Prevent updating state in the delay manager if the packet was reordered.

Currently, if the last packet was reordered (e.g. due to retransmission) then the next packet's inter-arrival time will be estimated incorrectly due to the jump in sequence numbers. This change prevents that by not resetting the stopwatch on reordered packets.

This will also better estimate inter-arrival times when we have multiple reordered packets in a burst. Currently we would only measure the iat of the first reordered packet correctly and not the ones coming after it.

There is a slight risk introducing this: If we would receive an out of order packet far into the future (in sequence numbers) and then continue getting packets in the normal order, then we would not update the current sequence number for these and incorrectly estimate their inter-arrival times since they would all be considered reordered.

Change-Id: Ic938a37cbddf1cb9c30b610218f56794568d3d01
Bug: webrtc:10178
Reviewed-on: https://webrtc-review.googlesource.com/c/119949
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26572}

modules/audio_coding/neteq/decision_logic_unittest.cc

@@ -31,7 +31,7 @@ TEST(DecisionLogic, CreateAndDestroy) {
   TickTimer tick_timer;
   PacketBuffer packet_buffer(10, &tick_timer);
   DelayPeakDetector delay_peak_detector(&tick_timer, false);
-  DelayManager delay_manager(240, 0, &delay_peak_detector, &tick_timer);
+  DelayManager delay_manager(240, 0, false, &delay_peak_detector, &tick_timer);
   BufferLevelFilter buffer_level_filter;
   DecisionLogic* logic = DecisionLogic::Create(
       fs_hz, output_size_samples, false, &decoder_database, packet_buffer,
@@ -48,7 +48,7 @@ TEST(DecisionLogic, PostponeDecodingAfterExpansionSettings) {
   TickTimer tick_timer;
   PacketBuffer packet_buffer(10, &tick_timer);
   DelayPeakDetector delay_peak_detector(&tick_timer, false);
-  DelayManager delay_manager(240, 0, &delay_peak_detector, &tick_timer);
+  DelayManager delay_manager(240, 0, false, &delay_peak_detector, &tick_timer);
   BufferLevelFilter buffer_level_filter;
   {
     test::ScopedFieldTrials field_trial(

modules/audio_coding/neteq/delay_manager.cc

@@ -34,6 +34,8 @@ constexpr int kCumulativeSumDrift = 2;  // Drift term for cumulative sum
 // Steady-state forgetting factor for |iat_vector_|, 0.9993 in Q15.
 constexpr int kIatFactor_ = 32745;
 constexpr int kMaxIat = 64;  // Max inter-arrival time to register.
+constexpr int kMaxReorderedPackets =
+    10;  // Max number of consecutive reordered packets.
 
 absl::optional<int> GetForcedLimitProbability() {
   constexpr char kForceTargetDelayPercentileFieldTrial[] =
@@ -63,6 +65,7 @@ namespace webrtc {
 
 DelayManager::DelayManager(size_t max_packets_in_buffer,
                            int base_min_target_delay_ms,
+                           bool enable_rtx_handling,
                            DelayPeakDetector* peak_detector,
                            const TickTimer* tick_timer)
     : first_packet_received_(false),
@@ -85,7 +88,8 @@ DelayManager::DelayManager(size_t max_packets_in_buffer,
       last_pack_cng_or_dtmf_(1),
       frame_length_change_experiment_(
           field_trial::IsEnabled("WebRTC-Audio-NetEqFramelengthExperiment")),
-      forced_limit_probability_(GetForcedLimitProbability()) {
+      forced_limit_probability_(GetForcedLimitProbability()),
+      enable_rtx_handling_(enable_rtx_handling) {
   assert(peak_detector);  // Should never be NULL.
   RTC_DCHECK_GE(base_min_target_delay_ms_, 0);
   RTC_DCHECK_LE(minimum_delay_ms_, maximum_delay_ms_);
@@ -146,6 +150,7 @@ int DelayManager::Update(uint16_t sequence_number,
         rtc::saturated_cast<int>(1000 * packet_len_samp / sample_rate_hz);
   }
 
+  bool reordered = false;
   if (packet_len_ms > 0) {
     // Cannot update statistics unless |packet_len_ms| is valid.
     // Calculate inter-arrival time (IAT) in integer "packet times"
@@ -158,7 +163,6 @@ int DelayManager::Update(uint16_t sequence_number,
     }
 
     // Check for discontinuous packet sequence and re-ordering.
-    bool reordered = false;
     if (IsNewerSequenceNumber(sequence_number, last_seq_no_ + 1)) {
       // Compensate for gap in the sequence numbers. Reduce IAT with the
       // expected extra time due to lost packets, but ensure that the IAT is
@@ -183,6 +187,12 @@ int DelayManager::Update(uint16_t sequence_number,
     LimitTargetLevel();
   }  // End if (packet_len_ms > 0).
 
+  if (enable_rtx_handling_ && reordered &&
+      num_reordered_packets_ < kMaxReorderedPackets) {
+    ++num_reordered_packets_;
+    return 0;
+  }
+  num_reordered_packets_ = 0;
   // Prepare for next packet arrival.
   packet_iat_stopwatch_ = tick_timer_->GetNewStopwatch();
   last_seq_no_ = sequence_number;

modules/audio_coding/neteq/delay_manager.h

@@ -36,6 +36,7 @@ class DelayManager {
   // PeakDetector object to the DelayManager.
   DelayManager(size_t max_packets_in_buffer,
                int base_min_target_delay_ms,
+               bool enable_rtx_handling,
                DelayPeakDetector* peak_detector,
                const TickTimer* tick_timer);
 
@@ -177,6 +178,8 @@ class DelayManager {
   int last_pack_cng_or_dtmf_;
   const bool frame_length_change_experiment_;
   const absl::optional<int> forced_limit_probability_;
+  const bool enable_rtx_handling_;
+  int num_reordered_packets_ = 0;  // Number of consecutive reordered packets.
 
