789da89d37
The special cases for packet sizes <= 10 ms (one case for each sample rate) resulted in reading outside of the pw16_decoded vector. This is now fixed by making sure that WebRtcSpl_DownsampleFast gets correct input and output vector lengths. Review URL: http://webrtc-codereview.appspot.com/295008 git-svn-id: http://webrtc.googlecode.com/svn/trunk@1027 4adac7df-926f-26a2-2b94-8c16560cd09d |
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audio_coding | ||
audio_conference_mixer | ||
audio_device | ||
audio_processing | ||
interface | ||
media_file | ||
rtp_rtcp | ||
udp_transport | ||
utility | ||
video_capture | ||
video_coding | ||
video_processing/main | ||
video_render | ||
modules.gyp |