nrf: PDMIn: Implement
So far, this supports only 16kHz and 16-bit samples with a fixed gain. This is enough to support the basic functionality of e.g., sensing ambient audio levels.
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@ -25,9 +25,15 @@
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*/
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#include "common-hal/audiobusio/PDMIn.h"
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#include "shared-bindings/microcontroller/Pin.h"
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#include "py/runtime.h"
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__attribute__((used))
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NRF_PDM_Type *nrf_pdm = NRF_PDM;
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static uint32_t dummy_buffer[4];
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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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@ -35,30 +41,90 @@ void common_hal_audiobusio_pdmin_construct(audiobusio_pdmin_obj_t* self,
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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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mp_raise_RuntimeError(translate("not implemented"));
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assert_pin_free(clock_pin);
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assert_pin_free(data_pin);
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claim_pin(clock_pin);
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claim_pin(data_pin);
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self->mono = mono;
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self->clock_pin_number = clock_pin->number;
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self->data_pin_number = data_pin->number;
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if (sample_rate != 16000) {
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mp_raise_ValueError(translate("only sample_rate=16000 is supported"));
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}
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if (bit_depth != 16) {
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mp_raise_ValueError(translate("only bit_depth=16 is supported"));
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}
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nrf_pdm->PSEL.CLK = self->clock_pin_number;
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nrf_pdm->PSEL.DIN = self->data_pin_number;
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nrf_pdm->PDMCLKCTRL = PDM_PDMCLKCTRL_FREQ_Default; // For Ratio64
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nrf_pdm->RATIO = PDM_RATIO_RATIO_Ratio64;
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nrf_pdm->GAINL = PDM_GAINL_GAINL_DefaultGain;
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nrf_pdm->GAINR = PDM_GAINR_GAINR_DefaultGain;
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nrf_pdm->ENABLE = 1;
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nrf_pdm->SAMPLE.PTR = (uintptr_t)&dummy_buffer;
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nrf_pdm->SAMPLE.MAXCNT = 1;
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nrf_pdm->TASKS_START = 1;
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}
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bool common_hal_audiobusio_pdmin_deinited(audiobusio_pdmin_obj_t* self) {
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mp_raise_RuntimeError(translate("not implemented"));
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return !self->clock_pin_number;
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}
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void common_hal_audiobusio_pdmin_deinit(audiobusio_pdmin_obj_t* self) {
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mp_raise_RuntimeError(translate("not implemented"));
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nrf_pdm->ENABLE = 0;
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reset_pin_number(self->clock_pin_number);
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self->clock_pin_number = 0;
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reset_pin_number(self->data_pin_number);
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self->data_pin_number = 0;
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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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return 16;
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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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return 16000;
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}
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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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mp_raise_RuntimeError(translate("not implemented"));
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}
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// Note: Adafruit's module has SELECT pulled to GND, which makes the DATA
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// valid when the CLK is low, therefore it must be sampled on the rising edge.
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if (self->mono) {
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nrf_pdm->MODE = PDM_MODE_OPERATION_Stereo | PDM_MODE_EDGE_LeftRising;
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} else {
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nrf_pdm->MODE = PDM_MODE_OPERATION_Mono | PDM_MODE_EDGE_LeftRising;
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}
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void common_hal_audiobusio_pdmin_record_to_file(audiobusio_pdmin_obj_t* self, uint8_t* buffer, uint32_t length) {
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mp_raise_RuntimeError(translate("not implemented"));
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// step 1. Redirect to real buffer
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nrf_pdm->SAMPLE.PTR = (uintptr_t)output_buffer;
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nrf_pdm->SAMPLE.MAXCNT = output_buffer_length;
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// a delay is the safest simple way to ensure that the above requested sample has started
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mp_hal_delay_us(200);
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nrf_pdm->EVENTS_END = 0;
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// step 2. Registers are double buffered, so pre-redirect back to dummy buffer
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nrf_pdm->SAMPLE.PTR = (uintptr_t)&dummy_buffer;
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nrf_pdm->SAMPLE.MAXCNT = 1;
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// Step 3. wait for PDM to end
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while (!nrf_pdm->EVENTS_END) {
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MICROPY_VM_HOOK_LOOP;
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}
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// Step 4. They want unsigned
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for (uint32_t i=0; i<output_buffer_length; i++) {
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output_buffer[i] += 32768;
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}
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if (self->mono) {
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return (output_buffer_length / 2) * 2;
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} else {
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return (output_buffer_length / 4) * 4;
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}
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}
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@ -33,11 +33,8 @@
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typedef struct {
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mp_obj_base_t base;
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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 bytes_per_sample;
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uint8_t bit_depth;
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uint8_t clock_pin_number, data_pin_number;
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bool mono;
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} audiobusio_pdmin_obj_t;
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#endif
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