This will handle the scenario where, for example, the initial offer/answer only negotiates audio, and video is added later (to the same stream). Previously, there was absolutely no way to get a handle to the new track without hacking the SDP. Now, the stream will be updated after setRemoteDescription finishes. Bug: webrtc:5677 Change-Id: Iea31bb7744da6b82afdaf44c8f74d721298a9474 Reviewed-on: https://webrtc-review.googlesource.com/6261 Reviewed-by: Sami Kalliomäki <sakal@webrtc.org> Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20228}
Reland of Fix the video buffer size should take rtt into consideration (patchset #2 id:160001 of https://codereview.chromium.org/3002033002/ )
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
Description
The idea is to make CMake build for WebRTC m130 version - for audio processing module
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