down sampling

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down sampling

alex jamshedi
Hi,

Hopefully this is an appropriate question for the forums.

My goal is to receive a live audio stream that is being sampled at 131,072
Hz and re-sample it at 44.1 kHz before outputting it through my computers
speakers. Is this a task ffmpeg can perform?

Thank you.
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Re: down sampling

Carl Eugen Hoyos-2
2018-12-27 19:01 GMT+01:00, alex jamshedi <[hidden email]>:

> My goal is to receive a live audio stream that is being sampled at
> 131,072 Hz and re-sample it at 44.1 kHz before outputting it
> through my computers speakers. Is this a task ffmpeg can perform?

Yes, there is an output option "-ar" that accepts "44100" as argument.

Carl Eugen
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Re: down sampling

Michael Koch
Am 27.12.2018 um 20:18 schrieb Carl Eugen Hoyos:
> 2018-12-27 19:01 GMT+01:00, alex jamshedi <[hidden email]>:
>
>> My goal is to receive a live audio stream that is being sampled at
>> 131,072 Hz and re-sample it at 44.1 kHz before outputting it
>> through my computers speakers. Is this a task ffmpeg can perform?
> Yes, there is an output option "-ar" that accepts "44100" as argument.

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.

Michael

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Re: down sampling

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).

Indeed, probably a worthwhile task adding an ffmpeg "sdl audio" device.

Moritz
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Re: down sampling

Michael Koch
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. Can you please point me
to a working example?

Thanks,
Michael

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Re: down sampling

Carl Eugen Hoyos-2
2018-12-28 10:14 GMT+01:00, Michael Koch <[hidden email]>:
> 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.)

Carl Eugen
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Re: down sampling

Paul B Mahol
In reply to this post by Michael Koch
On 12/27/18, Michael Koch <[hidden email]> wrote:

> Am 27.12.2018 um 20:18 schrieb Carl Eugen Hoyos:
>> 2018-12-27 19:01 GMT+01:00, alex jamshedi <[hidden email]>:
>>
>>> My goal is to receive a live audio stream that is being sampled at
>>> 131,072 Hz and re-sample it at 44.1 kHz before outputting it
>>> through my computers speakers. Is this a task ffmpeg can perform?
>> Yes, there is an output option "-ar" that accepts "44100" as argument.
>
> 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 should really use mpv.
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Re: down sampling

Michael Koch
In reply to this post by Carl Eugen Hoyos-2
Am 28.12.2018 um 10:18 schrieb Carl Eugen Hoyos:

> 2018-12-28 10:14 GMT+01:00, Michael Koch <[hidden email]>:
>> 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


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Re: down sampling

Paul B Mahol
On 12/28/18, Michael Koch <[hidden email]> wrote:

> Am 28.12.2018 um 10:18 schrieb Carl Eugen Hoyos:
>> 2018-12-28 10:14 GMT+01:00, Michael Koch <[hidden email]>:
>>> 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.
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Re: down sampling

Carl Eugen Hoyos-2
In reply to this post by Michael Koch
2018-12-28 11:32 GMT+01:00, Michael Koch <[hidden email]>:

> Am 28.12.2018 um 10:18 schrieb Carl Eugen Hoyos:
>> 2018-12-28 10:14 GMT+01:00, Michael Koch <[hidden email]>:
>>> 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

The syntax looks broken here:
Do you want to use a named pipe "play"?
Iiuc, you have to create this pipe before launching FFmpeg,
no need to specify a second output url.

Or you want to use the pipe "-", in this case I believe you do
not consume it with your ffplay command:

> -i pipe:play

Finally, I would expect that you have to separate the call
to ffmpeg from the call to ffplay: "|"

Carl Eugen
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Re: down sampling

Michael Koch

>> 10000,lowpass=f=10000,lowpass=f=10000" -t 10 -f mp3 pipe:play -
>> //ffmpeg/ffplay
> The syntax looks broken here:
> Do you want to use a named pipe "play"?
> Iiuc, you have to create this pipe before launching FFmpeg,
> no need to specify a second output url.
>
> Or you want to use the pipe "-", in this case I believe you do
> not consume it with your ffplay command:
>
>> -i pipe:play
> Finally, I would expect that you have to separate the call
> to ffmpeg from the call to ffplay: "|"

You are right that the "|" was missing in my example. But when I include
it, it doesn't work either. I did a lot of Google searching for an
example how to pipe from ffmpeg to ffplay under Windows. I found 2 or 3
(see below), and those didn't work. Seems to be either impossible or
quite complicated.
If anyone has a working example, please add it to the ffmpeg documentation.

