/*************************************************************************** * Copyright (C) 2020 - 2025 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 * ***************************************************************************/ #ifndef DSP_H #define DSP_H #include #include #include #ifdef __cplusplus extern "C" { #endif 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. */ /** * Data structure holding the internal state of a filter. */ typedef struct { float u[3]; // input values u(k), u(k-1), u(k-2) float y[3]; // output values y(k), y(k-1), y(k-2) bool initialised; // state variables initialised } filter_state_t; /** * Reset the filter state variables. * * @param state: pointer to the data structure containing the filter state. */ void dsp_resetFilterState(filter_state_t *state); /** * Remove the DC offset from a collection of audio samples, processing data * in-place. * * @param state: pointer to the data structure containing the filter state. * @param buffer: buffer containing the audio samples. * @param length: number of samples contained in the buffer. */ void dsp_dcRemoval(filter_state_t *state, audio_sample_t *buffer, size_t length); /* * Inverts the phase of the audio buffer passed as paramenter. * 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_invertPhase(audio_sample_t *buffer, uint16_t length); #ifdef __cplusplus } #endif // __cplusplus #endif /* DSP_H */