Added RTCMediaStreamTrackStats.concealmentEvents

The number of concealment events. This counter increases every time a concealed sample is synthesized after a non-concealed sample. That is, multiple consecutive concealed samples will increase the concealedSamples count multiple times but is a single concealment event. Bug: webrtc:8246 Change-Id: I7ef404edab765218b1f11e3128072c5391e83049 Reviewed-on: https://webrtc-review.googlesource.com/1221 Commit-Queue: Gustaf Ullberg <[email protected]> Reviewed-by: Henrik Andreassson <[email protected]> Reviewed-by: Fredrik Solenberg <[email protected]> Reviewed-by: Henrik Lundin <[email protected]> Reviewed-by: Taylor Brandstetter <[email protected]> Reviewed-by: Henrik Boström <[email protected]> Cr-Commit-Position: refs/heads/master@{#19881}
kellaJL · Sep 18, 2017 · 9a2e906 · 9a2e906
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Showing 20 changed files with 90 additions and 36 deletions.
diff --git a/api/stats/rtcstats_objects.h b/api/stats/rtcstats_objects.h
@@ -278,6 +278,7 @@ class RTCMediaStreamTrackStats final : public RTCStats {
   RTCStatsMember<uint64_t> total_samples_received;
   RTCStatsMember<double> total_samples_duration;
   RTCStatsMember<uint64_t> concealed_samples;
+  RTCStatsMember<uint64_t> concealment_events;
 };
 
 // https://w3c.github.io/webrtc-stats/#pcstats-dict*

diff --git a/api/statstypes.cc b/api/statstypes.cc
@@ -373,6 +373,8 @@ const char* StatsReport::Value::display_name() const {
       return "bytesSent";
     case kStatsValueNameConcealedSamples:
       return "concealedSamples";
+    case kStatsValueNameConcealmentEvents:
+      return "concealmentEvents";
     case kStatsValueNamePacketsSent:
       return "packetsSent";
     case kStatsValueNameBytesReceived:

diff --git a/api/statstypes.h b/api/statstypes.h
@@ -105,6 +105,7 @@ class StatsReport {
     kStatsValueNameBytesSent,
     kStatsValueNameCodecImplementationName,
     kStatsValueNameConcealedSamples,
+    kStatsValueNameConcealmentEvents,
     kStatsValueNameDataChannelId,
     kStatsValueNameFramesDecoded,
     kStatsValueNameFramesEncoded,

diff --git a/audio/audio_receive_stream.cc b/audio/audio_receive_stream.cc
@@ -196,6 +196,7 @@ webrtc::AudioReceiveStream::Stats AudioReceiveStream::GetStats() const {
   stats.jitter_buffer_preferred_ms = ns.preferredBufferSize;
   stats.total_samples_received = ns.totalSamplesReceived;
   stats.concealed_samples = ns.concealedSamples;
+  stats.concealment_events = ns.concealmentEvents;
   stats.expand_rate = Q14ToFloat(ns.currentExpandRate);
   stats.speech_expand_rate = Q14ToFloat(ns.currentSpeechExpandRate);
   stats.secondary_decoded_rate = Q14ToFloat(ns.currentSecondaryDecodedRate);

diff --git a/audio/audio_receive_stream_unittest.cc b/audio/audio_receive_stream_unittest.cc
@@ -64,9 +64,9 @@ const CallStatistics kCallStats = {
     345,  678,  901, 234, -12, 3456, 7890, 567, 890, 123};
 const CodecInst kCodecInst = {
     123, "codec_name_recv", 96000, -187, 0, -103};
-const NetworkStatistics kNetworkStats = {123, 456, false, 789012, 3456, 0, {},
-                                         789, 12,  345,   678,    901,  0, -1,
-                                         -1,  -1,  -1,    -1,     0};
+const NetworkStatistics kNetworkStats = {123, 456, false, 789012, 3456, 123, 0,
+                                         {},  789, 12,    345,    678,  901, 0,
+                                         -1,  -1,  -1,    -1,     -1,   0};
 const AudioDecodingCallStats kAudioDecodeStats = MakeAudioDecodeStatsForTest();
 
