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Migrate CallSimulator to the new perf metrics logging API

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Bug: b/246095034
Change-Id: I613f702d2f469b6bc8d1634f8dda40d444ff7cf2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276632
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38213}
This commit is contained in:
Artem Titov 2022-09-26 16:29:48 +02:00 committed by WebRTC LUCI CQ
parent e2f2cae3fb
commit 7fee2f7908
4 changed files with 27 additions and 45 deletions
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4
api/numerics/samples_stats_counter.cc
View file

@ -19,6 +19,10 @@
namespace webrtc {
SamplesStatsCounter::SamplesStatsCounter() = default;
SamplesStatsCounter::SamplesStatsCounter(size_t expected_samples_count) {
samples_.reserve(expected_samples_count);
}
SamplesStatsCounter::~SamplesStatsCounter() = default;
SamplesStatsCounter::SamplesStatsCounter(const SamplesStatsCounter&) = default;
SamplesStatsCounter& SamplesStatsCounter::operator=(

1
api/numerics/samples_stats_counter.h
View file

@ -34,6 +34,7 @@ class SamplesStatsCounter {
};
SamplesStatsCounter();
explicit SamplesStatsCounter(size_t expected_samples_count);
~SamplesStatsCounter();
SamplesStatsCounter(const SamplesStatsCounter&);
SamplesStatsCounter& operator=(const SamplesStatsCounter&);

4
modules/audio_processing/BUILD.gn
View file

@ -498,13 +498,15 @@ if (rtc_include_tests) {
":audio_processing",
":audioproc_test_utils",
"../../api:array_view",
"../../api/numerics",
"../../api/test/metrics:global_metrics_logger_and_exporter",
"../../api/test/metrics:metric",
"../../rtc_base:platform_thread",
"../../rtc_base:protobuf_utils",
"../../rtc_base:random",
"../../rtc_base:rtc_event",
"../../rtc_base:safe_conversions",
"../../system_wrappers",
"../../test:perf_test",
"../../test:test_support",
]
absl_deps = [ "//third_party/abseil-cpp/absl/strings" ]

63
modules/audio_processing/audio_processing_performance_unittest.cc
View file

@ -16,6 +16,9 @@
#include "absl/strings/string_view.h"
#include "api/array_view.h"
#include "api/numerics/samples_stats_counter.h"
#include "api/test/metrics/global_metrics_logger_and_exporter.h"
#include "api/test/metrics/metric.h"
#include "modules/audio_processing/audio_processing_impl.h"
#include "modules/audio_processing/test/audio_processing_builder_for_testing.h"
#include "modules/audio_processing/test/test_utils.h"
@ -25,13 +28,14 @@
#include "rtc_base/random.h"
#include "system_wrappers/include/clock.h"
#include "test/gtest.h"
#include "test/testsupport/perf_test.h"
namespace webrtc {
namespace {
static const bool kPrintAllDurations = false;
using ::webrtc::test::GetGlobalMetricsLogger;
using ::webrtc::test::ImprovementDirection;
using ::webrtc::test::Metric;
using ::webrtc::test::Unit;
class CallSimulator;
@ -203,11 +207,11 @@ class TimedThreadApiProcessor {
frame_data_(kMaxFrameSize),
clock_(webrtc::Clock::GetRealTimeClock()),
num_durations_to_store_(num_durations_to_store),
api_call_durations_(num_durations_to_store_ - kNumInitializationFrames),
samples_count_(0),
input_level_(input_level),
processor_type_(processor_type),
num_channels_(num_channels) {
api_call_durations_.reserve(num_durations_to_store_);
}
num_channels_(num_channels) {}
// Implements the callback functionality for the threads.
bool Process();
@ -219,21 +223,17 @@ class TimedThreadApiProcessor {
const std::string sample_rate_name =
"_" + std::to_string(simulation_config_->sample_rate_hz) + "Hz";
webrtc::test::PrintResultMeanAndError(
"apm_timing", sample_rate_name, processor_name, GetDurationAverage(),
GetDurationStandardDeviation(), "us", false);
if (kPrintAllDurations) {
webrtc::test::PrintResultList("apm_call_durations", sample_rate_name,
processor_name, api_call_durations_, "us",
false);
}
GetGlobalMetricsLogger()->LogMetric(
"apm_timing" + sample_rate_name, processor_name, api_call_durations_,
Unit::kMilliseconds, ImprovementDirection::kNeitherIsBetter);
}
void AddDuration(int64_t duration) {
if (api_call_durations_.size() < num_durations_to_store_) {
api_call_durations_.push_back(duration);
if (samples_count_ >= kNumInitializationFrames &&
samples_count_ < num_durations_to_store_) {
api_call_durations_.AddSample(duration);
}
samples_count_++;
}
private:
@ -241,32 +241,6 @@ class TimedThreadApiProcessor {
static const int kMaxFrameSize = 480;
static const int kNumInitializationFrames = 5;
int64_t GetDurationStandardDeviation() const {
double variance = 0;
const int64_t average_duration = GetDurationAverage();
for (size_t k = kNumInitializationFrames; k < api_call_durations_.size();
k++) {
int64_t tmp = api_call_durations_[k] - average_duration;
variance += static_cast<double>(tmp * tmp);
}
const int denominator = rtc::checked_cast<int>(api_call_durations_.size()) -
kNumInitializationFrames;
return (denominator > 0
? rtc::checked_cast<int64_t>(sqrt(variance / denominator))
: -1);
}
int64_t GetDurationAverage() const {
int64_t average_duration = 0;
for (size_t k = kNumInitializationFrames; k < api_call_durations_.size();
k++) {
average_duration += api_call_durations_[k];
}
const int denominator = rtc::checked_cast<int>(api_call_durations_.size()) -
kNumInitializationFrames;
return (denominator > 0 ? average_duration / denominator : -1);
}
int ProcessCapture() {
// Set the stream delay.
apm_->set_stream_delay_ms(30);
@ -382,7 +356,8 @@ class TimedThreadApiProcessor {
AudioFrameData frame_data_;
webrtc::Clock* clock_;
const size_t num_durations_to_store_;
std::vector<double> api_call_durations_;
SamplesStatsCounter api_call_durations_;
size_t samples_count_ = 0;
const float input_level_;
bool first_process_call_ = true;
const ProcessorType processor_type_;