| commit | b4977de306bd80dbf8c20930943c318f01e037ac | [log] [tgz] |
|---|---|---|
| author | Alex Loiko <aleloi@webrtc.org> | Mon Jan 28 15:38:38 2019 |
| committer | Commit Bot <commit-bot@chromium.org> | Tue Jan 29 12:43:23 2019 |
| tree | 4d0f19db2c12bb74e22cdef568ab2d24d07a3256 | |
| parent | 7a3e43a5d7920438b4a3d2b8d983db5ff429cb88 [diff] |
Receive-side ready for multiple channels. Made path from NetEq to AudioTransport ready for many-channel audio. If there is one stream, we can handle anything that fits in an AudioFrame. For many streams, the current limit is 6. Some multi-channel combinations are not supported: e.g. if we get stereo audio and attempt to play out 6 channels. Changes: * AudioFrameOperations - replaced the MonoTo* and *ToMono methods by UpmixChannels & DownmixChannels. * AudioMixer: removed DCHECKs for <= 2 channels and tweaked the mixing algorithm to handle many channels. Bug: webrtc:8649 Change-Id: Ib83e16d463694e35658caa09c27849e853d508fb Reviewed-on: https://webrtc-review.googlesource.com/c/106040 Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Commit-Queue: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26446}
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.
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.