citra/src/audio_core/dsp_interface.cpp
Weiyi Wang 7d8f115185 Prefix all size_t with std::
done automatically by executing regex replace `([^:0-9a-zA-Z_])size_t([^0-9a-zA-Z_])` -> `$1std::size_t$2`
2018-09-06 16:03:28 -04:00

84 lines
2.5 KiB
C++

// Copyright 2017 Citra Emulator Project
// Licensed under GPLv2 or any later version
// Refer to the license.txt file included.
#include <cstddef>
#include "audio_core/dsp_interface.h"
#include "audio_core/sink.h"
#include "audio_core/sink_details.h"
#include "common/assert.h"
#include "core/settings.h"
namespace AudioCore {
DspInterface::DspInterface() = default;
DspInterface::~DspInterface() {
if (perform_time_stretching) {
FlushResidualStretcherAudio();
}
}
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);
time_stretcher.SetOutputSampleRate(sink->GetNativeSampleRate());
}
Sink& DspInterface::GetSink() {
ASSERT(sink);
return *sink.get();
}
void DspInterface::EnableStretching(bool enable) {
if (perform_time_stretching == enable)
return;
if (!enable) {
FlushResidualStretcherAudio();
}
perform_time_stretching = enable;
}
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());
}
}
void DspInterface::FlushResidualStretcherAudio() {
if (!sink)
return;
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);
}
}
} // namespace AudioCore