PCM Wave 文件 - 立体声到单声道

Question

我有一个立体声音频文件。将其转换为单声道只是跳过每隔一个字节（在标头之后）的情况吗？它以 16 位签名 PCM 格式编码。我有 javax.sound.sampled 可用。

以下是我尝试过但不起作用的代码：

WaveFileWriter wfw = new WaveFileWriter();
AudioFormat format = new AudioFormat(Encoding.PCM_SIGNED, 44100, 16, 2, 2, 44100, false);
AudioFormat monoFormat = new AudioFormat(Encoding.PCM_SIGNED, 44100, 16, 1, 2, 44100, false);

byte[] audioData = dataout.toByteArray();
int length = audioData.length;
ByteArrayInputStream bais = new ByteArrayInputStream(audioData);

AudioInputStream stereoStream = new AudioInputStream(bais,format,length);
AudioInputStream monoStream = new AudioInputStream(stereoStream,format,length/2);

wfw.write(monoStream, Type.WAVE, new File(Environment.
                 getExternalStorageDirectory().getAbsolutePath()+"/stegDroid/un-ogged.wav"));

此代码在使用 Jorbis 读取 .ogg 文件后使用，将其转换为 PCM 数据。唯一的问题是结果是立体声，而我需要它是单声道，所以如果有其他解决方案，我很高兴听到它！

Answer 1

我有一个立体声音频文件。
将其转换为单声道只是一个案例
跳过每隔一个字节（之后
标题）？

几乎 - 你想跳过所有其他样本，而不是字节。在您的情况下，每个样本的大小似乎为 16 位 = 2 字节。因此，您可能想要获取 2 个字节、跳过 2 个字节、获取 2 个字节等等。

AudioInputStream monoStream = new AudioInputStream(stereoStream,format,length/2);

wfw.write(monoStream, Type.WAVE, new File(Environment.getExternalStorageDirectory().getAbsolutePath()+"/stegDroid/un-ogged.wav"));

看起来您只是写出了文件的前半部分，而不是写出了所有其他示例。此外，您还必须修复 WAV 标头，以指定单个通道（请参阅您的 monoFormat）。

Answer 2

看看这段代码。当我需要弄乱 wav 文件中的字节时，它对我很有帮助。

package GlobalUtilities;

import java.applet.Applet;
import java.applet.AudioClip;
import java.net.URISyntaxException;
import java.util.logging.Level;
import java.util.logging.Logger;
import java.io.*;
import java.io.File;
import java.net.MalformedURLException;
import java.net.URL;
import javax.sound.sampled.*;

/**
 * This class handles the reading, writing, and playing of wav files. It is
 * also capable of converting the file to its raw byte [] form.
 *
 * based on code by Evan Merz modified by Dan Vargo
 * @author dvargo
 */
public class Wav {
    /*
    WAV File Specification
    FROM http://ccrma.stanford.edu/courses/422/projects/WaveFormat/
    The canonical WAVE format starts with the RIFF header:
    0         4   ChunkID          Contains the letters "RIFF" in ASCII form
    (0x52494646 big-endian form).
    4         4   ChunkSize        36 + SubChunk2Size, or more precisely:
    4 + (8 + SubChunk1Size) + (8 + SubChunk2Size)
    This is the size of the rest of the chunk
    following this number.  This is the size of the
    entire file in bytes minus 8 bytes for the
    two fields not included in this count:
    ChunkID and ChunkSize.
    8         4   Format           Contains the letters "WAVE"
    (0x57415645 big-endian form).

    The "WAVE" format consists of two subchunks: "fmt " and "data":
    The "fmt " subchunk describes the sound data's format:
    12        4   Subchunk1ID      Contains the letters "fmt "
    (0x666d7420 big-endian form).
    16        4   Subchunk1Size    16 for PCM.  This is the size of the
    rest of the Subchunk which follows this number.
    20        2   AudioFormat      PCM = 1 (i.e. Linear quantization)
    Values other than 1 indicate some
    form of compression.
    22        2   NumChannels      Mono = 1, Stereo = 2, etc.
    24        4   SampleRate       8000, 44100, etc.
    28        4   ByteRate         == SampleRate * NumChannels * BitsPerSample/8
    32        2   BlockAlign       == NumChannels * BitsPerSample/8
    The number of bytes for one sample including
    all channels. I wonder what happens when
    this number isn't an integer?
    34        2   BitsPerSample    8 bits = 8, 16 bits = 16, etc.

