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Play incoming voice through TeamSpeak's adaptive jitter buffer

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Each speaker's stream held a fixed 60 ms and gave up after 200 ms of
concealment, so a jittery connection concealed constantly and a stall
ended the burst. It now runs through a port of the libspeex jitter
buffer the official client links, configured and driven as it does:

- the delay adapts to the measured jitter, aiming for at most 1% of
  packets arriving late
- a speaker keeps one timeline, never re-anchored between talk bursts,
  so what the buffer learned carries over; only 5 s without packets
  drops it
- the talker's stop marker never enters the buffer; playback ends when
  it reaches it, and its slot is played as silence, not concealed

A pull now always hands out one 20 ms frame: what a longer packet or a
delay increase produces beyond that is played on the following pulls,
which Android's mixer used to cut off.

Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com>
This commit is contained in:
ericek111
2026-09-25 10:46:08 +00:00
parent 7cb4f47a81
commit 0083dc8ad6
5 changed files with 1226 additions and 129 deletions
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584
ts3-client/core/src/main/java/com/ts3client/audio/JitterBuffer.java Normal file
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@@ -0,0 +1,584 @@
package com.ts3client.audio;
/**
* Adaptive jitter buffer for incoming voice packets.
*
* <p>This is a port of the jitter buffer TeamSpeak itself uses: the official client links
* libspeex's {@code libspeex/jitter.c} verbatim (its diagnostics are still in the shipped
* binary) and drives it through the wrapper described in {@link #forTeamSpeakVoice}. The
* algorithm is what makes a voice stream survive a bad network: packets are reordered by
* timestamp, losses are reported so the codec can conceal them, and — most importantly —
* the playback delay continuously re-tunes itself to the measured jitter instead of being
* a fixed guess.
*
* <h2>How the delay adapts</h2>
* Every packet's arrival is recorded as a <em>timing</em>: how early (positive) or late
* (negative) it was relative to the moment it was needed. Timings are kept in three
* rotating sub-windows, each holding only the {@value #MAX_TIMINGS} <em>latest</em> arrivals
* it saw, which makes the set a cheap running estimate of the distribution's late tail.
* {@link #computeOptimalDelay()} then walks the {@value #TOP_DELAY} latest timings and picks
* the delay minimising {@code delay + lateCost * (packets that would then be late)}. The
* result is applied a whole {@link #delayStep} at a time, by dropping a frame to shrink the
* delay or inserting a concealed one to grow it, so the correction is inaudible.
*
* <h2>Timestamp units</h2>
* Timestamps and spans are in caller-defined ticks; the buffer never converts them to time.
* All comparisons are wrap-safe, so a tick counter may overflow freely.
*
* <p>Not thread-safe: every call must be externally serialised.
*/
public final class JitterBuffer {
/** Outcome of a {@link #get} call. */
public enum Status {
/** A packet was available; see {@link #payload()}. */
OK,
/** Nothing available — the caller should conceal {@link #span()} ticks of loss. */
MISSING,
/** The buffer is growing its delay; the caller should insert {@link #span()} ticks. */
INSERTION
}
/** How many packets may be held at once, awaiting playback or reordering. */
private static final int MAX_PACKETS = 200;
/** Timings kept per sub-window. Only the latest arrivals are retained. */
private static final int MAX_TIMINGS = 40;
/** Rotating sub-windows the timings are spread over. */
private static final int SUB_WINDOWS = 3;
/** How many of the latest timings the delay search considers. */
private static final int TOP_DELAY = 40;
/**
* The latest arrival timings seen within one sub-window, sorted ascending.
*
* <p>Only the {@value #MAX_TIMINGS} latest are kept: an arrival too early to displace
* any of them is counted but not stored, since the delay search only ever looks at the
* late tail.
*/
private static final class TimingWindow {
final int[] timing = new int[MAX_TIMINGS];
/** Timings actually stored. */
int filled;
/** Arrivals seen, including those too early to store. */
int seen;
void clear() {
filled = 0;
seen = 0;
}
void add(int value) {
if (filled >= MAX_TIMINGS && value >= timing[filled - 1]) {
seen++;
return;
}
int pos = 0;
while (pos < filled && value >= timing[pos]) pos++;
if (pos < filled) {
int moveSize = filled - pos;
if (filled == MAX_TIMINGS) moveSize--;
System.arraycopy(timing, pos, timing, pos + 1, moveSize);
}
timing[pos] = value;
seen++;
if (filled < MAX_TIMINGS) filled++;
}
}
private final byte[][] data = new byte[MAX_PACKETS][];
private final int[] timestamp = new int[MAX_PACKETS];
private final int[] span = new int[MAX_PACKETS];
/** Value of {@link #nextStop} when the packet arrived, or 0 if that is unknown. */
private final int[] arrival = new int[MAX_PACKETS];
private final TimingWindow[] windows = new TimingWindow[SUB_WINDOWS];
/** {@link #windows} in age order, newest first; rotated in place. */
private final TimingWindow[] byAge = new TimingWindow[SUB_WINDOWS];
private final int delayStep;
private final int concealmentSize;
private int bufferMargin;
private int maxLateRate;
private int lateCost;
private int windowSize;
private int subWindowSize;
/** Playback position: the timestamp the next {@link #get} wants. */
private int pointerTimestamp;
/** Timestamp playback will have reached by the time the next packet is needed. */
private int nextStop;
/** Ticks handed out by the last {@link #get} beyond what was asked for. */
private int buffered;
/** Ticks of concealment {@link #get} has been told to insert before reading on. */
private int interpolationRequested;
/** Consecutive {@link #get} calls that found nothing. */
private int lostCount;
/** True until the first packet arrives and defines the timeline. */
private boolean resetState;
/** Self-tuned cost of a late packet, used when {@link #lateCost} is 0. */
private int autoTradeoff;
// Result of the most recent get(), kept here so the per-frame path allocates nothing.
private byte[] resultPayload;
private int resultTimestamp;
private int resultSpan;
/**
* @param stepSize granularity of delay changes, and the span of one concealed frame;
* normally the span of a single packet
* @param margin extra delay held on top of the measured optimum
* @param maxLateRate percentage of packets allowed to arrive too late, which also sets
* how long a window the delay is estimated over
* @param lateCost cost of one late packet relative to delay, or 0 to self-tune
*/
public JitterBuffer(int stepSize, int margin, int maxLateRate, int lateCost) {
this.delayStep = stepSize;
this.concealmentSize = stepSize;
this.bufferMargin = margin;
this.lateCost = lateCost;
setMaxLateRate(maxLateRate);
for (int i = 0; i < SUB_WINDOWS; i++) {
windows[i] = new TimingWindow();
byAge[i] = windows[i];
}
reset();
}
/**
* A buffer configured the way the official TeamSpeak client configures its own.
