Allright, I put something together which should get you started. I'll post the full code below but I'll first try and explain the steps involved.
The interesting part here is to create you're own audio "mixer" class which allows consumers of that class to schedule audio blocks at specific points in the (near) future. The specific-point-in-time part is important here: i'm assuming you receive network voices in packets where each packet needs to start exactly at the end of the previous one in order to play back a continuous sound for a single voice. Also since you say voices can overlap I'm assuming (yes, lots of assumptions) a new one can come in over the network while one or more old ones are still playing. So it seems reasonable to allow audio blocks to be scheduled from any thread. Note that there's only one thread actually writing to the dataline, it's just that any thread can submit audio packets to the mixer.
So for the submit-audio-packet part we now have this:
private final ConcurrentLinkedQueue<QueuedBlock> scheduledBlocks;
public void mix(long when, short[] block) {
scheduledBlocks.add(new QueuedBlock(when, Arrays.copyOf(block, block.length)));
}
The QueuedBlock class is just used to tag a byte array (the audio buffer) with the "when": the point in time where the block should be played.
Points in time are expressed relative to the current position of the audio stream. It is set to zero when the stream is created and updated with the buffer size each time an audio buffer is written to the dataline:
private final AtomicLong position = new AtomicLong();
public long position() {
return position.get();
}
Apart from all the hassle to set up the data line, the interesting part of the mixer class is obviously where the mixdown happens. For each scheduled audio block, it's split up into 3 cases:
- The block is already played in it's entirety. Remove from the scheduledBlocks list.
- The block is scheduled to start at some point in time after the current buffer. Do nothing.
- (Part of) the block should be mixed down into the current buffer. Note that the beginning of the block may (or may not) be already played in previous buffer(s). Similarly, the end of the scheduled block may exceed the end of the current buffer in which case we mix down the first part of it and leave the rest for the next round, untill all of it has been played an the entire block is removed.
Also note that there's no reliable way to start playing audio data immediately, when you submit packets to the mixer be sure to always have them start at least the duration of 1 audio buffer from now otherwise you'll risk losing the beginning of your sound. Here's the mixdown code:
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);
And finally a complete, self-contained sample which spawns a number of threads submitting audio blocks representing sinewaves of random duration and frequency to the mixer (called AudioConsumer in this sample). Replace sinewaves by incoming network packets and you should be halfway to a solution.
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();
}
}
