386 lines
16 KiB
C
386 lines
16 KiB
C
/*
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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) 2017 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/gc.h"
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#include "py/mperrno.h"
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#include "py/runtime.h"
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#include "common-hal/analogio/AnalogOut.h"
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#include "common-hal/audiobusio/PDMIn.h"
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#include "shared-bindings/analogio/AnalogOut.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 "samd21_pins.h"
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#include "shared_dma.h"
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#include "tick.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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void pdmin_reset(void) {
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while (I2S->SYNCBUSY.reg & I2S_SYNCBUSY_ENABLE) {}
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I2S->INTENCLR.reg = I2S_INTENCLR_MASK;
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I2S->INTFLAG.reg = I2S_INTFLAG_MASK;
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I2S->CTRLA.reg &= ~I2S_SYNCBUSY_ENABLE;
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while (I2S->SYNCBUSY.reg & I2S_SYNCBUSY_ENABLE) {}
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I2S->CTRLA.reg = I2S_CTRLA_SWRST;
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}
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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 frequency,
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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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self->clock_pin = clock_pin; // PA10, PA20 -> SCK0, PB11 -> SCK1
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if (clock_pin == &pin_PA10
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#ifdef PIN_PA20
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|| clock_pin == &pin_PA20
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#endif
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) {
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self->clock_unit = 0;
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#ifdef PIN_PB11
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} else if (clock_pin == &pin_PB11) {
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self->clock_unit = 1;
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#endif
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} else {
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mp_raise_ValueError("Invalid clock pin");
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}
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self->data_pin = data_pin; // PA07, PA19 -> SD0, PA08, PB16 -> SD1
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if (data_pin == &pin_PA07 || data_pin == &pin_PA19) {
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self->serializer = 0;
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} else if (data_pin == &pin_PA08
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#ifdef PB16
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|| data_pin == &pin_PB16) {
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#else
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) {
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#endif
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self->serializer = 1;
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} else {
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mp_raise_ValueError("Invalid data pin");
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}
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claim_pin(clock_pin);
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claim_pin(data_pin);
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if (MP_STATE_VM(audiodma_block_counter) == NULL &&
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!allocate_block_counter()) {
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mp_raise_RuntimeError("Unable to allocate audio DMA block counter.");
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}
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if (!(bit_depth == 16 || bit_depth == 8) || !mono || oversample != OVERSAMPLING) {
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mp_raise_NotImplementedError("Only 8 or 16 bit mono with " MP_STRINGIFY(OVERSAMPLING) "x oversampling is supported.");
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}
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// TODO(tannewt): Use the DPLL to get a more precise sampling rate.
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// DFLL -> GCLK (/600 for 8khz, /300 for 16khz and /150 for 32khz) -> DPLL (*(63 + 1)) -> GCLK ( / 10) -> 512khz
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i2s_init(&self->i2s_instance, I2S);
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struct i2s_clock_unit_config config_clock_unit;
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i2s_clock_unit_get_config_defaults(&config_clock_unit);
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config_clock_unit.clock.gclk_src = GCLK_GENERATOR_3;
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config_clock_unit.clock.mck_src = I2S_MASTER_CLOCK_SOURCE_GCLK;
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config_clock_unit.clock.mck_out_enable = false;
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config_clock_unit.clock.sck_src = I2S_SERIAL_CLOCK_SOURCE_MCKDIV;
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uint32_t clock_divisor = (uint32_t) roundf( 8000000.0f / frequency / oversample);
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config_clock_unit.clock.sck_div = clock_divisor;
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float mic_clock_freq = 8000000.0f / clock_divisor;
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self->frequency = mic_clock_freq / oversample;
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if (mic_clock_freq < MIN_MIC_CLOCK || clock_divisor == 0 || clock_divisor > 255) {
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mp_raise_ValueError("sampling frequency out of range");
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}
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config_clock_unit.frame.number_slots = 2;
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config_clock_unit.frame.slot_size = I2S_SLOT_SIZE_16_BIT;
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config_clock_unit.frame.data_delay = I2S_DATA_DELAY_0;
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config_clock_unit.frame.frame_sync.width = I2S_FRAME_SYNC_WIDTH_SLOT;
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config_clock_unit.mck_pin.enable = false;
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config_clock_unit.sck_pin.enable = true;
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config_clock_unit.sck_pin.gpio = self->clock_pin->pin;
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// Mux is always the same.
