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• FFMPEG : Creating video stream from non-consecutively numbered png sequence [closed]

  8 mars 2016, par frageDE

  I have used VLC to decompose a video stream into frames, and it decomposed in a way that it leaps 5 frames per save, so the images are ordered like this :

  scene04436.png
  
  scene04441.png
  
  scene04446.png
  
  scene04451.png
  
  scene04456.png
  
  scene04461.png
  
  scene04466.png

  Now I want them to be a video stream again. I have already tried FFMPEG and AVCONV commands, they didn’t work out for me. The following command gives the error below :

  ffmpeg -framerate 1/5 -i scene%05d.png -c:v libx264 -r 30 -pix_fmt yuv420p out.mp4

  EDIT : Yes, I tried Googleing the ffmpeg commands, they did not work, mostly gave the error of :

      ffmpeg version 0.8.17-4:0.8.17-0ubuntu0.12.04.1, Copyright (c) 2000-2014 the Libav developers built on Mar 16 2015 13:26:50 with gcc 4.6.3
  
  
  
  The ffmpeg program is only provided for script compatibility and will be removed in a future release. It has been deprecated in the Libav project to allow for incompatible command line syntax improvements in its replacement called avconv
  
  
  
  (see Changelog for details). Please use avconv instead.
  
  Input #0, image2, from 'scene%05d.png':
  
    Duration: 00:00:05.00, start: 0.000000, bitrate: N/A
  
      Stream #0.0: Video: png, bgra, 720x480, 0.20 tbr, 0.20 tbn, 0.20 tbc
  
  
  
  Unrecognized option 'c:v'
  
  Failed to set value 'libx264' for option 'c:v'

  EDIT2 : This is the version output on my console for the FFMPEG

  ffmpeg version 3.0 Copyright (c) 2000-2016 the FFmpeg developers
  
  built with gcc 4.6 (Ubuntu/Linaro 4.6.3-1ubuntu5)
  
  configuration: --disable-yasm
  
  libavutil      55. 17.103 / 55. 17.103
  
  libavcodec     57. 24.102 / 57. 24.102
  
  libavformat    57. 25.100 / 57. 25.100
  
  libavdevice    57.  0.101 / 57.  0.101
  
  libavfilter     6. 31.100 /  6. 31.100
  
  libswscale      4.  0.100 /  4.  0.100
  
  libswresample   2.  0.101 /  2.  0.101

  Any suggestions ?

• ffmpeg replace part of audio file with looped audio

  1er décembre 2015, par user1202648

  I am quite new to ffmpeg and I am trying to replace a part of a first audio file with another second file. The second file can be too short, so some sort of loop should exist.

  After some research I came up with the following command arguments and it gives me the output as long as I only do one replacement. But I would like to do multiple replacements. So any help on what I am doing wrong ? Any suggestions/remarks on the way of working are also very welcome.

  (Any typos in the commands below can be ignored, I generate the command by script and for ease of use I simplified the names.)

  Works (One replacement) :

  "ffmpeg.exe" -y -i "first.wav" -i "second.wav" -filter_complex "[1:a][1:a][1:a]concat=n=3:v=0:a=1,asetpts=PTS-STARTPTS[replaceBase];[0:a]atrim=0:3,asetpts=PTS-STARTPTS[partA];[replaceBase]atrim=0:2,asetpts=PTS-STARTPTS[replaceA];[0:a]atrim=start=5,asetpts=PTS-STARTPTS[partB];[partA][replaceA][partB]concat=n=3:v=0:a=1[aout]" -map "[aout]" Out.wav

  Works Not (Multiple replacements) :

  "ffmpeg.exe" -y -i "first.wav" -i "second.wav" -filter_complex "[1:a][1:a][1:a]concat=n=3:v=0:a=1,asetpts=PTS-STARTPTS[replaceBase];[0:a]atrim=0:3,asetpts=PTS-STARTPTS[partA];[replaceBase]atrim=0:2,asetpts=PTS-STARTPTS[replaceA];[0:a]atrim=5:4,asetpts=PTS-STARTPTS[partB];[replaceBase]atrim=0:2,asetpts=PTS-STARTPTS[replaceB];[0:a]atrim=start=6,asetpts=PTS-STARTPTS[partC];[partA][replaceA][partB][replaceB][PartC]concat=n=4:v=0:a=1[aout]" -map "[aout]" Out.wav

