PSA: WebRTC M58 Release Notes

[Anatoli Davidson]

M58

WebRTC M58 branch (cut at r16937)

Summary

WebRTC M58, currently available in Chrome's beta channel and as native libraries for Android and iOS, contains over 20 new features and over 60 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

Fixed M57 regression in bandwidth limitation

A regression has been fixed that was introduced M57 and affects in how the "b=AS" SDP attribute is handled. This SDP attribute can be used to limit the maximum total bandwidth used by a media stream, and it's should be possible to set different values for different m sections. For details and a workaround for M57, see this PSA.

Features

Spec-compliant RTCPeerConnection.getStats

The promise-based getStats is released which (unlike the callback-based getStats) returns stats that follow the spec (Statistics Model, Identifiers for WebRTC's Statistics API). Most but not all stats are supported. The callback-based getStats is still available but we aim to deprecate it in a future release.

Adding support of Opus 120ms encoding in WebRTC

Starting from version 1.2-alpha, Opus supports direct encoding of audio frames with a duration up to 120 ms. Chromium does not include this Opus version in its third party dependency yet, but WebRTC has built up infrastructure to support 120ms encoding. Longer frames reduce transport overhead on low capacity networks. To utilize the upcoming Opus 120ms frame encoding, audio network adaptor has also been updated to allow frame length to adapt to 120ms.

RTCPeerConnection.setConfiguration

The setConfiguration method allows an application to modify the RTCConfiguration of an RTCPeerConnection. Specifically, it allows changes to the ICE servers and ICE transport policy, e.g., to specify new TURN credentials when the existing credentials expire. Previously, there was no workaround for this scenario aside from a full teardown of the connection. Another use case is changing the ICE transport policy depending on the phase of a call. For example, a call may begin with only relay connections (either to speed up call setup or to protect the user's privacy), then later switch to the "all" policy. Note that the iceCandidatePoolSize member of RTCConfiguration is still unsupported, but is planned to be implemented in M59.

Authenticated HTTP proxy

WebRTC can now make use of proxies that require explicit credentials, as long as the user has authenticated against said proxy at least once before initiating a WebRTC connection. Proxies that use Single-Sign-On or that do not require authentication continue to work as expected.

Script for generating WebRTC Android Library (.aar)

There now exists a script for generating WebRTC Android library (.aar). The script exists in tools-webrtc/android/build_aar.py and can be used to generate .aar-file that can be included in an Android Studio project that uses WebRTC. More information on how to use the script can be found at the top of the script.

Enable VP9 support in WebRTC HW decoders

There are existing VP9 HW decoder implementations in Chrome, i.e. DXVA based in Windows and V4L2/VAAPI based in Chrome OS. It is now possible to use these as external video accelerators in WebRTC calls where VP9 is used.

New video jitter buffer

The video jitter buffer has been rewritten from scratch. The new jitter buffer has been implemented as five classes (PacketBuffer, NackModule, FrameObject, RtpFrameReferenceFinder, FrameBuffer). The benefits of the new jitter buffer are lower code complexity resulting in easier maintenance and tuning. It opens up the possibility to implement transports other than RTP. The new design will enable the reduction of memcpys, but this requires further work. There should be no significant difference in behavior between the current and the new video jitter buffer.

Audio output debug recording

A debug recording of audio output is now generated when checking “Enable diagnostic audio recordings” in chrome://webrtc-internals. File recording of playout audio happens in the browser process close to the OS/sound system.

Deprecations

Platform	Issue	Description	Component
Chrome	7198	Remove support for manual reference counting in ObjC code	Cleanup
Chrome	684741	Removed Wave-based audio capture implementation on Windows	Audio
Chrome	7025	Remove outdated openssl defines.	Cleanup, Network
Chrome	7124	Delete VideoReceiveStream::Config::pre_render_callback	Video
Chrome	4306	Remove shared global SSRC database	Video
Native	7183	Remove Wave legacy audio capture implementation on Windows	Audio
Native	7158	Remove VideoRendererGui	Mobile, Video
Native	PSA	Delete header file webrtc/base/common.h	Internal
Native	4690	Removed VoEExternalMedia interface.	Audio
Native	4690	Removed VoEVideoSync interface.	Audio
Native	4690	Removed VoEAudioProcessing interface.	Audio
Native	4690	Removed unused and untested methods from VoERTP_RTCP interface.	Audio
Native	6829	Removed base/linux.cc/.h and linuxfdwalk.c/.h.	Internal
Native	6829	Removed base/dbus.cc/.h.	Internal

