Add jitterBufferTargetDelay as RTCNonStandardStatsMember to new GetStats API
Bug: webrtc:11381
Change-Id: I7df3450e50da49d178e1e3a5d9f4970672d91aac
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169120
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30758}
diff --git a/api/neteq/neteq.h b/api/neteq/neteq.h
index 67dbd50..d632e03 100644
--- a/api/neteq/neteq.h
+++ b/api/neteq/neteq.h
@@ -68,6 +68,7 @@
uint64_t concealment_events = 0;
uint64_t jitter_buffer_delay_ms = 0;
uint64_t jitter_buffer_emitted_count = 0;
+ uint64_t jitter_buffer_target_delay_ms = 0;
uint64_t inserted_samples_for_deceleration = 0;
uint64_t removed_samples_for_acceleration = 0;
uint64_t silent_concealed_samples = 0;
diff --git a/api/stats/rtcstats_objects.h b/api/stats/rtcstats_objects.h
index 197a810..e83c83d 100644
--- a/api/stats/rtcstats_objects.h
+++ b/api/stats/rtcstats_objects.h
@@ -327,6 +327,14 @@
RTCNonStandardStatsMember<uint64_t> jitter_buffer_flushes;
RTCNonStandardStatsMember<uint64_t> delayed_packet_outage_samples;
RTCNonStandardStatsMember<double> relative_packet_arrival_delay;
+ // Non-standard metric showing target delay of jitter buffer.
+ // This value is increased by the target jitter buffer delay every time a
+ // sample is emitted by the jitter buffer. The added target is the target
+ // delay, in seconds, at the time that the sample was emitted from the jitter
+ // buffer. (https://github.com/w3c/webrtc-provisional-stats/pull/20)
+ // Currently it is implemented only for audio.
+ // TODO(titovartem) implement for video streams when will be requested.
+ RTCNonStandardStatsMember<double> jitter_buffer_target_delay;
// TODO(henrik.lundin): Add description of the interruption metrics at
// https://github.com/henbos/webrtc-provisional-stats/issues/17
RTCNonStandardStatsMember<uint32_t> interruption_count;
diff --git a/audio/audio_receive_stream.cc b/audio/audio_receive_stream.cc
index 7c6adcd..ce1b344 100644
--- a/audio/audio_receive_stream.cc
+++ b/audio/audio_receive_stream.cc
@@ -222,6 +222,9 @@
static_cast<double>(ns.jitterBufferDelayMs) /
static_cast<double>(rtc::kNumMillisecsPerSec);
stats.jitter_buffer_emitted_count = ns.jitterBufferEmittedCount;
+ stats.jitter_buffer_target_delay_seconds =
+ static_cast<double>(ns.jitterBufferTargetDelayMs) /
+ static_cast<double>(rtc::kNumMillisecsPerSec);
stats.inserted_samples_for_deceleration = ns.insertedSamplesForDeceleration;
stats.removed_samples_for_acceleration = ns.removedSamplesForAcceleration;
stats.expand_rate = Q14ToFloat(ns.currentExpandRate);
diff --git a/audio/audio_receive_stream_unittest.cc b/audio/audio_receive_stream_unittest.cc
index b8eff0a..0b2cae5 100644
--- a/audio/audio_receive_stream_unittest.cc
+++ b/audio/audio_receive_stream_unittest.cc
@@ -69,9 +69,9 @@
123,
{"codec_name_recv", 96000, 0}};
const NetworkStatistics kNetworkStats = {
- 123, 456, false, 789012, 3456, 123, 456, 789, 543, 432,
- 321, 123, 101, 0, {}, 789, 12, 345, 678, 901,
- 0, -1, -1, -1, -1, 0, 0, 0, 0};
+ 123, 456, false, 789012, 3456, 123, 456, 789, 543, 123,
+ 432, 321, 123, 101, 0, {}, 789, 12, 345, 678,
+ 901, 0, -1, -1, -1, -1, 0, 0, 0, 0};
const AudioDecodingCallStats kAudioDecodeStats = MakeAudioDecodeStatsForTest();
struct ConfigHelper {
@@ -294,6 +294,9 @@
stats.jitter_buffer_delay_seconds);
EXPECT_EQ(kNetworkStats.jitterBufferEmittedCount,
stats.jitter_buffer_emitted_count);
+ EXPECT_EQ(static_cast<double>(kNetworkStats.jitterBufferTargetDelayMs) /
+ static_cast<double>(rtc::kNumMillisecsPerSec),
+ stats.jitter_buffer_target_delay_seconds);
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 090fb82..4a50cdb 100644
--- a/call/audio_receive_stream.h
+++ b/call/audio_receive_stream.h
@@ -58,6 +58,7 @@
uint64_t concealment_events = 0;
double jitter_buffer_delay_seconds = 0.0;
