2021-02-23 18:50:00 -05:00
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/*
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* This file is part of the MicroPython project, http://micropython.org/
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*
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* The MIT License (MIT)
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*
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* Copyright (c) 2021 Scott Shawcroft for Adafruit Industries
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*
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* Permission is hereby granted, free of charge, to any person obtaining a copy
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* of this software and associated documentation files (the "Software"), to deal
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* in the Software without restriction, including without limitation the rights
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* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
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* copies of the Software, and to permit persons to whom the Software is
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* furnished to do so, subject to the following conditions:
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*
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* The above copyright notice and this permission notice shall be included in
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* all copies or substantial portions of the Software.
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*
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* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
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* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
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* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
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* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
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* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
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* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
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* THE SOFTWARE.
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*/
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#include <stdint.h>
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#include <string.h>
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#include <math.h>
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#include "py/mperrno.h"
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#include "py/runtime.h"
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#include "shared-bindings/audiobusio/PDMIn.h"
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#include "shared-bindings/microcontroller/Pin.h"
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#include "supervisor/shared/translate.h"
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#include "audio_dma.h"
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#define OVERSAMPLING 64
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#define SAMPLES_PER_BUFFER 32
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// MEMS microphones must be clocked at at least 1MHz.
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#define MIN_MIC_CLOCK 1000000
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const uint16_t pdmin[] = {
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// in pins 1 side 0b1
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0x5001,
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// push iffull side 0b0
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0x8040
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};
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// Caller validates that pins are free.
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void common_hal_audiobusio_pdmin_construct(audiobusio_pdmin_obj_t *self,
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const mcu_pin_obj_t *clock_pin,
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const mcu_pin_obj_t *data_pin,
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uint32_t sample_rate,
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uint8_t bit_depth,
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bool mono,
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uint8_t oversample) {
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if (!(bit_depth == 16 || bit_depth == 8) || !mono || oversample != OVERSAMPLING) {
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mp_raise_NotImplementedError(translate("Only 8 or 16 bit mono with " MP_STRINGIFY(OVERSAMPLING) "x oversampling is supported."));
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}
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// Use the state machine to manage pins.
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common_hal_rp2pio_statemachine_construct(&self->state_machine,
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pdmin, sizeof(pdmin) / sizeof(pdmin[0]),
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44100 * 32 * 2, // Clock at 44.1 khz to warm the DAC up.
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NULL, 0,
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NULL, 1, 0, 0xffffffff, // out pin
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data_pin, 1, // in pins
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0, 0, // in pulls
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NULL, 0, 0, 0x1f, // set pins
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clock_pin, 1, 0, 0x1f, // sideset pins
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false, // No sideset enable
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NULL, // jump pin
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0, // wait gpio pins
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true, // exclusive pin use
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false, 32, false, // out settings
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false, // Wait for txstall
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false, 32, true, // in settings
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false); // Not user-interruptible.
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uint32_t actual_frequency = common_hal_rp2pio_statemachine_get_frequency(&self->state_machine);
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if (actual_frequency < MIN_MIC_CLOCK) {
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mp_raise_ValueError(translate("sampling rate out of range"));
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}
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self->sample_rate = actual_frequency / oversample;
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self->bit_depth = bit_depth;
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}
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bool common_hal_audiobusio_pdmin_deinited(audiobusio_pdmin_obj_t *self) {
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return common_hal_rp2pio_statemachine_deinited(&self->state_machine);
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}
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void common_hal_audiobusio_pdmin_deinit(audiobusio_pdmin_obj_t *self) {
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if (common_hal_audiobusio_pdmin_deinited(self)) {
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return;
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}
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return common_hal_rp2pio_statemachine_deinit(&self->state_machine);
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}
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uint8_t common_hal_audiobusio_pdmin_get_bit_depth(audiobusio_pdmin_obj_t *self) {
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return self->bit_depth;
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}
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uint32_t common_hal_audiobusio_pdmin_get_sample_rate(audiobusio_pdmin_obj_t *self) {
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return self->sample_rate;
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}
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// a windowed sinc filter for 44 khz, 64 samples
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//
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// This filter is good enough to use for lower sample rates as
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// well. It does not increase the noise enough to be a problem.
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//
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// In the long run we could use a fast filter like this to do the
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// decimation and initial filtering in real time, filtering to a
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// higher sample rate than specified. Then after the audio is
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// recorded, a more expensive filter non-real-time filter could be
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// used to down-sample and low-pass.
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const uint16_t sinc_filter[OVERSAMPLING] = {
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0, 2, 9, 21, 39, 63, 94, 132,
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179, 236, 302, 379, 467, 565, 674, 792,
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920, 1055, 1196, 1341, 1487, 1633, 1776, 1913,
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2042, 2159, 2263, 2352, 2422, 2474, 2506, 2516,
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2506, 2474, 2422, 2352, 2263, 2159, 2042, 1913,
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1776, 1633, 1487, 1341, 1196, 1055, 920, 792,
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674, 565, 467, 379, 302, 236, 179, 132,
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94, 63, 39, 21, 9, 2, 0, 0
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};
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#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)
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static uint16_t filter_sample(uint32_t pdm_samples[2]) {
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uint16_t running_sum = 0;
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const uint16_t *filter_ptr = sinc_filter;
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for (uint8_t i = 0; i < 2; i++) {
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uint32_t pdm_sample = pdm_samples[i];
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REPEAT_32_TIMES({
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if (pdm_sample & 0x1) {
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running_sum += *filter_ptr;
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}
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filter_ptr++;
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pdm_sample >>= 1;
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}
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);
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}
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return running_sum;
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}
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// output_buffer may be a byte buffer or a halfword buffer.
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// output_buffer_length is the number of slots, not the number of bytes.
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uint32_t common_hal_audiobusio_pdmin_record_to_buffer(audiobusio_pdmin_obj_t *self,
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uint16_t *output_buffer, uint32_t output_buffer_length) {
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uint32_t samples[2];
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size_t output_count = 0;
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common_hal_rp2pio_statemachine_clear_rxfifo(&self->state_machine);
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// Do one read to get the mic going and throw it away.
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common_hal_rp2pio_statemachine_readinto(&self->state_machine, (uint8_t *)samples, 2 * sizeof(uint32_t), sizeof(uint32_t));
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while (output_count < output_buffer_length && !common_hal_rp2pio_statemachine_get_rxstall(&self->state_machine)) {
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common_hal_rp2pio_statemachine_readinto(&self->state_machine, (uint8_t *)samples, 2 * sizeof(uint32_t), sizeof(uint32_t));
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// Call filter_sample just one place so it can be inlined.
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uint16_t value = filter_sample(samples);
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if (self->bit_depth == 8) {
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// Truncate to 8 bits.
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((uint8_t *)output_buffer)[output_count] = value >> 8;
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} else {
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output_buffer[output_count] = value;
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}
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output_count++;
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}
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return output_count;
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}
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