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• How to add album art with ffmpeg ?

  17 juillet 2016, par Adi Ricky k

  Everybody, I’ve stuck to add album art on mp3 files.

  I already research and Googling about this issue but not solution yet, on ffmpeg documentation to add image (album art) to mp3 use this script :

  ffmpeg -i input.mp3 -i cover.png -c copy -map 0 -map 1 -metadata:s:v title="Album cover" -metadata:s:v comment="Cover (Front)" out.mp3

  Source from : http://www.ffmpeg.org/ffmpeg-all.html#mp3

  But it’s doesn’t work on my console output is :

  Unrecognized option 'c'
  
  Failed to set value 'copy' for option 'c'

  I looking for another solution and I get from : http://trac.ffmpeg.org/ticket/2221 :

  ffmpeg -i input.mp3 -i cover.png -map 0:0 -map 1:0 -c copy -id3v2_version 3 -metadata:s:v title="Album cover" -metadata:s:v comment="Cover (Front)" out.mp3

  Also return same output, I already research why return :

  Unrecognized option 'c'
  
  Failed to set value 'copy' for option 'c'

  But there are no body have same problem.
  anybody can help me, please ?

  nb : I use ubuntu 12.04 and ffmpeg version 0.8.6-4:0.8.6-0.
  thanks

• Is it possibility to merge FLV files by using javascript ?

  26 juin 2020, par DingXuan

  I know that I can merge multiple FLV files by using FFMPEG, but I want I can do this on front-end.

  


  I have looking for this question for a long time, but can't find any method to do that.

  


  Anyone can help me ?

  


• `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...!