commit | 1ff16c87aa63d4f9b63a238fc0664b5e625e9884 | [log] [tgz] |
---|---|---|
author | Guido Urdaneta <guidou@webrtc.org> | Mon May 20 19:31:53 2019 +0200 |
committer | Commit Bot <commit-bot@chromium.org> | Mon May 20 18:38:06 2019 +0000 |
tree | 17fbbb8886385f16b74363e533068bc06456f661 | |
parent | eb166972591504a230baae416bc37b674bb20c2a [diff] |
Add RtpSenderInterface.SetStreams This is a reland of df5731e44d510e9f23a35b77e9e102eb41919bf4 with fixes to avoid existing chromium tests to fail. Instead of replacing the existing RtpSender::set_stream_ids() to also fire OnRenegotiationNeeded(), this CL keeps the old set_stream_ids() and adds the new RtpSender::SetStreams() which sets the stream IDs and fires the callback. This allows existing callsites to maintain behavior, and reserve SetStreams() for the cases when we want OnRenegotiationNeeded() to fire. Using the SetStreams() name instead of SetStreamIDs() to match the W3C spec and to make it more different that the existing set_stream_ids(). Original change's description: > Improve spec compliance of SetStreamIDs in RtpSenderInterface > > This CL makes RtpSender::SetStreamIDs fire fire negotiationneeded > event if needed and exposes the method on RtpSenderInterface. > > This is a spec-compliance change. > > Bug: webrtc:10129 > Change-Id: I2b98b92665c847102838b094516a79b24de0e47e > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135121 > Commit-Queue: Guido Urdaneta <guidou@webrtc.org> > Reviewed-by: Steve Anton <steveanton@webrtc.org> > Reviewed-by: Henrik Boström <hbos@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#27974} Bug: webrtc:10129 Change-Id: Ic0b322bfa25c157e3a39465ef8b486f898eaf6bd Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137439 Commit-Queue: Guido Urdaneta <guidou@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27992}
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.