commit | 191bf5c6534da633c91d460798cf607b25a09294 | [log] [tgz] |
---|---|---|
author | Tomas Gunnarsson <tommi@chromium.org> | Fri Mar 30 10:44:43 2018 |
committer | Commit Bot <commit-bot@chromium.org> | Fri Mar 30 10:44:53 2018 |
tree | 0eabb624694d3e63f679a57365959f44966f8f06 | |
parent | ef3e28a2b7b10e9043a50b4852593fe524107c51 [diff] |
Revert "Reland "Adds support for multiple or no media stream ids."" This reverts commit f351c3408a0c7f695447a2a9f4e6a1719a0d6a26. Reason for revert: Breaks chromium import https://ci.chromium.org/p/chromium/builders/luci.chromium.try/linux_chromium_rel_ng/58012 Failin tests: WebRtcRtpBrowserTest.TrackAddedToSecondStream WebRtcRtpBrowserTest.TrackSwitchingStream 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=steveanton@webrtc.org,deadbeef@webrtc.org,shampson@webrtc.org Change-Id: I1835419f963762bc308a91d81c423d8e7bf65026 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:7932, webrtc:7933 Reviewed-on: https://webrtc-review.googlesource.com/65700 Reviewed-by: Tomas Gunnarsson <tommi@chromium.org> Commit-Queue: Tomas Gunnarsson <tommi@chromium.org> Cr-Commit-Position: refs/heads/master@{#22690}
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.