dea374a467f196e5ec7dadc45e41e93e654f8472

Normally, packet/frame info is delivered to AudioReceiveStream's source_tracker_ when an audio frame is pulled out of the stream (as a side-effect of GetAudioFrameWithInfo). When playout is muted, though, packets are thrown away in ChannelReceive::OnReceivedPayloadData, so AudioRtpReceiver stops seeing updates to its RtpSources and any related information (e.g. CSRCs and associated timestamps, levels). Skipping the playout path here has a downside of being misaligned with whatever playout delay would normally be, but it allows clients that want to consume RtpSource information to be able to do so while playout is muted. Bug: None Change-Id: Id00566b645de4196c2341611cd9e8b94b35aa157 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203500 Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org> Reviewed-by: Noah Richards <noahric@chromium.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Commit-Queue: Ranveer Aggarwal <ranvr@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33236}
…
…
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
Description
Languages
C++
88.6%
C
3.3%
Java
3%
Objective-C++
1.9%
Python
1.9%
Other
1%