| commit | 92220ffe9fe31698b9113f9e5347f7bd85b557e4 | [log] [tgz] |
|---|---|---|
| author | oprypin <oprypin@webrtc.org> | Thu Mar 23 03:40:03 2017 -0700 |
| committer | Commit bot <commit-bot@chromium.org> | Thu Mar 23 10:40:03 2017 +0000 |
| tree | 5d6afdb79192f8785ebd70f52031adc094380d1f | |
| parent | 2aa463f721c922ec93bdd3306650b0e24968b92d [diff] |
Low-bandwidth audio testing The C++ part of the test uses CallTest to set up an audio-only call. It reads an audio file, plays it through a FakeAudioDevice which transfers data through a FakeNetworkPipe for another FakeAudioDevice to receive it and write it to a file. Information about these files is printed to stdout. The test cases are meant to try different network and audio configs (more are planned in the future). The Python part of the test runs the C++ part and scans stdout for tests to perform, runs the pairs of files (original and degraded) through the PESQ tool to receive a score and writes that to perf dashboard. BUG=webrtc:7229 NOTRY=True Review-Url: https://codereview.webrtc.org/2694203002 Cr-Commit-Position: refs/heads/master@{#17356}
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. This page is maintained by the Google Chrome team.
See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.