65 lines
2.6 KiB
C++
65 lines
2.6 KiB
C++
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// Copyright 2018 yuzu 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 <algorithm>
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#include <cmath>
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#include <cstddef>
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#include "audio_core/time_stretch.h"
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#include "common/logging/log.h"
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namespace AudioCore {
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TimeStretcher::TimeStretcher(u32 sample_rate, u32 channel_count)
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: m_sample_rate(sample_rate), m_channel_count(channel_count) {
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m_sound_touch.setChannels(channel_count);
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m_sound_touch.setSampleRate(sample_rate);
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m_sound_touch.setPitch(1.0);
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m_sound_touch.setTempo(1.0);
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}
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void TimeStretcher::Clear() {
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m_sound_touch.clear();
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}
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size_t TimeStretcher::Process(const s16* in, size_t num_in, s16* out, size_t num_out) {
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const double time_delta = static_cast<double>(num_out) / m_sample_rate; // seconds
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// We were given actual_samples number of samples, and num_samples were requested from us.
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double current_ratio = static_cast<double>(num_in) / static_cast<double>(num_out);
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const double max_latency = 0.3; // seconds
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const double max_backlog = m_sample_rate * max_latency / 1000.0 / m_stretch_ratio;
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const double backlog_fullness = m_sound_touch.numSamples() / max_backlog;
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if (backlog_fullness > 5.0) {
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// Too many samples in backlog: Don't push anymore on
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num_in = 0;
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}
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// We ideally want the backlog to be about 50% full.
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// This gives some headroom both ways to prevent underflow and overflow.
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// We tweak current_ratio to encourage this.
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constexpr double tweak_time_scale = 0.05; // seconds
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const double tweak_correction = (backlog_fullness - 0.5) * (time_delta / tweak_time_scale);
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current_ratio *= std::pow(1.0 + 2.0 * tweak_correction, tweak_correction < 0 ? 3.0 : 1.0);
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// This low-pass filter smoothes out variance in the calculated stretch ratio.
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// The time-scale determines how responsive this filter is.
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constexpr double lpf_time_scale = 2.0; // seconds
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const double lpf_gain = 1.0 - std::exp(-time_delta / lpf_time_scale);
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m_stretch_ratio += lpf_gain * (current_ratio - m_stretch_ratio);
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// Place a lower limit of 10% speed. When a game boots up, there will be
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// many silence samples. These do not need to be timestretched.
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m_stretch_ratio = std::max(m_stretch_ratio, 0.1);
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m_sound_touch.setTempo(m_stretch_ratio);
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LOG_DEBUG(Audio, "Audio Stretching: samples:{}/{} ratio:{} backlog:{} gain: {}", num_in, num_out,
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m_stretch_ratio, backlog_fullness, lpf_gain);
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m_sound_touch.putSamples(in, num_in);
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return m_sound_touch.receiveSamples(out, num_out);
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}
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} // namespace AudioCore
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