commit | 6a0aad926044c2ae4eed6c26268e3f7723ea7736 | [log] [tgz] |
---|---|---|
author | Artem Titov <titovartem@webrtc.org> | Mon Apr 15 08:35:51 2019 |
committer | Commit Bot <commit-bot@chromium.org> | Mon Apr 15 09:22:28 2019 |
tree | 28f9049b1fc2f712e43085267a305e19213935d6 | |
parent | 9466b66ed9ecf58fafa952baf50cc4cdb4d3c991 [diff] |
Temporary switch back to generated audio to fix test flakes Wav file capturer won't repeat file or produce silence after file end and WebRTC pipeline will crash in such case. In future we need to make it possible to continue audio after file was ended to behalf in the same way as video files capturer. Bug: webrtc:10138 Change-Id: I35f5bd33790cd430a56002a44af0abb894a96d29 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132795 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27609}
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.