| commit | 5b4f075f9c599d49785430e8619f47b7cff1c102 | [log] [tgz] |
|---|---|---|
| author | Seth Hampson <shampson@webrtc.org> | Mon Apr 02 16:31:36 2018 -0700 |
| committer | Commit Bot <commit-bot@chromium.org> | Tue Apr 03 01:10:07 2018 +0000 |
| tree | 35962d20233ebe29de8eb8f112818ab171d2efe4 | |
| parent | 3d954a6962e5084d428f61b36bf189a9f9c9b0e9 [diff] |
Reland "Reland "Adds support for multiple or no media stream ids."" This is a reland of f351c3408a0c7f695447a2a9f4e6a1719a0d6a26 Reland history: The original CL broke tests in chromium which were manually tested in the first reland. Another small fix was added to the reland to fix a downstream bug, which caused separate tests to fail in chromium. These were not caught because the chromium trybot was down. These are temporarily disabled in chrome to allow this change to roll in. Original change's description: > Reland "Adds support for multiple or no media stream ids." > > This is a reland of 1550292efe680ac79a18004705c908b1cdca54cb > > 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} > > Bug: webrtc:7932, webrtc:7933 > Change-Id: Ica272ac18088103e65cccf6b96a6d3ecccb178ed > Reviewed-on: https://webrtc-review.googlesource.com/65560 > 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@{#22687} TBR=deadbeef@webrtc.org Bug: webrtc:7932, webrtc:7933 Change-Id: Ideb30219b2f952dd51428cd4e8bd43ef49df5b17 Reviewed-on: https://webrtc-review.googlesource.com/66280 Commit-Queue: Seth Hampson <shampson@webrtc.org> Reviewed-by: Seth Hampson <shampson@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22699}
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.