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• ffmpeg - unknown keyword [duplicate]

  20 juin 2017, par Jublo

  This question already has an answer here :

  • ffmpeg concat comand error
    
    2 answers
    

  When calling ffmpeg with the following command line :

  ffmpeg -f concat
  
    -i "/Users/demo/mono128fixed.mp3"
  
    -i "/Users/demo/stereo256fixed.mp3" 
  
    -acodec copy "/Users/Demo/outfile 2017-06-20 20-56-23.mp3"    

  I’m getting the following error message :

  [concat @ 0x102800000] Line 1: unknown keyword '\377\373\220\304'
  
  /Users/demo/mono128fixed.mp3: Invalid data found when processing input

  Why would this be ?

  ffmpeg config is like this :

  ffmpeg version 3.3.2 Copyright (c) 2000-2017 the FFmpeg developers
  
    built with Apple LLVM version 8.1.0 (clang-802.0.42)
  
    configuration: --prefix=/Volumes/Ramdisk/sw --enable-pthreads --enable-version3 --disable-decoder=libvpx --enable-libmp3lame --enable-avfilter --enable-libopencore_amrwb --enable-libopencore_amrnb --enable-filters --arch=x86_64 --enable-runtime-cpudetect
  
    libavutil      55. 58.100 / 55. 58.100
  
    libavcodec     57. 89.100 / 57. 89.100
  
    libavformat    57. 71.100 / 57. 71.100
  
    libavdevice    57.  6.100 / 57.  6.100
  
    libavfilter     6. 82.100 /  6. 82.100
  
    libswscale      4.  6.100 /  4.  6.100
  
    libswresample   2.  7.100 /  2.  7.100

• ffmpeg : Streamcopy requested for output stream 0:0, which is fed from a complex filtergraph. Filtering and streamcopy cannot be used together

  7 juin 2017, par DmitrySemenov

  I’m on Fedora25 running the following command expecting live streaming to youtube and at the same time saving of the final combined stream (screencast + webcam + logo + audio) into local mp4 file.

  ffmpeg \
  
  -f x11grab -video_size 3840x2160 -framerate 30 -i :1.0 \
  
  -f v4l2 -video_size 320x240 -framerate 30 -i /dev/video1 \
  
  -f alsa -ac 2 -i pulse -i logo.png -filter_complex \
  
  "[0:v]scale=1920:-1,setpts=PTS-STARTPTS[bg]; \
  
   [1:v]scale=160:-1,setpts=PTS-STARTPTS[fg]; \
  
   [bg][fg]overlay=W-w-10:10[bg2]; \
  
   [bg2][3:v]overlay=W-w-10:H-h-10,format=yuv420p[out]" \
  
  -map "[out]" -map 2:a -c:v libx264 -preset veryfast \
  
  -maxrate 9000k -bufsize 12000k -c:a aac -b:a 160k -ar 44100 -b:a 128k \
  
  -force_key_frames "expr:gte(t,n_forced*2)" \
  
  -c copy -f flv rtmp://a.rtmp.youtube.com/live2/TOKEN
  
  -c copy -f mp4 out.mp4

  getting the following output

  ffmpeg version 3.1.7 Copyright (c) 2000-2017 the FFmpeg developers
  
    built with gcc 6.3.1 (GCC) 20161221 (Red Hat 6.3.1-1)
  
    configuration: --prefix=/usr --bindir=/usr/bin --datadir=/usr/share/ffmpeg --incdir=/usr/include/ffmpeg --libdir=/usr/lib64 --mandir=/usr/share/man --arch=x86_64 --optflags='-O2 -g -pipe -Wall -Werror=format-security -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector-strong --param=ssp-buffer-size=4 -grecord-gcc-switches -specs=/usr/lib/rpm/redhat/redhat-hardened-cc1 -m64 -mtune=generic' --extra-ldflags='-Wl,-z,relro -specs=/usr/lib/rpm/redhat/redhat-hardened-ld' --enable-bzlib --disable-crystalhd --enable-fontconfig --enable-frei0r --enable-gcrypt --enable-gnutls --enable-ladspa --enable-libass --enable-libbluray --enable-libcdio --enable-indev=jack --enable-libfreetype --enable-libfribidi --enable-libgsm --enable-libmp3lame --enable-nvenc --extra-cflags=-I/usr/include/nvenc --enable-openal --enable-opencl --enable-opengl --enable-libopenjpeg --enable-libopus --enable-libpulse --enable-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libvorbis --enable-libv4l2 --enable-libvpx --enable-libx264 --enable-libx265 --enable-libxvid --enable-x11grab --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. 28.100 / 55. 28.100
  
