| commit | 6c278491adae678cdd95c71c444a02103012df56 | [log] [tgz] |
|---|---|---|
| author | aleloi <aleloi@webrtc.org> | Thu Oct 20 14:24:39 2016 -0700 |
| committer | Commit bot <commit-bot@chromium.org> | Thu Oct 20 21:24:46 2016 +0000 |
| tree | 74e43399d79b3b6e8ad431cabcaf364dd5dd7499 | |
| parent | 920d30bc743b76d30c6cae7ad4c97af2e6c24e4c [diff] |
Move audio frame memory handling inside AudioMixer. Simplify the AudioMixer::Source interface and update the mixer implementation to the new interface. Instead of asking a mixer source to provide a pointer to an AudioFrame during each mixing iteration, a mixer should supply a pointer to its own AudioFrame. This simplifies lifetime issues as sources do not give away an internal pointer. Implementation: when an audio source is added, the mixer allocates a new AudioFrame. The audio frame is kept together in the internal class SourceStatus together with the audio source pointer until the source is removed. NOTRY=True BUG=webrtc:6346 Review-Url: https://codereview.webrtc.org/2420913002 Cr-Commit-Position: refs/heads/master@{#14713}
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.