好的，我整理了一些内容，可以帮助您入门。我将在下面发布完整的代码，但我将首先尝试解释所涉及的步骤。
这里有趣的部分是创建您自己的音频“混音器”类，它允许该类的消费者在（不久的）将来的特定时间点安排音频块。特定时间点部分在这里很重要：我假设您在数据包中接收网络语音，其中每个数据包需要在前一个数据包的末尾准确开始，以便为单个语音播放连续的声音。此外，由于您说声音可以重叠，我假设（是的，很多假设）一个新声音可以通过网络进入，而一个或多个旧声音仍在播放。因此，允许从任何线程调度音频块似乎是合理的。请注意，只有一个线程实际写入数据线，只是任何线程都可以向混音器提交音频包。
所以对于提交音频包部分，我们现在有了这个：
private final ConcurrentLinkedQueue<QueuedBlock> scheduledBlocks;
public void mix(long when, short[] block) {
scheduledBlocks.add(new QueuedBlock(when, Arrays.copyOf(block, block.length)));
}
QueuedBlock 类仅用于用“when”标记字节数组（音频缓冲区）：应该播放块的时间点。
时间点相对于音频流的当前位置表示。每次将音频缓冲区写入数据线时，当创建流并使用缓冲区大小更新流时，它设置为零：
private final AtomicLong position = new AtomicLong();
public long position() {
return position.get();
}
除了设置数据线的所有麻烦之外，混音器类的有趣部分显然是混音发生的地方。对于每个预定的音频块，它分为 3 种情况：
- 该块已完整播放。从 scheduleBlocks 列表中删除。
- 块计划在当前缓冲区之后的某个时间点开始。什么都不做。
- （部分）块应混合到当前缓冲区中。请注意，块的开头可能（或可能不）已在先前的缓冲区中播放。类似地，计划块的结尾可能会超过当前缓冲区的结尾，在这种情况下，我们混合它的第一部分并将其余部分留给下一轮，直到所有的块都被播放完，整个块被删除。
另外请注意，没有可靠的方法可以立即开始播放音频数据，当您将数据包提交到混音器时，请确保始终让它们从现在开始至少 1 个音频缓冲区的持续时间，否则您将面临丢失开头的风险你的声音。这是混音代码：
private static final double MIXDOWN_VOLUME = 1.0 / NUM_PRODUCERS;
private final List<QueuedBlock> finished = new ArrayList<>();
private final short[] mixBuffer = new short[BUFFER_SIZE_FRAMES * CHANNELS];
private final byte[] audioBuffer = new byte[BUFFER_SIZE_FRAMES * CHANNELS * 2];
private final AtomicLong position = new AtomicLong();
Arrays.fill(mixBuffer, (short) 0);
long bufferStartAt = position.get();
for (QueuedBlock block : scheduledBlocks) {
int blockFrames = block.data.length / CHANNELS;
// block fully played - mark for deletion
if (block.when + blockFrames <= bufferStartAt) {
finished.add(block);
continue;
}
// block starts after end of current buffer
if (bufferStartAt + BUFFER_SIZE_FRAMES <= block.when)
continue;
// mix in part of the block which overlaps current buffer
int blockOffset = Math.max(0, (int) (bufferStartAt - block.when));
int blockMaxFrames = blockFrames - blockOffset;
int bufferOffset = Math.max(0, (int) (block.when - bufferStartAt));
int bufferMaxFrames = BUFFER_SIZE_FRAMES - bufferOffset;
for (int f = 0; f < blockMaxFrames && f < bufferMaxFrames; f++)
for (int c = 0; c < CHANNELS; c++) {
int bufferIndex = (bufferOffset + f) * CHANNELS + c;
int blockIndex = (blockOffset + f) * CHANNELS + c;
mixBuffer[bufferIndex] += (short)
(block.data[blockIndex]*MIXDOWN_VOLUME);
}
}
scheduledBlocks.removeAll(finished);
finished.clear();
ByteBuffer
.wrap(audioBuffer)
.order(ByteOrder.LITTLE_ENDIAN)
.asShortBuffer()
.put(mixBuffer);
line.write(audioBuffer, 0, audioBuffer.length);
position.addAndGet(BUFFER_SIZE_FRAMES);
最后是一个完整的、自包含的示例，它生成多个线程，将代表随机持续时间和频率的正弦波的音频块提交给混音器（在此示例中称为 AudioConsumer）。用传入的网络数据包替换正弦波，您应该已经成功了。
package test;
import java.nio.ByteBuffer;
import java.nio.ByteOrder;
import java.util.ArrayList;
import java.util.Arrays;
import java.util.List;
import java.util.concurrent.ConcurrentLinkedQueue;
import java.util.concurrent.ThreadLocalRandom;
import java.util.concurrent.atomic.AtomicBoolean;
import java.util.concurrent.atomic.AtomicLong;
import javax.sound.sampled.AudioFormat;
import javax.sound.sampled.AudioSystem;
import javax.sound.sampled.Line;
import javax.sound.sampled.Mixer;
import javax.sound.sampled.SourceDataLine;
public class Test {
public static final int CHANNELS = 2;
