| commit | 133cff009b7677af4649c282f1e2546da4dba975 | [log] [tgz] |
|---|---|---|
| author | Karl Wiberg <kwiberg@webrtc.org> | Fri Jul 06 13:40:14 2018 |
| committer | Commit Bot <commit-bot@chromium.org> | Wed Aug 08 01:38:05 2018 |
| tree | 4ab04b21687ba85b52ff6c2dfa4538a8705aab46 | |
| parent | 191f46c5c1ce4853b1a05b2b337a85bee204bec7 [diff] |
AudioCodingModuleTest.TestAllCodecs: Create audio encoders the new way Specifically, don't expect the ACM to be able to create encoders; we have to give it an encoder that we make ourselves. To make it work, I had to add support for the "ptime" parameter to the L16 codec. Bug: webrtc:8396 Change-Id: I3869422882611d2eed65d6c849ea7cd3ad6bd126 Reviewed-on: https://webrtc-review.googlesource.com/87423 Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24217}
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.