首先感谢大家帮助我解决我之前的问题。
在下面的代码中，我交替采用两个频率并将它们写入 .wav 格式，以便在我的 Windows Media Player 中运行它，时间由用户指定。我想要了解如何循环这些频率以在指定的时间内交替运行，例如救护车的警报器，并且在我的程序中，两个频率只播放一次，或者。例如，如果我将时间指定为 10 秒，那么两个频率都将分别运行 5 秒。但我想要的是第一个频率运行一秒或两秒（如用户指定），然后第二个频率运行类似的秒，然后再次运行第一个频率，它应该一直持续到指定的时间.
import java.io.ByteArrayInputStream;
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;
import javax.sound.sampled.LineUnavailableException;
public class AudioWrite2New {
public static void main(String[] args) throws IOException, InterruptedException, LineUnavailableException {
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 time2 = in.nextByte();
int frequency1 = in.nextInt(); //Frequency1 specified by the user in hz
int frequency2 = in.nextInt(); //Frequency2 specified by the user in hz
float buffer[] = new float[(int) (time/2 * SAMPLING_RATE)]; //Size of buffer[], which in case of 10 seconds is 441000
float buffer1[] = new float[(int) (time/2 * SAMPLING_RATE)]; //Size of buffer1[], which in case of 10 seconds is 441000
for (int sample = 0; sample < buffer.length; sample++) {
double cycle = sample / SAMPLING_RATE; //Fraction of cycle between samples
buffer[sample] = (float) (Math.sin(2 * Math.PI * frequency1 * cycle));
//buffer1[sample] = (float) (Math.sin(2 * Math.PI * frequency2 * cycle));
}
for (int sample = 0; sample < buffer1.length; sample++) {
double cycle = sample / SAMPLING_RATE; //Fraction of cycle between samples
//buffer[sample] = (float) (Math.sin(2 * Math.PI * frequency1 * cycle));
buffer1[sample] = (float) (Math.sin(2 * Math.PI * frequency2 * cycle));
}
//System.out.println(buffer[1]);
byte byteBuffer[] = new byte[buffer.length * 2]; //Size of byteBuffer, in this case 882000
byte byteBuffer1[] = new byte[buffer1.length * 2]; //Size of byteBuffer, in this case 882000
int count = 0;
for (int i = 0; i < byteBuffer.length; i++) {
final int x = (int) (buffer[count++] * Short.MAX_VALUE);
byteBuffer[i++] = (byte) x;
byteBuffer[i] = (byte) (x / 256);
}
count = 0;
for (int i = 0; i < byteBuffer1.length; i++) {
final int x = (int) (buffer1[count++] * Short.MAX_VALUE);
byteBuffer1[i++] = (byte) x;
byteBuffer1[i] = (byte) (x / 256);
}
//For merging the two frequencies
byte[] merge = new byte[byteBuffer.length + byteBuffer1.length];
System.arraycopy(byteBuffer, 0, merge, 0, byteBuffer.length);
System.arraycopy(byteBuffer1, 0, merge, byteBuffer.length, byteBuffer1.length);
File out = new File("E:/RecordAudio17.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(merge);
//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, buffer1.length + 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
}
}
由于我是 Java 和 JavaSounds 的新手，因此有时我可能会问一些愚蠢或不相关的问题。所以请耐心等待，因为这是我可以学习的唯一方法。谢谢。