| commit | ef9daee934107d9ff911a0db3b4582fd6a62d4ec | [log] [tgz] |
|---|---|---|
| author | Sebastian Jansson <srte@webrtc.org> | Thu Feb 22 14:49:02 2018 +0100 |
| committer | Commit Bot <commit-bot@chromium.org> | Thu Feb 22 17:32:25 2018 +0000 |
| tree | 915e6b35c18b0ddeb5d1bbd39dfc923219ed9570 | |
| parent | 89c79383e45f9c46951f74e8971fb9fedfe0137c [diff] |
Using mock transport controller in audio unit tests. Using a mock of rtp transport controller send in audio send stream unit tests. This reduces the dependencies and makes the tests more focused on testing the functionality of audio send stream itself. Bug: webrtc:8415 Change-Id: Ia8d9cf47d93decc74b10ca75a6771f39df658dc2 Reviewed-on: https://webrtc-review.googlesource.com/56600 Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Commit-Queue: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22161}
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.