Added audio sender test and disconnect reasons

examples/audio/pom.xml

@@ -0,0 +1,60 @@
+<?xml version="1.0" encoding="UTF-8"?>
+<project xmlns="http://maven.apache.org/POM/4.0.0"
+         xmlns:xsi="http://www.w3.org/2001/XMLSchema-instance"
+         xsi:schemaLocation="http://maven.apache.org/POM/4.0.0 http://maven.apache.org/xsd/maven-4.0.0.xsd">
+    <modelVersion>4.0.0</modelVersion>
+
+    <groupId>com.github.manevolent.ts3j</groupId>
+    <artifactId>audio-example</artifactId>
+    <version>1.0-SNAPSHOT</version>
+
+    <repositories>
+        <repository>
+            <id>tomp2p.net</id>
+            <name>tomp2p</name>
+            <url>http://tomp2p.net/dev/mvn/</url>
+        </repository>
+    </repositories>
+
+    <build>
+        <plugins>
+            <plugin>
+                <groupId>org.apache.maven.plugins</groupId>
+                <artifactId>maven-compiler-plugin</artifactId>
+                <configuration>
+                    <source>1.8</source>
+                    <target>1.8</target>
+                </configuration>
+            </plugin>
+        </plugins>
+    </build>
+
+    <dependencies>
+        <dependency>
+            <groupId>org.apache.commons</groupId>
+            <artifactId>commons-lang3</artifactId>
+            <version>3.8.1</version>
+        </dependency>
+
+        <!-- Opus playback -->
+        <dependency>
+            <groupId>net.tomp2p</groupId>
+            <artifactId>opus-wrapper</artifactId>
+            <version>1.4</version>
+        </dependency>
+
+        <!-- Careful since this dependency may conflict -->
+        <dependency>
+            <groupId>com.sun.jna</groupId>
+            <artifactId>jna</artifactId>
+            <version>3.0.9</version>
+        </dependency>
+
+        <!-- if you've loaded the entire repo into your IDE, and ran the maven "install" step this should work -->
+        <dependency>
+            <groupId>com.github.manevolent</groupId>
+            <artifactId>ts3j</artifactId>
+            <version>1.0-SNAPSHOT</version>
+        </dependency>
+    </dependencies>
+</project>

examples/audio/src/main/java/com/github/manevolent/ts3j/examples/audio/AudioSender.java

@@ -0,0 +1,73 @@
+package com.github.manevolent.ts3j.examples.audio;
+
+import com.github.manevolent.ts3j.identity.LocalIdentity;
+import com.github.manevolent.ts3j.protocol.client.ClientConnectionState;
+import com.github.manevolent.ts3j.protocol.socket.client.LocalTeamspeakClientSocket;
+import com.github.manevolent.ts3j.util.Ts3Debugging;
+
+import static com.github.manevolent.ts3j.examples.audio.TeamspeakFastMixerSink.AUDIO_FORMAT;
+
+public class AudioSender {
+    public static void main(String[] args) throws Exception {
+        Ts3Debugging.setEnabled(false);
+
+        LocalTeamspeakClientSocket client = new LocalTeamspeakClientSocket();
+
+        LocalIdentity identity = LocalIdentity.generateNew(10);
+
+        client.setIdentity(identity);
+        client.setNickname(AudioSender.class.getSimpleName());
+        client.setHWID("TestTestTest");
+
+        TeamspeakFastMixerSink sink = new TeamspeakFastMixerSink(
+                AUDIO_FORMAT,
+                (int) AUDIO_FORMAT.getSampleRate() * AUDIO_FORMAT.getChannels() * 4 /*2=16bit float*/,
+                new OpusParameters(
+                        20,
+                        96000, // 96kbps
+                        10, // max complexity
+                        0, // 0 expected packet loss
+                        false, // no VBR
+                        false, // no FEC
+                        true // OPUS MUSIC - channel doesn't have to be Opus Music ;)
+                )
+        );
+
+        client.setMicrophone(sink);
+
+        // -1 to 1 (signed) positions of samples
+        float[] thirtySecondsOfAudio = new float[(int) AUDIO_FORMAT.getSampleRate() * AUDIO_FORMAT.getChannels()];
+
+        // Generate a sine wave
+        for (int i = 0; i < thirtySecondsOfAudio.length; i ++) {
+            thirtySecondsOfAudio[i] = (float) Math.sin(i / 60f);
+        }
+
+        sink.start();
+
+        // Write into the sink
+        sink.write(thirtySecondsOfAudio, thirtySecondsOfAudio.length);
+
+        client.connect(
+                "your.teamspeak.server",
+                10000L
+        );
+
+        sink.drain(); // this will block until the sink is stopped
+
+        System.err.println("Encoder time taken = " + sink.getNanotime() + " ns");
+        System.err.println("Audio position = " + sink.getPosition() + " samples sank");
+        System.err.println("Encoder position = " + sink.getEncoderPosition() + " samples encoded");
+        System.err.println("Network position = " + sink.getNetworkPosition() + " bytes sent");
+        System.err.println("Packets encoded = " + sink.getPacketsEncoded() + " packets encoded");
+        System.err.println("Packets sent = " + sink.getPacketsSent() + " packets sent");
+        System.err.println("Overflow events = " + sink.getOverflows());
+
+        // (resource starvation; can cause lost packets)
+        System.err.println("Underflow events = " + sink.getUnderflows());
+
+        client.disconnect("BYE");
+
+        System.exit(0);
+    }
+}