 struct ConfigHelper {
@@ -322,6 +322,7 @@ TEST(AudioReceiveStreamTest, GetStats) {
   EXPECT_EQ(kNetworkStats.totalSamplesReceived, stats.total_samples_received);
   EXPECT_EQ(kTotalOutputDuration, stats.total_output_duration);
   EXPECT_EQ(kNetworkStats.concealedSamples, stats.concealed_samples);
+  EXPECT_EQ(kNetworkStats.concealmentEvents, stats.concealment_events);
   EXPECT_EQ(Q14ToFloat(kNetworkStats.currentExpandRate), stats.expand_rate);
   EXPECT_EQ(Q14ToFloat(kNetworkStats.currentSpeechExpandRate),
             stats.speech_expand_rate);

diff --git a/call/audio_receive_stream.h b/call/audio_receive_stream.h
@@ -50,18 +50,14 @@ class AudioReceiveStream {
     uint32_t jitter_buffer_preferred_ms = 0;
     uint32_t delay_estimate_ms = 0;
     int32_t audio_level = -1;
-    // See description of "totalAudioEnergy" in the WebRTC stats spec:
-    // https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaudioenergy
+    // Stats below correspond to similarly-named fields in the WebRTC stats
+    // spec. https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats
     double total_output_energy = 0.0;
-    // See description of "totalSamplesReceived" in the WebRTC stats spec:
-    // https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalsamplesreceived
     uint64_t total_samples_received = 0;
-    // See description of "totalSamplesDuration" in the WebRTC stats spec:
-    // https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalsamplesduration
     double total_output_duration = 0.0;
-    // See description of "concealedSamples" in the WebRTC stats spec:
-    // https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-concealedsamples
     uint64_t concealed_samples = 0;
+    uint64_t concealment_events = 0;
+    // Stats below DO NOT correspond directly to anything in the WebRTC stats
     float expand_rate = 0.0f;
     float speech_expand_rate = 0.0f;
     float secondary_decoded_rate = 0.0f;

diff --git a/common_types.h b/common_types.h
@@ -375,6 +375,10 @@ struct NetworkStatistics {
   // conceal packet loss.
   // https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-concealedsamples
   uint64_t concealedSamples;
+  // Number of times a concealed sample is synthesized after a non-concealed
+  // sample.
+  // https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-concealmentevents
+  uint64_t concealmentEvents;
   // Loss rate (network + late); fraction between 0 and 1, scaled to Q14.
   uint16_t currentPacketLossRate;
   // Late loss rate; fraction between 0 and 1, scaled to Q14.

diff --git a/media/base/mediachannel.h b/media/base/mediachannel.h
@@ -657,6 +657,7 @@ struct VoiceReceiverInfo : public MediaReceiverInfo {
         total_samples_received(0),
         total_output_duration(0.0),
         concealed_samples(0),
+        concealment_events(0),
         expand_rate(0),
         speech_expand_rate(0),
         secondary_decoded_rate(0),
@@ -678,18 +679,14 @@ struct VoiceReceiverInfo : public MediaReceiverInfo {
   int jitter_buffer_preferred_ms;
   int delay_estimate_ms;
   int audio_level;
-  // See description of "totalAudioEnergy" in the WebRTC stats spec:
-  // https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaudioenergy
+  // Stats below correspond to similarly-named fields in the WebRTC stats spec.
+  // https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats
   double total_output_energy;
-  // See description of "totalSamplesReceived" in the WebRTC stats spec:
-  // https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalsamplesreceived
   uint64_t total_samples_received;
-  // See description of "totalSamplesDuration" in the WebRTC stats spec:
-  // https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalsamplesduration
   double total_output_duration;
-  // See description of "concealedSamples" in the WebRTC stats spec:
-  // https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-concealedsamples
   uint64_t concealed_samples;
+  uint64_t concealment_events;
+  // Stats below DO NOT correspond directly to anything in the WebRTC stats
   // fraction of synthesized audio inserted through expansion.
   float expand_rate;
   // fraction of synthesized speech inserted through expansion.

