forked from dolphin-emu/dolphin
		
	
		
			
				
	
	
		
			90 lines
		
	
	
		
			3.1 KiB
		
	
	
	
		
			C++
		
	
	
	
	
	
			
		
		
	
	
			90 lines
		
	
	
		
			3.1 KiB
		
	
	
	
		
			C++
		
	
	
	
	
	
// Copyright 2017 Dolphin Emulator Project
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// Licensed under GPLv2+
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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 "AudioCommon/AudioStretcher.h"
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#include "Common/Logging/Log.h"
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#include "Core/ConfigManager.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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  m_sound_touch.setSetting(SETTING_USE_QUICKSEEK, 0);
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  m_sound_touch.setSetting(SETTING_SEQUENCE_MS, 62);
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  m_sound_touch.setSetting(SETTING_SEEKWINDOW_MS, 28);
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  m_sound_touch.setSetting(SETTING_OVERLAP_MS, 8);
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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 = SConfig::GetInstance().m_audio_stretch_max_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(AUDIO, "Audio stretching: samples:%u/%u ratio:%f backlog:%f gain: %f", 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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}
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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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