Add relative_packet_arrival_delay and jitter_buffer_packets_received statistics.
Bug: webrtc:10333
Change-Id: I415e2286b426cbca940fe3a187957531847272ec
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/124780
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26976}diff --git a/modules/audio_coding/neteq/decision_logic_unittest.cc b/modules/audio_coding/neteq/decision_logic_unittest.cc
index 17840d1..14b5842 100644
--- a/modules/audio_coding/neteq/decision_logic_unittest.cc
+++ b/modules/audio_coding/neteq/decision_logic_unittest.cc
@@ -16,6 +16,7 @@
#include "modules/audio_coding/neteq/delay_manager.h"
#include "modules/audio_coding/neteq/delay_peak_detector.h"
#include "modules/audio_coding/neteq/packet_buffer.h"
+#include "modules/audio_coding/neteq/statistics_calculator.h"
#include "modules/audio_coding/neteq/tick_timer.h"
#include "test/field_trial.h"
#include "test/gtest.h"
@@ -29,10 +30,11 @@
DecoderDatabase decoder_database(
new rtc::RefCountedObject<MockAudioDecoderFactory>, absl::nullopt);
TickTimer tick_timer;
+ StatisticsCalculator stats;
PacketBuffer packet_buffer(10, &tick_timer);
DelayPeakDetector delay_peak_detector(&tick_timer, false);
- auto delay_manager =
- DelayManager::Create(240, 0, false, &delay_peak_detector, &tick_timer);
+ auto delay_manager = DelayManager::Create(240, 0, false, &delay_peak_detector,
+ &tick_timer, &stats);
BufferLevelFilter buffer_level_filter;
DecisionLogic* logic = DecisionLogic::Create(
fs_hz, output_size_samples, false, &decoder_database, packet_buffer,
diff --git a/modules/audio_coding/neteq/delay_manager.cc b/modules/audio_coding/neteq/delay_manager.cc
index 1c7ad19..ab2d48d 100644
--- a/modules/audio_coding/neteq/delay_manager.cc
+++ b/modules/audio_coding/neteq/delay_manager.cc
@@ -20,6 +20,7 @@
#include "absl/memory/memory.h"
#include "modules/audio_coding/neteq/delay_peak_detector.h"
#include "modules/audio_coding/neteq/histogram.h"
+#include "modules/audio_coding/neteq/statistics_calculator.h"
#include "modules/include/module_common_types_public.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
@@ -114,6 +115,7 @@
bool enable_rtx_handling,
DelayPeakDetector* peak_detector,
const TickTimer* tick_timer,
+ StatisticsCalculator* statistics,
std::unique_ptr<Histogram> histogram)
: first_packet_received_(false),
max_packets_in_buffer_(max_packets_in_buffer),
@@ -121,6 +123,7 @@
histogram_quantile_(histogram_quantile),
histogram_mode_(histogram_mode),
tick_timer_(tick_timer),
+ statistics_(statistics),
base_minimum_delay_ms_(base_minimum_delay_ms),
effective_minimum_delay_ms_(base_minimum_delay_ms),
base_target_level_(4), // In Q0 domain.
