kopia lustrzana https://github.com/OpenRTX/OpenRTX
76 wiersze
2.9 KiB
C
76 wiersze
2.9 KiB
C
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/***************************************************************************
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* Copyright (C) 2020 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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extern "C" {
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#endif
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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 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_pwmCompensate(audio_sample_t *buffer, uint16_t length);
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/*
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* Remove any DC bias from the audio buffer passed as parameter.
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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_dcRemoval(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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* Applies a generic FIR filter on the audio buffer passed as parameter.
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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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* @param taps: an array of coefficients which defines the transfer function.
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*/
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template<size_t order>
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void dsp_applyFIR(audio_sample_t *buffer,
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uint16_t length,
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std::array<float, order> taps);
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#endif // __cplusplus
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#endif /* DSP_H */
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