commit | 10aeb2a5dc2e84db0eea0ebd64384be3834ca920 | [log] [tgz] |
---|---|---|
author | Piotr (Peter) Slatala <psla@webrtc.org> | Wed Nov 14 18:57:24 2018 |
committer | Commit Bot <commit-bot@chromium.org> | Thu Nov 15 15:15:09 2018 |
tree | ab66d9426e1965c756d1d82f465174fa8d3785eb | |
parent | 0462948c9cb959cc36e33d129b1d4315f55accc1 [diff] |
MediaTransportTests should use audio-only peer connection. Currently (and this has to change), media transport is created two times if audio&video is used (even if bundling is enabled). The second time it's destroyed really quickly (but given lack of 'Connect' method, the connection has already started). This change adds a TODO and modifies existing tests to prevent creation of 2 media transports. Bug: webrtc:9719 Change-Id: I872e98dcd10685beb0326d501f0e0abf36c0fdfc Reviewed-on: https://webrtc-review.googlesource.com/c/110887 Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Commit-Queue: Peter Slatala <psla@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25660}
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.