commit | 1aee0b5bd98eade203d8d4c0a05a85414ad15896 | [log] [tgz] |
---|---|---|
author | maxmorin <maxmorin@webrtc.org> | Mon Aug 15 11:46:19 2016 -0700 |
committer | Commit bot <commit-bot@chromium.org> | Mon Aug 15 18:46:28 2016 +0000 |
tree | 0cbc5c86fdf41ffe90f044ac1489c93d2f37762e | |
parent | c8c71f484ecae4a77117221af82a59adaf0dc2b9 [diff] |
Remove old methods in AudioTransport, make it pass a gn build when building with default warnings. This is in preparation for making a gn target for audio_device_tests. BUG=webrtc:6170, webrtc:163 NOTRY=True Review-Url: https://codereview.webrtc.org/2219653004 Cr-Commit-Position: refs/heads/master@{#13759}
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.