| commit | c7a8b08a7cd8d8f37d7f5fb9930d0cdc74baba35 | [log] [tgz] |
|---|---|---|
| author | solenberg <solenberg@webrtc.org> | Fri Oct 16 21:35:07 2015 |
| committer | Commit bot <commit-bot@chromium.org> | Fri Oct 16 21:35:11 2015 |
| tree | 49320db07164c889ad54905559ac8b72b0b49eed | |
| parent | 9781152e043e35e2f676ddcf5de079c9548f3b37 [diff] |
Add webrtc::AudioSendStream and methods on webrtc::Call to create and delete AudioSendStreams. AudioSendStream will be replacing the send side of VoiceEngine channels and associated APIs. Hence, they will be used transform recorded audio into RTP/RTCP packets that can be transmitted to another party, according to given parameters. BUG=webrtc:4690 Review URL: https://codereview.webrtc.org/1397123003 Cr-Commit-Position: refs/heads/master@{#10307}
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.