commit | 054e3bbbe7921b8b84213ecbbb58f68cd5712f6d | [log] [tgz] |
---|---|---|
author | Chen Xing <chxg@google.com> | Fri Aug 02 10:29:26 2019 |
committer | Commit Bot <commit-bot@chromium.org> | Fri Aug 02 11:45:34 2019 |
tree | d538d7eaa46a41d2f537e910c942789586dfe96a | |
parent | 59bbd65561111fe25e8f0d481e64962598e7c87f [diff] |
Reland "Replace the implementation of `GetContributingSources()` on the audio side." This reverts commit 67008dfb366237469fe088a61b62c0cad852c024. Reason for revert: Tests in the Chromium repo have been changed to accomodate this CL: https://chromium-review.googlesource.com/c/chromium/src/+/1728565 Original change's description: > Revert "Replace the implementation of `GetContributingSources()` on the audio side." > > This reverts commit 8fa7151e4bbad40fec1f964fe0c003b8787bb78a. > > Reason for revert: Speculative revert to fix roll of webrtc into chrome. Right now tests related to RTCRtpReceiver failing and looks like it is main candidate, who can affect that behavior. > > Original change's description: > > Replace the implementation of `GetContributingSources()` on the audio side. > > > > This change replaces the `ContributingSources`-implementation of `GetContributingSources()` and `GetSynchronizationSources()` on the audio side with the spec-compliant `SourceTracker`-implementation. > > > > The most noticeable impact is that the per-frame dictionaries are now updated when frames are delivered to the RTCRtpReceiver's MediaStreamTrack rather than when RTP packets are received on the network. > > > > This change is almost identical to the previous video side change at: https://webrtc-review.googlesource.com/c/src/+/143177 > > > > Bug: webrtc:10545 > > Change-Id: Ife7f08ee8ca1346099b7466837a3756947085fc5 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144422 > > Reviewed-by: Oskar Sundbom <ossu@webrtc.org> > > Commit-Queue: Chen Xing <chxg@google.com> > > Cr-Commit-Position: refs/heads/master@{#28459} > > TBR=ossu@webrtc.org,chxg@google.com > > Change-Id: I5c631d4dcfb39601055ffce9b104f45eea871fd3 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:10545 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144562 > Reviewed-by: Artem Titov <titovartem@webrtc.org> > Commit-Queue: Artem Titov <titovartem@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#28478} TBR=ossu@webrtc.org,titovartem@webrtc.org,chxg@google.com # Not skipping CQ checks because original CL landed > 1 day ago. Bug: webrtc:10545 Change-Id: I609cca4f0ca4e1d31a156ba9eb44407518409f57 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147865 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Chen Xing <chxg@google.com> Commit-Queue: Chen Xing <chxg@google.com> Cr-Commit-Position: refs/heads/master@{#28746}
WebRTC is a free, open software project that provides browsers and mobile applications with Real-Time Communications (RTC) capabilities via simple APIs. The WebRTC components have been optimized to best serve this purpose.
Our mission: To enable rich, high-quality RTC applications to be developed for the browser, mobile platforms, and IoT devices, and allow them all to communicate via a common set of protocols.
The WebRTC initiative is a project supported by Google, Mozilla and Opera, amongst others.
See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.
Authoritative list of directories that contain the native API header files.