[FFmpeg-user] Multiply audio samples

[Paul B Mahol]

On 9/16/18, Michael Koch <astroelectronic at t-online.de> wrote:
>> Filter inputs will be auto-converted from any non-float formats to
>> float or double one. Float/double formats are in [-1 .. +1] interval.
>
> Thank you very much for the amultiply filter, it works great!
>
> Here is an example for an ultrasonic converter which downconverts the
> 15kHz - 25kHz band to the 0 - 10kHz band.
>
> First make an input file for testing. This is a 2 second 19kHz tone
> followed by 2 seconds silence:
>
> c://ffmpeg/ffmpeg -f lavfi -i
> "sine=frequency=19000:sample_rate=48000:duration=2" -af apad -t 4
> ultrasonic.wav
>
> Then use the downconverter:
>
> c://ffmpeg/ffmpeg -i ultrasonic.wav -f lavfi -i
> "sine=frequency=15000:sample_rate=48000" -filter_complex
> "[0]highpass=f=15000,highpass=f=15000,highpass=f=15000,highpass=f=15000[sound];[1]volume=8[sine];[sound][sine]amultiply[mixed];[mixed]lowpass=f=10000,lowpass=f=10000,lowpass=f=10000,lowpass=f=10000"
> out.wav

Nice, it might be better to use aevalsrc, because it have multichannel output.

For stronger highpass/lowpass you can use aiir filter if you know
correct parameters.