PSA: WebRTC M59 Release Notes

[Anatoli Davidson]

M59

WebRTC M59 branch (cut at r17657)

 

Summary

WebRTC M59, currently available in Chrome's beta channel and as native libraries for Android and iOS, contains 15 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

Security vulnerability in M58 fixed

A fix for a security vulnerability in M58 has been merged into M58 (available in Chrome 58.0.3029.96) and is also present in M59. See the details in the public PSA on the discuss-webrtc forum.

Features

ICE candidate pooling

Normally, ICE candidates are only gathered after RTCPeerConnection.setLocalDescription is called, because this is the point at which the PeerConnection knows how many media components it needs to gather candidates for. However, the iceCandidatePoolSize member of RTCConfiguration, implemented in M59, instructs the PeerConnection to gather ICE candidates before this occurs, in order to speed up call setup. For more information, see its description in the spec.

AEC3

A major update to the echo cancellation functionality (AEC3) is now available for testing. The main benefits with the new functionality are improved echo cancellation performance and more rapid recovery from issues in the audio buffer pipeline. For this release, AEC3 is still off by default and its performance will be evaluated in experiments to apply further tuning. In Chrome M59 it can be controlled via the Chrome flag “WebRTC Echo Canceller 3”.

AppRTCMobile can now be developed inside Android Studio

It is now possible to compile and debug AppRTCMobile inside Android Studio. WebRTC Java code can also be developed inside Android Studio using this approach. This gives access to Android Studio’s extensive development and debugging tools. More information on how to do this can be found here: https://webrtc.org/native-code/android/

Partial RTCRtpReceiver support with getContributingSources()

It is now possible to get the RTP packet contributing sources (CSRCs) of receiving tracks. RTCPeerConnection.getReceivers() has been added, which returns RTCRtpReceiver with .track and .getContributingSources() support. See spec.

More robust Bandwidth Estimation for cellular and WiFi connections.

To improve the robustness of the bandwidth estimator on cellular and WiFi networks we've replaced the approach previously used (described here) with a less complex filter based on estimating the trend of the queuing delay. In addition, we've changed the way we're estimating the throughput of the network to make it more stable to short periods of no or very little throughput, which could throw off the old mechanism. Our experiments show that the impact of delay transients is reduced. The rmcat draft will be updated later to reflect these changes in the algorithm.

Spec-compliant video constraints processing in getUserMedia

getUserMedia now supports the standard algorithm for processing video constraints, including support for 'ideal' values. See spec.

VP9 Denoiser enabled by default

The internal denoiser of VP9 has been enabled by default. This will lead to improved visual quality in low light scenarios, with only slightly higher CPU utilization. More information is available in the feature issue.

 

Deprecations

 

Platform	Issue	Description	Component
All	7338	Delete support for RTCP RPSI and SLI messages	Video
Native	5549	iOS: remove use of deprecated APIs when moving from ios_deployment_target 7.0 to 9.0.	Build, Mobile (iOS)
Native	7035	Remove webrtc::CriticalSectionWrapper and migrate to rtc::CriticalSection	Internals

 

Features and Bugfixes

Chrome

 