public static final int SAMPLE_RATE = 48000;
public static final int NUM_PRODUCERS = 10;
public static final int BUFFER_SIZE_FRAMES = 4800;
// generates some random sine wave
public static class ToneGenerator {
private static final double[] NOTES = {261.63, 311.13, 392.00};
private static final double[] OCTAVES = {1.0, 2.0, 4.0, 8.0};
private static final double[] LENGTHS = {0.05, 0.25, 1.0, 2.5, 5.0};
private double phase;
private int framesProcessed;
private final double length;
private final double frequency;
public ToneGenerator() {
ThreadLocalRandom rand = ThreadLocalRandom.current();
length = LENGTHS[rand.nextInt(LENGTHS.length)];
frequency = NOTES[rand.nextInt(NOTES.length)] * OCTAVES[rand.nextInt(OCTAVES.length)];
}
// make sound
public void fill(short[] block) {
for (int f = 0; f < block.length / CHANNELS; f++) {
double sample = Math.sin(phase * 2.0 * Math.PI);
for (int c = 0; c < CHANNELS; c++)
block[f * CHANNELS + c] = (short) (sample * Short.MAX_VALUE);
phase += frequency / SAMPLE_RATE;
}
framesProcessed += block.length / CHANNELS;
}
// true if length of tone has been generated
public boolean done() {
return framesProcessed >= length * SAMPLE_RATE;
}
}
// dummy audio producer, based on sinewave generator
// above but could also be incoming network packets
public static class AudioProducer {
final Thread thread;
final AudioConsumer consumer;
final short[] buffer = new short[BUFFER_SIZE_FRAMES * CHANNELS];
public AudioProducer(AudioConsumer consumer) {
this.consumer = consumer;
thread = new Thread(() -> run());
thread.setDaemon(true);
}
public void start() {
thread.start();
}
// repeatedly play random sine and sleep for some time
void run() {
try {
ThreadLocalRandom rand = ThreadLocalRandom.current();
while (true) {
long pos = consumer.position();
ToneGenerator g = new ToneGenerator();
// if we schedule at current buffer position, first part of the tone will be
// missed so have tone start somewhere in the middle of the next buffer
pos += BUFFER_SIZE_FRAMES + rand.nextInt(BUFFER_SIZE_FRAMES);
while (!g.done()) {
g.fill(buffer);
consumer.mix(pos, buffer);
pos += BUFFER_SIZE_FRAMES;
// we can generate audio faster than it's played
// sleep a while to compensate - this more closely
// corresponds to playing audio coming in over the network
double bufferLengthMillis = BUFFER_SIZE_FRAMES * 1000.0 / SAMPLE_RATE;
Thread.sleep((int) (bufferLengthMillis * 0.9));
}
// sleep a while in between tones
Thread.sleep(1000 + rand.nextInt(2000));
}
} catch (Throwable t) {
System.out.println(t.getMessage());
t.printStackTrace();
}
}
}
// audio consumer - plays continuously on a background
// thread, allows audio to be mixed in from arbitrary threads
public static class AudioConsumer {
// audio block with "when to play" tag
private static class QueuedBlock {
final long when;
final short[] data;
public QueuedBlock(long when, short[] data) {
this.when = when;
this.data = data;
}
}
// need not normally be so low but in this example
// we're mixing down a bunch of full scale sinewaves
private static final double MIXDOWN_VOLUME = 1.0 / NUM_PRODUCERS;
private final List<QueuedBlock> finished = new ArrayList<>();
private final short[] mixBuffer = new short[BUFFER_SIZE_FRAMES * CHANNELS];