diff --git a/media/engine/webrtcvoiceengine.cc b/media/engine/webrtcvoiceengine.cc
@@ -2284,6 +2284,7 @@ bool WebRtcVoiceMediaChannel::GetStats(VoiceMediaInfo* info) {
     rinfo.total_samples_received = stats.total_samples_received;
     rinfo.total_output_duration = stats.total_output_duration;
     rinfo.concealed_samples = stats.concealed_samples;
+    rinfo.concealment_events = stats.concealment_events;
     rinfo.expand_rate = stats.expand_rate;
     rinfo.speech_expand_rate = stats.speech_expand_rate;
     rinfo.secondary_decoded_rate = stats.secondary_decoded_rate;

diff --git a/media/engine/webrtcvoiceengine_unittest.cc b/media/engine/webrtcvoiceengine_unittest.cc
@@ -622,6 +622,7 @@ class WebRtcVoiceEngineTestFake : public testing::Test {
     stats.audio_level = 1234;
     stats.total_samples_received = 5678901;
     stats.concealed_samples = 234;
+    stats.concealment_events = 12;
     stats.expand_rate = 5.67f;
     stats.speech_expand_rate = 8.90f;
     stats.secondary_decoded_rate = 1.23f;
@@ -661,6 +662,7 @@ class WebRtcVoiceEngineTestFake : public testing::Test {
     EXPECT_EQ(info.audio_level, stats.audio_level);
     EXPECT_EQ(info.total_samples_received, stats.total_samples_received);
     EXPECT_EQ(info.concealed_samples, stats.concealed_samples);
+    EXPECT_EQ(info.concealment_events, stats.concealment_events);
     EXPECT_EQ(info.expand_rate, stats.expand_rate);
     EXPECT_EQ(info.speech_expand_rate, stats.speech_expand_rate);
     EXPECT_EQ(info.secondary_decoded_rate, stats.secondary_decoded_rate);

diff --git a/modules/audio_coding/acm2/acm_receiver.cc b/modules/audio_coding/acm2/acm_receiver.cc
@@ -336,6 +336,7 @@ void AcmReceiver::GetNetworkStatistics(NetworkStatistics* acm_stat) {
   NetEqLifetimeStatistics neteq_lifetime_stat = neteq_->GetLifetimeStatistics();
   acm_stat->totalSamplesReceived = neteq_lifetime_stat.total_samples_received;
   acm_stat->concealedSamples = neteq_lifetime_stat.concealed_samples;
+  acm_stat->concealmentEvents = neteq_lifetime_stat.concealment_events;
 }
 
 int AcmReceiver::DecoderByPayloadType(uint8_t payload_type,

diff --git a/modules/audio_coding/neteq/include/neteq.h b/modules/audio_coding/neteq/include/neteq.h
@@ -61,13 +61,11 @@ struct NetEqNetworkStatistics {
 // NetEq statistics that persist over the lifetime of the class.
 // These metrics are never reset.
 struct NetEqLifetimeStatistics {
-  // Total number of audio samples received, including synthesized samples.
-  // https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalsamplesreceived
+  // Stats below correspond to similarly-named fields in the WebRTC stats spec.
+  // https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats
   uint64_t total_samples_received = 0;
-  // Total number of inbound audio samples that are based on synthesized data to
-  // conceal packet loss.
-  // https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-concealedsamples
   uint64_t concealed_samples = 0;
+  uint64_t concealment_events = 0;
 };
 
 enum NetEqPlayoutMode {

diff --git a/modules/audio_coding/neteq/neteq_impl.cc b/modules/audio_coding/neteq/neteq_impl.cc
@@ -847,7 +847,7 @@ int NetEqImpl::GetAudioInternal(AudioFrame* audio_frame, bool* muted) {
             : timestamp_scaler_->ToExternal(playout_timestamp_) -
                   static_cast<uint32_t>(audio_frame->samples_per_channel_);
     audio_frame->num_channels_ = sync_buffer_->Channels();
-    stats_.ExpandedNoiseSamples(output_size_samples_);
+    stats_.ExpandedNoiseSamples(output_size_samples_, false);
     *muted = true;
     return 0;
   }
@@ -1573,14 +1573,15 @@ int NetEqImpl::DoExpand(bool play_dtmf) {
     algorithm_buffer_->Clear();
     int return_value = expand_->Process(algorithm_buffer_.get());
     size_t length = algorithm_buffer_->Size();
+    bool is_new_concealment_event = (last_mode_ != kModeExpand);
 
