9cf9f758fc73bd4dde477cf02c7beeee669b8a39
If this is not done, the RTC_DCHECK_CALLED_SEQUENTIALLY might fire if the encoder is used on a new VideoStreamEncoder. This happens after VideoSendStream recreations due to changes in the SDP. BUG=b/66590444 Change-Id: I086370526afbbe2ba629805f97f89e512ba3f472 Reviewed-on: https://webrtc-review.googlesource.com/4360 Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org> Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Commit-Queue: Rasmus Brandt <brandtr@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20020}
Reland of Fix the video buffer size should take rtt into consideration (patchset #2 id:160001 of https://codereview.chromium.org/3002033002/ )
Revert of Add full stack tests for MediaCodec. (patchset #10 id:180001 of https://codereview.webrtc.org/3005253002/ )
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.
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
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