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https://github.com/danog/libtgvoip.git
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5caaaafa42
I'm now using the entire audio processing module from WebRTC as opposed to individual DSP algorithms pulled from there before. Seems to work better this way. |
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.. | ||
include | ||
vad_core.c | ||
vad_core.h | ||
vad_filterbank.c | ||
vad_filterbank.h | ||
vad_gmm.c | ||
vad_gmm.h | ||
vad_sp.c | ||
vad_sp.h | ||
vad.cc | ||
webrtc_vad.c |