3b69817e62
This reverts commit 87bed4793ff8f463202f442381339626d0b27f0d. Reason for revert: <INSERT REASONING HERE> Original change's description: > Reland "Preserve min and max playout delay from RTP header extension" > > This reverts commit f31cc08ba01ed403e89255b5f3f38d5dbdde855e. > > Reason for revert: Reland with fixes > > Original change's description: > > Revert "Preserve min and max playout delay from RTP header extension" > > > > This reverts commit 85ba9972c42544c4771e394c9aa1d20bf5d09a91. > > > > Reason for revert: Audio might be more out of sync than needed due to jitter in received video stream. > > > > Original change's description: > > > Preserve min and max playout delay from RTP header extension > > > > > > Audio and video synchronization can sometimes override the minimum > > > and maximum playout delay that is set through the RTP header > > > extension. This CL makes sure that the playout delay always is > > > within the limits set by the RTP header extension. > > > > > > Bug: webrtc:10886 > > > Change-Id: Ie2dd4b901c4ed178759b555a8be04bd8b8f63bda > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150645 > > > Commit-Queue: Johannes Kron <kron@webrtc.org> > > > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > > > Cr-Commit-Position: refs/heads/master@{#28980} > > > > TBR=stefan@webrtc.org,kron@webrtc.org > > > > Change-Id: I417e440d8a7e04ab3e19faa4454b704d2b971cd7 > > No-Presubmit: true > > No-Tree-Checks: true > > No-Try: true > > Bug: webrtc:10886 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150652 > > Reviewed-by: Johannes Kron <kron@webrtc.org> > > Commit-Queue: Johannes Kron <kron@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#28984} > > TBR=stefan@webrtc.org,kron@webrtc.org > > Change-Id: I5a3908a8c45f7faedab6f009b22df81d674e13a0 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:10886 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150653 > Reviewed-by: Johannes Kron <kron@webrtc.org> > Commit-Queue: Johannes Kron <kron@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#28985} TBR=stefan@webrtc.org,kron@webrtc.org Change-Id: Id2e5d1ff804881e956a07fa4ae0f8301895dcc95 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:10886 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150654 Reviewed-by: Johannes Kron <kron@webrtc.org> Commit-Queue: Johannes Kron <kron@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28986}
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