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config_clock_unit.sck_pin.mux = 6L;
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config_clock_unit.fs_pin.enable = false;
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i2s_clock_unit_set_config(&self->i2s_instance, self->clock_unit, &config_clock_unit);
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struct i2s_serializer_config config_serializer;
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i2s_serializer_get_config_defaults(&config_serializer);
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config_serializer.clock_unit = self->clock_unit;
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config_serializer.mode = I2S_SERIALIZER_PDM2;
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config_serializer.data_size = I2S_DATA_SIZE_32BIT;
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config_serializer.data_pin.gpio = self->data_pin->pin;
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// Mux is always the same.
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config_serializer.data_pin.mux = 6L;
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config_serializer.data_pin.enable = true;
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i2s_serializer_set_config(&self->i2s_instance, self->serializer, &config_serializer);
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i2s_enable(&self->i2s_instance);
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// Run the serializer all the time. This eliminates startup delay for the microphone.
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i2s_clock_unit_enable(&self->i2s_instance, self->clock_unit);
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i2s_serializer_enable(&self->i2s_instance, self->serializer);
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self->bytes_per_sample = oversample >> 3;
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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 self->clock_pin == mp_const_none;
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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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i2s_disable(&self->i2s_instance);
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i2s_serializer_disable(&self->i2s_instance, self->serializer);
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i2s_clock_unit_disable(&self->i2s_instance, self->clock_unit);
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i2s_reset(&self->i2s_instance);
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reset_pin(self->clock_pin->pin);
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reset_pin(self->data_pin->pin);
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self->clock_pin = mp_const_none;
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self->data_pin = mp_const_none;
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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_frequency(audiobusio_pdmin_obj_t* self) {
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return self->frequency;
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}
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static void setup_dma(audiobusio_pdmin_obj_t* self, uint32_t length,
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DmacDescriptor* second_descriptor,
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uint8_t words_per_buffer, uint8_t words_per_sample,
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uint32_t* first_buffer, uint32_t* second_buffer) {
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// Set up the DMA
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struct dma_descriptor_config descriptor_config;
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dma_descriptor_get_config_defaults(&descriptor_config);
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descriptor_config.beat_size = DMA_BEAT_SIZE_WORD;
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descriptor_config.step_selection = DMA_STEPSEL_SRC;
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descriptor_config.source_address = (uint32_t)&I2S->DATA[self->serializer];
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descriptor_config.src_increment_enable = false;
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// Block transfer count is the number of beats per block (aka descriptor).
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// In this case there are two bytes per beat so divide the length by two.
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uint16_t block_transfer_count = words_per_buffer;
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if (length * words_per_sample < words_per_buffer) {
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block_transfer_count = length * words_per_sample;
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}
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descriptor_config.block_transfer_count = block_transfer_count;
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descriptor_config.destination_address = ((uint32_t) first_buffer + sizeof(uint32_t) * block_transfer_count);
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descriptor_config.event_output_selection = DMA_EVENT_OUTPUT_BLOCK;
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descriptor_config.next_descriptor_address = 0;
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if (length * words_per_sample > words_per_buffer) {
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descriptor_config.next_descriptor_address = ((uint32_t)second_descriptor);
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}
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dma_descriptor_create(audio_dma.descriptor, &descriptor_config);
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// Do we need more values than will fit in the first buffer?
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// If so, set up a second buffer chained to be filled after the first buffer.
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if (length * words_per_sample > words_per_buffer) {
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block_transfer_count = words_per_buffer;
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descriptor_config.next_descriptor_address = ((uint32_t)audio_dma.descriptor);
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if (length * words_per_sample < 2 * words_per_buffer) {
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// Length needed is more than one buffer but less than two.
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// Subtract off the size of the first buffer, and what remains is the count we need.
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block_transfer_count = length * words_per_sample - words_per_buffer;
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descriptor_config.next_descriptor_address = 0;
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}
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descriptor_config.block_transfer_count = block_transfer_count;
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descriptor_config.destination_address = ((uint32_t) second_buffer + sizeof(uint32_t) * block_transfer_count);
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dma_descriptor_create(second_descriptor, &descriptor_config);
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}
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switch_audiodma_trigger(I2S_DMAC_ID_RX_0 + self->serializer);
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}
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void start_dma(audiobusio_pdmin_obj_t* self) {
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dma_start_transfer_job(&audio_dma);
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tc_start_counter(MP_STATE_VM(audiodma_block_counter));
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I2S->DATA[1].reg = I2S->DATA[1].reg;
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}
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void stop_dma(audiobusio_pdmin_obj_t* self) {
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// Shutdown the DMA: serializer keeps running.
