commit | 4f26a3c7e8e20e0e0ca4ca67a6ebdf3f5543dc3f | [log] [tgz] |
---|---|---|
author | Philipp Hancke <philipp.hancke@googlemail.com> | Thu May 27 11:47:47 2021 |
committer | WebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com> | Mon May 31 13:35:44 2021 |
tree | 0f62ad0f2f98801dcce2107175aa002e7db171db | |
parent | 5d4c3c51282b2177ae9d027d20d3e09c2162304a [diff] |
red: assign payload type 63 to audio/RED for opus Starting new audio codecs from the top of the lower range reduces collisions with video codecs which are assigned from the bottom of the lower range BUG=webrtc:11640 Change-Id: If6d2b849b8e1de777a1d4352df533e4f1845fde9 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/220022 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34166}
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 here for instructions on how to get started developing with the native code.
Authoritative list of directories that contain the native API header files.