| commit | 1896cece014c6e76c4e2d020388df7ea9704c34a | [log] [tgz] |
|---|---|---|
| author | Sebastian Jansson <srte@webrtc.org> | Tue Feb 20 09:06:11 2018 +0100 |
| committer | Commit Bot <commit-bot@chromium.org> | Tue Feb 20 15:05:57 2018 +0000 |
| tree | b1a171cda0ac10e5198b76d308749ab6b189ce06 | |
| parent | 6bd3cddcef0c3676750a85b84ca0a0662bac756d [diff] |
Removed dependencies from audio send stream unit test The audio send stream unit tests did not use the mocks injected to the fake rtp transport controller send. This CL prepares for removing the fake controller which makes it harder to refactor the rtp transport controller interface. Bug: webrt:8415 Change-Id: I73f7d105e66f9beb80aeaa92f3490cd61c80c5b8 Reviewed-on: https://webrtc-review.googlesource.com/54302 Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Commit-Queue: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22102}
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.
Authoritative list of directories that contain the native API header files.