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399 lines
13 KiB
C++
399 lines
13 KiB
C++
//
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// LowpassSpeaker.hpp
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// Clock Signal
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//
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// Created by Thomas Harte on 15/12/2017.
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// Copyright 2017 Thomas Harte. All rights reserved.
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//
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#ifndef LowpassSpeaker_hpp
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#define LowpassSpeaker_hpp
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#include "../Speaker.hpp"
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#include "../../../SignalProcessing/FIRFilter.hpp"
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#include "../../../ClockReceiver/ClockReceiver.hpp"
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#include "../../../Concurrency/AsyncTaskQueue.hpp"
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#include <algorithm>
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#include <mutex>
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#include <cstring>
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#include <cmath>
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namespace Outputs {
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namespace Speaker {
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template <typename ConcreteT, bool is_stereo> class LowpassBase: public Speaker {
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public:
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/*!
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Sets the clock rate of the input audio.
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*/
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void set_input_rate(float cycles_per_second) {
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std::lock_guard lock_guard(filter_parameters_mutex_);
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if(filter_parameters_.input_cycles_per_second == cycles_per_second) {
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return;
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}
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filter_parameters_.input_cycles_per_second = cycles_per_second;
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filter_parameters_.parameters_are_dirty = true;
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filter_parameters_.input_rate_changed = true;
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}
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/*!
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Allows a cut-off frequency to be specified for audio. Ordinarily this low-pass speaker
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will determine a cut-off based on the output audio rate. A caller can manually select
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an alternative cut-off. This allows machines with a low-pass filter on their audio output
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path to be explicit about its effect, and get that simulation for free.
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*/
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void set_high_frequency_cutoff(float high_frequency) {
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std::lock_guard lock_guard(filter_parameters_mutex_);
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if(filter_parameters_.high_frequency_cutoff == high_frequency) {
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return;
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}
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filter_parameters_.high_frequency_cutoff = high_frequency;
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filter_parameters_.parameters_are_dirty = true;
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}
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private:
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float get_ideal_clock_rate_in_range(float minimum, float maximum) final {
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std::lock_guard lock_guard(filter_parameters_mutex_);
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// Return twice the cut off, if applicable.
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if( filter_parameters_.high_frequency_cutoff > 0.0f &&
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filter_parameters_.input_cycles_per_second >= filter_parameters_.high_frequency_cutoff * 3.0f &&
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filter_parameters_.input_cycles_per_second <= filter_parameters_.high_frequency_cutoff * 3.0f)
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return filter_parameters_.high_frequency_cutoff * 3.0f;
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// Return exactly the input rate if possible.
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if( filter_parameters_.input_cycles_per_second >= minimum &&
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filter_parameters_.input_cycles_per_second <= maximum)
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return filter_parameters_.input_cycles_per_second;
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// If the input rate is lower, return the minimum...
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if(filter_parameters_.input_cycles_per_second < minimum)
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return minimum;
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// ... otherwise, return the maximum.
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return maximum;
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}
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// Implemented as per Speaker.
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void set_computed_output_rate(float cycles_per_second, int buffer_size, bool) final {
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std::lock_guard lock_guard(filter_parameters_mutex_);
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if(filter_parameters_.output_cycles_per_second == cycles_per_second && size_t(buffer_size) == output_buffer_.size()) {
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return;
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}
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filter_parameters_.output_cycles_per_second = cycles_per_second;
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filter_parameters_.parameters_are_dirty = true;
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output_buffer_.resize(std::size_t(buffer_size) * (is_stereo + 1));
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}
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// MARK: - Filtering.
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std::size_t output_buffer_pointer_ = 0;
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std::size_t input_buffer_depth_ = 0;
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std::vector<int16_t> input_buffer_;
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std::vector<int16_t> output_buffer_;
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float step_rate_ = 0.0f;
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float position_error_ = 0.0f;
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std::unique_ptr<SignalProcessing::FIRFilter> filter_;
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std::mutex filter_parameters_mutex_;
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struct FilterParameters {
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float input_cycles_per_second = 0.0f;
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float output_cycles_per_second = 0.0f;
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float high_frequency_cutoff = -1.0;
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bool parameters_are_dirty = true;
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bool input_rate_changed = false;
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} filter_parameters_;
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void update_filter_coefficients(const FilterParameters &filter_parameters) {
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float high_pass_frequency = filter_parameters.output_cycles_per_second / 2.0f;
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if(filter_parameters.high_frequency_cutoff > 0.0) {
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high_pass_frequency = std::min(filter_parameters.high_frequency_cutoff, high_pass_frequency);
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}
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// Make a guess at a good number of taps.