*
* <p>Recovered from {@code ts3client_linux_amd64} (3.6.x): it calls
* {@code jitter_buffer_init(60)} and then sets margin 60 and max late rate 1, pulling
* one 60-tick frame per {@code jitter_buffer_get}. One tick is therefore a sixtieth of
* a 20&nbsp;ms voice frame, so the delay moves in whole-frame steps, one frame of slack
* is held on top of the measured optimum, and the target is for no more than 1% of
* packets to arrive late — a deliberately conservative setting that estimates over a
* ~1.3&nbsp;s window and is the reason TeamSpeak rides out network trouble smoothly.
*
* @param ticksPerFrame ticks spanned by one voice frame; see {@link #TEAMSPEAK_TICKS_PER_FRAME}
*/
public static JitterBuffer forTeamSpeakVoice(int ticksPerFrame) {
return new JitterBuffer(ticksPerFrame, ticksPerFrame, 1, 0);
}
/** Ticks per voice frame in TeamSpeak's own timebase, i.e. a 20 ms frame in 1/3 ms units. */
public static final int TEAMSPEAK_TICKS_PER_FRAME = 60;
/**
* Sets the share of packets allowed to arrive too late, as a percentage.
*
* <p>This doubles as the estimator's time constant: a stricter rate needs a
* proportionally longer window to observe that many late packets.
*/
public void setMaxLateRate(int rate) {
this.maxLateRate = Math.max(1, rate);
this.windowSize = 100 * TOP_DELAY / this.maxLateRate;
this.subWindowSize = this.windowSize / SUB_WINDOWS;
}
/** Sets the extra delay held on top of the measured optimum. */
public void setMargin(int margin) {
this.bufferMargin = margin;
}
public int margin() {
return bufferMargin;
}
/** Ticks currently held ahead of the playback position, i.e. the live buffering depth. */
public int bufferedSpan() {
int total = 0;
for (int i = 0; i < MAX_PACKETS; i++) {
if (data[i] != null && ge(timestamp[i], pointerTimestamp)) total += span[i];
}
return total;
}
/** True when no packet is held and playback has nothing left to drain. */
public boolean isEmpty() {
for (int i = 0; i < MAX_PACKETS; i++) {
if (data[i] != null) return false;
}
return true;
}
/**
* Drops every held packet and forgets both the timeline and the network estimate.
*
* <p>Reserved for the cases where a speaker's position genuinely stopped meaning
* anything — a new speaker, or a silence long enough that the sequence may have been
* recycled. Notably <em>not</em> something to do between talk bursts: the timeline stays
* valid across a pause because the sender stops numbering while the receiver stops
* playing, and the estimate needs far longer than one burst to be worth anything.
*/
public void reset() {
for (int i = 0; i < MAX_PACKETS; i++) data[i] = null;
pointerTimestamp = 0;
nextStop = 0;
resetState = true;
lostCount = 0;
buffered = 0;
interpolationRequested = 0;
autoTradeoff = 32000;
for (TimingWindow w : windows) w.clear();
System.arraycopy(windows, 0, byAge, 0, SUB_WINDOWS);
}
/**
* Offers an arrived packet to the buffer.
*
* @param payload the packet, retained by reference until played or dropped
* @param timestamp the packet's position on the sender's timeline, in ticks
* @param span how many ticks the packet covers
*/
public void put(byte[] payload, int timestamp, int span) {
// Drop anything playback has already moved past.
if (!resetState) {
for (int i = 0; i < MAX_PACKETS; i++) {
if (data[i] != null && le(this.timestamp[i] + this.span[i], pointerTimestamp)) {
data[i] = null;
}
}
}
// A packet that missed its slot still tells us how far behind the clock we are.
boolean late = !resetState && lt(timestamp, nextStop);
if (late) {
updateTimings(timestamp - nextStop - bufferMargin);
}
// The consumer has stopped fetching; don't sit on packets it will never take.
if (lostCount > 20) reset();
// Ignore packets so late that even concealment has moved past them.
if (!resetState && !ge(timestamp + span + delayStep, pointerTimestamp)) return;
int slot = -1;
for (int i = 0; i < MAX_PACKETS; i++) {
if (data[i] == null) {
slot = i;
break;
}
}
if (slot < 0) {
// Full: make room by discarding the oldest packet.
slot = 0;
int earliest = this.timestamp[0];
for (int i = 1; i < MAX_PACKETS; i++) {
if (lt(this.timestamp[i], earliest)) {
earliest = this.timestamp[i];
slot = i;
}
}
}
data[slot] = payload;
this.timestamp[slot] = timestamp;
this.span[slot] = span;
this.arrival[slot] = (resetState || late) ? 0 : nextStop;
}
/**
* Takes the next {@code desiredSpan} ticks of audio.
*
* <p>On {@link Status#OK} the packet is in {@link #payload()}. On {@link Status#MISSING}
* or {@link Status#INSERTION} the caller must produce {@link #span()} ticks of
* concealment instead — for Opus that means decoding a null packet.
*/
public Status get(int desiredSpan) {
resultPayload = null;
// The first packet to arrive defines where the timeline starts.
if (resetState) {
int oldest = 0;
boolean found = false;
for (int i = 0; i < MAX_PACKETS; i++) {
if (data[i] != null && (!found || lt(timestamp[i], oldest))) {
oldest = timestamp[i];
found = true;
}
}
if (!found) {
resultTimestamp = 0;
resultSpan = interpolationRequested;
return Status.MISSING;
}
resetState = false;
pointerTimestamp = oldest;
nextStop = oldest;
}
// A delay increase decided during the last tick(): pad before reading on.
if (interpolationRequested != 0) {
resultTimestamp = pointerTimestamp;
resultSpan = interpolationRequested;
pointerTimestamp += interpolationRequested;
interpolationRequested = 0;
buffered = resultSpan - desiredSpan;
return Status.INSERTION;
}
int found = findPacket(desiredSpan);
if (found >= 0) {
lostCount = 0;
// An on-time arrival: record how much slack it had.
if (arrival[found] != 0) {
updateTimings(timestamp[found] - arrival[found] - bufferMargin);
}
resultPayload = data[found];
resultTimestamp = timestamp[found];
resultSpan = span[found];
data[found] = null;
pointerTimestamp = resultTimestamp + resultSpan;
buffered = resultSpan - desiredSpan;
return Status.OK;
}
lostCount++;
int optimal = computeOptimalDelay();
if (optimal < 0) {
// The measured jitter wants a deeper buffer: insert without advancing playback.
shiftTimings(-optimal);
resultTimestamp = pointerTimestamp;
resultSpan = -optimal;
buffered = resultSpan - desiredSpan;
return Status.INSERTION;
}
// Ordinary loss: conceal a whole number of frames and move on.
int concealed = roundDown(desiredSpan, concealmentSize);
resultTimestamp = pointerTimestamp;
resultSpan = concealed;
pointerTimestamp += concealed;
buffered = concealed - desiredSpan;
return Status.MISSING;
}
/**
* The held packet sitting at the playback position, without consuming it.
