| commit | bc609eaab1f6811ac6176cb3e2516961b7f6c29e | [log] [tgz] |
|---|---|---|
| author | Emircan Uysaler <emircan@webrtc.org> | Tue Mar 27 21:57:18 2018 |
| committer | Commit Bot <commit-bot@chromium.org> | Tue Mar 27 23:01:55 2018 |
| tree | 1c535fa9cc3ad8f9edd255018c8c9646ea2fb466 | |
| parent | ca0fcc2df9ca19dc7df3e3fdf732cc1b76e79337 [diff] |
Revert "Adds support for multiple or no media stream ids." This reverts commit 1550292efe680ac79a18004705c908b1cdca54cb. Reason for revert: webkit_layout_tests:fast/peerconnection/RTCPeerConnection-sdpSemantics.html is broken, see below. WebRTC roll isn't going through because of it. This CL looks the most suspicious within the 5 in the range. https://chromium-review.googlesource.com/c/chromium/src/+/981899 https://webrtc.googlesource.com/src.git/+log/bb50ce5bb6d5..27f3bf512827 https://ci.chromium.org/p/chromium/builders/luci.chromium.try/linux_chromium_rel_ng/54616 Original change's description: > Adds support for multiple or no media stream ids. > > With Unified Plan SDP semantics, this adds support for specifying > either no media stream ids or multiple media stream ids for a > transceiver/sender/receiver. This includes serializing/deserializing > SDPs with multiple a=msid lines in a m section, or an "a=msid:- > <appdata>" line to indicate the no stream case. Note that this does > not synchronize between multiple streams, this is still just supported > based upon the first media stream id. > > Bug: webrtc:7932, webrtc:7933 > Change-Id: Ib7433929af7b2925abe2824b485b360cec12f275 > Reviewed-on: https://webrtc-review.googlesource.com/61341 > Commit-Queue: Seth Hampson <shampson@webrtc.org> > Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> > Reviewed-by: Steve Anton <steveanton@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#22611} TBR=steveanton@webrtc.org,deadbeef@webrtc.org,shampson@webrtc.org # Not skipping CQ checks because original CL landed > 1 day ago. Bug: webrtc:7932, webrtc:7933 Change-Id: I1d4e4332b518ec32b305c8af07679650059d02bb Reviewed-on: https://webrtc-review.googlesource.com/65000 Reviewed-by: Emircan Uysaler <emircan@webrtc.org> Commit-Queue: Emircan Uysaler <emircan@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22634}
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.