如何解决混合两个 16 位编码立体声 PCM 样本，导致生成的音频出现噪音和失真
我从两个来源获得两个不同的音频样本。
-
对于麦克风声音：
audioRecord = new AudioRecord(MediaRecorder.AudioSource.DEFAULT,44100,AudioFormat.CHANNEL_IN_STEREO,AudioFormat.ENCODING_PCM_16BIT,(AudioRecord.getMinBufferSize(44100,AudioFormat.ENCODING_PCM_16BIT)*5)); -
对于内部声音：
audioRecord = new AudioRecord.Builder() .setAudioPlaybackCaptureConfig(config) .setAudioFormat(new AudioFormat.Builder() .setEncoding(AudioFormat.ENCODING_PCM_16BIT) .setSampleRate(44100) .setChannelMask(AudioFormat.CHANNEL_IN_STEREO) .build()) .setBufferSizeInBytes((AudioRecord.getMinBufferSize(44100,AudioFormat.ENCODING_PCM_16BIT)*5)) .build();
为了从 audioRecord 对象中读取，我们创建了单独的框架对象（称为框架的自定义对象）-
private ByteBuffer pcmBuffer = ByteBuffer.allocateDirect(4096);
private Frame read() {
pcmBuffer.rewind();
int size = audioRecord.read(pcmBuffer,pcmBuffer.remaining());
if (size <= 0) {
return null;
}
return new Frame(pcmBuffer.array(),pcmBuffer.arrayOffset(),size);
}
我们创建了两个单独的 LL（Linked List）来添加我们从 read 函数中获得的这些帧。
private LinkedList internalAudioQueue = new LinkedList(); private LinkedListmicAudioQueue = new LinkedList();
public void onFrameReceived(Frame frame,boolean isInternalAudio) {
if (isInternalAudio) {
internalAudioQueue.add(frame);
} else {
microphoneAudioQueue.add(frame);
}
checkAndPoll();
}
每次我们在相应的 LL 中添加一个帧时，我们都会调用以下 checkAndPoll() 函数，并根据情况将帧传递给 audioEncoder。
public void checkAndPoll() {
Frame frame1 = internalAudioQueue.poll();
Frame frame2 = microphoneAudioQueue.poll();
if (frame1 == null && frame2 != null) {
audioEncoder.inputPCMData(frame2);
} else if (frame1 != null && frame2 == null) {
audioEncoder.inputPCMData(frame1);
} else if (frame1 != null && frame2 != null) {
Frame frame = new Frame(PCMUtil.mix(frame1.getBuffer(),frame2.getBuffer(),frame1.getSize(),frame2.getSize(),false),frame1.getOrientation(),frame1.getSize());
audioEncoder.inputPCMData(frame);
}
}
现在我们在 Hendrik 的帮助下以这种方式混合来自两个源的 ByteBuffer 形式的音频样本。
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,aa,a.length);
System.arraycopy(b,bb,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,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,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 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;
}
我得到的混合样本既有失真又有噪声。无法弄清楚需要做什么才能删除它。任何帮助在这里表示赞赏。 提前致谢！
解决方法
我认为如果你要混合，你应该取两者的（加权）平均值。
如果您有样本 128 和 128，那么结果仍然是 128，而不是可能超出范围的 256。
所以只需将您的代码更改为：
// mix by adding
final int[] mix = new int[aSamples.length];
for (int i=0; i<mix.length; i++) {
// calculating the average
mix[i] = (aSamples[i] + bSamples[i]) >> 1;
}
这对你有用吗？
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