Type	Issue	Description	Component
Feature	6018	AEC3: major AEC functionality update	Audio
Feature	7058	Calculate first-order-FEC-recoverable packet loss rate for audio and use it to adapt Opus in real time.	Audio
Feature	7175	Support N unsignaled audio streams	Audio
Feature	7392	Set max bitrate for audio send stream based on RtpParameters	Audio
Feature	6984	Add probe logging to RtcEventLog (and UMA?).	BWE
Feature	6566	BWE can be too sensitive to delay spikes.	BWE
Feature	5222	Support GCM encryption for SRTP	Network
Feature	6541	QP statistics for WebRTC codecs	Stats
Feature	6822	Improve bitrate probing at high bitrates.	Video
Feature	7208	Perform probing when network route change.	Video
Feature	7412	Enable vp9 denoiser by default	Video
Bugfix	7169	Render signal is not stored to aecdump when AEC is disabled	Audio
Bugfix	7334	AudioEncoderOpus::RecreateEncoderInstance uses old config_ for num of samples and num of channels	Audio
Bugfix	7270	Memory leak in Windows Core Audio	Audio
Bugfix	7257	Avoid cyclic dependency between RTC event log and audio network adaptor	Audio
Bugfix	7282	WebRTC might pick the wrong CN payload id for sending	Audio
Bugfix	7290	VoiceProcessThread busy	Audio
Bugfix	7347	Feedback about packets sent on an old route may temporarily have a large effect on BWE.	BWE
Bugfix	7059	webrtc::field_trial::FindFullName called too often	Cleanup
Bugfix	7305	XGetImage errors are not always handled properly	DesktopCapture
Bugfix	7304	CPU adaptation for HW encode	HardwareCodec
Bugfix	4508	Make ProcessThread::DeRegisterModule and RegisterModule be called consistently on the same thread	Internals
Bugfix	7341	QueueTask on windows don't execute DelayedTasks then under heavy load.	Internals
Bugfix	7288	Incorrectly masked seq number for GCM IV.	Internals, Network
Bugfix	7220	4 microphones not supported in Windows core audio	Mic
Bugfix	7072	Error thrown if Chrome receives subsequent offer with DTLS role that's not "actpass"	Network
Bugfix	3416	PeerConnectionFactory requires initializing voice & video engines even in datachannel-only apps	PeerConnection
Bugfix	6959	SCTP DataChannel packets with source ports 32768-49151 get filtered out	PeerConnection
Bugfix	1928	Support Suite B cryptosuites for SRTP	PeerConnection
Bugfix	7181	RTCStatsCollector retains pointers to closed data channels	Stats
Bugfix	7432	Quality limited resolution stats are reported when quality scaler is disabled.	Stats, Video
Bugfix	7400	VideoReceiveStream::Stats.total_bitrate_bps does not report the received bitrate when there is packet loss	Stats, Video
Bugfix	7362	DCHECK fails in RtpFrameReferenceFinder::Vp9PidTl0Fix	Video
Bugfix	7027	Video isn't played if codec name isn't correctly cased ("vp8" instead of "VP8")	Video
Bugfix	7293	Improve performance of ParseRbsp() function.	Video
Bugfix	6202	Cannot remove b=AS restriction once it has been applied	Video
Bugfix	6139	Probe fails due to negative send delta time	Video
Bugfix	7331	FrameBuffer can cause busy loops during shutdown	Video
Bugfix	7335	PacedSender can busy loop while paused	Video
Bugfix	7330	ThreadSanitizer: lock-order-inversion in RespectsMinTransmitBitrateAfterContentSwitch	Video
Bugfix	714386	User-preferred audio device in content settings is not preferred over default device	Blink>GetUserMedia
Bugfix	682887	getUserMedia gives 640x480 when constraints say 1280x720	Blink>GetUserMedia
Bugfix	651910	Chrome does not throw getUserMedia error when any other application is using camera on Windows	Blink>GetUserMedia, Blink>WebRTC
Bugfix	697780	chrome.desktopCapture broken on Mac: does not mirror context menus and shows two cursor	Blink>GetUserMedia>Desktop
Bugfix	702745	DesktopCapturer::Create(Window|Screen)Capturer() do not handle failures correctly.	Blink>GetUserMedia>Desktop, Services>Chromoting
Bugfix	695440	"Always allow https://... to access your microphone" is not sticky.	Blink>GetUserMedia>Mic, UI
Bugfix	710040	tab mirroring broken. local content flashes white and blue when page updates	Blink>GetUserMedia>Tab
Bugfix	684288	Stream with disabled video track reports 2x2 resolution on loadedmetadata	Blink>MediaStream
Bugfix	704326	CHECK failure: false in jsepsessiondescription.cc	Blink>WebRTC
Bugfix	705505	can't cast from tab9	Blink>WebRTC, Internals>Cast
Bugfix	651512	Disabling (wait to enable) #enumerate-audio-devices feature on ChromeOS	Blink>WebRTC>Audio
Bugfix	703122	Partial RTCRtpReceiver and RTCRtpContributingSource support	Blink>WebRTC>Network
Bugfix	673395	Implement RTCConfiguration.iceCandidatePoolSize	Blink>WebRTC>PeerConnection
Bugfix	712311	Using DXVA HW decoder for WebRTC fails	Blink>WebRTC>Video, Internals>Media>Codecs

 

Native Android/iOS

 

Type	Issue	Description	Component
Feature	7456	Add support for 64-bit architectures in build_aar.py.	Mobile (Android)
Feature	6328	Make AppRTCDemo easy to use from Android Studio	Mobile (Android)
Feature	7316	iOS: Add an option to specify preferred codec in AppRTCMobile	Mobile (iOS)
Feature	CL	Add loopback launch argument functionality in AppRTCMobile	Mobile (iOS)
Bugfix	7267	Incorrect frame size from Android HW H264 decoder	Video, Mobile (Android)
Bugfix	7431	Implement getter for configuration in RTCPeerConnection	PeerConnection (iOS)
Bugfix	7139	Chrome on Android: H264 HW decoding error since #447978	Video (Android)
Bugfix	674959	Inconsistent UI for Android screensharing	Blink>MediaStream, Blink>WebRTC (Android)

 

Regards

Anatoli

[Kushtrim Pacaj]

Hi, is there a way to turn on AEC3 for testing in Android native ? Is it intended to be used in mobile ?

[Eric M]

AEC3 can be used on mobile?  how about it performance?

在 2017年5月10日星期三 UTC+8下午8:07:24，Anatoli Davidson写道：

[bl...@webrtc.org]

Our AEC3 effort focuses on improving echo cancelation in Chrome on desktop platforms. It's not adapted for mobile platforms.

/Niklas