Java：将4个独立的音频字节数组组合成单个wav音频文件

Question

我曾尝试将4个单独的字节数组合并到一个文件中，但我只得到空指针异常，我不知道为什么。我的音频格式是16位PCM签名的，我知道我应该使用short而不是bytes，但老实说，我完全迷路了

private short[] mixByteBuffers(byte[] bufferA, byte[] bufferB) {
    short[] first_array = new short[bufferA.length/2];
    short[] second_array = new short [bufferB.length/2];
    short[] final_array = null;

    if(first_array.length > second_array.length) {
        short[] temp_array = new short[bufferA.length];

        for (int i = 0; i < temp_array.length; i++) {
            int mixed=(int)first_array[i] + (int)second_array[i];
            if (mixed>32767) mixed=32767;
            if (mixed<-32768) mixed=-32768;
            temp_array[i] = (short)mixed;
            final_array = temp_array;
        }
    }
    else {
        short[] temp_array = new short[bufferB.length];

        for (int i = 0; i < temp_array.length; i++) {
            int mixed=(int)first_array[i] + (int)second_array[i];
            if (mixed>32767) mixed=32767;
            if (mixed<-32768) mixed=-32768;
            temp_array[i] = (short)mixed;
            final_array = temp_array;
        }        
    }
    return final_array;
}

这就是我目前正在尝试的，但它在第行返回java.lang.ArrayIndexOutOfBoundsException: 0

int mixed = (int)first_array[i] + (int)second_array[i];

我的数组不都是相同的长度，这就是我调用函数的方式：

public void combineAudio() {
    short[] combinationOne = mixByteBuffers(tempByteArray1, tempByteArray2);
    short[] combinationTwo = mixByteBuffers(tempByteArray3, tempByteArray4);
    short[] channelsCombinedAll = mixShortBuffers(combinationOne, combinationTwo);
    byte[] bytesCombined = new byte[channelsCombinedAll.length * 2];
    ByteBuffer.wrap(bytesCombined).order(ByteOrder.LITTLE_ENDIAN)
        .asShortBuffer().put(channelsCombinedAll);

    mixedByteArray = bytesCombined;
}

一定有比我现在所做的更好的方法，这让我发疯

Answer 1

要将两个byte数组与16位声音样本混合，应首先将这些数组转换为int数组，即基于样本的数组，然后添加它们（以混合），然后再转换回字节数组。从byte数组转换为int数组时，需要确保使用正确的endianness (byte order)

这里有一些代码可以让您混合使用两个数组。最后有一些示例代码（使用正弦波）演示了该方法。请注意，这可能不是对其进行编码的理想方式，而是演示该概念的工作示例。使用流或线，如Phil recommends可能是更明智的总体方法

祝你好运

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

public class MixDemo {

    public static byte[] mix(final byte[] a, final byte[] b, final boolean bigEndian) {
        final byte[] aa;
        final byte[] bb;

        final int length = Math.max(a.length, b.length);
        // ensure same lengths
        if (a.length != b.length) {
            aa = new byte[length];
            bb = new byte[length];
            System.arraycopy(a, 0, aa, 0, a.length);
            System.arraycopy(b, 0, bb, 0, b.length);
        } else {
            aa = a;
            bb = b;
        }

        // convert to samples
        final int[] aSamples = toSamples(aa, bigEndian);
        final int[] bSamples = toSamples(bb, bigEndian);

        // mix by adding
        final int[] mix = new int[aSamples.length];
        for (int i=0; i<mix.length; i++) {
            mix[i] = aSamples[i] + bSamples[i];
            // enforce min and max (may introduce clipping)
            mix[i] = Math.min(Short.MAX_VALUE, mix[i]);
            mix[i] = Math.max(Short.MIN_VALUE, mix[i]);
        }

        // convert back to bytes
        return toBytes(mix, bigEndian);
    }

    private static int[] toSamples(final byte[] byteSamples, final boolean bigEndian) {
        final int bytesPerChannel = 2;
        final int length = byteSamples.length / bytesPerChannel;
        if ((length % 2) != 0) throw new IllegalArgumentException("For 16 bit audio, length must be even: " + length);
        final int[] samples = new int[length];
        for (int sampleNumber = 0; sampleNumber < length; sampleNumber++) {
            final int sampleOffset = sampleNumber * bytesPerChannel;
            final int sample = bigEndian
                    ? byteToIntBigEndian(byteSamples, sampleOffset, bytesPerChannel)
                    : byteToIntLittleEndian(byteSamples, sampleOffset, bytesPerChannel);
            samples[sampleNumber] = sample;
        }
        return samples;
    }

    private static byte[] toBytes(final int[] intSamples, final boolean bigEndian) {
        final int bytesPerChannel = 2;
        final int length = intSamples.length * bytesPerChannel;
        final byte[] bytes = new byte[length];
        for (int sampleNumber = 0; sampleNumber < intSamples.length; sampleNumber++) {
            final byte[] b = bigEndian
                    ? intToByteBigEndian(intSamples[sampleNumber], bytesPerChannel)
                    : intToByteLittleEndian(intSamples[sampleNumber], bytesPerChannel);
            System.arraycopy(b, 0, bytes, sampleNumber * bytesPerChannel, bytesPerChannel);
        }
        return bytes;
    }

