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• `ffmpeg record audio output device

  11 juillet 2020, par Roy

  i am trying to capture my pulse output device :

  


  alsa_output.pci-0000_00_1b.0.analog-stereo.monitor

  


  with this command :

  


  /usr/bin/ffmpeg -f pulse -i alsa_output.pci-0000_00_1b.0.analog-stereo.monitor -c:a aac -b:a 128k /roee_files/test.wav


  


  i get this output with those warnings, and there is an audio delay :

  


  ffmpeg version 3.4.7 Copyright (c) 2000-2019 the FFmpeg developers
  built with gcc 4.8.5 (GCC) 20150623 (Red Hat 4.8.5-39)
  configuration: --prefix=/usr --bindir=/usr/bin --datadir=/usr/share/ffmpeg --docdir=/usr/share/doc/ffmpeg --incdir=/usr/include/ffmpeg --libdir=/usr/lib64 --mandir=/usr/share/man --arch=x86_64 --optflags='-O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector-strong --param=ssp-buffer-size=4 -grecord-gcc-switches -m64 -mtune=generic' --extra-ldflags='-Wl,-z,relro ' --extra-cflags=' ' --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libvo-amrwbenc --enable-version3 --enable-bzlib --disable-crystalhd --enable-fontconfig --enable-gcrypt --enable-gnutls --enable-ladspa --enable-libass --enable-libbluray --enable-libcdio --enable-libdrm --enable-indev=jack --enable-libfreetype --enable-libfribidi --enable-libgsm --enable-libmp3lame --enable-nvenc --enable-openal --enable-opencl --enable-opengl --enable-libopenjpeg --enable-libopus --disable-encoder=libopus --enable-libpulse --enable-librsvg --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libvorbis --enable-libv4l2 --enable-libvidstab --enable-libx264 --enable-libx265 --enable-libxvid --enable-libzvbi --enable-avfilter --enable-avresample --enable-postproc --enable-pthreads --disable-static --enable-shared --enable-gpl --disable-debug --disable-stripping --shlibdir=/usr/lib64 --enable-libmfx --enable-runtime-cpudetect
  libavutil      55. 78.100 / 55. 78.100
  libavcodec     57.107.100 / 57.107.100
  libavformat    57. 83.100 / 57. 83.100
  libavdevice    57. 10.100 / 57. 10.100
  libavfilter     6.107.100 /  6.107.100
  libavresample   3.  7.  0 /  3.  7.  0
  libswscale      4.  8.100 /  4.  8.100
  libswresample   2.  9.100 /  2.  9.100
  libpostproc    54.  7.100 / 54.  7.100
Guessed Channel Layout for Input Stream #0.0 : stereo
Input #0, pulse, from 'alsa_output.pci-0000_00_1b.0.analog-stereo.monitor':
  Duration: N/A, start: 1594460535.507972, bitrate: 1536 kb/s
    Stream #0:0: Audio: pcm_s16le, 48000 Hz, stereo, s16, 1536 kb/s
File '/roee_files/test.wav' already exists. Overwrite ? [y/N] y
Stream mapping:
  Stream #0:0 -> #0:0 (pcm_s16le (native) -> aac (native))
Press [q] to stop, [?] for help
Output #0, wav, to '/roee_files/test.wav':
  Metadata:
    ISFT            : Lavf57.83.100
    Stream #0:0: Audio: aac (LC) ([255][0][0][0] / 0x00FF), 48000 Hz, stereo, fltp, 128 kb/s
    Metadata:
      encoder         : Lavc57.107.100 aac
[aac @ 0x1aec1a0] Queue input is backward in time
[wav @ 0x1aeabc0] Non-monotonous DTS in output stream 0:0; previous: 120076, current: 116419; changing to 120076. This may result in incorrect timestamps in the output file.
[wav @ 0x1aeabc0] Non-monotonous DTS in output stream 0:0; previous: 120076, current: 117443; changing to 120076. This may result in incorrect timestamps in the output file.
[wav @ 0x1aeabc0] Non-monotonous DTS in output stream 0:0; previous: 120076, current: 118467; changing to 120076. This may result in incorrect timestamps in the output file.
[wav @ 0x1aeabc0] Non-monotonous DTS in output stream 0:0; previous: 120076, current: 119491; changing to 120076. This may result in incorrect timestamps in the output file.
[aac @ 0x1aec1a0] Queue input is backward in time
[wav @ 0x1aeabc0] Non-monotonous DTS in output stream 0:0; previous: 123587, current: 118096; changing to 123587. This may result in incorrect timestamps in the output file.
[wav @ 0x1aeabc0] Non-monotonous DTS in output stream 0:0; previous: 123587, current: 119120; changing to 123587. This may result in incorrect timestamps in the output file.


  


  but,
when i am trying to capture my pulse input device with the same command :

  


  /usr/bin/ffmpeg -f pulse -i alsa_input.pci-0000_00_1b.0.analog-stereo -c:a aac -b:a 128k /roee_files/test.wav


  


  everything works fine without any warning or delay as you can see in this output :

