|
| 1 | +package synchronizer_test |
| 2 | + |
| 3 | +import ( |
| 4 | + "testing" |
| 5 | + "time" |
| 6 | + |
| 7 | + "github.com/pion/rtcp" |
| 8 | + "github.com/pion/rtp" |
| 9 | + "github.com/pion/webrtc/v4" |
| 10 | + "github.com/stretchr/testify/require" |
| 11 | + |
| 12 | + "github.com/livekit/mediatransportutil" |
| 13 | + "github.com/livekit/server-sdk-go/v2/pkg/synchronizer" |
| 14 | + "github.com/livekit/server-sdk-go/v2/pkg/synchronizer/synchronizerfakes" |
| 15 | +) |
| 16 | + |
| 17 | +const timeTolerance = time.Millisecond * 10 |
| 18 | + |
| 19 | +// ---------- helpers ---------- |
| 20 | + |
| 21 | +func near(t *testing.T, got, want, tol time.Duration) { |
| 22 | + t.Helper() |
| 23 | + d := got - want |
| 24 | + if d < 0 { |
| 25 | + d = -d |
| 26 | + } |
| 27 | + require.LessOrEqualf(t, d, tol, "got %v, want ~%v (±%v)", got, want, tol) |
| 28 | +} |
| 29 | + |
| 30 | +func pkt(ts uint32) *rtp.Packet { |
| 31 | + return &rtp.Packet{Header: rtp.Header{Timestamp: ts}} |
| 32 | +} |
| 33 | + |
| 34 | +func fakeAudio48k(ssrc uint32) *synchronizerfakes.FakeTrackRemote { |
| 35 | + f := &synchronizerfakes.FakeTrackRemote{} |
| 36 | + f.IDReturns("audio-1") |
| 37 | + f.KindReturns(webrtc.RTPCodecTypeAudio) |
| 38 | + f.SSRCReturns(webrtc.SSRC(ssrc)) |
| 39 | + f.CodecReturns(webrtc.RTPCodecParameters{ |
| 40 | + RTPCodecCapability: webrtc.RTPCodecCapability{ |
| 41 | + MimeType: webrtc.MimeTypeOpus, |
| 42 | + ClockRate: 48000, |
| 43 | + }, |
| 44 | + }) |
| 45 | + return f |
| 46 | +} |
| 47 | + |
| 48 | +func fakeVideo90k(ssrc uint32) *synchronizerfakes.FakeTrackRemote { |
| 49 | + f := &synchronizerfakes.FakeTrackRemote{} |
| 50 | + f.IDReturns("video-1") |
| 51 | + f.KindReturns(webrtc.RTPCodecTypeVideo) |
| 52 | + f.SSRCReturns(webrtc.SSRC(ssrc)) |
| 53 | + f.CodecReturns(webrtc.RTPCodecParameters{ |
| 54 | + RTPCodecCapability: webrtc.RTPCodecCapability{ |
| 55 | + MimeType: webrtc.MimeTypeVP8, // codec value not important here |
| 56 | + ClockRate: 90000, |
| 57 | + }, |
| 58 | + }) |
| 59 | + return f |
| 60 | +} |
| 61 | + |
| 62 | +// ---------- tests ---------- |
| 63 | + |
| 64 | +// Initialize + same timestamp returns previous adjusted value (near zero right after init) |
| 65 | +func TestInitialize_AndSameTimestamp(t *testing.T) { |
| 66 | + s := synchronizer.NewSynchronizerWithOptions(synchronizer.WithMaxTsDiff(50 * time.Millisecond)) |
| 67 | + tr := fakeAudio48k(0xA0010001) |
| 68 | + |
| 69 | + ts := uint32(1_000_000) |
| 70 | + tsync := s.AddTrack(tr, "p1") |
| 71 | + |
| 72 | + tsync.Initialize(pkt(ts)) |
| 73 | + |
| 74 | + adj0, err := tsync.GetPTS(pkt(ts)) |
| 75 | + require.NoError(t, err) |
| 76 | + near(t, adj0, 0, timeTolerance) |
| 77 | + |
| 78 | + adj1, err := tsync.GetPTS(pkt(ts)) |
| 79 | + require.NoError(t, err) |
| 80 | + require.Equal(t, adj0, adj1) |
| 81 | +} |
| 82 | + |
| 83 | +// 20 ms RTP step at 48 kHz → ~20 ms adjusted PTS (with small tolerance) |
| 84 | +func TestMonotonicPTS_SmallRTPDelta(t *testing.T) { |
| 85 | + s := synchronizer.NewSynchronizerWithOptions(synchronizer.WithMaxTsDiff(50 * time.Millisecond)) |
| 86 | + tr := fakeAudio48k(0xA0010002) |
| 87 | + |
| 88 | + ts0 := uint32(500_000) |
| 89 | + delta20ms := uint32(48000 * 20 / 1000) // 960 ticks |
| 90 | + |
| 91 | + tsync := s.AddTrack(tr, "p1") |
| 92 | + tsync.Initialize(pkt(ts0)) |
| 93 | + |
