/***************************************************************************
 *   Copyright (C) 2020 by Federico Amedeo Izzo IU2NUO,                    *
 *                         Niccolò Izzo IU2KIN,                            *
 *                         Silvano Seva IU2KWO,                            *
 *                         Frederik Saraci IU2NRO                          *
 *                                                                         *
 *   This program is free software; you can redistribute it and/or modify  *
 *   it under the terms of the GNU General Public License as published by  *
 *   the Free Software Foundation; either version 3 of the License, or     *
 *   (at your option) any later version.                                   *
 *                                                                         *
 *   This program is distributed in the hope that it will be useful,       *
 *   but WITHOUT ANY WARRANTY; without even the implied warranty of        *
 *   MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the         *
 *   GNU General Public License for more details.                          *
 *                                                                         *
 *   You should have received a copy of the GNU General Public License     *
 *   along with this program; if not, see <http://www.gnu.org/licenses/>   *
 ***************************************************************************/

#ifndef DSP_H
#define DSP_H

#include <inttypes.h>
#include <stdlib.h>

typedef int16_t audio_sample_t;

/*
 * This header contains various DSP utilities which can be used to condition
 * input or output signals when implementing digital modes on OpenRTX.
 */

#ifdef __cplusplus
#include <array>
extern "C" {
#endif

/*
 * Compensate for the filtering applied by the PWM output over the modulated
 * signal. The buffer will be processed in place to save memory.
 *
 * @param buffer: the buffer to be used as both source and destination.
 * @param length: the length of the input buffer.
 */
void dsp_pwmCompensate(audio_sample_t *buffer, uint16_t length);

/*
 * Remove any DC bias from the audio buffer passed as parameter.
 * The buffer will be processed in place to save memory.
 *
 * @param buffer: the buffer to be used as both source and destination.
 * @param length: the length of the input buffer.
 */
void dsp_dcRemoval(audio_sample_t *buffer, uint16_t length);

#ifdef __cplusplus
}

/*
 * Applies a generic FIR filter on the audio buffer passed as parameter.
 * The buffer will be processed in place to save memory.
 *
 * @param buffer: the buffer to be used as both source and destination.
 * @param length: the length of the input buffer.
 * @param taps: an array of coefficients which defines the transfer function.
 */
template<size_t order>
void dsp_applyFIR(audio_sample_t *buffer,
                  uint16_t length,
                  std::array<float, order> taps);

#endif // __cplusplus

#endif /* DSP_H */