| commit | b572768efbc1e52b97a5ad98932c667956aba4b8 | [log] [tgz] |
|---|---|---|
| author | Fredrik Solenberg <solenberg@webrtc.org> | Fri Dec 04 15:22:19 2015 +0100 |
| committer | Fredrik Solenberg <solenberg@webrtc.org> | Fri Dec 04 14:22:30 2015 +0000 |
| tree | 64ff145b6a47859a8860218c2748196cc32075a0 | |
| parent | fcdcf4a92796bd915f0abb343f99f79256ec7d38 [diff] |
- Remove calls to VoEDtmf from WVoE/MC. - Flatten logic and make the relevant calls on VoE::Channel from AudioSendStream::SendTelephoneEvent(). - Store current payload type for telephone events in WVoMC, instead of setting it on the Channel. This should be refactored to be an AudioSendStream::Config parameter when we redo WVoMC::SetSendCodecs(). BUG=webrtc:4690 R=pthatcher@webrtc.org, tina.legrand@webrtc.org Review URL: https://codereview.webrtc.org/1491743004 . Cr-Commit-Position: refs/heads/master@{#10895}
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.