mirror of
https://github.com/TomHarte/CLK.git
synced 2024-11-25 16:31:42 +00:00
212 lines
5.3 KiB
C++
212 lines
5.3 KiB
C++
//
|
|
// Audio.hpp
|
|
// Clock Signal
|
|
//
|
|
// Created by Thomas Harte on 20/03/2024.
|
|
// Copyright © 2024 Thomas Harte. All rights reserved.
|
|
//
|
|
|
|
#pragma once
|
|
|
|
#include "../../../Concurrency/AsyncTaskQueue.hpp"
|
|
#include "../../../Outputs/Speaker/Implementation/LowpassSpeaker.hpp"
|
|
|
|
#include <array>
|
|
#include <cstdint>
|
|
|
|
namespace Archimedes {
|
|
|
|
// Generate lookup table for sound output levels, and hold it only once regardless
|
|
// of how many template instantiations there are of @c Sound.
|
|
static constexpr std::array<int16_t, 256> generate_levels() {
|
|
std::array<int16_t, 256> result{};
|
|
|
|
// There are 8 segments of 16 steps; each segment is a linear
|
|
// interpolation from its start level to its end level and
|
|
// each level is double the previous.
|
|
//
|
|
// Bit 7 provides a sign.
|
|
|
|
for(size_t c = 0; c < 256; c++) {
|
|
// This is the VIDC1 rule.
|
|
// const bool is_negative = c & 128;
|
|
// const auto point = static_cast<int>(c & 0xf);
|
|
// const auto chord = static_cast<int>((c >> 4) & 7);
|
|
|
|
// VIDC2 rule, which seems to be effective. I've yet to spot the rule by which
|
|
// VIDC1/2 is detected.
|
|
const bool is_negative = c & 1;
|
|
const auto point = static_cast<int>((c >> 1) & 0xf);
|
|
const auto chord = static_cast<int>((c >> 5) & 7);
|
|
|
|
const int start = (1 << chord) - 1;
|
|
const int end = (chord == 7) ? 247 : ((start << 1) + 1);
|
|
|
|
const int level = start * (16 - point) + end * point;
|
|
result[c] = static_cast<int16_t>((level * 32767) / 3832);
|
|
if(is_negative) result[c] = -result[c];
|
|
}
|
|
|
|
return result;
|
|
}
|
|
struct SoundLevels {
|
|
static constexpr auto levels = generate_levels();
|
|
};
|
|
|
|
/// Models the Archimedes sound output; in a real machine this is a joint efort between the VIDC and the MEMC.
|
|
template <typename InterruptObserverT>
|
|
struct Sound: private SoundLevels {
|
|
Sound(InterruptObserverT &observer, const uint8_t *ram) : ram_(ram), observer_(observer) {
|
|
speaker_.set_input_rate(1'000'000);
|
|
speaker_.set_high_frequency_cutoff(2'200.0f);
|
|
}
|
|
|
|
void set_next_end(uint32_t value) {
|
|
next_.end = value;
|
|
}
|
|
|
|
void set_next_start(uint32_t value) {
|
|
next_.start = value;
|
|
set_buffer_valid(true); // My guess: this is triggered on next buffer start write.
|
|
|
|
// Definitely wrong; testing.
|
|
// set_halted(false);
|
|
}
|
|
|
|
bool interrupt() const {
|
|
return !next_buffer_valid_;
|
|
}
|
|
|
|
void swap() {
|
|
current_.start = next_.start;
|
|
std::swap(current_.end, next_.end);
|
|
set_buffer_valid(false);
|
|
set_halted(false);
|
|
}
|
|
|
|
void set_frequency(uint8_t frequency) {
|
|
divider_ = reload_ = frequency;
|
|
}
|
|
|
|
void set_stereo_image(uint8_t channel, uint8_t value) {
|
|
if(!value) {
|
|
positions_[channel].left =
|
|
positions_[channel].right = 0;
|
|
return;
|
|
}
|
|
|
|
positions_[channel].right = value - 1;
|
|
positions_[channel].left = 6 - positions_[channel].right;
|
|
}
|
|
|
|
void set_dma_enabled(bool enabled) {
|
|
dma_enabled_ = enabled;
|
|
}
|
|
|
|
void tick() {
|
|
// Write silence if not currently outputting.
