Expose jitter buffer flushes metric in new getStats api.
Origin trial experiment proposal (new statistic part):
https://docs.google.com/document/d/1stYIZhEmDZ7NJF9gjjsM66eLFJUdc-14a3QutrFbIwI/edit?ts=5bf5535c#
Bug: chromium:907113
Change-Id: I1d005291f9b47665f70c26148dbdcbb55564bef8
Reviewed-on: https://webrtc-review.googlesource.com/c/111505
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Ruslan Burakov <kuddai@google.com>
Cr-Commit-Position: refs/heads/master@{#25768}diff --git a/api/stats/rtcstats.h b/api/stats/rtcstats.h
index f65df28..7fd0d84 100644
--- a/api/stats/rtcstats.h
+++ b/api/stats/rtcstats.h
@@ -295,15 +295,6 @@
is_defined_ = true;
return value_;
}
- T& operator=(const RTCStatsMember<T>& other) {
- RTC_DCHECK(other.is_defined_);
- // Shouldn't be attempting to assign an RTCNonStandardStatsMember to an
- // RTCStatsMember or vice versa.
- RTC_DCHECK(is_standardized() == other.is_standardized());
- value_ = other.value_;
- is_defined_ = true;
- return value_;
- }
// Value getters.
T& operator*() {
@@ -348,6 +339,11 @@
: RTCStatsMember<T>(std::move(other)) {}
bool is_standardized() const override { return false; }
+
+ T& operator=(const T& value) { return RTCStatsMember<T>::operator=(value); }
+ T& operator=(const T&& value) {
+ return RTCStatsMember<T>::operator=(std::move(value));
+ }
};
} // namespace webrtc
diff --git a/api/stats/rtcstats_objects.h b/api/stats/rtcstats_objects.h
index dc9da72..8496f8c 100644
--- a/api/stats/rtcstats_objects.h
+++ b/api/stats/rtcstats_objects.h
@@ -308,6 +308,9 @@
RTCStatsMember<double> total_samples_duration;
RTCStatsMember<uint64_t> concealed_samples;
RTCStatsMember<uint64_t> concealment_events;
+ // Non-standard audio-only member
+ // TODO(kuddai): Add descriptoin to standard. crbug.com/webrtc/10042
+ RTCNonStandardStatsMember<uint64_t> jitter_buffer_flushes;
};
// https://w3c.github.io/webrtc-stats/#pcstats-dict*
diff --git a/audio/audio_receive_stream.cc b/audio/audio_receive_stream.cc
index 4250765..2da3877 100644
--- a/audio/audio_receive_stream.cc
+++ b/audio/audio_receive_stream.cc
@@ -213,6 +213,7 @@
stats.secondary_discarded_rate = Q14ToFloat(ns.currentSecondaryDiscardedRate);
stats.accelerate_rate = Q14ToFloat(ns.currentAccelerateRate);
stats.preemptive_expand_rate = Q14ToFloat(ns.currentPreemptiveRate);
+ stats.jitter_buffer_flushes = ns.packetBufferFlushes;
auto ds = channel_receive_->GetDecodingCallStatistics();
stats.decoding_calls_to_silence_generator = ds.calls_to_silence_generator;
diff --git a/call/audio_receive_stream.h b/call/audio_receive_stream.h
index f792aa2..fe609de 100644
--- a/call/audio_receive_stream.h
+++ b/call/audio_receive_stream.h
@@ -71,6 +71,7 @@
int32_t decoding_plc_cng = 0;
int32_t decoding_muted_output = 0;
int64_t capture_start_ntp_time_ms = 0;
+ uint64_t jitter_buffer_flushes = 0;
};
struct Config {
diff --git a/media/base/mediachannel.h b/media/base/mediachannel.h
index 3ab333e..04e70c5 100644
--- a/media/base/mediachannel.h
+++ b/media/base/mediachannel.h
@@ -476,6 +476,8 @@
int decoding_muted_output = 0;
// Estimated capture start time in NTP time in ms.
int64_t capture_start_ntp_time_ms = -1;
+ // Count of the number of buffer flushes.
