| commit | 61a208b1b8e88716747971fe4ba1da8ddf521bb1 | [log] [tgz] |
|---|---|---|
| author | ossu <ossu@webrtc.org> | Tue Sep 20 08:38:00 2016 |
| committer | Commit bot <commit-bot@chromium.org> | Tue Sep 20 08:38:09 2016 |
| tree | 8832f407de8efa5e1d4c2c2cd755705e83c3a9e4 | |
| parent | 02bd5125e9a90f1516b83ce001a7262869a096ef [diff] |
Added a ParsePayload method to AudioDecoder. It allows the decoder to split the input up into usable chunks before they are put into NetEq's PacketBuffer. Eventually, all packet splitting will move into ParsePayload. There's currently a base implementation of ParsePayload. It will generate a single Frame that calls the underlying AudioDecoder for getting Duration() and to Decode. BUG=webrtc:5805 BUG=chromium:428099 Review-Url: https://codereview.webrtc.org/2326953003 Cr-Commit-Position: refs/heads/master@{#14300}
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.