*
* <p>After a {@link Status#MISSING} the position has already stepped past the gap, so
* this is the packet immediately following the lost one — the one carrying the FEC copy
* that can rebuild it. {@code null} when that packet is missing too.
*/
public byte[] nextPayload() {
for (int i = 0; i < MAX_PACKETS; i++) {
if (data[i] != null && timestamp[i] == pointerTimestamp) return data[i];
}
return null;
}
/** The timestamp the next {@link #get} will play from. */
public int position() {
return pointerTimestamp;
}
/** Whole frames currently held, given a frame spans {@code frameSpan} ticks. */
public int heldFrames(int frameSpan) {
int held = 0;
for (int i = 0; i < MAX_PACKETS; i++) {
if (data[i] != null) held += Math.max(1, span[i] / frameSpan);
}
return held;
}
/** The packet returned by the last {@link #get}, or {@code null} if it was not {@link Status#OK}. */
public byte[] payload() {
return resultPayload;
}
/** Ticks covered by the last {@link #get}, whether a real packet or concealment. */
public int span() {
return resultSpan;
}
/** Timestamp of the last {@link #get}. */
public int timestamp() {
return resultTimestamp;
}
/**
* Advances the playback clock by one frame, after the audio from {@link #get} has been
* handed to the output. This is also where the delay is re-tuned.
*/
public void tick() {
updateDelay();
nextStop = pointerTimestamp - Math.max(0, buffered);
buffered = 0;
}
/**
* Advances the playback clock by one frame that the caller played from audio it still had
* from an earlier {@link #get}, instead of calling {@link #get} and {@link #tick}.
*
* @param remaining ticks of that audio still left after this frame
*/
public void remainingSpan(int remaining) {
updateDelay();
nextStop = pointerTimestamp - remaining;
}
/** Picks the held packet that best covers the chunk starting at the playback position. */
private int findPacket(int desiredSpan) {
// Exact start, covering the whole chunk.
for (int i = 0; i < MAX_PACKETS; i++) {
if (data[i] != null && timestamp[i] == pointerTimestamp
&& ge(timestamp[i] + span[i], pointerTimestamp + desiredSpan)) {
return i;
}
}
// An earlier packet still covering the whole chunk.
for (int i = 0; i < MAX_PACKETS; i++) {
if (data[i] != null && le(timestamp[i], pointerTimestamp)
&& ge(timestamp[i] + span[i], pointerTimestamp + desiredSpan)) {
return i;
}
}
// An earlier packet covering part of it.
for (int i = 0; i < MAX_PACKETS; i++) {
if (data[i] != null && le(timestamp[i], pointerTimestamp)
&& gt(timestamp[i] + span[i], pointerTimestamp)) {
return i;
}
}
// Anything starting inside the chunk: prefer the earliest, then the longest.
int best = -1;
for (int i = 0; i < MAX_PACKETS; i++) {
if (data[i] == null) continue;
if (!(lt(timestamp[i], pointerTimestamp + desiredSpan) && ge(timestamp[i], pointerTimestamp))) continue;
if (best < 0 || lt(timestamp[i], timestamp[best])
|| (timestamp[i] == timestamp[best] && gt(span[i], span[best]))) {
best = i;
}
}
return best;
}
/**
* Applies the delay the timings call for, by scheduling an insertion to grow the buffer
* or skipping ahead to shrink it. The timings are shifted by the same amount so the
* next estimate is made relative to the new delay.
*/
private void updateDelay() {
int optimal = computeOptimalDelay();
if (optimal == 0) return;
shiftTimings(-optimal);
pointerTimestamp += optimal;
if (optimal < 0) interpolationRequested = -optimal;
}
/**
* Finds the delay minimising {@code delay + lateCost * latePackets} over the observed
* timings, quantised to whole {@link #delayStep}s.
*
* @return ticks to shift playback by: negative to buffer more, positive to catch up
*/
private int computeOptimalDelay() {
int totalSeen = 0;
for (TimingWindow w : windows) totalSeen += w.seen;
if (totalSeen == 0) return 0;
float lateFactor = lateCost != 0
? lateCost * 100.0f / totalSeen
: (float) autoTradeoff * windowSize / totalSeen;
int[] pos = new int[SUB_WINDOWS];
int optimal = 0;
int bestCost = Integer.MAX_VALUE;
int late = 0;
boolean penaltyTaken = false;
int earliestSeen = 0;
int latestSeen = 0;
// Walk the timings from latest to earliest, costing each as a candidate delay.
for (int i = 0; i < TOP_DELAY; i++) {
int next = -1;
int timing = Integer.MAX_VALUE;
for (int j = 0; j < SUB_WINDOWS; j++) {
if (pos[j] < windows[j].filled && windows[j].timing[pos[j]] < timing) {
next = j;
timing = windows[j].timing[pos[j]];
}
}
if (next < 0) break;
if (i == 0) latestSeen = timing;
earliestSeen = timing;
timing = roundDown(timing, delayStep);
pos[next]++;
int cost = (int) (-timing + lateFactor * late);
if (cost < bestCost) {
bestCost = cost;
optimal = timing;
}
late++;
// Hysteresis: charge extra the first time we would start making packets late.
if (timing >= 0 && !penaltyTaken) {
penaltyTaken = true;
late += 4;
}
}
autoTradeoff = 1 + (earliestSeen - latestSeen) / TOP_DELAY;
// Too little evidence to justify shrinking the buffer.
if (totalSeen < TOP_DELAY && optimal > 0) return 0;
return optimal;
}
/** Records one arrival's slack, rotating the sub-windows when the newest fills up. */
private void updateTimings(int timing) {
timing = Math.max(-32767, Math.min(32767, timing));
if (byAge[0].seen >= subWindowSize) {
TimingWindow oldest = byAge[SUB_WINDOWS - 1];
System.arraycopy(byAge, 0, byAge, 1, SUB_WINDOWS - 1);
byAge[0] = oldest;
oldest.clear();
}
byAge[0].add(timing);
}
/** Re-bases every recorded timing after the delay moved. */
private void shiftTimings(int amount) {
for (TimingWindow w : windows) {
for (int i = 0; i < w.filled; i++) w.timing[i] += amount;
}
}
private static int roundDown(int value, int step) {
return value < 0 ? (value - step + 1) / step * step : value / step * step;
}
// Wrap-safe comparisons, so tick counters may overflow.
private static boolean lt(int a, int b) {
return a - b < 0;
}
private static boolean le(int a, int b) {
return a - b <= 0;
}
private static boolean gt(int a, int b) {
return a - b > 0;
}
private static boolean ge(int a, int b) {
return a - b >= 0;
}
}

327
ts3-client/core/src/main/java/com/ts3client/audio/VoiceStream.java
View File

@@ -4,13 +4,21 @@ import com.github.manevolent.ts3j.enums.CodecType;
import com.ts3client.audio.opus.OpusCodec;
import com.ts3client.audio.opus.OpusDecoder;
import java.util.HashMap;
import java.util.Map;
import java.util.function.BiConsumer;
/**
* One speaker's incoming voice: packets go into a small jitter buffer as they arrive, and
* every {@link #pull} plays out the next one in sequence, concealing any that went missing.
