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• Trying loop an mp3 and output the mp3 with ffmpeg

  12 février 2023, par Chewie The Chorkie

  I have a 7 second mp3. I want to loop it 3 times and give me back the output. It only plays once in the output, though.

  


  I've tried setting the duration to 21 seconds instead (the atrim parameter), but in that case the output will only play once, and the remaining 14 seconds is silent.

  


  ffmpeg -i explosion.mp3 -filter_complex "[0:a]aresample=async=1:min_hard_comp=0.100000:first_pts=0,apad,atrim=end=7,aloop=loop=3:size=0[outa]" -map "[outa]" -c:a libmp3lame -q:a 2 output.mp3


  


• wrong play audio samples

  8 août 2014, par Ivan Lisovich

  I have a problem with a ffmpeg and NAudio libs.
  I worked with the old ffmpeg library and there the audio plays correctly.
  read video in manage c++

  // Read frames and save to list audio frames
  
  while(av_read_frame(pFormatCtx, &packet) >= 0) 
  
  {
  
      if(packet.stream_index == videoStream) 
  
      {
  
          // reade image
  
      }
  
      else if(packet.stream_index == audioStream) 
  
      {
  
          int b = av_dup_packet(&packet);
  
          if(b >= 0) {
  
              int audio_pkt_size = packet.size;
  
              libffmpeg::uint8_t* audio_pkt_data = packet.data;
  
              while(audio_pkt_size > 0) 
  
              {
  
                  int got_frame = 0;
  
                  int len1 = libffmpeg::avcodec_decode_audio4(aCodecCtx, &frame, &got_frame, &packet);
  
                  if(len1 < 0)
  
                  {
  
                      /* if error, skip frame */
  
                      audio_pkt_size = 0;
  
                      break;
  
                  }
  
                  audio_pkt_data += len1;
  
                  audio_pkt_size -= len1;
  
                  if (got_frame)
  
                  {
  
                      int data_size = libffmpeg::av_samples_get_buffer_size ( NULL, aCodecCtx->channels, frame.nb_samples, aCodecCtx->sample_fmt, 1 );
  
                      array<byte>^ managedBuf = gcnew array<byte>(data_size);
  
                      System::IntPtr iptr = System::IntPtr( frame.data[0] );
  
                      System::Runtime::InteropServices::Marshal::Copy( iptr, managedBuf, 0, data_size );
  
                      audioData->Add(managedBuf);
  
                  }
  
              }
  
          }
  
      }
  
      // Free the packet that was allocated by av_read_frame
  
      libffmpeg::av_free_packet(&amp;packet);
  
  }
  
  </byte></byte>

  I return audioData to c# code and play in NAudio library
  play in c#

  var recordingFormat = new WaveFormat(reader.SampleRate, 16, reader.Channels);
  
  var waveProvider = new BufferedWaveProvider(recordingFormat) { DiscardOnBufferOverflow = true, BufferDuration = TimeSpan.FromMilliseconds(10000) };
  
  var waveOut = new DirectSoundOut();
  
  waveOut.Init(waveProvider);
  
  waveOut.Play();
  
  foreach (byte[] data in audioData)
  
  {
  
      waveProvider.AddSamples(data, 0, data.Length);
  
  }

  but audio not playing.
  what am I doing wrong ?

• CoreAudio : how to retrieve actual sampling rate of raw data ?

  13 août 2014, par jyavenard

  When attempting to play AAC-HE content in an mp4 container, the reported sampling rate found in the mp4 container appears to be half of the actual sampling rate.

  E.g it appears as 24kHz instead of 48kHz.

  Using the FFmpeg AAC decoder, retrieving the actual sampling rate can be done by simply decoding an audio packet using

  avcodec_decode_audio4

  And looking at AVCodecContext::sample_rate which will be updated appropriately. From that it’s easy to adapt the output.

  With CoreAudio decoder, I would use a AudioConverterRef set the input and output AudioStreamBasicDescription
  and call AudioConverterFillComplexBuffer

  As the converter performs all the required internal conversion including resampling it’s fine. But it plays the content after resampling it to 24kHz (as that’s what the input AudioStreamBasicDescription contains.

  Would there be a way to retrieve the actual sampling rate as found be the decoder (rather than the demuxer) in a similar fashion as one can with FFmpeg ?

  Would prefer to avoid losing audio quality if at all possible, and not downmix data

  Thanks