Features and Bugfixes

Chrome

Type	Issue	Description	Component
Feature	7093	Add support for 120ms frames in the audio network adaptor.	Audio
Feature	7097	Adding support of Opus 120ms direct encoding in WebRTC.	Audio
Feature	6303	Adding network adaptation to audio encoder.	Audio
Feature	6845	Logging audio network adaptor decisions in the event log	Audio
Feature	7160	Add tool for visualisation of the Audio network adaptor decisions	Audio
Feature	7199	Frame length controller should not switch to shorter frame lengths when it will cause a overuse.	Audio
Feature	7167	Optionally disable APM limiter in AudioMixer.	Audio
Feature	7215	Use FallbackDesktopCapturerWrapper for ScreenCapturerWinMagnifier	DesktopCapture
Feature	7196	Values stored in rtc::Optional does not print properly in tests	Internals
Feature	7118	Basic command line tool to print packet information from an RTC event log	Internals
Feature	6805	Implement support for setting constraints-related properties on MediaStreamTrack	SpecConformance
Feature	6991	video_replay: output RTP header contents when delivery fails	Video
Feature	6904	Adding transport feedback based packet loss estimation.	Video
Feature	5514	Jitter Buffer Rewrite.	Video
Feature	6423	Log delay-based bandwidth estimate	Video
Feature	6984	Add probe logging to RtcEventLog (and UMA?).	Video
Feature	685655	Launch bug for promise-based RTCPeerConnection.getStats	Blink>WebRTC
Feature	531883	Add debug recording to file of speaker output	Blink>WebRTC>Audio
Feature	Design doc	Use DirectX for faster screen capture on Windows	Blink>GetUserMedia>Desktop
Feature	687650	Enable VP9 support in WebRTC HW encoder and decoders	Internals>Media>Codecs
Feature	587453	PeerConnection.setConfiguration	Blink>WebRTC>Network
Feature	600399	Use assembler for OpenH264 encoding on Windows and Linux	Blink>WebRTC>Video
Bugfix	7272	Make Opus decoder go to CNG mode for 2-byte payloads	Audio
Bugfix	7144	Transmit correct number of bytes returned by the Opus encoder in DTX mode	Audio
Bugfix	7136	Allow residual echo detector to be enabled/disabled using AudioOptions	Audio
Bugfix	7087	AudioEncoderOpus can set bitrate that no audio can be encoded.	Audio
Bugfix	7083	Stop sending audio packets when the connection becomes unwritable	Audio
Bugfix	7039	disable APM clobbering for WebKit build	Audio
Bugfix	7258	StartupShutdownWithExternalADM sometimes enters a busy loop	Audio
Bugfix	7179	Don't recreate audio receive streams if they were previously unsignaled	Audio, PeerConnection
Bugfix	7043	Enable bitrate probing for audio streams.	Audio, Video
Bugfix	3062	GetStaticInstance subject to race	Audio, Video
Bugfix	6986	"stereo" parameter isn't handled correctly for Opus decoder	Audio
Bugfix	7237	FakeAudioDeviceModule runs a busy loop by default.	Audio
Bugfix	7116	Potential use are free in h264 packetizer	Internals
Bugfix	7054	current webrtc's NO_RETURN definition is conflicting with WebKit one	Internals
Bugfix	7151	Add support for multimedia timers to TaskQueue	Internals
Bugfix	7216	Add support for priorities to TaskQueue	Internals
Bugfix	7099	Allow applications to control the DTLS timeout	Network
Bugfix	7109	Allow STUN RTTs up to 60 seconds	Network
Bugfix	4947	Change UpdateIce to SetConfiguration in the JS PeerConnection API	PeerConnection
Bugfix	6972	Having a=crypto in an SDP offer causes exception in Canary	PeerConnection
Bugfix	7131	Add better comments to interfaces in api/ subdirectory	PeerConnection
Bugfix	3914	Data races in AsyncInvoker destructor [linux_tsan2]	PeerConnection
Bugfix	4180	Expose DtmSender from RtpSender (in C++, Java interfaces)	PeerConnection
Bugfix	6487	Should VideoCapturerTrackSource increase its reference count when sending message on the thread?	PeerConnection
Bugfix	6857	peerconnection_client tool not working	PeerConnection, SampleApps, Video
Bugfix	5208	webrtc stops playing video when SSRC of incoming RTP stream has changed	PeerConnection, Video
Bugfix	7011	RTX payload types are not updated in VideoReceiveStream when WebRtcVideoChannel2::SetRecvParameters is called.	Video
Bugfix	7017	FlexfecReceiveStreams are leaked when WebRtcVideoChannel2 is destroyed	Video
Bugfix	7211	Video scaling stuck at low resolution	Video
Bugfix	7255	Busy loop in PacedSender+BitrateProber	Video
Bugfix	7269	Busy loop in VideoReceiveStream	Video
Bugfix	7178	Video streams can get stuck in suspended state	Video
Bugfix	7125	Log which delay estimator type is picked by BWE.	Video
Bugfix	7246	NackModule will enter a busy loop when running_ == false	Video
Bugfix	7045	AV sync issue in WebRTC in 20% of streams	Video
Bugfix	6953	Video call no longer starts with scaled resolution on low initial bandwidth.	Video
Bugfix	6876	protection bitrate should include overhead as well	Video
Bugfix	7020	webrtcvideoencoderfactory grows codec list every time codecs/supported_codecs is called	Video
Bugfix	7154	Video decoder software fallback not reinitialized after ::Release()	Video
Bugfix	6998	Improve computational performance of bandwidth estimator	Video
Bugfix	7022	Add new simulations to test BWE at low bitrates	Video
Bugfix	7200	|event_| in RembWithSendSideBwe is never set	Video
Bugfix	7259	FakeNetworkPipe can cause a busy loop in PlatformThread	Video
Bugfix	697516	Picker list is too small when sharing tab only	Blink>GetUserMedia>Desktop
Bugfix	684162	DirectX screen capturer doesn't handle screen size changes properly	Blink>GetUserMedia>Desktop
Bugfix	692261	Very old frame emitted from tab capture after resume()	Blink>GetUserMedia>Tab
Bugfix	687750	Media Remoting should never admit initialization failed	Blink>GetUserMedia>Tab
Bugfix	693746	Cannot select camera in Chrome Capture Settings	Blink>GetUserMedia>Webcam
Bugfix	687502	webrtc::Ramp in webrtc/modules/audio_mixer/audio_frame_manipulator.cc is slow	Blink>WebRTC, Blink>WebRTC>Audio
Bugfix	612294	PowerSaveBlock activated even though there are no active PeerConnections	Blink>WebRTC
Bugfix	694278	Move AudioDebugFileWriter from content/ to media/	Blink>WebRTC>Audio
Bugfix	675193	Integer-overflow in webrtc::TimeStretch::SpeechDetection	Blink>WebRTC>Audio
Bugfix	686212	WebRTC data channel responds with old sctp draft on new sctp offer format	Blink>WebRTC>Network
Bugfix	439560	WebRTC cannot connect to TURN servers through HTTP proxies that require auth	Blink>WebRTC>Network
Bugfix	CL	Black boxes around cropped videos	Blink>GetUserMedia>Desktop

