[FFmpeg-user] down sampling

[Paul B Mahol]

On 12/28/18, Michael Koch <astroelectronic at t-online.de> wrote:
> Am 28.12.2018 um 10:18 schrieb Carl Eugen Hoyos:
>> 2018-12-28 10:14 GMT+01:00, Michael Koch <astroelectronic at t-online.de>:
>>> Am 28.12.2018 um 00:50 schrieb Moritz Barsnick:
>>>> On Thu, Dec 27, 2018 at 20:29:34 +0100, Michael Koch wrote:
>>>>> I think the bigger problem is "outputting through the computers
>>>>> speakers". As far as I know it depends on the operating system, and
>>>>> under Windows it's impossible.
>>>> You can always pipe to ffplay, which plays audio also under Windows
>>>> (using SDL audio).
>>> I did try that some time ago, without success.
>> What did you try? (Command line and complete, uncut console output
>> missing.)
>
> Below is the console output. It's an ultrasonic converter and it works
> fine when I send the output to a file.
>
> F:\Sound\Ultraschall_Konvertierung>c://ffmpeg/ffmpeg -f dshow -channels
> 2 -i aud
> io="Mikrofon (Realtek High Definiti" -f lavfi -i
> aevalsrc="sin(3000*2*PI*t):c=st
> ereo:s=44100" -filter_complex
> "[0]volume=3,highpass=f=3000,highpass=f=3000,highp
> ass=f=3000,highpass=f=3000[sound];[sound][1]amultiply,lowpass=f=10000,lowpass=f=
> 10000,lowpass=f=10000,lowpass=f=10000" -t 10 -f mp3 pipe:play -
> //ffmpeg/ffplay
> -i pipe:play
> ffmpeg version N-91960-g63c69d51c7 Copyright (c) 2000-2018 the FFmpeg
> developers
>
>    built with gcc 8.2.1 (GCC) 20180813
>    configuration: --enable-gpl --enable-version3 --enable-sdl2
> --enable-fontconfi
> g --enable-gnutls --enable-iconv --enable-libass --enable-libbluray
> --enable-lib
> freetype --enable-libmp3lame --enable-libopencore-amrnb
> --enable-libopencore-amr
> wb --enable-libopenjpeg --enable-libopus --enable-libshine
> --enable-libsnappy --
> enable-libsoxr --enable-libtheora --enable-libtwolame --enable-libvpx
> --enable-l
> ibwavpack --enable-libwebp --enable-libx264 --enable-libx265
> --enable-libxml2 --
> enable-libzimg --enable-lzma --enable-zlib --enable-gmp
> --enable-libvidstab --en
> able-libvorbis --enable-libvo-amrwbenc --enable-libmysofa
> --enable-libspeex --en
> able-libxvid --enable-libaom --enable-libmfx --enable-amf
> --enable-ffnvcodec --e
> nable-cuvid --enable-d3d11va --enable-nvenc --enable-nvdec
> --enable-dxva2 --enab
> le-avisynth
>    libavutil      56. 19.101 / 56. 19.101
>    libavcodec     58. 30.100 / 58. 30.100
>    libavformat    58. 18.101 / 58. 18.101
>    libavdevice    58.  4.103 / 58.  4.103
>    libavfilter     7. 31.100 /  7. 31.100
>    libswscale      5.  2.100 /  5.  2.100
>    libswresample   3.  2.100 /  3.  2.100
>    libpostproc    55.  2.100 / 55.  2.100
> Guessed Channel Layout for Input Stream #0.0 : stereo
> Input #0, dshow, from 'audio=Mikrofon (Realtek High Definiti':
>    Duration: N/A, start: 6979.682000, bitrate: 1411 kb/s
>      Stream #0:0: Audio: pcm_s16le, 44100 Hz, stereo, s16, 1411 kb/s
> Input #1, lavfi, from 'aevalsrc=sin(3000*2*PI*t):c=stereo:s=44100':
>    Duration: N/A, start: 0.000000, bitrate: 5644 kb/s
>      Stream #1:0: Audio: pcm_f64le, 44100 Hz, stereo, dbl, 5644 kb/s
> pipe:play: Cannot allocate memory

One can now use afftfilt to shift frequencies around in frequency domain.
It should be easier than using amultiply filter.