    libavcodec     57. 48.101 / 57. 48.101
  
    libavformat    57. 41.100 / 57. 41.100
  
    libavdevice    57.  0.101 / 57.  0.101
  
    libavfilter     6. 47.100 /  6. 47.100
  
    libavresample   3.  0.  0 /  3.  0.  0
  
    libswscale      4.  1.100 /  4.  1.100
  
    libswresample   2.  1.100 /  2.  1.100
  
    libpostproc    54.  0.100 / 54.  0.100
  
  [x11grab @ 0x562f6e929c20] Stream #0: not enough frames to estimate rate; consider increasing probesize
  
  Input #0, x11grab, from ':1.0':
  
    Duration: N/A, start: 1496800335.695598, bitrate: N/A
  
      Stream #0:0: Video: rawvideo (BGR[0] / 0x524742), bgr0, 3840x2160, 30 fps, 1000k tbr, 1000k tbn, 1000k tbc
  
  Input #1, video4linux2,v4l2, from '/dev/video1':
  
    Duration: N/A, start: 17665.258327, bitrate: 36864 kb/s
  
      Stream #1:0: Video: rawvideo (YUY2 / 0x32595559), yuyv422, 320x240, 36864 kb/s, 30 fps, 30 tbr, 1000k tbn, 1000k tbc
  
  Guessed Channel Layout for Input Stream #2.0 : stereo
  
  Input #2, alsa, from 'pulse':
  
    Duration: N/A, start: 1496800336.972575, bitrate: 1536 kb/s
  
      Stream #2:0: Audio: pcm_s16le, 48000 Hz, 2 channels, s16, 1536 kb/s
  
  Input #3, png_pipe, from 'logo.png':
  
    Duration: N/A, bitrate: N/A
  
      Stream #3:0: Video: png, rgba(pc), 190x105 [SAR 2835:2835 DAR 38:21], 25 tbr, 25 tbn, 25 tbc
  
  Streamcopy requested for output stream 0:0, which is fed from a complex filtergraph. Filtering and streamcopy cannot be used together.
  
  ➜  tmp 

• How to use codec type properly in NPM

  3 juillet 2017, par shamaleyte

  Trying to use ’-acodec libopus’ in my npm project as I use in the command line like in the following format ;

  ffmpeg -acodec libopus -i 1.webm 1.wav

  This works perfectly ! But I would like to make it possible in my NPM project.

  How can I set the parameters ?
  This is what I have , but not working. The output file is broken in a way that some of the frames of the audio file are missing. It is like there is sound and then there is not. And vice versa.

  var proc = new ffmpeg({
  
          source: file,
  
          nolog: false       
  
      });
  
  
  
  
  
  format = "opus"; // or could be wav as well!   
  
  
  
  
  
      proc.addOptions([
  
          '-f ' + format,          
  
          '-acodec libopus',
  
          '-vn'
  
      ]);

  The purpose is to take audio file from the video file seamlessly.

  Without the codec libopus, I get the following errors in the command prompt, so I assume I should handle the same issue in my NPM project as well.

  [opus @ 00000000006d4520] LBRR frames is not implemented. Update your FFmpeg version to the newest one from Git. If the problem still occurs, it means that your file has a feature which has not been implemented.
  
  
  
  
  
  [opus @ 00000000006d4520] Error decoding a SILK frame.
  
  
  
  [opus @ 00000000006d4520] Error decoding an Opus frame.

  My library is up to date, I just need to use the codec libopus properly.
  Any suggestions ?

   \node-js>ffmpeg -version
  
   ffmpeg version N-86175-g64ea4d1 Copyright (c) 2000-2017 the FFmpeg 
  
   developers
  
   built with gcc 6.3.0 (GCC)
  
  
  
  Output in command line;
  
  xtranscribe transcodeWatson: file : ./data/that/2.webm
  
  progress 62.625273103421605%
  
  progress 100.01224534515762%
  
  SAVED - transcodeWatson : .mp3
  
  out of transcode!
  
  fileSizeInBytes  : 16284033