ff571c60a9
Internal counters in the RenderDelayBuffer can slip out of sync with external counters, leading to buffer misalignment. This CL gives the RenderDelayBuffer an opportunity to update its counters. Tested: Passes: modules_unittests --gtest_filter=BlockProcessor.* Fails as expected due to new unit test: modules_unittests --gtest_filter=BlockProcessor.* --force_fieldtrials="WebRTC-Aec3RenderBufferCallCounterUpdateKillSwitch/Enabled/" audioproc_f with default AEC settings has been verified to be bit-exact on a large number of aecdumps. Bug: webrtc:11803 Change-Id: I9363b834c8c8c934add0335013df60bf131da4bc Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180126 Reviewed-by: Per Åhgren <peah@webrtc.org> Commit-Queue: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31795}
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