private final byte[] audioBuffer = new byte[BUFFER_SIZE_FRAMES * CHANNELS * 2];
private final Thread thread;
private final AtomicLong position = new AtomicLong();
private final AtomicBoolean running = new AtomicBoolean(true);
private final ConcurrentLinkedQueue<QueuedBlock> scheduledBlocks = new ConcurrentLinkedQueue<>();
public AudioConsumer() {
thread = new Thread(() -> run());
}
public void start() {
thread.start();
}
public void stop() {
running.set(false);
}
// gets the play cursor. note - this is not accurate and
// must only be used to schedule blocks relative to other blocks
// (e.g., for splitting up continuous sounds into multiple blocks)
public long position() {
return position.get();
}
// put copy of audio block into queue so we don't
// have to worry about caller messing with it afterwards
public void mix(long when, short[] block) {
scheduledBlocks.add(new QueuedBlock(when, Arrays.copyOf(block, block.length)));
}
// better hope mixer 0, line 0 is output
private void run() {
Mixer.Info[] mixerInfo = AudioSystem.getMixerInfo();
try (Mixer mixer = AudioSystem.getMixer(mixerInfo[0])) {
Line.Info[] lineInfo = mixer.getSourceLineInfo();
try (SourceDataLine line = (SourceDataLine) mixer.getLine(lineInfo[0])) {
line.open(new AudioFormat(SAMPLE_RATE, 16, CHANNELS, true, false), BUFFER_SIZE_FRAMES);
line.start();
while (running.get())
processSingleBuffer(line);
line.stop();
}
} catch (Throwable t) {
System.out.println(t.getMessage());
t.printStackTrace();
}
}
// mix down single buffer and offer to the audio device
private void processSingleBuffer(SourceDataLine line) {
Arrays.fill(mixBuffer, (short) 0);
long bufferStartAt = position.get();
// mixdown audio blocks
for (QueuedBlock block : scheduledBlocks) {
int blockFrames = block.data.length / CHANNELS;
// block fully played - mark for deletion
if (block.when + blockFrames <= bufferStartAt) {
finished.add(block);
continue;
}
// block starts after end of current buffer
if (bufferStartAt + BUFFER_SIZE_FRAMES <= block.when)
continue;
// mix in part of the block which overlaps current buffer
// note that block may have already started in the past
// but extends into the current buffer, or that it starts
// in the future but before the end of the current buffer
int blockOffset = Math.max(0, (int) (bufferStartAt - block.when));
int blockMaxFrames = blockFrames - blockOffset;
int bufferOffset = Math.max(0, (int) (block.when - bufferStartAt));
int bufferMaxFrames = BUFFER_SIZE_FRAMES - bufferOffset;
for (int f = 0; f < blockMaxFrames && f < bufferMaxFrames; f++)
for (int c = 0; c < CHANNELS; c++) {
int bufferIndex = (bufferOffset + f) * CHANNELS + c;
int blockIndex = (blockOffset + f) * CHANNELS + c;
mixBuffer[bufferIndex] += (short) (block.data[blockIndex] * MIXDOWN_VOLUME);
}
}
scheduledBlocks.removeAll(finished);
finished.clear();
ByteBuffer.wrap(audioBuffer).order(ByteOrder.LITTLE_ENDIAN).asShortBuffer().put(mixBuffer);
line.write(audioBuffer, 0, audioBuffer.length);
position.addAndGet(BUFFER_SIZE_FRAMES);
}
}
public static void main(String[] args) {
System.out.print("Press return to exit...");
AudioConsumer consumer = new AudioConsumer();
consumer.start();
for (int i = 0; i < NUM_PRODUCERS; i++)
new AudioProducer(consumer).start();
System.console().readLine();
consumer.stop();
}
}