c35c7dedc0
All of the buffer size returned by Windows Core Audio APIs are in unit of audio frames (which is sample times number of channels), while WebRTC's AudioDeviceBuffer RequestPlayoutData method takes in samples per channel (equivalent to frames per channel) but returns number of audio samples in all the channels. This CL makes sure that we compare playout block size in frames with frames and size in samples with samples, which should fix the excessive logging issues and audio quality problems due to the mismatch when comparing. BUG=webrtc:7797 Review-Url: https://codereview.webrtc.org/2933953003 Cr-Commit-Position: refs/heads/master@{#18546}
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.