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• avformat/format : silence -Wdiscarded-qualifiers

  19 septembre 2015, par Ganesh Ajjanagadde

  avformat/format : silence -Wdiscarded-qualifiers
  lpd.buf is non-const and discards the const qualifier of zerobuffer.
  
  This fixes -Wdiscarded-qualifiers observed with a variety of compilers, including GCC 5.2.
  
  Note that this does not change the type of zerobuffer, and merely makes the intent explicit.
  Signed-off-by : Ganesh Ajjanagadde <gajjanagadde@gmail.com>
  
  Signed-off-by : Michael Niedermayer <michael@niedermayer.cc>
  

  • [DH] libavformat/format.c

• 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>