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• Add a timestamp overlay using ffmpeg by Node Js

  20 septembre 2015, par Nazmul Hossain Bilash

  I am working on video editing. Currently I want to add time stamp on a video. I have found some command line code from a website. But how can I add this by Node Js FFMPEG module.

  ffmpeg -f dshow -i video="HD Pro Webcam C920" -vf "drawtext=fontfile=arial.ttf:text='%m';fontcolor=white@0.8:x=7:y=460" -vcodec libx264 -vb 2000k -preset ultrafast -f mp4 output.mp4

  Some notes : arial.tff lives in the same directory as ffmpeg

  Here is the website link.

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

  2 juillet 2013, 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.

• undefined reference to 'av_rdft_init(int, RDFTransformType)'

  25 juin 2015, par mavixce

  I’m trying to integrate ffmpeg library to an android ndk project. I followed this documentation. When i try to call a method from avcodec library like as av_malloc, there is no problem. But when i try to call av_rdft_init or av_rdft_calc methods, ndk gives errors like as ;

   error: undefined reference to 'av_rdft_init(int, RDFTransformType)'  
  
   error: undefined reference to 'av_rdft_calc(RDFTContext*)'

  I can see those methods in header file and also see in libavcodec.so file’s symbol table but could not build with ndk.

  Any ideas to fix this problem ?
  Thanks in advance.