commit | eb166972591504a230baae416bc37b674bb20c2a | [log] [tgz] |
---|---|---|
author | Karl Wiberg <kwiberg@webrtc.org> | Thu May 16 13:14:01 2019 |
committer | Commit Bot <commit-bot@chromium.org> | Mon May 20 17:33:56 2019 |
tree | 1b024286b09502e4830b5d5c0485b4eb5467d066 | |
parent | 94079f84523b8c12f1732594cb5995a7fdaa9d36 [diff] |
AudioEncoderOpus: Don't mix up sample rate and RTP timestamp rate A later change will allow them to differ. Bug: webrtc:10631 Change-Id: I4e13f41980261990b3bbbc6897cd754369265ca0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137046 Reviewed-by: Minyue Li <minyue@webrtc.org> Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27991}
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.