     // Update in-call and post-call statistics.
     if (expand_->MuteFactor(0) == 0) {
       // Expand operation generates only noise.
-      stats_.ExpandedNoiseSamples(length);
+      stats_.ExpandedNoiseSamples(length, is_new_concealment_event);
     } else {
       // Expand operation generates more than only noise.
-      stats_.ExpandedVoiceSamples(length);
+      stats_.ExpandedVoiceSamples(length, is_new_concealment_event);
     }
 
     last_mode_ = kModeExpand;

diff --git a/modules/audio_coding/neteq/neteq_unittest.cc b/modules/audio_coding/neteq/neteq_unittest.cc
@@ -1634,4 +1634,38 @@ TEST_F(NetEqDecodingTest, LastDecodedTimestampsTwoDecoded) {
             neteq_->LastDecodedTimestamps());
 }
 
+TEST_F(NetEqDecodingTest, TestConcealmentEvents) {
+  const int kNumConcealmentEvents = 19;
+  const size_t kSamples = 10 * 16;
+  const size_t kPayloadBytes = kSamples * 2;
+  int seq_no = 0;
+  RTPHeader rtp_info;
+  rtp_info.ssrc = 0x1234;     // Just an arbitrary SSRC.
+  rtp_info.payloadType = 94;  // PCM16b WB codec.
+  rtp_info.markerBit = 0;
+  const uint8_t payload[kPayloadBytes] = {0};
+  bool muted;
+
+  for (int i = 0; i < kNumConcealmentEvents; i++) {
+    // Insert some packets of 10 ms size.
+    for (int j = 0; j < 10; j++) {
+      rtp_info.sequenceNumber = seq_no++;
+      rtp_info.timestamp = rtp_info.sequenceNumber * kSamples;
+      neteq_->InsertPacket(rtp_info, payload, 0);
+      neteq_->GetAudio(&out_frame_, &muted);
+    }
+
+    // Lose a number of packets.
+    int num_lost = 1 + i;
+    for (int j = 0; j < num_lost; j++) {
+      seq_no++;
+      neteq_->GetAudio(&out_frame_, &muted);
+    }
+  }
+
+  // Check number of concealment events.
+  NetEqLifetimeStatistics stats = neteq_->GetLifetimeStatistics();
+  EXPECT_EQ(kNumConcealmentEvents, static_cast<int>(stats.concealment_events));
+}
+
 }  // namespace webrtc
diff --git a/modules/audio_coding/neteq/statistics_calculator.cc b/modules/audio_coding/neteq/statistics_calculator.cc
@@ -151,14 +151,18 @@ void StatisticsCalculator::ResetMcu() {
   timestamps_since_last_report_ = 0;
 }
 
-void StatisticsCalculator::ExpandedVoiceSamples(size_t num_samples) {
+void StatisticsCalculator::ExpandedVoiceSamples(size_t num_samples,
+                                                bool is_new_concealment_event) {
   expanded_speech_samples_ += num_samples;
   lifetime_stats_.concealed_samples += num_samples;
+  lifetime_stats_.concealment_events += is_new_concealment_event;
 }
 
-void StatisticsCalculator::ExpandedNoiseSamples(size_t num_samples) {
+void StatisticsCalculator::ExpandedNoiseSamples(size_t num_samples,
+                                                bool is_new_concealment_event) {
   expanded_noise_samples_ += num_samples;
   lifetime_stats_.concealed_samples += num_samples;
+  lifetime_stats_.concealment_events += is_new_concealment_event;
 }
 
 void StatisticsCalculator::ExpandedVoiceSamplesCorrection(int num_samples) {

diff --git a/modules/audio_coding/neteq/statistics_calculator.h b/modules/audio_coding/neteq/statistics_calculator.h
@@ -39,11 +39,11 @@ class StatisticsCalculator {
 
   // Reports that |num_samples| samples were produced through expansion, and
   // that the expansion produced other than just noise samples.
-  void ExpandedVoiceSamples(size_t num_samples);
+  void ExpandedVoiceSamples(size_t num_samples, bool is_new_concealment_event);
 
   // Reports that |num_samples| samples were produced through expansion, and
   // that the expansion produced only noise samples.
-  void ExpandedNoiseSamples(size_t num_samples);
+  void ExpandedNoiseSamples(size_t num_samples, bool is_new_concealment_event);
 