@@ -150,7 +153,8 @@
int base_minimum_delay_ms,
bool enable_rtx_handling,
DelayPeakDetector* peak_detector,
- const TickTimer* tick_timer) {
+ const TickTimer* tick_timer,
+ StatisticsCalculator* statistics) {
int quantile;
std::unique_ptr<Histogram> histogram;
HistogramMode mode;
@@ -168,7 +172,8 @@
}
return absl::make_unique<DelayManager>(
max_packets_in_buffer, base_minimum_delay_ms, quantile, mode,
- enable_rtx_handling, peak_detector, tick_timer, std::move(histogram));
+ enable_rtx_handling, peak_detector, tick_timer, statistics,
+ std::move(histogram));
}
DelayManager::~DelayManager() {}
@@ -234,16 +239,18 @@
reordered = true;
}
+ int iat_delay = iat_ms - packet_len_ms;
+ int relative_delay;
+ if (reordered) {
+ relative_delay = std::max(iat_delay, 0);
+ } else {
+ UpdateDelayHistory(iat_delay);
+ relative_delay = CalculateRelativePacketArrivalDelay();
+ }
+ statistics_->RelativePacketArrivalDelay(relative_delay);
+
switch (histogram_mode_) {
case RELATIVE_ARRIVAL_DELAY: {
- int iat_delay = iat_ms - packet_len_ms;
- int relative_delay;
- if (reordered) {
- relative_delay = std::max(iat_delay, 0);
- } else {
- UpdateDelayHistory(iat_delay);
- relative_delay = CalculateRelativePacketArrivalDelay();
- }
const int index = relative_delay / kBucketSizeMs;
if (index < histogram_->NumBuckets()) {
// Maximum delay to register is 2000 ms.
diff --git a/modules/audio_coding/neteq/delay_manager.h b/modules/audio_coding/neteq/delay_manager.h
index 11dfeb9..e54e950 100644
--- a/modules/audio_coding/neteq/delay_manager.h
+++ b/modules/audio_coding/neteq/delay_manager.h
@@ -18,6 +18,7 @@
#include "absl/types/optional.h"
#include "modules/audio_coding/neteq/histogram.h"
+#include "modules/audio_coding/neteq/statistics_calculator.h"
#include "modules/audio_coding/neteq/tick_timer.h"
#include "rtc_base/constructor_magic.h"
@@ -40,6 +41,7 @@
bool enable_rtx_handling,
DelayPeakDetector* peak_detector,
const TickTimer* tick_timer,
+ StatisticsCalculator* statistics,
std::unique_ptr<Histogram> histogram);
// Create a DelayManager object. Notify the delay manager that the packet
@@ -51,7 +53,8 @@
int base_minimum_delay_ms,
bool enable_rtx_handling,
DelayPeakDetector* peak_detector,
- const TickTimer* tick_timer);
+ const TickTimer* tick_timer,
+ StatisticsCalculator* statistics);
virtual ~DelayManager();
@@ -174,6 +177,7 @@
const int histogram_quantile_;
const HistogramMode histogram_mode_;
const TickTimer* tick_timer_;
+ StatisticsCalculator* statistics_;
int base_minimum_delay_ms_;
// Provides delay which is used by LimitTargetLevel as lower bound on target
// delay.
diff --git a/modules/audio_coding/neteq/delay_manager_unittest.cc b/modules/audio_coding/neteq/delay_manager_unittest.cc
index 7b57324..c57f074 100644
--- a/modules/audio_coding/neteq/delay_manager_unittest.cc
+++ b/modules/audio_coding/neteq/delay_manager_unittest.cc
@@ -18,6 +18,7 @@
#include "modules/audio_coding/neteq/histogram.h"
#include "modules/audio_coding/neteq/mock/mock_delay_peak_detector.h"
#include "modules/audio_coding/neteq/mock/mock_histogram.h"
+#include "modules/audio_coding/neteq/mock/mock_statistics_calculator.h"
#include "rtc_base/checks.h"
#include "test/field_trial.h"
#include "test/gmock.h"
@@ -53,6 +54,7 @@
std::unique_ptr<DelayManager> dm_;
TickTimer tick_timer_;
+ MockStatisticsCalculator stats_;
MockDelayPeakDetector detector_;
MockHistogram* mock_histogram_;
uint16_t seq_no_;
@@ -81,10 +83,11 @@