Thanks,
Michael


https://ffmpeg.zeranoe.com/forum/viewtopic.php?t=1414

|ffmpeg -i abc.avi -f rawvideo - | ffplay -f rawvideo -s 624x352 -pix_fmt
yuv420p -|


https://jonlabelle.com/snippets/view/shell/ffmpeg-command

|ffmpeg -ss 00:34:24.85 -t 10 -i path||/to/file||.mp4 -f mp3 pipe:play |
ffplay -i pipe:play -autoexit|



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Re: down sampling

Moritz Barsnick
On Sat, Dec 29, 2018 at 19:45:29 +0100, Michael Koch wrote:
> (see below), and those didn't work. Seems to be either impossible or
> quite complicated.

"Didn't work" is not a concise error description.

> https://ffmpeg.zeranoe.com/forum/viewtopic.php?t=1414
>
> |ffmpeg -i abc.avi -f rawvideo - | ffplay -f rawvideo -s 624x352 -pix_fmt yuv420p -|

This is wrong in many ways. It only works if you know your resolution
and format (because rawvideo doesn't carry any meta-information).

I would use the practical nut container, and do:

$ ffmpeg -i input -f nut - | ffplay -

> |ffmpeg -ss 00:34:24.85 -t 10 -i path||/to/file||.mp4 -f mp3 pipe:play |
> ffplay -i pipe:play -autoexit|

IMO, this shouldn't be piped with '|', but executed as two separate
shell commands.

Please try the former. And post the actually used command and the
complete, uncut console output.

(Sorry, if I had my Windows machine ready, I would just simply try.)

Moritz
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Re: down sampling

Carl Eugen Hoyos-2
In reply to this post by Michael Koch


> Am 29.12.2018 um 19:45 schrieb Michael Koch <[hidden email]>:

> ffmpeg -i abc.avi -f rawvideo - | ffplay -f rawvideo -s 624x352 -pix_fmt yuv420p -

Complete, uncut console output missing / works fine with ffmpeg here.

Carl Eugen
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Re: down sampling

Michael Koch
In reply to this post by Moritz Barsnick

> I would use the practical nut container, and do:
>
> $ ffmpeg -i input -f nut - | ffplay -

F:\Sound\Ultraschall_Konvertierung>c://ffmpeg/ffmpeg -i 699.mp4 -f nut -
| c://f
fmpeg/ffplay -
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
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '699.mp4':
   Metadata:
     major_brand     : isom
     minor_version   : 512
     compatible_brands: isomiso2avc1mp41
     encoder         : Lavf58.18.101
   Duration: 00:00:53.02, start: 0.000000, bitrate: 7558 kb/s
     Stream #0:0(und): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz,
stereo, flt
p, 128 kb/s (default)
     Metadata:
       handler_name    : SoundHandler
     Stream #0:1(eng): Video: h264 (High) (avc1 / 0x31637661),
yuvj420p(pc), 1920
x1080, 7426 kb/s, 25 fps, 25 tbr, 12800 tbn, 50 tbc (default)
     Metadata:
       handler_name    : VideoHandler
       timecode        : 00:00:46:17
     Stream #0:2(eng): Data: none (tmcd / 0x64636D74)
     Metadata:
       handler_name    : TimeCodeHandler
       timecode        : 00:00:46:17
[NULL @ 0000000000544dc0] Unable to find a suitable output format for '|'
|: Invalid argument
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Re: down sampling

Michael Koch
Am 29.12.2018 um 20:19 schrieb Michael Koch:

>
>> I would use the practical nut container, and do:
>>
>> $ ffmpeg -i input -f nut - | ffplay -
>
> F:\Sound\Ultraschall_Konvertierung>c://ffmpeg/ffmpeg -i 699.mp4 -f nut
> - | c://f
> fmpeg/ffplay -
> 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
> Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '699.mp4':
>   Metadata:
>     major_brand     : isom
>     minor_version   : 512
>     compatible_brands: isomiso2avc1mp41
>     encoder         : Lavf58.18.101
>   Duration: 00:00:53.02, start: 0.000000, bitrate: 7558 kb/s
>     Stream #0:0(und): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz,
> stereo, flt
> p, 128 kb/s (default)
>     Metadata:
>       handler_name    : SoundHandler
>     Stream #0:1(eng): Video: h264 (High) (avc1 / 0x31637661),
> yuvj420p(pc), 1920
> x1080, 7426 kb/s, 25 fps, 25 tbr, 12800 tbn, 50 tbc (default)
>     Metadata:
>       handler_name    : VideoHandler
>       timecode        : 00:00:46:17
>     Stream #0:2(eng): Data: none (tmcd / 0x64636D74)
>     Metadata:
>       handler_name    : TimeCodeHandler
>       timecode        : 00:00:46:17
> [NULL @ 0000000000544dc0] Unable to find a suitable output format for '|'
> |: Invalid argument

Might it be part of the problem that I'm starting ffmpeg from a batch file?
This is the content of the batch file:

c://ffmpeg/ffmpeg -i 699.mp4 -f nut - ^| c://ffmpeg/ffplay -
pause

Michael

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Re: down sampling

Reino Wijnsma
In reply to this post by Michael Koch
On 29-12-2018 20:19, Michael Koch <[hidden email]> wrote:
> F:\Sound\Ultraschall_Konvertierung>c://ffmpeg/ffmpeg -i 699.mp4 -f nut - | c://f
> fmpeg/ffplay -

Have you actually tested this at all? Forward slashes don't work on Windows!