examples/audio/src/main/java/com/github/manevolent/ts3j/examples/audio/OpusDecoder.java

@@ -0,0 +1,83 @@
+package com.github.manevolent.ts3j.examples.audio;
+
+import com.sun.jna.ptr.PointerByReference;
+import net.tomp2p.opuswrapper.Opus;
+
+import java.nio.FloatBuffer;
+import java.nio.IntBuffer;
+
+public class OpusDecoder implements AutoCloseable {
+    private final PointerByReference decoder;
+    private final int sampleRate, frameSize, channels;
+
+    public OpusDecoder(int sampleRate, int frameSize, int channels) {
+        this.channels = channels;
+        this.sampleRate = sampleRate;
+        this.frameSize = frameSize;
+
+        IntBuffer errorBuffer = IntBuffer.allocate(1);
+
+        decoder = Opus.INSTANCE.opus_decoder_create(
+                sampleRate,
+                channels,
+                errorBuffer
+        );
+
+        OpusUtil.checkError("opus_decoder_create", errorBuffer.get());
+    }
+
+    public int reset() {
+        return OpusUtil.checkError(
+                "opus_decoder_ctl/OPUS_RESET_STATE",
+                Opus.INSTANCE.opus_decoder_ctl(decoder, Opus.OPUS_RESET_STATE)
+        );
+    }
+
+    public int setDecoderValue(int field, Object... values) {
+        return OpusUtil.checkError(
+                "opus_decoder_ctl/" + field,
+                Opus.INSTANCE.opus_decoder_ctl(decoder, field, values)
+        );
+    }
+
+    public int decodePLC(float[] floats) {
+        return decode(null, floats);
+    }
+
+    public int decode(byte[] packet, float[] floats) {
+        if (floats == null)
+            throw new NullPointerException("floats");
+        else if (floats.length < frameSize * channels)
+            throw new IllegalArgumentException(floats.length + " < " + frameSize * channels);
+
+        int result = Opus.INSTANCE.opus_decode_float(
+                decoder,
+                packet,
+                packet == null ? 0 : packet.length,
+                FloatBuffer.wrap(floats),
+                frameSize,
+                0
+        );
+
+        OpusUtil.checkError("opus_decode_float", result);
+
+        return result;
+    }
+
+    public int getSampleRate() {
+        return sampleRate;
+    }
+
+    public int getChannels() {
+        return channels;
+    }
+
+    @Override
+    public void close() {
+        Opus.INSTANCE.opus_decoder_destroy(decoder);
+    }
+
+    public static String getVersion() {
+        return Opus.INSTANCE.opus_get_version_string();
+    }
+}

examples/audio/src/main/java/com/github/manevolent/ts3j/examples/audio/OpusEncoder.java

