轉換音頻立體聲爲音頻字節

Question

我試圖做一些音頻處理，我真的堅持立體聲單聲道轉換。我在互聯網上查看有關立體聲轉換爲單聲道。

據我所知，我可以把左聲道，右聲道，總和它們除以2.但是當我把結果再次轉儲到WAV文件中時，我得到了很多前景噪聲。我知道處理數據時可能會引起噪聲，字節變量中會出現一些溢出。

這是從一個MP3文件中檢索字節[]數據塊我的課：

公共類InputSoundDecoder {

private int BUFFER_SIZE = 128000; 
private String _inputFileName; 
private File _soundFile; 
private AudioInputStream _audioInputStream; 
private AudioFormat _audioInputFormat; 
private AudioFormat _decodedFormat; 
private AudioInputStream _audioInputDecodedStream; 

public InputSoundDecoder(String fileName) throws UnsuportedSampleRateException{ 
    this._inputFileName = fileName; 
    this._soundFile = new File(this._inputFileName); 
    try{ 
     this._audioInputStream = AudioSystem.getAudioInputStream(this._soundFile); 
    } 
    catch (Exception e){ 
     e.printStackTrace(); 
     System.err.println("Could not open file: " + this._inputFileName); 
     System.exit(1); 
    } 

    this._audioInputFormat = this._audioInputStream.getFormat(); 

    this._decodedFormat = new AudioFormat(AudioFormat.Encoding.PCM_SIGNED, 44100, 16, 2, 1, 44100, false); 
    this._audioInputDecodedStream = AudioSystem.getAudioInputStream(this._decodedFormat, this._audioInputStream); 

    /** Supported sample rates */ 
    switch((int)this._audioInputFormat.getSampleRate()){ 
     case 22050: 
       this.BUFFER_SIZE = 2304; 
      break; 

     case 44100: 
       this.BUFFER_SIZE = 4608; 
      break; 

     default: 
      throw new UnsuportedSampleRateException((int)this._audioInputFormat.getSampleRate()); 
    } 

    System.out.println ("# Channels: " + this._decodedFormat.getChannels()); 
    System.out.println ("Sample size (bits): " + this._decodedFormat.getSampleSizeInBits()); 
    System.out.println ("Frame size: " + this._decodedFormat.getFrameSize()); 
    System.out.println ("Frame rate: " + this._decodedFormat.getFrameRate()); 

} 

public byte[] getSamples(){ 
    byte[] abData = new byte[this.BUFFER_SIZE]; 
    int bytesRead = 0; 

    try{ 
     bytesRead = this._audioInputDecodedStream.read(abData,0,abData.length); 
    } 
    catch (Exception e){ 
     e.printStackTrace(); 
     System.err.println("Error getting samples from file: " + this._inputFileName); 
     System.exit(1); 
    } 

    if (bytesRead > 0) 
     return abData; 
    else 
     return null; 
} 

}

這意味着，每次我打電話getSamples時間，它返回一個數組，如：

buff = {Lchannel，Rchannel，Lchannel，Rchannel，Lchannel，Rchannel，Lchannel，Rchannel ...}

的處理例行程序的轉換到單聲道的樣子：

byte[] buff = null; 
     while((buff = _input.getSamples()) != null){ 

      /** Convert to mono */ 
      byte[] mono = new byte[buff.length/2]; 

      for (int i = 0 ; i < mono.length/2; ++i){ 
       int left = (buff[i * 4] << 8) | (buff[i * 4 + 1] & 0xff); 
       int right = (buff[i * 4 + 2] <<8) | (buff[i * 4 + 3] & 0xff); 
       int avg = (left + right)/2; 
       short m = (short)avg; /*Mono is an average between 2 channels (stereo)*/ 
       mono[i * 2] = (byte)((short)(m >> 8)); 
       mono[i * 2 + 1] = (byte)(m & 0xff); 
      } 

}

和寫入到使用wav文件：

 public static void writeWav(byte [] theResult, int samplerate, File outfile) { 
     // now convert theResult into a wav file 
     // probably should use a file if samplecount is too big! 
     int theSize = theResult.length; 


     InputStream is = new ByteArrayInputStream(theResult); 
     //Short2InputStream sis = new Short2InputStream(theResult); 

     AudioFormat audioF = new AudioFormat(
       AudioFormat.Encoding.PCM_SIGNED, 
       samplerate, 
       16, 
       1,   // channels 
       2,   // framesize 
       samplerate, 
       false 
     ); 

     AudioInputStream ais = new AudioInputStream(is, audioF, theSize); 

     try { 
      AudioSystem.write(ais, AudioFileFormat.Type.WAVE, outfile); 
     } catch (IOException ioe) { 
      System.err.println("IO Exception; probably just done with file"); 
      return; 
     } 


    } 

隨着44100作爲採樣率。

考慮採取實際的byte []數組，我已經得到它已經PCM，所以MP3 - > PCM轉換它通過指定

this._decodedFormat = new AudioFormat(AudioFormat.Encoding.PCM_SIGNED, 44100, 16, 2, 1, 44100, false); 
this._audioInputDecodedStream = AudioSystem.getAudioInputStream(this._decodedFormat, this._audioInputStream); 

當我做說，寫入Wav文件時，我聽到很多噪音。我假裝對每一個字節應用一個FFT，但我認爲由於噪聲很大，結果是不正確的。

因爲我拍了兩首歌，其中一首是另一首20秒的作品，當比較作品的fft結果與原始的20秒子集時，它完全不匹配。

我認爲這是不正確的轉換stereo-> mono的原因。

希望有人知道這件事，

問候。

Answer 1

正如在評論中指出的，排序可能是錯誤的。另外，轉換爲有符號的short並將其移位可能會導致第一個字節爲0xFF。

嘗試：

int HI = 0; int LO = 1; 
int left = (buff[i * 4 + HI] << 8) | (buff[i * 4 + LO] & 0xff); 
int right = (buff[i * 4 + 2 + HI] << 8) | (buff[i * 4 + 2 + LO] & 0xff); 
int avg = (left + right)/2; 
mono[i * 2 + HI] = (byte)((avg >> 8) & 0xff); 
mono[i * 2 + LO] = (byte)(avg & 0xff); 

然後切換HI和LO的值，看它是否變得更好。