* One speaker's incoming voice: packets go into an adaptive {@link JitterBuffer} as they
* arrive, and every {@link #pull} plays out the next 20&nbsp;ms, concealing what went missing.
*
* <p>The buffer is driven the way the official client drives it. A speaker has one timeline,
* {@code unwrap16(packetId) * 60} ticks, which is never re-anchored between talk bursts: the
* sender stops numbering when it stops talking and playback stops at the same point, so the
* next burst lands where the last one left off and the delay the buffer has learned carries
* over. Only a long silence drops the timeline.
*
* <p>An empty packet is the talker letting go of the key. It never enters the buffer; its
* position is remembered, the burst ends once playback reaches it, and its slot is played as
* silence rather than concealed, since nothing was lost there.
*
* <p>The stream keeps no clock of its own: whoever pulls sets the pace, one call per
* 20&nbsp;ms frame. {@link #offer} and {@link #stop} may be called from any thread; pulls
@@ -23,36 +31,51 @@ final class VoiceStream {
/** Longest Opus frame (120 ms @ 48 kHz) a packet may decode to, per channel. */
static final int MAX_FRAME = 5760;
/** Initial playout delay: enough room for common UDP jitter without feeling laggy. */
private static final int JITTER_DELAY_FRAMES = 3;
/** Cap PLC during a lost end-of-talk marker or a hard network stall. */
private static final int MAX_CONSECUTIVE_PLC_FRAMES = 10;
/** Keep memory bounded when a talk burst runs far ahead of playback. */
private static final int MAX_PENDING_PACKETS = 32;
private record VoiceFrame(int packetId, CodecType codec, byte[] data, boolean end) {
}
private static final int FRAME = VoiceFormat.FRAME_SIZE;
private static final int FRAME_TICKS = JitterBuffer.TEAMSPEAK_TICKS_PER_FRAME;
/** Pulls in a row with nothing held before a burst whose end marker got lost is given up. */
private static final int EMPTY_PULLS_TO_END = 10;
/** How far back a talker's stop is remembered, as in the official client. */
private static final int STOP_MARK_WINDOW_TICKS = 3000;
/** Silence after which the timeline and the delay estimate are dropped, as in the official client. */
private static final long IDLE_RESET_NANOS = 5_000_000_000L;
private static final int EPOCH_UNSET = Integer.MIN_VALUE;
/** What a pull plays, decided under the lock and produced outside it. */
private enum Source { PACKET, CONCEAL, SILENCE, SURPLUS }
final int clientId;
private final OpusCodec opus;
private final BiConsumer<Integer, Boolean> talkListener;
private final Object lock = new Object();
private final Map<Integer, VoiceFrame> pending = new HashMap<>();
// Guarded by lock.
private final JitterBuffer jitter = JitterBuffer.forTeamSpeakVoice(FRAME_TICKS);
private int epoch = EPOCH_UNSET;
private int lastSeq;
private int[] stopMarks = new int[8];
private int stopMarkCount;
/** Channels of the latest packet: TeamSpeak streams {@code OPUS_MUSIC} in stereo. */
private int channels = 1;
private boolean playing;
private int expectedPacketId;
private int activeChannels = 1;
private int consecutivePlcFrames;
/** Pulls still to sit out before the burst starts playing. */
private int warmup;
private int emptyPulls;
/** Frames of decoded audio left over from the last packet, still to be played. */
private int surplusFrames;
/** Frames of concealment still owed to a delay increase. */
private int concealOwed;
/** Set when the timeline was dropped; the decoder follows on the next pull. */
private boolean resetDecoder;
/** Guards the decoder: a stop or close from another thread may reach it mid-pull. */
/** Guards the decoder: a close from another thread may reach it mid-pull. */
private final Object decoderLock = new Object();
private final float[] pcm = new float[MAX_FRAME * VoiceFormat.MAX_CHANNELS];
/** Audio a packet decoded to beyond its first frame; only the pulling thread touches it. */
private final float[] surplus = new float[MAX_FRAME * VoiceFormat.MAX_CHANNELS];
private int surplusOffset;
private int surplusChannels;
private OpusDecoder decoder;
private int decoderChannels;
/** Length of the last packet, which a lost one most likely had too. */
private int lastFrameSize = VoiceFormat.FRAME_SIZE;
private boolean closed;
private volatile boolean talking;
@@ -64,36 +87,48 @@ final class VoiceStream {
this.talkListener = talkListener;
}
/** TeamSpeak streams {@code OPUS_MUSIC} in stereo and everything else in mono. */
private static int channelsFor(CodecType codec) {
return codec == CodecType.OPUS_MUSIC ? 2 : 1;
}
/**
* Queues an incoming packet; an empty one marks the end of the talk burst.
* Places an incoming packet on the speaker's timeline; an empty one marks the end of the
* talk burst.
*
* @return true if the packet started a new burst, so playout has to begin
*/
boolean offer(int packetId, CodecType codec, byte[] data) {
lastPacketNanos = System.nanoTime();
VoiceFrame frame = new VoiceFrame(packetId & 0xFFFF, codec, data, data == null || data.length == 0);
long now = System.nanoTime();
boolean longIdle = lastPacketNanos != 0 && now - lastPacketNanos >= IDLE_RESET_NANOS;
lastPacketNanos = now;
boolean end = data == null || data.length == 0;
boolean started = false;
synchronized (lock) {
if (longIdle && !playing) {
jitter.reset();
epoch = EPOCH_UNSET;
stopMarkCount = 0;
resetDecoder = true;
}
boolean fresh = epoch == EPOCH_UNSET;
int ticks = unwrap(packetId & 0xFFFF) * FRAME_TICKS;
if (end) {
markStop(ticks);
return false;
}
// Playback already went past it; the buffer would only drop it.
if (!fresh && ticks - jitter.position() < 0) return false;
channels = channelsFor(codec);
jitter.put(data, ticks, FRAME_TICKS);
if (!playing) {
playing = true;
started = true;
expectedPacketId = frame.packetId();
activeChannels = channelsFor(codec);
consecutivePlcFrames = 0;
warmup = JITTER_DELAY_FRAMES - 1;
} else if (packetDistance(expectedPacketId, frame.packetId()) < 0) {
return false; // too late for this burst; do not replay stale audio
emptyPulls = 0;
}
pending.putIfAbsent(frame.packetId(), frame);
trimPending();
}
if (!frame.end()) markTalking(true);
markTalking(true);
return started;
}
@@ -101,78 +136,139 @@ final class VoiceStream {
* Plays out the next frame into {@code out}, interleaved over {@code outChannels} and
* scaled by {@code gain} but not clipped, so it can still be mixed.
