commit | b6bf0b2546f3f5eeec88112431c8a58e86a2e19a | [log] [tgz] |
---|---|---|
author | Danil Chapovalov <danilchap@webrtc.org> | Tue Jan 28 17:36:57 2020 |
committer | Commit Bot <commit-bot@chromium.org> | Tue Jan 28 19:26:28 2020 |
tree | a9f636c8f4473910adf96a5cec47a26c7cf9ace5 | |
parent | f417238217396b52fd1e03a7ed6439e03cbc43f8 [diff] |
Pass picture_id from generic packetizer through codec-specific field To free up RtpVideoHeader::generic field for codec agnostic details from an rtp header extension. Bug: webrtc:10342 Change-Id: I7b9d869b2ecfedb96dfd860be47ed8dffa058749 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166175 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30396}
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.