    // from https://github.com/hendriks73/jipes/blob/master/src/main/java/com/tagtraum/jipes/audio/AudioSignalSource.java#L238
    private static int byteToIntLittleEndian(final byte[] buf, final int offset, final int bytesPerSample) {
        int sample = 0;
        for (int byteIndex = 0; byteIndex < bytesPerSample; byteIndex++) {
            final int aByte = buf[offset + byteIndex] & 0xff;
            sample += aByte << 8 * (byteIndex);
        }
        return (short)sample;
    }

    // from https://github.com/hendriks73/jipes/blob/master/src/main/java/com/tagtraum/jipes/audio/AudioSignalSource.java#L247
    private static int byteToIntBigEndian(final byte[] buf, final int offset, final int bytesPerSample) {
        int sample = 0;
        for (int byteIndex = 0; byteIndex < bytesPerSample; byteIndex++) {
            final int aByte = buf[offset + byteIndex] & 0xff;
            sample += aByte << (8 * (bytesPerSample - byteIndex - 1));
        }
        return (short)sample;
    }

    private static byte[] intToByteLittleEndian(final int sample, final int bytesPerSample) {
        byte[] buf = new byte[bytesPerSample];
        for (int byteIndex = 0; byteIndex < bytesPerSample; byteIndex++) {
            buf[byteIndex] = (byte)((sample >>> (8 * byteIndex)) & 0xFF);
        }
        return buf;
    }

    private static byte[] intToByteBigEndian(final int sample, final int bytesPerSample) {
        byte[] buf = new byte[bytesPerSample];
        for (int byteIndex = 0; byteIndex < bytesPerSample; byteIndex++) {
            buf[byteIndex] = (byte)((sample >>> (8 * (bytesPerSample - byteIndex - 1))) & 0xFF);
        }
        return buf;
    }

    public static void main(final String[] args) throws IOException {
        final int sampleRate = 44100;
        final boolean bigEndian = true;
        final int sampleSizeInBits = 16;
        final int channels = 1;
        final boolean signed = true;
        final AudioFormat targetAudioFormat = new AudioFormat(sampleRate, sampleSizeInBits, channels, signed, bigEndian);

        final byte[] a = new byte[sampleRate * 10];
        final byte[] b = new byte[sampleRate * 5];

        // create sine waves
        for (int i=0; i<a.length/2; i++) {
            System.arraycopy(intToByteBigEndian((int)(30000*Math.sin(i*0.5)),2), 0, a, i*2, 2);
        }
        for (int i=0; i<b.length/2; i++) {
            System.arraycopy(intToByteBigEndian((int)(30000*Math.sin(i*0.1)),2), 0, b, i*2, 2);
        }

        final File aFile = new File("a.wav");
        AudioSystem.write(new AudioInputStream(new ByteArrayInputStream(a), targetAudioFormat, a.length),
                AudioFileFormat.Type.WAVE, aFile);
        final File bFile = new File("b.wav");
        AudioSystem.write(new AudioInputStream(new ByteArrayInputStream(b), targetAudioFormat, b.length),
                AudioFileFormat.Type.WAVE, bFile);

        // mix a and b
        final byte[] mixed = mix(a, b, bigEndian);
        final File outFile = new File("out.wav");
        AudioSystem.write(new AudioInputStream(new ByteArrayInputStream(mixed), targetAudioFormat, mixed.length),
                AudioFileFormat.Type.WAVE, outFile);
    }
}

Answer 2

else子句for循环中的temp_array.length值为bufferB.length。但是if子句中的值是bufferA.length/2。你忽略了在else子句中除以2了吗

不管怎样，通常只将音频数据（信号）处理为流。在每一行打开时，从每一行获取预定义缓冲区的字节值，足以从每一行获得相同数量的PCM值。如果一行在其他行之前用完，可以用0值填充该行

除非有足够的理由添加长度不等的数组，否则我认为最好避免这样做。相反，使用指针（如果您是从数组中绘制）或渐进式读取（）方法（如果是从音频输入行）来获取每次循环迭代的固定数量的PCM值。否则，我认为你是在自找麻烦，不必要地使事情复杂化

我见过一些可行的解决方案，每次只从每个源处理一个PCM值，甚至更多，比如1000甚至半秒（如果是44100 fps，则为22050）。最主要的是在每次迭代中从每个源获得相同数量的PCM，如果一个源的数据用完，则用0填充