| commit | 8886c816582a7c6190c5429222cb8096fca302a6 | [log] [tgz] |
|---|---|---|
| author | solenberg <solenberg@webrtc.org> | Wed Mar 09 11:32:44 2016 |
| committer | Commit bot <commit-bot@chromium.org> | Wed Mar 09 11:32:53 2016 |
| tree | f336c49e6656900039384139fc353aed240ae5be | |
| parent | 16daaa5a407e0b7c6918ee0974d1b00c5a7f0db3 [diff] |
- Clean up unused voice engine DTMF code following removal of VoEDtmf APIs. - Use better types in AudioSendStream::SendTelephoneEvent() and related methods. BUG=webrtc:4690 Review URL: https://codereview.webrtc.org/1722253002 Cr-Commit-Position: refs/heads/master@{#11927}
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.