Newest 'ffmpeg' Questions - Stack Overflow

Les articles publiés sur le site

• C# and FFmpeg preferably without shell commands ?

  14 août 2018, par Dominic Bou-Samra

  I'd like to be able to use FFmpeg to convert a video file from within my C# program. I know I can just call a shell command, but is there a better way?

  The issue with invoking a command via the shell, is I'm not sure you could do things like a progress bar, etc... or could you?

  If there isn't a way, can anyone suggest the best way to layout some framework for executing shell commands. Passing one big long string is very cumbersome atm.

• ffmpeg run from shell runs properly, but does not when called from within .NET

  14 août 2018, par Bevin

  I'm attempting to use ffmpeg (compiled on Windows with Cygwin) in a C# program, by using the Process class to spawn an ffmpeg instance. However, I've hit a rather odd bug that doesn't make much sense.

  When I run ffmpeg directly from a shell (be it Cygwin's bash, PowerShell, cmd), ffmpeg can properly decode and reencode files without any issues:

  PS C:\audio> ffmpeg -i .\sound1.wav -acodec libvorbis -f ogg abc.ogg
  ffmpeg version 1.2 Copyright (c) 2000-2013 the FFmpeg developers
    built on Apr  8 2013 15:10:40 with gcc 4.5.3 (GCC)
    configuration: --disable-encoder=vorbis --enable-libvorbis
    libavutil      52. 18.100 / 52. 18.100
    libavcodec     54. 92.100 / 54. 92.100
    libavformat    54. 63.104 / 54. 63.104
    libavdevice    54.  3.103 / 54.  3.103
    libavfilter     3. 42.103 /  3. 42.103
    libswscale      2.  2.100 /  2.  2.100
    libswresample   0. 17.102 /  0. 17.102
  [wav @ 0x800538a0] max_analyze_duration 5000000 reached at 5015510 microseconds
  Guessed Channel Layout for  Input Stream #0.0 : stereo
  Input #0, wav, from '.\sound1.wav':
    Metadata:
      encoder         : Lavf54.63.104
    Duration: 00:00:05.76, bitrate: 1411 kb/s
      Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, stereo, s16, 1411 kb/s
  Output #0, ogg, to 'abc.ogg':
    Metadata:
      encoder         : Lavf54.63.104
      Stream #0:0: Audio: vorbis, 44100 Hz, stereo, fltp
  Stream mapping:
    Stream #0:0 -> #0:0 (pcm_s16le -> libvorbis)
  Press [q] to stop, [?] for help
  size=      55kB time=00:00:05.74 bitrate=  78.5kbits/s
  video:0kB audio:51kB subtitle:0 global headers:4kB muxing overhead 0.817473%
  

  The file plays fine, and I can encode to WAV or any other format I like. However, when I call ffmpeg from C# with the following code:

  string tempfile = Path.GetTempFileName();
  FileStream tempfilestr = File.OpenWrite(tempfile);
  input.CopyTo(tempfilestr);
  
  ProcessStartInfo pstart = new ProcessStartInfo("ffmpeg", string.Format("-i \"{0}\" -v verbose -y -f wav -", tempfile));
  pstart.CreateNoWindow = true;
  pstart.ErrorDialog = false;
  pstart.RedirectStandardOutput = true;
  pstart.RedirectStandardError = true;
  pstart.UseShellExecute = false;
  
  
  Process proc = new Process();
  proc.StartInfo = pstart;
  proc.Start();
  StreamReader stdout = proc.StandardOutput;
  StreamReader stderr = proc.StandardError;
  
  outtempfilestr = File.OpenRead(outtempfile);
  MemoryStream output = new MemoryStream();
  
  stdout.BaseStream.CopyTo(output);
  
  try {
      proc.Kill();
  }
  catch(InvalidOperationException) { }
  catch(Win32Exception) { }
  
  File.Delete(tempfile);
  
  return output.ToArray();
  