@@ -0,0 +1,201 @@
+package com.github.manevolent.ts3j.examples.audio;
+
+import com.sun.jna.ptr.PointerByReference;
+import net.tomp2p.opuswrapper.Opus;
+import org.apache.commons.lang3.ArrayUtils;
+
+import java.nio.ByteBuffer;
+import java.nio.ByteOrder;
+import java.nio.IntBuffer;
+import java.nio.ShortBuffer;
+
+/**
+ * Thread-safe opus encoder
+ */
+public class OpusEncoder implements AutoCloseable {
+    private static final int[] OPUS_PERMITTED_SAMPLE_RATES =
+            { 8000, 12000, 16000, 24000, 48000 };
+
+    private static final int[] OPUS_PERMITTED_FRAME_SIZES =
+            { 120, 240, 480, 960, 1920, 2880 };
+
+    private static final int[] OPUS_PERMITTED_CHANNEL_COUNTS =
+            { 1, 2 };
+
+    private final PointerByReference encoder;
+    private final int sampleRate, frameSize, channels;
+    private final int expectedByteSize;
+    private final boolean bigEndian;
+
+    private final ShortBuffer sourceShortBuffer;
+    private final ByteBuffer sourceByteBuffer;
+    private final ByteBuffer targetBuffer;
+
+    private final Object encoderLock = new Object();
+
+    private boolean closed = false;
+
+    public OpusEncoder(int sampleRate, int frameSize, int channels, boolean bigEndian) {
+        this(sampleRate, frameSize, channels, bigEndian, 4096);
+    }
+
+    public OpusEncoder(int sampleRate, int frameSize,
+                       int channels, boolean bigEndian,
+                       int maxPacketLength) {
+        this.channels = channels;
+        this.sampleRate = sampleRate;
+        this.frameSize = frameSize;
+        this.bigEndian = bigEndian;
+        this.expectedByteSize = frameSize * channels * 2;
+
+        if (!ArrayUtils.contains(OPUS_PERMITTED_FRAME_SIZES, frameSize))
+            throw new IllegalArgumentException("Invalid Opus frame size: " + frameSize);
+
+        if (!ArrayUtils.contains(OPUS_PERMITTED_SAMPLE_RATES, sampleRate))
+            throw new IllegalArgumentException("Invalid Opus sample rate: " + sampleRate);
+
+        if (!ArrayUtils.contains(OPUS_PERMITTED_CHANNEL_COUNTS, channels))
+            throw new IllegalArgumentException("Invalid Opus channel count: " + channels);
+
+        this.sourceShortBuffer = ShortBuffer.allocate(frameSize * channels);
+        this.sourceByteBuffer = ByteBuffer.allocate(frameSize * channels * 2);
+        this.sourceByteBuffer.order(bigEndian ? ByteOrder.BIG_ENDIAN : ByteOrder.LITTLE_ENDIAN);
+
+        this.targetBuffer = ByteBuffer.allocate(maxPacketLength);
+
+        IntBuffer errorBuffer = IntBuffer.allocate(1);
+
+        encoder = Opus.INSTANCE.opus_encoder_create(
+                sampleRate,
+                channels,
+                Opus.OPUS_APPLICATION_AUDIO,
+                errorBuffer
+        );
+
+        if (encoder == null) throw new NullPointerException("encoder");
+
+        OpusUtil.checkError(
+                "opus_encoder_create",
+                errorBuffer.get()
+        );
+    }
+
+    public void reset() {
+        synchronized (encoderLock) {
+            OpusUtil.checkError(
+                    "opus_encoder_ctl/OPUS_RESET_STATE",
+                    Opus.INSTANCE.opus_encoder_ctl(encoder, Opus.OPUS_RESET_STATE)
+            );
+        }
+    }
+
+    public int setEncoderValue(int field, Object... values) {
+        synchronized (encoderLock) {
+            if (closed) throw new IllegalStateException("encoder closed");
+
+            return OpusUtil.checkError(
+                    "opus_encoder_ctl/" + field,
+                    Opus.INSTANCE.opus_encoder_ctl(encoder, field, values)
+            );
+        }
+    }
+
+    public byte[] encode(float[] floats, int len) {
+        if (len != getFrameSize() * getChannels())
+            throw new IllegalArgumentException(len + " != " + getFrameSize() * getChannels());
+
+        targetBuffer.clear();
+
+        int result;
+        synchronized (encoderLock) {
+            if (closed) throw new IllegalStateException("encoder closed");
+
+            if (encoder == null) throw new IllegalStateException("encoder is null");
+
+            result = Opus.INSTANCE.opus_encode_float(
+                    encoder,
+                    floats,
+                    frameSize, // this is STATIC.
+                    targetBuffer,
+                    targetBuffer.capacity()
+            );
+        }
+
+        OpusUtil.checkError(
+                "opus_encode_float," +
+                        " ch=" + channels +
+                        " smprate=" + sampleRate +
+                        " rate=" + sampleRate +
+                        " len=" + len + "/" + floats.length + ", frame_size=" + frameSize +
+                        ", max_data_bytes=" + targetBuffer.capacity(),
+                result
+        );
+
+        byte[] encoded = new byte[result];
+        targetBuffer.get(encoded);
+
+        return encoded;
+    }
+
+    public byte[] encode(float[] floats) {
+        return encode(floats, floats.length);
+    }
+
+    public byte[] encode(byte[] pcm) {
+        if (pcm.length != expectedByteSize)
+            throw new IllegalArgumentException(pcm.length + " != " + expectedByteSize);
+
+        // Clear source and input the PCM samples
+        sourceByteBuffer.clear();
+        sourceByteBuffer.put(pcm);
+        sourceByteBuffer.flip();
+
+        sourceShortBuffer.clear();
+        while (sourceByteBuffer.position() < sourceByteBuffer.limit())
+            sourceShortBuffer.put(sourceByteBuffer.getShort());
+
+        // Clear target
+        sourceShortBuffer.flip();
+        targetBuffer.clear();
+
+        int result;
+
+        synchronized (encoderLock) {
+            if (closed) throw new IllegalStateException("encoder closed");
+
+            result = Opus.INSTANCE.opus_encode(
+                    encoder,
+                    sourceShortBuffer,
+                    frameSize, // this is STATIC.
+                    targetBuffer,
+                    targetBuffer.capacity()
+            );
+        }
+
+        OpusUtil.checkError("opus_encode", result);
+
+        byte[] encoded = new byte[result];
+        targetBuffer.get(encoded);
+
+        return encoded;
+    }
+
+    public int getFrameSize() { return frameSize; }
+
+    public int getSampleRate() {
+        return sampleRate;
+    }
+
+    public int getChannels() {
+        return channels;
+    }
+
+    @Override
+    public void close() {
+        synchronized (encoderLock) {
+            if (closed) throw new IllegalStateException("already closed");
+            Opus.INSTANCE.opus_encoder_destroy(encoder);
+            closed = true;
+        }
+    }
+}

