| commit | 6d8e011b648900f24c3c7b4488d3694ce4deca70 | [log] [tgz] |
|---|---|---|
| author | henrik.lundin <henrik.lundin@webrtc.org> | Fri Mar 04 18:34:21 2016 |
| committer | Commit bot <commit-bot@chromium.org> | Fri Mar 04 18:34:26 2016 |
| tree | 1a49c1c70d634f11c2eb4eee3e7be1abf5a64fd0 | |
| parent | 6459f84766dd6ece67e70c8743aeb2378ac267be [diff] |
Change NetEq::GetAudio to use AudioFrame With this change, NetEq now uses AudioFrame as output type, like the surrounding functions in ACM and VoiceEngine already do. The computational savings is probably slim, since one memcpy is removed while another one is added (both in AcmReceiver::GetAudio). More simplifications and clean-up will be done in AcmReceiver::GetAudio in future CLs. BUG=webrtc:5607 Review URL: https://codereview.webrtc.org/1750353002 Cr-Commit-Position: refs/heads/master@{#11874}
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.