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 <gustaf@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19881}diff --git a/api/stats/rtcstats_objects.h b/api/stats/rtcstats_objects.h
index a600fb9..903d266 100644
--- a/api/stats/rtcstats_objects.h
+++ b/api/stats/rtcstats_objects.h
@@ -278,6 +278,7 @@
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
index 29b3e5e..37e8aac 100644
--- a/api/statstypes.cc
+++ b/api/statstypes.cc
@@ -373,6 +373,8 @@
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
index 431a1ce..7f69b02 100644
--- a/api/statstypes.h
+++ b/api/statstypes.h
@@ -105,6 +105,7 @@
kStatsValueNameBytesSent,
kStatsValueNameCodecImplementationName,
kStatsValueNameConcealedSamples,
+ kStatsValueNameConcealmentEvents,
kStatsValueNameDataChannelId,
kStatsValueNameFramesDecoded,
kStatsValueNameFramesEncoded,
diff --git a/audio/audio_receive_stream.cc b/audio/audio_receive_stream.cc
index 0c83319..8ff1d58 100644
--- a/audio/audio_receive_stream.cc
+++ b/audio/audio_receive_stream.cc
@@ -196,6 +196,7 @@
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
index 7bcb85a..b79ae19 100644
--- a/audio/audio_receive_stream_unittest.cc
+++ b/audio/audio_receive_stream_unittest.cc
@@ -64,9 +64,9 @@
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 @@
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
index 096f196..baf2b67 100644
--- a/call/audio_receive_stream.h
+++ b/call/audio_receive_stream.h
@@ -50,18 +50,14 @@
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
index 4bd9313..69fc761 100644
--- a/common_types.h
+++ b/common_types.h
@@ -375,6 +375,10 @@
// 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
index 631e695..103240e 100644
--- a/media/base/mediachannel.h
+++ b/media/base/mediachannel.h
@@ -657,6 +657,7 @@
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 @@
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
index 8fa5fe2..00c7fd6 100644
--- a/media/engine/webrtcvoiceengine.cc
+++ b/media/engine/webrtcvoiceengine.cc
@@ -2284,6 +2284,7 @@
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
index 2c31253..4e80788 100644
--- a/media/engine/webrtcvoiceengine_unittest.cc
+++ b/media/engine/webrtcvoiceengine_unittest.cc
@@ -622,6 +622,7 @@
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 @@
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
index 8be2471..d999df0 100644
--- a/modules/audio_coding/acm2/acm_receiver.cc
+++ b/modules/audio_coding/acm2/acm_receiver.cc
@@ -336,6 +336,7 @@
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
index 56f8204..b349f20 100644
--- a/modules/audio_coding/neteq/include/neteq.h
+++ b/modules/audio_coding/neteq/include/neteq.h
@@ -61,13 +61,11 @@
// 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
index 3cdcf45..cdf590b 100644
--- a/modules/audio_coding/neteq/neteq_impl.cc
+++ b/modules/audio_coding/neteq/neteq_impl.cc
@@ -847,7 +847,7 @@
: 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 @@
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
index 657b8ed..764a505 100644
--- a/modules/audio_coding/neteq/neteq_unittest.cc
+++ b/modules/audio_coding/neteq/neteq_unittest.cc
@@ -1634,4 +1634,38 @@
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
index 163cfff..bbd6e24 100644
--- a/modules/audio_coding/neteq/statistics_calculator.cc
+++ b/modules/audio_coding/neteq/statistics_calculator.cc
@@ -151,14 +151,18 @@
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
index e3e2749..cec02bbf 100644
--- a/modules/audio_coding/neteq/statistics_calculator.h
+++ b/modules/audio_coding/neteq/statistics_calculator.h
@@ -39,11 +39,11 @@
// 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
index a498dc8..62d316d 100644
--- a/pc/rtcstats_integrationtest.cc
+++ b/pc/rtcstats_integrationtest.cc
@@ -560,17 +560,20 @@
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
index 9a6b26c..161d224 100644
--- a/pc/rtcstatscollector.cc
+++ b/pc/rtcstatscollector.cc
@@ -417,6 +417,8 @@
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
index cb5a79f..14f669c 100644
--- a/pc/rtcstatscollector_unittest.cc
+++ b/pc/rtcstatscollector_unittest.cc
@@ -1555,6 +1555,7 @@
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 @@
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
index f407c06..e643e12 100644
--- a/stats/rtcstats_objects.cc
+++ b/stats/rtcstats_objects.cc
@@ -383,7 +383,8 @@
&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 @@
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 @@
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() {
}
