commit | ee5ccbc57fcbd14032ec28dd1bbd74f1baaf4324 | [log] [tgz] |
---|---|---|
author | Niels Möller <nisse@webrtc.org> | Wed Mar 06 16:47:29 2019 +0100 |
committer | Commit Bot <commit-bot@chromium.org> | Wed Mar 06 17:15:00 2019 +0000 |
tree | b890a364a7b51e7bc17ba65636313f5b72e6840a | |
parent | 232b3fda921a475e873f09cc58fbc8ceffdbe4ac [diff] |
Move ownership of RTPSenderAudio to ChannelSend. This change takes out responsibility for packetization from the RtpRtcp class, and deletes the method RtpRtcp::SendOutgoingData. Video packetization was similarly moved in cl https://webrtc-review.googlesource.com/c/src/+/123187 Bug: webrtc:7135 Change-Id: I0953125a5ca22a2ce51761b83693e0bb8ea74cd8 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125721 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27000}
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.