00bceb1eda
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}
Name: WebRTC URL: http://www.webrtc.org Version: 90 License: BSD License File: LICENSE Description: WebRTC provides real time voice and video processing functionality to enable the implementation of PeerConnection/MediaStream. Third party code used in this project is described in the file LICENSE_THIRD_PARTY.