commit | 9c31ac23231a3494a794b3ba0a6b018969eaa7aa | [log] [tgz] |
---|---|---|
author | Alex Loiko <aleloi@webrtc.org> | Fri Feb 15 09:44:28 2019 |
committer | Commit Bot <commit-bot@chromium.org> | Mon Feb 18 17:09:59 2019 |
tree | 97a3f3eb03b27ea4867076da3929ee4fafa88444 | |
parent | f2727fb8d36be5135472fd865833ee9714a5d684 [diff] |
Tests for multi-stream Opus. This CL (mainly) adds bit-exactness tests for multi-stream Opus. The tests are in audio_coding_unittest.cc. Some refactoring of AcmSendTestOldApi, AcmSenderBitExactnessOldApi is done to make it possible. A few checks for "channels \in {1, 2}" are replaced with "channels \in {1, 2, 4, 6, 8}" in the WebRTC Opus codec wrapper. A few other changes are made to be able to write and read multi-channel WAV files. The SDP changes are NOT included; as of this CL there is no way to set up a multi-stream opus en/de-coder from SDP strings. Bug: webrtc:8649 Change-Id: I1d93a9b8eecc3f6e19896ff2e2ce9b63da77a23c Reviewed-on: https://webrtc-review.googlesource.com/c/114883 Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Commit-Queue: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26742}
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.