    The "data" subchunk contains the size of the data and the actual sound:
    36        4   Subchunk2ID      Contains the letters "data"
    (0x64617461 big-endian form).
    40        4   Subchunk2Size    == NumSamples * NumChannels * BitsPerSample/8
    This is the number of bytes in the data.
    You can also think of this as the size
    of the read of the subchunk following this
    number.
    44        *   Data             The actual sound data.


    The thing that makes reading wav files tricky is that java has no unsigned types.  This means that the
    binary data can't just be read and cast appropriately.  Also, we have to use larger types
    than are normally necessary.

    In many languages including java, an integer is represented by 4 bytes.  The issue here is
    that in most languages, integers can be signed or unsigned, and in wav files the  integers
    are unsigned.  So, to make sure that we can store the proper values, we have to use longs
    to hold integers, and integers to hold shorts.

    Then, we have to convert back when we want to save our wav data.

    It's complicated, but ultimately, it just results in a few extra functions at the bottom of
    this file.  Once you understand the issue, there is no reason to pay any more attention
    to it.

    ALSO:

    This code won't read ALL wav files.  This does not use to full specification.  It just uses
    a trimmed down version that most wav files adhere to.

     */

    ByteArrayOutputStream byteArrayOutputStream;
    AudioFormat audioFormat;
    TargetDataLine targetDataLine;
    AudioInputStream audioInputStream;
    SourceDataLine sourceDataLine;
    float frequency = 8000.0F;  //8000,11025,16000,22050,44100
    int samplesize = 16;
    private String myPath;
    private long myChunkSize;
    private long mySubChunk1Size;
    private int myFormat;
    private long myChannels;
    private long mySampleRate;
    private long myByteRate;
    private int myBlockAlign;
    private int myBitsPerSample;
    private long myDataSize;
    // I made this public so that you can toss whatever you want in here
    // maybe a recorded buffer, maybe just whatever you want
    public byte[] myData;



    public Wav()
    {
        myPath = "";
    }

    // constructor takes a wav path
    public Wav(String tmpPath) {
        myPath = tmpPath;
    }


    // get/set for the Path property
    public String getPath()
    {
        return myPath;
    }

    public void setPath(String newPath)
    {
        myPath = newPath;
    }

    // read a wav file into this class
    public boolean read() {
        DataInputStream inFile = null;
        myData = null;
        byte[] tmpLong = new byte[4];
        byte[] tmpInt = new byte[2];

        try {
            inFile = new DataInputStream(new FileInputStream(myPath));

            //System.out.println("Reading wav file...\n"); // for debugging only

            String chunkID = "" + (char) inFile.readByte() + (char) inFile.readByte() + (char) inFile.readByte() + (char) inFile.readByte();

            inFile.read(tmpLong); // read the ChunkSize
            myChunkSize = byteArrayToLong(tmpLong);

            String format = "" + (char) inFile.readByte() + (char) inFile.readByte() + (char) inFile.readByte() + (char) inFile.readByte();

            // print what we've read so far
            //System.out.println("chunkID:" + chunkID + " chunk1Size:" + myChunkSize + " format:" + format); // for debugging only



            String subChunk1ID = "" + (char) inFile.readByte() + (char) inFile.readByte() + (char) inFile.readByte() + (char) inFile.readByte();

            inFile.read(tmpLong); // read the SubChunk1Size
            mySubChunk1Size = byteArrayToLong(tmpLong);

            inFile.read(tmpInt); // read the audio format.  This should be 1 for PCM
            myFormat = byteArrayToInt(tmpInt);

            inFile.read(tmpInt); // read the # of channels (1 or 2)
            myChannels = byteArrayToInt(tmpInt);

            inFile.read(tmpLong); // read the samplerate
            mySampleRate = byteArrayToLong(tmpLong);

            inFile.read(tmpLong); // read the byterate
            myByteRate = byteArrayToLong(tmpLong);

            inFile.read(tmpInt); // read the blockalign
            myBlockAlign = byteArrayToInt(tmpInt);

            inFile.read(tmpInt); // read the bitspersample
            myBitsPerSample = byteArrayToInt(tmpInt);

