WebRTC M53 branch (cut at r13341)
Chrome M53, currently available in Chrome's dev channel (soon in the beta channel), contains over 15 bug fixes, enhancements and stability 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.
Camera2 capture implementation is now available. Using the Camera2 API has improved performance on newer hardware compared to using the old Camera API on Android. It’s available in AppRTCDemo as a config option. Please try it and report back.
navigator.webkitGetUserMedia has now been unprefixed to navigator.getUserMedia since it now conforms to the spec. More details can be found in this intent to ship post.
We have added a promise based navigator.mediaDevices.getUserMedia() API. This is part of our ongoing spec compliance work and as a result updating the WebRTC API’s to be promise based. Details are in Issue 503227. Note that navigator.getUserMedia will remain callback based.
Issue 595428 Unmute source audio when sharing tab audio
Issue 597034 Regression: blob recorded using MediaStream Recording freezes playback if it has Audio
Issue 597334 Finch experiment for controlling H264 HW enc
Issue 603261 WebrtcEventLogApiTest tests failing on Site Isolation FYI bot
Issue 605340 desktopCapture: Active tab in foreground is behind the desktopCapture dialog
Issue 611698 Experiment flag WebRTC-EnableWebRtcEcdsa relies on rtc::KT_DEFAULT being KT_RSA
Issue 612198 Hardware encoding causing WebRTC problems on devices Big, Blaze and kitty
Issue 616680 Dtor of V4L2VEA causes a crash on Blaze
Issue 620565 RtcVideoEncoder should fallback if encoder doesn't provide timestamp
Issue 503227 Add promise-based navigator.mediaDevices.getUserMedia()
Issue 607439 Unprefix navigator.getUserMedia
Issue 610967 Log OS glitches to WebRTC log
Issue 622773 New audio rendering mixing strategy
Issue 623607 Add UMA stats for AEC filter divergence metric
Issue 618215 will make --use-file-for-fake-audio-capture loop the supplied .wav file by default, unless %noloop is appended to the path of the wav file (if so it will only play once). The previous default was to play once.
Issue 5917 Event log does not capture all outgoing packets
Issue 6076 A connection switch can lead to BWE being reset to zero.
Issue 6080 Merge to 53: fix for poor bandwidth when switching from video to screen sharing