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-rw-r--r--src/audio_core/CMakeLists.txt3
-rw-r--r--src/audio_core/cubeb_sink.cpp25
-rw-r--r--src/audio_core/sdl2_sink.cpp4
-rw-r--r--src/audio_core/time_stretch.cpp68
-rw-r--r--src/audio_core/time_stretch.h34
-rw-r--r--src/yuzu_cmd/default_ini.h6
6 files changed, 3 insertions, 137 deletions
diff --git a/src/audio_core/CMakeLists.txt b/src/audio_core/CMakeLists.txt
index 090dd19b1..e553b8203 100644
--- a/src/audio_core/CMakeLists.txt
+++ b/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)
diff --git a/src/audio_core/cubeb_sink.cpp b/src/audio_core/cubeb_sink.cpp
index 93c35e785..13de3087c 100644
--- a/src/audio_core/cubeb_sink.cpp
+++ b/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),
diff --git a/src/audio_core/sdl2_sink.cpp b/src/audio_core/sdl2_sink.cpp
index 62d3716a6..2d14ce2cb 100644
--- a/src/audio_core/sdl2_sink.cpp
+++ b/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) {
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
diff --git a/src/audio_core/time_stretch.h b/src/audio_core/time_stretch.h
deleted file mode 100644
index bb2270b96..000000000
--- a/src/audio_core/time_stretch.h
+++ /dev/null
@@ -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
diff --git a/src/yuzu_cmd/default_ini.h b/src/yuzu_cmd/default_ini.h
index 34782c378..f34d6b728 100644
--- a/src/yuzu_cmd/default_ini.h
+++ b/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 =
generated by cgit v1.2.3 (git 2.25.1) at 2024-11-18 13:11:54 +0100