*
* @param out room for {@link #MAX_FRAME} samples per channel
* @return the samples written per channel; 0 while the burst is still buffering or a
* packet could not be decoded, or {@link #IDLE} once the burst is over
* @param out room for {@link VoiceFormat#FRAME_SIZE} samples per channel
* @return the samples written per channel, at most one frame; 0 if a packet could not be
* decoded, or {@link #IDLE} once the burst is over
*/
int pull(float[] out, int outChannels, double gain) {
VoiceFrame frame;
int channels;
Source source;
byte[] packet = null;
int surplusTake = 0;
int packetChannels;
boolean reset;
synchronized (lock) {
if (!playing) return IDLE;
if (warmup > 0) {
warmup--;
return 0;
}
reset = resetDecoder;
resetDecoder = false;
packetChannels = channels;
frame = pending.remove(expectedPacketId);
boolean conceal = frame == null;
if (conceal) {
consecutivePlcFrames++;
if (surplusFrames > 0 || concealOwed > 0) {
// Still playing out what the last get() handed over.
if (surplusFrames > 0) {
source = Source.SURPLUS;
surplusTake = Math.min(FRAME, surplusFrames);
surplusFrames -= surplusTake;
} else {
source = Source.CONCEAL;
concealOwed--;
}
jitter.remainingSpan(remainingTicks());
} else {
consecutivePlcFrames = 0;
}
if ((frame != null && frame.end()) || consecutivePlcFrames > MAX_CONSECUTIVE_PLC_FRAMES) {
stopLocked();
frame = null;
channels = 0;
} else {
channels = frame != null ? channelsFor(frame.codec()) : activeChannels;
activeChannels = channels;
expectedPacketId = (expectedPacketId + 1) & 0xFFFF;
boolean empty = jitter.isEmpty();
emptyPulls = empty ? emptyPulls + 1 : 0;
if (empty && (stoppedAt(jitter.position()) || emptyPulls >= EMPTY_PULLS_TO_END)) {
stopLocked();
source = null;
} else {
JitterBuffer.Status status = jitter.get(FRAME_TICKS);
switch (status) {
case OK -> {
source = Source.PACKET;
packet = jitter.payload();
}
case MISSING -> source = isStopMark(jitter.timestamp()) ? Source.SILENCE : Source.CONCEAL;
default -> {
// The buffer grows its delay by inserting this many frames.
source = Source.CONCEAL;
concealOwed = Math.max(1, jitter.span() / FRAME_TICKS) - 1;
}
}
jitter.tick();
}
}
}
if (channels == 0) {
finish();
if (source == null) {
markTalking(false);
return IDLE;
}
return decode(frame != null ? frame.data() : null, channels, out, outChannels, gain);
}
private int decode(byte[] data, int channels, float[] out, int outChannels, double gain) {
int frames;
synchronized (decoderLock) {
if (closed) return 0;
try {
OpusDecoder d = decoderFor(channels);
if (data == null) {
frames = d.conceal(pcm, lastFrameSize);
} else {
frames = d.decode(data, pcm);
lastFrameSize = frames;
}
if (reset && decoder != null) decoder.reset();
frames = produce(source, packet, surplusTake, packetChannels, out, outChannels, gain);
} catch (RuntimeException e) {
return 0;
}
}
// Match the decoded stream to the output: duplicate mono across stereo, fold a
// stereo (music) stream down onto mono.
return frames;
}
private int produce(Source source, byte[] packet, int surplusTake, int packetChannels, float[] out,
int outChannels, double gain) {
switch (source) {
case SURPLUS -> {
emit(surplus, surplusOffset, surplusTake, surplusChannels, out, outChannels, gain);
surplusOffset += surplusTake;
return surplusTake;
}
case SILENCE -> {
java.util.Arrays.fill(out, 0, FRAME * outChannels, 0f);
return FRAME;
}
case CONCEAL -> {
int frames = decoderFor(decoderChannels == 0 ? packetChannels : decoderChannels).conceal(pcm, FRAME);
emit(pcm, 0, frames, decoderChannels, out, outChannels, gain);
return frames;
}
default -> {
int frames = decoderFor(packetChannels).decode(packet, pcm);
int played = Math.min(frames, FRAME);
emit(pcm, 0, played, packetChannels, out, outChannels, gain);
if (frames > played) keepSurplus(played, frames - played, packetChannels);
return played;
}
}
}
/** Holds back what a packet decoded to beyond one frame, for the following pulls. */
private void keepSurplus(int from, int frames, int channels) {
System.arraycopy(pcm, from * channels, surplus, 0, frames * channels);
surplusOffset = 0;
surplusChannels = channels;
synchronized (lock) {
if (playing) surplusFrames = frames;
}
}
/** Ticks of audio already handed out but not yet played; guarded by the lock. */
private int remainingTicks() {
return (surplusFrames * FRAME_TICKS + FRAME - 1) / FRAME + concealOwed * FRAME_TICKS;
}
/**
* Matches decoded audio to the output: duplicates mono across stereo and folds a stereo
* (music) stream down onto mono.
*/
private static void emit(float[] src, int fromFrame, int frames, int channels, float[] out, int outChannels,
double gain) {
int base = fromFrame * channels;
for (int i = 0, k = 0; i < frames; i++) {
for (int c = 0; c < outChannels; c++, k++) {
double v;
if (channels == outChannels) {
v = pcm[i * channels + c];
v = src[base + i * channels + c];
} else if (channels == 1) {
v = pcm[i];
v = src[base + i];
} else {
double sum = 0;
for (int s = 0; s < channels; s++) sum += pcm[i * channels + s];
for (int s = 0; s < channels; s++) sum += src[base + i * channels + s];
v = sum / channels;
}
out[k] = (float) (v * gain);
}
}
return frames;
}
/**
@@ -188,24 +284,66 @@ final class VoiceStream {
return decoder;
}
/**
* Widens the 16-bit packet id into a continuous sequence. A packet that arrives after the
* counter wrapped, but was sent before it, counts back into the previous epoch.
