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4f23d5d69e
audio: implement DSP LLE
89 lines
2.8 KiB
C++
89 lines
2.8 KiB
C++
// Copyright 2017 Citra Emulator Project
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// Licensed under GPLv2 or any later version
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// Refer to the license.txt file included.
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#include <cstddef>
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#include "audio_core/dsp_interface.h"
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#include "audio_core/sink.h"
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#include "audio_core/sink_details.h"
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#include "common/assert.h"
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#include "core/settings.h"
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namespace AudioCore {
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DspInterface::DspInterface() = default;
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DspInterface::~DspInterface() = default;
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void DspInterface::SetSink(const std::string& sink_id, const std::string& audio_device) {
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sink = CreateSinkFromID(Settings::values.sink_id, Settings::values.audio_device_id);
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sink->SetCallback(
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[this](s16* buffer, std::size_t num_frames) { OutputCallback(buffer, num_frames); });
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time_stretcher.SetOutputSampleRate(sink->GetNativeSampleRate());
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}
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Sink& DspInterface::GetSink() {
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ASSERT(sink);
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return *sink.get();
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}
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void DspInterface::EnableStretching(bool enable) {
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if (perform_time_stretching == enable)
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return;
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if (!enable) {
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flushing_time_stretcher = true;
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}
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perform_time_stretching = enable;
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}
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void DspInterface::OutputFrame(StereoFrame16& frame) {
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if (!sink)
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return;
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fifo.Push(frame.data(), frame.size());
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}
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void DspInterface::OutputSample(std::array<s16, 2> sample) {
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if (!sink)
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return;
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fifo.Push(&sample, 1);
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}
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void DspInterface::OutputCallback(s16* buffer, std::size_t num_frames) {
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std::size_t frames_written;
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if (perform_time_stretching) {
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const std::vector<s16> in{fifo.Pop()};
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const std::size_t num_in{in.size() / 2};
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frames_written = time_stretcher.Process(in.data(), num_in, buffer, num_frames);
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} else if (flushing_time_stretcher) {
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time_stretcher.Flush();
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frames_written = time_stretcher.Process(nullptr, 0, buffer, num_frames);
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frames_written += fifo.Pop(buffer, num_frames - frames_written);
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flushing_time_stretcher = false;
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} else {
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frames_written = fifo.Pop(buffer, num_frames);
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}
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if (frames_written > 0) {
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std::memcpy(&last_frame[0], buffer + 2 * (frames_written - 1), 2 * sizeof(s16));
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}
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// Hold last emitted frame; this prevents popping.
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for (std::size_t i = frames_written; i < num_frames; i++) {
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std::memcpy(buffer + 2 * i, &last_frame[0], 2 * sizeof(s16));
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}
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// Implementation of the hardware volume slider with a dynamic range of 60 dB
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const float linear_volume = std::clamp(Settings::values.volume, 0.0f, 1.0f);
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if (linear_volume != 1.0) {
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const float volume_scale_factor = std::exp(6.90775f * linear_volume) * 0.001f;
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for (std::size_t i = 0; i < num_frames; i++) {
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buffer[i * 2 + 0] = static_cast<s16>(buffer[i * 2 + 0] * volume_scale_factor);
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buffer[i * 2 + 1] = static_cast<s16>(buffer[i * 2 + 1] * volume_scale_factor);
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}
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}
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}
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} // namespace AudioCore
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