commit | 171df9326200d1e01bce530e2ff01ac5890e6cb7 | [log] [tgz] |
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author | Niels Möller <nisse@webrtc.org> | Thu Jan 24 08:44:14 2019 |
committer | Commit Bot <commit-bot@chromium.org> | Thu Jan 24 10:47:21 2019 |
tree | a1a449752b4994c1991506f168b7c2a2c92db298 | |
parent | 2820d174b5532e134643c681e2c525034f3266a8 [diff] |
Delete RtpUtility::Payload, and refactor RTPSender to not use it Replaced by a payload type --> video codec map in RTPSenderVideo, where it is used to select the right packetizer. Bug: webrtc:6883 Change-Id: I43a635d5135c5d519df860a2f4287a4478870b0f Reviewed-on: https://webrtc-review.googlesource.com/c/119263 Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26380}
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.