dm_ = absl::make_unique<DelayManager>(
kMaxNumberOfPackets, kMinDelayMs, kDefaultHistogramQuantile,
histogram_mode_, enable_rtx_handling_, &detector_, &tick_timer_,
- std::move(histogram));
+ &stats_, std::move(histogram));
} else {
dm_ = DelayManager::Create(kMaxNumberOfPackets, kMinDelayMs,
- enable_rtx_handling_, &detector_, &tick_timer_);
+ enable_rtx_handling_, &detector_, &tick_timer_,
+ &stats_);
}
}
@@ -709,4 +712,17 @@
EXPECT_EQ(0, dm_->Update(seq_no_, ts_, kFs));
}
+TEST_F(DelayManagerTest, RelativeArrivalDelayStatistic) {
+ SetPacketAudioLength(kFrameSizeMs);
+ InsertNextPacket();
+
+ IncreaseTime(kFrameSizeMs);
+ EXPECT_CALL(stats_, RelativePacketArrivalDelay(0));
+ InsertNextPacket();
+
+ IncreaseTime(2 * kFrameSizeMs);
+ EXPECT_CALL(stats_, RelativePacketArrivalDelay(20));
+ InsertNextPacket();
+}
+
} // namespace webrtc
diff --git a/modules/audio_coding/neteq/include/neteq.h b/modules/audio_coding/neteq/include/neteq.h
index a1c0b52..57fd349 100644
--- a/modules/audio_coding/neteq/include/neteq.h
+++ b/modules/audio_coding/neteq/include/neteq.h
@@ -71,9 +71,18 @@
uint64_t concealment_events = 0;
uint64_t jitter_buffer_delay_ms = 0;
uint64_t jitter_buffer_emitted_count = 0;
- // Below stat is not part of the spec.
+ // Below stats are not part of the spec.
uint64_t voice_concealed_samples = 0;
uint64_t delayed_packet_outage_samples = 0;
+ // This is sum of relative packet arrival delays of received packets so far.
+ // Since end-to-end delay of a packet is difficult to measure and is not
+ // necessarily useful for measuring jitter buffer performance, we report a
+ // relative packet arrival delay. The relative packet arrival delay of a
+ // packet is defined as the arrival delay compared to the first packet
+ // received, given that it had zero delay. To avoid clock drift, the "first"
+ // packet can be made dynamic.
+ uint64_t relative_packet_arrival_delay_ms = 0;
+ uint64_t jitter_buffer_packets_received = 0;
};
// Metrics that describe the operations performed in NetEq, and the internal
diff --git a/modules/audio_coding/neteq/mock/mock_delay_manager.h b/modules/audio_coding/neteq/mock/mock_delay_manager.h
index 63dd575..3a128ce 100644
--- a/modules/audio_coding/neteq/mock/mock_delay_manager.h
+++ b/modules/audio_coding/neteq/mock/mock_delay_manager.h
@@ -15,6 +15,7 @@
#include "modules/audio_coding/neteq/delay_manager.h"
#include "modules/audio_coding/neteq/histogram.h"
+#include "modules/audio_coding/neteq/statistics_calculator.h"
#include "test/gmock.h"
namespace webrtc {
@@ -28,6 +29,7 @@
bool enable_rtx_handling,
DelayPeakDetector* peak_detector,
const TickTimer* tick_timer,
+ StatisticsCalculator* stats,
std::unique_ptr<Histogram> histogram)
: DelayManager(max_packets_in_buffer,
base_min_target_delay_ms,
@@ -36,6 +38,7 @@
enable_rtx_handling,
peak_detector,
tick_timer,
+ stats,
std::move(histogram)) {}
virtual ~MockDelayManager() { Die(); }
MOCK_METHOD0(Die, void());
diff --git a/modules/audio_coding/neteq/mock/mock_statistics_calculator.h b/modules/audio_coding/neteq/mock/mock_statistics_calculator.h
index 85f2620..aedb1df 100644
--- a/modules/audio_coding/neteq/mock/mock_statistics_calculator.h
+++ b/modules/audio_coding/neteq/mock/mock_statistics_calculator.h
@@ -21,6 +21,7 @@
public:
MOCK_METHOD1(PacketsDiscarded, void(size_t num_packets));
MOCK_METHOD1(SecondaryPacketsDiscarded, void(size_t num_packets));
+ MOCK_METHOD1(RelativePacketArrivalDelay, void(size_t delay_ms));