examples/audio/src/main/java/com/github/manevolent/ts3j/examples/audio/OpusParameters.java

@@ -0,0 +1,51 @@
+package com.github.manevolent.ts3j.examples.audio;
+
+public class OpusParameters {
+    private final int opusFrameRate;
+    private final int opusBitrate;
+    private final int opusComplexity;
+    private final int opusPacketLoss;
+    private final boolean opusVbr;
+    private final boolean opusFec;
+    private final boolean opusMusic;
+
+    public OpusParameters(int opusFrameRate, int opusBitrate,
+                          int opusComplexity, int opusPacketLoss,
+                          boolean opusVbr, boolean opusFec, boolean opusMusic) {
+        this.opusFrameRate = opusFrameRate;
+        this.opusBitrate = opusBitrate;
+        this.opusComplexity = opusComplexity;
+        this.opusPacketLoss = opusPacketLoss;
+        this.opusVbr = opusVbr;
+        this.opusFec = opusFec;
+        this.opusMusic = opusMusic;
+    }
+
+    public int getOpusFrameTime() {
+        return opusFrameRate;
+    }
+
+    public int getOpusBitrate() {
+        return opusBitrate;
+    }
+
+    public int getOpusComplexity() {
+        return opusComplexity;
+    }
+
+    public int getOpusPacketLoss() {
+        return opusPacketLoss;
+    }
+
+    public boolean isOpusVbr() {
+        return opusVbr;
+    }
+
+    public boolean isOpusFec() {
+        return opusFec;
+    }
+
+    public boolean isOpusMusic() {
+        return opusMusic;
+    }
+}

examples/audio/src/main/java/com/github/manevolent/ts3j/examples/audio/OpusUtil.java

@@ -0,0 +1,26 @@
+package com.github.manevolent.ts3j.examples.audio;
+
+import net.tomp2p.opuswrapper.Opus;
+
+public class OpusUtil {
+    public static int checkError(String description, int returnCode) throws RuntimeException {
+        if (returnCode < Opus.OPUS_OK)
+            throw new RuntimeException("Opus error [" + description + "] (" + returnCode + "): " +
+                    Opus.INSTANCE.opus_strerror(returnCode).trim()
+            );
+        else return returnCode;
+    }
+
+    public static int checkError(int returnCode) throws RuntimeException {
+        if (returnCode < Opus.OPUS_OK)
+            throw new RuntimeException(
+                    "Opus error (" + returnCode + "): " +
+                    Opus.INSTANCE.opus_strerror(returnCode).trim()
+            );
+        else return returnCode;
+    }
+
+    public static String getVersion() {
+        return Opus.INSTANCE.opus_get_version_string();
+    }
+}

examples/audio/src/main/java/com/github/manevolent/ts3j/examples/audio/TeamspeakFastMixerSink.java

