| commit | ffbbcac4c6111095a11536b96b600c6813444562 | [log] [tgz] |
|---|---|---|
| author | solenberg <solenberg@webrtc.org> | Thu Nov 17 05:25:37 2016 -0800 |
| committer | Commit bot <commit-bot@chromium.org> | Thu Nov 17 13:25:45 2016 +0000 |
| tree | ca618bb6d3a4f1230f0a621b0656613f84d0f650 | |
| parent | 2779bab02a3c9e576cdb9927ef86d3fc40772bdc [diff] |
Support multiple timestamp rates for sending DTMF. We support multiple payload types, and one which matches the audio codec the closest, is picked (or the one with lowest clock rate, if no perfect match is found). The exact clock rate is then ignored and DTMF packets are time stamped with the rate of the current audio codec. This is exactly the way the code has worked up to this point, but until now we have been under the impression that we were in fact sending 8k DTMF. In other words, this is an improvement over the current situation, since we will most likely find a payload type which matches the codec clock rate. This CL also does a little cleaning of the DTMFQueue and RTPSenderAudio classes. BUG=webrtc:2795 Review-Url: https://codereview.webrtc.org/2392883002 Cr-Commit-Position: refs/heads/master@{#15129}
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.