commit | 22821deb38d4c54920b37ad446eb9855ae666e85 | [log] [tgz] |
---|---|---|
author | Jakob Ivarsson <jakobi@webrtc.org> | Fri Jan 20 21:09:29 2023 |
committer | WebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com> | Mon Jan 23 17:29:06 2023 |
tree | 3f1bed0b8c4e93db68e33261938fad18f41d4000 | |
parent | 6e627290bfea927ea21a552fab64ce85459222ee [diff] |
Make capture timestamp optional in ADM. This is to avoid using 0 as a default value. Also fix a bug in audio_device_buffer where the timestamp aligner used the wrong input timestamp. Bug: webrtc:13609 Change-Id: I00016e68ab50d052990c2b9f80aa1e2d7e167b93 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291118 Reviewed-by: Olov Brändström <brandstrom@google.com> Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org> Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39177}
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.