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media / base / audio_limiter_perftest.cc [blame]
// Copyright 2024 The Chromium Authors
// Use of this source code is governed by a BSD-style license that can be
// found in the LICENSE file.
#include "media/base/audio_limiter.h"
#include <memory>
#include "base/functional/callback_helpers.h"
#include "base/time/time.h"
#include "media/audio/simple_sources.h"
#include "media/base/audio_timestamp_helper.h"
#include "media/base/fake_audio_render_callback.h"
#include "testing/gtest/include/gtest/gtest.h"
#include "testing/perf/perf_result_reporter.h"
#ifdef UNSAFE_BUFFERS_BUILD
// TODO(crbug.com/41494069): Update these tests once AudioBus is spanified..
#pragma allow_unsafe_buffers
#endif
namespace media {
constexpr int kSampleRate = 48000;
constexpr int kBenchmarkIterations = 20000;
constexpr base::TimeDelta kBufferDuration = base::Milliseconds(20);
void RunConvertBenchmark(const AudioParameters& params,
float amplitude,
const std::string& trace_name) {
constexpr int kFrequency = 440;
SineWaveAudioSource sine_source(params.channels(), kFrequency,
params.sample_rate());
auto input_bus = AudioBus::Create(params);
auto output_bus = AudioBus::Create(params);
sine_source.OnMoreData(base::TimeDelta(), base::TimeTicks(), {},
input_bus.get());
for (int ch = 0; ch < input_bus->channels(); ++ch) {
float* channel_data = input_bus->channel(ch);
for (int i = 0; i < input_bus->frames(); ++i) {
channel_data[i] *= amplitude;
}
}
AudioLimiter::OutputChannels output_channels;
for (int ch = 0; ch < output_bus->channels(); ++ch) {
output_channels.emplace_back(
reinterpret_cast<uint8_t*>(output_bus->channel(ch)),
output_bus->frames() * sizeof(float));
}
AudioLimiter limiter(params.sample_rate(), params.channels());
base::TimeTicks start = base::TimeTicks::Now();
for (int i = 0; i < kBenchmarkIterations; ++i) {
limiter.LimitPeaks(*input_bus, output_channels, base::DoNothing());
}
base::TimeDelta elapsed_time = base::TimeTicks::Now() - start;
base::TimeDelta benchmark_data_duration =
kBenchmarkIterations * kBufferDuration;
// How may ms of data can the AudioLimiter process in 1ms. Higher is better.
double processing_ratio = std::round(benchmark_data_duration / elapsed_time);
perf_test::PerfResultReporter reporter("audio_limiter", trace_name);
reporter.RegisterImportantMetric("", "ms_of_data/ms");
reporter.AddResult("", processing_ratio);
}
TEST(AudioLimiterBenchmark, LimitPeaksBenchmark_NoLimiting) {
// Create input and output parameters to convert between the two most common
// sets of parameters (as indicated via UMA data).
AudioParameters input_params(
AudioParameters::AUDIO_PCM_LINEAR, media::ChannelLayoutConfig::Mono(),
kSampleRate,
AudioTimestampHelper::TimeToFrames(kBufferDuration, kSampleRate));
RunConvertBenchmark(input_params, 0.5, "NoLimitting");
}
TEST(AudioLimiterBenchmark, LimitPeaksBenchmark_Limiting) {
// Create input and output parameters to convert between the two most common
// sets of parameters (as indicated via UMA data).
AudioParameters input_params(
AudioParameters::AUDIO_PCM_LINEAR, media::ChannelLayoutConfig::Mono(),
kSampleRate,
AudioTimestampHelper::TimeToFrames(kBufferDuration, kSampleRate));
RunConvertBenchmark(input_params, 2.0, "Limitting");
}
} // namespace media