commit | 6298b5689048c81e1aa711d980b7c5e1248f676c | [log] [tgz] |
---|---|---|
author | Sebastian Jansson <srte@webrtc.org> | Tue Jan 14 16:55:19 2020 |
committer | Commit Bot <commit-bot@chromium.org> | Wed Jan 15 19:01:50 2020 |
tree | 94781d9652a7ca53ddf67b2c42c8b924070eb90c | |
parent | c2509fec7c0cff95612a837e043032b5c85322c5 [diff] |
Cleanup: Using RtpRtcp directly from AudioSendStream This reduces indirection and makes it easier to follow code. It also fits into a long term strategy of reducing the scope of ChannelSend. Bug: webrtc:9883 Change-Id: I2661c4aa6c561f7691beaaa289636254f7a58b72 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166042 Reviewed-by: Sam Zackrisson <saza@webrtc.org> Commit-Queue: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30273}
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.