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dsp.c
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dsp.c
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/*
* Copyright (c) 2019-2020, Dmitry (DiSlord) [email protected]
* Based on TAKAHASHI Tomohiro (TTRFTECH) [email protected]
* All rights reserved.
*
* This 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, or (at your option)
* any later version.
*
* The software 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 GNU Radio; see the file COPYING. If not, write to
* the Free Software Foundation, Inc., 51 Franklin Street,
* Boston, MA 02110-1301, USA.
*/
#include "nanovna.h"
#ifdef USE_VARIABLE_OFFSET
static int16_t sincos_tbl[AUDIO_SAMPLES_COUNT][2];
void generate_DSP_Table(int offset){
float audio_freq = AUDIO_ADC_FREQ;
// N = offset * AUDIO_SAMPLES_COUNT / audio_freq; should be integer
// AUDIO_SAMPLES_COUNT = N * audio_freq / offset; N - minimum integer value for get integer AUDIO_SAMPLES_COUNT
// Bandwidth on one step = audio_freq / AUDIO_SAMPLES_COUNT
float step = offset / audio_freq;
float w = step/2;
for (int i=0; i<AUDIO_SAMPLES_COUNT; i++){
float s, c;
vna_sincosf(w, &s, &c);
sincos_tbl[i][0] = s*32700.0f;
sincos_tbl[i][1] = c*32700.0f;
w+=step;
}
}
#elif FREQUENCY_OFFSET==7000*(AUDIO_ADC_FREQ/AUDIO_SAMPLES_COUNT/1000)
// static Table for 28kHz IF and 192kHz ADC (or 7kHz IF and 48kHz ADC) audio ADC
static const int16_t sincos_tbl[48][2] = {
{ 14493, 29389}, { 32138, 6393}, { 24636,-21605}, { -2143,-32698},
{-27246,-18205}, {-31029, 10533}, {-10533, 31029}, { 18205, 27246},
{ 32698, 2143}, { 21605,-24636}, { -6393,-32138}, {-29389,-14493},
{-29389, 14493}, { -6393, 32138}, { 21605, 24636}, { 32698, -2143},
{ 18205,-27246}, {-10533,-31029}, {-31029,-10533}, {-27246, 18205},
{ -2143, 32698}, { 24636, 21605}, { 32138, -6393}, { 14493,-29389},
{-14493,-29389}, {-32138, -6393}, {-24636, 21605}, { 2143, 32698},
{ 27246, 18205}, { 31029,-10533}, { 10533,-31029}, {-18205,-27246},
{-32698, -2143}, {-21605, 24636}, { 6393, 32138}, { 29389, 14493},
{ 29389,-14493}, { 6393,-32138}, {-21605,-24636}, {-32698, 2143},
{-18205, 27246}, { 10533, 31029}, { 31029, 10533}, { 27246,-18205},
{ 2143,-32698}, {-24636,-21605}, {-32138, 6393}, {-14493, 29389}
};
#elif FREQUENCY_OFFSET==6000*(AUDIO_ADC_FREQ/AUDIO_SAMPLES_COUNT/1000)
// static Table for 12kHz IF and 96kHz ADC (or 6kHz IF and 48kHz ADC) audio ADC
static const int16_t sincos_tbl[48][2] = {
{ 6393, 32138}, { 27246, 18205}, { 32138,-6393}, { 18205,-27246},
{-6393,-32138}, {-27246,-18205}, {-32138, 6393}, {-18205, 27246},
{ 6393, 32138}, { 27246, 18205}, { 32138,-6393}, { 18205,-27246},
{-6393,-32138}, {-27246,-18205}, {-32138, 6393}, {-18205, 27246},
{ 6393, 32138}, { 27246, 18205}, { 32138,-6393}, { 18205,-27246},
{-6393,-32138}, {-27246,-18205}, {-32138, 6393}, {-18205, 27246},
{ 6393, 32138}, { 27246, 18205}, { 32138,-6393}, { 18205,-27246},
{-6393,-32138}, {-27246,-18205}, {-32138, 6393}, {-18205, 27246},
{ 6393, 32138}, { 27246, 18205}, { 32138,-6393}, { 18205,-27246},
{-6393,-32138}, {-27246,-18205}, {-32138, 6393}, {-18205, 27246},
{ 6393, 32138}, { 27246, 18205}, { 32138,-6393}, { 18205,-27246},
{-6393,-32138}, {-27246,-18205}, {-32138, 6393}, {-18205, 27246}
