| commit | 9fea80f50daab46f20d4a6fc67b0144fbbbf56cd | [log] [tgz] |
|---|---|---|
| author | Stefan Holmer <stefan@webrtc.org> | Thu Jan 07 16:43:18 2016 |
| committer | Stefan Holmer <stefan@webrtc.org> | Thu Jan 07 16:43:31 2016 |
| tree | bfc21d8098a47a6948e5678f7328e2c8debdc9c2 | |
| parent | ecd21b481fc2f01b19044b23856ef6b95826f237 [diff] |
Add audio streams to CallTest and a first A/V call test. Add audio send and receive streams to CallTest and call the necessary voice engine APIs for the streams to be usable. Verifies the implementation by adding a simple test which monitors outgoing packets and checks that both audio and video is being sent with transport sequence numbers. Audio streams are using a fake audio device with file input. The CallTest implementation is to a big degree based on call_perf_tests.cc and should in the future replace a lot of that code. R=pbos@webrtc.org TBR=kjellander@webrtc.org BUG=webrtc:5263 Review URL: https://codereview.webrtc.org/1542653002 . Cr-Commit-Position: refs/heads/master@{#11171}
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.