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@hauck-jvsh noticed we are converting audios to 16Khz, but without resampling, what can give some unexpected results depending on original frequency. We should use mplayer resample parameter to guarantee audio samples are taken equidistantly in time, using interpolation in resampling if necessary.
The text was updated successfully, but these errors were encountered:
@hauck-jvsh noticed we are converting audios to 16Khz, but without resampling, what can give some unexpected results depending on original frequency. We should use mplayer resample parameter to guarantee audio samples are taken equidistantly in time, using interpolation in resampling if necessary.
The text was updated successfully, but these errors were encountered: