M45
We are excited to announce that the audio output device selection JavaScript API has been implemented behind a flag. Try it out using the following the demo pages, just make sure to follow the in-page instructions on how to enable it:
Our new delay-agnostic echo canceller is on for a majority of users on Windows, Linux and ChromeOS. Mac OS X will be ramping up later this quarter but is still behind a flag. This new AEC should result in fewer users hearing echo from the remote side. Do help us test it, and please let us know if you encounter a scenario where it doesn't work well.
A first initial step has been taking in assisting users that want to limit the IP addresses used by WebRTC, through the release of an official Chrome extension. More details in the discuss-webrtc PSA.
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!
For some pointers on how to file a good bug report, please take a look at this page.
Issue 450193: New delay agnostic echo canceller active on stable for Windows, Linux, and ChromeOS. Mac still behind a flag. Do help us test it.
Issue 457492: Creating an option to directly control multiple routes for WebRTC
Issue 438023: Implement an audio output device API for the web
Issue 4611: Add RtcpMuxPolicy based on https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-09#section-4.1.1
Issue 511082: Put navigator.mediaDevices behind a flag again.
Issue 504346: Hangout services extension preferences not updated
Issue 501120: Crash after getUserMedia()
Issue 280726: PeerConnection's RTP-based Data Channel allows bypassing rate limiter via SDP mangling, allowing for sending data with no congestion control.
Issue 447431: WebRTC does not respect packet boundaries with DTLS.
Issue 506206: Screenshare notification horizontal dimension is to small to display the text properly
Issue 507994: Using audio SourceID constraint with getUserMedia() causes video to fail.
Issue 1675: If application is send only there shouldn't be a thread waiting for decode
Issue 4042: Crash in SrtpSession::~SrtpSession() if srtp_create fails
Issue 49989004: Ensure mediasession generated offers with RTX contain an RTX ssrc for each video ssrc.
Issue 45439004: Add support for iOS http proxy detection.
Issue 4366: Adapted frames have wrong width and height and are cropped
Issue 4316: googTrack stats reports don't set timestamp
Issue 4617: RTX-TIME parameter is disappeared
Issue 4717: WebRTC ISAC audio codec fails on ARM7
Issue 4722: Stats for decoded framerate is always zero if frame is backed by a texture
Issue 4724: Candidate could be resurrected with incorrect candidate type
Issue 4778: Initial bitrate probing doesn't work with combined audio/video BWE.
Issue 4651: Change name of ReportedDelay
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