| 94 | + // establish startTime |
| 95 | + _, _ = tsync.GetPTS(pkt(ts0)) |
| 96 | + |
| 97 | + adj, err := tsync.GetPTS(pkt(ts0 + delta20ms)) |
| 98 | + require.NoError(t, err) |
| 99 | + near(t, adj, 20*time.Millisecond, timeTolerance) |
| 100 | +} |
| 101 | + |
| 102 | +// Large RTP jump with tight MaxTsDiff should reset to small estimatedPTS (not ~2s) |
| 103 | +func TestUnacceptableDrift_ResetsToEstimatedPTS(t *testing.T) { |
| 104 | + s := synchronizer.NewSynchronizerWithOptions(synchronizer.WithMaxTsDiff(10 * time.Millisecond)) |
| 105 | + tr := fakeAudio48k(0xA0010003) |
| 106 | + |
| 107 | + ts0 := uint32(777_000) |
| 108 | + delta2s := uint32(48000 * 2) // 96,000 ticks |
| 109 | + |
| 110 | + tsync := s.AddTrack(tr, "p1") |
| 111 | + tsync.Initialize(pkt(ts0)) |
| 112 | + |
| 113 | + // establish startTime |
| 114 | + _, _ = tsync.GetPTS(pkt(ts0)) |
| 115 | + |
| 116 | + adj, err := tsync.GetPTS(pkt(ts0 + delta2s)) |
| 117 | + require.NoError(t, err) |
| 118 | + require.LessOrEqual(t, adj, 150*time.Millisecond, "expected reset to small estimatedPTS, not ~2s") |
| 119 | +} |
| 120 | + |
| 121 | +func TestOnSenderReport_SlewsTowardDesiredOffset(t *testing.T) { |
| 122 | + const ( |
| 123 | + maxAdjustment = 5 * time.Millisecond |
| 124 | + ts0 = uint32(900_000) |
| 125 | + stepRTP = uint32(48000 * 20 / 1000) // 20 ms @ 48 kHz |
| 126 | + stepDur = 20 * time.Millisecond |
| 127 | + desired = 50 * time.Millisecond // target offset from SR |
| 128 | + ) |
| 129 | + |
| 130 | + s := synchronizer.NewSynchronizerWithOptions(synchronizer.WithMaxTsDiff(1 * time.Second)) |
| 131 | + tr := fakeAudio48k(0xA0010004) |
| 132 | + |
| 133 | + tsync := s.AddTrack(tr, "p1") |
| 134 | + tsync.Initialize(pkt(ts0)) |
| 135 | + |
| 136 | + // Anchor wall-clock just before startTime is set. |
| 137 | + t0 := time.Now() |
| 138 | + _, _ = tsync.GetPTS(pkt(ts0)) // sets startTime ≈ t0 |
| 139 | + |
| 140 | + cur := ts0 + stepRTP |
| 141 | + baseAdj, _ := tsync.GetPTS(pkt(cur)) // baseline adjusted PTS with 20ms content |
| 142 | + |
| 143 | + // Craft SR so offset ≈ desired: |
| 144 | + // offset := NTP - (startTime + pts_at_SR) |
| 145 | + // pts_at_SR here is 20ms, startTime ≈ t0 → set NTP to t0 + 20ms + desired |
| 146 | + ntp := mediatransportutil.ToNtpTime(t0.Add(stepDur + desired)) |
| 147 | + tsync.OnSenderReport(func(d time.Duration) {}) |
| 148 | + s.OnRTCP(&rtcp.SenderReport{ |
| 149 | + SSRC: uint32(tr.SSRC()), |
| 150 | + NTPTime: uint64(ntp), |
| 151 | + RTPTime: cur, |
| 152 | + }) |
| 153 | + |
| 154 | + // Converge in N = ceil(desired / 5ms) steps (5ms maxAdjustment) |
| 155 | + N := int((desired + 5*time.Millisecond - 1) / (5 * time.Millisecond)) |
| 156 | + |
| 157 | + for i := 0; i < N; i++ { |
| 158 | + cur += stepRTP |
| 159 | + _, err := tsync.GetPTS(pkt(cur)) |
| 160 | + require.NoError(t, err) |
| 161 | + } |
| 162 | + |
| 163 | + // After N steps, total adjusted delta over base should be: |
| 164 | + // content progression (N * 20ms) + desired (slew) |
| 165 | + finalAdj, err := tsync.GetPTS(pkt(cur)) // same TS → returns last adjusted |
| 166 | + require.NoError(t, err) |
| 167 | + |
| 168 | + gotDelta := finalAdj - baseAdj |
| 169 | + wantDelta := time.Duration(N)*stepDur + desired |
| 170 | + |
| 171 | + near(t, gotDelta, wantDelta, timeTolerance) |
| 172 | +} |
| 173 | + |