};
} // namespace webrtc
diff --git a/modules/audio_coding/neteq/neteq_impl.cc b/modules/audio_coding/neteq/neteq_impl.cc
index 5e6b8bc..f1e8527 100644
--- a/modules/audio_coding/neteq/neteq_impl.cc
+++ b/modules/audio_coding/neteq/neteq_impl.cc
@@ -41,6 +41,7 @@
#include "modules/audio_coding/neteq/post_decode_vad.h"
#include "modules/audio_coding/neteq/preemptive_expand.h"
#include "modules/audio_coding/neteq/red_payload_splitter.h"
+#include "modules/audio_coding/neteq/statistics_calculator.h"
#include "modules/audio_coding/neteq/sync_buffer.h"
#include "modules/audio_coding/neteq/tick_timer.h"
#include "modules/audio_coding/neteq/time_stretch.h"
@@ -58,6 +59,7 @@
const NetEq::Config& config,
const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory)
: tick_timer(new TickTimer),
+ stats(new StatisticsCalculator),
buffer_level_filter(new BufferLevelFilter),
decoder_database(
new DecoderDatabase(decoder_factory, config.codec_pair_id)),
@@ -67,7 +69,8 @@
config.min_delay_ms,
config.enable_rtx_handling,
delay_peak_detector.get(),
- tick_timer.get())),
+ tick_timer.get(),
+ stats.get())),
dtmf_buffer(new DtmfBuffer(config.sample_rate_hz)),
dtmf_tone_generator(new DtmfToneGenerator),
packet_buffer(
@@ -97,6 +100,7 @@
expand_factory_(std::move(deps.expand_factory)),
accelerate_factory_(std::move(deps.accelerate_factory)),
preemptive_expand_factory_(std::move(deps.preemptive_expand_factory)),
+ stats_(std::move(deps.stats)),
last_mode_(kModeNormal),
decoded_buffer_length_(kMaxFrameSize),
decoded_buffer_(new int16_t[decoded_buffer_length_]),
@@ -233,7 +237,7 @@
const std::vector<int> changed_payload_types =
decoder_database_->SetCodecs(codecs);
for (const int pt : changed_payload_types) {
- packet_buffer_->DiscardPacketsWithPayloadType(pt, &stats_);
+ packet_buffer_->DiscardPacketsWithPayloadType(pt, stats_.get());
}
}
@@ -251,7 +255,8 @@
rtc::CritScope lock(&crit_sect_);
int ret = decoder_database_->Remove(rtp_payload_type);
if (ret == DecoderDatabase::kOK || ret == DecoderDatabase::kDecoderNotFound) {
- packet_buffer_->DiscardPacketsWithPayloadType(rtp_payload_type, &stats_);
+ packet_buffer_->DiscardPacketsWithPayloadType(rtp_payload_type,
+ stats_.get());
return kOK;
}
return kFail;
@@ -329,20 +334,21 @@
assert(decision_logic_.get());
const int ms_per_packet = rtc::dchecked_cast<int>(
decision_logic_->packet_length_samples() / (fs_hz_ / 1000));
- stats_.PopulateDelayManagerStats(ms_per_packet, *delay_manager_.get(), stats);
- stats_.GetNetworkStatistics(fs_hz_, total_samples_in_buffers,
- decoder_frame_length_, stats);
+ stats_->PopulateDelayManagerStats(ms_per_packet, *delay_manager_.get(),
+ stats);
+ stats_->GetNetworkStatistics(fs_hz_, total_samples_in_buffers,
+ decoder_frame_length_, stats);
return 0;
}
NetEqLifetimeStatistics NetEqImpl::GetLifetimeStatistics() const {
rtc::CritScope lock(&crit_sect_);
- return stats_.GetLifetimeStatistics();
+ return stats_->GetLifetimeStatistics();
}
NetEqOperationsAndState NetEqImpl::GetOperationsAndState() const {
rtc::CritScope lock(&crit_sect_);
- auto result = stats_.GetOperationsAndState();
+ auto result = stats_->GetOperationsAndState();
result.current_buffer_size_ms =
(packet_buffer_->NumSamplesInBuffer(decoder_frame_length_) +
sync_buffer_->FutureLength()) *
@@ -469,6 +475,7 @@
RTC_LOG_F(LS_ERROR) << "payload is empty";
return kInvalidPointer;
}
+ stats_->ReceivedPacket();
PacketList packet_list;
// Insert packet in a packet list.
