forked from DiSlord/NanoVNA-D
-
Notifications
You must be signed in to change notification settings - Fork 0
/
dsp.c
263 lines (253 loc) · 10.2 KB
/
dsp.c
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
184
185
186
187
188
189
190
191
192
193
194
195
196
197
198
199
200
201
202
203
204
205
206
207
208
209
210
211
212
213
214
215
216
217
218
219
220
221
222
223
224
225
226
227
228
229
230
231
232
233
234
235
236
237
238
239
240
241
242
243
244
245
246
247
248
249
250
251
252
253
254
255
256
257
258
259
260
261
262
263
/*
* Copyright (c) 2019-2020, Dmitry (DiSlord) dislordlive@gmail.com
* Based on 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 "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[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 = vna_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;
}