PSA: Chrome M53 WebRTC Release Notes

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Christoffer Jansson

Jul 28, 2016, 7:52:22 AM7/28/16
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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.

Important PSAs


Camera2 API support on Android

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.

Remove webkit prefix for navigator.webKitGetUserMedia

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.

Promise-based navigator.mediaDevices.getUserMedia() has been added

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.

Bugfixes and Features

  • 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

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Mathieu Hofman

Aug 3, 2016, 6:01:05 PM8/3/16
to discuss-webrtc
Issue 622773 doesn't seem to be public.
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