uint64_t jitter_buffer_emitted_count = 0;
+ double jitter_buffer_target_delay_seconds = 0.0;
uint64_t inserted_samples_for_deceleration = 0;
uint64_t removed_samples_for_acceleration = 0;
// Stats below DO NOT correspond directly to anything in the WebRTC stats
diff --git a/media/base/media_channel.h b/media/base/media_channel.h
index a62c618..7ca6002 100644
--- a/media/base/media_channel.h
+++ b/media/base/media_channel.h
@@ -512,6 +512,7 @@
uint64_t concealment_events = 0;
double jitter_buffer_delay_seconds = 0.0;
uint64_t jitter_buffer_emitted_count = 0;
+ double jitter_buffer_target_delay_seconds = 0.0;
uint64_t inserted_samples_for_deceleration = 0;
uint64_t removed_samples_for_acceleration = 0;
uint64_t fec_packets_received = 0;
diff --git a/media/engine/webrtc_voice_engine.cc b/media/engine/webrtc_voice_engine.cc
index 7da9abd..45b46bf 100644
--- a/media/engine/webrtc_voice_engine.cc
+++ b/media/engine/webrtc_voice_engine.cc
@@ -2230,6 +2230,8 @@
rinfo.concealment_events = stats.concealment_events;
rinfo.jitter_buffer_delay_seconds = stats.jitter_buffer_delay_seconds;
rinfo.jitter_buffer_emitted_count = stats.jitter_buffer_emitted_count;
+ rinfo.jitter_buffer_target_delay_seconds =
+ stats.jitter_buffer_target_delay_seconds;
rinfo.inserted_samples_for_deceleration =
stats.inserted_samples_for_deceleration;
rinfo.removed_samples_for_acceleration =
diff --git a/modules/audio_coding/acm2/acm_receiver.cc b/modules/audio_coding/acm2/acm_receiver.cc
index 9cecb98..29eff19 100644
--- a/modules/audio_coding/acm2/acm_receiver.cc
+++ b/modules/audio_coding/acm2/acm_receiver.cc
@@ -272,6 +272,8 @@
neteq_lifetime_stat.silent_concealed_samples;
acm_stat->concealmentEvents = neteq_lifetime_stat.concealment_events;
acm_stat->jitterBufferDelayMs = neteq_lifetime_stat.jitter_buffer_delay_ms;
+ acm_stat->jitterBufferTargetDelayMs =
+ neteq_lifetime_stat.jitter_buffer_target_delay_ms;
acm_stat->jitterBufferEmittedCount =
neteq_lifetime_stat.jitter_buffer_emitted_count;
acm_stat->delayedPacketOutageSamples =
diff --git a/modules/audio_coding/include/audio_coding_module_typedefs.h b/modules/audio_coding/include/audio_coding_module_typedefs.h
index 2f40acd..a5d4b24 100644
--- a/modules/audio_coding/include/audio_coding_module_typedefs.h
+++ b/modules/audio_coding/include/audio_coding_module_typedefs.h
@@ -90,6 +90,8 @@
uint64_t concealmentEvents;
uint64_t jitterBufferDelayMs;
uint64_t jitterBufferEmittedCount;
+ // Non standard stats propagated to spec complaint GetStats API.
+ uint64_t jitterBufferTargetDelayMs;
uint64_t insertedSamplesForDeceleration;
uint64_t removedSamplesForAcceleration;
uint64_t fecPacketsReceived;
diff --git a/modules/audio_coding/neteq/neteq_impl.cc b/modules/audio_coding/neteq/neteq_impl.cc
index 265043a..0b7510d 100644
--- a/modules/audio_coding/neteq/neteq_impl.cc
+++ b/modules/audio_coding/neteq/neteq_impl.cc
@@ -1987,7 +1987,9 @@
}
extracted_samples = packet->timestamp - first_timestamp + packet_duration;
- stats_->JitterBufferDelay(packet_duration, waiting_time_ms);
+ RTC_DCHECK(controller_);
+ stats_->JitterBufferDelay(packet_duration, waiting_time_ms,
+ controller_->TargetLevelMs());
packet_list->push_back(std::move(*packet)); // Store packet in list.
packet = absl::nullopt; // Ensure it's never used after the move.
diff --git a/modules/audio_coding/neteq/neteq_unittest.cc b/modules/audio_coding/neteq/neteq_unittest.cc
index 701a3c5..d78e2c6 100644
--- a/modules/audio_coding/neteq/neteq_unittest.cc
+++ b/modules/audio_coding/neteq/neteq_unittest.cc
@@ -986,6 +986,7 @@
int packets_sent = 0;
int packets_received = 0;
int expected_delay = 0;
+ int expected_target_delay = 0;
uint64_t expected_emitted_count = 0;
while (packets_received < kNumPackets) {
// Insert packet.