@@ -0,0 +1,410 @@
+package com.github.manevolent.ts3j.examples.audio;
+
+import com.github.manevolent.ts3j.enums.CodecType;
+import net.tomp2p.opuswrapper.Opus;
+
+import javax.sound.sampled.AudioFormat;
+import java.util.NoSuchElementException;
+import java.util.Queue;
+import java.util.concurrent.ConcurrentLinkedQueue;
+import java.util.logging.Level;
+import java.util.logging.Logger;
+
+public class TeamspeakFastMixerSink implements TeamspeakMixerSink {
+    public static final AudioFormat AUDIO_FORMAT =
+            new AudioFormat(48000, 32, 2, true, false);
+
+    private final Object stateLock = new Object();
+    private final Object drainLock = new Object();
+
+    // Audio format
+    private final AudioFormat audioFormat; // Audio output format (mostly just ensured to match certain parameters)
+
+    // OPUS variables
+    private final OpusParameters opusParameters; // Opus parameters set by configurations
+    private volatile OpusEncoder encoder = null; // Opus encoder instance
+    private final int opusFrameSize; // Opus frame size (PER CHANNEL!) (also the ASIO chunk size)
+    private long opusPacketsEncoded = 0, opusPacketsSent = 0;
+    private long opusTime = 0;
+    private long networkTime = 0;
+    private long opusPosition = 0;
+    private long opusBytePosition = 0;
+
+    private long waitTime = 0;
+
+    // Statistics variables
+    private long underflowed = 0; // Underflowed samples (ASIO has to return 0 samples to the encoder)
+    private long overflowed = 0; // Overflowed samples (write returns 0)
+    private long clipped = 0; // Clipped samples (outside -1F,1F bounds)
+    private long position = 0; // Position in samples
+
+    // I/O variables
+    private final Queue<OpusPacket> packetQueue = new ConcurrentLinkedQueue<>();
+    private final int bufferSize; // Buffer size, in samples
+    private volatile int availableInput;
+    private volatile int samplePosition; // Position, in samples, of the ASIO buffer (available samples)
+    private final float[] sampleBuffer; // ASIO buffer
+
+    // Mixer sink state variables
+    private volatile boolean running = false, opening = false;
+
+    public TeamspeakFastMixerSink(AudioFormat audioFormat,
+                                  int bufferSizeInBytes,
+                                  OpusParameters opusParameters) {
+
+        this.audioFormat = audioFormat;
+
+        if (audioFormat.getSampleRate() != 48000)
+            throw new IllegalArgumentException("Invalid audio sample rate: " +
+                    audioFormat.getSampleRate() + " != 48000: OPUS requires 48KHz audio");
+        else if (audioFormat.getChannels() != 2)
+            throw new IllegalArgumentException("Invalid audio channel count: " +
+                    audioFormat.getChannels() + " != 2: OPUS requires stereo audio");
+
+        this.bufferSize = bufferSizeInBytes / (audioFormat.getSampleSizeInBits() / 8);
+        if (bufferSize <= 0 || bufferSize % audioFormat.getChannels() != 0)
+            throw new IllegalArgumentException("invalid bufferSize: " + bufferSize);
+
+        this.sampleBuffer = new float[bufferSize];
+        this.availableInput = 0;
+
+        this.opusParameters = opusParameters;
+        this.opusFrameSize = (int) getAudioFormat().getSampleRate() / (1000 / opusParameters.getOpusFrameTime());
+    }
+
+    public AudioFormat getAudioFormat() {
+        return audioFormat;
+    }
+
+    public void write(float[] buffer, int len) {
+        synchronized (stateLock) {
+            if (!running) throw new IllegalStateException("not running");
+
+            if (len > availableInput()) { // This should never happen
+                overflowed++; // Probably never will run
+                throw new IllegalArgumentException(len + " > " + availableInput());
+            }
+
+            if (len > buffer.length) {
+                overflowed++;
+                throw new IllegalArgumentException(len + " > " + buffer.length);
+            }
+
+            if (len <= 0)
+                throw new IllegalArgumentException(len + " <= 0");
+
+            if (len % getChannels() != 0)
+                throw new IllegalArgumentException("not a full frame");
+
+            // Write to the buffer
+            System.arraycopy(buffer, 0, sampleBuffer, samplePosition, len);
+            samplePosition += len;
+            availableInput -= len;
+
+            // Read samples from the buffer and encode them into packets
+            position += encode(false);
+        }
+    }
+
+    /**
+     * Encodes the buffer to packets for the packet queue
+     * @return Packets encoded
+     */
+    private int encode(boolean flush) {
+        if (flush) Logger.getGlobal().log(Level.FINE, "Flushing TeamspeakFastMixerSink...");
+
+        int written = 0;
+        int frameSize = opusFrameSize * getChannels();
+        byte[] encoded;
+        long now;
+        int copy;
+
+        while (samplePosition >= (flush ? 1 : frameSize)) {
+            copy = Math.min(samplePosition, frameSize);
+            if (flush && copy < frameSize) {
+                for (int i = copy; i < frameSize; i++)
+                    sampleBuffer[i] = 0f;
+
+                samplePosition = frameSize;
+            }
+
+            if (encoder == null) openOpusEncoder();
+
+            now = System.nanoTime();
+            encoded = encoder.encode(sampleBuffer, frameSize);
+            opusTime += (System.nanoTime() - now);
+            opusPacketsEncoded ++;
+            opusPosition += frameSize;
+
+            System.arraycopy(sampleBuffer, frameSize, sampleBuffer, 0, samplePosition - frameSize);
+            samplePosition -= frameSize;
+            if (packetQueue.add(new OpusPacket(frameSize, encoded)))
+                written+=frameSize;
+        }
+
+        if (written > 0) opening = false;
+
+        if (flush) Logger.getGlobal().log(Level.FINE, "Flushed TeamspeakFastMixerSink.");
+
+        return written;
+    }
+
+    public long getNanotime() {
+        return opusTime;
+    }
+    public long getFrameSize() {
+        return opusFrameSize;
+    }
+    public long getEncoderPosition() {
+        return opusPosition;
+    }
+    public long getNetworkPosition() {
+        return opusBytePosition;
+    }
+    public OpusParameters getEncoderParameters() {
+        return opusParameters;
+    }
+
+    @Override
+    public long getPacketsEncoded() {
+        return opusPacketsEncoded;
+    }
+
+    @Override
+    public long getPacketsSent() {
+        return opusPacketsSent;
+    }
+
+    /**
+     * Finds the count of available samples to be written to write() in the len param
+     * @return Available sample count
+     */
+    public int availableInput() {
+        return availableInput;
+    }
+
+    public boolean isRunning() {
+        return running;
+    }
+
+    /**
+     * Opens the Opus encoder
+     */
