162 lines
4.9 KiB
C++
162 lines
4.9 KiB
C++
/***************************************************************************
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* Copyright (C) 2020 - 2022 by Federico Amedeo Izzo IU2NUO, *
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* Niccolò Izzo IU2KIN, *
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* Silvano Seva IU2KWO, *
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* Frederik Saraci IU2NRO *
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* *
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* This program is free software; you can redistribute it and/or modify *
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* it under the terms of the GNU General Public License as published by *
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* the Free Software Foundation; either version 3 of the License, or *
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* (at your option) any later version. *
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* *
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* This program is distributed in the hope that it will be useful, *
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* but WITHOUT ANY WARRANTY; without even the implied warranty of *
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the *
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* GNU General Public License for more details. *
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* *
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* You should have received a copy of the GNU General Public License *
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* along with this program; if not, see <http://www.gnu.org/licenses/> *
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***************************************************************************/
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#ifndef DSP_H
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#define DSP_H
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#include <inttypes.h>
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#include <stdlib.h>
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typedef int16_t audio_sample_t;
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/*
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* This header contains various DSP utilities which can be used to condition
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* input or output signals when implementing digital modes on OpenRTX.
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*/
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#ifdef __cplusplus
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#include <array>
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#include <cstddef>
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extern "C" {
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#endif
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/**
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* Data structure holding the internal state of a filter.
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*/
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typedef struct
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{
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float u[3]; // input values u(k), u(k-1), u(k-2)
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float y[3]; // output values y(k), y(k-1), y(k-2)
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bool initialised; // state variables initialised
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}
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filter_state_t;
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/**
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* Reset the filter state variables.
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*
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* @param state: pointer to the data structure containing the filter state.
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*/
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void dsp_resetFilterState(filter_state_t *state);
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/**
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* Compensate for the filtering applied by the PWM output over the modulated
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* signal. The buffer is be processed in place to save memory.
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*
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* @param state: pointer to the data structure containing the filter state.
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* @param buffer: the buffer to be used as both source and destination.
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* @param length: the length of the input buffer.
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*/
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void dsp_pwmCompensate(filter_state_t *state, audio_sample_t *buffer,
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size_t length);
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/**
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* Remove the DC offset from a collection of audio samples, processing data
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* in-place.
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*
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* @param state: pointer to the data structure containing the filter state.
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* @param buffer: buffer containing the audio samples.
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* @param length: number of samples contained in the buffer.
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*/
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void dsp_dcRemoval(filter_state_t *state, audio_sample_t *buffer, size_t length);
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/*
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* Inverts the phase of the audio buffer passed as paramenter.
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* The buffer will be processed in place to save memory.
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*
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* @param buffer: the buffer to be used as both source and destination.
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* @param length: the length of the input buffer.
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*/
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void dsp_invertPhase(audio_sample_t *buffer, uint16_t length);
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#ifdef __cplusplus
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}
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/**
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* Class for FIR filter with configurable coefficients.
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* Adapted from the original implementation by Rob Riggs, Mobilinkd LLC.
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*/
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template < size_t N >
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class Fir
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{
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public:
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/**
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* Constructor.
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*
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* @param taps: reference to a std::array of floating poing values representing
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* the FIR filter coefficients.
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*/
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Fir(const std::array< float, N >& taps) : taps(taps), pos(0)
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{
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reset();
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}
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/**
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* Destructor.
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*/
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~Fir() { }
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/**
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* Perform one step of the FIR filter, computing a new output value given
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* the input value and the history of previous input values.
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*
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* @param input: FIR input value for the current time step.
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* @return FIR output as a function of the current and past input values.
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*/
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float operator()(const float& input)
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{
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hist[pos++] = input;
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if(pos >= N) pos = 0;
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float result = 0.0;
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size_t index = pos;
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for(size_t i = 0; i < N; i++)
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{
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index = (index != 0 ? index - 1 : N - 1);
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result += hist[index] * taps[i];
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}
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return result;
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}
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/**
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* Reset FIR history, clearing the memory of past values.
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*/
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void reset()
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{
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hist.fill(0);
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pos = 0;
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}
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private:
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const std::array< float, N >& taps; ///< FIR filter coefficients.
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std::array< float, N > hist; ///< History of past inputs.
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size_t pos; ///< Current position in history.
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};
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#endif // __cplusplus
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#endif /* DSP_H */
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