*/
private int unwrap(int seq) {
if (epoch == EPOCH_UNSET) {
epoch = 0;
lastSeq = seq;
} else {
int delta = (short) (seq - lastSeq);
if (delta > 0) {
if (seq < lastSeq) epoch++;
lastSeq = seq;
} else if (seq > lastSeq) {
epoch--;
}
}
return (epoch << 16) | seq;
}
private void markStop(int ticks) {
int position = jitter.position();
int kept = 0;
for (int i = 0; i < stopMarkCount; i++) {
if (position - stopMarks[i] < STOP_MARK_WINDOW_TICKS) stopMarks[kept++] = stopMarks[i];
}
stopMarkCount = kept;
if (stopMarkCount == stopMarks.length) stopMarks = java.util.Arrays.copyOf(stopMarks, stopMarkCount * 2);
stopMarks[stopMarkCount++] = ticks;
}
private boolean isStopMark(int ticks) {
for (int i = 0; i < stopMarkCount; i++) {
if (stopMarks[i] == ticks) return true;
}
return false;
}
/** Whether the talker let go at or shortly before {@code position}, so the burst is spoken out. */
private boolean stoppedAt(int position) {
for (int i = 0; i < stopMarkCount; i++) {
int behind = position - stopMarks[i];
if (behind >= 0 && behind < STOP_MARK_WINDOW_TICKS) return true;
}
return false;
}
/** Ends the current talk burst, if any; the next packet starts a new one. */
void stop() {
synchronized (lock) {
stopLocked();
}
finish();
markTalking(false);
}
private void stopLocked() {
playing = false;
pending.clear();
}
private void finish() {
synchronized (decoderLock) {
if (decoder != null) decoder.reset();
}
markTalking(false);
surplusFrames = 0;
concealOwed = 0;
emptyPulls = 0;
}
boolean isPlaying() {
@@ -238,25 +376,4 @@ final class VoiceStream {
this.talking = talking;
talkListener.accept(clientId, talking);
}
private static int packetDistance(int from, int to) {
int distance = (to - from) & 0xFFFF;
return distance >= 0x8000 ? distance - 0x10000 : distance;
}
private void trimPending() {
while (pending.size() > MAX_PENDING_PACKETS) {
Integer furthest = null;
int furthestDistance = Integer.MIN_VALUE;
for (Integer packetId : pending.keySet()) {
int distance = packetDistance(expectedPacketId, packetId);
if (distance > furthestDistance) {
furthestDistance = distance;
furthest = packetId;
}
}
if (furthest == null) return;
pending.remove(furthest);
}
}
}

7
ts3-client/core/src/test/java/com/ts3client/audio/FakeOpus.java
View File

@@ -9,7 +9,7 @@ import java.util.Arrays;
import java.util.List;
/**
* Stands in for libopus: a "packet" is one byte, decoded to a 20 ms frame holding that byte
* Stands in for libopus: a "packet" of n bytes decodes to n 20 ms frames holding the first byte
* divided by 100 in every sample. Concealment yields -1, so it is easy to spot.
*/
final class FakeOpus implements OpusCodec {
@@ -33,8 +33,9 @@ final class FakeOpus implements OpusCodec {
return new OpusDecoder() {
@Override
public int decode(byte[] packet, float[] out) {
Arrays.fill(out, 0, VoiceFormat.FRAME_SIZE * channels, packet[0] / 100f);
return VoiceFormat.FRAME_SIZE;
int frames = VoiceFormat.FRAME_SIZE * packet.length;
Arrays.fill(out, 0, frames * channels, packet[0] / 100f);
return frames;
}
@Override

353
ts3-client/core/src/test/java/com/ts3client/audio/JitterBufferTest.java Normal file
View File

@@ -0,0 +1,353 @@
package com.ts3client.audio;
import org.junit.jupiter.api.Test;
import java.util.ArrayList;
import java.util.List;
import java.util.Random;
import static org.junit.jupiter.api.Assertions.assertEquals;
import static org.junit.jupiter.api.Assertions.assertFalse;
import static org.junit.jupiter.api.Assertions.assertNotNull;
import static org.junit.jupiter.api.Assertions.assertNull;
import static org.junit.jupiter.api.Assertions.assertTrue;
class JitterBufferTest {
private static final int SPAN = JitterBuffer.TEAMSPEAK_TICKS_PER_FRAME;
private static JitterBuffer buffer() {
return JitterBuffer.forTeamSpeakVoice(SPAN);
}
/** Frame {@code n} of a stream, tagged so it can be identified on the way out. */
private static byte[] frame(int n) {
return new byte[]{(byte) n};
}
private static void put(JitterBuffer b, int n) {
b.put(frame(n), n * SPAN, SPAN);
}
/** Pulls one frame, returning its tag, or -1 for concealment. */
private static int pull(JitterBuffer b) {
JitterBuffer.Status status = b.get(SPAN);
int tag = status == JitterBuffer.Status.OK ? b.payload()[0] : -1;
b.tick();
return tag;
}
@Test
void playsAnUndisturbedStreamInOrder() {
JitterBuffer b = buffer();
List<Integer> played = new ArrayList<>();
for (int i = 0; i < 20; i++) {
put(b, i);
played.add(pull(b));
}
// The buffer holds some delay back, so the tail is still queued; nothing is reordered.
List<Integer> real = played.stream().filter(n -> n >= 0).toList();
assertEquals(real.stream().sorted().toList(), real);
assertTrue(real.size() >= 15, "most frames should have played: " + played);
}
@Test
void reordersPacketsThatArriveOutOfOrder() {
JitterBuffer b = buffer();
// Prime a delay so a swapped pair still lands in time.
for (int i = 0; i < 12; i++) {
put(b, i);
pull(b);
}
put(b, 13);
put(b, 12);
put(b, 14);
List<Integer> played = new ArrayList<>();
for (int i = 0; i < 6; i++) played.add(pull(b));
List<Integer> real = played.stream().filter(n -> n >= 0).toList();
assertEquals(real.stream().sorted().toList(), real, "played out of order: " + played);
assertTrue(real.contains(12) && real.contains(13) && real.contains(14), "lost a frame: " + played);
}
@Test
void reportsMissingFramesInsteadOfSkippingAhead() {
JitterBuffer b = buffer();
for (int i = 0; i < 30; i++) {
if (i != 20) put(b, i);
}
int concealed = 0;
int played = 0;
for (int i = 0; i < 30; i++) {
if (pull(b) < 0) concealed++;
else played++;
}
assertTrue(concealed >= 1, "the gap should have been concealed");
assertEquals(29, played, "every delivered frame should still be played");
}
@Test
void everyGetYieldsExactlyOneFrameOfAudio() {
JitterBuffer b = buffer();
Random random = new Random(7);
long ticks = 0;
for (int i = 0; i < 200; i++) {
if (random.nextInt(10) != 0) put(b, i);
b.get(SPAN);
assertEquals(0, b.span() % SPAN, "a partial frame cannot be rendered");
assertTrue(b.span() > 0);
ticks += b.span();
b.tick();
}
// Concealment for a growing delay may run long, but audio never stops or doubles up.
assertTrue(ticks >= 200L * SPAN, "output underran: " + ticks);
}
@Test
void growsTheDelayWhenArrivalsAreJittery() {
JitterBuffer b = buffer();
Random random = new Random(11);
// Deliver in bursts: frames pile up, then nothing arrives for a while.
