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• audioCodec doesn't recognize 'aac' or 'libfdk-aac'

  24 août 2015, par Mayur Tanna

  I want to convert mp3 to aac using ffmpeg and stream it to s3 in my node application. My code is as below. Now when I try to use audioCodec as ’aac’ or ’libfdk-aac’. It throws the error. Note that ffmpeg is compiled with ’libfdk-aac’ and working fine from command line.

  Any inputs/fixes ?

  var proc = new ffmpeg({source:uploadedFiles[0].fd})
  
  .audioCodec('aac') // .audioCodec('libfdk-aac')
  
  .audioChannels(1);
  
  
  
  proc.on('error', function(err) { 
  
  console.log('An error occurred: ' + err.message);
  
  })
  
  .on('end', function() {
  
      console.log('audio uploaded successfully!');
  
  }).pipe(upload(opts));

  Thanks.

• ffmpeg issue with mp4 play and time convert for webm [on hold]

  19 mai 2015, par ali rah

  I use this code to convert mp4 to mp4 (with watermark) and webm and ogv

  and I view it via a standard html5 code.

  exec("$ffmpegPath  -i $inputPath -i $watermark -filter_complex overlay=10:10 $outPath");

  But I found 2 important problem.

  first : it generate mp4 but when open it on a browser it just play voice (no video with a black screen)

  note : I can play orginall mp4 in browsers.

  Second : I saw it can convert mp4 and ovg in a few second but for webm it takes about 3 min !!!...

  please advice me to solve my problems...

• C# library for audio resampling that has the same abilities as FFmpeg

  21 avril 2013, par Designation

  I have to use a pure C# solution for resampling audio, which can produce me the exact same results as FFmpeg's audio sampling can.

  FFmpeg first builds some kind of polyphase filter bank, and then uses that for the sampling process (sorry for the vague phrasing, but I'm not too familiar with this topic). According to this brief documentation, the initialization can be customized this way :

  AVResampleContext* av_resample_init(
  
      int     out_rate,
  
      int     in_rate,
  
      int     filter_length,
  
      int     log2_phase_count,
  
      int     linear,
  
      double  cutoff   
  
      )

  The parameters are :

  • out_rate : output sample rate
  • in_rate : input sample rate
  • filter_length : length of each FIR filter in the filterbank relative to the cutoff freq
  • log2_phase_count : log2 of the number of entries in the polyphase filterbank
  • linear : if 1 then the used FIR filter will be linearly interpolated between the 2 closest, if 0 the closest will be used
  • cutoff : cutoff frequency, 1.0 corresponds to half the output sampling rate

  I'd need to use a C# library that is configurable in the same depth. I've been trying to use NAudio (more specifically, its WaveFormatConversionStream class), but there, I could only set the input and output sample rates, so I didn't get the expected results.

  So, is there a C# lib that could resample with the same settings as FFmpeg can ? Or one that has almost all of these settings or similar ones ? Note : I need a C# solution, not a wrapper !