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• FFMPEG Create internal pipeline for adding raw frames to AVI file (no input file)

  2 avril 2015, par Katie

  I have an application that reads in a raw video file, does some image processing to each frame, then feeds the resulting BGRA-format byte[] frames to the FFMPEG container to eventually create an AVI file. Since this process works slightly differently than any other FFMPEG example I’ve seen in that it does not have an existing input file, I’m wondering if anyone knows how to do this.

  I initialize the FFMPEG container :

  ProcessBuilder pBuilder = new ProcessBuilder(raid.getLocation()
  
                  + "\\ffmpeg\\bin\\ffmpeg.exe", "-r", "30", "-vcodec",
  
                  "rawvideo", "-f", "rawvideo", "-pix_fmt", "bgra", "-s",
  
                  size, "-i", "pipe:0", "-r", "30", "-y", "-c:v", "libx264",
  
                  "C:\export\2015-02-03\1500\EXPORT6.avi");
  
  
  
   try
  
   {
  
       process = pBuilder.start();
  
   }
  
   catch (IOException e)
  
   {
  
       e.printStackTrace();
  
   }
  
  
  
    ffmpegInput = process.getOutputStream();

  For each incoming byte[] array frame, I add the frame to the container ("src" is a BufferedImage that I’m converting to a byte array) :

  try
  
  {
  
       ByteArrayOutputStream baos = new ByteArrayOutputStream();
  
       ImageIO.write(src, ".png", baos);
  
       ffmpegInput.write(baos.toByteArray());
  
  }
  
  catch (IOException e)
  
  {
  
       e.printStackTrace();
  
  }

  And once the video is finished loading frames, I close the container :

  try
  
  {
  
       ffmpegInput.flush();
  
       ffmpegInput.close();
  
  }
  
  catch (IOException e)
  
  {
  
       e.printStackTrace();
  
  }

  The AVI file is created but it displays an error when opening. The FFMPEG logger displays this as the error :

  ffmpeg version N-71102-g1f5d1ee Copyright (c) 2000-2015 the FFmpeg developers built with gcc 4.9.2 (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-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-lzma --enable-decklink --enable-zlib
  
    libavutil      54. 20.101 / 54. 20.101
  
    libavcodec     56. 30.100 / 56. 30.100
  
    libavformat    56. 26.101 / 56. 26.101
  
    libavdevice    56.  4.100 / 56.  4.100
  
    libavfilter     5. 13.101 /  5. 13.101
  
    libswscale      3.  1.101 /  3.  1.101
  
    libswresample   1.  1.100 /  1.  1.100
  
    libpostproc    53.  3.100 / 53.  3.100
  
  Input #0, rawvideo, from 'pipe:0':
  
    Duration: N/A, bitrate: 294912 kb/s
  
      Stream #0:0: Video: rawvideo (BGRA / 0x41524742), bgra, 640x480, 294912 kb/s, 30 tbr, 30 tbn, 30 tbc
  
  No pixel format specified, yuv444p for H.264 encoding chosen.
  
  Use -pix_fmt yuv420p for compatibility with outdated media players.
  
  [libx264 @ 00000000003bcbe0] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2
  
  [libx264 @ 00000000003bcbe0] profile High 4:4:4 Predictive, level 3.0, 4:4:4 8-bit
  
  Output #0, avi, to 'C:\export\2015-02-03\1500\EXPORT6.avi':
  
    Metadata:
  
      ISFT            : Lavf56.26.101
  
      Stream #0:0: Video: h264 (libx264) (H264 / 0x34363248), yuv444p, 640x480, q=-1--1, 30 fps, 30 tbn, 30 tbc
  
      Metadata:
  
        encoder         : Lavc56.30.100 libx264
  
  Stream mapping:
  
    Stream #0:0 -> #0:0 (rawvideo (native) -> h264 (libx264))
  
  frame=    0 fps=0.0 q=0.0 Lsize=       6kB time=00:00:00.00 bitrate=N/A    
  
  video:0kB audio:0kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: unknown
  
  Output file is empty, nothing was encoded (check -ss / -t / -frames parameters if used)

  Any insight or ideas would be greatly appreciated !