@@ -654,7 +661,7 @@
// Insert packets in buffer.
const int ret = packet_buffer_->InsertPacketList(
&parsed_packet_list, *decoder_database_, ¤t_rtp_payload_type_,
- ¤t_cng_rtp_payload_type_, &stats_);
+ ¤t_cng_rtp_payload_type_, stats_.get());
if (ret == PacketBuffer::kFlushed) {
// Reset DSP timestamp etc. if packet buffer flushed.
new_codec_ = true;
@@ -751,8 +758,8 @@
*muted = false;
last_decoded_timestamps_.clear();
tick_timer_->Increment();
- stats_.IncreaseCounter(output_size_samples_, fs_hz_);
- const auto lifetime_stats = stats_.GetLifetimeStatistics();
+ stats_->IncreaseCounter(output_size_samples_, fs_hz_);
+ const auto lifetime_stats = stats_->GetLifetimeStatistics();
expand_uma_logger_.UpdateSampleCounter(lifetime_stats.concealed_samples,
fs_hz_);
speech_expand_uma_logger_.UpdateSampleCounter(
@@ -772,7 +779,7 @@
: 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_, false);
+ stats_->ExpandedNoiseSamples(output_size_samples_, false);
*muted = true;
return 0;
}
@@ -981,7 +988,7 @@
if (!new_codec_) {
const uint32_t five_seconds_samples = 5 * fs_hz_;
packet_buffer_->DiscardOldPackets(end_timestamp, five_seconds_samples,
- &stats_);
+ stats_.get());
}
const Packet* packet = packet_buffer_->PeekNextPacket();
@@ -1001,12 +1008,14 @@
(end_timestamp >= packet->timestamp ||
end_timestamp + generated_noise_samples > packet->timestamp)) {
// Don't use this packet, discard it.
- if (packet_buffer_->DiscardNextPacket(&stats_) != PacketBuffer::kOK) {
+ if (packet_buffer_->DiscardNextPacket(stats_.get()) !=
+ PacketBuffer::kOK) {
assert(false); // Must be ok by design.
}
// Check buffer again.
if (!new_codec_) {
- packet_buffer_->DiscardOldPackets(end_timestamp, 5 * fs_hz_, &stats_);
+ packet_buffer_->DiscardOldPackets(end_timestamp, 5 * fs_hz_,
+ stats_.get());
}
packet = packet_buffer_->PeekNextPacket();
}
@@ -1088,7 +1097,7 @@
decision_logic_->SoftReset();
buffer_level_filter_->Reset();
delay_manager_->Reset();
- stats_.ResetMcu();
+ stats_->ResetMcu();
}
size_t required_samples = output_size_samples_;
@@ -1193,7 +1202,7 @@
// if comfort noise is not played. If comfort noise was just played,
// this adjustment of timestamp is only done to get back in sync with the
// stream timestamp; no loss to report.
- stats_.LostSamples(packet->timestamp - end_timestamp);
+ stats_->LostSamples(packet->timestamp - end_timestamp);
}
if (*operation != kRfc3389Cng) {
@@ -1460,10 +1469,10 @@
// Update in-call and post-call statistics.
if (expand_->MuteFactor(0) == 0) {
// Expand generates only noise.
