Start counting NetEq stats after first packet is decoded.

A slight behavior change is that we only increment total samples received when GetAudio is successful.

Bug: webrtc:370424996
Change-Id: I8607418c179ca3bc22963b98792a9e8b9af2d451
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/364220
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43139}

modules/audio_coding/neteq/neteq_impl.cc

@@ -201,6 +201,7 @@ int NetEqImpl::GetAudio(AudioFrame* audio_frame,
   if (GetAudioInternal(audio_frame, action_override) != 0) {
     return kFail;
   }
+  stats_->IncreaseCounter(output_size_samples_, fs_hz_);
   RTC_DCHECK_EQ(
       audio_frame->sample_rate_hz_,
       rtc::dchecked_cast<int>(audio_frame->samples_per_channel_ * 100));
@@ -733,7 +734,6 @@ int NetEqImpl::GetAudioInternal(AudioFrame* audio_frame,
   bool play_dtmf;
   last_decoded_packet_infos_.clear();
   tick_timer_->Increment();
-  stats_->IncreaseCounter(output_size_samples_, fs_hz_);
 
   // Check for muted state.
   if (enable_muted_state_ && expand_->Muted() && packet_buffer_->Empty()) {

modules/audio_coding/neteq/neteq_impl_unittest.cc

@@ -384,6 +384,35 @@ TEST_F(NetEqImplTest, InsertPacket) {
                        /*receive_time=*/clock_.CurrentTime());
 }
 
+TEST_F(NetEqImplTest, CountStatsAfterFirstDecodedPacket) {
+  UseNoMocks();
+  CreateInstance();
+  const uint8_t kPayloadType = 17;  // Just an arbitrary number.
+  EXPECT_TRUE(neteq_->RegisterPayloadType(kPayloadType,
+                                          SdpAudioFormat("l16", 8000, 1)));
+  const size_t kPayloadLengthSamples = 80;
+  const size_t kPayloadLengthBytes = 2 * kPayloadLengthSamples;  // PCM 16-bit.
+  uint8_t payload[kPayloadLengthBytes] = {0};
+  RTPHeader rtp_header;
+  rtp_header.payloadType = kPayloadType;
+  rtp_header.sequenceNumber = 0x1234;
+  rtp_header.timestamp = 0x12345678;
+  rtp_header.ssrc = 0x87654321;
+  AudioFrame frame;
+  // Get audio a couple of times to make sure that samples received remains
+  // zero.
+  for (int i = 0; i < 3; ++i) {
+    neteq_->GetAudio(&frame);
+    EXPECT_EQ(neteq_->GetLifetimeStatistics().concealed_samples, 0u);
+    EXPECT_EQ(neteq_->GetLifetimeStatistics().total_samples_received, 0u);
+  }
+  neteq_->InsertPacket(rtp_header, payload, clock_.CurrentTime());
+  neteq_->GetAudio(&frame);
+  EXPECT_EQ(neteq_->GetLifetimeStatistics().concealed_samples, 0u);
+  EXPECT_EQ(neteq_->GetLifetimeStatistics().total_samples_received,
+            kPayloadLengthSamples);
+}
+
 TEST_F(NetEqImplTest, InsertPacketsUntilBufferIsFull) {
   UseNoMocks();
   CreateInstance();

modules/audio_coding/neteq/neteq_unittest.cc

@@ -81,7 +81,7 @@ TEST_F(NetEqDecodingTest, MAYBE_TestOpusBitExactness) {
       "cefd2de4adfa8f6a9b66a3639ad63c2f6779d0cd";
 
   const std::string network_stats_checksum =
-      "5f2c8e3dff9cff55dd7a9f4167939de001566d95|"
+      "06f6b9a86aeae6317fd25a36edf9ed16f35e798f|"
       "80ab17c17da030d4f2dfbf314ac44aacdadd7f0c";
 
   DecodeAndCompare(input_rtp_file, output_checksum, network_stats_checksum,
@@ -103,7 +103,7 @@ TEST_F(NetEqDecodingTest, MAYBE_TestOpusDtxBitExactness) {
       "5d13affec87bf4cc8c7667f0cd0d25e1ad09c7c3";
 
   const std::string network_stats_checksum =
-      "92b0fdcbf8bb9354d40140b7312f2fb76a078555";
+      "6af74a713749cc4343464718b6af54f1e5b06ad9";
 
