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Merge pull request #4194 from MerryMage/audiofifo

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audio_core: Simplify sink interface
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Merry 2018-09-21 13:30:51 +01:00 committed by GitHub
commit bb9e92c77c
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12 changed files with 316 additions and 314 deletions
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127
src/audio_core/cubeb_sink.cpp
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@ -2,6 +2,7 @@
// Licensed under GPLv2 or any later version
// Refer to the license.txt file included.
#include <cstdarg>
#include <mutex>
#include <vector>
#include <cubeb/cubeb.h>
@ -13,17 +14,16 @@ namespace AudioCore {
struct CubebSink::Impl {
unsigned int sample_rate = 0;
std::vector<std::string> device_list;
cubeb* ctx = nullptr;
cubeb_stream* stream = nullptr;
std::mutex queue_mutex;
std::vector<s16> queue;
std::function<void(s16*, std::size_t)> cb;
static long DataCallback(cubeb_stream* stream, void* user_data, const void* input_buffer,
void* output_buffer, long num_frames);
static void StateCallback(cubeb_stream* stream, void* user_data, cubeb_state state);
static void LogCallback(char const* fmt, ...);
};
CubebSink::CubebSink(std::string target_device_name) : impl(std::make_unique<Impl>()) {
@ -31,21 +31,23 @@ CubebSink::CubebSink(std::string target_device_name) : impl(std::make_unique<Imp
LOG_CRITICAL(Audio_Sink, "cubeb_init failed");
return;
}
cubeb_devid output_device = nullptr;
cubeb_stream_params params;
params.rate = native_sample_rate;
params.channels = 2;
params.format = CUBEB_SAMPLE_S16NE;
params.layout = CUBEB_LAYOUT_STEREO;
cubeb_set_log_callback(CUBEB_LOG_NORMAL, &Impl::LogCallback);
impl->sample_rate = native_sample_rate;
u32 minimum_latency = 0;
if (cubeb_get_min_latency(impl->ctx, &params, &minimum_latency) != CUBEB_OK)
LOG_CRITICAL(Audio_Sink, "Error getting minimum latency");
cubeb_stream_params params;
params.rate = impl->sample_rate;
params.channels = 2;
params.layout = CUBEB_LAYOUT_STEREO;
params.format = CUBEB_SAMPLE_S16NE;
params.prefs = CUBEB_STREAM_PREF_NONE;
u32 minimum_latency = 100 * impl->sample_rate / 1000; // Firefox default
if (cubeb_get_min_latency(impl->ctx, &params, &minimum_latency) != CUBEB_OK) {
LOG_CRITICAL(Audio_Sink, "Error getting minimum latency");
}
cubeb_devid output_device = nullptr;
if (target_device_name != auto_device_name && !target_device_name.empty()) {
cubeb_device_collection collection;
if (cubeb_enumerate_devices(impl->ctx, CUBEB_DEVICE_TYPE_OUTPUT, &collection) != CUBEB_OK) {
@ -63,10 +65,22 @@ CubebSink::CubebSink(std::string target_device_name) : impl(std::make_unique<Imp
}
}
if (cubeb_stream_init(impl->ctx, &impl->stream, "Citra Audio Output", nullptr, nullptr,
int stream_err = cubeb_stream_init(impl->ctx, &impl->stream, "CitraAudio", nullptr, nullptr,
output_device, &params, std::max(512u, minimum_latency),
&Impl::DataCallback, &Impl::StateCallback, impl.get()) != CUBEB_OK) {
LOG_CRITICAL(Audio_Sink, "Error initializing cubeb stream");
&Impl::DataCallback, &Impl::StateCallback, impl.get());
if (stream_err != CUBEB_OK) {
switch (stream_err) {
case CUBEB_ERROR:
default:
LOG_CRITICAL(Audio_Sink, "Error initializing cubeb stream ({})", stream_err);
break;
case CUBEB_ERROR_INVALID_FORMAT:
LOG_CRITICAL(Audio_Sink, "Invalid format when initializing cubeb stream");
break;
case CUBEB_ERROR_DEVICE_UNAVAILABLE:
LOG_CRITICAL(Audio_Sink, "Device unavailable when initializing cubeb stream");
break;
}
return;
}
@ -77,8 +91,11 @@ CubebSink::CubebSink(std::string target_device_name) : impl(std::make_unique<Imp
}
CubebSink::~CubebSink() {
if (!impl->ctx)
if (!impl->ctx) {
return;
}
impl->cb = nullptr;
if (cubeb_stream_stop(impl->stream) != CUBEB_OK) {
LOG_CRITICAL(Audio_Sink, "Error stopping cubeb stream");
@ -95,56 +112,62 @@ unsigned int CubebSink::GetNativeSampleRate() const {
