audio - 如何查找二维字节数组的长度以及如何在 ByteArrayInputStream 中使用该二维字节数组

Question

首先，在下面的代码中，我试图做的是使用“byteBuffer[0].length”找到二维字节数组的长度，但它实际上不起作用。当我打印 'byteBuffer[0].length' 时，它给出的输出为 4 而不是 882000，根据我传递的参数，(后者)应该是正确的输出。那么如何在我的循环中迭代它呢？

其次，我想在“ByteArrayInputStream”中传递“byteBuffer”，但在“ByteArrayInputStream”中我们不能传递二维数组。那么有没有办法附加值并在那里使用它？而且我还需要交替传递“Frequency1”和“Frequency2”的值并将它们保存为 .wav 格式，以便我可以在我的媒体播放器中相应地播放它们。例如:救护车的警报器。

import java.io.ByteArrayInputStream;
import java.io.ByteArrayOutputStream;
import java.io.File;
import java.io.IOException;
import java.util.Scanner;
import javax.sound.sampled.AudioFileFormat;
import javax.sound.sampled.AudioFormat;
import javax.sound.sampled.AudioInputStream;
import javax.sound.sampled.AudioSystem;

public class AudioPlay {

public static void main(String[] args) throws IOException {

    Scanner in = new Scanner(System.in);
    final double SAMPLING_RATE = 44100;             // Audio sampling rate
    int time = in.nextInt();                        //Time specified by user in seconds
    int frequency1 = in.nextInt();                  //Frequency specified by the user in hz
    int frequency2 = in.nextInt();

    //Size of buffer, in case time is 10 seconds it will be [2][441000]
    float buffer[][] = new float[2][(int) (time * SAMPLING_RATE)];   

    for (int sample = 0; sample < buffer[0].length; sample++) {
        double cycle = sample / SAMPLING_RATE;                  //Fraction of cycle between samples
        buffer[0][sample] = (float) (Math.sin(2 * Math.PI * frequency1 * cycle));   //Storing value at every index of 1st row
        buffer[1][sample] = (float) (Math.sin(2 * Math.PI * frequency2 * cycle));   //Storing value at every index of 2nd row
    }

    //Size of byteBuffer, in case time is 10sec it will be [2][882000]
    byte byteBuffer[][] = new byte[2][(int)(buffer.length * 2)];
    System.out.println(byteBuffer[0].length);      // Giving wrong output

    int count = 0;

    for (int j = 0; j < byteBuffer.length; j++) {
        for (int i = 0; i < byteBuffer[0].length; i++) {
            final int x = (int) ((buffer[j][count++]) * Short.MAX_VALUE);
            byteBuffer[j][i++] = (byte) x;
            byteBuffer[j][i] = (byte) (x / 256);       //Total Value of Byte
        }
    }
    File out = new File("E:/RecordAudio7.wav"); //The path where user want the file data to be written

    //Construct an audio format, using 44100hz sampling rate, 16 bit samples, mono, and big 
    // endian byte ordering
    AudioFormat format = new AudioFormat((float) SAMPLING_RATE, 16, 1, true, false);

    // It uses bytebuffer as its buffer array that contains bytes that may be read from the stream.
    ByteArrayInputStream bais = new ByteArrayInputStream(byteBuffer[0]);

    //Constructs an audio input stream that has the requested format and length in sample frames, using audio data 
    //from the specified input stream.
    AudioInputStream audioInputStream = new AudioInputStream(bais, format, buffer.length);

    //Writes a stream of bytes representing an audio file of the specified file type to the external file provided.
    AudioSystem.write(audioInputStream, AudioFileFormat.Type.WAVE, out);

    audioInputStream.close();       //Closes this audio input stream
}

Answer 1

我找到了问题第一部分的答案。代码只有一个小改动:

byte byteBuffer[][] = new byte[2][(int)(buffer[0].length * 2)];

代替

 byte byteBuffer[][] = new byte[2][(int)(buffer.length * 2)];