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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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.. | ||
common.h | ||
gmm.cc | ||
gmm.h | ||
noise_gmm_tables.h | ||
pitch_based_vad.cc | ||
pitch_based_vad.h | ||
pitch_internal.cc | ||
pitch_internal.h | ||
pole_zero_filter.cc | ||
pole_zero_filter.h | ||
standalone_vad.cc | ||
standalone_vad.h | ||
vad_audio_proc_internal.h | ||
vad_audio_proc.cc | ||
vad_audio_proc.h | ||
vad_circular_buffer.cc | ||
vad_circular_buffer.h | ||
voice_activity_detector.cc | ||
voice_activity_detector.h | ||
voice_gmm_tables.h |