};
#elif FREQUENCY_OFFSET==5000*(AUDIO_ADC_FREQ/AUDIO_SAMPLES_COUNT/1000)
// static Table for 10kHz IF and 96kHz ADC (or 5kHz IF and 48kHz ADC) audio ADC
static const int16_t sincos_tbl[48][2] = {
{ 10533, 31029 }, { 27246, 18205 }, { 32698, -2143 }, { 24636, -21605 },
{ 6393, -32138 }, {-14493, -29389 }, {-29389, -14493 }, {-32138, 6393 },
{-21605, 24636 }, { -2143, 32698 }, { 18205, 27246 }, { 31029, 10533 },
{ 31029, -10533 }, { 18205, -27246 }, { -2143, -32698 }, {-21605, -24636 },
{-32138, -6393 }, {-29389, 14493 }, {-14493, 29389 }, { 6393, 32138 },
{ 24636, 21605 }, { 32698, 2143 }, { 27246, -18205 }, { 10533, -31029 },
{-10533, -31029 }, {-27246, -18205 }, {-32698, 2143 }, {-24636, 21605 },
{ -6393, 32138 }, { 14493, 29389 }, { 29389, 14493 }, { 32138, -6393 },
{ 21605, -24636 }, { 2143, -32698 }, {-18205, -27246 }, {-31029, -10533 },
{-31029, 10533 }, {-18205, 27246 }, { 2143, 32698 }, { 21605, 24636 },
{ 32138, 6393 }, { 29389, -14493 }, { 14493, -29389 }, { -6393, -32138 },
{-24636, -21605 }, {-32698, -2143 }, {-27246, 18205 }, {-10533, 31029 }
};
#elif FREQUENCY_OFFSET==4000*(AUDIO_ADC_FREQ/AUDIO_SAMPLES_COUNT/1000)
// static Table for 8kHz IF and 96kHz audio ADC (or 4kHz IF and 48kHz ADC) audio ADC
static const int16_t sincos_tbl[48][2] = {
{ 4277, 32488}, { 19948, 25997}, { 30274, 12540}, { 32488, -4277},
{ 25997,-19948}, { 12540,-30274}, { -4277,-32488}, {-19948,-25997},
{-30274,-12540}, {-32488, 4277}, {-25997, 19948}, {-12540, 30274},
{ 4277, 32488}, { 19948, 25997}, { 30274, 12540}, { 32488, -4277},
{ 25997,-19948}, { 12540,-30274}, { -4277,-32488}, {-19948,-25997},
{-30274,-12540}, {-32488, 4277}, {-25997, 19948}, {-12540, 30274},
{ 4277, 32488}, { 19948, 25997}, { 30274, 12540}, { 32488, -4277},
{ 25997,-19948}, { 12540,-30274}, { -4277,-32488}, {-19948,-25997},
{-30274,-12540}, {-32488, 4277}, {-25997, 19948}, {-12540, 30274},
{ 4277, 32488}, { 19948, 25997}, { 30274, 12540}, { 32488, -4277},
{ 25997,-19948}, { 12540,-30274}, { -4277,-32488}, {-19948,-25997},
{-30274,-12540}, {-32488, 4277}, {-25997, 19948}, {-12540, 30274}
};
#elif FREQUENCY_OFFSET==3000*(AUDIO_ADC_FREQ/AUDIO_SAMPLES_COUNT/1000)
// static Table for 6kHz IF and 96kHz audio ADC (or 3kHz IF and 48kHz ADC) audio ADC
static const int16_t sincos_tbl[48][2] = {
{ 3212, 32610}, { 15447, 28899}, { 25330, 20788}, { 31357, 9512},
{ 32610, -3212}, { 28899,-15447}, { 20788,-25330}, { 9512,-31357},
{ -3212,-32610}, {-15447,-28899}, {-25330,-20788}, {-31357, -9512},
{-32610, 3212}, {-28899, 15447}, {-20788, 25330}, { -9512, 31357},
{ 3212, 32610}, { 15447, 28899}, { 25330, 20788}, { 31357, 9512},
{ 32610, -3212}, { 28899,-15447}, { 20788,-25330}, { 9512,-31357},
{ -3212,-32610}, {-15447,-28899}, {-25330,-20788}, {-31357, -9512},
{-32610, 3212}, {-28899, 15447}, {-20788, 25330}, { -9512, 31357},
{ 3212, 32610}, { 15447, 28899}, { 25330, 20788}, { 31357, 9512},
{ 32610, -3212}, { 28899,-15447}, { 20788,-25330}, { 9512,-31357},
{ -3212,-32610}, {-15447,-28899}, {-25330,-20788}, {-31357, -9512},
{-32610, 3212}, {-28899, 15447}, {-20788, 25330}, { -9512, 31357}
};
#elif FREQUENCY_OFFSET==2000*(AUDIO_ADC_FREQ/AUDIO_SAMPLES_COUNT/1000)
// static Table
static const int16_t sincos_tbl[48][2] = {