return impl->sample_rate;
}
void CubebSink::EnqueueSamples(const s16* samples, std::size_t sample_count) {
if (!impl->ctx)
return;
std::lock_guard lock{impl->queue_mutex};
impl->queue.reserve(impl->queue.size() + sample_count * 2);
std::copy(samples, samples + sample_count * 2, std::back_inserter(impl->queue));
}
size_t CubebSink::SamplesInQueue() const {
if (!impl->ctx)
return 0;
std::lock_guard lock{impl->queue_mutex};
return impl->queue.size() / 2;
void CubebSink::SetCallback(std::function<void(s16*, std::size_t)> cb) {
impl->cb = cb;
}
long CubebSink::Impl::DataCallback(cubeb_stream* stream, void* user_data, const void* input_buffer,
void* output_buffer, long num_frames) {
Impl* impl = static_cast<Impl*>(user_data);
u8* buffer = reinterpret_cast<u8*>(output_buffer);
s16* buffer = reinterpret_cast<s16*>(output_buffer);
if (!impl)
return 0;
std::lock_guard lock{impl->queue_mutex};
std::size_t frames_to_write =
std::min(impl->queue.size() / 2, static_cast<std::size_t>(num_frames));
memcpy(buffer, impl->queue.data(), frames_to_write * sizeof(s16) * 2);
impl->queue.erase(impl->queue.begin(), impl->queue.begin() + frames_to_write * 2);
if (frames_to_write < num_frames) {
// Fill the rest of the frames with silence
memset(buffer + frames_to_write * sizeof(s16) * 2, 0,
(num_frames - frames_to_write) * sizeof(s16) * 2);
if (!impl || !impl->cb) {
LOG_DEBUG(Audio_Sink, "Emitting zeros");
std::memset(output_buffer, 0, num_frames * 2 * sizeof(s16));
return num_frames;
}
impl->cb(buffer, num_frames);
return num_frames;
}
void CubebSink::Impl::StateCallback(cubeb_stream* stream, void* user_data, cubeb_state state) {}
void CubebSink::Impl::StateCallback(cubeb_stream* stream, void* user_data, cubeb_state state) {
switch (state) {
case CUBEB_STATE_STARTED:
LOG_INFO(Audio_Sink, "Cubeb Audio Stream Started");
break;
case CUBEB_STATE_STOPPED:
LOG_INFO(Audio_Sink, "Cubeb Audio Stream Stopped");
break;
case CUBEB_STATE_DRAINED:
LOG_INFO(Audio_Sink, "Cubeb Audio Stream Drained");
break;
case CUBEB_STATE_ERROR:
LOG_CRITICAL(Audio_Sink, "Cubeb Audio Stream Errored");
break;
}
}
void CubebSink::Impl::LogCallback(char const* format, ...) {
std::array<char, 512> buffer;
std::va_list args;
va_start(args, format);
#ifdef _MSC_VER
vsprintf_s(buffer.data(), buffer.size(), format, args);
#else
vsnprintf(buffer.data(), buffer.size(), format, args);
#endif
va_end(args);
buffer.back() = '\0';
LOG_INFO(Audio_Sink, "{}", buffer.data());
}
std::vector<std::string> ListCubebSinkDevices() {
std::vector<std::string> device_list;
cubeb* ctx;
if (cubeb_init(&ctx, "Citra Device Enumerator", nullptr) != CUBEB_OK) {
if (cubeb_init(&ctx, "CitraEnumerator", nullptr) != CUBEB_OK) {
LOG_CRITICAL(Audio_Sink, "cubeb_init failed");
return {};
}

4
src/audio_core/cubeb_sink.h
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@ -17,9 +17,7 @@ public:
unsigned int GetNativeSampleRate() const override;
void EnqueueSamples(const s16* samples, std::size_t sample_count) override;
std::size_t SamplesInQueue() const override;
void SetCallback(std::function<void(s16*, std::size_t)> cb) override;
private:
struct Impl;

73
src/audio_core/dsp_interface.cpp
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@ -12,16 +12,13 @@
namespace AudioCore {
DspInterface::DspInterface() = default;
DspInterface::~DspInterface() {
if (perform_time_stretching) {
FlushResidualStretcherAudio();
}
}
DspInterface::~DspInterface() = default;
void DspInterface::SetSink(const std::string& sink_id, const std::string& audio_device) {
const SinkDetails& sink_details = GetSinkDetails(sink_id);
sink = sink_details.factory(audio_device);
sink->SetCallback(
[this](s16* buffer, std::size_t num_frames) { OutputCallback(buffer, num_frames); });
time_stretcher.SetOutputSampleRate(sink->GetNativeSampleRate());
}
@ -35,7 +32,7 @@ void DspInterface::EnableStretching(bool enable) {
return;
if (!enable) {
FlushResidualStretcherAudio();
