Recherche avancée

Sur d’autres sites (13481)

• 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 ?

• wrong play audio sasamples

  7 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 ?

• find the timestamp of a sound sample of an mp3 with linux or python

  23 juin 2020, par cardamom

  I am slowly working on a project which where it would be very useful if the computer could find where in an mp3 file a certain sample occurs. I would restrict this problem to meaning a fairly exact snippet of the audio, not just for example the chorus in a song on a different recording by the same band where it would become more some kind of machine learning problem. Am thinking if it has no noise added and comes from the same file, it should somehow be possible to locate the time at which it occurs without machine learning, just like grep can find the lines in a textfile where a word occurs.

  


  In case you don't have an mp3 lying around, can set up the problem with some music available on the net which is in the public domain, so nobody complains :

  


  curl https://web.archive.org/web/20041019004300/http://www.navyband.navy.mil/anthems/ANTHEMS/United%20Kingdom.mp3 --output godsavethequeen.mp3


  


  It's a minute long :

  


  exiftool godsavethequeen.mp3 | grep Duration
Duration                        : 0:01:03 (approx)


  


  Now cut out a bit between 30 and 33 seconds (the bit which goes la la la la..) :

  


  ffmpeg -ss 30 -to 33 -i godsavethequeen.mp3 gstq_sample.mp3


  


  both files in the folder :

  


  $ ls -la
-rw-r--r-- 1 cardamom cardamom   48736 Jun 23 00:08 gstq_sample.mp3
-rw-r--r-- 1 cardamom cardamom 1007055 Jun 22 23:57 godsavethequeen.mp3


  


  This is what am after :

  


  $ findsoundsample gstq_sample.mp3 godsavethequeen.mp3
start 30 end 33


  


  Am happy if it is a bash script or a python solution, even using some kind of python library. Sometimes if you use the wrong tool, the solution might work but look horrible, so whichever tool is more suitable. This is a one minute mp3, have not thought yet about performance just about getting it done at all, but would like some scalability, eg find ten seconds somewhere in half an hour.