b732bd5fb5
Add timestamps to the function AudioDeviceBuffer::SetRecordedBuffer. This will be used to store audio timestaps in future changes. This is a part of the A/V sync metric metric feature for mobile. The metric have already launched for web clients. Bug: webrtc:13609 Change-Id: I0031843476ff1b573b262308fca52d587fae30b7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249085 Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Reviewed-by: Minyue Li <minyue@google.com> Commit-Queue: Olov Brändström <brandstrom@google.com> Cr-Commit-Position: refs/heads/main@{#35851}
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: https://ci.chromium.org/p/webrtc/g/ci/console
- Coding style guide
- Code of conduct
- Reporting bugs
- Documentation