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tc_stop_counter(MP_STATE_VM(audiodma_block_counter));
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dma_abort_job(&audio_dma);
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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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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_16_TIMES(X) X X X X X X X X X X X X X X X X
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static uint16_t filter_sample(uint32_t pdm_samples[4]) {
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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 < OVERSAMPLING/16; i++) {
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// The sample is 16-bits right channel in the upper two bytes and 16-bits left channel
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// in the lower two bytes.
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// We just ignore the upper bits
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uint32_t pdm_sample = pdm_samples[i];
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REPEAT_16_TIMES( {
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if (pdm_sample & 0x8000) {
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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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// We allocate two buffers on the stack to use for double buffering.
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const uint8_t samples_per_buffer = SAMPLES_PER_BUFFER;
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// For every word we record, we throw away 2 bytes of a phantom second channel.
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const uint8_t words_per_sample = self->bytes_per_sample / 2;
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const uint8_t words_per_buffer = samples_per_buffer * words_per_sample;
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uint32_t first_buffer[words_per_buffer];
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uint32_t second_buffer[words_per_buffer];
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COMPILER_ALIGNED(16) DmacDescriptor second_descriptor;
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setup_dma(self, output_buffer_length, &second_descriptor, words_per_buffer,
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words_per_sample, first_buffer, second_buffer);
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start_dma(self);
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// Record
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uint32_t buffers_processed = 0;
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uint32_t values_output = 0;
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uint32_t remaining_samples_needed = output_buffer_length;
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while (values_output < output_buffer_length) {
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// Wait for the next buffer to fill
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uint32_t block_counter;
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while ((block_counter = tc_get_count_value(MP_STATE_VM(audiodma_block_counter))) == buffers_processed) {
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#ifdef MICROPY_VM_HOOK_LOOP
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MICROPY_VM_HOOK_LOOP
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#endif
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}
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if (block_counter != (buffers_processed + 1)) {
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// Looks like we aren't keeping up. We shouldn't skip a buffer.
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break;
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}
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// The mic is running all the time, so we don't need to wait the usual 10msec or 100msec
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// for it to start up.
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// Flip back and forth between processing the first and second buffers.
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uint32_t *buffer = first_buffer;
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DmacDescriptor* descriptor = audio_dma.descriptor;
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if (buffers_processed % 2 == 1) {
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buffer = second_buffer;
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descriptor = &second_descriptor;
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}
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// Decimate and filter the buffer that was just filled.
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uint32_t samples_gathered = descriptor->BTCNT.reg / words_per_sample;
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// Don't run off the end of output buffer. Process only as many as needed.
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uint32_t samples_to_process = min(remaining_samples_needed, samples_gathered);
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for (uint32_t i = 0; i < samples_to_process; i++) {
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// Call filter_sample just one place so it can be inlined.
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uint16_t value = filter_sample(buffer + i * words_per_sample);
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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)[values_output] = value >> 8;
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} else {
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output_buffer[values_output] = value;
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}
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values_output++;
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}
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buffers_processed++;
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// Compute how many more samples we need, and if the last buffer is the last
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// set of samples needed, adjust the DMA count to only fetch as necessary.
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remaining_samples_needed = output_buffer_length - values_output;
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if (remaining_samples_needed <= samples_per_buffer*2 &&
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remaining_samples_needed > samples_per_buffer) {
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// Adjust the DMA settings for the current buffer, which will be processed
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// after the other buffer, which is now receiving samples via DMA.
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// We don't adjust the DMA in progress, but the one after that.
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// Timeline:
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// 1. current buffer (already processed)
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// 2. alternate buffer (DMA in progress)
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// 3. current buffer (last set of samples needed)
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// Set up to receive the last set of samples (don't include the alternate buffer, now in use).
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uint32_t samples_needed_for_last_buffer = remaining_samples_needed - samples_per_buffer;
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descriptor->BTCNT.reg = samples_needed_for_last_buffer * words_per_sample;
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descriptor->DSTADDR.reg = ((uint32_t) buffer)
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+ samples_needed_for_last_buffer * words_per_sample * sizeof(buffer[0]);
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// Break chain to alternate buffer.
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descriptor->DESCADDR.reg = 0;
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}
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}
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stop_dma(self);
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return values_output;
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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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}
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