+    private void openOpusEncoder() {
+        synchronized (stateLock) {
+            if (encoder != null) return;
+            Logger.getGlobal().log(Level.FINE, "Opening TeamspeakFastMixerSink encoder...");
+
+            encoder = new OpusEncoder(
+                    (int) getAudioFormat().getSampleRate(), // smp rate (always 48kHz)
+                    opusFrameSize,
+                    getChannels(),
+                    audioFormat.isBigEndian()
+                    //,MAXIMUM_OPUS_PACKET_SIZE // Prevent IP fragmentation
+            );
+
+            encoder.setEncoderValue(
+                    Opus.OPUS_SET_SIGNAL_REQUEST,
+                    opusParameters.isOpusMusic() ? Opus.OPUS_SIGNAL_MUSIC : Opus.OPUS_SIGNAL_VOICE
+            );
+
+            encoder.setEncoderValue(Opus.OPUS_SET_BITRATE_REQUEST, opusParameters.getOpusBitrate());
+            encoder.setEncoderValue(Opus.OPUS_SET_COMPLEXITY_REQUEST, opusParameters.getOpusComplexity());
+            encoder.setEncoderValue(Opus.OPUS_SET_PACKET_LOSS_PERC_REQUEST, opusParameters.getOpusPacketLoss());
+            encoder.setEncoderValue(Opus.OPUS_SET_VBR_REQUEST, opusParameters.isOpusVbr() ? 1 : 0);
+            encoder.setEncoderValue(Opus.OPUS_SET_INBAND_FEC_REQUEST, opusParameters.isOpusFec() ? 1 : 0);
+
+            Logger.getGlobal().log(Level.FINE, "Opened TeamspeakFastMixerSink encoder.");
+        }
+    }
+
+    private void closeOpusEncoder() {
+        synchronized (stateLock) {
+            if (encoder != null) {
+                Logger.getGlobal().log(Level.FINE, "Closing TeamspeakFastMixerSink encoder...");
+                encoder.close();
+                encoder = null;
+                Logger.getGlobal().log(Level.FINE, "Closed TeamspeakFastMixerSink encoder.");
+            }
+        }
+    }
+
+    public void close() {
+        if (isRunning()) stop();
+
+        closeOpusEncoder();
+    }
+
+    public boolean start() {
+        synchronized (stateLock) {
+            if (running) return false;
+
+            Logger.getGlobal().log(Level.FINE, "Starting TeamspeakFastMixerSink...");
+
+            // Flush buffers, clear outgoing packet queues.
+            for (int i = 0; i < sampleBuffer.length; i ++) sampleBuffer[i] = 0f;
+            samplePosition = 0;
+            availableInput = bufferSize;
+            packetQueue.clear();
+
+            // Open (or re-open) Opus encoder.
+            openOpusEncoder();
+
+            // Reset the encoder
+            if (encoder != null) {
+                Logger.getGlobal().log(Level.FINE, "Resetting TeamspeakFastMixerSink...");
+                encoder.reset(); // Reset encoder output
+                Logger.getGlobal().log(Level.FINE, "Reset TeamspeakFastMixerSink.");
+            }
+
+            // Mark as running.
+            opening = true;
+            running = true;
+
+            Logger.getGlobal().log(Level.FINE, "Started TeamspeakFastMixerSink.");
+
+            return true;
+        }
+    }
+
+    public void drain() throws InterruptedException {
+        synchronized (drainLock) {
+            while (running && packetQueue.peek() != null) {
+                drainLock.wait();
+            }
+        }
+    }
+
+    /**
+     * Closes mixer sink
+     *
+     * Note that we don't close the Opus encoder - just in case there are more samples to write/flush out.
+     */
+    public boolean stop() {
+        synchronized (stateLock) {
+            if (!running) return false;
+
+            Logger.getGlobal().log(Level.FINE, "Stopping TeamspeakFastMixerSink...");
+
+            try {
+                encode(true);
+            } catch (RuntimeException e) {
+                Logger.getGlobal().log(Level.WARNING, "Problem flushing audio buffer upon close", e);
+            }
+
+            running = false;
+            Logger.getGlobal().log(Level.FINE, "Stopped TeamspeakFastMixerSink.");
+
+            return true;
+        }
+    }
+
+    public int getBufferSize() {
+        return bufferSize;
+    }
+
+    public long getPosition() {
+        return position;
+    }
+
+    public long getClippedSamples() {
+        return clipped;
+    }
+
+    public long getUnderflows() {
+        return underflowed;
+    }
+
+    public long getOverflows() {
+        return overflowed;
+    }
+
+    /**
+     * Called by the TS3J encoder thread to determine if the sink is ready to send samples.
+     * @return true if the sink has samples to write, or if samples will become available.
+     */
+    @Override
+    public boolean isReady() {
+        return (running && !opening) || packetQueue.size() > 0;
+    }
+
+    @Override
+    public CodecType getCodec() {
+        return CodecType.OPUS_MUSIC;
+    }
+
+    /**
+     * Provides OPUS encoded audio (as a packet) to the TS3J UDP audio network thread.
+     * @return Encoded OPUS audio packet (zero-length packet if there is an underflow).
+     */
+    @Override
+    public byte[] provide() {
+        long start = System.nanoTime();
+
+        try {
+            if (packetQueue.peek() == null) {
+                underflowed ++;
+                return new byte[0];
+            }
+
+            OpusPacket packet = packetQueue.remove();
+
+            if (packet == null) {
+                underflowed ++;
+                return new byte[0];
+            }
+
+            availableInput += packet.getSamples();
+            opusBytePosition += packet.getBytes().length;
+            opusPacketsSent ++;
+
+            return packet.getBytes();
+        } catch (NoSuchElementException ex) {
+            underflowed ++;
+            return new byte[0];
+        } finally {
+            long networkTime = System.nanoTime() - start;
+
+            if (networkTime >= (opusParameters.getOpusFrameTime() * 1000000L)) {
+                underflowed++;
+
+                Logger.getGlobal().warning("[TeamspeakFastMixerSink] provide() took longer than the expected " +
+                        opusParameters.getOpusFrameTime() + "ms: " + ((double)networkTime / 1_000_000D) + "ms");
+            }
+
+            this.networkTime += networkTime;
+
+            if (packetQueue.peek() == null) {
+                synchronized (drainLock) {
+                    drainLock.notifyAll();
+                }
+            }
+        }
+    }
+
+    public int getChannels() {
+        return getAudioFormat().getChannels();
+    }
+
+    public long getWaitTime() {
+        return waitTime;
+    }
+
+    private final class OpusPacket {
+        private final int samples;
+        private final byte[] bytes;
+
+        protected OpusPacket(int samples, byte[] bytes) {
+            this.samples = samples;
+            this.bytes = bytes;
+        }
+
+        public int getSamples() {
+            return samples;
+        }
+
+        public byte[] getBytes() {
+            return bytes;
+        }
+    }
+}