flushing_time_stretcher = true;
}
perform_time_stretching = enable;
}
@ -44,39 +41,41 @@ void DspInterface::OutputFrame(StereoFrame16& frame) {
if (!sink)
return;
// Implementation of the hardware volume slider with a dynamic range of 60 dB
double volume_scale_factor = std::exp(6.90775 * Settings::values.volume) * 0.001;
for (std::size_t i = 0; i < frame.size(); i++) {
frame[i][0] = static_cast<s16>(frame[i][0] * volume_scale_factor);
frame[i][1] = static_cast<s16>(frame[i][1] * volume_scale_factor);
}
if (perform_time_stretching) {
time_stretcher.AddSamples(&frame[0][0], frame.size());
std::vector<s16> stretched_samples = time_stretcher.Process(sink->SamplesInQueue());
sink->EnqueueSamples(stretched_samples.data(), stretched_samples.size() / 2);
} else {
constexpr std::size_t maximum_sample_latency = 2048; // about 64 miliseconds
if (sink->SamplesInQueue() > maximum_sample_latency) {
// This can occur if we're running too fast and samples are starting to back up.
// Just drop the samples.
return;
}
sink->EnqueueSamples(&frame[0][0], frame.size());
}
fifo.Push(frame.data(), frame.size());
}
void DspInterface::FlushResidualStretcherAudio() {
if (!sink)
return;
void DspInterface::OutputCallback(s16* buffer, std::size_t num_frames) {
std::size_t frames_written;
if (perform_time_stretching) {
const std::vector<s16> in{fifo.Pop()};
const std::size_t num_in{in.size() / 2};
frames_written = time_stretcher.Process(in.data(), num_in, buffer, num_frames);
} else if (flushing_time_stretcher) {
time_stretcher.Flush();
while (true) {
std::vector<s16> residual_audio = time_stretcher.Process(sink->SamplesInQueue());
if (residual_audio.empty())
break;
sink->EnqueueSamples(residual_audio.data(), residual_audio.size() / 2);
frames_written = time_stretcher.Process(nullptr, 0, buffer, num_frames);
frames_written += fifo.Pop(buffer, num_frames - frames_written);
flushing_time_stretcher = false;
} else {
frames_written = fifo.Pop(buffer, num_frames);
}
if (frames_written > 0) {
std::memcpy(&last_frame[0], buffer + 2 * (frames_written - 1), 2 * sizeof(s16));
}
// Hold last emitted frame; this prevents popping.
for (std::size_t i = frames_written; i < num_frames; i++) {
std::memcpy(buffer + 2 * i, &last_frame[0], 2 * sizeof(s16));
}
// Implementation of the hardware volume slider with a dynamic range of 60 dB
const float linear_volume = std::clamp(Settings::values.volume, 0.0f, 1.0f);
if (linear_volume != 1.0) {
const float volume_scale_factor = std::exp(6.90775f * linear_volume) * 0.001f;
for (std::size_t i = 0; i < num_frames; i++) {
buffer[i * 2 + 0] = static_cast<s16>(buffer[i * 2 + 0] * volume_scale_factor);
buffer[i * 2 + 1] = static_cast<s16>(buffer[i * 2 + 1] * volume_scale_factor);
}
}
}

7
src/audio_core/dsp_interface.h
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@ -9,6 +9,7 @@
#include "audio_core/audio_types.h"
#include "audio_core/time_stretch.h"
#include "common/common_types.h"
#include "common/ring_buffer.h"
#include "core/memory.h"
namespace Service {
@ -81,9 +82,13 @@ protected:
private:
void FlushResidualStretcherAudio();
void OutputCallback(s16* buffer, std::size_t num_frames);
std::unique_ptr<Sink> sink;
bool perform_time_stretching = false;
std::atomic<bool> perform_time_stretching = false;
std::atomic<bool> flushing_time_stretcher = false;
Common::RingBuffer<s16, 0x2000, 2> fifo;
std::array<s16, 2> last_frame{};
TimeStretcher time_stretcher;
};

6
src/audio_core/null_sink.h
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@ -19,11 +19,7 @@ public:
return native_sample_rate;
}
void EnqueueSamples(const s16*, std::size_t) override {}
std::size_t SamplesInQueue() const override {
return 0;
}
void SetCallback(std::function<void(s16*, std::size_t)>) override {}
};
} // namespace AudioCore

58
src/audio_core/sdl2_sink.cpp
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@ -2,8 +2,8 @@
// Licensed under GPLv2 or any later version
// Refer to the license.txt file included.
