WebRTC 113 Release Notes

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Harald Alvestrand

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Apr 24, 2023, 8:46:52 AM4/24/23
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WebRTC M113 is going to be released as part of Chrome M113 , currently planned for release on  May 2nd 2023.


We had two PSAs:


Please note that the legacy callback-based getStats variant is deprecated and scheduled for removal in M117.


The following issues were marked as fixed or verified and had at least one commit in M113 (build, test and trivial code changes are not included):



Issue

Summary

Component

aomedia:3391

Bitrate overshoot encoding AV1 on Windows

Video

chromium:1423361

Using VP9/SVC and webrtc-svc, the video consumes more bitrate when we scale down the encoding.

Blink>WebRTC>Video,
Blink>WebRTC

chromium:1424238

Failed to set local offer sdp errors on Chrome v111

Blink>WebRTC,
Blink>WebRTC>PeerConnection

chromium:1412584

Switch WGC to ScreenCaptureFrameQueue

Internals>Media>ScreenCapture

chromium:1425362

CloneSenderVideoFrame() always sets frameType to kVideoFrameDelta

Blink>WebRTC

chromium:1428098

VP9 {L3T3_KEY,inactive,inactive} not sending correctly in M113

Blink>WebRTC>Video

webrtc:13089

Android MediaCodec Encoder requires 16-pixel alignment

Mobile

webrtc:14367

Refactor ICE controller to support an active interface

Network>ICE

webrtc:14931

Add option to configure AV1 EncoderInfo resolution_bitrate_limits

Video

webrtc:14367

Refactor ICE controller to support an active interface

Network>ICE

chromium:1423365

Chrome tab crashes using VP9/SVC and setting a low maxBitrate in the encoding parameters

Blink>WebRTC>Video,
Blink>WebRTC

webrtc:13429

Tracking bug for better Linux/Wayland screensharing support

DesktopCapture

chromium:1420245

Remove TransformableVideoFrame::GetMetadata in favour of TransformableVideoFrame::Metadata()

Blink>WebRTC

chromium:1421534

Chrome: Crash Report - [Assert] webrtc::SctpDataChannel::SetSctpSid

Blink>WebRTC>DataChannel

webrtc:14884

Support VP9 simulcast

Video

webrtc:15016

VP9: Changing from legacy SVC to standard 1/3 active encodings /w scalability_mode set crashes

Video

webrtc:15028

Single-layer-VP9 simulcast should behave like VP9 SVC.

Video

webrtc:14967

h264 sps log2_max_frame_num_minus4/log2_max_pic_order_cnt_lsb_minus4 max value is incorrect in sps parser

Video

webrtc:15005

Support AV1 simulcast

Video

webrtc:14973

stats: rename RTCInboundRTPStreamStats and RTCOutboundRTPStreamStats

Stats

webrtc:15005

Support AV1 simulcast

Video

webrtc:14948

stats: RTCOutboundRtpStreamStats should inherit from RTCSentRtpStreamStats

Stats

webrtc:14989

Make RTCRtpSentRtpStreamStats.packetsSent unsigned long long

Stats

webrtc:7660

Remove VideoCodec types from webrtc/common_types.h

Build,Video

webrtc:15029

PipeWire capturer: drop old DmaBuf support

DesktopCapture

webrtc:15032

NativeToJavaPointer should support const pointers

Video

webrtc:14859

Ensure ReceiveSideCongestionController use modern types

Cleanup,BWE

webrtc:14813

Deprecate and remove stats timestamp_us()

Stats

webrtc:14884

Support VP9 simulcast

Video

webrtc:14931

Add option to configure AV1 EncoderInfo resolution_bitrate_limits

Video

webrtc:5773

Bringing AppRTCDemo to the background and back to foreground can cause BWE to get stuck at low levels.

Video

Issue

Summary

Component

chromium:1302689

Align lifecycles of MediaStreamSource and MediaStreamComponent with their Platform members

Blink>MediaStream

chromium:1420245

Remove TransformableVideoFrame::GetMetadata in favour of TransformableVideoFrame::Metadata()

Blink>WebRTC

chromium:1425373

Cloning an already sent RTCEncodedVideoFrame crashes

Blink>WebRTC

chromium:1427193

Unstable fps when using MediaStream for video (Starting from Chrome v111)

Blink>MediaStream

webrtc:14884

Support VP9 simulcast

Video

chromium:1375217

tracking bug for webrtc-internals ui/ux improvements

Blink>WebRTC>Tools

chromium:1422441

Software AVC decoder is used for RTC decoding when negotiated packetization-mode is 0




For the full list of commits please refer to the git log between this branch and the previous branch.


We strongly recommend WebRTC developers to fully test their services in Chrome Beta to ensure stability for end-users.

 

The Chrome release schedule can be found here.


These release notes were prepared by Philipp Hancke.



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