebRTC> PeerConnection
Bug	741619	RTCPeerConnection should not rely on remote stream added/remove events	Blink>WebRTC> PeerConnection
Bug	7826	Certain functions in modules/pacing and modules/congestion_controller should be callable by multiple threads.	Cleanup
Bug	8157	Window border cannot be captured when window is not in foreground on Windows 8+	DesktopCapture
Bug	8325	Potential race condition in AudioSendStream constructor	Internals
Bug	8142	Config events lost while logging to file if the file is full	Internals
Bug	8143	RtcEventLogHelperThread::StartLogFile() can skip events	Internals
Bug	8145	Migrate RtcEventLog to use a task-queue rather than a helper thread	Internals
Bug	8188	TaskQueue::PostTask() overloading confused when posting subclass of PostedTask	Internals
Bug	8260	OnRemoveTrack	Internals, PeerConnection
Bug	7996	WebRTC send NACK request at constant period 100ms if remote no RTCP ReceiveReport send	Network>RTP
Bug	8335	RtpPacketizerVp8 causes hard to spot issues for incorrect input	Network>RTP, Video
Bug	8332	Existing RTX codec added to the end of the list when APT codec's payload type was remapped	PeerConnection
Bug	8266	Stop supporting fallback from DTLS to SDES.	PeerConnection
Bug	8253	The concealedSamples counter should not go down.	Stats
Bug	8162	Fix temporal layer flags used for vp8 with 4 layers	Video
Bug	8070	x-google-min-bitrate doesn't seem to work with H264	Video
Bug	5677	Remote Java MediaStream isn't updated when tracks are added/removed via SRD	PeerConnection

Native Android/iOS

Type	Issue	Description	Component
Feature	7825	Add a scoped smart pointer type for CoreFoundation objects	Mobile (iOS, Mac)
Feature	7749	Android: Support RGB texture frames	Mobile (Android)
Feature	761472	Add support in Android GlShader for uploads of vertex arrays with non-zero stride.	Blink>WebRTC (Android)
Bug	761336	Win7 Chrome can't decode H264 stream from Android Chrome	Blink>WebRTC>Video (Android)
Bug	8306	Memory leak in RTCVideoEncoderH264	Mobile (iOS)
Bug	8308	Memory leak in RTCMTLNV12Renderer	Mobile (iOS0
Bug	8303	Byte buffer objects returned by getDataY/U/V are shared between threads unsafely.	Mobile (Android)
Bug	8217	Android AAR generation is broken by Java 8	Mobile (Android)
Bug	8220	iOS video decoding fails after switching to background and coming back to active	Mobile, Video (iOS)
Bug	8302	Hangs in VTDecompressionSessionInvalidate in iOS 11.	Mobile, Video (iOS)
Bug	8256	Continual gathering logic isn't regathering candidate when "getifaddrs" thinks interface is active but ConnectivityManager doesn't	Network>ICE (Android)

[Archer Chen]

I am so sorry to reply in here. But if possible, can someone help me answer How to build WebRTC with DataChannel only in iOS ? Thank you in advance.