| commit | 20cb0c1c85d4571919106019a562bbcbd35041fd | [log] [tgz] |
|---|---|---|
| author | deadbeef <deadbeef@webrtc.org> | Wed Feb 01 20:27:00 2017 -0800 |
| committer | Commit bot <commit-bot@chromium.org> | Thu Feb 02 04:27:00 2017 +0000 |
| tree | 43f6a6d5fb2d77035b54ecc1520faa880c1298f3 | |
| parent | 2e03c6611927af6ed08cf967b95342744ff329b6 [diff] |
Move DTMF sender to RtpSender (as opposed to WebRtcSession). Previously in the spec, there was a createDtmfSender method on PeerConnection, but that's been replaced by a "dtmf" attribute on RtpSender, which allows getting a DTMF sender without having an audio track. This also simplifies the code slightly, since tracks are now not necessary for identification. BUG=webrtc:4180 Review-Url: https://codereview.webrtc.org/2666853002 Cr-Commit-Position: refs/heads/master@{#16409}
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.