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• FFMPEG - Struggling to find correct input audio codec parameters on macOS

  20 septembre 2024, par Xavi

  I am trying to read my external stereo microphone with ffmpeg within my Qt Windows+macOs application, but I am struggling to obtain consistent correct input codec parameters on macOs. My findings and suspicions so far :

  


  The code I'm using in macOs is the following, where everything returns a successful return code :

  


   avdevice_register_all();
 
 //macOs only, the same code in windows looks for "dshow" 
 const AVInputFormat *inputFormat = av_find_input_format("avfoundation");
 
 AVFormatContext* inputFormatContext;
 avformat_open_input(&inputFormatContext, inputDevice, inputFormat, NULL);

 avformat_find_stream_info(inputFormatContext, NULL);
 
 //... allocate the codec context for the single input stream and
 // copy the parameters from the stream to the context



  


  In my standalone minimal reproducer this always results on the codec ID of the single stream being AV_CODEC_ID_PCM_F32LE, in both macOS and Windows. When I integrate this code in my Qt application on Windows, I get the same result. However, on macOS, most of the times results in the codec id of the stream being AV_CODEC_ID_PCM_S16LE (via AV_CODEC_ID_FIRST_AUDIO) and sometimes AV_CODEC_ID_PCM_F32LE. Both sample formats are supported by my microphone.

  


  AV_CODEC_ID_PCM_F32LE always results in a correct output. AV_CODEC_ID_PCM_S16LE results on buzzy noisy audio slowed down to 0.5x, and If in this case I decode with AV_CODEC_ID_PCM_F32LE instead of copying the codec parameters from the stream, the output sounds correct again.

  


  I am trying to write generic code, so while enforcing the AV_CODEC_ID_PCM_F32LE codec works, I'd rather understand what is happening.

  


  What am I missing ? Is Qt interacting in some way that I can't think of ? I am compiling and linking my own ffmpeg libraries (6.1.1) and not using Qt's ones.

  


• FFMpegCore .Net cuts off the fist few ( 1.5) seconds of audio

  28 mai 2024, par Tim

  I want to use FFMpegCore to convert some audio files to raw pcm. I noticed that this always cuts off 1.5 seconds of my audio from the start. I check my input stream, saved it to HD all good. If use it from cli with the same arguments everything seem fine. I tried -ss 0, no luck. This behavior is observed with .wav (RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, stereo 44100 Hz), same issue with different sample rate. I tested mp3 works fine.

  


  public async Task<memorystream> ConvertToPcmStreamAsync(Stream inputStream)&#xA;{&#xA;    var outputStream = new MemoryStream();&#xA;    &#xA;    var audioInput = new StreamPipeSource(inputStream);&#xA;    var audioOutput = new StreamPipeSink(outputStream);&#xA;&#xA;    await FFMpegArguments&#xA;        .FromPipeInput(audioInput)&#xA;        .OutputToPipe(audioOutput, options => options&#xA;            .WithCustomArgument("-ss 0 -f s16le -acodec pcm_s16le -ac 1"))&#xA;        .ProcessAsynchronously();&#xA;&#xA;    // Reset the position of the memory stream to the beginning&#xA;    outputStream.Position = 0;&#xA;&#xA;    return outputStream;&#xA;}&#xA;</memorystream>

  


• avformat/mpegts : skip non-PMT tids earlier

  9 mai 2018, par Aman Gupta

  avformat/mpegts : skip non-PMT tids earlier
  This mimics the logic flow in all the other callbacks
  
  (pat_cb, sdt_cb, m4sl_cb), and avoids calling skip_identical()
  
  for non PMT_TID packets.
  Since skip_identical modifies internal state like
  
  MpegTSSectionFilter.last_ver, this change prevents unnecessary
  
  reprocessing on some streams which contain multiple tables in
  
  the PMT pid. This can be observed with streams from certain US
  
  cable providers, which include both tid=0x2 and another unspecified
  
  tid=0xc0.
  Signed-off-by : Aman Gupta <aman@tmm1.net>
  

  • [DH] libavformat/mpegts.c