examples/audio/src/main/java/com/github/manevolent/ts3j/examples/audio/TeamspeakMixerSink.java

@@ -0,0 +1,54 @@
+package com.github.manevolent.ts3j.examples.audio;
+
+import com.github.manevolent.ts3j.audio.Microphone;
+
+public interface TeamspeakMixerSink extends Microphone {
+    /**
+     * Gets the Encoder parameters
+     * @return Encoder parameters
+     */
+    OpusParameters getEncoderParameters();
+
+    /**
+     * Gets the count of encoded Encoder packets
+     * @return Encoder packets
+     */
+    long getPacketsEncoded();
+
+    /**
+     * Gets the count of written encoded Encoder packets
+     * @return Encoder packets
+     */
+    long getPacketsSent();
+
+    /**
+     * Gets the time spent encoding Encoder packets, in nanoseconds
+     * @return Encoder time
+     */
+    long getNanotime();
+
+    /**
+     * Gets the Encoder frame size for packets
+     * @return Encoder frame size
+     */
+    long getFrameSize();
+
+    /**
+     * Gets the Encoder position in samples
+     * @return Encoder position
+     */
+    long getEncoderPosition();
+
+    /**
+     * Gets the Encoder network position in bytes
+     * @return Encoder network position
+     */
+    long getNetworkPosition();
+
+    /**
+     * Gets the channel count
+     * @return Channel count
+     */
+    int getChannels();
+
+}

