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254 lines
8.8 KiB
C++
254 lines
8.8 KiB
C++
//
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// Speaker.hpp
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// Clock Signal
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//
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// Created by Thomas Harte on 12/01/2016.
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// Copyright © 2016 Thomas Harte. All rights reserved.
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//
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#ifndef Speaker_hpp
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#define Speaker_hpp
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#include <cstdint>
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#include <cstdio>
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#include <ctime>
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#include <memory>
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#include <list>
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#include <vector>
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#include "../SignalProcessing/Stepper.hpp"
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#include "../SignalProcessing/FIRFilter.hpp"
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#include "../Concurrency/AsyncTaskQueue.hpp"
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#include "../ClockReceiver/ClockReceiver.hpp"
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namespace Outputs {
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/*!
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Provides the base class for an audio output source, with an input rate (the speed at which the source will
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provide data), an output rate (the speed at which the destination will receive data), a delegate to receive
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the output and some help for the output in picking an appropriate rate once the input rate is known.
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Intended to be a parent class, allowing descendants to pick the strategy by which input samples are mapped to
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output samples.
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*/
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class Speaker {
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public:
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class Delegate {
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public:
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virtual void speaker_did_complete_samples(Speaker *speaker, const std::vector<int16_t> &buffer) = 0;
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};
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float get_ideal_clock_rate_in_range(float minimum, float maximum) {
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// return twice the cut off, if applicable
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if(high_frequency_cut_off_ > 0.0f && input_cycles_per_second_ >= high_frequency_cut_off_ * 3.0f && input_cycles_per_second_ <= high_frequency_cut_off_ * 3.0f) return high_frequency_cut_off_ * 3.0f;
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// return exactly the input rate if possible
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if(input_cycles_per_second_ >= minimum && input_cycles_per_second_ <= maximum) return input_cycles_per_second_;
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// if the input rate is lower, return the minimum
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if(input_cycles_per_second_ < minimum) return minimum;
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// otherwise, return the maximum
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return maximum;
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}
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void set_output_rate(float cycles_per_second, int buffer_size) {
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output_cycles_per_second_ = cycles_per_second;
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buffer_in_progress_.resize((size_t)buffer_size);
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set_needs_updated_filter_coefficients();
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}
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void set_output_quality(int number_of_taps) {
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requested_number_of_taps_ = (size_t)number_of_taps;
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set_needs_updated_filter_coefficients();
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}
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void set_delegate(Delegate *delegate) {
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delegate_ = delegate;
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}
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void set_input_rate(float cycles_per_second) {
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input_cycles_per_second_ = cycles_per_second;
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set_needs_updated_filter_coefficients();
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}
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/*!
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Sets the cut-off frequency for a low-pass filter attached to the output of this speaker; optional.
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*/
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void set_high_frequency_cut_off(float high_frequency) {
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high_frequency_cut_off_ = high_frequency;
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set_needs_updated_filter_coefficients();
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}
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Speaker() : buffer_in_progress_pointer_(0), requested_number_of_taps_(0), high_frequency_cut_off_(-1.0), _queue(new Concurrency::AsyncTaskQueue) {}
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/*!
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Ensures any deferred processing occurs now.
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*/
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void flush() {
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if(!queued_functions_) return;
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std::shared_ptr<std::list<std::function<void(void)>>> queued_functions = queued_functions_;
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queued_functions_.reset();
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_queue->enqueue([queued_functions] {
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for(auto function : *queued_functions) {
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function();
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}
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});
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}
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protected:
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void enqueue(std::function<void(void)> function) {
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if(!queued_functions_) queued_functions_.reset(new std::list<std::function<void(void)>>);
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queued_functions_->push_back(function);
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}
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std::shared_ptr<std::list<std::function<void(void)>>> queued_functions_;
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std::vector<int16_t> buffer_in_progress_;
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float high_frequency_cut_off_;
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size_t buffer_in_progress_pointer_;
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size_t number_of_taps_, requested_number_of_taps_;
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bool coefficients_are_dirty_;
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Delegate *delegate_;
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float input_cycles_per_second_, output_cycles_per_second_;
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void set_needs_updated_filter_coefficients() {
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coefficients_are_dirty_ = true;
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}
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void get_samples(unsigned int quantity, int16_t *target) {}
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void skip_samples(unsigned int quantity) {
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int16_t throwaway_samples[quantity];
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get_samples(quantity, throwaway_samples);
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}
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std::unique_ptr<Concurrency::AsyncTaskQueue> _queue;
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};
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/*!
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A concrete descendant of Speaker that uses a FIR filter to map from input data to output data when scaling
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and a copy-through buffer when input and output rates are the same.
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Audio sources should use @c Filter as both a template and a parent, implementing at least
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`get_samples(unsigned int quantity, int16_t *target)` and ideally also `skip_samples(unsigned int quantity)`
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to provide source data.
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Call `run_for` to request that the next period of input data is collected.