#include <list>
#include <numeric>
#include <string>
#include <vector>
#include <SDL.h>
#include "audio_core/audio_types.h"
#include "audio_core/sdl2_sink.h"
@ -17,7 +17,7 @@ struct SDL2Sink::Impl {
SDL_AudioDeviceID audio_device_id = 0;
std::list<std::vector<s16>> queue;
std::function<void(s16*, std::size_t)> cb;
static void Callback(void* impl_, u8* buffer, int buffer_size_in_bytes);
};
@ -74,58 +74,18 @@ unsigned int SDL2Sink::GetNativeSampleRate() const {
return impl->sample_rate;
}
void SDL2Sink::EnqueueSamples(const s16* samples, std::size_t sample_count) {
if (impl->audio_device_id <= 0)
return;
SDL_LockAudioDevice(impl->audio_device_id);
impl->queue.emplace_back(samples, samples + sample_count * 2);
SDL_UnlockAudioDevice(impl->audio_device_id);
}
size_t SDL2Sink::SamplesInQueue() const {
if (impl->audio_device_id <= 0)
return 0;
SDL_LockAudioDevice(impl->audio_device_id);
std::size_t total_size =
std::accumulate(impl->queue.begin(), impl->queue.end(), static_cast<std::size_t>(0),
[](std::size_t sum, const auto& buffer) {
// Division by two because each stereo sample is made of
// two s16.
return sum + buffer.size() / 2;
});
SDL_UnlockAudioDevice(impl->audio_device_id);
return total_size;
void SDL2Sink::SetCallback(std::function<void(s16*, std::size_t)> cb) {
impl->cb = cb;
}
void SDL2Sink::Impl::Callback(void* impl_, u8* buffer, int buffer_size_in_bytes) {
Impl* impl = reinterpret_cast<Impl*>(impl_);
if (!impl || !impl->cb)
return;
std::size_t remaining_size = static_cast<std::size_t>(buffer_size_in_bytes) /
sizeof(s16); // Keep track of size in 16-bit increments.
const size_t num_frames = buffer_size_in_bytes / (2 * sizeof(s16));
while (remaining_size > 0 && !impl->queue.empty()) {
if (impl->queue.front().size() <= remaining_size) {
memcpy(buffer, impl->queue.front().data(), impl->queue.front().size() * sizeof(s16));
buffer += impl->queue.front().size() * sizeof(s16);
remaining_size -= impl->queue.front().size();
impl->queue.pop_front();
} else {
memcpy(buffer, impl->queue.front().data(), remaining_size * sizeof(s16));
buffer += remaining_size * sizeof(s16);
impl->queue.front().erase(impl->queue.front().begin(),
impl->queue.front().begin() + remaining_size);
remaining_size = 0;
}
}
if (remaining_size > 0) {
memset(buffer, 0, remaining_size * sizeof(s16));
}
impl->cb(reinterpret_cast<s16*>(buffer), num_frames);
}
std::vector<std::string> ListSDL2SinkDevices() {

4
src/audio_core/sdl2_sink.h
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@ -17,9 +17,7 @@ public:
unsigned int GetNativeSampleRate() const override;
void EnqueueSamples(const s16* samples, std::size_t sample_count) override;
std::size_t SamplesInQueue() const override;
void SetCallback(std::function<void(s16*, std::size_t)> cb) override;
private:
struct Impl;

13
src/audio_core/sink.h
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@ -4,7 +4,7 @@
#pragma once
#include <vector>
#include <functional>
#include "common/common_types.h"
namespace AudioCore {
@ -20,19 +20,16 @@ class Sink {
public:
virtual ~Sink() = default;
/// The native rate of this sink. The sink expects to be fed samples that respect this. (Units:
/// samples/sec)
/// The native rate of this sink. The sink expects to be fed samples that respect this.
