commit | 1391bd472ce478ffccd0ffdba8a2065649248466 | [log] [tgz] |
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author | Henrik Lundin <henrik.lundin@webrtc.org> | Fri Nov 24 08:28:57 2017 |
committer | Commit Bot <commit-bot@chromium.org> | Fri Nov 24 09:05:48 2017 |
tree | 3ed553905f2ff45c90b1dd70fa9401ce25b6d4ac | |
parent | bb5ca2ef4115c013ba43ad71759d2799840375eb [diff] |
Replacing the legacy tool RTPencode with a new rtp_encode This new tool provides the same functionality as the legacy tool, but it is implemented using AudioCodingModule and AudioEncoder APIs instead of the naked codecs. Bug: webrtc:2692 Change-Id: I29accd77d4ba5c7b5e1559853cbaf20ee812e6bc Reviewed-on: https://webrtc-review.googlesource.com/24861 Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20857}
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.