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我正在尝试进行一些音频处理，我真的被立体声到单声道转换所困扰。我在互联网上查看了有关立体声到单声道转换的信息。

据我所知，我可以取左声道、右声道，将它们相加并除以 2。但是当我再次将结果转储到 WAV 文件中时，我得到了很多前景噪音。我知道在处理数据时可能会引起噪音，字节变量中有一些溢出。

这是我从 MP3 文件中检索 byte[] 数据块的课程：

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，每次我调用 getSamples 时，它都会返回一个数组，如下所示：

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

转换为单声道的处理例程如下所示：

    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);
            }

}

并使用以下命令写入 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;
        }


    }

以 44100 作为采样率。

请记住，实际上我得到的 byte[] 数组已经是 pcm，所以 mp3 -> pcm 转换是通过指定来完成的

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

正如我所说，在写入 Wav 文件时，我遇到了很多噪音。我假装将 FFT 应用于每个字节块，但我认为由于嘈杂的声音，结果是不正确的。

因为我正在拍摄两首歌曲，其中一首是 20 秒的裁剪，而当将裁剪 fft 结果与原始的 20 秒子集进行比较时，它根本不匹配。

我认为这是不正确的转换立体声->单声道的原因。

希望有人对此有所了解，

问候。

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1 回答 1

7

正如评论中指出的那样，字节序可能是错误的。此外，转换为有符号的 short 并对其进行移位可能会导致第一个字节为 0xFF。

尝试：

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);

然后切换 HI 和 LO 的值，看看它是否变得更好。

于 2013-05-09T19:04:07.850 回答