PSA: Chrome M44 WebRTC Release Notes

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Christoffer Jansson

Jun 17, 2015, 4:03:41 AM6/17/15



Guess what? We continue on our focus of speed and stability for Chrome 44.

With M44 we’ve done major low-level refactoring to the WebRTC video stack which aims to decrease complexity and enable future work to increase performance and improve overall stability.

For our Mac users out there we have identified an initialization problem for AVFoundation preventing the camera from starting properly, this has now been fixed. The results should translate into a reduction in camera initialization failures.

We encourage all developers to run dev versions frequently and quickly report any issues found. The help we have received has been invaluable! On to dev and canary everyone!

Finally, we encourage you to test our new echo canceller. While it is on for a percentage of users on beta and stable, you can force it on through chrome://flags and set the “Enable Delay Agnostic AEC in WebRTC.” to enable.



  • Issue 473336: Desktop/Window/Tab capture engines to provide both variable- and fixed- resolution modes

  • Issue 1708:  Add ViE callback for encoded resolution

  • Issue 1788: Port libjingle video to webrtc::Call APIs.

  • Issue 3872: patch: enable DTLS in peerconnection_client sample

  • Issue 4529: Bump default microphone volume to 33% at startup (Chrom[e|ium] only)

  • Issue 4609: add flag to unpack_aecdump to store mic level in a text file


  • Issue 434744: googframewidthsent and googframeheightsent should report dimensions after downscaling, not set codec dimensions

  • Issue 477749: webrtc on MIPS cannot build without -fomit-frame-pointer

  • git hash 7252a2ba8035c4128917a9558a3e34fc9dbe7c44 Add HW fallback option to software decoding

  • Issue 1574: Update BUNDLE support to match draft-ietf-mmusic-sdp-bundle-negotiation-03

  • Issue 2335: TSAN errors in libjingle_p2p_unittest/libjingle_peerconnection_unittest after ljbjingle roll 4595 -> 4612

  • Issue 2920: Support ability to fall back to built-in codec factory when external codecs develop issues

  • Issue 4020: Logic error found by /analyze

  • Issue 4033: Report encoded (internally downscaled) resolution in VideoSendStream::GetStats

  • Issue 4077:  New VIE API - setRemoteDescription updates causes UI locks

  • Issue 4084: Chrome crashes when sending and receiving intensively over multiple data channels

  • Issue 4175: Assert fails when a not-connected data channel, whose SID is not explicitly set, is closed.

  • Issue 4208: "Unsupported candidate type" created by Chrome itself

  • Issue 4535: Potential deadlock in CPU adaptation vs. CreateChannel

  • Issue 4546: Frequently fluctuating rtt values with ViE2

  • Issue 4558: Received packet loss missing with ViE2.

  • Issue 4568: Deadlock between MediaOptimization::UpdateWithEncodedData and ViEChannel::ProtectionRequest

  • Issue 4569:  VPx specific settings not correctly set in the new API

  • Issue 4574: Combined audio/video BWE with ViE2 ("Call") API

  • Issue 4605: CPU adaptation should be disabled for screen content in the new API

  • Issue 4678: Receivers created before first send stream doesn't use the RTCP SSRC of the first sender.

  • Issue 4745: Reduced-size RTCP packets sent without negotiation

Aelix Smith

Jun 20, 2015, 2:31:00 AM6/20/15
What do you mean by "While it is on for a percentage of users on beta and stable"? I am assume echo canceller is either ON or OFF for all beta or stable users.
Since we are having echo problem with Chrome version 43, I would appreciate if you can clarify your statement.


Jun 20, 2015, 3:12:10 AM6/20/15
to WebRTC-discuss
Chrome has some experiments that toggle a feature only for some of the users. This new echo canceller is one of those experiments.
What you need is a way to tell whether it is on or not, I do not know if such a way exists, others might.



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Ashok Pitambar

Sep 16, 2015, 8:47:38 AM9/16/15
to discuss-webrtc
Hi All, 

How do I directly download M44 webRTC release?


Peter Boström

Sep 16, 2015, 8:48:07 AM9/16/15
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Ashok Pitambar

Sep 18, 2015, 7:26:29 AM9/18/15
to discuss-webrtc
Thanks Peter

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