| commit | 00bceb1eda186848a7883b43d024d42a67d724b5 | [log] [tgz] |
|---|---|---|
| author | ossu <ossu@webrtc.org> | Fri Dec 02 02:40:02 2016 -0800 |
| committer | Commit bot <commit-bot@chromium.org> | Fri Dec 02 10:40:12 2016 +0000 |
| tree | 6e2b6b9fde4f4ae095e7a05ad1fdec644050e889 | |
| parent | e066b302ab18c2b4a9545b1fb907b9d65c72a4b8 [diff] |
Deprecated SetAudioPacketSize from RTPSender and removed calls to it. The packet size was only used to control how often to output DTMF packets. However, it likely did not work as intended, since that interval was only set during initialization. No changes to the packet size, like what AudioEncoder::Num10MsFramesInNextPacket could indicate, were picked up. The value was instead taken from an entry in ACMCodecDB. Since it was not-fully-functional, its exact value didn't seem to matter and it was getting in the way of making it possible to supply an external audio encoder factory, I've decided to remove it altogether. The DTMF code now uses an interval of 50 ms regardless, which is a value recommended by the RFC. BUG=webrtc:5806 Review-Url: https://codereview.webrtc.org/2545753002 Cr-Commit-Position: refs/heads/master@{#15380}
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.