  This randomly produces errors in the output:

  ffmpeg version 1.2 Copyright (c) 2000-2013 the FFmpeg developers
    built on Apr  8 2013 15:10:40 with gcc 4.5.3 (GCC)
    configuration: --disable-encoder=vorbis --enable-libvorbis
    libavutil      52. 18.100 / 52. 18.100
    libavcodec     54. 92.100 / 54. 92.100
    libavformat    54. 63.104 / 54. 63.104
    libavdevice    54.  3.103 / 54.  3.103
    libavfilter     3. 42.103 /  3. 42.103
    libswscale      2.  2.100 /  2.  2.100
    libswresample   0. 17.102 /  0. 17.102
  [wav @ 0x80053860] parser not found for codec pcm_s16le, packets or times may be invalid.
      Last message repeated 1 times
  [wav @ 0x80053860] max_analyze_duration 5000000 reached at 5015510 microseconds
  Guessed Channel Layout for  Input Stream #0.0 : stereo
  Input #0, wav, from 'C:\Users\Bevin\AppData\Local\Temp\tmp1CCE.tmp':
    Duration: 00:00:05.20, bitrate: 1411 kb/s
      Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, stereo, s16, 1411 kb/s
  [graph 0 input from stream 0:0 @ 0x8011f320] tb:1/44100 samplefmt:s16 samplerate:44100 chlayout:0x3
  Output #0, wav, to 'pipe:':
    Metadata:
      ISFT            : Lavf54.63.104
      Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, stereo, s16, 1411 kb/s
  Stream mapping:
    Stream #0:0 -> #0:0 (pcm_s16le -> pcm_s16le)
  Press [q] to stop, [?] for help
  Multiple frames in a packet from stream 0
  [pcm_s16le @ 0x8005c160] Invalid PCM packet, data has size 3 but at least a size of 4 was expected
  Error while decoding stream #0:0: Invalid data found when processing input
  No more output streams to write to, finishing.
  size=     896kB time=00:00:05.20 bitrate=1411.3kbits/s    
  video:0kB audio:896kB subtitle:0 global headers:0kB muxing overhead 0.008719%
  

  Note that these errors don't always occur. Sometimes they happen for certain files, sometimes they don't. I've tried various combinations of stream redirects and temp files, none of them work. I've also verified the integrity of the temp files, and it all checks out. I've even extracted the temp file before it was deleted, and decoded it in shell without a hitch.

  Any ideas?

  Edit: I've tried running ffmpeg from a shell script that's run through C#. It gives the same issues. Compiling ffmpeg via MinGW gives the same issue as well.

• How to use pipe in ffmpeg within c#

  14 août 2018, par Andrew Simpson

  I have 100 jpegs.

  I use ffmpeg to encode to a video file which is written to a hard drive.

  Is there a way to pipe it directly to a byte/stream?

  I am using C# and I am using the process class to initate ffmpeg.

  Thanks

• HLS stream from multiple FFMPEG to RTMP command at VideoJS keep repeating for segments

  14 août 2018, par Lokesh Kumawat

  I am building on demand video streaming application based on user interaction at frontend using FFMPEG and RTMP, which eventually converted to HLS using nginx-rtmp-module, with hls_continuous flag set to true.

  While running back to back FFMPEG command to RTMP(i.e. once one FFMPEG command done with execution at RTMP stream, another FFMPEG command is executed at same stream), observation at VideoJs player that some of the HLS segment keeps repeating.

  Would be great help if someone could help me to figure out what could be possible reason, and how to fix the same?

  Thanks in Advance.

• How to change bitrate mode : VBR to CBR with MPEG4 of H264 file ?

  14 août 2018, par The Bird

  I've tried to convert bitrate mode from VBR to CBR with FFMPEG library,but bitrate mode cannot change. My command line:

  ffmpeg -i -f mpegts -acodec libmp3lame -ar 48000 -ab 64k -s 320×240 -vcodec libx264 -b 96k -flags +loop -cmp +chroma -partitions +parti4x4+partp8x8+partb8x8 -subq 5 -trellis 1 -refs 1 -coder 0 -me_range 16 -keyint_min 25 -sc_threshold 40 -i_qfactor 0.71 -bt 200k -maxrate 96k -bufsize 96k -rc_eq 'blurCplx^(1-qComp)' -qcomp 0.6 -qmin 10 -qmax 51 -qdiff 4 -level 30 -aspect 320:240 -g 30 -async 2

  I found some apps but don't have any app that i can change bitrate mode. Can anyone point me why my cmd cannot change mode or the app can does that?. I think CBR is a mode that bitrate is the same all time,it's true? Thanks