/// (Units: samples/sec)
virtual unsigned int GetNativeSampleRate() const = 0;
/**
* Feed stereo samples to sink.
* Set callback for samples
* @param samples Samples in interleaved stereo PCM16 format.
* @param sample_count Number of samples.
*/
virtual void EnqueueSamples(const s16* samples, std::size_t sample_count) = 0;
/// Samples enqueued that have not been played yet.
virtual std::size_t SamplesInQueue() const = 0;
virtual void SetCallback(std::function<void(s16*, std::size_t)> cb) = 0;
};
} // namespace AudioCore

172
src/audio_core/time_stretch.cpp
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@ -3,143 +3,75 @@
// Refer to the license.txt file included.
#include <algorithm>
#include <chrono>
#include <cmath>
#include <vector>
#include <cstddef>
#include <memory>
#include <SoundTouch.h>
#include "audio_core/audio_types.h"
#include "audio_core/time_stretch.h"
#include "common/common_types.h"
#include "common/logging/log.h"
using steady_clock = std::chrono::steady_clock;
namespace AudioCore {
constexpr double MIN_RATIO = 0.1;
constexpr double MAX_RATIO = 100.0;
static double ClampRatio(double ratio) {
return std::clamp(ratio, MIN_RATIO, MAX_RATIO);
TimeStretcher::TimeStretcher()
: sample_rate(native_sample_rate), sound_touch(std::make_unique<soundtouch::SoundTouch>()) {
sound_touch->setChannels(2);
sound_touch->setSampleRate(native_sample_rate);
sound_touch->setPitch(1.0);
sound_touch->setTempo(1.0);
}
constexpr double MIN_DELAY_TIME = 0.05; // Units: seconds
constexpr double MAX_DELAY_TIME = 0.25; // Units: seconds
constexpr std::size_t DROP_FRAMES_SAMPLE_DELAY = 16000; // Units: samples
constexpr double SMOOTHING_FACTOR = 0.007;
struct TimeStretcher::Impl {
soundtouch::SoundTouch soundtouch;
steady_clock::time_point frame_timer = steady_clock::now();
std::size_t samples_queued = 0;
double smoothed_ratio = 1.0;
double sample_rate = static_cast<double>(native_sample_rate);
};
std::vector<s16> TimeStretcher::Process(std::size_t samples_in_queue) {
// This is a very simple algorithm without any fancy control theory. It works and is stable.
double ratio = CalculateCurrentRatio();
ratio = CorrectForUnderAndOverflow(ratio, samples_in_queue);
impl->smoothed_ratio =
(1.0 - SMOOTHING_FACTOR) * impl->smoothed_ratio + SMOOTHING_FACTOR * ratio;
impl->smoothed_ratio = ClampRatio(impl->smoothed_ratio);
// SoundTouch's tempo definition the inverse of our ratio definition.
impl->soundtouch.setTempo(1.0 / impl->smoothed_ratio);
std::vector<s16> samples = GetSamples();
if (samples_in_queue >= DROP_FRAMES_SAMPLE_DELAY) {
samples.clear();
LOG_DEBUG(Audio, "Dropping frames!");
}
return samples;
}
TimeStretcher::TimeStretcher() : impl(std::make_unique<Impl>()) {
impl->soundtouch.setPitch(1.0);
impl->soundtouch.setChannels(2);
impl->soundtouch.setSampleRate(native_sample_rate);
Reset();
}
TimeStretcher::~TimeStretcher() {
impl->soundtouch.clear();
}
TimeStretcher::~TimeStretcher() = default;
void TimeStretcher::SetOutputSampleRate(unsigned int sample_rate) {
impl->sample_rate = static_cast<double>(sample_rate);
impl->soundtouch.setRate(static_cast<double>(native_sample_rate) / impl->sample_rate);
sound_touch->setSampleRate(sample_rate);
sample_rate = native_sample_rate;
}
void TimeStretcher::AddSamples(const s16* buffer, std::size_t num_samples) {
impl->soundtouch.putSamples(buffer, static_cast<uint>(num_samples));
impl->samples_queued += num_samples;
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) / sample_rate; // seconds
double current_ratio = static_cast<double>(num_in) / static_cast<double>(num_out);
const double max_latency = 0.25; // seconds
const double max_backlog = sample_rate * max_latency;
const double backlog_fullness = 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 = 0.050; // 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);
stretch_ratio += lpf_gain * (current_ratio - 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.