            // print what we've read so far
            //System.out.println("SubChunk1ID:" + subChunk1ID + " SubChunk1Size:" + mySubChunk1Size + " AudioFormat:" + myFormat + " Channels:" + myChannels + " SampleRate:" + mySampleRate);


            // read the data chunk header - reading this IS necessary, because not all wav files will have the data chunk here - for now, we're just assuming that the data chunk is here
            String dataChunkID = "" + (char) inFile.readByte() + (char) inFile.readByte() + (char) inFile.readByte() + (char) inFile.readByte();

            inFile.read(tmpLong); // read the size of the data
            myDataSize = byteArrayToLong(tmpLong);


            // read the data chunk
            myData = new byte[(int) myDataSize];
            inFile.read(myData);

            // close the input stream
            inFile.close();
        } catch (Exception e) {
            return false;
        }

        return true; // this should probably be something more descriptive
    }

    // write out the wav file
    public boolean save() {
        try {
            DataOutputStream outFile = new DataOutputStream(new FileOutputStream(myPath + "temp"));

            // write the wav file per the wav file format
            outFile.writeBytes("RIFF");                 // 00 - RIFF
            outFile.write(intToByteArray((int) myChunkSize), 0, 4);     // 04 - how big is the rest of this file?
            outFile.writeBytes("WAVE");                 // 08 - WAVE
            outFile.writeBytes("fmt ");                 // 12 - fmt
            outFile.write(intToByteArray((int) mySubChunk1Size), 0, 4); // 16 - size of this chunk
            outFile.write(shortToByteArray((short) myFormat), 0, 2);        // 20 - what is the audio format? 1 for PCM = Pulse Code Modulation
            outFile.write(shortToByteArray((short) myChannels), 0, 2);  // 22 - mono or stereo? 1 or 2?  (or 5 or ???)
            outFile.write(intToByteArray((int) mySampleRate), 0, 4);        // 24 - samples per second (numbers per second)
            outFile.write(intToByteArray((int) myByteRate), 0, 4);      // 28 - bytes per second
            outFile.write(shortToByteArray((short) myBlockAlign), 0, 2);    // 32 - # of bytes in one sample, for all channels
            outFile.write(shortToByteArray((short) myBitsPerSample), 0, 2); // 34 - how many bits in a sample(number)?  usually 16 or 24
            outFile.writeBytes("data");                 // 36 - data
            outFile.write(intToByteArray((int) myDataSize), 0, 4);      // 40 - how big is this data chunk
            outFile.write(myData);                      // 44 - the actual data itself - just a long string of numbers
        } catch (Exception e) {
            System.out.println(e.getMessage());
            return false;
        }

        return true;
    }

    // return a printable summary of the wav file
    public String getSummary() {
        //String newline = System.getProperty("line.separator");
        String newline = "
";
        String summary = "Format: " + myFormat + newline + "Channels: " + myChannels + newline + "SampleRate: " + mySampleRate + newline + "ByteRate: " + myByteRate + newline + "BlockAlign: " + myBlockAlign + newline + "BitsPerSample: " + myBitsPerSample + newline + "DataSize: " + myDataSize + "";
        return summary;
    }

    public byte[] getBytes() {
        read();
        return myData;
    }



    /**
     * Plays back audio stored in the byte array using an audio format given by
     * freq, sample rate, ect.
     * @param data The byte array to play
     */
    public void playAudio(byte[] data) {
        try {
            byte audioData[] = data;
            //Get an input stream on the byte array containing the data
            InputStream byteArrayInputStream = new ByteArrayInputStream(audioData);
            AudioFormat audioFormat = getAudioFormat();
            audioInputStream = new AudioInputStream(byteArrayInputStream, audioFormat, audioData.length / audioFormat.getFrameSize());
            DataLine.Info dataLineInfo = new DataLine.Info(SourceDataLine.class, audioFormat);
            sourceDataLine = (SourceDataLine) AudioSystem.getLine(dataLineInfo);
            sourceDataLine.open(audioFormat);
            sourceDataLine.start();