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*/
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template <class T> class Filter: public Speaker {
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public:
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~Filter() {
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_queue->flush();
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}
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void run_for(const Cycles cycles) {
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enqueue([=]() {
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unsigned int cycles_remaining = (unsigned int)cycles.as_int();
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if(coefficients_are_dirty_) update_filter_coefficients();
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// if input and output rates exactly match, just accumulate results and pass on
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if(input_cycles_per_second_ == output_cycles_per_second_ && high_frequency_cut_off_ < 0.0) {
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while(cycles_remaining) {
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unsigned int cycles_to_read = (unsigned int)(buffer_in_progress_.size() - (size_t)buffer_in_progress_pointer_);
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if(cycles_to_read > cycles_remaining) cycles_to_read = cycles_remaining;
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static_cast<T *>(this)->get_samples(cycles_to_read, &buffer_in_progress_[(size_t)buffer_in_progress_pointer_]);
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buffer_in_progress_pointer_ += cycles_to_read;
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// announce to delegate if full
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if(buffer_in_progress_pointer_ == buffer_in_progress_.size()) {
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buffer_in_progress_pointer_ = 0;
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if(delegate_) {
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delegate_->speaker_did_complete_samples(this, buffer_in_progress_);
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}
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}
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cycles_remaining -= cycles_to_read;
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}
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return;
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}
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// if the output rate is less than the input rate, use the filter
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if(input_cycles_per_second_ > output_cycles_per_second_ || (input_cycles_per_second_ == output_cycles_per_second_ && high_frequency_cut_off_ >= 0.0)) {
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while(cycles_remaining) {
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unsigned int cycles_to_read = (unsigned int)std::min((size_t)cycles_remaining, number_of_taps_ - input_buffer_depth_);
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static_cast<T *>(this)->get_samples(cycles_to_read, &input_buffer_[(size_t)input_buffer_depth_]);
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cycles_remaining -= cycles_to_read;
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input_buffer_depth_ += cycles_to_read;
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if(input_buffer_depth_ == number_of_taps_) {
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buffer_in_progress_[(size_t)buffer_in_progress_pointer_] = filter_->apply(input_buffer_.data());
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buffer_in_progress_pointer_++;
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// announce to delegate if full
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if(buffer_in_progress_pointer_ == buffer_in_progress_.size()) {
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buffer_in_progress_pointer_ = 0;
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if(delegate_) {
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delegate_->speaker_did_complete_samples(this, buffer_in_progress_);
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}
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}
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// If the next loop around is going to reuse some of the samples just collected, use a memmove to
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// preserve them in the correct locations (TODO: use a longer buffer to fix that) and don't skip
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// anything. Otherwise skip as required to get to the next sample batch and don't expect to reuse.
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uint64_t steps = stepper_->step();
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if(steps < number_of_taps_) {
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int16_t *input_buffer = input_buffer_.data();
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memmove(input_buffer, &input_buffer[steps], sizeof(int16_t) * ((size_t)number_of_taps_ - (size_t)steps));
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input_buffer_depth_ -= steps;
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} else {
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if(steps > number_of_taps_)
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static_cast<T *>(this)->skip_samples((unsigned int)steps - (unsigned int)number_of_taps_);
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input_buffer_depth_ = 0;
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}
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}
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}
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return;
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}
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// TODO: input rate is less than output rate
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});
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}
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private:
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std::unique_ptr<SignalProcessing::Stepper> stepper_;
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std::unique_ptr<SignalProcessing::FIRFilter> filter_;
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std::vector<int16_t> input_buffer_;
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size_t input_buffer_depth_;
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void update_filter_coefficients() {
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// make a guess at a good number of taps if this hasn't been provided explicitly
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if(requested_number_of_taps_) {
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number_of_taps_ = requested_number_of_taps_;
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} else {
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number_of_taps_ = (size_t)ceilf((input_cycles_per_second_ + output_cycles_per_second_) / output_cycles_per_second_);
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number_of_taps_ *= 2;
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number_of_taps_ |= 1;
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}
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coefficients_are_dirty_ = false;
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buffer_in_progress_pointer_ = 0;
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stepper_.reset(new SignalProcessing::Stepper((uint64_t)input_cycles_per_second_, (uint64_t)output_cycles_per_second_));
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float high_pass_frequency;
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if(high_frequency_cut_off_ > 0.0) {
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high_pass_frequency = std::min(output_cycles_per_second_ / 2.0f, high_frequency_cut_off_);
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} else {
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high_pass_frequency = output_cycles_per_second_ / 2.0f;
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}
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filter_.reset(new SignalProcessing::FIRFilter((unsigned int)number_of_taps_, (float)input_cycles_per_second_, 0.0, high_pass_frequency, SignalProcessing::FIRFilter::DefaultAttenuation));
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input_buffer_.resize((size_t)number_of_taps_);
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input_buffer_depth_ = 0;
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}
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};
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}
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#endif /* Speaker_hpp */
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