1 files changed, 0 insertions, 68 deletions
diff --git a/src/audio_core/time_stretch.cpp b/src/audio_core/time_stretch.cpp
deleted file mode 100644
index 726591fce..000000000
--- a/src/audio_core/time_stretch.cpp
+++ /dev/null
@@ -1,68 +0,0 @@
-// Copyright 2018 yuzu Emulator Project
-// Licensed under GPLv2 or any later version
-// Refer to the license.txt file included.
-
-#include <algorithm>
-#include <cmath>
-#include <cstddef>
-#include "audio_core/time_stretch.h"
-#include "common/logging/log.h"
-
-namespace AudioCore {
-
-TimeStretcher::TimeStretcher(u32 sample_rate, u32 channel_count) : m_sample_rate{sample_rate} {
-    m_sound_touch.setChannels(channel_count);
-    m_sound_touch.setSampleRate(sample_rate);
-    m_sound_touch.setPitch(1.0);
-    m_sound_touch.setTempo(1.0);
-}
-
-void TimeStretcher::Clear() {
-    m_sound_touch.clear();
-}
-
-void TimeStretcher::Flush() {
-    m_sound_touch.flush();
-}
-
-std::size_t TimeStretcher::Process(const s16* in, std::size_t num_in, s16* out,
-                                   std::size_t 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 = 0.25; // seconds
-    const double max_backlog = m_sample_rate * max_latency;
-    const double backlog_fullness = m_sound_touch.numSamples() / max_backlog;
-    if (backlog_fullness > 4.0) {
-        // 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 = 0.05; // seconds
-    const double tweak_correction = (backlog_fullness - 0.5) * (time_delta / tweak_time_scale);
-    current_ratio *= std::pow(1.0 + 2.0 * tweak_correction, tweak_correction < 0 ? 3.0 : 1.0);
-
-    // 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 = 0.712; // 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 5% 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, 0.05);
-    m_sound_touch.setTempo(m_stretch_ratio);
-
-    LOG_TRACE(Audio, "{:5}/{:5} ratio:{:0.6f} backlog:{:0.6f}", num_in, num_out, m_stretch_ratio,
-              backlog_fullness);
-
-    m_sound_touch.putSamples(in, static_cast<u32>(num_in));
-    return m_sound_touch.receiveSamples(out, static_cast<u32>(num_out));
-}
-
-} // namespace AudioCore