mirror of
https://github.com/Lime3DS/Lime3DS
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146 lines
4.3 KiB
C++
146 lines
4.3 KiB
C++
// Copyright 2016 Citra Emulator Project
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// Licensed under GPLv2 or any later version
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// Refer to the license.txt file included.
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#include <chrono>
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#include <cmath>
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#include <vector>
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#include <SoundTouch.h>
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#include "audio_core/audio_core.h"
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#include "audio_core/time_stretch.h"
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#include "common/common_types.h"
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#include "common/logging/log.h"
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#include "common/math_util.h"
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using steady_clock = std::chrono::steady_clock;
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namespace AudioCore {
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constexpr double MIN_RATIO = 0.1;
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constexpr double MAX_RATIO = 100.0;
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static double ClampRatio(double ratio) {
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return MathUtil::Clamp(ratio, MIN_RATIO, MAX_RATIO);
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}
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constexpr double MIN_DELAY_TIME = 0.05; // Units: seconds
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constexpr double MAX_DELAY_TIME = 0.25; // Units: seconds
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constexpr size_t DROP_FRAMES_SAMPLE_DELAY = 16000; // Units: samples
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constexpr double SMOOTHING_FACTOR = 0.007;
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struct TimeStretcher::Impl {
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soundtouch::SoundTouch soundtouch;
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steady_clock::time_point frame_timer = steady_clock::now();
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size_t samples_queued = 0;
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double smoothed_ratio = 1.0;
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double sample_rate = static_cast<double>(native_sample_rate);
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};
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std::vector<s16> TimeStretcher::Process(size_t samples_in_queue) {
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// This is a very simple algorithm without any fancy control theory. It works and is stable.
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double ratio = CalculateCurrentRatio();
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ratio = CorrectForUnderAndOverflow(ratio, samples_in_queue);
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impl->smoothed_ratio =
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(1.0 - SMOOTHING_FACTOR) * impl->smoothed_ratio + SMOOTHING_FACTOR * ratio;
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impl->smoothed_ratio = ClampRatio(impl->smoothed_ratio);
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// SoundTouch's tempo definition the inverse of our ratio definition.
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impl->soundtouch.setTempo(1.0 / impl->smoothed_ratio);
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std::vector<s16> samples = GetSamples();
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if (samples_in_queue >= DROP_FRAMES_SAMPLE_DELAY) {
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samples.clear();
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LOG_DEBUG(Audio, "Dropping frames!");
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}
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return samples;
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}
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TimeStretcher::TimeStretcher() : impl(std::make_unique<Impl>()) {
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impl->soundtouch.setPitch(1.0);
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impl->soundtouch.setChannels(2);
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impl->soundtouch.setSampleRate(native_sample_rate);
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Reset();
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}
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TimeStretcher::~TimeStretcher() {
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impl->soundtouch.clear();
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}
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void TimeStretcher::SetOutputSampleRate(unsigned int sample_rate) {
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impl->sample_rate = static_cast<double>(sample_rate);
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impl->soundtouch.setRate(static_cast<double>(native_sample_rate) / impl->sample_rate);
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}
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void TimeStretcher::AddSamples(const s16* buffer, size_t num_samples) {
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impl->soundtouch.putSamples(buffer, static_cast<uint>(num_samples));
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impl->samples_queued += num_samples;
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}
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void TimeStretcher::Flush() {
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impl->soundtouch.flush();
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}
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void TimeStretcher::Reset() {
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impl->soundtouch.setTempo(1.0);
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impl->soundtouch.clear();
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impl->smoothed_ratio = 1.0;
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impl->frame_timer = steady_clock::now();
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impl->samples_queued = 0;
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SetOutputSampleRate(native_sample_rate);
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}
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double TimeStretcher::CalculateCurrentRatio() {
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const steady_clock::time_point now = steady_clock::now();
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const std::chrono::duration<double> duration = now - impl->frame_timer;
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const double expected_time =
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static_cast<double>(impl->samples_queued) / static_cast<double>(native_sample_rate);
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const double actual_time = duration.count();
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double ratio;
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if (expected_time != 0) {
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ratio = ClampRatio(actual_time / expected_time);
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} else {
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ratio = impl->smoothed_ratio;
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}
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impl->frame_timer = now;
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impl->samples_queued = 0;
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return ratio;
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}
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double TimeStretcher::CorrectForUnderAndOverflow(double ratio, size_t sample_delay) const {
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const size_t min_sample_delay = static_cast<size_t>(MIN_DELAY_TIME * impl->sample_rate);
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const size_t max_sample_delay = static_cast<size_t>(MAX_DELAY_TIME * impl->sample_rate);
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if (sample_delay < min_sample_delay) {
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// Make the ratio bigger.
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ratio = ratio > 1.0 ? ratio * ratio : sqrt(ratio);
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} else if (sample_delay > max_sample_delay) {
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// Make the ratio smaller.
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ratio = ratio > 1.0 ? sqrt(ratio) : ratio * ratio;
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}
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return ClampRatio(ratio);
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}
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std::vector<s16> TimeStretcher::GetSamples() {
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uint available = impl->soundtouch.numSamples();
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std::vector<s16> output(static_cast<size_t>(available) * 2);
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impl->soundtouch.receiveSamples(output.data(), available);
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return output;
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}
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} // namespace AudioCore
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