int next = 0;
int concealedEarly = 0;
int concealedLate = 0;
for (int i = 0; i < 400; i++) {
if (i % 5 == 0) {
int burst = 3 + random.nextInt(4);
for (int j = 0; j < burst && next < 400; j++) put(b, next++);
}
boolean concealed = pull(b) < 0;
if (i < 100) concealedEarly += concealed ? 1 : 0;
else if (i >= 300) concealedLate += concealed ? 1 : 0;
}
assertTrue(concealedLate < concealedEarly,
"the buffer should settle: " + concealedEarly + " -> " + concealedLate);
}
@Test
void staysStableAcrossTimestampWraparound() {
JitterBuffer b = buffer();
// Start just below the point where the tick counter overflows.
int base = Integer.MAX_VALUE - 5 * SPAN;
for (int i = 0; i < 40; i++) {
b.put(frame(i), base + i * SPAN, SPAN);
}
int played = 0;
for (int i = 0; i < 40; i++) {
if (b.get(SPAN) == JitterBuffer.Status.OK) {
assertNotNull(b.payload());
played++;
}
b.tick();
}
assertTrue(played >= 35, "wraparound lost frames: " + played);
}
@Test
void resetForgetsTheTimelineSoTheNextBurstResyncs() {
JitterBuffer b = buffer();
for (int i = 0; i < 10; i++) {
put(b, i);
pull(b);
}
b.reset();
assertTrue(b.isEmpty());
// A new burst numbered far away must still play, not be treated as hopelessly late.
b.put(frame(1), 500 * SPAN, SPAN);
assertEquals(JitterBuffer.Status.OK, b.get(SPAN));
assertEquals(1, b.payload()[0]);
}
@Test
void dropsPacketsThatArriveHopelesslyLate() {
JitterBuffer b = buffer();
for (int i = 0; i < 40; i++) {
put(b, i);
pull(b);
}
assertFalse(b.isEmpty() && b.bufferedSpan() != 0);
// A frame from far in the past has nowhere to go and must not stall the stream.
b.put(frame(99), 0, SPAN);
for (int i = 0; i < 5; i++) {
assertTrue(pull(b) != 99, "a hopelessly late frame was played");
}
}
@Test
void teamSpeakConfigurationMatchesTheShippedClient() {
JitterBuffer b = JitterBuffer.forTeamSpeakVoice(JitterBuffer.TEAMSPEAK_TICKS_PER_FRAME);
assertEquals(60, JitterBuffer.TEAMSPEAK_TICKS_PER_FRAME);
assertEquals(60, b.margin());
}
/** Plays until nothing is held, leaving the position just past the last real frame. */
private static void drain(JitterBuffer b) {
for (int i = 0; i < 100 && !b.isEmpty(); i++) pull(b);
}
@Test
void offersTheFollowingPacketToRebuildALostOne() {
JitterBuffer b = buffer();
for (int i = 0; i < 12; i++) put(b, i);
drain(b);
// The frame due now never arrives, but the one after it is already here,
// carrying the FEC copy that can rebuild it.
int due = b.position();
b.put(frame(13), due + SPAN, SPAN);
assertEquals(JitterBuffer.Status.MISSING, b.get(SPAN));
assertNotNull(b.nextPayload(), "the successor should be offered for FEC recovery");
assertEquals(13, b.nextPayload()[0]);
b.tick();
// And it is still there to be played normally afterwards.
assertEquals(13, pull(b));
}
@Test
void offersNothingWhenTheSuccessorIsAlsoLost() {
JitterBuffer b = buffer();
for (int i = 0; i < 12; i++) put(b, i);
drain(b);
// Two frames in a row are gone; FEC only reaches back one.
int due = b.position();
b.put(frame(15), due + 2 * SPAN, SPAN);
assertEquals(JitterBuffer.Status.MISSING, b.get(SPAN));
assertNull(b.nextPayload(), "a double loss cannot be rebuilt from FEC");
}
@Test
void aPauseInSpeechCostsNothingWhenTheClockStopsToo() {
JitterBuffer b = buffer();
for (int i = 0; i < 40; i++) {
put(b, i);
pull(b);
}
drain(b);
// Silence. No packets arrive and nothing is pulled, so the position stays put —
// which is what lets the sender's numbering still line up when speech resumes.
int paused = b.position();
assertEquals(paused, b.position());
b.put(frame(7), paused, SPAN);
assertEquals(JitterBuffer.Status.OK, b.get(SPAN), "the next burst should play, not be judged late");
assertEquals(7, b.payload()[0]);
}
@Test
void keepsItsEstimateAcrossAPause() {
JitterBuffer b = buffer();
Random random = new Random(5);
int next = 0;
for (int i = 0; i < 300; i++) {
if (i % 4 == 0) {
for (int j = 0, burst = 2 + random.nextInt(4); j < burst; j++) put(b, next++);
}
pull(b);
}
drain(b);
JitterBuffer fresh = buffer();
assertTrue(concealedInBurst(b, b.position() / SPAN) <= concealedInBurst(fresh, 1000),
"a buffer that survived the pause should conceal no more than a cold one");
}
/** Plays a short jittery burst through a buffer and counts the concealed frames. */
private static int concealedInBurst(JitterBuffer b, int firstFrame) {
Random random = new Random(9);
int concealed = 0;
int next = firstFrame;
for (int i = 0; i < 60; i++) {
if (i % 4 == 0) {
for (int j = 0, burst = 2 + random.nextInt(4); j < burst; j++) {
b.put(frame(next), next * SPAN, SPAN);
next++;
}
}
if (pull(b) < 0) concealed++;
}
return concealed;
}
@Test
void resetForgetsTheNetworkEstimateToo() {
JitterBuffer b = buffer();
for (int i = 0; i < 100; i++) {
put(b, i);
pull(b);
}
b.reset();
// With no timings recorded there is nothing to justify moving the delay.
assertTrue(b.isEmpty());
assertEquals(0, b.heldFrames(SPAN));
}
@Test
void heldFramesCountsWhatIsWaiting() {
JitterBuffer b = buffer();
assertEquals(0, b.heldFrames(SPAN));
put(b, 0);
put(b, 1);
put(b, 2);
assertEquals(3, b.heldFrames(SPAN));
}
@Test
void aShortUtteranceComesOutWhole() {
JitterBuffer b = buffer();
int played = 0;
// Five frames, one per frame period — a single short word.
for (int i = 0; i < 5; i++) {
put(b, i);
if (pull(b) >= 0) played++;
}
// Then whatever the delay is still holding back.
for (int i = 0; i < 20 && !b.isEmpty(); i++) {
if (pull(b) >= 0) played++;
}
assertEquals(5, played, "every frame of a short utterance must be heard");
}
@Test
void pullingOnlyWhenAPacketIsWaitingStarvesTheStream() {
// The buffer adapts its delay in tick(), which only runs on a pull. A consumer that
// skips the pull whenever the buffer happens to be empty pins the delay at zero —
// and at zero delay the buffer is empty much of the time, so it never recovers.