|
|
if(halted_ || !dma_enabled_) {
|
|
post_sample(Outputs::Speaker::StereoSample());
|
|
return;
|
|
}
|
|
|
|
// Apply user-programmed clock divider.
|
|
--divider_;
|
|
if(!divider_) {
|
|
divider_ = reload_ + 2;
|
|
|
|
// Grab a single byte from the FIFO.
|
|
const uint8_t raw = ram_[static_cast<std::size_t>(current_.start) + static_cast<std::size_t>(byte_)];
|
|
sample_ = Outputs::Speaker::StereoSample( // TODO: pan, volume.
|
|
static_cast<int16_t>((levels[raw] * positions_[byte_ & 7].left) / 6),
|
|
static_cast<int16_t>((levels[raw] * positions_[byte_ & 7].right) / 6)
|
|
);
|
|
++byte_;
|
|
|
|
// If the FIFO is exhausted, consider triggering a DMA request.
|
|
if(byte_ == 16) {
|
|
byte_ = 0;
|
|
|
|
current_.start += 16;
|
|
if(current_.start == current_.end) {
|
|
if(next_buffer_valid_) {
|
|
swap();
|
|
} else {
|
|
set_halted(true);
|
|
}
|
|
}
|
|
}
|
|
}
|
|
post_sample(sample_);
|
|
}
|
|
|
|
Outputs::Speaker::Speaker *speaker() {
|
|
return &speaker_;
|
|
}
|
|
|
|
~Sound() {
|
|
while(is_posting_.test_and_set());
|
|
}
|
|
|
|
private:
|
|
const uint8_t *ram_ = nullptr;
|
|
|
|
uint8_t divider_ = 0, reload_ = 0;
|
|
int byte_ = 0;
|
|
|
|
void set_buffer_valid(bool valid) {
|
|
next_buffer_valid_ = valid;
|
|
observer_.update_interrupts();
|
|
}
|
|
|
|
void set_halted(bool halted) {
|
|
if(halted_ != halted && !halted) {
|
|
byte_ = 0;
|
|
divider_ = reload_;
|
|
}
|
|
halted_ = halted;
|
|
}
|
|
|
|
bool next_buffer_valid_ = false;
|
|
bool halted_ = true; // This is a bit of a guess.
|
|
bool dma_enabled_ = false;
|
|
|
|
struct Buffer {
|
|
uint32_t start = 0, end = 0;
|
|
};
|
|
Buffer current_, next_;
|
|
|
|
struct StereoPosition {
|
|
// These are maintained as sixths, i.e. a value of 6 means 100%.
|
|
int left, right;
|
|
} positions_[8];
|
|
|
|
InterruptObserverT &observer_;
|
|
Outputs::Speaker::PushLowpass<true> speaker_;
|
|
Concurrency::AsyncTaskQueue<true> queue_;
|
|
|
|
void post_sample(Outputs::Speaker::StereoSample sample) {
|
|
samples_[sample_target_][sample_pointer_++] = sample;
|
|
if(sample_pointer_ == samples_[0].size()) {
|
|
while(is_posting_.test_and_set());
|
|
|
|
const auto post_source = sample_target_;
|
|
sample_target_ ^= 1;
|
|
sample_pointer_ = 0;
|
|
queue_.enqueue([this, post_source]() {
|
|
speaker_.push(reinterpret_cast<int16_t *>(samples_[post_source].data()), samples_[post_source].size());
|
|
is_posting_.clear();
|
|
});
|
|
}
|
|
}
|
|
std::size_t sample_pointer_ = 0;
|
|
std::size_t sample_target_ = 0;
|
|
Outputs::Speaker::StereoSample sample_;
|
|
|
|
using SampleBuffer = std::array<Outputs::Speaker::StereoSample, 4096>;
|
|
std::array<SampleBuffer, 2> samples_;
|
|
std::atomic_flag is_posting_ = ATOMIC_FLAG_INIT;
|
|
};
|
|
|
|
}
|