   // Corrects the statistics for number of samples produced through non-noise
   // expansion by adding |num_samples| (negative or positive) to the current

diff --git a/pc/rtcstats_integrationtest.cc b/pc/rtcstats_integrationtest.cc
@@ -560,17 +560,20 @@ class RTCStatsReportVerifier {
       verifier.MarkMemberTested(
           media_stream_track.echo_return_loss_enhancement, true);
     }
-    // totalSamplesReceived and concealedSamples are only present on inbound
-    // audio tracks.
+    // totalSamplesReceived, concealedSamples and concealmentEvents are only
+    // present on inbound audio tracks.
     if (*media_stream_track.kind == RTCMediaStreamTrackKind::kAudio &&
         *media_stream_track.remote_source) {
       verifier.TestMemberIsNonNegative<uint64_t>(
           media_stream_track.total_samples_received);
       verifier.TestMemberIsNonNegative<uint64_t>(
           media_stream_track.concealed_samples);
+      verifier.TestMemberIsNonNegative<uint64_t>(
+          media_stream_track.concealment_events);
     } else {
       verifier.TestMemberIsUndefined(media_stream_track.total_samples_received);
       verifier.TestMemberIsUndefined(media_stream_track.concealed_samples);
+      verifier.TestMemberIsUndefined(media_stream_track.concealment_events);
     }
     return verifier.ExpectAllMembersSuccessfullyTested();
   }

diff --git a/pc/rtcstatscollector.cc b/pc/rtcstatscollector.cc
@@ -417,6 +417,8 @@ ProduceMediaStreamTrackStatsFromVoiceReceiverInfo(
   audio_track_stats->total_samples_duration =
       voice_receiver_info.total_output_duration;
   audio_track_stats->concealed_samples = voice_receiver_info.concealed_samples;
+  audio_track_stats->concealment_events =
+      voice_receiver_info.concealment_events;
   return audio_track_stats;
 }
 

diff --git a/pc/rtcstatscollector_unittest.cc b/pc/rtcstatscollector_unittest.cc
@@ -1555,6 +1555,7 @@ TEST_F(RTCStatsCollectorTest,
   voice_receiver_info.total_samples_received = 4567;
   voice_receiver_info.total_output_duration = 0.25;
   voice_receiver_info.concealed_samples = 123;
+  voice_receiver_info.concealment_events = 12;
 
   test_->CreateMockRtpSendersReceiversAndChannels(
       { std::make_pair(local_audio_track.get(), voice_sender_info_ssrc1),
@@ -1631,6 +1632,7 @@ TEST_F(RTCStatsCollectorTest,
   expected_remote_audio_track.total_samples_received = 4567;
   expected_remote_audio_track.total_samples_duration = 0.25;
   expected_remote_audio_track.concealed_samples = 123;
+  expected_remote_audio_track.concealment_events = 12;
   ASSERT_TRUE(report->Get(expected_remote_audio_track.id()));
   EXPECT_EQ(expected_remote_audio_track,
             report->Get(expected_remote_audio_track.id())->cast_to<

diff --git a/stats/rtcstats_objects.cc b/stats/rtcstats_objects.cc
@@ -383,7 +383,8 @@ WEBRTC_RTCSTATS_IMPL(RTCMediaStreamTrackStats, RTCStats, "track",
                      &echo_return_loss_enhancement,
                      &total_samples_received,
                      &total_samples_duration,
-                     &concealed_samples);
+                     &concealed_samples,
+                     &concealment_events);
 // clang-format on
 
 RTCMediaStreamTrackStats::RTCMediaStreamTrackStats(
@@ -416,7 +417,8 @@ RTCMediaStreamTrackStats::RTCMediaStreamTrackStats(std::string&& id,
       echo_return_loss_enhancement("echoReturnLossEnhancement"),
       total_samples_received("totalSamplesReceived"),
       total_samples_duration("totalSamplesDuration"),
-      concealed_samples("concealedSamples") {
+      concealed_samples("concealedSamples"),
+      concealment_events("concealmentEvents") {
   RTC_DCHECK(kind == RTCMediaStreamTrackKind::kAudio ||
              kind == RTCMediaStreamTrackKind::kVideo);
 }
@@ -445,7 +447,8 @@ RTCMediaStreamTrackStats::RTCMediaStreamTrackStats(
       echo_return_loss_enhancement(other.echo_return_loss_enhancement),
       total_samples_received(other.total_samples_received),
       total_samples_duration(other.total_samples_duration),
-      concealed_samples(other.concealed_samples) {}
+      concealed_samples(other.concealed_samples),
+      concealment_events(other.concealment_events) {}
 
 RTCMediaStreamTrackStats::~RTCMediaStreamTrackStats() {
 }