This is part of the work towards implementing Unified Plan. In Unified Plan, an RtpSender/MediaStreamTrack can be a part of multiple streams. This changes RtpSender to internally store stream ids as a vector so that later CL's can update downstream consumers to support multiple streams. For now, the vector can only have exactly one element in it, but once the supporting code has been updated the singular getter/setter will be removed and any number of streams can be associated with an RtpSender. Bug: webrtc:8183 Change-Id: I499a77ce79198861d35c91328b40ced2eb913cc0 Reviewed-on: https://chromium-review.googlesource.com/646712 Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Commit-Queue: Steve Anton <steveanton@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19733}
Revert of Fix the video buffer size should take rtt into consideration (patchset #3 id:40001 of https://codereview.chromium.org/2980413002/ )
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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://chromium.googlesource.com/external/webrtc
- 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
Description
The idea is to make CMake build for WebRTC m130 version - for audio processing module
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