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Corrects clocking issues around audio, and cuts down queue costs.

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This commit is contained in:
Thomas Harte 2019-07-17 14:41:36 -04:00
parent ee8d853fcb
commit 22ee51c12c
3 changed files with 34 additions and 50 deletions
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62
Machines/Apple/Macintosh/Audio.cpp
View File

@ -11,7 +11,10 @@
using namespace Apple::Macintosh;
namespace {
const std::size_t sample_length = 352;
// The sample_length is coupled with the clock rate selected within the Macintosh proper.
const std::size_t sample_length = 352 / 2;
}
Audio::Audio(Concurrency::DeferringAsyncTaskQueue &task_queue) : task_queue_(task_queue) {}
@ -19,20 +22,17 @@ Audio::Audio(Concurrency::DeferringAsyncTaskQueue &task_queue) : task_queue_(tas
// MARK: - Inputs
void Audio::post_sample(uint8_t sample) {
// Grab the read and write pointers, ensure there's room for a new sample and, if not,
// drop this one.
const auto write_pointer = sample_queue_.write_pointer.load();
const auto read_pointer = sample_queue_.read_pointer.load();
const decltype(write_pointer) next_write_pointer = (write_pointer + 1) % sample_queue_.buffer.size();
if(next_write_pointer == read_pointer) {
return;
}
sample_queue_.buffer[write_pointer] = sample;
sample_queue_.write_pointer.store(next_write_pointer);
// Store sample directly indexed by current write pointer; this ensures that collected samples
// directly map to volume and enabled/disabled states.
sample_queue_.buffer[sample_queue_.write_pointer] = sample;
sample_queue_.write_pointer = (sample_queue_.write_pointer + 1) % sample_queue_.buffer.size();
}
void Audio::set_volume(int volume) {
// Do nothing if the volume hasn't changed.
if(posted_volume_ == volume) return;
posted_volume_ = volume;
// Post the volume change as a deferred event.
task_queue_.defer([=] () {
volume_ = volume;
@ -40,9 +40,13 @@ void Audio::set_volume(int volume) {
}
void Audio::set_enabled(bool on) {
// Do nothing if the mask hasn't changed.
if(posted_enable_mask_ == int(on)) return;
posted_enable_mask_ = int(on);
// Post the enabled mask change as a deferred event.
task_queue_.defer([=] () {
enabled_mask_ = on ? 1 : 0;
enabled_mask_ = int(on);
});
}
@ -58,46 +62,18 @@ void Audio::set_sample_volume_range(std::int16_t range) {
}
void Audio::get_samples(std::size_t number_of_samples, int16_t *target) {
const auto write_pointer = sample_queue_.write_pointer.load();
auto read_pointer = sample_queue_.read_pointer.load();
// TODO: the implementation below acts as if the hardware uses pulse-amplitude modulation;
// in fact it uses pulse-width modulation. But the scale for pulses isn't specified, so
// that's something to return to.
// TODO: temporary implementation. Very inefficient. Replace.
for(std::size_t sample = 0; sample < number_of_samples; ++sample) {
// if(volume_ && enabled_mask_) printf("%d\n", sample_queue_.buffer[read_pointer]);
target[sample] = volume_multiplier_ * int16_t(sample_queue_.buffer[read_pointer] * volume_ * enabled_mask_);
target[sample] = volume_multiplier_ * int16_t(sample_queue_.buffer[sample_queue_.read_pointer] * volume_ * enabled_mask_);
++subcycle_offset_;
if(subcycle_offset_ == sample_length) {
// printf("%d: %d\n", sample_queue_.buffer[read_pointer], volume_multiplier_ * int16_t(sample_queue_.buffer[read_pointer]));
subcycle_offset_ = 0;
const unsigned int next_read_pointer = (read_pointer + 1) % sample_queue_.buffer.size();
if(next_read_pointer != write_pointer) {
read_pointer = next_read_pointer;
}
sample_queue_.read_pointer = (sample_queue_.read_pointer + 1) % sample_queue_.buffer.size();
}
}
sample_queue_.read_pointer.store(read_pointer);
}
void Audio::skip_samples(std::size_t number_of_samples) {
const auto write_pointer = sample_queue_.write_pointer.load();
auto read_pointer = sample_queue_.read_pointer.load();
// Number of samples that would be consumed is (number_of_samples + subcycle_offset_) / sample_length.
const unsigned int samples_passed = static_cast<unsigned int>((number_of_samples + subcycle_offset_) / sample_length);
subcycle_offset_ = (number_of_samples + subcycle_offset_) % sample_length;
// Get also number of samples available.
const unsigned int samples_available = static_cast<unsigned int>((write_pointer + sample_queue_.buffer.size() - read_pointer) % sample_queue_.buffer.size());
// Advance by whichever of those is the lower number.
const auto samples_to_consume = std::min(samples_available, samples_passed);
read_pointer = (read_pointer + samples_to_consume) % sample_queue_.buffer.size();
sample_queue_.read_pointer.store(read_pointer);
}

16
Machines/Apple/Macintosh/Audio.hpp
View File

@ -20,8 +20,10 @@ namespace Apple {
namespace Macintosh {
/*!
Implements the Macintosh's audio output hardware, using a
combination
Implements the Macintosh's audio output hardware.
Designed to be clocked at half the rate of the real hardware — i.e.
a shade less than 4Mhz.
*/
class Audio: public ::Outputs::Speaker::SampleSource {
public:
@ -51,7 +53,6 @@ class Audio: public ::Outputs::Speaker::SampleSource {
// to satisfy ::Outputs::Speaker (included via ::Outputs::Filter.
void get_samples(std::size_t number_of_samples, int16_t *target);
void skip_samples(std::size_t number_of_samples);
bool is_zero_level();
void set_sample_volume_range(std::int16_t range);
@ -61,10 +62,15 @@ class Audio: public ::Outputs::Speaker::SampleSource {
// A queue of fetched samples; read from by one thread,
// written to by another.
struct {
std::array<uint8_t, 4096> buffer;
std::atomic<unsigned int> read_pointer, write_pointer;
std::array<uint8_t, 740> buffer;
size_t read_pointer = 0, write_pointer = 0;
} sample_queue_;
// Emulator-thread stateful variables, to avoid work posting
// deferral updates if possible.
int posted_volume_ = 0;
int posted_enable_mask_ = 0;
// Stateful variables, modified from the audio generation
// thread only.
int volume_ = 0;

6
Machines/Apple/Macintosh/Macintosh.cpp
View File

@ -112,7 +112,7 @@ template <Analyser::Static::Macintosh::Target::Model model> class ConcreteMachin
// The Mac runs at 7.8336mHz.
set_clock_rate(double(CLOCK_RATE));
audio_.speaker.set_input_rate(float(CLOCK_RATE));
audio_.speaker.set_input_rate(float(CLOCK_RATE) / 2.0f);
// Insert any supplied media.
insert_media(target.media);
@ -543,7 +543,9 @@ template <Analyser::Static::Macintosh::Target::Model model> class ConcreteMachin
}
void run_for(HalfCycles duration) {
audio_.time_since_update += duration;
// The 6522 enjoys a divide-by-ten, so multiply back up here to make the
// divided-by-two clock the audio works on.
audio_.time_since_update += HalfCycles(duration.as_int() * 5);
}
void flush() {