| commit | b2d1e0d1daf41a158cde8a4519eb81d44603d78c | [log] [tgz] |
|---|---|---|
| author | ossu <ossu@webrtc.org> | Wed Oct 05 14:51:44 2016 |
| committer | Commit bot <commit-bot@chromium.org> | Wed Oct 05 14:51:50 2016 |
| tree | 9fc81d28528ed3ee96301b0a5a7689ba03df9abd | |
| parent | b1fff9264489a6f5680fcecfef9d77a30750b1be [diff] |
Resurrected test_api_audio.cc I'll be doing some changes to code it tests (rtp_receiver_audio, specifically) and want to make sure there are tests in place before I touch anything. Fixed test_api_audio not properly checking payload data. Required a fix to LoopBackTransport in test_api to as to act like the regular audio and video parts of WebRTC and separate payload from header data. Also added a test for CNG and cleaned up constants. BUG=webrtc:5805 Review-Url: https://codereview.webrtc.org/2378403004 Cr-Commit-Position: refs/heads/master@{#14529}
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.