+ uint64_t jitter_buffer_flushes = 0;
};
struct VideoSenderInfo : public MediaSenderInfo {
diff --git a/media/engine/webrtcvoiceengine.cc b/media/engine/webrtcvoiceengine.cc
index 51f4582..c6751bb 100644
--- a/media/engine/webrtcvoiceengine.cc
+++ b/media/engine/webrtcvoiceengine.cc
@@ -2242,6 +2242,8 @@
rinfo.decoding_plc_cng = stats.decoding_plc_cng;
rinfo.decoding_muted_output = stats.decoding_muted_output;
rinfo.capture_start_ntp_time_ms = stats.capture_start_ntp_time_ms;
+ rinfo.jitter_buffer_flushes = stats.jitter_buffer_flushes;
+
info->receivers.push_back(rinfo);
}
diff --git a/modules/audio_coding/acm2/acm_receiver.cc b/modules/audio_coding/acm2/acm_receiver.cc
index e3aa4ed..d02e50d 100644
--- a/modules/audio_coding/acm2/acm_receiver.cc
+++ b/modules/audio_coding/acm2/acm_receiver.cc
@@ -345,6 +345,11 @@
acm_stat->concealedSamples = neteq_lifetime_stat.concealed_samples;
acm_stat->concealmentEvents = neteq_lifetime_stat.concealment_events;
acm_stat->jitterBufferDelayMs = neteq_lifetime_stat.jitter_buffer_delay_ms;
+
+ NetEqOperationsAndState neteq_operations_and_state =
+ neteq_->GetOperationsAndState();
+ acm_stat->packetBufferFlushes =
+ neteq_operations_and_state.packet_buffer_flushes;
}
int AcmReceiver::DecoderByPayloadType(uint8_t payload_type,
diff --git a/modules/audio_coding/include/audio_coding_module_typedefs.h b/modules/audio_coding/include/audio_coding_module_typedefs.h
index 8946afd..3c746de 100644
--- a/modules/audio_coding/include/audio_coding_module_typedefs.h
+++ b/modules/audio_coding/include/audio_coding_module_typedefs.h
@@ -117,6 +117,8 @@
int maxWaitingTimeMs;
// added samples in off mode due to packet loss
size_t addedSamples;
+ // count of the number of buffer flushes
+ uint64_t packetBufferFlushes;
};
} // namespace webrtc
diff --git a/pc/rtcstats_integrationtest.cc b/pc/rtcstats_integrationtest.cc
index f9375f7..46687ba 100644
--- a/pc/rtcstats_integrationtest.cc
+++ b/pc/rtcstats_integrationtest.cc
@@ -604,11 +604,14 @@
media_stream_track.concealed_samples);
verifier.TestMemberIsNonNegative<uint64_t>(
media_stream_track.concealment_events);
+ verifier.TestMemberIsNonNegative<uint64_t>(
+ media_stream_track.jitter_buffer_flushes);
} else {
verifier.TestMemberIsUndefined(media_stream_track.jitter_buffer_delay);
verifier.TestMemberIsUndefined(media_stream_track.total_samples_received);
verifier.TestMemberIsUndefined(media_stream_track.concealed_samples);
verifier.TestMemberIsUndefined(media_stream_track.concealment_events);
+ verifier.TestMemberIsUndefined(media_stream_track.jitter_buffer_flushes);
}
return verifier.ExpectAllMembersSuccessfullyTested();
}
diff --git a/pc/rtcstatscollector.cc b/pc/rtcstatscollector.cc
index 871cff9..33d4af1 100644
--- a/pc/rtcstatscollector.cc
+++ b/pc/rtcstatscollector.cc
@@ -454,6 +454,8 @@
audio_track_stats->concealed_samples = voice_receiver_info.concealed_samples;
audio_track_stats->concealment_events =
voice_receiver_info.concealment_events;
+ audio_track_stats->jitter_buffer_flushes =
+ voice_receiver_info.jitter_buffer_flushes;
return audio_track_stats;
}
diff --git a/pc/rtcstatscollector_unittest.cc b/pc/rtcstatscollector_unittest.cc
index 3fc0127..e16c7e3 100644
--- a/pc/rtcstatscollector_unittest.cc
+++ b/pc/rtcstatscollector_unittest.cc
@@ -1426,6 +1426,7 @@
voice_receiver_info.concealed_samples = 123;
voice_receiver_info.concealment_events = 12;
voice_receiver_info.jitter_buffer_delay_seconds = 3456;
+ voice_receiver_info.jitter_buffer_flushes = 7;
stats_->CreateMockRtpSendersReceiversAndChannels(
{}, {std::make_pair(remote_audio_track.get(), voice_receiver_info)}, {},
@@ -1459,6 +1460,7 @@
expected_remote_audio_track.concealed_samples = 123;
expected_remote_audio_track.concealment_events = 12;
expected_remote_audio_track.jitter_buffer_delay = 3456;
+ expected_remote_audio_track.jitter_buffer_flushes = 7;
ASSERT_TRUE(report->Get(expected_remote_audio_track.id()));
EXPECT_EQ(expected_remote_audio_track,
report->Get(expected_remote_audio_track.id())
diff --git a/stats/rtcstats_objects.cc b/stats/rtcstats_objects.cc
index dcc1180..669db14 100644
--- a/stats/rtcstats_objects.cc
+++ b/stats/rtcstats_objects.cc
@@ -374,7 +374,8 @@
&total_samples_received,
&total_samples_duration,
&concealed_samples,
- &concealment_events);
+ &concealment_events,
+ &jitter_buffer_flushes);
// clang-format on
RTCMediaStreamTrackStats::RTCMediaStreamTrackStats(const std::string& id,
@@ -410,7 +411,8 @@
total_samples_received("totalSamplesReceived"),
total_samples_duration("totalSamplesDuration"),
concealed_samples("concealedSamples"),
- concealment_events("concealmentEvents") {
+ concealment_events("concealmentEvents"),
+ jitter_buffer_flushes("jitterBufferFlushes") {
RTC_DCHECK(kind == RTCMediaStreamTrackKind::kAudio ||
kind == RTCMediaStreamTrackKind::kVideo);
}
@@ -442,7 +444,8 @@
total_samples_received(other.total_samples_received),
total_samples_duration(other.total_samples_duration),
concealed_samples(other.concealed_samples),
- concealment_events(other.concealment_events) {}
+ concealment_events(other.concealment_events),
+ jitter_buffer_flushes(other.jitter_buffer_flushes) {}
RTCMediaStreamTrackStats::~RTCMediaStreamTrackStats() {}