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std::size_t number_of_taps = std::size_t(
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ceilf((filter_parameters.input_cycles_per_second + high_pass_frequency) / high_pass_frequency)
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);
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number_of_taps = (number_of_taps * 2) | 1;
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step_rate_ = filter_parameters.input_cycles_per_second / filter_parameters.output_cycles_per_second;
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position_error_ = 0.0f;
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filter_ = std::make_unique<SignalProcessing::FIRFilter>(
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unsigned(number_of_taps),
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filter_parameters.input_cycles_per_second,
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0.0,
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high_pass_frequency,
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SignalProcessing::FIRFilter::DefaultAttenuation);
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// Pick the new conversion function.
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if( filter_parameters.input_cycles_per_second == filter_parameters.output_cycles_per_second &&
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filter_parameters.high_frequency_cutoff < 0.0) {
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// If input and output rates exactly match, and no additional cut-off has been specified,
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// just accumulate results and pass on.
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conversion_ = Conversion::Copy;
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} else if( filter_parameters.input_cycles_per_second > filter_parameters.output_cycles_per_second ||
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(filter_parameters.input_cycles_per_second == filter_parameters.output_cycles_per_second && filter_parameters.high_frequency_cutoff >= 0.0)) {
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// If the output rate is less than the input rate, or an additional cut-off has been specified, use the filter.
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conversion_ = Conversion::ResampleSmaller;
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} else {
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conversion_ = Conversion::ResampleLarger;
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}
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// Do something sensible with any dangling input, if necessary.
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const int scale = static_cast<ConcreteT *>(this)->get_scale();
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switch(conversion_) {
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// Neither direct copying nor resampling larger currently use any temporary input.
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// Although in the latter case that's just because it's unimplemented. But, regardless,
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// that means nothing to do.
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default: break;
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case Conversion::ResampleSmaller: {
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// Reize the input buffer only if absolutely necessary; if sizing downward
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// such that a sample would otherwise be lost then output it now. Keep anything
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// currently in the input buffer that hasn't yet been processed.
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const size_t required_buffer_size = size_t(number_of_taps) * (is_stereo + 1);
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if(input_buffer_.size() != required_buffer_size) {
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if(input_buffer_depth_ >= required_buffer_size) {
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resample_input_buffer(scale);
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input_buffer_depth_ %= required_buffer_size;
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}
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input_buffer_.resize(required_buffer_size);
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}
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} break;
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}
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}
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inline void resample_input_buffer(int scale) {
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if constexpr (is_stereo) {
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output_buffer_[output_buffer_pointer_ + 0] = filter_->apply(input_buffer_.data(), 2);
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output_buffer_[output_buffer_pointer_ + 1] = filter_->apply(input_buffer_.data() + 1, 2);
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output_buffer_pointer_+= 2;
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} else {
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output_buffer_[output_buffer_pointer_] = filter_->apply(input_buffer_.data());
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output_buffer_pointer_++;
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}
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// Apply scale, if supplied, clamping appropriately.
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if(scale != 65536) {
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#define SCALE(x) x = int16_t(std::clamp((int(x) * scale) >> 16, -32768, 32767))
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if constexpr (is_stereo) {
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SCALE(output_buffer_[output_buffer_pointer_ - 2]);
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SCALE(output_buffer_[output_buffer_pointer_ - 1]);
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} else {
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SCALE(output_buffer_[output_buffer_pointer_ - 1]);
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}
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#undef SCALE
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}
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// Announce to delegate if full.
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if(output_buffer_pointer_ == output_buffer_.size()) {
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output_buffer_pointer_ = 0;
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did_complete_samples(this, output_buffer_, is_stereo);
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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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const size_t steps = size_t(step_rate_ + position_error_) * (is_stereo + 1);
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position_error_ = fmodf(step_rate_ + position_error_, 1.0f);
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if(steps < input_buffer_.size()) {
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auto *const input_buffer = input_buffer_.data();
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std::memmove( input_buffer,
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&input_buffer[steps],
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sizeof(int16_t) * (input_buffer_.size() - steps));
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input_buffer_depth_ -= steps;
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} else {
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if(steps > input_buffer_.size()) {
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static_cast<ConcreteT *>(this)->skip_samples((steps - input_buffer_.size()) / (1 + is_stereo));
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}
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input_buffer_depth_ = 0;
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}
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}
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enum class Conversion {
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ResampleSmaller,
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Copy,
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ResampleLarger
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} conversion_ = Conversion::Copy;
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bool recalculate_filter_if_dirty() {
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FilterParameters filter_parameters;
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{
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std::lock_guard lock_guard(filter_parameters_mutex_);
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filter_parameters = filter_parameters_;
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filter_parameters_.parameters_are_dirty = false;
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filter_parameters_.input_rate_changed = false;
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}
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if(filter_parameters.parameters_are_dirty) update_filter_coefficients(filter_parameters);
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return filter_parameters.input_rate_changed;
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}
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protected:
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void process(size_t length) {
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const auto delegate = delegate_.load(std::memory_order::memory_order_relaxed);
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if(!delegate) return;
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const int scale = static_cast<ConcreteT *>(this)->get_scale();
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if(recalculate_filter_if_dirty()) {
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delegate->speaker_did_change_input_clock(this);
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}
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switch(conversion_) {
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case Conversion::Copy:
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while(length) {
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const auto samples_to_read = std::min((output_buffer_.size() - output_buffer_pointer_) / (1 + is_stereo), length);
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static_cast<ConcreteT *>(this)->get_samples(samples_to_read, &output_buffer_[output_buffer_pointer_ ]);
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output_buffer_pointer_ += samples_to_read * (1 + is_stereo);
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// TODO: apply scale.