Native Android/iOS

Type	Issue	Description	Component
Feature	6937	Android AppRTCMobile: Swap between local and remote feed on the fullscreen.	Mobile (Android, iOS)
Feature	7023	Script to generate AAR-files for Android Studio.	Mobile (Android)
Feature	7088	Support new peerconnection certificate policy API in ObjC	Mobile (iOS), PeerConnection (Mac)
Feature	6545	AppRTCMobile - Capture from file instead of camera and save rendering to file	SampleApps (Android)
Feature	6290	Remove unnecessary thread in MediaCodecVideoEncoder on Android	Video (Android)
Bugfix	5582	Consolidate helpers_ios into webrtc/base/objc	Audio (iOS)
Bugfix	6382	Apple Non-Public API usage	Audio (iOS)
Bugfix	7187	PlatformThread on posix has a potential for a costly busy loop	Internals
Bugfix	6738	Handle multiple H264 codecs in SDP when reordering the codec list priority	Mobile (Android, iOS)
Bugfix	6751	AppRTCMobile not working on Galaxy S3 mini running Android 4.2.2	Mobile (Android)
Bugfix	7015	Camera1Session: Fatal error: Camera is being used after Camera.release() was called	Mobile (Android)
Bugfix	6715	Android log spam in loopback call: Assuming benign blocking error : [0x0000000e] Bad address	Mobile (Android), Network
Bugfix	7082	Allow applications to control ICE timers from RTCConfiguration	Mobile, PeerConnection
Bugfix	6651	Check failed: 0 == slice_height (0 vs. 1)	Mobile (Android), Video
Bugfix	6396	SecRandomCopyBytes does not build using Xcode 8 on iOS	SampleApps (iOS)
Bugfix	7206	MediaCodec decoder has trouble with low resolutions	Video (Android)
Bugfix	7008	Camera2Session calls onCameraError instead of onCameraDisconnected if the session gets disconnected during startup	Video (Android)
Bugfix	6841	webrtc/sdk/objc/Framework/Headers/WebRTC/WebRTC.h doesn't compile on macOS	Video (iOS)

Regards

Anatoli

[dc_neo]

wow, it's impressive to do such big improvement.our team will keep tracking this project.

在 2017年4月6日星期四 UTC+8上午2:56:36，Anatoli Davidson写道：

[Philipp Hancke]

would you folks mind sending webrtc relevant "intent to ship" like the unflagging of getStats to this list as well?

Also, is work scheduled for fixing https://github.com/webrtc/samples/issues/869?

[Crossle Song]

It said File recording of playout audio happens in the browser process close to the OS/sound system. How to file recording playout on WebRTC native Android or iOS

在 2017年4月6日星期四 UTC+8上午2:56:36，Anatoli Davidson写道：

M58

Regards

Anatoli

[Henrik Andreasson]

There is no suitable high-level API but one possible way is to use the VoEFile API.

Can't say for sure how long that API will be exist in its current shape.

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[Crossle Song]

But it can not work record playout, only work record microphone audio

在 2017年4月18日星期二 UTC+8下午6:11:20，Henrik Andreassson写道：

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[Henrik Andreasson]

Please report any issues here adding as much details as possible:

platform, how to reproduce etc. This thread is not the right context for discussions.

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[Anton Venema]

Chrome 58 seems to be sending multi-time aggregation packets (MTAP) for H.264 while in packetization mode 1 (non-interleaved), which does not allow MTAP. It looks like this is already fixed in Chrome 59, but I wanted to mention it here as well in case a minor update is coming to resolve this.

[Zoron]

This interface is no longer available in the latest version. Is there any other interface that can replace it?

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