9b1f24d552
This reverts commit 8b3e4e2d1166464f6b309f4fc533a29607d2771f. Reason for revert: The culprit was https://webrtc-review.googlesource.com/c/src/+/133169. Original change's description: > Revert "Reland "Add ability to set RTCP sender ssrc at construction time"" > > This reverts commit 6f420e424885dab1d9f885365ea9abea5cc4a901. > > Reason for revert: Speculative revert (some perf test are failing) > > Original change's description: > > Reland "Add ability to set RTCP sender ssrc at construction time" > > > > This is a reland of 94c58fd815f0c7c6429aa53a79621ea9ef39c770 > > > > Patch set 1 is the original CL. > > Patch set 2 introduced a trivial fix. In RtcpSender::SetSSRC(), check > > if either current SSRC is 0 or if the SSRC is identical to the current > > one. If so, don't schedule an early report. > > This prevents a regression in which audio jitter became too low(?) > > > > Original change's description: > > > Add ability to set RTCP sender ssrc at construction time > > > > > > Bug: webrtc:10774 > > > Change-Id: Iaf5857e24359e9795434bcd0cdbe1658a2f9f5d3 > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144632 > > > Reviewed-by: Åsa Persson <asapersson@webrtc.org> > > > Commit-Queue: Erik Språng <sprang@webrtc.org> > > > Cr-Commit-Position: refs/heads/master@{#28506} > > > > Bug: webrtc:10774 > > Change-Id: I103dfa48719aa41d6ab633cdac8b3a5c46b54843 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144565 > > Commit-Queue: Erik Språng <sprang@webrtc.org> > > Reviewed-by: Åsa Persson <asapersson@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#28520} > > TBR=asapersson@webrtc.org,sprang@webrtc.org,ilnik@webrtc.org > > # Not skipping CQ checks because original CL landed > 1 day ago. > > Bug: webrtc:10774 > Change-Id: I39238d942b2bbe0a9c8ca752387a35ed9dd70650 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145327 > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#28555} TBR=mbonadei@webrtc.org,ilnik@webrtc.org,asapersson@webrtc.org,sprang@webrtc.org Change-Id: I2e5c17e8edfd938424f901222158643baa04866e No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:10774 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145400 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28562}
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.
Development
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.
More info
- Official web site: http://www.webrtc.org
- Master source code repo: https://webrtc.googlesource.com/src
- Samples and reference apps: https://github.com/webrtc
- Mailing list: http://groups.google.com/group/discuss-webrtc
- Continuous build: http://build.chromium.org/p/client.webrtc
- Coding style guide
- Code of conduct