  


  ffmpeg version 3.4.7 Copyright (c) 2000-2019 the FFmpeg developers
  built with gcc 4.8.5 (GCC) 20150623 (Red Hat 4.8.5-39)
  configuration: --prefix=/usr --bindir=/usr/bin --datadir=/usr/share/ffmpeg --docdir=/usr/share/doc/ffmpeg --incdir=/usr/include/ffmpeg --libdir=/usr/lib64 --mandir=/usr/share/man --arch=x86_64 --optflags='-O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector-strong --param=ssp-buffer-size=4 -grecord-gcc-switches -m64 -mtune=generic' --extra-ldflags='-Wl,-z,relro ' --extra-cflags=' ' --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libvo-amrwbenc --enable-version3 --enable-bzlib --disable-crystalhd --enable-fontconfig --enable-gcrypt --enable-gnutls --enable-ladspa --enable-libass --enable-libbluray --enable-libcdio --enable-libdrm --enable-indev=jack --enable-libfreetype --enable-libfribidi --enable-libgsm --enable-libmp3lame --enable-nvenc --enable-openal --enable-opencl --enable-opengl --enable-libopenjpeg --enable-libopus --disable-encoder=libopus --enable-libpulse --enable-librsvg --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libvorbis --enable-libv4l2 --enable-libvidstab --enable-libx264 --enable-libx265 --enable-libxvid --enable-libzvbi --enable-avfilter --enable-avresample --enable-postproc --enable-pthreads --disable-static --enable-shared --enable-gpl --disable-debug --disable-stripping --shlibdir=/usr/lib64 --enable-libmfx --enable-runtime-cpudetect
  libavutil      55. 78.100 / 55. 78.100
  libavcodec     57.107.100 / 57.107.100
  libavformat    57. 83.100 / 57. 83.100
  libavdevice    57. 10.100 / 57. 10.100
  libavfilter     6.107.100 /  6.107.100
  libavresample   3.  7.  0 /  3.  7.  0
  libswscale      4.  8.100 /  4.  8.100
  libswresample   2.  9.100 /  2.  9.100
  libpostproc    54.  7.100 / 54.  7.100
Guessed Channel Layout for Input Stream #0.0 : stereo
Input #0, pulse, from 'alsa_input.pci-0000_00_1b.0.analog-stereo':
  Duration: N/A, start: 1594460998.540483, bitrate: 1536 kb/s
    Stream #0:0: Audio: pcm_s16le, 48000 Hz, stereo, s16, 1536 kb/s
File '/roee_files/test.wav' already exists. Overwrite ? [y/N] y
Stream mapping:
  Stream #0:0 -> #0:0 (pcm_s16le (native) -> aac (native))
Press [q] to stop, [?] for help
Output #0, wav, to '/roee_files/test.wav':
  Metadata:
    ISFT            : Lavf57.83.100
    Stream #0:0: Audio: aac (LC) ([255][0][0][0] / 0x00FF), 48000 Hz, stereo, fltp, 128 kb/s
    Metadata:
      encoder         : Lavc57.107.100 aac
size=     822kB time=00:00:52.45 bitrate= 128.4kbits/s speed=1.07x


  


  i tried what someone suggest here- to uncomment and change the line, default-sample-rate = 44100 , to default-sample-rate = 48000. (in /etc/pulse/daemon.conf file, and restart PA/computer) withot success.

  


  this is my whole section in this file after changing :

  


  ; default-sample-format = s16le
 default-sample-rate = 48000
; alternate-sample-rate = 48000
; default-sample-channels = 2
; default-channel-map = front-left,front-right

; default-fragments = 4
; default-fragment-size-msec = 25

; enable-deferred-volume = yes
; deferred-volume-safety-margin-usec = 8000
; deferred-volume-extra-delay-usec = 0


  


  Does someone can help me solve this issue PLEASE...

  


  does it relate to sample rate, or maybe other problem ?

  


  thank you all...!

  


• How to convert 4:3 stream to 16:9 using ffmpeg

  13 février 2020, par DavidBill7

  Can somebody please help me ?

  I need to convert an analog camera stream to 16:9. At the moment it’s portraying at 4:3.

  The code I’m using is below.

  ffmpeg -hide_banner -re -i  {{OUTPUT_FILENAME}} -vcodec h264_qsv -preset slower -s 1280x720 -vsync 1 -async 1 -r 25 -b:v 650K -minrate:v 650K -maxrate:v 650K -acodec aac -ac 2 -ar 48000 -ab 128k -bsf:v 'filter_units=pass_types=1-5'  -map 0 -g 50 -flags -global_header -f flv rtmp://

• recording live stream video from tv card using ffmpeg at window [on hold]

  6 décembre 2013, par user2688423

  I want to record live stream every 1 second from tv card(tv signal) using ffmpeg in window.

  first of all, to record live video from tv card, I tried below.

  1. First I tried this.

  ffmpeg -list_devices true -f dshow -i dummy

  then the result is

  " [dshow @ 000000000024e6fe0] DirectShow video devices
  [dshow @000000000024e6fe0] "SKYTV HD USB Maxx Video Capture"
  [dshow @
  000000000024e6fe0] DirectShow audio devices
  [dshow @
  000000000024e6fe0] "Analog Audio In(SKYTV HD USB Ma" "

  so I tried

  ffmpeg -f dshow -i video="SKYTV HD USB Maxx Video Capture" -r 20
  -threads 0 D ://test.mkv

  But it didn't work. the Error message is

  "[dshow@000000000034d920] Could not run filter
  video=SKYTV HD USB
  Maxx Video Capture : Input/output error"

  I use the device called 'SKYTV HD USB Maxx Video Capture' for getting tv signal(TV card).

  1. The First way deosn't work, I tried different way.

  ffmpeg -y -f vfwcap -i list

  then the result is
  "

  [dshow @ 00000000003fd760] Driver 0

  [dshow @ 00000000003fd760] Microsoft WDM Image Capture (Win32)
  [dshow @ 00000000003fd760] Version : 6.1.7601.17514 list : Input/output error

  "

  so I tried

  ffmpeg -y -f vfwcap -r 25 -i 0 D ://out.mp4

  then, there is some out.mp4 file in D drive but the file is nothing.
  (I think it is not TV signal)

  what should i do to record live video every 1 second from tv card(tv signal) using ffmpeg in window ? And How can I set channel at tvcard(Because I want to get tv signal, there are many channels).

  Please help..!