• Screen capture WITH audio in ffmpeg

  6 avril 2015, par Adam Kurkiewicz

  I am using ffmpeg built from source at this revision in Fedora 20.

  I am able to record audio perfectly fine with the command :

  FFmpeg/ffmpeg -f alsa -ac 2 -i pulse -- output.wav

  However, when I try both screen capture and audio, like this :

  FFmpeg/ffmpeg -video_size 800x600 -framerate 25 -ac 2 -f x11grab -i :0.0+0,0 -ac 2 -f alsa -i pulse -ac 2 -acodec copy output.mpeg -ac 2

  I’m getting a video with 0 audio channels, which I can check running FFmpeg/ffplay output.mpeg

  Input #0, mpeg, from 'output.mpeg':
  
    Duration: 00:00:09.44, start: 0.540000, bitrate: 2743 kb/s
  
      Stream #0:0[0x1e0]: Video: mpeg1video, yuv420p(tv), 800x600 [SAR 1:1 DAR 4:3], 104857 kb/s, 25 fps, 25 tbr, 90k tbn, 25 tbc
  
      Stream #0:1[0x1c0]: Audio: mp2, 0 channels, s16p
  
     7.26 M-V:  0.000 fd=   0 aq=    0KB vq=   33KB sq=    0B f=0/0   

  and no audible sound is played. The position of -ac 2 doesn’t change the behaviour, nor does repeating it only once. This is despite ffmpeg claims to record the audio with 2 channels, here is the input :

  ffmpeg version N-71312-ga66dcfe Copyright (c) 2000-2015 the FFmpeg developers
  
    built with gcc 4.8.3 (GCC) 20140911 (Red Hat 4.8.3-7)
  
    configuration: 
  
    libavutil      54. 22.100 / 54. 22.100
  
    libavcodec     56. 34.100 / 56. 34.100
  
    libavformat    56. 29.100 / 56. 29.100
  
    libavdevice    56.  4.100 / 56.  4.100
  
    libavfilter     5. 13.101 /  5. 13.101
  
    libswscale      3.  1.101 /  3.  1.101
  
    libswresample   1.  1.100 /  1.  1.100
  
  Trailing options were found on the commandline.
  
  Input #0, x11grab, from ':0.0+0,0':
  
    Duration: N/A, start: 1428348285.201679, bitrate: N/A
  
      Stream #0:0: Video: rawvideo (BGR[0] / 0x524742), bgr0, 800x600, 25 fps, 25 tbr, 1000k tbn, 25 tbc
  
  Guessed Channel Layout for  Input Stream #1.0 : stereo
  
  Input #1, alsa, from 'pulse':
  
    Duration: N/A, start: 1428348285.225901, bitrate: 1536 kb/s
  
      Stream #1:0: Audio: pcm_s16le, 48000 Hz, 2 channels, s16, 1536 kb/s
  
   File 'output.mpeg' already exists. Overwrite ? [y/N] y
  
  [mpeg @ 0x2d079a0] VBV buffer size not set, using default size of 130KB
  
  If you want the mpeg file to be compliant to some specification
  
  Like DVD, VCD or others, make sure you set the correct buffer size
  
  Output #0, mpeg, to 'output.mpeg':
  
    Metadata:
  
      encoder         : Lavf56.29.100
  
      Stream #0:0: Video: mpeg1video, yuv420p, 800x600, q=2-31, 200     kb/s, 25 fps, 90k tbn, 25 tbc
  
      Metadata:
  
        encoder         : Lavc56.34.100 mpeg1video
  
      Stream #0:1: Audio: pcm_s16le, 48000 Hz, stereo, 1536 kb/s
  
  Stream mapping:
  
    Stream #0:0 -> #0:0 (rawvideo (native) -> mpeg1video (native))
  
    Stream #1:0 -> #0:1 (copy)

  I’m clueless. If anybody can tell me how to get this to work by either :

  • fixing my mistakes in the command line flags
  • if it’s a regression, which revision to build from.
  • any other way

  I’ll buy him/her a pint.