@@ -1010,6 +1011,7 @@
// number of samples that are sent for play out.
int current_delay_ms = packets_delay * kPacketLenMs;
expected_delay += current_delay_ms * kSamples;
+ expected_target_delay += neteq_->TargetDelayMs() * kSamples;
expected_emitted_count += kSamples;
}
}
@@ -1021,8 +1023,11 @@
// Check jitter buffer delay.
NetEqLifetimeStatistics stats = neteq_->GetLifetimeStatistics();
- EXPECT_EQ(expected_delay, static_cast<int>(stats.jitter_buffer_delay_ms));
+ EXPECT_EQ(expected_delay,
+ rtc::checked_cast<int>(stats.jitter_buffer_delay_ms));
EXPECT_EQ(expected_emitted_count, stats.jitter_buffer_emitted_count);
+ EXPECT_EQ(expected_target_delay,
+ rtc::checked_cast<int>(stats.jitter_buffer_target_delay_ms));
}
TEST_F(NetEqDecodingTestFaxMode, TestJitterBufferDelayWithoutLoss) {
@@ -1043,6 +1048,7 @@
rtp_info.markerBit = 0;
const uint8_t payload[kPayloadBytes] = {0};
+ int expected_target_delay = neteq_->TargetDelayMs() * kSamples;
neteq_->InsertPacket(rtp_info, payload);
bool muted;
@@ -1055,6 +1061,7 @@
rtp_info.timestamp += kSamples;
neteq_->InsertPacket(rtp_info, payload);
+ expected_target_delay += neteq_->TargetDelayMs() * 2 * kSamples;
// We have two packets in the buffer and kAccelerate operation will
// extract 20 ms of data.
neteq_->GetAudio(&out_frame_, &muted, NetEq::Operation::kAccelerate);
@@ -1063,6 +1070,8 @@
NetEqLifetimeStatistics stats = neteq_->GetLifetimeStatistics();
EXPECT_EQ(10 * kSamples * 3, stats.jitter_buffer_delay_ms);
EXPECT_EQ(kSamples * 3, stats.jitter_buffer_emitted_count);
+ EXPECT_EQ(expected_target_delay,
+ rtc::checked_cast<int>(stats.jitter_buffer_target_delay_ms));
}
namespace test {
diff --git a/modules/audio_coding/neteq/statistics_calculator.cc b/modules/audio_coding/neteq/statistics_calculator.cc
index 081ec33..fa2925c 100644
--- a/modules/audio_coding/neteq/statistics_calculator.cc
+++ b/modules/audio_coding/neteq/statistics_calculator.cc
@@ -275,8 +275,11 @@
}
void StatisticsCalculator::JitterBufferDelay(size_t num_samples,
- uint64_t waiting_time_ms) {
+ uint64_t waiting_time_ms,
+ uint64_t target_delay_ms) {
lifetime_stats_.jitter_buffer_delay_ms += waiting_time_ms * num_samples;
+ lifetime_stats_.jitter_buffer_target_delay_ms +=
+ target_delay_ms * num_samples;
lifetime_stats_.jitter_buffer_emitted_count += num_samples;
}
diff --git a/modules/audio_coding/neteq/statistics_calculator.h b/modules/audio_coding/neteq/statistics_calculator.h
index 38e463c..333f4a7 100644
--- a/modules/audio_coding/neteq/statistics_calculator.h
+++ b/modules/audio_coding/neteq/statistics_calculator.h
@@ -83,7 +83,9 @@
void IncreaseCounter(size_t num_samples, int fs_hz);
// Update jitter buffer delay counter.
- void JitterBufferDelay(size_t num_samples, uint64_t waiting_time_ms);
+ void JitterBufferDelay(size_t num_samples,
+ uint64_t waiting_time_ms,
+ uint64_t target_delay_ms);
// Stores new packet waiting time in waiting time statistics.
