| commit | 2fe1cb0f0acb66e6f8df47365aac816cb69eb911 | [log] [tgz] |
|---|---|---|
| author | andrew <andrew@webrtc.org> | Fri Nov 27 17:27:35 2015 -0800 |
| committer | Commit bot <commit-bot@chromium.org> | Sat Nov 28 01:27:40 2015 +0000 |
| tree | 3506658fb7f97c94db4e3f689a7a48d2d7fbea77 | |
| parent | 7e43138c0890bd99f627fa061b122c8d5716a99d [diff] |
Don't overwrite audio stats when they're not available. Chromium implements AudioProcessorInterface::GetStats(), but other clients may not. The existing stats were getting overwritten with default AudioProcessorStats values in that case. Now, we only overwrite the stats if the track has an AudioProcessorInterface. Also, move signal level out of SetAudioProcessingStats() to avoid the "don't set if it's -1" pattern. Review URL: https://codereview.webrtc.org/1469803004 Cr-Commit-Position: refs/heads/master@{#10831}
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.