• How to join AAC files with AVCONV CONCAT using an input txt file ?

  26 mars 2015, par user27284

  I have a number of AAC files and want to concatenate them into one AAC file. When I do this with AVCONV at the command line and I am specific with the file names, it works. When I try to do this with a text file that holds a list of files, it fails. What led me here is the FFMPEG tutorial page at Concatenating media files. I get a variety of results depending on what I try and I cannot find any info on what I am doing wrong with the syntax.

  My file list (list.txt in the same directory)...

  file 'sr_program_2015_03_23_05_44_01.aac'
  
  file 'sr_program_2015_03_23_07_44_58.aac'

  When I follow the example at the page I mentioned above, I get an error. I included both the use of ffmpeg AND avconv but the results are the same.

  ffmpeg...

  ffmpeg -f concat -i list.txt -c copy output
  
  ffmpeg version 0.8.17-6:0.8.17-1, Copyright (c) 2000-2014 the Libav developers
  
    built on Mar 15 2015 17:00:31 with gcc 4.7.2
  
  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.
  
  Unknown input format: 'concat'

  avconv...

  avconv -f concat -i list.txt -c copy output
  
  avconv version 0.8.17-6:0.8.17-1, Copyright (c) 2000-2014 the Libav developers
  
    built on Mar 15 2015 17:00:31 with gcc 4.7.2
  
  Unknown input format: 'concat'

  When I remove the "-f" and use the filenames directly, it works. I have no idea why. The resulting file also plays as expected.

  avconv -i concat:sr_program_2015_03_23_05_44_01.aac\|sr_program_2015_03_23_07_44_58.aac -c copy output.aac
  
  avconv version 0.8.17-6:0.8.17-1, Copyright (c) 2000-2014 the Libav developers
  
    built on Mar 15 2015 17:00:31 with gcc 4.7.2
  
  [aac @ 0xcb6cc0] channel element 3.5 is not allocated
  
  [aac @ 0xcb4b20] max_analyze_duration reached
  
  [aac @ 0xcb4b20] Estimating duration from bitrate, this may be inaccurate
  
  Input #0, aac, from 'concat:sr_program_2015_03_23_05_44_01.aac|sr_program_2015_03_23_07_44_58.aac':
  
    Duration: 01:58:34.29, bitrate: 65 kb/s
  
      Stream #0.0: Audio: aac, 44100 Hz, stereo, s16, 65 kb/s
  
  Output #0, adts, to 'output.aac':
  
    Metadata:
  
      encoder         : Lavf53.21.1
  
      Stream #0.0: Audio: aac, 44100 Hz, stereo, 65 kb/s
  
  Stream mapping:
  
    Stream #0:0 -> #0:0 (copy)
  
  Press ctrl-c to stop encoding
  
  size=   57148kB time=7315.03 bitrate=  64.0kbits/s
  
  video:0kB audio:57148kB global headers:0kB muxing overhead 0.000000%

  Noticing that the working approach does not use the ’-f’ option, I tried that again with my first attempt and got a completely different error.

  avconv concat -i list.txt -c copy output.aac
  
  avconv version 0.8.17-6:0.8.17-1, Copyright (c) 2000-2014 the Libav developers
  
    built on Mar 15 2015 17:00:31 with gcc 4.7.2
  
  Unable to find a suitable output format for 'concat'

  The reason I want to process these ’broken stream’ aac files from a file list in a file is because I want to create the list from a script/code and then process it as part of a daily automated process. When the streams have broken, there are numerous files with the same date. When all goes well, there is only one file. It happens every other week or so. I want to automate the fix that I usually do manually. I have already accomplished creating the target file list, ironically, from examples on the same page mentioned above when I became stuck with this odd behavior.

  I also want to know what it is I am doing wrong. I see the example I used first everywhere. I have tried this on two different machines running debian but different architectures (arm and x86) and received the same results.

  Also, to make sure I had the latest ffmeg, I compiled it on each system using this page... Compile FFmpeg on Ubuntu, Debian, or Mint

  Thank you for your time.