   DecodeAndCompare(input_rtp_file, output_checksum, network_stats_checksum,
                    absl::GetFlag(FLAGS_gen_ref));
@@ -660,7 +660,7 @@ TEST_F(NetEqDecodingTestWithMutedState, MutedState) {
   // NetEqNetworkStatistics::expand_rate tells the fraction of samples that were
   // concealment samples, in Q14 (16384 = 100%) .The vast majority should be
   // concealment samples in this test.
-  EXPECT_GT(stats.expand_rate, 14000);
+  EXPECT_GT(stats.expand_rate, 13000);
   // And, it should be greater than the speech_expand_rate.
   EXPECT_GT(stats.expand_rate, stats.speech_expand_rate);
 }

modules/audio_coding/neteq/statistics_calculator.cc

@@ -155,6 +155,9 @@ void StatisticsCalculator::ResetMcu() {
 
 void StatisticsCalculator::ExpandedVoiceSamples(size_t num_samples,
                                                 bool is_new_concealment_event) {
+  if (!decoded_output_played_) {
+    return;
+  }
   expanded_speech_samples_ += num_samples;
   ConcealedSamplesCorrection(rtc::dchecked_cast<int>(num_samples), true);
   lifetime_stats_.concealment_events += is_new_concealment_event;
@@ -162,18 +165,27 @@ void StatisticsCalculator::ExpandedVoiceSamples(size_t num_samples,
 
 void StatisticsCalculator::ExpandedNoiseSamples(size_t num_samples,
                                                 bool is_new_concealment_event) {
+  if (!decoded_output_played_) {
+    return;
+  }
   expanded_noise_samples_ += num_samples;
   ConcealedSamplesCorrection(rtc::dchecked_cast<int>(num_samples), false);
   lifetime_stats_.concealment_events += is_new_concealment_event;
 }
 
 void StatisticsCalculator::ExpandedVoiceSamplesCorrection(int num_samples) {
+  if (!decoded_output_played_) {
+    return;
+  }
   expanded_speech_samples_ =
       AddIntToSizeTWithLowerCap(num_samples, expanded_speech_samples_);
   ConcealedSamplesCorrection(num_samples, true);
 }
 
 void StatisticsCalculator::ExpandedNoiseSamplesCorrection(int num_samples) {
+  if (!decoded_output_played_) {
+    return;
+  }
   expanded_noise_samples_ =
       AddIntToSizeTWithLowerCap(num_samples, expanded_noise_samples_);
   ConcealedSamplesCorrection(num_samples, false);
@@ -184,6 +196,9 @@ void StatisticsCalculator::DecodedOutputPlayed() {
 }
 
 void StatisticsCalculator::EndExpandEvent(int fs_hz) {
+  if (!decoded_output_played_) {
+    return;
+  }
   RTC_DCHECK_GE(lifetime_stats_.concealed_samples,
                 concealed_samples_at_event_end_);
   const int event_duration_ms =
@@ -201,6 +216,9 @@ void StatisticsCalculator::EndExpandEvent(int fs_hz) {
 
 void StatisticsCalculator::ConcealedSamplesCorrection(int num_samples,
                                                       bool is_voice) {
+  if (!decoded_output_played_) {
+    return;
+  }
   if (num_samples < 0) {
     // Store negative correction to subtract from future positive additions.
     // See also the function comment in the header file.
@@ -226,18 +244,27 @@ void StatisticsCalculator::ConcealedSamplesCorrection(int num_samples,
 }
 
 void StatisticsCalculator::PreemptiveExpandedSamples(size_t num_samples) {
+  if (!decoded_output_played_) {
+    return;
+  }
   preemptive_samples_ += num_samples;
   operations_and_state_.preemptive_samples += num_samples;
   lifetime_stats_.inserted_samples_for_deceleration += num_samples;
 }
 
 void StatisticsCalculator::AcceleratedSamples(size_t num_samples) {
+  if (!decoded_output_played_) {
+    return;
+  }
   accelerate_samples_ += num_samples;
   operations_and_state_.accelerate_samples += num_samples;
   lifetime_stats_.removed_samples_for_acceleration += num_samples;
 }
 
 void StatisticsCalculator::GeneratedNoiseSamples(size_t num_samples) {
+  if (!decoded_output_played_) {
+    return;
+  }
   lifetime_stats_.generated_noise_samples += num_samples;
 }
 
