本文介绍了在将声音写入文件之前，如何在Java中不使用“剪辑"的情况下循环播放声音的处理方法，对大家解决问题具有一定的参考价值，需要的朋友们下面随着小编来一起学习吧！
问题描述
首先，感谢大家帮助我解决之前的问题.
First of all thanks everyone for helping me with my previous questions.
在下面的代码中，我交替使用两个频率，并将它们写入.wav格式，以便在Windows Media Player中在用户指定的时间段内运行它.我要了解的是如何使这些频率循环运行，以便在指定的时间交替运行，例如救护车的警笛声，而在我的程序中，这两种频率只能交替播放一次.例如，如果我将时间指定为10秒，则两个频率都将连续运行5秒.但是我想要的是第一个频率运行一秒钟或两秒钟(根据用户指定)，然后第二个频率运行相同的秒数，然后再次运行第一个频率，它应该一直持续到指定的时间
In the following code I am taking two frequencies alternatively and writing them into a .wav format, to run it in my Windows Media Player, for a specific time given by the user.What I want is to understand how to loop those frequencies to run alternatively for that specified time, like the Ambulance's siren, and in my program both frequencies are getting played just once, alternatively.For example, if I am specifying the time as 10 seconds, then both the frequencies run for 5 seconds each at a stretch. But what I want is that the 1st frequency to run for a second or two seconds(as the user specifies) and then the 2nd frequency to run for that similar second and then again the 1st frequency and it should keep on going till the specified time.
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.因此，请多多包涵，因为这是我学习的唯一途径.谢谢.
And as I am quite new in Java and JavaSounds, therefore sometimes I might ask few foolish or irrelevant questions. So please bear with me as that is the only way in which I can learn.Thanks.
推荐答案
我解决了它.尽管我需要处理一小部分.完成后，我会立即对其进行更新.
I resolved it. Although there is one minor part that I need to handle. I will update it as soon as I am done with it.
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 milliseconds
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 * SAMPLING_RATE))/1000]; //Size of buffer[], which in case of 10 seconds is 441000
float buffer1[] = new float[((int) (time * SAMPLING_RATE))/1000]; //Size of buffer1[], which in case of 10 seconds is 441000
//for (int a = 1; a <= time2 / 2; a++) {
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);
}
int iterations = (1000*time2)/(2*time);
byte[] merge = new byte[iterations*(byteBuffer.length + byteBuffer1.length)];
for (int i = 0; i<iterations; i++)
{
//arraycopy(Object src, int srcPos, Object dest, int destPos, int length)
//For merging the two frequencies
System.arraycopy(byteBuffer, 0, merge, 0+i*(byteBuffer.length + byteBuffer1.length), byteBuffer.length);
System.arraycopy(byteBuffer1, 0, merge, byteBuffer.length+i*(byteBuffer.length + byteBuffer1.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) * (time2/4));
//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
}
}
这篇关于在将声音写入文件之前，如何在Java中不使用“剪辑"的情况下循环播放声音的文章就介绍到这了，希望我们推荐的答案对大家有所帮助，也希望大家多多支持！