| commit | c62f6c712175244db6b1e761f4b7afb6b3b75147 | [log] [tgz] |
|---|---|---|
| author | Karl Wiberg <kwiberg@webrtc.org> | Wed Oct 04 10:38:53 2017 |
| committer | Commit Bot <commit-bot@chromium.org> | Wed Oct 04 11:30:14 2017 |
| tree | 645da88d886bfa8fd5ed2616a7b096337519087f | |
| parent | 83ccca1864fc4da6f98761223946694a437532d6 [diff] |
RTPPayloadRegistry: Use SdpAudioFormat to represent audio codecs This is needed in the general case, now that we aim to support codecs other than those built-in to WebRTC. BUG=webrtc:8159 Change-Id: I40a41252bf69ad5d4d0208e3c1e8918da7394706 Reviewed-on: https://webrtc-review.googlesource.com/5380 Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20136}
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.
See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.