AudioCore: Implement time stretcher (#1737)

* AudioCore: Implement time stretcher

* fixup! AudioCore: Implement time stretcher

* fixup! fixup! AudioCore: Implement time stretcher

* fixup! fixup! fixup! AudioCore: Implement time stretcher

* fixup! fixup! fixup! fixup! AudioCore: Implement time stretcher

* fixup! fixup! fixup! fixup! fixup! AudioCore: Implement time stretcher
This commit is contained in:
Maribel 2016-05-15 03:04:03 +01:00 committed by bunnei
parent d299f7ed28
commit 6f6af6928f
4 changed files with 219 additions and 0 deletions

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@ -7,6 +7,7 @@ set(SRCS
hle/source.cpp
interpolate.cpp
sink_details.cpp
time_stretch.cpp
)
set(HEADERS
@ -21,6 +22,7 @@ set(HEADERS
null_sink.h
sink.h
sink_details.h
time_stretch.h
)
include_directories(../../externals/soundtouch/include)

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@ -9,6 +9,7 @@
#include "audio_core/hle/pipe.h"
#include "audio_core/hle/source.h"
#include "audio_core/sink.h"
#include "audio_core/time_stretch.h"
namespace DSP {
namespace HLE {
@ -48,15 +49,29 @@ static std::array<Source, num_sources> sources = {
};
static std::unique_ptr<AudioCore::Sink> sink;
static AudioCore::TimeStretcher time_stretcher;
void Init() {
DSP::HLE::ResetPipes();
for (auto& source : sources) {
source.Reset();
}
time_stretcher.Reset();
if (sink) {
time_stretcher.SetOutputSampleRate(sink->GetNativeSampleRate());
}
}
void Shutdown() {
time_stretcher.Flush();
while (true) {
std::vector<s16> residual_audio = time_stretcher.Process(sink->SamplesInQueue());
if (residual_audio.empty())
break;
sink->EnqueueSamples(residual_audio);
}
}
bool Tick() {
@ -77,6 +92,7 @@ bool Tick() {
void SetSink(std::unique_ptr<AudioCore::Sink> sink_) {
sink = std::move(sink_);
time_stretcher.SetOutputSampleRate(sink->GetNativeSampleRate());
}
} // namespace HLE

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@ -0,0 +1,144 @@
// Copyright 2016 Citra Emulator Project
// Licensed under GPLv2 or any later version
// Refer to the license.txt file included.
#include <chrono>
#include <cmath>
#include <vector>
#include <SoundTouch.h>
#include "audio_core/audio_core.h"
#include "audio_core/time_stretch.h"
#include "common/common_types.h"
#include "common/logging/log.h"
#include "common/math_util.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 MathUtil::Clamp(ratio, MIN_RATIO, MAX_RATIO);
}
constexpr double MIN_DELAY_TIME = 0.05; // Units: seconds
constexpr double MAX_DELAY_TIME = 0.25; // Units: seconds
constexpr 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();
size_t samples_queued = 0;
double smoothed_ratio = 1.0;
double sample_rate = static_cast<double>(native_sample_rate);
};
std::vector<s16> TimeStretcher::Process(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();
}
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);
}
void TimeStretcher::AddSamples(const s16* buffer, size_t num_samples) {
impl->soundtouch.putSamples(buffer, static_cast<uint>(num_samples));
impl->samples_queued += num_samples;
}
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, size_t sample_delay) const {
const size_t min_sample_delay = static_cast<size_t>(MIN_DELAY_TIME * impl->sample_rate);
const size_t max_sample_delay = static_cast<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<size_t>(available) * 2);
impl->soundtouch.receiveSamples(output.data(), available);
return output;
}
} // namespace AudioCore

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@ -0,0 +1,57 @@
// Copyright 2016 Citra Emulator Project
// Licensed under GPLv2 or any later version
// Refer to the license.txt file included.
#include <cstddef>
#include <memory>
#include <vector>
#include "common/common_types.h"
namespace AudioCore {
class TimeStretcher final {
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_sample Number of samples.
*/
void AddSamples(const s16* sample_buffer, size_t num_samples);
/// 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(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, size_t sample_delay) const;
/// INTERNAL: Gets the time-stretched samples from SoundTouch.
std::vector<s16> GetSamples();
};
} // namespace AudioCore