NetEq: Implement logging of Delayed Packet Outage Events
Measures the duration of each packet loss concealment (a.k.a. expand)
event that is not followed by a merge operation.
Having decoded and played packet m−1, the next expected packet is
m. If packet m arrives after some time of packet loss concealment, we
have a delayed packet outage event. However, if instead packet n>m
arrives, we have a lost packet outage event. In NetEq, the two outage
types results in different operations. Both types start with expand
operations to generate audio to play while the buffer is empty. When a
lost packet outage happens, the expand operation(s) are followed by
one merge operation. For delayed packet outages, merge is not done,
and the expand operations are immediately followed by normal
operations.
This change also includes unit tests for the new statistics.
BUG=webrtc:4915, chromium:488124
R=minyue@webrtc.org
Review URL: https://codereview.webrtc.org/1290113002 .
Cr-Commit-Position: refs/heads/master@{#9725}
diff --git a/webrtc/modules/audio_coding/neteq/expand.cc b/webrtc/modules/audio_coding/neteq/expand.cc
index ae12e50..d01465a 100644
--- a/webrtc/modules/audio_coding/neteq/expand.cc
+++ b/webrtc/modules/audio_coding/neteq/expand.cc
@@ -16,10 +16,12 @@
#include <algorithm> // min, max
#include <limits> // numeric_limits<T>
+#include "webrtc/base/safe_conversions.h"
#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
#include "webrtc/modules/audio_coding/neteq/background_noise.h"
#include "webrtc/modules/audio_coding/neteq/dsp_helper.h"
#include "webrtc/modules/audio_coding/neteq/random_vector.h"
+#include "webrtc/modules/audio_coding/neteq/statistics_calculator.h"
#include "webrtc/modules/audio_coding/neteq/sync_buffer.h"
namespace webrtc {
@@ -27,6 +29,7 @@
Expand::Expand(BackgroundNoise* background_noise,
SyncBuffer* sync_buffer,
RandomVector* random_vector,
+ StatisticsCalculator* statistics,
int fs,
size_t num_channels)
: random_vector_(random_vector),
@@ -36,10 +39,12 @@
num_channels_(num_channels),
consecutive_expands_(0),
background_noise_(background_noise),
+ statistics_(statistics),
overlap_length_(5 * fs / 8000),
lag_index_direction_(0),
current_lag_index_(0),
stop_muting_(false),
+ expand_duration_samples_(0),
channel_parameters_(new ChannelParameters[num_channels_]) {
assert(fs == 8000 || fs == 16000 || fs == 32000 || fs == 48000);
assert(fs <= kMaxSampleRate); // Should not be possible.
@@ -78,6 +83,7 @@
// Perform initial setup if this is the first expansion since last reset.
AnalyzeSignal(random_vector);
first_expand_ = false;
+ expand_duration_samples_ = 0;
} else {
// This is not the first expansion, parameters are already estimated.
// Extract a noise segment.
@@ -298,6 +304,10 @@
// Increase call number and cap it.
consecutive_expands_ = consecutive_expands_ >= kMaxConsecutiveExpands ?
kMaxConsecutiveExpands : consecutive_expands_ + 1;
+ expand_duration_samples_ += output->Size();
+ // Clamp the duration counter at 2 seconds.
+ expand_duration_samples_ =
+ std::min(expand_duration_samples_, rtc::checked_cast<size_t>(fs_hz_ * 2));
return 0;
}
@@ -305,6 +315,8 @@
current_lag_index_ = 0;
lag_index_direction_ = 0;
stop_muting_ = true; // Do not mute signal any more.
+ statistics_->LogDelayedPacketOutageEvent(
+ rtc::checked_cast<int>(expand_duration_samples_) / (fs_hz_ / 1000));
}
void Expand::SetParametersForMergeAfterExpand() {
@@ -833,10 +845,11 @@
Expand* ExpandFactory::Create(BackgroundNoise* background_noise,
SyncBuffer* sync_buffer,
RandomVector* random_vector,
+ StatisticsCalculator* statistics,
int fs,
size_t num_channels) const {
- return new Expand(background_noise, sync_buffer, random_vector, fs,
- num_channels);
+ return new Expand(background_noise, sync_buffer, random_vector, statistics,
+ fs, num_channels);
}
// TODO(turajs): This can be moved to BackgroundNoise class.