| commit | 0b69826ffbf74d58fc26834c56aeb57a8ab7ce1c | [log] [tgz] |
|---|---|---|
| author | Sebastian Jansson <srte@webrtc.org> | Thu Mar 07 09:17:19 2019 +0100 |
| committer | Commit Bot <commit-bot@chromium.org> | Thu Mar 07 09:42:26 2019 +0000 |
| tree | 6761f4587e88f102f0e5130dd238ab682494123b | |
| parent | de3360ec1dd38c702415875a81e9fe3bb42cb935 [diff] |
Don't inject worker queue into send streams. This prepares for making AudioSendStream use its own task queue. In the future more of the functionality that depends on running on the task queue is planned to be moved directly into RtpTransportControllerSend. They should instead get it from the transport controller. This affects the media transport tests which previously assumed that the transport controller could be missing. However, this is not something that is used in production, so this is an improvement of the tests as they will behave more like production code. Bug: webrtc:9883 Change-Id: Ie32f4c2f6433ec37ac16a08d531ceb690ea9c0b5 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126000 Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27010}
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.