| commit | db208317eb8c9275653fb7168484f5d25e34e374 | [log] [tgz] |
|---|---|---|
| author | Jakob Ivarsson <jakobi@webrtc.org> | Fri Jan 27 14:13:22 2023 |
| committer | WebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com> | Fri Jan 27 15:46:32 2023 |
| tree | fcd8fc01bb968a3a129baa0acd086f252cb8d356 | |
| parent | 9d21eb3e3eb279d587f9726a7a03e168b688e340 [diff] |
Update RTP timestamp based on capture timestamp when audio send stream is resumed. This removes the previous approach where we continued to update the timestamp when the capturer is running but the send stream is stopped in favor of a more general approach that also works when the capturer is paused. Some assumptions for this change to be correct: input sample rate and frame size will be the same before/after the stream is paused. Bug: webrtc:12397 Change-Id: I3b03741cd6d3285cbc9aee3893800729852e6cfa Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291526 Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39213}
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.