Newest 'ffmpeg' Questions - Stack Overflow

Les articles publiés sur le site

• Distributed video decoding over a network

  26 mars 2015, par tkcast

  i'm developing a videowall controller. I can use any technology of programming language needed, and I want to decode videos of arbitrarily high resolution on my videowall.

  One possible solution is: -split the ultra high video into several slices using ffmpeg and have one computer to decode each tile of the videowall separately. I'd use the network only to control the playback

  Another interesting solution: -only a master computer would have the huge video, and it would control a distributed decoding over the network. Is it even possible? how?

  Thanks!

• MKV to MP4 conversion using ffmpeg gives error

  26 mars 2015, par ketan

  I am trying to convert MKV to MP4 then it gives error in some file convention.

  [matroska,webm @ 0000000004417a80] Read error at pos. 17896 (0x45e8)
  [matroska,webm @ 0000000004417a80] Could not find codec parameters for stream 0
  (Video: h264, 1280x720): unspecified pixel format
  

  
  I am using it in Windows 7.

  I use: ffmpeg -i "filename.mkv" -vcodec copy -acodec copy "filename.mp4" to convert video from MKV to mp4 with same quality.

  Some Video converted properly but some gives error like:

  libxavs --enable-libxvid --enable-decklink --enable-zlib
    libavutil      52. 92.100 / 52. 92.100
    libavcodec     55. 69.100 / 55. 69.100
    libavformat    55. 45.100 / 55. 45.100
    libavdevice    55. 13.102 / 55. 13.102
    libavfilter     4. 10.100 /  4. 10.100
    libswscale      2.  6.100 /  2.  6.100
    libswresample   0. 19.100 /  0. 19.100
    libpostproc    52.  3.100 / 52.  3.100
  [matroska,webm @ 0000000004417a80] Read error at pos. 17896 (0x45e8)
  [matroska,webm @ 0000000004417a80] Could not find codec parameters for stream 0
  (Video: h264, 1280x720): unspecified pixel format
  Consider increasing the value for the 'analyzeduration' and 'probesize' options
  Input #0, matroska,webm, from 'filename.mkv':
    Metadata:
      encoder         : libebml v1.3.0 + libmatroska v1.4.1
      creation_time   : 2014-10-18 00:32:40
    Duration: 00:42:09.40, start: 0.000000, bitrate: 3321 kb/s
      Stream #0:0: Video: h264, 1280x720, SAR 1:1 DAR 16:9, 23.98 fps, 23.98 tbr,
  1k tbn, 2k tbc
      Stream #0:1: Audio: ac3, 48000 Hz, 5.1(side), fltp, 384 kb/s (default)
  [mp4 @ 000000000442ec60] Using AVStream.codec.time_base as a timebase hint to th
  e muxer is deprecated. Set AVStream.time_base instead.
      Last message repeated 1 times
  [mp4 @ 000000000442ec60] track 1: codec frame size is not set
  Output #0, mp4, to 'filename.mp4':
    Metadata:
      encoder         : Lavf55.45.100
      Stream #0:0: Video: h264 ([33][0][0][0] / 0x0021), 1280x720 [SAR 1:1 DAR 16:
  9], q=2-31, 23.98 fps, 16k tbn, 1k tbc
      Stream #0:1: Audio: ac3 ([165][0][0][0] / 0x00A5), 48000 Hz, 5.1(side), 384
  kb/s (default)
  Stream mapping:
    Stream #0:0 -> #0:0 (copy)
    Stream #0:1 -> #0:1 (copy)
  Press [q] to stop, [?] for help
  frame=    0 fps=0.0 q=-1.0 Lsize=      13kB time=00:00:00.25 bitrate= 406.1kbits
  /s
  video:0kB audio:12kB subtitle:0kB other streams:0kB global headers:0kB muxing ov
  erhead: 5.745443%
  

  How can i solved using ffmpeg in Windows?
  Any help is appreciate.
  
  Thanks.

• How to increase compression speed for FFmpeg ?

  26 mars 2015, par Shailesh

  I have used following command for compress video in android.

  ffmpeg -y -i /sdcard/DCIM/Camera/VID_20150326_125017.mp4 -strict experimental -s 640x360 -r 25 -vcodec mpeg4 -b 1000k -ab 48000 -ac 2 -ar 22050 /sdcard/videokit/out.mp4
  

  Above command is a work properly, but it's taking too much time for compression. An 80MB video file takes around 3 minutes to compress to an 8MB file. Is there any way I can reduce this time.

• How to reduce bit rate of an audio file using ffmpeg in java

  26 mars 2015, par Mehandi Hassan

  How to reduce bit rate of an audio file using ffmpeg in java

  I am not able to pass more then 4 argument in process builder.

  Here the code for video and i want audio.

   ProcessBuilder pb = new ProcessBuilder(
                              "ffmpeg",
                              "-i",
                              "C:/Users/Amit/Videos/Wildlife.wmv",
                              "C:/Users/Amit/Videos/Wildlife12.avi");
  

• 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.