| commit | 63fb95a68d70cd7036db65c3378e91cb77faa44d | [log] [tgz] |
|---|---|---|
| author | ossu <ossu@webrtc.org> | Wed Jul 06 16:34:22 2016 |
| committer | Commit bot <commit-bot@chromium.org> | Wed Jul 06 16:34:31 2016 |
| tree | 5d75d94938b97bc5a3883f80af92c335e4976937 | |
| parent | 1d4fefbbaf056492096e9e8a689550c6b7c49fe9 [diff] |
Fixed time moving backwards in the AudioCodingModule. There was a fast path in PreprocessToAddData that would just use the input timestamps if the input format was equal to the required format of the encoder. This works well as long as the codec never changes. If we are first doing resampling (specifically upsampling) and then change to a codec that does not require resampling, we'll need to stick to whatever input timestamp we left off at, rather than silently accepting whatever we're sent. BUG=622435 Review-Url: https://codereview.webrtc.org/2119393002 Cr-Commit-Position: refs/heads/master@{#13398}
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.