Merge pull request #13352 from Sam-Belliveau/granular_synthesis

AudioCommon: Granule Synthesis Audio System

1 files changed, 0 insertions, 85 deletions

diff --git a/Source/Core/AudioCommon/AudioStretcher.cpp b/Source/Core/AudioCommon/AudioStretcher.cpp
deleted file mode 100644
index a5e54b8163..0000000000
--- a/Source/Core/AudioCommon/AudioStretcher.cpp
+++ /dev/null
@@ -1,85 +0,0 @@
-// Copyright 2017 Dolphin Emulator Project
-// SPDX-License-Identifier: GPL-2.0-or-later
-
-#include "AudioCommon/AudioStretcher.h"
-
-#include <algorithm>
-#include <cmath>
-#include <cstddef>
-
-#include "Common/Logging/Log.h"
-#include "Core/Config/MainSettings.h"
-
-namespace AudioCommon
-{
-AudioStretcher::AudioStretcher(unsigned int sample_rate) : m_sample_rate(sample_rate)
-{
-  m_sound_touch.setChannels(2);
-  m_sound_touch.setSampleRate(sample_rate);
-  m_sound_touch.setPitch(1.0);
-  m_sound_touch.setTempo(1.0);
-}
-
-void AudioStretcher::Clear()
-{
-  m_sound_touch.clear();
-}
-
-void AudioStretcher::ProcessSamples(const short* in, unsigned int num_in, unsigned int num_out)
-{
-  const double time_delta = static_cast<double>(num_out) / m_sample_rate;  // seconds
-
-  // We were given actual_samples number of samples, and num_samples were requested from us.
-  double current_ratio = static_cast<double>(num_in) / static_cast<double>(num_out);
-
-  const double max_latency = Config::Get(Config::MAIN_AUDIO_STRETCH_LATENCY);
-  const double max_backlog = m_sample_rate * max_latency / 1000.0 / m_stretch_ratio;
-  const double backlog_fullness = m_sound_touch.numSamples() / max_backlog;
-  if (backlog_fullness > 5.0)
-  {
-    // Too many samples in backlog: Don't push anymore on
-    num_in = 0;
-  }
-
-  // We ideally want the backlog to be about 50% full.
-  // This gives some headroom both ways to prevent underflow and overflow.
-  // We tweak current_ratio to encourage this.
-  constexpr double tweak_time_scale = 0.5;  // seconds
-  current_ratio *= 1.0 + 2.0 * (backlog_fullness - 0.5) * (time_delta / tweak_time_scale);
-
-  // This low-pass filter smoothes out variance in the calculated stretch ratio.
-  // The time-scale determines how responsive this filter is.
-  constexpr double lpf_time_scale = 1.0;  // seconds
-  const double lpf_gain = 1.0 - std::exp(-time_delta / lpf_time_scale);
-  m_stretch_ratio += lpf_gain * (current_ratio - m_stretch_ratio);
-
-  // Place a lower limit of 10% speed.  When a game boots up, there will be
-  // many silence samples.  These do not need to be timestretched.
-  m_stretch_ratio = std::max(m_stretch_ratio, 0.1);
-  m_sound_touch.setTempo(m_stretch_ratio);
-
-  DEBUG_LOG_FMT(AUDIO, "Audio stretching: samples:{}/{} ratio:{} backlog:{} gain: {}", num_in,
-                num_out, m_stretch_ratio, backlog_fullness, lpf_gain);
-
-  m_sound_touch.putSamples(in, num_in);
-}
-
-void AudioStretcher::GetStretchedSamples(short* out, unsigned int num_out)
-{
-  const size_t samples_received = m_sound_touch.receiveSamples(out, num_out);
-
-  if (samples_received != 0)
-  {
-    m_last_stretched_sample[0] = out[samples_received * 2 - 2];
-    m_last_stretched_sample[1] = out[samples_received * 2 - 1];
-  }
-
-  // Perform padding if we've run out of samples.
-  for (size_t i = samples_received; i < num_out; i++)
-  {
-    out[i * 2 + 0] = m_last_stretched_sample[0];
-    out[i * 2 + 1] = m_last_stretched_sample[1];
-  }
-}
-
-}  // namespace AudioCommon