- stats_.ExpandedNoiseSamplesCorrection(expand_length_correction);
+ stats_->ExpandedNoiseSamplesCorrection(expand_length_correction);
} else {
// Expansion generates more than only noise.
- stats_.ExpandedVoiceSamplesCorrection(expand_length_correction);
+ stats_->ExpandedVoiceSamplesCorrection(expand_length_correction);
}
last_mode_ = kModeMerge;
@@ -1504,12 +1513,12 @@
if (std::all_of(concealment_audio_.cbegin(), concealment_audio_.cend(),
[](int16_t i) { return i == 0; })) {
// Expand operation generates only noise.
- stats_.ExpandedNoiseSamples(concealed_samples_per_channel,
- is_new_concealment_event);
+ stats_->ExpandedNoiseSamples(concealed_samples_per_channel,
+ is_new_concealment_event);
} else {
// Expand operation generates more than only noise.
- stats_.ExpandedVoiceSamples(concealed_samples_per_channel,
- is_new_concealment_event);
+ stats_->ExpandedVoiceSamples(concealed_samples_per_channel,
+ is_new_concealment_event);
}
last_mode_ = kModeCodecPlc;
if (!generated_noise_stopwatch_) {
@@ -1530,10 +1539,10 @@
// Update in-call and post-call statistics.
if (expand_->MuteFactor(0) == 0) {
// Expand operation generates only noise.
- stats_.ExpandedNoiseSamples(length, is_new_concealment_event);
+ stats_->ExpandedNoiseSamples(length, is_new_concealment_event);
} else {
// Expand operation generates more than only noise.
- stats_.ExpandedVoiceSamples(length, is_new_concealment_event);
+ stats_->ExpandedVoiceSamples(length, is_new_concealment_event);
}
last_mode_ = kModeExpand;
@@ -1582,7 +1591,7 @@
Accelerate::ReturnCodes return_code =
accelerate_->Process(decoded_buffer, decoded_length, fast_accelerate,
algorithm_buffer_.get(), &samples_removed);
- stats_.AcceleratedSamples(samples_removed);
+ stats_->AcceleratedSamples(samples_removed);
switch (return_code) {
case Accelerate::kSuccess:
last_mode_ = kModeAccelerateSuccess;
@@ -1660,7 +1669,7 @@
PreemptiveExpand::ReturnCodes return_code = preemptive_expand_->Process(
decoded_buffer, decoded_length, old_borrowed_samples_per_channel,
algorithm_buffer_.get(), &samples_added);
- stats_.PreemptiveExpandedSamples(samples_added);
+ stats_->PreemptiveExpandedSamples(samples_added);
switch (return_code) {
case PreemptiveExpand::kSuccess:
last_mode_ = kModePreemptiveExpandSuccess;
@@ -1875,7 +1884,7 @@
return -1;
}
const uint64_t waiting_time_ms = packet->waiting_time->ElapsedMs();
- stats_.StoreWaitingTime(waiting_time_ms);
+ stats_->StoreWaitingTime(waiting_time_ms);
RTC_DCHECK(!packet->empty());
if (first_packet) {
@@ -1899,7 +1908,7 @@
packet_duration = packet->frame->Duration();
// TODO(ossu): Is this the correct way to track Opus FEC packets?
if (packet->priority.codec_level > 0) {
- stats_.SecondaryDecodedSamples(
+ stats_->SecondaryDecodedSamples(
rtc::dchecked_cast<int>(packet_duration));
}
} else if (!has_cng_packet) {
@@ -1915,7 +1924,7 @@
}
extracted_samples = packet->timestamp - first_timestamp + packet_duration;
- stats_.JitterBufferDelay(packet_duration, waiting_time_ms);
+ stats_->JitterBufferDelay(packet_duration, waiting_time_ms);
packet_list->push_back(std::move(*packet)); // Store packet in list.
packet = absl::nullopt; // Ensure it's never used after the move.
