| commit | 5fac3f089280bb8f3ad433e8fdf4b52b2d4e0ba5 | [log] [tgz] |
|---|---|---|
| author | henrik.lundin <henrik.lundin@webrtc.org> | Wed Aug 24 18:18:49 2016 |
| committer | Commit bot <commit-bot@chromium.org> | Wed Aug 24 18:18:54 2016 |
| tree | 744917767afc5d3d244b85dfca517a6711bf62b9 | |
| parent | d1a10a0f7795213210f9a9f5720167f97bade8c9 [diff] |
NetEq: Don't check sample rate and frame size upon error If an error happens in the GetAudio call, for instance when corrupt payloads are inserted, GetAudio wil return an error. In this case, the audio frame is not always correctly populated, which is to be expected. BUG=webrtc:5447 Review-Url: https://codereview.webrtc.org/2272963002 Cr-Commit-Position: refs/heads/master@{#13902}
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.