Merge pull request #5311 from ligfx/mixerdpl2

AudioCommon: Move DPL2 decoding into Mixer

1 files changed, 89 insertions, 0 deletions

diff --git a/Source/Core/AudioCommon/AudioStretcher.cpp b/Source/Core/AudioCommon/AudioStretcher.cpp
new file mode 100644
index 0000000000..88721b0e35
--- /dev/null
+++ b/Source/Core/AudioCommon/AudioStretcher.cpp
@@ -0,0 +1,89 @@
+// Copyright 2017 Dolphin Emulator Project
+// Licensed under GPLv2+
+// Refer to the license.txt file included.
+
+#include <algorithm>
+#include <cmath>
+#include <cstddef>
+
+#include "AudioCommon/AudioStretcher.h"
+#include "Common/Logging/Log.h"
+#include "Core/ConfigManager.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);
+  m_sound_touch.setSetting(SETTING_USE_QUICKSEEK, 0);
+  m_sound_touch.setSetting(SETTING_SEQUENCE_MS, 62);
+  m_sound_touch.setSetting(SETTING_SEEKWINDOW_MS, 28);
+  m_sound_touch.setSetting(SETTING_OVERLAP_MS, 8);
+}
+
+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 = SConfig::GetInstance().m_audio_stretch_max_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(AUDIO, "Audio stretching: samples:%u/%u ratio:%f backlog:%f gain: %f", 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