  ffmpeg version N-76860-g72eaf72 Copyright (c) 2000-2015 the FFmpeg developers
  built with gcc 5.2.0 (GCC)
  configuration : —enable-gpl —enable-version3 —disable-w32threads —enable-avisynth —enable-bzlib —enable-fontconfig —enable-frei0r —enable-gnutls —enable-iconv —enable-libass —enable-libbluray —enable-libbs2b —enable-libcaca —enable-libdcadec —enable-libfreetype —enable-libgme —enable-libgsm —enable-libilbc —enable-libmodplug —enable-libmp3lame —enable-libopencore-amrnb —enable-libopencore-amrwb —enable-libopenjpeg —enable-libopus —enable-librtmp —enable-libschroedinger —enable-libsoxr —enable-libspeex —enable-libtheora —enable-libtwolame —enable-libvidstab —enable-libvo-aacenc —enable-libvo-amrwbenc —enable-libvorbis —enable-libvpx —enable-libwavpack —enable-libwebp —enable-libx264 —enable-libx265 —enable-libxavs —enable-libxvid —enable-libzimg —enable-lzma —enable-decklink —enable-zlib
  libavutil 55. 9.100 / 55. 9.100
  libavcodec 57. 16.100 / 57. 16.100
  libavformat 57. 19.100 / 57. 19.100
  libavdevice 57. 0.100 / 57. 0.100
  libavfilter 6. 15.100 / 6. 15.100
  libswscale 4. 0.100 / 4. 0.100
  libswresample 2. 0.101 / 2. 0.101
  libpostproc 54. 0.100 / 54. 0.100
  Guessed Channel Layout for Input Stream #0.0 : stereo
  Input #0, wav, from ’3897583stereo.wav’ :
  Duration : 00:00:12.07, bitrate : 256 kb/s
  Stream #0:0 : Audio : pcm_s16le ([1][0][0][0] / 0x0001), 8000 Hz, 2 channels, s16, 256 kb/s
  Guessed Channel Layout for Input Stream #1.0 : stereo
  Input #1, wav, from ’beep-021.wav’ :
  Metadata :
  encoder : Lavf57.19.100
  Duration : 00:00:00.30, bitrate : 1413 kb/s
  Stream #1:0 : Audio : pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, 2 channels, s16, 1411 kb/s
  [wav @ 057242c0] Invalid stream specifier : replaceBase.
  Last message repeated 1 times
  Stream specifier ’STREAM CUT matches no streams.

  Thanks in advance !

• ffmpeg does not record audio exactly after setting duration with -t

  25 janvier 2016, par rudyshihhh

  I tried to record the audio of web page by ffmpeg and it worked well with the comment as below.

  ~$ ffmpeg -f alsa -ac 2 -i pulse output5.wav

  and could get the audio file which was saved in my local device. However, recording could not work properly after I added the parameter "-t" to limit the duration of recording. For example, if I set -t 20, the saved file only 0 or 20 KB but the comment without -t can get about 4MB. Is there any problem with my comment ? or Is there any other manner to record the audio by ffmpeg in limited time ?

  ~$ ffmpeg -f alsa -ac 2 -i pulse -t 20 output5.wav
  
  
  
  ffmpeg version 0.8.17-4:0.8.17-0ubuntu0.12.04.1, Copyright (c) 2000-2014 the Libav developers
  
  built on Mar 16 2015 13:26:50 with gcc 4.6.3
  
  The ffmpeg program is only provided for script compatibility and will be removed
  
  in a future release. It has been deprecated in the Libav project to allow for
  
  incompatible command line syntax improvements in its replacement called avconv
  
  (see Changelog for details). Please use avconv instead.
  
  [alsa @ 0x212c7a0] capture with some ALSA plugins, especially dsnoop, may hang.
  
  [alsa @ 0x212c7a0] Estimating duration from bitrate, this may be inaccurate 
  
  Input #0, alsa, from 'pulse':
  
    Duration: N/A, start: 1453692638.847782, bitrate: N/A
  
      Stream #0.0: Audio: pcm_s16le, 48000 Hz, 2 channels, s16, 1536 kb/s
  
  Output #0, wav, to 'output5.wav':
  
    Metadata:
  
      encoder         : Lavf53.21.1
  
      Stream #0.0: Audio: pcm_s16le, 48000 Hz, 2 channels, s16, 1536 kb/s
  
  Stream mapping:
  
    Stream #0.0 -> #0.0
  
  Press ctrl-c to stop encoding
  
  size=      28kB time=0.15 bitrate=1538.5kbits/s    
  
  video:0kB audio:28kB global headers:0kB muxing overhead 0.162636%