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
|
/*
* This file is part of the MicroPython project, http://micropython.org/
*
* The MIT License (MIT)
*
* Copyright (c) 2021 Scott Shawcroft for Adafruit Industries
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
#include <stdint.h>
#include <string.h>
#include <math.h>
#include "py/mperrno.h"
#include "py/runtime.h"
#include "shared-bindings/audiobusio/PDMIn.h"
#include "shared-bindings/microcontroller/Pin.h"
#include "supervisor/shared/translate.h"
#include "audio_dma.h"
#define OVERSAMPLING 64
#define SAMPLES_PER_BUFFER 32
// MEMS microphones must be clocked at at least 1MHz.
#define MIN_MIC_CLOCK 1000000
const uint16_t pdmin[] = {
// in pins 1 side 0b1
0x5001,
// push iffull side 0b0
0x8040
};
// Caller validates that pins are free.
void common_hal_audiobusio_pdmin_construct(audiobusio_pdmin_obj_t *self,
const mcu_pin_obj_t *clock_pin,
const mcu_pin_obj_t *data_pin,
uint32_t sample_rate,
uint8_t bit_depth,
bool mono,
uint8_t oversample) {
if (!(bit_depth == 16 || bit_depth == 8) || !mono || oversample != OVERSAMPLING) {
mp_raise_NotImplementedError(translate("Only 8 or 16 bit mono with " MP_STRINGIFY(OVERSAMPLING) "x oversampling is supported."));
}
// Use the state machine to manage pins.
common_hal_rp2pio_statemachine_construct(&self->state_machine,
pdmin, sizeof(pdmin) / sizeof(pdmin[0]),
44100 * 32 * 2, // Clock at 44.1 khz to warm the DAC up.
NULL, 0,
NULL, 1, 0, 0xffffffff, // out pin
data_pin, 1, // in pins
0, 0, // in pulls
NULL, 0, 0, 0x1f, // set pins
clock_pin, 1, 0, 0x1f, // sideset pins
true, // exclusive pin use
false, 32, false, // out settings
false, // Wait for txstall
false, 32, true); // in settings
uint32_t actual_frequency = common_hal_rp2pio_statemachine_get_frequency(&self->state_machine);
if (actual_frequency < MIN_MIC_CLOCK) {
mp_raise_ValueError(translate("sampling rate out of range"));
}
self->sample_rate = actual_frequency / oversample;
self->bit_depth = bit_depth;
}
bool common_hal_audiobusio_pdmin_deinited(audiobusio_pdmin_obj_t *self) {
return common_hal_rp2pio_statemachine_deinited(&self->state_machine);
}
void common_hal_audiobusio_pdmin_deinit(audiobusio_pdmin_obj_t *self) {
if (common_hal_audiobusio_pdmin_deinited(self)) {
return;
}
return common_hal_rp2pio_statemachine_deinit(&self->state_machine);
}
uint8_t common_hal_audiobusio_pdmin_get_bit_depth(audiobusio_pdmin_obj_t *self) {
return self->bit_depth;
}
uint32_t common_hal_audiobusio_pdmin_get_sample_rate(audiobusio_pdmin_obj_t *self) {
return self->sample_rate;
}
// a windowed sinc filter for 44 khz, 64 samples
//
// This filter is good enough to use for lower sample rates as
// well. It does not increase the noise enough to be a problem.
//
// In the long run we could use a fast filter like this to do the
// decimation and initial filtering in real time, filtering to a
// higher sample rate than specified. Then after the audio is
// recorded, a more expensive filter non-real-time filter could be
// used to down-sample and low-pass.
const uint16_t sinc_filter[OVERSAMPLING] = {
0, 2, 9, 21, 39, 63, 94, 132,
179, 236, 302, 379, 467, 565, 674, 792,
920, 1055, 1196, 1341, 1487, 1633, 1776, 1913,
2042, 2159, 2263, 2352, 2422, 2474, 2506, 2516,
2506, 2474, 2422, 2352, 2263, 2159, 2042, 1913,
1776, 1633, 1487, 1341, 1196, 1055, 920, 792,
674, 565, 467, 379, 302, 236, 179, 132,
94, 63, 39, 21, 9, 2, 0, 0
};
#define REPEAT_32_TIMES(X) do { X X X X X X X X X X X X X X X X X X X X X X X X X X X X X X X X } while (0)
static uint16_t filter_sample(uint32_t pdm_samples[2]) {
uint16_t running_sum = 0;
const uint16_t *filter_ptr = sinc_filter;
for (uint8_t i = 0; i < 2; i++) {
uint32_t pdm_sample = pdm_samples[i];
REPEAT_32_TIMES({
if (pdm_sample & 0x1) {
running_sum += *filter_ptr;
}
filter_ptr++;
pdm_sample >>= 1;
}
);
}
return running_sum;
}
// output_buffer may be a byte buffer or a halfword buffer.
// output_buffer_length is the number of slots, not the number of bytes.
uint32_t common_hal_audiobusio_pdmin_record_to_buffer(audiobusio_pdmin_obj_t *self,
uint16_t *output_buffer, uint32_t output_buffer_length) {
uint32_t samples[2];
size_t output_count = 0;
common_hal_rp2pio_statemachine_clear_rxfifo(&self->state_machine);
// Do one read to get the mic going and throw it away.
common_hal_rp2pio_statemachine_readinto(&self->state_machine, (uint8_t *)samples, 2 * sizeof(uint32_t), sizeof(uint32_t));
while (output_count < output_buffer_length && !common_hal_rp2pio_statemachine_get_rxstall(&self->state_machine)) {
common_hal_rp2pio_statemachine_readinto(&self->state_machine, (uint8_t *)samples, 2 * sizeof(uint32_t), sizeof(uint32_t));
// Call filter_sample just one place so it can be inlined.
uint16_t value = filter_sample(samples);
if (self->bit_depth == 8) {
// Truncate to 8 bits.
((uint8_t *)output_buffer)[output_count] = value >> 8;
} else {
output_buffer[output_count] = value;
}
output_count++;
}
return output_count;
}
void common_hal_audiobusio_pdmin_record_to_file(audiobusio_pdmin_obj_t *self, uint8_t *buffer, uint32_t length) {
}
|
