| commit | deac4dea4fa3fde805c1b988a846e776a62b7b2c | [log] [tgz] |
|---|---|---|
| author | Philipp Hancke <philipp.hancke@googlemail.com> | Fri Jul 23 05:08:33 2021 |
| committer | WebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com> | Tue Aug 03 14:26:02 2021 |
| tree | 3751f1d512a5cd052f0e16a6811ccb02593f4a61 | |
| parent | 8d564722d7c3af1948c345175675106a75b75d3f [diff] |
red: copy audio level from main packet for recovery packet fill the audio level of the recovery packets from the main packet. While not exact, this should be close enough. Without this, the audio level in getStats() will be filled but the audio level in getSynchronizationSources() will be empty. In chrome this is easy to test, the audio level graph on https://webrtc.github.io/samples/src/content/peerconnection/audio/ will be empty all the time prior to this fix. BUG=webrtc:11640 Change-Id: Ia1e61fd1852445239021a76d08032120a92d83aa Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226840 Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34635}
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.