src/main/java/com/github/manevolent/ts3j/protocol/socket/client/LocalTeamspeakClientSocket.java

@@ -1088,16 +1088,34 @@ public class LocalTeamspeakClientSocket
             throws
             IOException, TimeoutException,
             ExecutionException, InterruptedException {
+        disconnect(null);
+    }
+
+    public void disconnect(String reason)
+            throws
+            IOException, TimeoutException,
+            ExecutionException, InterruptedException {
         try {
             switch (getState()) {
                 case CONNECTING:
                 case RETRIEVING_DATA:
                     waitForState(ClientConnectionState.CONNECTED, 10000L);
                 case CONNECTED:
-                    writePacket(new PacketBody2Command(
+                    if (reason == null) {
-                                    ProtocolRole.CLIENT,
+                        writePacket(new PacketBody2Command(
-                                    new SingleCommand("clientdisconnect", ProtocolRole.CLIENT))
+                                ProtocolRole.CLIENT,
-                    );
+                                new SingleCommand("clientdisconnect", ProtocolRole.CLIENT))
+                        );
+                    } else {
+                        writePacket(new PacketBody2Command(
+                                ProtocolRole.CLIENT,
+                                new SingleCommand(
+                                        "clientdisconnect", ProtocolRole.CLIENT,
+                                        new CommandSingleParameter("reasonid", "8"),
+                                        new CommandSingleParameter("reasonmsg", reason)
+                                ))
+                        );
+                    }
 
                     waitForState(ClientConnectionState.DISCONNECTED, 30000L);
 

src/test/java/com/github/manevolent/ts3j/ServerConnectionTest.java

@@ -10,6 +10,7 @@ import com.github.manevolent.ts3j.protocol.client.ClientConnectionState;
 import com.github.manevolent.ts3j.protocol.packet.PacketBody0Voice;
 import com.github.manevolent.ts3j.protocol.socket.client.LocalTeamspeakClientSocket;
 import com.github.manevolent.ts3j.util.Ts3Debugging;
+import com.sun.security.ntlm.Server;
 
 import java.math.BigInteger;
 import java.net.InetSocketAddress;
@@ -20,34 +21,29 @@ import static org.junit.Assert.assertEquals;
 
 public class ServerConnectionTest  {
     public static void main(String[] args) throws Exception {
+        Ts3Debugging.setEnabled(true);
 
         LocalTeamspeakClientSocket client = new LocalTeamspeakClientSocket();
 
-        Ts3Debugging.setEnabled(false);
-
         LocalIdentity identity = LocalIdentity.generateNew(10);
 
         client.setIdentity(identity);
-        client.setNickname("ts3j issue 2");
+        client.setNickname(ServerConnectionTest.class.getSimpleName());
         client.setHWID("TestTestTest");
 
-        client.setVoiceHandler(packet ->
+        try {
-                System.out.println(packet.getServerFlag0())
+            client.connect(
-        );
+                    "teamlixo.net",
+                    10000L
+            );
 
-        while (true) {
+            assertEquals(client.getState(), ClientConnectionState.CONNECTED);
-            try {
-                client.connect(
-                        "teamlixo.net",
-                        10000L
-                );
 
-                assertEquals(client.getState(), ClientConnectionState.CONNECTED);
+            client.disconnect("BYE");
-            } catch (Exception ex) {
+        } catch (Exception ex) {
-                ex.printStackTrace();
+            ex.printStackTrace();
-            }
-
-            Thread.sleep(30000);
         }
+
+        Thread.sleep(3000L);
     }
 }