This CL changes WebRtcVideoChannel::ResetUnsignaledRecvStream so that it deletes all default streams created by WebRtcVideoChannel::AddRecvStream. This is needed for the case that there are lingering default streams, whose SSRCs are different from the SSRCs that were subsequently signaled. This can happen when there are multiple "m= sections" and the early media is sent to an "m= section" that is later not supposed to be the sink for that particular SSRC. Default streams whose SSRC match the subsequently signaled SSRC is already handled here: https://source.chromium.org/chromium/chromium/src/+/master:third_party/webrtc/media/engine/webrtc_video_engine.cc;l=1386;drc=22387b44ff173d263b434889d394cea90368ab06?originalUrl=https:%2F%2Fcs.chromium.org%2F Bug: webrtc:11477 Change-Id: I96ed7e35b4904fb0757fe5824f8afa6f1b9a565e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172622 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Magnus Flodman <mflodman@webrtc.org> Commit-Queue: Rasmus Brandt <brandtr@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30971}
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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.
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
Description
The idea is to make CMake build for WebRTC m130 version - for audio processing module
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