@@ -1943,7 +1952,7 @@
// we could end up in the situation where we never decode anything, since
// all incoming packets are considered too old but the buffer will also
// never be flooded and flushed.
- packet_buffer_->DiscardAllOldPackets(timestamp_, &stats_);
+ packet_buffer_->DiscardAllOldPackets(timestamp_, stats_.get());
}
return rtc::dchecked_cast<int>(extracted_samples);
@@ -1953,7 +1962,7 @@
// Delete objects and create new ones.
expand_.reset(expand_factory_->Create(background_noise_.get(),
sync_buffer_.get(), &random_vector_,
- &stats_, fs_hz, channels));
+ stats_.get(), fs_hz, channels));
merge_.reset(new Merge(fs_hz, channels, expand_.get(), sync_buffer_.get()));
}
diff --git a/modules/audio_coding/neteq/neteq_impl.h b/modules/audio_coding/neteq/neteq_impl.h
index 23b63eb..34a5c71 100644
--- a/modules/audio_coding/neteq/neteq_impl.h
+++ b/modules/audio_coding/neteq/neteq_impl.h
@@ -99,6 +99,7 @@
~Dependencies();
std::unique_ptr<TickTimer> tick_timer;
+ std::unique_ptr<StatisticsCalculator> stats;
std::unique_ptr<BufferLevelFilter> buffer_level_filter;
std::unique_ptr<DecoderDatabase> decoder_database;
std::unique_ptr<DelayPeakDetector> delay_peak_detector;
@@ -361,6 +362,7 @@
RTC_GUARDED_BY(crit_sect_);
const std::unique_ptr<PreemptiveExpandFactory> preemptive_expand_factory_
RTC_GUARDED_BY(crit_sect_);
+ const std::unique_ptr<StatisticsCalculator> stats_ RTC_GUARDED_BY(crit_sect_);
std::unique_ptr<BackgroundNoise> background_noise_ RTC_GUARDED_BY(crit_sect_);
std::unique_ptr<DecisionLogic> decision_logic_ RTC_GUARDED_BY(crit_sect_);
@@ -375,7 +377,6 @@
RTC_GUARDED_BY(crit_sect_);
RandomVector random_vector_ RTC_GUARDED_BY(crit_sect_);
std::unique_ptr<ComfortNoise> comfort_noise_ RTC_GUARDED_BY(crit_sect_);
- StatisticsCalculator stats_ RTC_GUARDED_BY(crit_sect_);
int fs_hz_ RTC_GUARDED_BY(crit_sect_);
int fs_mult_ RTC_GUARDED_BY(crit_sect_);
int last_output_sample_rate_hz_ RTC_GUARDED_BY(crit_sect_);
diff --git a/modules/audio_coding/neteq/neteq_impl_unittest.cc b/modules/audio_coding/neteq/neteq_impl_unittest.cc
index 86fbe9c..5875493 100644
--- a/modules/audio_coding/neteq/neteq_impl_unittest.cc
+++ b/modules/audio_coding/neteq/neteq_impl_unittest.cc
@@ -26,6 +26,7 @@
#include "modules/audio_coding/neteq/mock/mock_red_payload_splitter.h"
#include "modules/audio_coding/neteq/neteq_impl.h"
#include "modules/audio_coding/neteq/preemptive_expand.h"
+#include "modules/audio_coding/neteq/statistics_calculator.h"
#include "modules/audio_coding/neteq/sync_buffer.h"
#include "modules/audio_coding/neteq/timestamp_scaler.h"
#include "rtc_base/numerics/safe_conversions.h"
@@ -100,7 +101,7 @@
config_.max_packets_in_buffer, config_.min_delay_ms, 1020054733,
DelayManager::HistogramMode::INTER_ARRIVAL_TIME,
config_.enable_rtx_handling, delay_peak_detector_, tick_timer_,
- absl::make_unique<Histogram>(50, 32745)));
+ deps.stats.get(), absl::make_unique<Histogram>(50, 32745)));
mock_delay_manager_ = mock.get();