int periods = 300;
assertEquals(periods, emitted(true), "a frame per period keeps the device fed");
int gated = emitted(false);
assertTrue(gated < periods * 9 / 10,
"gating the pull on a non-empty buffer should visibly starve the device: " + gated);
}
/** Runs a jittery arrival schedule for 300 periods, counting frames actually emitted. */
private static int emitted(boolean pullEveryPeriod) {
JitterBuffer b = buffer();
Random random = new Random(13);
int next = 0;
int frames = 0;
for (int i = 0; i < 300; i++) {
// Arrivals clump and stall, as they do on a real connection.
if (random.nextInt(4) != 0) {
for (int j = 0, burst = random.nextInt(3); j < burst; j++) put(b, next++);
}
if (pullEveryPeriod || !b.isEmpty()) {
b.get(SPAN);
b.tick();
frames++;
}
}
return frames;
}
}

84
ts3-client/core/src/test/java/com/ts3client/audio/VoiceStreamTest.java
View File

@@ -5,6 +5,7 @@ import org.junit.jupiter.api.Test;
import java.util.ArrayList;
import java.util.List;
import java.util.Random;
import static org.junit.jupiter.api.Assertions.assertEquals;
import static org.junit.jupiter.api.Assertions.assertFalse;
@@ -21,6 +22,10 @@ class VoiceStreamTest {
return stream.offer(packetId, CodecType.OPUS_VOICE, FakeOpus.packet(value));
}
private void offerEnd(int packetId) {
stream.offer(packetId, CodecType.OPUS_VOICE, new byte[0]);
}
/** Pulls one frame and returns its first sample, or NaN if nothing was played. */
private float pull() {
int frames = stream.pull(out, 1, 1.0);
@@ -28,13 +33,11 @@ class VoiceStreamTest {
}
@Test
void holdsBackTheStartOfABurstThenPlaysInOrder() {
void playsPacketsInOrderWhateverOrderTheyArriveIn() {
assertTrue(offer(10, 1));
assertFalse(offer(12, 3));
assertFalse(offer(11, 2));
assertEquals(Float.NaN, pull());
assertEquals(Float.NaN, pull());
assertEquals(0.01f, pull());
assertEquals(0.02f, pull());
assertEquals(0.03f, pull());
@@ -45,13 +48,10 @@ class VoiceStreamTest {
void concealsALostPacket() {
offer(0, 1);
offer(2, 3);
pull();
pull();
assertEquals(0.01f, pull());
assertEquals(FakeOpus.CONCEALED, pull());
assertEquals(0.03f, pull());
assertEquals(List.of(VoiceFormat.FRAME_SIZE), opus.concealedFrameSizes,
"a lost packet is concealed at the length of the one before, not the longest Opus frame");
assertEquals(List.of(VoiceFormat.FRAME_SIZE), opus.concealedFrameSizes);
}
@Test
@@ -60,38 +60,54 @@ class VoiceStreamTest {
offer(1, 2);
pull();
pull();
pull();
pull();
assertFalse(offer(0, 9));
assertEquals(FakeOpus.CONCEALED, pull());
}
@Test
void endPacketFinishesTheBurst() {
void theTalkersStopEndsTheBurst() {
offer(0, 1);
stream.offer(1, CodecType.OPUS_VOICE, new byte[0]);
pull();
pull();
pull();
offerEnd(1);
assertEquals(0.01f, pull());
assertEquals(VoiceStream.IDLE, stream.pull(out, 1, 1.0));
assertFalse(stream.isPlaying());
assertEquals(List.of(true, false), talk);
assertEquals(1, opus.resets);
assertTrue(offer(2, 1), "the next packet starts a new burst");
}
@Test
void givesUpAfterTooMuchConcealment() {
void theNextBurstCarriesOnWhereTheLastStopped() {
offer(0, 1);
offerEnd(1);
pull();
pull();
assertTrue(offer(2, 2), "the next packet starts a new burst");
assertEquals(0f, pull(), "the stop's slot is the talker's own silence, not a loss");
assertEquals(0.02f, pull());
assertEquals(0, opus.resets, "the decoder keeps its state between bursts");
assertEquals(List.of(), opus.concealedFrameSizes);
}
@Test
void givesUpOnABurstWhoseStopWasLost() {
offer(0, 1);
pull();
int concealed = 0;
while (stream.pull(out, 1, 1.0) != VoiceStream.IDLE) concealed++;
assertEquals(10, concealed);
assertTrue(concealed > 0 && concealed < 10, "concealed " + concealed + " frames");
assertEquals(List.of(true, false), talk);
}
@Test
void playsALongPacketOneFrameAtATime() {
stream.offer(0, CodecType.OPUS_VOICE, new byte[]{40, 0});
offer(1, 50);
assertEquals(VoiceFormat.FRAME_SIZE, stream.pull(out, 1, 1.0));
assertEquals(0.4f, out[0]);
assertEquals(VoiceFormat.FRAME_SIZE, stream.pull(out, 1, 1.0));
assertEquals(0.4f, out[0]);
assertEquals(0.5f, pull());
}
@Test
void stopFromAnotherThreadEndsTheBurst() {
offer(0, 1);
@@ -103,11 +119,37 @@ class VoiceStreamTest {
@Test
void spreadsMonoOverStereoWithGain() {
offer(0, 50);
stream.pull(out, 2, 0.5);
stream.pull(out, 2, 0.5);
assertEquals(VoiceFormat.FRAME_SIZE, stream.pull(out, 2, 0.5));
assertEquals(0.25f, out[0]);
assertEquals(0.25f, out[1]);
assertEquals(0.25f, out[2 * VoiceFormat.FRAME_SIZE - 1]);
}
@Test
void neverHandsOutMoreThanOneFrame() {
for (int i = 0; i < 50; i++) offer(i, 1);
for (int i = 0; i < 50; i++) assertTrue(stream.pull(out, 1, 1.0) <= VoiceFormat.FRAME_SIZE);
}
/**
* Packets sent every 20 ms arrive up to 60 ms late. A fixed delay short of that conceals
* constantly; the adaptive one should learn the spread and stop concealing.
*/
@Test
void learnsADelayThatRidesOutJitter() {
Random random = new Random(1);
int packets = 1500;
long[] arrival = new long[packets];
for (int i = 0; i < packets; i++) arrival[i] = i * 20L + random.nextInt(61);
int next = 0;
int concealedLate = 0;
for (long now = 0; now < packets * 20L; now += 20) {
for (; next < packets && arrival[next] <= now; next++) offer(next, 1);
// arrivals are not in order, but whatever is due by now has been offered
float sample = pull();
if (now >= packets * 10L && sample == FakeOpus.CONCEALED) concealedLate++;
}
assertTrue(concealedLate < packets / 2 / 50, "concealed " + concealedLate + " frames in the second half");
}
}