stretch_ratio = std::max(stretch_ratio, 0.05);
sound_touch->setTempo(stretch_ratio);
LOG_DEBUG(Audio, "{:5}/{:5} ratio:{:0.6f} backlog:{:0.6f}", num_in, num_out, stretch_ratio,
backlog_fullness);
sound_touch->putSamples(in, num_in);
return sound_touch->receiveSamples(out, num_out);
}
void TimeStretcher::Clear() {
sound_touch->clear();
}
void TimeStretcher::Flush() {
impl->soundtouch.flush();
}
void TimeStretcher::Reset() {
impl->soundtouch.setTempo(1.0);
impl->soundtouch.clear();
impl->smoothed_ratio = 1.0;
impl->frame_timer = steady_clock::now();
impl->samples_queued = 0;
SetOutputSampleRate(native_sample_rate);
}
double TimeStretcher::CalculateCurrentRatio() {
const steady_clock::time_point now = steady_clock::now();
const std::chrono::duration<double> duration = now - impl->frame_timer;
const double expected_time =
static_cast<double>(impl->samples_queued) / static_cast<double>(native_sample_rate);
const double actual_time = duration.count();
double ratio;
if (expected_time != 0) {
ratio = ClampRatio(actual_time / expected_time);
} else {
ratio = impl->smoothed_ratio;
}
impl->frame_timer = now;
impl->samples_queued = 0;
return ratio;
}
double TimeStretcher::CorrectForUnderAndOverflow(double ratio, std::size_t sample_delay) const {
const std::size_t min_sample_delay =
static_cast<std::size_t>(MIN_DELAY_TIME * impl->sample_rate);
const std::size_t max_sample_delay =
static_cast<std::size_t>(MAX_DELAY_TIME * impl->sample_rate);
if (sample_delay < min_sample_delay) {
// Make the ratio bigger.
ratio = ratio > 1.0 ? ratio * ratio : sqrt(ratio);
} else if (sample_delay > max_sample_delay) {
// Make the ratio smaller.
ratio = ratio > 1.0 ? sqrt(ratio) : ratio * ratio;
}
return ClampRatio(ratio);
}
std::vector<s16> TimeStretcher::GetSamples() {
uint available = impl->soundtouch.numSamples();
std::vector<s16> output(static_cast<std::size_t>(available) * 2);
impl->soundtouch.receiveSamples(output.data(), available);
return output;
sound_touch->flush();
}
} // namespace AudioCore

52
src/audio_core/time_stretch.h
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@ -4,57 +4,39 @@
#pragma once
#include <array>
#include <cstddef>
#include <memory>
#include <vector>
#include "common/common_types.h"
namespace soundtouch {
class SoundTouch;
}
namespace AudioCore {
class TimeStretcher final {
class TimeStretcher {
public:
TimeStretcher();
~TimeStretcher();
/**
* Set sample rate for the samples that Process returns.
* @param sample_rate The sample rate.
*/
void SetOutputSampleRate(unsigned int sample_rate);
/**
* Add samples to be processed.
* @param sample_buffer Buffer of samples in interleaved stereo PCM16 format.
* @param num_samples Number of samples.
*/
void AddSamples(const s16* sample_buffer, std::size_t num_samples);
/// @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();
/// Flush audio remaining in internal buffers.
void Flush();
/// Resets internal state and clears buffers.
void Reset();
/**
* Does audio stretching and produces the time-stretched samples.
* Timer calculations use sample_delay to determine how much of a margin we have.
* @param sample_delay How many samples are buffered downstream of this module and haven't been
* played yet.
* @return Samples to play in interleaved stereo PCM16 format.
