Question

如何同时播放多个（音频）字节数组？这个＆＃34;字节数组＆＃34;由TargetDataLine记录，使用服务器传输。

到目前为止我尝试了什么

使用SourceDataLine：

无法使用SourceDataLine播放多个流，因为write方法会一直阻塞，直到写入缓冲区。使用Threads无法修复此问题，因为只有一个SourceDataLine可以同时写入。

使用AudioPlayer类：

ByteInputStream stream2 = new ByteInputStream(data, 0, data.length);
AudioInputStream stream = new AudioInputStream(stream2, VoiceChat.format, data.length);
AudioPlayer.player.start(stream);

这只会对客户产生噪音。

EDIT 我不会同时收到语音数据包，不会同时收到语音数据包，更多地重复＃34;

Answer 1

显然，Java的Mixer接口不是为此而设计的。

http://docs.oracle.com/javase/7/docs/api/javax/sound/sampled/Mixer.html：

调音台是具有一条或多条线路的音频设备。它不一定是设计用于混合音频信号。

事实上，当我尝试在同一个调音台上打开多行时，这会失败并显示LineUnavailableException。但是，如果您的所有录音都具有相同的音频格式，则可以很容易地将它们手动混合在一起。例如，如果您有2个输入：

将两者转换为适当的数据类型（例如{8}音频为byte[]，16位为short[]，32位浮点为float[]等。
在另一个数组中求和它们。确保求和值不超过数据类型的范围。
将输出转换回字节并将其写入SourceDataLine

另见How is audio represented with numbers?

这是一个混合2个录音并输出为1个信号的示例，全部采用16位48Khz立体声。

    // print all devices (both input and output)
    int i = 0;
    Mixer.Info[] infos = AudioSystem.getMixerInfo();
    for (Mixer.Info info : infos)
        System.out.println(i++ + ": " + info.getName());

    // select 2 inputs and 1 output
    System.out.println("Select input 1: ");
    int in1Index = Integer.parseInt(System.console().readLine());
    System.out.println("Select input 2: ");
    int in2Index = Integer.parseInt(System.console().readLine());
    System.out.println("Select output: ");
    int outIndex = Integer.parseInt(System.console().readLine());

    // ugly java sound api stuff
    try (Mixer in1Mixer = AudioSystem.getMixer(infos[in1Index]);
            Mixer in2Mixer = AudioSystem.getMixer(infos[in2Index]);
            Mixer outMixer = AudioSystem.getMixer(infos[outIndex])) {
        in1Mixer.open();
        in2Mixer.open();
        outMixer.open();
        try (TargetDataLine in1Line = (TargetDataLine) in1Mixer.getLine(in1Mixer.getTargetLineInfo()[0]);
                TargetDataLine in2Line = (TargetDataLine) in2Mixer.getLine(in2Mixer.getTargetLineInfo()[0]);
                SourceDataLine outLine = (SourceDataLine) outMixer.getLine(outMixer.getSourceLineInfo()[0])) {

            // audio format 48khz 16 bit stereo (signed litte endian)
            AudioFormat format = new AudioFormat(48000.0f, 16, 2, true, false);

            // 4 bytes per frame (16 bit samples stereo)
            int frameSize = 4;
            int bufferSize = 4800;
            int bufferBytes = frameSize * bufferSize;

            // buffers for java audio
            byte[] in1Bytes = new byte[bufferBytes];
            byte[] in2Bytes = new byte[bufferBytes];
            byte[] outBytes = new byte[bufferBytes];

            // buffers for mixing
            short[] in1Samples = new short[bufferBytes / 2];
            short[] in2Samples = new short[bufferBytes / 2];
            short[] outSamples = new short[bufferBytes / 2];

            // how long to record & play
            int framesProcessed = 0;
            int durationSeconds = 10;
            int durationFrames = (int) (durationSeconds * format.getSampleRate());

