| commit | 940b6d648fab0e240d15fc1feb4ef45e0f418b5d | [log] [tgz] |
|---|---|---|
| author | solenberg <solenberg@webrtc.org> | Tue Oct 25 18:19:07 2016 |
| committer | Commit bot <commit-bot@chromium.org> | Tue Oct 25 18:19:11 2016 |
| tree | 5ec60fcc9e8863da3705d512571734a0ee26da61 | |
| parent | da389e351878d046f9eb6f305f04537774375f27 [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 Committed: https://crrev.com/1836fd6257a692959b3b49ba99ef587ad9995871 Review-Url: https://codereview.webrtc.org/2446963003 Cr-Original-Commit-Position: refs/heads/master@{#14771} Cr-Commit-Position: refs/heads/master@{#14780}
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.