Re: [FFmpeg-user] down sampling
Am 30.12.2018 um 13:20 schrieb Paul B Mahol: On 12/30/18, Michael Koch wrote: Am 28.12.2018 um 11:46 schrieb Paul B Mahol: One can now use afftfilt to shift frequencies around in frequency domain. It should be easier than using amultiply filter. c:\ffmpeg\ffmpeg -f dshow -channels 2 -i audio="Mikrofon (Realtek High Definiti" -af volume=30,afftfilt='real=real(b+300,ch)':'imag=imag(b+300,ch)' -f nut - | c:\\ffmpeg\ffplay - I've just tested this and it works fine. There is no error message when (b+300) becomes larger than the available number of bins. How does afftfilt handle this case? Does real(too_large_number,ch) return zero? No, it clips and return value in max bin. If you want explicit zero you will need to change your expression. yes, zero is better than using the value in the max bin. Here is the (Windows) batch file for live ultrasonic conversion: set "SR=44100" :: Sample Rate set "F=4000" :: Subtracted Frequency set "VOL=30" :: Volume Factor set /a "N=4096*%F%/%SR%" :: N = 4096 * F / SR c:\ffmpeg\ffmpeg -f dshow -channels 2 -i audio="Mikrofon (Realtek High Definiti" -af volume=%VOL%,afftfilt='real=if(lt(b+%N%,nb),real(b+%N%,ch),0)':'imag=if(lt(b+%N%,nb),imag(b+%N%,ch),0)' -f nut - | c:\ffmpeg\ffplay - Michael ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-user] down sampling
On 12/30/18, Michael Koch wrote: > Am 28.12.2018 um 11:46 schrieb Paul B Mahol: >> One can now use afftfilt to shift frequencies around in frequency domain. >> It should be easier than using amultiply filter. > > c:\ffmpeg\ffmpeg -f dshow -channels 2 -i audio="Mikrofon (Realtek High > Definiti" -af > volume=30,afftfilt='real=real(b+300,ch)':'imag=imag(b+300,ch)' -f nut - > | c:\\ffmpeg\ffplay - > > I've just tested this and it works fine. There is no error message when > (b+300) becomes larger than the available number of bins. How does > afftfilt handle this case? > Does real(too_large_number,ch) return zero? No, it clips and return value in max bin. If you want explicit zero you will need to change your expression. ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-user] down sampling
Am 28.12.2018 um 11:46 schrieb Paul B Mahol: One can now use afftfilt to shift frequencies around in frequency domain. It should be easier than using amultiply filter. c:\ffmpeg\ffmpeg -f dshow -channels 2 -i audio="Mikrofon (Realtek High Definiti" -af volume=30,afftfilt='real=real(b+300,ch)':'imag=imag(b+300,ch)' -f nut - | c:\\ffmpeg\ffplay - I've just tested this and it works fine. There is no error message when (b+300) becomes larger than the available number of bins. How does afftfilt handle this case? Does real(too_large_number,ch) return zero? Michael ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-user] down sampling
Works fine here (unless I add funny characters) here with Windows cmd, both with and without using a batch file. Finally it's working here. This is the content of the batch file: c:\\ffmpeg\ffmpeg -i 699.mp4 -f nut - | c:\\ffmpeg\ffplay - -- all backslashes -- no escape character before | -- but remember: If you have a % in the command line, then it must be escaped as %% Thanks to all who helped to solve this problem!!! Michael ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-user] down sampling
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 ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-user] down sampling
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 ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-user] down sampling
> Am 29.12.2018 um 20:41 schrieb 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 >> libavformat58. 18.101 / 58. 18.101 >> libavdevice58. 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 >> libpostproc55. 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: 0.00, 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 @ 00544dc0] 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 ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-user] down sampling
Am 29.12.2018 um 20:53 schrieb Reino Wijnsma: On 29-12-2018 20:19, Michael Koch 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 ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-user] down sampling
On 29-12-2018 20:19, Michael Koch 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 ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-user] down sampling
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: 0.00, 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 @ 00544dc0] 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 ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-user] down sampling
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: 0.00, 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 @ 00544dc0] Unable to find a suitable output format for '|' |: Invalid argument ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-user] down sampling
> Am 29.12.2018 um 19:45 schrieb Michael Koch : > 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 ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-user] down sampling
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 ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-user] down sampling
1,lowpass=f=1,lowpass=f=1" -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| ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-user] down sampling
2018-12-28 11:32 GMT+01:00, Michael Koch : > Am 28.12.2018 um 10:18 schrieb Carl Eugen Hoyos: >> 2018-12-28 10:14 GMT+01:00, 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. >> 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=1,lowpass=f= > 1,lowpass=f=1,lowpass=f=1" -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 ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-user] down sampling
On 12/28/18, Michael Koch wrote: > Am 28.12.2018 um 10:18 schrieb Carl Eugen Hoyos: >> 2018-12-28 10:14 GMT+01:00, 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. >> 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=1,lowpass=f= > 1,lowpass=f=1,lowpass=f=1" -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 >libavformat58. 18.101 / 58. 18.101 >libavdevice58. 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 >libpostproc55. 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: 0.00, 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. ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-user] down sampling
Am 28.12.2018 um 10:18 schrieb Carl Eugen Hoyos: 2018-12-28 10:14 GMT+01:00, 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. 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=1,lowpass=f= 1,lowpass=f=1,lowpass=f=1" -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: 0.00, bitrate: 5644 kb/s Stream #1:0: Audio: pcm_f64le, 44100 Hz, stereo, dbl, 5644 kb/s pipe:play: Cannot allocate memory ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-user] down sampling
On 12/27/18, Michael Koch wrote: > Am 27.12.2018 um 20:18 schrieb Carl Eugen Hoyos: >> 2018-12-27 19:01 GMT+01:00, alex jamshedi : >> >>> 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. ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-user] down sampling
2018-12-28 10:14 GMT+01:00, 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. What did you try? (Command line and complete, uncut console output missing.) Carl Eugen ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-user] down sampling
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 ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-user] down sampling
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 ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-user] down sampling
Am 27.12.2018 um 20:18 schrieb Carl Eugen Hoyos: 2018-12-27 19:01 GMT+01:00, alex jamshedi : 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 ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-user] down sampling
2018-12-27 19:01 GMT+01:00, alex jamshedi : > 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 ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
[FFmpeg-user] down sampling
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. ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
