351173c88c
New tests are: - AudioDeviceTest.StartStopPlayoutWithRealDevice - AudioDeviceTest.StartStopRecordingWithRealDevice (the comments below only affects ADM2 on Windows): When adding these tests it was found that we could hit the same known issue as in https://bugs.chromium.org/p/chromium/issues/detail?id=803056 and the same solution as in Chrome was therefore ported from Chrome to WebRTC. Hence, this change also adds support for core_audio_utility::WaveFormatWrapper to support devices that can return a format where only the WAVEFORMATEX parts is initialized. The old version would only DCHECK for these devices and that could lead to an unpredictable behavior. Tbr: minyue Bug: webrtc:11093 Change-Id: Icb238c5475100f251ce4e55e39a03653da04dbda Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159982 Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Commit-Queue: Henrik Andreassson <henrika@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29824}
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