void StoreWaitingTime(int waiting_time_ms);
diff --git a/pc/rtc_stats_collector.cc b/pc/rtc_stats_collector.cc
index 5b34f1c..29c7d09 100644
--- a/pc/rtc_stats_collector.cc
+++ b/pc/rtc_stats_collector.cc
@@ -648,6 +648,8 @@
voice_receiver_info.delayed_packet_outage_samples;
audio_track_stats->relative_packet_arrival_delay =
voice_receiver_info.relative_packet_arrival_delay_seconds;
+ audio_track_stats->jitter_buffer_target_delay =
+ voice_receiver_info.jitter_buffer_target_delay_seconds;
audio_track_stats->interruption_count =
voice_receiver_info.interruption_count >= 0
? voice_receiver_info.interruption_count
diff --git a/pc/rtc_stats_collector_unittest.cc b/pc/rtc_stats_collector_unittest.cc
index f886e14..12f6059 100644
--- a/pc/rtc_stats_collector_unittest.cc
+++ b/pc/rtc_stats_collector_unittest.cc
@@ -1547,6 +1547,7 @@
voice_receiver_info.silent_concealed_samples = 765;
voice_receiver_info.jitter_buffer_delay_seconds = 3456;
voice_receiver_info.jitter_buffer_emitted_count = 13;
+ voice_receiver_info.jitter_buffer_target_delay_seconds = 7.894;
voice_receiver_info.jitter_buffer_flushes = 7;
voice_receiver_info.delayed_packet_outage_samples = 15;
voice_receiver_info.relative_packet_arrival_delay_seconds = 16;
@@ -1591,6 +1592,7 @@
expected_remote_audio_track.silent_concealed_samples = 765;
expected_remote_audio_track.jitter_buffer_delay = 3456;
expected_remote_audio_track.jitter_buffer_emitted_count = 13;
+ expected_remote_audio_track.jitter_buffer_target_delay = 7.894;
expected_remote_audio_track.jitter_buffer_flushes = 7;
expected_remote_audio_track.delayed_packet_outage_samples = 15;
expected_remote_audio_track.relative_packet_arrival_delay = 16;
diff --git a/pc/rtc_stats_integrationtest.cc b/pc/rtc_stats_integrationtest.cc
index 93a46ef..fa7d56a 100644
--- a/pc/rtc_stats_integrationtest.cc
+++ b/pc/rtc_stats_integrationtest.cc
@@ -650,6 +650,8 @@
verifier.TestMemberIsUndefined(media_stream_track.interruption_count);
verifier.TestMemberIsUndefined(
media_stream_track.total_interruption_duration);
+ verifier.TestMemberIsUndefined(
+ media_stream_track.jitter_buffer_target_delay);
} else {
RTC_DCHECK_EQ(*media_stream_track.kind, RTCMediaStreamTrackKind::kAudio);
// The type of the referenced media source depends on kind.
@@ -660,6 +662,8 @@
media_stream_track.jitter_buffer_delay);
verifier.TestMemberIsNonNegative<uint64_t>(
media_stream_track.jitter_buffer_emitted_count);
+ verifier.TestMemberIsNonNegative<double>(
+ media_stream_track.jitter_buffer_target_delay);
verifier.TestMemberIsPositive<double>(media_stream_track.audio_level);
verifier.TestMemberIsPositive<double>(
media_stream_track.total_audio_energy);
@@ -694,6 +698,8 @@
verifier.TestMemberIsUndefined(media_stream_track.jitter_buffer_delay);
verifier.TestMemberIsUndefined(
media_stream_track.jitter_buffer_emitted_count);
+ verifier.TestMemberIsUndefined(
+ media_stream_track.jitter_buffer_target_delay);
verifier.TestMemberIsUndefined(media_stream_track.audio_level);
verifier.TestMemberIsUndefined(media_stream_track.total_audio_energy);
verifier.TestMemberIsUndefined(
diff --git a/stats/rtcstats_objects.cc b/stats/rtcstats_objects.cc
index 77cbc09..d99c9e7 100644
--- a/stats/rtcstats_objects.cc
+++ b/stats/rtcstats_objects.cc
@@ -395,6 +395,7 @@
&jitter_buffer_flushes,
&delayed_packet_outage_samples,
&relative_packet_arrival_delay,
+ &jitter_buffer_target_delay,
&interruption_count,
&total_interruption_duration,
&freeze_count,
@@ -454,6 +455,7 @@
relative_packet_arrival_delay(
"relativePacketArrivalDelay",
{NonStandardGroupId::kRtcStatsRelativePacketArrivalDelay}),
+ jitter_buffer_target_delay("jitterBufferTargetDelay"),
interruption_count("interruptionCount"),
total_interruption_duration("totalInterruptionDuration"),
freeze_count("freezeCount"),
@@ -503,6 +505,7 @@
jitter_buffer_flushes(other.jitter_buffer_flushes),
delayed_packet_outage_samples(other.delayed_packet_outage_samples),
relative_packet_arrival_delay(other.relative_packet_arrival_delay),
+ jitter_buffer_target_delay(other.jitter_buffer_target_delay),
interruption_count(other.interruption_count),
total_interruption_duration(other.total_interruption_duration),
freeze_count(other.freeze_count),