            //Create a thread to play back the data and start it running.  It will run \
            //until all the data has been played back.
            Thread playThread = new Thread(new PlayThread());
            playThread.start();
        } catch (Exception e) {
            System.out.println(e);
        }
    }

    /**
     * This method creates and returns an AudioFormat object for a given set
     * of format parameters.  If these parameters don't work well for
     * you, try some of the other allowable parameter values, which
     * are shown in comments following the declarations.
     * @return
     */
    private AudioFormat getAudioFormat() {
        float sampleRate = frequency;
        //8000,11025,16000,22050,44100
        int sampleSizeInBits = samplesize;
        //8,16
        int channels = 1;
        //1,2
        boolean signed = true;
        //true,false
        boolean bigEndian = false;
        //true,false
        //return new AudioFormat( AudioFormat.Encoding.PCM_SIGNED, 8000.0f, 8, 1, 1,
        //8000.0f, false );

        return new AudioFormat(sampleRate, sampleSizeInBits, channels, signed, bigEndian);
    }

    public void playWav(String filePath) {
        try {
            AudioClip clip = (AudioClip) Applet.newAudioClip(new File(filePath).toURI().toURL());
            clip.play();
        } catch (Exception e) {
            Logger.getLogger(Wav.class.getName()).log(Level.SEVERE, null, e);
        }

    }

// ===========================
// CONVERT BYTES TO JAVA TYPES
// ===========================
    // these two routines convert a byte array to a unsigned short
    public static int byteArrayToInt(byte[] b) {
        int start = 0;
        int low = b[start] & 0xff;
        int high = b[start + 1] & 0xff;
        return (int) (high > 8) & 0x000000FF);
        b[2] = (byte) ((i >> 16) & 0x000000FF);
        b[3] = (byte) ((i >> 24) & 0x000000FF);
        return b;
    }

    // convert a short to a byte array
    public static byte[] shortToByteArray(short data) {
        return new byte[]{(byte) (data & 0xff), (byte) ((data >>> 8) & 0xff)};
    }

    /**
     * Inner class to play back the data that was saved
     */
    class PlayThread extends Thread {

        byte tempBuffer[] = new byte[10000];

        public void run() {
            try {
                int cnt;
                //Keep looping until the input
                // read method returns -1 for
                // empty stream.
                while ((cnt = audioInputStream.read(tempBuffer, 0, tempBuffer.length)) != -1) {
                    if (cnt > 0) {
                        //Write data to the internal
                        // buffer of the data line
                        // where it will be delivered
                        // to the speaker.
                        sourceDataLine.write(tempBuffer, 0, cnt);
                    }
                }
                //Block and wait for internal
                // buffer of the data line to
                // empty.
                sourceDataLine.drain();
                sourceDataLine.close();
            } catch (Exception e) {
                System.out.println(e);
                System.exit(0);
            }
        }
    }
}

Answer 3

2018年回答这个问题。我也有类似的情况，并意识到自己犯了一个明显的错误。构造函数参数中的“格式”参数不正确。

    AudioFormat format = new AudioFormat(Encoding.PCM_SIGNED, 44100, 16, 2, 2, 44100, 
false);

第五个参数（在您的情况下，第二个“2”）表示帧大小。帧大小 = 样本大小 * 通道数。因为您的位深度是 16，所以您的样本大小是 2 个字节。

样本大小 = 2

通道 = 2

帧大小 = 样本大小 * 通道 = 4

因此，您的代码行应该为：

    AudioFormat format = new AudioFormat(Encoding.PCM_SIGNED, 44100, 16, 2, 4, 44100, 
false);

另外，您是否尝试过使用 FormatConversionProvider？

    javax.sound.sampled.spi.FormatConversionProvider

https://docs.oracle.com/javase/tutorial/sound/converters。 html
本教程对我有很大帮助，但我相信它假设您已经导入了上述类。

我没有看到此线程上发布的这些解决方案，但也许您已经弄清楚了。无论如何，希望这有帮助！