Previously, AudioEncoderCng required the speech encoder to not change its mind regarding the number of 10 ms frames in the next packet between calls to AudioEncoderCng::EncodeInternal()---specifically, it could handle an upward but not a downward adjustment. With this patch, it can handle a downward adjustment too, by simply saving the overshoot data for the next call to EncodeInternal(). It will still not handle the case where the encoder's reported number of 10 ms frames in the next packet is inconsistent with the behavior of its Encode() function when called with no intervening changes to the encoder. R=henrik.lundin@webrtc.org Review URL: https://webrtc-codereview.appspot.com/53469005 Cr-Commit-Position: refs/heads/master@{#9261}
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.