Add stats totalSamplesReceived and concealedSamples
Adds two new stats to RTCMediaStreamTrackStats:
* totalSamplesReceived is the total number of samples received on
the audio channel and includes real and synthetic samples.
* concealedSamples is the total number of synthetic samples
received on the audio channel used to conceal packet loss.
Bug: webrtc:8076
Change-Id: I36e9828525ec341490cf3310a972b56a8b443667
Reviewed-on: https://chromium-review.googlesource.com/615902
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19506}diff --git a/webrtc/modules/audio_coding/neteq/include/neteq.h b/webrtc/modules/audio_coding/neteq/include/neteq.h
index d05b76e..56c30e5 100644
--- a/webrtc/modules/audio_coding/neteq/include/neteq.h
+++ b/webrtc/modules/audio_coding/neteq/include/neteq.h
@@ -58,6 +58,18 @@
int max_waiting_time_ms;
};
+// 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
+ 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;
+};
+
enum NetEqPlayoutMode {
kPlayoutOn,
kPlayoutOff,
@@ -220,6 +232,10 @@
// after the call.
virtual int NetworkStatistics(NetEqNetworkStatistics* stats) = 0;
+ // Returns a copy of this class's lifetime statistics. These statistics are
+ // never reset.
+ virtual NetEqLifetimeStatistics GetLifetimeStatistics() const = 0;
+
// Writes the current RTCP statistics to |stats|. The statistics are reset
// and a new report period is started with the call.
virtual void GetRtcpStatistics(RtcpStatistics* stats) = 0;
diff --git a/webrtc/modules/audio_coding/neteq/neteq_impl.cc b/webrtc/modules/audio_coding/neteq/neteq_impl.cc
index 4b95253..7858e84 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_impl.cc
+++ b/webrtc/modules/audio_coding/neteq/neteq_impl.cc
@@ -380,6 +380,11 @@
return 0;
}
+NetEqLifetimeStatistics NetEqImpl::GetLifetimeStatistics() const {
+ rtc::CritScope lock(&crit_sect_);
+ return stats_.GetLifetimeStatistics();
+}
+
void NetEqImpl::GetRtcpStatistics(RtcpStatistics* stats) {
rtc::CritScope lock(&crit_sect_);
if (stats) {
diff --git a/webrtc/modules/audio_coding/neteq/neteq_impl.h b/webrtc/modules/audio_coding/neteq/neteq_impl.h
index 0eeff2e..f4b014a 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_impl.h
+++ b/webrtc/modules/audio_coding/neteq/neteq_impl.h
@@ -185,6 +185,8 @@
// and a new report period is started with the call.
void GetRtcpStatistics(RtcpStatistics* stats) override;
+ NetEqLifetimeStatistics GetLifetimeStatistics() const override;
+
// Same as RtcpStatistics(), but does not reset anything.
void GetRtcpStatisticsNoReset(RtcpStatistics* stats) override;
diff --git a/webrtc/modules/audio_coding/neteq/statistics_calculator.cc b/webrtc/modules/audio_coding/neteq/statistics_calculator.cc
index 3faed62..d7d1644 100644
--- a/webrtc/modules/audio_coding/neteq/statistics_calculator.cc
+++ b/webrtc/modules/audio_coding/neteq/statistics_calculator.cc
@@ -153,24 +153,29 @@
void StatisticsCalculator::ExpandedVoiceSamples(size_t num_samples) {
expanded_speech_samples_ += num_samples;
+ lifetime_stats_.concealed_samples += num_samples;
}
void StatisticsCalculator::ExpandedNoiseSamples(size_t num_samples) {
expanded_noise_samples_ += num_samples;
+ lifetime_stats_.concealed_samples += num_samples;
}
void StatisticsCalculator::ExpandedVoiceSamplesCorrection(int num_samples) {
expanded_speech_samples_ =
AddIntToSizeTWithLowerCap(num_samples, expanded_speech_samples_);
+ lifetime_stats_.concealed_samples += num_samples;
}
void StatisticsCalculator::ExpandedNoiseSamplesCorrection(int num_samples) {
expanded_noise_samples_ =
AddIntToSizeTWithLowerCap(num_samples, expanded_noise_samples_);
+ lifetime_stats_.concealed_samples += num_samples;
}
void StatisticsCalculator::PreemptiveExpandedSamples(size_t num_samples) {
preemptive_samples_ += num_samples;
+ lifetime_stats_.concealed_samples += num_samples;
}
void StatisticsCalculator::AcceleratedSamples(size_t num_samples) {
@@ -205,6 +210,7 @@
timestamps_since_last_report_ = 0;
discarded_packets_ = 0;
}
+ lifetime_stats_.total_samples_received += num_samples;
}
void StatisticsCalculator::SecondaryDecodedSamples(int num_samples) {
@@ -307,6 +313,10 @@
Reset();
}
+NetEqLifetimeStatistics StatisticsCalculator::GetLifetimeStatistics() const {
+ return lifetime_stats_;
+}
+
uint16_t StatisticsCalculator::CalculateQ14Ratio(size_t numerator,
uint32_t denominator) {
if (numerator == 0) {
diff --git a/webrtc/modules/audio_coding/neteq/statistics_calculator.h b/webrtc/modules/audio_coding/neteq/statistics_calculator.h
index 2877a16..f261a66 100644
--- a/webrtc/modules/audio_coding/neteq/statistics_calculator.h
+++ b/webrtc/modules/audio_coding/neteq/statistics_calculator.h
@@ -99,6 +99,10 @@
const DecisionLogic& decision_logic,
NetEqNetworkStatistics *stats);
+ // Returns a copy of this class's lifetime statistics. These statistics are
+ // never reset.
+ NetEqLifetimeStatistics GetLifetimeStatistics() const;
+
private:
static const int kMaxReportPeriod = 60; // Seconds before auto-reset.
static const size_t kLenWaitingTimes = 100;
@@ -158,6 +162,8 @@
// Calculates numerator / denominator, and returns the value in Q14.
static uint16_t CalculateQ14Ratio(size_t numerator, uint32_t denominator);
+ // TODO(steveanton): Add unit tests for the lifetime stats.
+ NetEqLifetimeStatistics lifetime_stats_;
size_t preemptive_samples_;
size_t accelerate_samples_;
size_t added_zero_samples_;