EXPECT_CALL(*mock_delay_manager_, set_streaming_mode(false)).Times(1);
deps.delay_manager = std::move(mock);
diff --git a/modules/audio_coding/neteq/statistics_calculator.cc b/modules/audio_coding/neteq/statistics_calculator.cc
index 7ad1a28..a0e9bca 100644
--- a/modules/audio_coding/neteq/statistics_calculator.cc
+++ b/modules/audio_coding/neteq/statistics_calculator.cc
@@ -258,6 +258,14 @@
buffer_full_counter_.RegisterSample();
}
+void StatisticsCalculator::ReceivedPacket() {
+ ++lifetime_stats_.jitter_buffer_packets_received;
+}
+
+void StatisticsCalculator::RelativePacketArrivalDelay(size_t delay_ms) {
+ lifetime_stats_.relative_packet_arrival_delay_ms += delay_ms;
+}
+
void StatisticsCalculator::LogDelayedPacketOutageEvent(int num_samples,
int fs_hz) {
int outage_duration_ms = num_samples / (fs_hz / 1000);
diff --git a/modules/audio_coding/neteq/statistics_calculator.h b/modules/audio_coding/neteq/statistics_calculator.h
index 1dee643..cb92f37 100644
--- a/modules/audio_coding/neteq/statistics_calculator.h
+++ b/modules/audio_coding/neteq/statistics_calculator.h
@@ -83,9 +83,15 @@
// Reports that |num_samples| samples were decoded from secondary packets.
void SecondaryDecodedSamples(int num_samples);
- // Rerport that the packet buffer was flushed.
+ // Reports that the packet buffer was flushed.
void FlushedPacketBuffer();
+ // Reports that the jitter buffer received a packet.
+ void ReceivedPacket();
+
+ // Reports that a received packet was delayed by |delay_ms| milliseconds.
+ virtual void RelativePacketArrivalDelay(size_t delay_ms);
+
// Logs a delayed packet outage event of |num_samples| expanded at a sample
// rate of |fs_hz|. A delayed packet outage event is defined as an expand
// period caused not by an actual packet loss, but by a delayed packet.
diff --git a/modules/audio_coding/neteq/statistics_calculator_unittest.cc b/modules/audio_coding/neteq/statistics_calculator_unittest.cc
index 0a4901d..1fb8e1c 100644
--- a/modules/audio_coding/neteq/statistics_calculator_unittest.cc
+++ b/modules/audio_coding/neteq/statistics_calculator_unittest.cc
@@ -104,4 +104,28 @@
EXPECT_EQ((50u << 14) / k10MsSamples, stats_output.speech_expand_rate);
}
+TEST(StatisticsCalculator, RelativePacketArrivalDelay) {
+ StatisticsCalculator stats;
+
+ stats.RelativePacketArrivalDelay(50);
+ NetEqLifetimeStatistics stats_output = stats.GetLifetimeStatistics();
+ EXPECT_EQ(50u, stats_output.relative_packet_arrival_delay_ms);
+
+ stats.RelativePacketArrivalDelay(20);
+ stats_output = stats.GetLifetimeStatistics();
+ EXPECT_EQ(70u, stats_output.relative_packet_arrival_delay_ms);
+}
+
+TEST(StatisticsCalculator, ReceivedPacket) {
+ StatisticsCalculator stats;
+
+ stats.ReceivedPacket();
+ NetEqLifetimeStatistics stats_output = stats.GetLifetimeStatistics();
+ EXPECT_EQ(1u, stats_output.jitter_buffer_packets_received);
+
+ stats.ReceivedPacket();
+ stats_output = stats.GetLifetimeStatistics();
+ EXPECT_EQ(2u, stats_output.jitter_buffer_packets_received);
+}
+
} // namespace webrtc