@@ -255,6 +282,9 @@ void StatisticsCalculator::SecondaryPacketsReceived(size_t num_packets) {
 }
 
 void StatisticsCalculator::IncreaseCounter(size_t num_samples, int fs_hz) {
+  if (!decoded_output_played_) {
+    return;
+  }
   const int time_step_ms =
       rtc::CheckedDivExact(static_cast<int>(1000 * num_samples), fs_hz);
   delayed_packet_outage_counter_.AdvanceClock(time_step_ms);
@@ -268,9 +298,13 @@ void StatisticsCalculator::IncreaseCounter(size_t num_samples, int fs_hz) {
   lifetime_stats_.total_samples_received += num_samples;
   expand_uma_logger_.UpdateSampleCounter(lifetime_stats_.concealed_samples,
                                          fs_hz);
-  speech_expand_uma_logger_.UpdateSampleCounter(
-      lifetime_stats_.concealed_samples -
-          lifetime_stats_.silent_concealed_samples,
+  uint64_t speech_concealed_samples = 0;
+  if (lifetime_stats_.concealed_samples >
+      lifetime_stats_.silent_concealed_samples) {
+    speech_concealed_samples = lifetime_stats_.concealed_samples -
+                               lifetime_stats_.silent_concealed_samples;
+  }
+  speech_expand_uma_logger_.UpdateSampleCounter(speech_concealed_samples,
                                                 fs_hz);
 }
 

modules/audio_coding/neteq/statistics_calculator_unittest.cc

@@ -17,6 +17,7 @@ namespace webrtc {
 TEST(LifetimeStatistics, TotalSamplesReceived) {
   TickTimer timer;
   StatisticsCalculator stats(&timer);
+  stats.DecodedOutputPlayed();
   for (int i = 0; i < 10; ++i) {
     stats.IncreaseCounter(480, 48000);  // 10 ms at 48 kHz.
   }
@@ -26,6 +27,7 @@ TEST(LifetimeStatistics, TotalSamplesReceived) {
 TEST(LifetimeStatistics, SamplesConcealed) {
   TickTimer timer;
   StatisticsCalculator stats(&timer);
+  stats.DecodedOutputPlayed();
   stats.ExpandedVoiceSamples(100, false);
   stats.ExpandedNoiseSamples(17, false);
   EXPECT_EQ(100u + 17u, stats.GetLifetimeStatistics().concealed_samples);
@@ -38,6 +40,7 @@ TEST(LifetimeStatistics, SamplesConcealed) {
 TEST(LifetimeStatistics, SamplesConcealedCorrection) {
   TickTimer timer;
   StatisticsCalculator stats(&timer);
+  stats.DecodedOutputPlayed();
   stats.ExpandedVoiceSamples(100, false);
   EXPECT_EQ(100u, stats.GetLifetimeStatistics().concealed_samples);
   stats.ExpandedVoiceSamplesCorrection(-10);
@@ -60,6 +63,7 @@ TEST(LifetimeStatistics, SamplesConcealedCorrection) {
 TEST(LifetimeStatistics, NoUpdateOnTimeStretch) {
   TickTimer timer;
   StatisticsCalculator stats(&timer);
+  stats.DecodedOutputPlayed();
   stats.ExpandedVoiceSamples(100, false);
   stats.AcceleratedSamples(4711);
   stats.PreemptiveExpandedSamples(17);
@@ -70,6 +74,7 @@ TEST(LifetimeStatistics, NoUpdateOnTimeStretch) {
 TEST(StatisticsCalculator, ExpandedSamplesCorrection) {
   TickTimer timer;
   StatisticsCalculator stats(&timer);
+  stats.DecodedOutputPlayed();
   NetEqNetworkStatistics stats_output;
   constexpr int kSampleRateHz = 48000;
   constexpr int k10MsSamples = kSampleRateHz / 100;
@@ -243,4 +248,19 @@ TEST(StatisticsCalculator, JitterBufferDelay) {
   EXPECT_EQ(lts.jitter_buffer_emitted_count, 960ul);
 }
 
+TEST(StatisticsCalculator, CountStatsAfterFirstDecodedPacket) {
+  TickTimer timer;
+  StatisticsCalculator stats(&timer);
+  stats.IncreaseCounter(/*num_samples=*/480, /*fs_hz=*/48000);
+  stats.ExpandedVoiceSamples(/*num_samples=*/480,
+                             /*is_new_concealment_event=*/true);
+  NetEqLifetimeStatistics lts = stats.GetLifetimeStatistics();
+  EXPECT_EQ(lts.total_samples_received, 0u);
+  EXPECT_EQ(lts.concealed_samples, 0u);
+  stats.DecodedOutputPlayed();
+  stats.IncreaseCounter(/*num_samples=*/480, /*fs_hz=*/48000);
+  lts = stats.GetLifetimeStatistics();
+  EXPECT_EQ(lts.total_samples_received, 480u);
+}
+
 }  // namespace webrtc