            // open devices
            in1Line.open(format, bufferBytes);
            in2Line.open(format, bufferBytes);
            outLine.open(format, bufferBytes);
            in1Line.start();
            in2Line.start();
            outLine.start();

            // start audio loop
            while (framesProcessed < durationFrames) {

                // record audio
                in1Line.read(in1Bytes, 0, bufferBytes);
                in2Line.read(in2Bytes, 0, bufferBytes);

                // convert input bytes to samples
                ByteBuffer.wrap(in1Bytes).order(ByteOrder.LITTLE_ENDIAN).asShortBuffer().get(in1Samples);
                ByteBuffer.wrap(in2Bytes).order(ByteOrder.LITTLE_ENDIAN).asShortBuffer().get(in2Samples);

                // mix samples - lower volume by 50% since we're mixing 2 streams
                for (int s = 0; s < bufferBytes / 2; s++)
                    outSamples[s] = (short) ((in1Samples[s] + in2Samples[s]) * 0.5);

                // convert output samples to bytes
                ByteBuffer.wrap(outBytes).order(ByteOrder.LITTLE_ENDIAN).asShortBuffer().put(outSamples);

                // play audio
                outLine.write(outBytes, 0, bufferBytes);

                framesProcessed += bufferBytes / frameSize;
            }

            in1Line.stop();
            in2Line.stop();
            outLine.stop();
        }
    }

Answer 2

好吧，我把一些东西放在一起，这应该可以让你开始。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你自己的音频＆＃34;混音器＆＃34;允许该类消费者在（近）未来的特定点安排音频块的类。具体的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准确地从前一个数据包的末尾开始，以便为单个数据播放连续的声音语音。此外，因为你说声音可以重叠我假设（是的，很多假设）新的一个可以通过网络进入，而一个或多个旧的仍在播放。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将音频数据包提交给调音台。

因此，对于submit-audio-packet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类仅用于标记字节数组（音频缓冲区）和＆＃34;当＆＃34;：应该播放该块的时间点。

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，在创建流时将其设置为零并使用缓冲区大小进行更新：

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：

该块已经全部播放。从scheduledBlocks列表中删除。
该块被安排在当前缓冲区之后的某个时间点开始。什么都不做。
（部分）块应该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，预定块的结束可能超过当前缓冲区的末尾，在这种情况下，我们将其第一部分混合并将剩余部分留给下一轮，直到所有它都被播放，整个块被移除。

另请注意，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调音台时，请务必始终让它们从现在开始至少持续1个音频缓冲区，否则您需要这样做。将失去声音开始的风险。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，自包含的样本，它产生了许多线程，提交代表随机持续时间和频率的正弦波的音频块到混音器（在此示例中称为AudioConsumer）。通过传入的网络数据包替换正弦波，您应该已经解决了一半。

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();
}
}

Answer 3

你可以使用Tritontus库来进行软件音频混音（它已经过时但效果还不错）。

将依赖项添加到项目中：

<dependency>
    <groupId>com.googlecode.soundlibs</groupId>
    <artifactId>tritonus-all</artifactId>
    <version>0.3.7.2</version>
</dependency>

使用org.tritonus.share.sampled.FloatSampleBuffer。在调用AudioFormat之前，两个缓冲区必须相同#mix。

// TODO instantiate these variables with real data
byte[] audio1, audio2;
AudioFormat af1, af2;
SourceDataLine sdl = AudioSystem.getSourceDataLine(af1);

FloatSampleBuffer fsb1 = new FloatSampleBuffer(audio1, 0, audio1.length, af1.getFormat());
FloatSampleBuffer fsb2 = new FloatSampleBuffer(audio2, 0, audio2.length, af2.getFormat());

fsb1.mix(fsb2);
byte[] result = fsb1.convertToByteArray(af1);

sdl.write(result, 0, result.length); // play it

在Java中同时播放多个字节数组

3 个答案: