NanoVNA/dsp.c
DiSlord 237a329226 Add sin_cos table for 6 or 12kHz offset for dsp
Add support direct clock for AIC3204 from si5351
Now possible made calibration and not reset old calibration data, just made another calibration (not need reset or disable correction).
  Open - Short calibration depend from self, need recalibrate it together.
  Load calibration possible made alone
  Isoln, Thru also depend from self, need recalibrate it together.
2020-06-04 20:26:06 +03:00

246 lines
8.9 KiB
C

/*
* Copyright (c) 2014-2015, TAKAHASHI Tomohiro (TTRFTECH) edy555@gmail.com
* 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 <arm_math.h>
#include "nanovna.h"
#ifdef ENABLED_DUMP
int16_t samp_buf[SAMPLE_LEN];
int16_t ref_buf[SAMPLE_LEN];
#endif
#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 = 2 * VNA_PI * offset / audio_freq;
float v = step/2;
for (int i=0; i<AUDIO_SAMPLES_COUNT; i++){
sincos_tbl[i][0] = sin(v)*32768.0 + 0.5;
sincos_tbl[i][1] = cos(v)*32768.0 + 0.5;
v+=step;
}
}
#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}
};
#else
#error "Need check/rebuild sin cos table for DAC"
#endif
#if 1
// 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(int16_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];
#ifdef ENABLED_DUMP
ref_buf[i] = ref;
samp_buf[i] = smp;
#endif
int16_t sin = ((int16_t *)sincos_tbl)[i+0];
int16_t cos = ((int16_t *)sincos_tbl)[i+1];
samp_s+= (smp * sin)>>4;
samp_c+= (smp * cos)>>4;
ref_s += (ref * sin)>>4;
ref_c += (ref * cos)>>4;
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;
acc_t acc_samp_s;
acc_t acc_samp_c;
acc_t acc_ref_s;
acc_t acc_ref_c;
// Cortex M4 DSP instruction use
#include "dsp.h"
void
dsp_process(int16_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;
i=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[2])
{
#if 1
// 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;
//rr = sqrtf(rr) * 1e8;
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 + 1.0;
gamma[0] = (sc_rc + ss_rc*rs_rc) / rr;
gamma[1] = (ss_rc - sc_rc*rs_rc) / rr;
#endif
#elif 0
gamma[0] = acc_samp_s;
gamma[1] = acc_samp_c;
#else
gamma[0] = acc_ref_s;
gamma[1] = acc_ref_c;
#endif
}
void
fetch_amplitude(float gamma[2])
{
gamma[0] = acc_samp_s * 1e-9;
gamma[1] = acc_samp_c * 1e-9;
}
void
fetch_amplitude_ref(float gamma[2])
{
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;
}