Merge pull request #8134 from Tachi107/remove-time-stretcher

audio_core: remove time stretcher

.gitmodules

@@ -7,9 +7,6 @@
 [submodule "dynarmic"]
     path = externals/dynarmic
     url = https://github.com/MerryMage/dynarmic.git
-[submodule "soundtouch"]
-    path = externals/soundtouch
-    url = https://github.com/citra-emu/ext-soundtouch.git
 [submodule "libressl"]
     path = externals/libressl
     url = https://github.com/citra-emu/ext-libressl-portable.git

externals/CMakeLists.txt

@@ -68,9 +68,6 @@ if (YUZU_USE_EXTERNAL_SDL2)
     add_library(SDL2 ALIAS SDL2-static)
 endif()
 
-# SoundTouch
-add_subdirectory(soundtouch)
-
 # Cubeb
 if(ENABLE_CUBEB)
     set(BUILD_TESTS OFF CACHE BOOL "")

externals/soundtouch

@@ -1 +0,0 @@
-Subproject commit 060181eaf273180d3a7e87349895bd0cb6ccbf4a

src/audio_core/CMakeLists.txt

@@ -36,8 +36,6 @@ add_library(audio_core STATIC
     splitter_context.h
     stream.cpp
     stream.h
-    time_stretch.cpp
-    time_stretch.h
     voice_context.cpp
     voice_context.h
 
@@ -63,7 +61,6 @@ if (NOT MSVC)
 endif()
 
 target_link_libraries(audio_core PUBLIC common core)
-target_link_libraries(audio_core PRIVATE SoundTouch)
 
 if(ENABLE_CUBEB)
     target_link_libraries(audio_core PRIVATE cubeb)

src/audio_core/cubeb_sink.cpp

@@ -7,7 +7,6 @@
 #include <cstring>
 #include "audio_core/cubeb_sink.h"
 #include "audio_core/stream.h"
-#include "audio_core/time_stretch.h"
 #include "common/assert.h"
 #include "common/logging/log.h"
 #include "common/ring_buffer.h"
@@ -23,8 +22,7 @@ class CubebSinkStream final : public SinkStream {
 public:
     CubebSinkStream(cubeb* ctx_, u32 sample_rate, u32 num_channels_, cubeb_devid output_device,
                     const std::string& name)
-        : ctx{ctx_}, num_channels{std::min(num_channels_, 6u)}, time_stretch{sample_rate,
-                                                                             num_channels} {
+        : ctx{ctx_}, num_channels{std::min(num_channels_, 6u)} {
 
         cubeb_stream_params params{};
         params.rate = sample_rate;
@@ -131,7 +129,6 @@ private:
     Common::RingBuffer<s16, 0x10000> queue;
     std::array<s16, 2> last_frame{};
     std::atomic<bool> should_flush{};
-    TimeStretcher time_stretch;
 
     static long DataCallback(cubeb_stream* stream, void* user_data, const void* input_buffer,
                              void* output_buffer, long num_frames);
@@ -205,25 +202,7 @@ long CubebSinkStream::DataCallback([[maybe_unused]] cubeb_stream* stream, void*
 
     const std::size_t num_channels = impl->GetNumChannels();
     const std::size_t samples_to_write = num_channels * num_frames;
-    std::size_t samples_written;
-
-    /*
-    if (Settings::values.enable_audio_stretching.GetValue()) {
-        const std::vector<s16> in{impl->queue.Pop()};
-        const std::size_t num_in{in.size() / num_channels};
-        s16* const out{reinterpret_cast<s16*>(buffer)};
-        const std::size_t out_frames =
-            impl->time_stretch.Process(in.data(), num_in, out, num_frames);
-        samples_written = out_frames * num_channels;
-
-        if (impl->should_flush) {
-            impl->time_stretch.Flush();
-            impl->should_flush = false;
-        }
-    } else {
-        samples_written = impl->queue.Pop(buffer, samples_to_write);
-    }*/
-    samples_written = impl->queue.Pop(buffer, samples_to_write);
+    const std::size_t samples_written = impl->queue.Pop(buffer, samples_to_write);
 
     if (samples_written >= num_channels) {
         std::memcpy(&impl->last_frame[0], buffer + (samples_written - num_channels) * sizeof(s16),

src/audio_core/sdl2_sink.cpp

@@ -7,7 +7,6 @@
 #include <cstring>
 #include "audio_core/sdl2_sink.h"
 #include "audio_core/stream.h"
-#include "audio_core/time_stretch.h"
 #include "common/assert.h"
 #include "common/logging/log.h"
 //#include "common/settings.h"
@@ -27,7 +26,7 @@ namespace AudioCore {
 class SDLSinkStream final : public SinkStream {
 public:
     SDLSinkStream(u32 sample_rate, u32 num_channels_, const std::string& output_device)
-        : num_channels{std::min(num_channels_, 6u)}, time_stretch{sample_rate, num_channels} {
+        : num_channels{std::min(num_channels_, 6u)} {
 
         SDL_AudioSpec spec;
         spec.freq = sample_rate;
@@ -116,7 +115,6 @@ private:
     SDL_AudioDeviceID dev = 0;
     u32 num_channels{};
     std::atomic<bool> should_flush{};
-    TimeStretcher time_stretch;
 };
 
 SDLSink::SDLSink(std::string_view target_device_name) {

src/audio_core/time_stretch.cpp

@@ -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

src/audio_core/time_stretch.h

@@ -1,34 +0,0 @@
-// Copyright 2018 yuzu Emulator Project
-// Licensed under GPLv2 or any later version
-// Refer to the license.txt file included.
-
-#pragma once
-
-#include <cstddef>
-#include <SoundTouch.h>
-#include "common/common_types.h"
-
-namespace AudioCore {
-
-class TimeStretcher {
-public:
-    TimeStretcher(u32 sample_rate, u32 channel_count);
-
-    /// @param in       Input sample buffer
-    /// @param num_in   Number of input frames in `in`
-    /// @param out      Output sample buffer
-    /// @param num_out  Desired number of output frames in `out`
-    /// @returns Actual number of frames written to `out`
-    std::size_t Process(const s16* in, std::size_t num_in, s16* out, std::size_t num_out);
-
-    void Clear();
-
-    void Flush();
-
-private:
-    u32 m_sample_rate;
-    soundtouch::SoundTouch m_sound_touch;
-    double m_stretch_ratio = 1.0;
-};
-
-} // namespace AudioCore

src/yuzu_cmd/default_ini.h

@@ -342,12 +342,6 @@ fps_cap =
 # null: No audio output
 output_engine =
 
-# Whether or not to enable the audio-stretching post-processing effect.
-# This effect adjusts audio speed to match emulation speed and helps prevent audio stutter,
-# at the cost of increasing audio latency.
-# 0: No, 1 (default): Yes
-enable_audio_stretching =
-
 # Which audio device to use.
 # auto (default): Auto-select
 output_device =