#error "Need check/rebuild sin cos table for DAC"
};
#elif FREQUENCY_OFFSET==1000*(AUDIO_ADC_FREQ/AUDIO_SAMPLES_COUNT/1000)
// static Table
static const int16_t sincos_tbl[48][2] = {
#error "Need check/rebuild sin cos table for DAC"
};
#else
#error "Need check/rebuild sin cos table for DAC"
#endif
#ifndef __USE_DSP__
// Define DSP accumulator value type
typedef float acc_t;
typedef float measure_t;
acc_t acc_samp_s;
acc_t acc_samp_c;
acc_t acc_ref_s;
acc_t acc_ref_c;
void
dsp_process(audio_sample_t *capture, size_t length)
{
int32_t samp_s = 0;
int32_t samp_c = 0;
int32_t ref_s = 0;
int32_t ref_c = 0;
uint32_t i = 0;
do{
int16_t ref = capture[i+0];
int16_t smp = capture[i+1];
int32_t sin = ((int16_t *)sincos_tbl)[i+0];
int32_t cos = ((int16_t *)sincos_tbl)[i+1];
samp_s+= (smp * sin)/16;
samp_c+= (smp * cos)/16;
ref_s += (ref * sin)/16;
ref_c += (ref * cos)/16;
i+=2;
}while (i < length);
acc_samp_s += samp_s;
acc_samp_c += samp_c;
acc_ref_s += ref_s;
acc_ref_c += ref_c;
}
#else
// Define DSP accumulator value type
typedef int64_t acc_t;
typedef float measure_t;
static acc_t acc_samp_s;
static acc_t acc_samp_c;
static acc_t acc_ref_s;
static acc_t acc_ref_c;
// Cortex M4 DSP instruction use
#include "dsp.h"
void
dsp_process(audio_sample_t *capture, size_t length)
{
uint32_t i = 0;
// int64_t samp_s = 0;
// int64_t samp_c = 0;
// int64_t ref_s = 0;
// int64_t ref_c = 0;
do{
int32_t sc = ((int32_t *)sincos_tbl)[i];
int32_t sr = ((int32_t *)capture)[i];
// int32_t acc DSP functions, but int32 can overflow
// samp_s = __smlatb(sr, sc, samp_s); // samp_s+= smp * sin
// samp_c = __smlatt(sr, sc, samp_c); // samp_c+= smp * cos
// ref_s = __smlabb(sr, sc, ref_s); // ref_s+= ref * sin
// ref_c = __smlabt(sr, sc, ref_c); // ref_s+= ref * cos
// int64_t acc DSP functions
acc_samp_s= __smlaltb(acc_samp_s, sr, sc ); // samp_s+= smp * sin
acc_samp_c= __smlaltt(acc_samp_c, sr, sc ); // samp_c+= smp * cos
acc_ref_s = __smlalbb( acc_ref_s, sr, sc ); // ref_s+= ref * sin
acc_ref_c = __smlalbt( acc_ref_c, sr, sc ); // ref_s+= ref * cos
i++;
} while (i < length/2);
// Accumulate result, for faster calc and prevent overflow reduce size to int32_t
// acc_samp_s+= (int32_t)(samp_s>>4);
// acc_samp_c+= (int32_t)(samp_c>>4);
// acc_ref_s += (int32_t)( ref_s>>4);
// acc_ref_c += (int32_t)( ref_c>>4);
}
#endif
void
calculate_gamma(float *gamma)
{
// calculate reflection coeff. by samp divide by ref
#if 0
measure_t rs = acc_ref_s;
measure_t rc = acc_ref_c;
measure_t rr = rs * rs + rc * rc;
measure_t ss = acc_samp_s;
measure_t sc = acc_samp_c;
gamma[0] = (sc * rc + ss * rs) / rr;
gamma[1] = (ss * rc - sc * rs) / rr;
#else
measure_t rs_rc = (measure_t) acc_ref_s / acc_ref_c;
measure_t sc_rc = (measure_t)acc_samp_c / acc_ref_c;
measure_t ss_rc = (measure_t)acc_samp_s / acc_ref_c;
measure_t rr = rs_rc * rs_rc + 1.0f;
gamma[0] = (sc_rc + ss_rc * rs_rc) / rr;
gamma[1] = (ss_rc - sc_rc * rs_rc) / rr;
#endif
}
void
fetch_amplitude(float *gamma)
{
gamma[0] = acc_samp_s * 1e-9;
gamma[1] = acc_samp_c * 1e-9;
}
void
fetch_amplitude_ref(float *gamma)
{
gamma[0] = acc_ref_s * 1e-9;
gamma[1] = acc_ref_c * 1e-9;
}
void
reset_dsp_accumerator(void)
{
acc_ref_s = 0;
acc_ref_c = 0;
acc_samp_s = 0;
acc_samp_c = 0;
}