| 174 | +// Regression: late video start (~2s) + tiny SR offset (~10ms) must NOT emit a big negative drift |
| 175 | +func TestOnSenderReport_LateVideoStart_SmallSROffset_NoHugeNegativeDrift(t *testing.T) { |
| 176 | + const ( |
| 177 | + lateStart = 2 * time.Second |
| 178 | + srOffset = 50 * time.Millisecond |
| 179 | + stepSlew = 5 * time.Millisecond // TrackSynchronizer's maxAdjustment |
| 180 | + ) |
| 181 | + |
| 182 | + s := synchronizer.NewSynchronizerWithOptions(synchronizer.WithMaxTsDiff(2 * time.Second)) |
| 183 | + |
| 184 | + // 1) Audio publishes immediately → establishes startedAt |
| 185 | + audio := fakeAudio48k(0xA0010005) |
| 186 | + tsA0 := uint32(1_000_000) |
| 187 | + aSync := s.AddTrack(audio, "p1") |
| 188 | + aSync.Initialize(pkt(tsA0)) |
| 189 | + _, _ = aSync.GetPTS(pkt(tsA0)) |
| 190 | + |
| 191 | + // Simulate a real late video publish |
| 192 | + time.Sleep(lateStart) |
| 193 | + |
| 194 | + // 2) Video publishes later |
| 195 | + video := fakeVideo90k(0x00BEEF01) |
| 196 | + tsV0 := uint32(2_000_000) |
| 197 | + vSync := s.AddTrack(video, "p1") |
| 198 | + vSync.Initialize(pkt(tsV0)) |
| 199 | + _, _ = vSync.GetPTS(pkt(tsV0)) |
| 200 | + |
| 201 | + // 3) First SR: small positive drift |
| 202 | + ntp := mediatransportutil.ToNtpTime(time.Now().Add(srOffset)) |
| 203 | + var observedDrift time.Duration |
| 204 | + vSync.OnSenderReport(func(d time.Duration) { observedDrift = d }) |
| 205 | + |
| 206 | + s.OnRTCP(&rtcp.SenderReport{ |
| 207 | + SSRC: uint32(video.SSRC()), |
| 208 | + NTPTime: uint64(ntp), |
| 209 | + RTPTime: tsV0, // equals lastTS → SR uses lastPTS at tsV0 |
| 210 | + }) |
| 211 | + |
| 212 | + near(t, observedDrift, srOffset, timeTolerance) |
| 213 | + |
| 214 | + // baseline adjusted PTS at the SR moment (same TS => returns last adjusted) |
| 215 | + baseAdj, err := vSync.GetPTS(pkt(tsV0)) |
| 216 | + require.NoError(t, err) |
| 217 | + |
| 218 | + step33ms := uint32(90000 * 33 / 1000) // ~33 ms per 30fps frame at 90 kHz |
| 219 | + stepDur := 33 * time.Millisecond |
| 220 | + |
| 221 | + // Converge in N = ceil(srOffset / stepSlew) steps (50ms / 5ms = 10) |
| 222 | + N := int((srOffset + stepSlew - 1) / stepSlew) |
| 223 | + |
| 224 | + cur := tsV0 |
| 225 | + var adj time.Duration |
| 226 | + |
| 227 | + // Drive N frames to converge |
| 228 | + for i := 1; i <= N; i++ { |
| 229 | + cur += step33ms |
| 230 | + adj, err = vSync.GetPTS(pkt(cur)) |
| 231 | + require.NoError(t, err) |
| 232 | + } |
| 233 | + |
| 234 | + // After N steps, the extra beyond content cadence should be ~srOffset |
| 235 | + gotDelta := adj - baseAdj |
| 236 | + wantDelta := time.Duration(N)*stepDur + srOffset |
| 237 | + near(t, gotDelta, wantDelta, timeTolerance) |
| 238 | + |
| 239 | + // Now push more frames and ensure we DON'T keep slewing (stays near srOffset) |
| 240 | + const extraFrames = 8 |
| 241 | + for i := 1; i <= extraFrames; i++ { |
| 242 | + cur += step33ms |
| 243 | + adj, err = vSync.GetPTS(pkt(cur)) |
| 244 | + require.NoError(t, err) |
| 245 | + |
| 246 | + // Extra slew measured from the SR moment: |
| 247 | + totalContent := time.Duration(N+i) * stepDur |
| 248 | + extra := (adj - baseAdj) - totalContent |
| 249 | + |
| 250 | + // Stay within a small band around srOffset (no steady growth) |
| 251 | + near(t, extra, srOffset, timeTolerance) |
| 252 | + } |
| 253 | +} |
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