我试图做一些音频处理,我真的坚持立体声单声道转换。我在互联网上查看有关立体声转换为单声道。转换音频立体声为音频字节
据我所知,我可以把左声道,右声道,总和它们除以2.但是当我把结果再次转储到WAV文件中时,我得到了很多前景噪声。我知道处理数据时可能会引起噪声,字节变量中会出现一些溢出。
这是从一个MP3文件中检索字节[]数据块我的课:
公共类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;
}
}
这意味着,每次我打电话getSamples时间,它返回一个数组,如:
buff = {Lchannel,Rchannel,Lchannel,Rchannel,Lchannel,Rchannel,Lchannel,Rchannel ...}
的处理例行程序的转换到单声道的样子:
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);
}
}
和写入到使用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;
}
}
随着44100作为采样率。
考虑采取实际的byte []数组,我已经得到它已经PCM,所以MP3 - > PCM转换它通过指定
this._decodedFormat = new AudioFormat(AudioFormat.Encoding.PCM_SIGNED, 44100, 16, 2, 1, 44100, false); this._audioInputDecodedStream = AudioSystem.getAudioInputStream(this._decodedFormat, this._audioInputStream);
当我做说,写入Wav文件时,我听到很多噪音。我假装对每一个字节应用一个FFT,但我认为由于噪声很大,结果是不正确的。
因为我拍了两首歌,其中一首是另一首20秒的作品,当比较作品的fft结果与原始的20秒子集时,它完全不匹配。
我认为这是不正确的转换stereo-> mono的原因。
希望有人知道这件事,
问候。
如果是由溢出引起的,为什么不除以2然后求和? – James 2013-05-09 16:26:55
您可能会错误地获取数据的字节序。试着做一些像没有转换的读写操作,或者更好的办法是通过一个已知的干净的数据源(也许是一个只使用2个不同振幅值的方波)并检查输出的原始字节。有了一点经验,如果音频软件中的信号图表可能会很快被识别出来。 – 2013-05-09 16:29:05
如果我不转换,所有我从一个MP3文件它是原始编码字节。转换它不是一个可选的步骤,它必须完成才能将真实的声音值输入到数组中。 划分和求和有相同的结果... – Mario 2013-05-09 16:33:43