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// Announce to delegate if full.
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if(output_buffer_pointer_ == output_buffer_.size()) {
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output_buffer_pointer_ = 0;
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did_complete_samples(this, output_buffer_, is_stereo);
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}
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length -= samples_to_read;
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}
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break;
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case Conversion::ResampleSmaller:
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while(length) {
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const auto cycles_to_read = std::min((input_buffer_.size() - input_buffer_depth_) / (1 + is_stereo), length);
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static_cast<ConcreteT *>(this)->get_samples(cycles_to_read, &input_buffer_[input_buffer_depth_]);
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input_buffer_depth_ += cycles_to_read * (1 + is_stereo);
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if(input_buffer_depth_ == input_buffer_.size()) {
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resample_input_buffer(scale);
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}
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length -= cycles_to_read;
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}
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break;
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case Conversion::ResampleLarger:
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// TODO: input rate is less than output rate.
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break;
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}
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}
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};
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/*!
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Provides a low-pass speaker to which blocks of samples are pushed.
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*/
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template <bool is_stereo> class PushLowpass: public LowpassBase<PushLowpass<is_stereo>, is_stereo> {
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private:
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using BaseT = LowpassBase<PushLowpass<is_stereo>, is_stereo>;
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friend BaseT;
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using BaseT::process;
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std::atomic<uint16_t> scale_ = 32767;
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int get_scale() {
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return scale_;
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}
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const int16_t *buffer_ = nullptr;
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void skip_samples(size_t count) {
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buffer_ += count;
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}
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void get_samples(size_t length, int16_t *target) {
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memcpy(target, buffer_, length);
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buffer_ += length;
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}
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public:
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void set_output_volume(float volume) final {
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scale_.load(std::clamp(volume * 65535.0f, 0.0f, 65535.0f));
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}
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bool get_is_stereo() final {
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return is_stereo;
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}
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/*!
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Filters and posts onward the provided buffer, on the calling thread.
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*/
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void push(const int16_t *buffer, size_t length) {
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buffer_ = buffer;
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process(length);
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}
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};
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/*!
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The low-pass speaker expects an Outputs::Speaker::SampleSource-derived
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template class, and uses the instance supplied to its constructor as the
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source of a high-frequency stream of audio which it filters down to a
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lower-frequency output.
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*/
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template <typename SampleSource> class PullLowpass: public LowpassBase<PullLowpass<SampleSource>, SampleSource::get_is_stereo()> {
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public:
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PullLowpass(SampleSource &sample_source) : sample_source_(sample_source) {
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// Propagate an initial volume level.
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sample_source.set_sample_volume_range(32767);
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}
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void set_output_volume(float volume) final {
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// Clamp to the acceptable range, and set.
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volume = std::clamp(volume, 0.0f, 1.0f);
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sample_source_.set_sample_volume_range(int16_t(32767.0f * volume));
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}
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bool get_is_stereo() final {
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return SampleSource::get_is_stereo();
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}
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/*!
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Schedules an advancement by the number of cycles specified on the provided queue.
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The speaker will advance by obtaining data from the sample source supplied
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at construction, filtering it and passing it on to the speaker's delegate if there is one.
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*/
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void run_for(Concurrency::DeferringAsyncTaskQueue &queue, const Cycles cycles) {
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queue.defer([this, cycles] {
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run_for(cycles);
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});
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}
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private:
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using BaseT = LowpassBase<PullLowpass<SampleSource>, SampleSource::get_is_stereo()>;
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friend BaseT;
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using BaseT::process;
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/*!
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Advances by the number of cycles specified, obtaining data from the sample source supplied
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at construction, filtering it and passing it on to the speaker's delegate if there is one.
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*/
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void run_for(const Cycles cycles) {
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std::size_t cycles_remaining = size_t(cycles.as_integral());
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if(!cycles_remaining) return;
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process(cycles_remaining);
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}
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SampleSource &sample_source_;
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void skip_samples(size_t count) {
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sample_source_.skip_samples(count);
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}
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int get_scale() {
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return int(65536.0 / sample_source_.get_average_output_peak());
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}
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void get_samples(size_t length, int16_t *target) {
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sample_source_.get_samples(length, target);
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}
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};
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}
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}
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#endif /* LowpassSpeaker_hpp */
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