| commit | b2328d11dcc86fba1661ee3fa0d51fc126939764 | [log] [tgz] |
|---|---|---|
| author | aluebs <aluebs@webrtc.org> | Tue Jan 12 04:32:29 2016 |
| committer | Commit bot <commit-bot@chromium.org> | Tue Jan 12 04:32:32 2016 |
| tree | 54514d04c1037e1bce85076e3c30ba6c13c469b0 | |
| parent | e93ad1b12913981eaf2c8ba278921a30167bf77f [diff] |
Remove additional channel constraints when Beamforming is enabled in AudioProcessing The general constraints on number of channels for AudioProcessing is: num_in_channels == num_out_channels || num_out_channels == 1 When Beamforming is enabled and additional constraint was added forcing: num_out_channels == 1 This artificial constraint was removed by adding upmixing support in CopyTo, since it was already supported for the AudioFrame interface using InterleaveTo. Review URL: https://codereview.webrtc.org/1571013002 Cr-Commit-Position: refs/heads/master@{#11215}
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. This page is maintained by the Google Chrome team.
See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.