| commit | 1836fd6257a692959b3b49ba99ef587ad9995871 | [log] [tgz] |
|---|---|---|
| author | solenberg <solenberg@webrtc.org> | Tue Oct 25 06:44:45 2016 -0700 |
| committer | Commit bot <commit-bot@chromium.org> | Tue Oct 25 13:44:49 2016 +0000 |
| tree | 723aa5f5883386470ade2a93af5db3826d7c3106 | |
| parent | 701d628f5f392a80436bd53cb118800b7845cb9d [diff] |
Clean up logging in AudioSendStream::SetupSendCodec(). As a side effect: - Moved the AudioSendStream::Config::SendCodecSpec methods into the .cc. - Which exposed an issue with event_visualizer_utils not having a dependency on api:call_api set up. - Which further exposed clang warnings about large inlined default methods in webrtc/config.h. BUG=webrtc:4690 Review-Url: https://codereview.webrtc.org/2446963003 Cr-Commit-Position: refs/heads/master@{#14771}
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.