   RTC_DISALLOW_COPY_AND_ASSIGN(DelayManager);
 };

modules/audio_coding/neteq/delay_manager_unittest.cc

@@ -46,6 +46,7 @@ class DelayManagerTest : public ::testing::Test {
   MockDelayPeakDetector detector_;
   uint16_t seq_no_;
   uint32_t ts_;
+  bool enable_rtx_handling_ = false;
 };
 
 DelayManagerTest::DelayManagerTest()
@@ -60,8 +61,8 @@ void DelayManagerTest::SetUp() {
 
 void DelayManagerTest::RecreateDelayManager() {
   EXPECT_CALL(detector_, Reset()).Times(1);
-  dm_.reset(new DelayManager(kMaxNumberOfPackets, kMinDelayMs, &detector_,
-                             &tick_timer_));
+  dm_.reset(new DelayManager(kMaxNumberOfPackets, kMinDelayMs,
+                             enable_rtx_handling_, &detector_, &tick_timer_));
 }
 
 void DelayManagerTest::SetPacketAudioLength(int lengt_ms) {
@@ -320,6 +321,31 @@ TEST_F(DelayManagerTest, UpdateReorderedPacket) {
   EXPECT_EQ(0, dm_->Update(seq_no_ - 1, ts_ - kFrameSizeMs, kFs));
 }
 
+TEST_F(DelayManagerTest, EnableRtxHandling) {
+  enable_rtx_handling_ = true;
+  RecreateDelayManager();
+
+  // Insert first packet.
+  SetPacketAudioLength(kFrameSizeMs);
+  InsertNextPacket();
+
+  // Insert reordered packet.
+  // TODO(jakobi): Test estimated inter-arrival time by mocking the histogram
+  // instead of checking the call to the peak detector.
+  EXPECT_CALL(detector_, Update(3, true, _));
+  EXPECT_EQ(0, dm_->Update(seq_no_ - 3, ts_ - 3 * kFrameSizeMs, kFs));
+
+  // Insert another reordered packet.
+  EXPECT_CALL(detector_, Update(2, true, _));
+  EXPECT_EQ(0, dm_->Update(seq_no_ - 2, ts_ - 2 * kFrameSizeMs, kFs));
+
+  // Insert the next packet in order and verify that the inter-arrival time is
+  // estimated correctly.
+  IncreaseTime(kFrameSizeMs);
+  EXPECT_CALL(detector_, Update(1, false, _));
+  InsertNextPacket();
+}
+
 // Tests that skipped sequence numbers (simulating empty packets) are handled
 // correctly.
 TEST_F(DelayManagerTest, EmptyPacketsReported) {

modules/audio_coding/neteq/mock/mock_delay_manager.h

@@ -21,10 +21,12 @@ class MockDelayManager : public DelayManager {
  public:
   MockDelayManager(size_t max_packets_in_buffer,
                    int base_min_target_delay_ms,
+                   bool enable_rtx_handling,
                    DelayPeakDetector* peak_detector,
                    const TickTimer* tick_timer)
       : DelayManager(max_packets_in_buffer,
                      base_min_target_delay_ms,
+                     enable_rtx_handling,
                      peak_detector,
                      tick_timer) {}
   virtual ~MockDelayManager() { Die(); }

modules/audio_coding/neteq/neteq_impl.cc

@@ -65,6 +65,7 @@ NetEqImpl::Dependencies::Dependencies(
           new DelayPeakDetector(tick_timer.get(), config.enable_rtx_handling)),
       delay_manager(new DelayManager(config.max_packets_in_buffer,
                                      config.min_delay_ms,
+                                     config.enable_rtx_handling,
                                      delay_peak_detector.get(),
                                      tick_timer.get())),
       dtmf_buffer(new DtmfBuffer(config.sample_rate_hz)),

modules/audio_coding/neteq/neteq_impl_unittest.cc

@@ -97,7 +97,7 @@ class NetEqImplTest : public ::testing::Test {
     if (use_mock_delay_manager_) {
       std::unique_ptr<MockDelayManager> mock(new MockDelayManager(
           config_.max_packets_in_buffer, config_.min_delay_ms,
-          delay_peak_detector_, tick_timer_));
+          config_.enable_rtx_handling, delay_peak_detector_, tick_timer_));
       mock_delay_manager_ = mock.get();
       EXPECT_CALL(*mock_delay_manager_, set_streaming_mode(false)).Times(1);
       deps.delay_manager = std::move(mock);

modules/audio_coding/neteq/tools/neteq_test.cc

@@ -136,6 +136,7 @@ NetEqTest::SimulationStepResult NetEqTest::RunToNextGetAudio() {
                 : -1;
         *text_log_ << "Packet   - wallclock: " << std::setw(5) << time_now_ms
                    << ", delta wc: " << std::setw(4) << delta_wallclock
+                   << ", seq_no: " << packet_data->header.sequenceNumber
                    << ", timestamp: " << std::setw(10)
                    << packet_data->header.timestamp
                    << ", delta ts: " << std::setw(4) << delta_timestamp