yuzu-android/src/audio_core/time_stretch.cpp

144 lines
4.3 KiB
C++

// Copyright 2016 Citra Emulator Project
// Licensed under GPLv2 or any later version
// Refer to the license.txt file included.
#include <chrono>
#include <cmath>
#include <vector>
#include <SoundTouch.h>
#include "audio_core/audio_core.h"
#include "audio_core/time_stretch.h"
#include "common/common_types.h"
#include "common/logging/log.h"
#include "common/math_util.h"
using steady_clock = std::chrono::steady_clock;
namespace AudioCore {
constexpr double MIN_RATIO = 0.1;
constexpr double MAX_RATIO = 100.0;
static double ClampRatio(double ratio) {
return MathUtil::Clamp(ratio, MIN_RATIO, MAX_RATIO);
}
constexpr double MIN_DELAY_TIME = 0.05; // Units: seconds
constexpr double MAX_DELAY_TIME = 0.25; // Units: seconds
constexpr size_t DROP_FRAMES_SAMPLE_DELAY = 16000; // Units: samples
constexpr double SMOOTHING_FACTOR = 0.007;
struct TimeStretcher::Impl {
soundtouch::SoundTouch soundtouch;
steady_clock::time_point frame_timer = steady_clock::now();
size_t samples_queued = 0;
double smoothed_ratio = 1.0;
double sample_rate = static_cast<double>(native_sample_rate);
};
std::vector<s16> TimeStretcher::Process(size_t samples_in_queue) {
// This is a very simple algorithm without any fancy control theory. It works and is stable.
double ratio = CalculateCurrentRatio();
ratio = CorrectForUnderAndOverflow(ratio, samples_in_queue);
impl->smoothed_ratio =
(1.0 - SMOOTHING_FACTOR) * impl->smoothed_ratio + SMOOTHING_FACTOR * ratio;
impl->smoothed_ratio = ClampRatio(impl->smoothed_ratio);
// SoundTouch's tempo definition the inverse of our ratio definition.
impl->soundtouch.setTempo(1.0 / impl->smoothed_ratio);
std::vector<s16> samples = GetSamples();
if (samples_in_queue >= DROP_FRAMES_SAMPLE_DELAY) {
samples.clear();
LOG_DEBUG(Audio, "Dropping frames!");
}
return samples;
}
TimeStretcher::TimeStretcher() : impl(std::make_unique<Impl>()) {
impl->soundtouch.setPitch(1.0);
impl->soundtouch.setChannels(2);
impl->soundtouch.setSampleRate(native_sample_rate);
Reset();
}
TimeStretcher::~TimeStretcher() {
impl->soundtouch.clear();
}
void TimeStretcher::SetOutputSampleRate(unsigned int sample_rate) {
impl->sample_rate = static_cast<double>(sample_rate);
impl->soundtouch.setRate(static_cast<double>(native_sample_rate) / impl->sample_rate);
}
void TimeStretcher::AddSamples(const s16* buffer, size_t num_samples) {
impl->soundtouch.putSamples(buffer, static_cast<uint>(num_samples));
impl->samples_queued += num_samples;
}
void TimeStretcher::Flush() {
impl->soundtouch.flush();
}
void TimeStretcher::Reset() {
impl->soundtouch.setTempo(1.0);
impl->soundtouch.clear();
impl->smoothed_ratio = 1.0;
impl->frame_timer = steady_clock::now();
impl->samples_queued = 0;
SetOutputSampleRate(native_sample_rate);
}
double TimeStretcher::CalculateCurrentRatio() {
const steady_clock::time_point now = steady_clock::now();
const std::chrono::duration<double> duration = now - impl->frame_timer;
const double expected_time =
static_cast<double>(impl->samples_queued) / static_cast<double>(native_sample_rate);
const double actual_time = duration.count();
double ratio;
if (expected_time != 0) {
ratio = ClampRatio(actual_time / expected_time);
} else {
ratio = impl->smoothed_ratio;
}
impl->frame_timer = now;
impl->samples_queued = 0;
return ratio;
}
double TimeStretcher::CorrectForUnderAndOverflow(double ratio, size_t sample_delay) const {
const size_t min_sample_delay = static_cast<size_t>(MIN_DELAY_TIME * impl->sample_rate);
const size_t max_sample_delay = static_cast<size_t>(MAX_DELAY_TIME * impl->sample_rate);
if (sample_delay < min_sample_delay) {
// Make the ratio bigger.
ratio = ratio > 1.0 ? ratio * ratio : sqrt(ratio);
} else if (sample_delay > max_sample_delay) {
// Make the ratio smaller.
ratio = ratio > 1.0 ? sqrt(ratio) : ratio * ratio;
}
return ClampRatio(ratio);
}
std::vector<s16> TimeStretcher::GetSamples() {
uint available = impl->soundtouch.numSamples();
std::vector<s16> output(static_cast<size_t>(available) * 2);
impl->soundtouch.receiveSamples(output.data(), available);
return output;
}
} // namespace AudioCore