WebRTC M74 Release Notes
WebRTC M74 branch (cut at r26981)
WebRTC M74, currently available in Chrome's beta channel and as native libraries for Android and iOS, contains feature additions like RID/MID based Simulcast, multiple 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. Native libraries for Android and iOS are built on a weekly basis and are available on JCenter and CocoaPods; the Changelog is available here.
We made the RTCPeerConnections iceConnectionState (mostly) spec-compliant. The main change is that the state is now entirely computed from the state of the ICE transports, instead of including signals from the DTLS layer. That is, we report “connected” as soon as ICE is connected instead of waiting for the DTLS handshake which we used to do.
All default IDs for RTP extensions are removed from the header file. One-byte RTP extensions may only have IDs in the range 1-14. For higher IDs, the two-byte format must be used. If default IDs are set for all extensions, once 15 extensions are defined by the code, some extensions will have IDs greater than 14. This will happen even if only one extension actually ends up being offered, so long as it's that unfortunate RTP extension. It's better to dynamically assign IDs to those extensions we actually offer
The new echo canceller solution AEC3 has now replaced the legacy AEC (AEC2) in the audio
processing module as the desktop AEC in WebRTC
We've restricted the host candidate obfuscation to peer connections for which the user has not provided
getUserMedia permission. The intent is to expand the experiment also to include Chrome Beta and
Developer editions over the next few days. As mentioned before, this feature and experiment only affects
Chrome on Windows, MacOS and Linux
PSA: Spec-compliant simulcast using addTransciever API
Simulcast allows sending multiple encoded versions of the same source.
Spec-compliant simulcast uses MID and RID extension headers to identify the layers and does not signal
SSRCs. This is a new API surface to replace the SDP-munging scenario. No action is needed,
SDP-munging applications will not break. Users are encouraged to migrate to the new API
The encoding parameter scaleResolutionDownBy to dynamically change the size of your layers has been
Implemented. It is supported in simulcast mode too and can be paired with the spec-compliant simulcast
using addTransceiver API to resize each layer as needed
Improved spec compliance of RTCIceCandidate. Now all the information about a candidate is available in
separate fields and the constructor works as specified by the standard
The new echo canceller solution AEC3 has now replaced the legacy AEC (AEC2) in the audio processing module as the desktop AEC in WebRTC. The work for this was tracked in issue 10366.
Type | Issue | Description | Component |
Feature | Rid support and wiring (for Simulcast use) | ||
Feature | Always rewrite/add VUI and set max_num_reorder_frames=0 | Video | |
Feature | Implement RtpEncodingParameters.scale_resolution_down_by | PeerConnection | |
Feature | Handle reordered packets in NetEq | Audio | |
Feature | Probe controller should cap all probes at max allocated bitrate | BWE | |
Feature | RtpStreamId (rid) headers in mux/demux | PeerConnection | |
Bug | Remove dependencies on audio codec implementations | Audio | |
Bug | VP8 at one temporal layer unnecessarily predicts from very old frames | Video | |
Bug | The IDs of RTP extensions should not be hard-coded | Network>RTP | |
Bug | Toggling of retransmissions can lead to spurious mid-call probes | Video | |
Bug | Video simulcast hysteresis not accounted for in padding calculation | Video | |
Bug | Desktop capture frame can have corrupted data when using multiple screens | DesktopCapture | |
Bug | RttBackoff misfires if TWCC-stamped sending ceases. | ||
Bug | setRemoteDescription called twice throws an error if SDP don't have media sections with mid attributes | PeerConnection | |
Bug | Jitter buffer delay statistic reporting incorrect value | Audio | |
Bug | RTC_CHECK failure in VideoStreamEncoder | ||
Bug | Restore the ability to test two peerconnection_clients on one machine | PeerConnection | |
Bug | Timeout in neteq_signal_fuzzer | Blink>WebRTC>Audio | |
Bug | close() may get stuck because of pending getStats() requests | Blink>GetUserMedia>Webcam | |
Bug | RTCIceCandidate: Add missing attributes | Blink>WebRTC | |
Bug | WebRTC doesn't share Office documents correctly | Blink>GetUserMedia>Desktop | |
Bug | Create Loss Notification RTCP feedback message | Network>RTP | |
Bug | RtpParameters are invalidated in calls to SLD/SRD | PeerConnection | |
Bug | Obsolete cricket::VideoCapturer | Video | |
Bug | Incorrect spectral features computations in the RNN VAD | Audio | |
Bug | Reported network queue is growing when packet loss is simulated | Network | |
Bug | mDNS obfuscation of host candidates should be turned off with getUserMedia permission | Blink>WebRTC>Network | |
Bug | SimulatedNetwork does not reflect bandwidth changes correctly. | BWE | |
Bug | iceConnectionState doesn't go to failed if connection does not establish | Blink>WebRTC>Network | |
Bug | RTCPeerConnection calls onIceConnectionStateChanged(Connected) even if no connection is possible | Blink>WebRTC>Network | |
Bug | NetEq produces audio that sounds like a repeating pattern in certain circumstances | Video | |
Bug | No distinction between simulcast layer paused and disabled | PeerConnection | |
Bug | mdns may generate invalid c= line | Blink>WebRTC>Network | |
Bug | Incorrect FPS measured when frame dropper kicks in | Video | |
Bug | Mic audio is silent without error on macOS 10.14 (Mojave) if system permission is denied | Blink>GetUserMedia>Mic Blink>WebRTC>Audio | |
Bug | Issue in the camera and microphone permissions flow with Mac Mojave OSX | Blink>GetUserMedia |
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Did the DTLS 1.0/1.1 deprecation make it in? Will M74 only support DTLS 1.2? I don't see https://bugs.chromium.org/p/webrtc/issues/detail?id=10261 mentioned in the release notes.
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