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• motion vectors using FFmpeg gives lots of noise

  18 juin 2015, par taansari

  I have a video recorded from an IP surveillance camera. Link :
  https://www.dropbox.com/s/4gtcbge08nlv9rx/motion0.mp4?dl=0

  I want to extract motion info, so I use following command :

  ffmpeg -vismv pf -i motion0.mp4 -y output.mp4

  This converts the original file, marking motion vectors as well (as required). But the end result is : it is marking stationary places as noise as well (random arrows everywhere).

  I want to know why this is happening. I have already tried this command with other video(s), and random noise does not show there.

  Any help is much appreciated...

• Noise when decoding mp3 using libavcodec on iOS

  11 janvier 2014, par Trenskow

  I have this methods. It first sets up an audio input if necessary. Then it reads a frame, and returns the frame converted using an AudioConverterRef (wrapped in an KFInlineConverterUnit class) to my Core Audio graph. At the end of the method it tears down the input, if audio output has stopped.

  - (NSData *)readNextChunk {
  
  
  
      /* Setup audio input */
  
      if (_formatContext == NULL && self.runLoopActive) {
  
  
  
          static dispatch_once_t onceToken;
  
          dispatch_once(&onceToken, ^{
  
              av_register_all();
  
          });
  
  
  
          avformat_network_init();
  
  
  
          av_init_packet(&_packet);
  
  
  
          _formatContext = avformat_alloc_context();
  
  
  
          [self handleError:avformat_open_input(&_formatContext, [[_url absoluteString] cStringUsingEncoding:NSUTF8StringEncoding], NULL, NULL)];
  
  
  
          [self handleError:avformat_find_stream_info(_formatContext, NULL)];
  
  
  
          _audioStreamIndex = -1;
  
          for (NSInteger index = 0 ; index < _formatContext->nb_streams ; index++)
  
              if (_formatContext->streams[index]->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
  
                  _audioStreamIndex = index;
  
                  break;
  
              }
  
  
  
          if (_audioStreamIndex > -1) {
  
  
  
              _codec = avcodec_find_decoder(_formatContext->streams[_audioStreamIndex]->codec->codec_id);
  
  
  
              _codecContext = _formatContext->streams[_audioStreamIndex]->codec;
  
  
  
              [self handleError:avcodec_open2(_codecContext, _codec, NULL)];
  
  
  
          }
  
  
  
          if (self.timeOffset > .0) {
  
              int64_t timestamp = AV_TIME_BASE * self.timeOffset;
  
              av_seek_frame(_formatContext, -1, timestamp, AVSEEK_FLAG_ANY);
  
          }
  
  
  
      }
  
  
  
      /* Read and decode next frame */
  
  
  
      NSData *chunk = nil;        
  
      AVFrame *frame = avcodec_alloc_frame();
  
      while (self.runLoopActive && _formatContext != NULL && av_read_frame(_formatContext, &_packet) >= 0) {
  
  
  
          if (_packet.stream_index == _audioStreamIndex) {
  
  
  
              int gotFrame = 0;
  
              int len = avcodec_decode_audio4(_codecContext, frame, &gotFrame, &_packet);
  
              [self handleError:len];
  
  
  
              if (gotFrame) {
  
  
  
                  /* Setup a converter for my Audio Graph */
  
                  AudioStreamBasicDescription inputAudioDescription = [self audioDescriptionForCodecContext];
  
  
  
                  if (!_converter || memcmp(&inputAudioDescription, _converter.inputAudioDescription, sizeof(AudioStreamBasicDescription)) != 0)
  
                      _converter = [KFInlineConverterUnit newWithInputAudioDescription:inputAudioDescription
  
                                                                outputAudioDescription:self.audioDescription];
  
  
  
                  /* Convert frames */
  
                  if (inputAudioDescription.mFormatFlags & kAudioFormatFlagIsNonInterleaved) {
  
                      NSMutableArray *buffers = [@[] mutableCopy];
  
                      for (NSInteger i = 0 ; i < frame->channels ; i++) {
  
                          [buffers addObject:[NSData dataWithBytes:frame->data[i] length:frame->nb_samples * inputAudioDescription.mBytesPerFrame]];
  
                      }
  
                      chunk = [_converter convertBuffer:buffers];
  
                  } else
  
                      chunk = [_converter convertBuffer:@[[NSData dataWithBytes:frame->data[0] length:frame->nb_samples * inputAudioDescription.mBytesPerFrame]]];
  
  
  
                  if (_packet.data != NULL)
  
                      av_free_packet(&_packet);
  
  
  
                  break;
  
  
  
              }
  
  
  
          }
  
  
  
      }
  
  
  
      avcodec_free_frame(&frame);
  
  
  
      /* Tear down if output has stopped */
  
      if (!self.runLoopActive) {
  
  
  
          avformat_free_context(_formatContext);
  
          _formatContext = NULL;
  
  
  
          avformat_network_deinit();
  
  
  
      }
  
  
  
      return chunk;
  
  
  
  }

  Basically everything is fine. I am developing this for iOS, and when I test it in the iOS Simulator everything works smoothly. It plays all audio formats with no problems.

  The problem arises, when I run the code on an actual device. Then mp3 files are audible - but VERY distorted. I have tried everything in my mind to make this work, but I cannot figure what's going wrong.

  I can say that the problem does NOT lay in the converter. I've extracted the audio before it hits the decoder, and it sounds the same. So the converter just converts the distorted audio. So I think it might be an MP3 decoder issue.

  Anyone experienced this before ?

• how to remove noise from buffer

  7 décembre 2013, par Ari

  I am integrating an media player kind of app

  I have converted the audio file format into buffer, and I have applied bass,treble filters with high gain then i got noise.How to minimise or eliminate that noise from audio in ffmpeg or is there any another way such that i can remove noise using java code by doing changes in byte i.e(audio buffer)

  here is my code :

      void playSound(byte[] buf, final int size) 
  
   {     
  
       trackStereo.write(buf, 0,size);
  
       trackStereo.play();
  
  
  
      buf = null;
  
   }
  
  bufSizeStereo =  AudioTrack.getMinBufferSize(44100,
  
    AudioFormat.CHANNEL_OUT_STEREO,
  
   AudioFormat.ENCODING_PCM_16BIT);
  
      trackStereo = new AudioTrack(AudioManager.STREAM_MUSIC, 44100,
  
  AudioFormat.CHANNEL_OUT_STEREO,
  
  AudioFormat.ENCODING_PCM_16BIT, bufSizeStereo,
  
  AudioTrack.MODE_STREAM);
  
      bytes = new byte[bufSizeStereo];

  how remove noise from byte before playing the audio

  Thanks