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8ea6bef98f
While trying to work on adding audiodump support for CLI, I was alerted that it was important to first try moving the DSP configs to the new config before continuing, as that makes it substantially easier to write clean code to add such a feature. This commit aims to allow for Dolphin to only rely on the new config for DSP-related settings.
85 lines
2.9 KiB
C++
85 lines
2.9 KiB
C++
// Copyright 2017 Dolphin Emulator Project
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// SPDX-License-Identifier: GPL-2.0-or-later
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#include <algorithm>
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#include <cmath>
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#include <cstddef>
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#include "AudioCommon/AudioStretcher.h"
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#include "Common/Logging/Log.h"
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#include "Core/Config/MainSettings.h"
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namespace AudioCommon
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{
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AudioStretcher::AudioStretcher(unsigned int sample_rate) : m_sample_rate(sample_rate)
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{
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m_sound_touch.setChannels(2);
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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 AudioStretcher::Clear()
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{
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m_sound_touch.clear();
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}
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void AudioStretcher::ProcessSamples(const short* in, unsigned int num_in, unsigned int num_out)
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{
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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 = Config::Get(Config::MAIN_AUDIO_STRETCH_LATENCY);
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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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{
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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 = 0.5; // seconds
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current_ratio *= 1.0 + 2.0 * (backlog_fullness - 0.5) * (time_delta / tweak_time_scale);
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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 = 1.0; // 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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DEBUG_LOG_FMT(AUDIO, "Audio stretching: samples:{}/{} ratio:{} backlog:{} gain: {}", num_in,
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num_out, m_stretch_ratio, backlog_fullness, lpf_gain);
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m_sound_touch.putSamples(in, num_in);
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}
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void AudioStretcher::GetStretchedSamples(short* out, unsigned int num_out)
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{
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const size_t samples_received = m_sound_touch.receiveSamples(out, num_out);
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if (samples_received != 0)
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{
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m_last_stretched_sample[0] = out[samples_received * 2 - 2];
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m_last_stretched_sample[1] = out[samples_received * 2 - 1];
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}
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// Perform padding if we've run out of samples.
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for (size_t i = samples_received; i < num_out; i++)
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{
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out[i * 2 + 0] = m_last_stretched_sample[0];
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out[i * 2 + 1] = m_last_stretched_sample[1];
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}
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}
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} // namespace AudioCommon
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