cf327b45b9
WebRTC works on 10ms buffer sizes in both directions but this class has contained support for any size (with some limits) and for changes on the fly. It makes no sense to maintain such code and we have no tests to test it. This CL ensures that only 10ms audio buffers are supported and that nothing can be changed on the fly. It also updates the style to follow the Google C++ style guide. Finally, I remove very old (not tested and not maintained) support for file handling since the code is never used. It was more or less dead code. BUG=NONE R=magjed@webrtc.org Review URL: https://codereview.webrtc.org/2256833003 . Cr-Commit-Position: refs/heads/master@{#13833}
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.