c35c7dedc0
All of the buffer size returned by Windows Core Audio APIs are in unit of audio frames (which is sample times number of channels), while WebRTC's AudioDeviceBuffer RequestPlayoutData method takes in samples per channel (equivalent to frames per channel) but returns number of audio samples in all the channels. This CL makes sure that we compare playout block size in frames with frames and size in samples with samples, which should fix the excessive logging issues and audio quality problems due to the mismatch when comparing. BUG=webrtc:7797 Review-Url: https://codereview.webrtc.org/2933953003 Cr-Commit-Position: refs/heads/master@{#18546}
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.
More info
- Official web site: http://www.webrtc.org
- Master source code repo: https://chromium.googlesource.com/external/webrtc
- 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