4552e8f2d4
Current VoIP Engine logic stops ADM from polling registered audio channel when caller invokes StopPlay which can leads to incoming RTP to be flushed and undesirable statistics report. Instead, VoipBase::StopPlay should silence the decoded audio sample from NetEq as muted to avoid mixing while allowing it go through prior process for correct ingress statistic values. The ADM stop playing logic will be triggered when all audio channels are released by VoipBase::ReleaseChannel API. Bug: webrtc:12121 Change-Id: I410eea4ea13f93acb465ab162a3c14c9819e2b92 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/191140 Commit-Queue: Tim Na <natim@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32553}
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 here 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
- Reporting bugs