*/
std::vector<s16> Process(std::size_t sample_delay);
private:
struct Impl;
std::unique_ptr<Impl> impl;
/// INTERNAL: ratio = wallclock time / emulated time
double CalculateCurrentRatio();
/// INTERNAL: If we have too many or too few samples downstream, nudge ratio in the appropriate
/// direction.
double CorrectForUnderAndOverflow(double ratio, std::size_t sample_delay) const;
/// INTERNAL: Gets the time-stretched samples from SoundTouch.
std::vector<s16> GetSamples();
unsigned int sample_rate;
std::unique_ptr<soundtouch::SoundTouch> sound_touch;
double stretch_ratio = 1.0;
};
} // namespace AudioCore

1
src/common/CMakeLists.txt
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@ -72,6 +72,7 @@ add_library(common STATIC
param_package.cpp
param_package.h
quaternion.h
ring_buffer.h
scm_rev.cpp
scm_rev.h
scope_exit.h

111
src/common/ring_buffer.h Normal file
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@ -0,0 +1,111 @@
// Copyright 2018 Citra Emulator Project
// Licensed under GPLv2 or any later version
// Refer to the license.txt file included.
#pragma once
#include <algorithm>
#include <array>
#include <atomic>
#include <cstddef>
#include <cstring>
#include <type_traits>
#include <vector>
#include "common/common_types.h"
namespace Common {
/// SPSC ring buffer
/// @tparam T Element type
/// @tparam capacity Number of slots in ring buffer
/// @tparam granularity Slot size in terms of number of elements
template <typename T, std::size_t capacity, std::size_t granularity = 1>
class RingBuffer {
/// A "slot" is made of `granularity` elements of `T`.
static constexpr std::size_t slot_size = granularity * sizeof(T);
// T must be safely memcpy-able and have a trivial default constructor.
static_assert(std::is_trivial_v<T>);
// Ensure capacity is sensible.
static_assert(capacity < std::numeric_limits<std::size_t>::max() / 2 / granularity);
static_assert((capacity & (capacity - 1)) == 0, "capacity must be a power of two");
// Ensure lock-free.
static_assert(std::atomic<std::size_t>::is_always_lock_free);
public:
/// Pushes slots into the ring buffer
/// @param new_slots Pointer to the slots to push
/// @param slot_count Number of slots to push
/// @returns The number of slots actually pushed
std::size_t Push(const void* new_slots, std::size_t slot_count) {
const std::size_t write_index = m_write_index.load();
const std::size_t slots_free = capacity + m_read_index.load() - write_index;
const std::size_t push_count = std::min(slot_count, slots_free);
const std::size_t pos = write_index % capacity;
const std::size_t first_copy = std::min(capacity - pos, push_count);
const std::size_t second_copy = push_count - first_copy;
const char* in = static_cast<const char*>(new_slots);
std::memcpy(m_data.data() + pos * granularity, in, first_copy * slot_size);
in += first_copy * slot_size;
std::memcpy(m_data.data(), in, second_copy * slot_size);
m_write_index.store(write_index + push_count);
return push_count;
}
std::size_t Push(const std::vector<T>& input) {
return Push(input.data(), input.size() / granularity);
}
/// Pops slots from the ring buffer
/// @param output Where to store the popped slots
/// @param max_slots Maximum number of slots to pop
/// @returns The number of slots actually popped
std::size_t Pop(void* output, std::size_t max_slots = ~std::size_t(0)) {
const std::size_t read_index = m_read_index.load();
const std::size_t slots_filled = m_write_index.load() - read_index;
const std::size_t pop_count = std::min(slots_filled, max_slots);
const std::size_t pos = read_index % capacity;
const std::size_t first_copy = std::min(capacity - pos, pop_count);
const std::size_t second_copy = pop_count - first_copy;
char* out = static_cast<char*>(output);
std::memcpy(out, m_data.data() + pos * granularity, first_copy * slot_size);
out += first_copy * slot_size;
std::memcpy(out, m_data.data(), second_copy * slot_size);
m_read_index.store(read_index + pop_count);
return pop_count;
}
std::vector<T> Pop(std::size_t max_slots = ~std::size_t(0)) {
std::vector<T> out(std::min(max_slots, capacity) * granularity);
const std::size_t count = Pop(out.data(), out.size() / granularity);
out.resize(count * granularity);
return out;
}
/// @returns Number of slots used
std::size_t Size() const {
return m_write_index.load() - m_read_index.load();
}
/// @returns Maximum size of ring buffer
constexpr std::size_t Capacity() const {
return capacity;
}
private:
// It is important to align the below variables for performance reasons:
// Having them on the same cache-line would result in false-sharing between them.
alignas(128) std::atomic<std::size_t> m_read_index{0};
alignas(128) std::atomic<std::size_t> m_write_index{0};
std::array<T, granularity * capacity> m_data;
};
} // namespace Common