C:\ffmpeg\ffmpeg.exe -f lavfi -i aevalsrc="sin(864*2*PI*t):c=stereo:s=131072" -ar 44.1k -f wav - | C:\ffmpeg\ffplay.exe -i -
or
C:\ffmpeg\ffmpeg.exe -f lavfi -i aevalsrc="sin(864*2*PI*t):c=stereo:s=131072" -af "aresample=resampler=soxr:osr=48000:precision=28" -f wav - | C:\ffmpeg\ffplay.exe -i -

-- Reino

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Re: down sampling

Michael Koch
Am 29.12.2018 um 20:53 schrieb Reino Wijnsma:
> On 29-12-2018 20:19, Michael Koch <[hidden email]> wrote:
>> F:\Sound\Ultraschall_Konvertierung>c://ffmpeg/ffmpeg -i 699.mp4 -f nut - | c://f
>> fmpeg/ffplay -
> Have you actually tested this at all? Forward slashes don't work on Windows!

Sure I tested this. I posted the console output, you can see that ffmpeg
is found.

Michael

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Re: down sampling

Carl Eugen Hoyos-2
In reply to this post by Michael Koch


> Am 29.12.2018 um 20:41 schrieb Michael Koch <[hidden email]>:
>
>> Am 29.12.2018 um 20:19 schrieb Michael Koch:
>>
>>> I would use the practical nut container, and do:
>>>
>>> $ ffmpeg -i input -f nut - | ffplay -
>>
>> F:\Sound\Ultraschall_Konvertierung>c://ffmpeg/ffmpeg -i 699.mp4 -f nut - | c://f
>> fmpeg/ffplay -
>> 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
>> Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '699.mp4':
>>   Metadata:
>>     major_brand     : isom
>>     minor_version   : 512
>>     compatible_brands: isomiso2avc1mp41
>>     encoder         : Lavf58.18.101
>>   Duration: 00:00:53.02, start: 0.000000, bitrate: 7558 kb/s
>>     Stream #0:0(und): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, stereo, flt
>> p, 128 kb/s (default)
>>     Metadata:
>>       handler_name    : SoundHandler
>>     Stream #0:1(eng): Video: h264 (High) (avc1 / 0x31637661), yuvj420p(pc), 1920
>> x1080, 7426 kb/s, 25 fps, 25 tbr, 12800 tbn, 50 tbc (default)
>>     Metadata:
>>       handler_name    : VideoHandler
>>       timecode        : 00:00:46:17
>>     Stream #0:2(eng): Data: none (tmcd / 0x64636D74)
>>     Metadata:
>>       handler_name    : TimeCodeHandler
>>       timecode        : 00:00:46:17
>> [NULL @ 0000000000544dc0] Unable to find a suitable output format for '|'
>> |: Invalid argument
>
> Might it be part of the problem that I'm starting ffmpeg from a batch file?
> This is the content of the batch file:
>
> c://ffmpeg/ffmpeg -i 699.mp4 -f nut - ^| c://ffmpeg/ffplay -

Why did you add the caret?
And even more important, why did you not post your actual command line when we asked?

Works fine here (unless I add funny characters) here with Windows cmd, both with and without using a batch file.

Carl Eugen
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Re: down sampling

Michael Koch
In reply to this post by Michael Koch

> Might it be part of the problem that I'm starting ffmpeg from a batch
> file?
> This is the content of the batch file:
>
> c://ffmpeg/ffmpeg -i 699.mp4 -f nut - ^| c://ffmpeg/ffplay -
> pause
>

I just found out that when using this command in a batch file, it
doesn't matter if the slashes are forward or backward. But when the
command is typed into the console window (without the ^ character), then
it works only if backslashes are used. And then piping from ffmpeg to
ffplay works fine!
Why doesn't the same command line work in a batch file? What must I change?

Michael

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Re: down sampling

Michael Koch
In reply to this post by Carl Eugen Hoyos-2

>> Might it be part of the problem that I'm starting ffmpeg from a batch file?
>> This is the content of the batch file:
>>
>> c://ffmpeg/ffmpeg -i 699.mp4 -f nut - ^| c://ffmpeg/ffplay -
> Why did you add the caret?

I thought that in a batch file the | character must be escaped with a ^
character. As documented here:
https://www.robvanderwoude.com/escapechars.php

You see that the ^ doesn't appear in the console output.

> And even more important, why did you not post your actual command line when we asked?

But the command line that ffmpeg got is correct, isn't it?

> Works fine here (unless I add funny characters) here with Windows cmd, both with and without using a batch file.

It's not yet working here with a batch file. Please post your batch file.

Michael

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