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circuitpython/atmel-samd/common-hal/audiobusio/PDMIn.c

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/*
* This file is part of the MicroPython project, http://micropython.org/
*
* The MIT License (MIT)
*
* Copyright (c) 2017 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/gc.h"
#include "py/mperrno.h"
#include "py/runtime.h"
#include "common-hal/analogio/AnalogOut.h"
#include "common-hal/audiobusio/PDMIn.h"
#include "shared-bindings/analogio/AnalogOut.h"
#include "shared-bindings/audiobusio/PDMIn.h"
#include "shared-bindings/microcontroller/Pin.h"
#include "asf/sam0/drivers/port/port.h"
#include "samd21_pins.h"
#include "shared_dma.h"
#include "tick.h"
#define OVERSAMPLING 64
#define SAMPLES_PER_BUFFER 32
// MEMS microphones must be clocked at at least 1MHz.
#define MIN_MIC_CLOCK 1000000
void pdmin_reset(void) {
while (I2S->SYNCBUSY.reg & I2S_SYNCBUSY_ENABLE) {}
I2S->INTENCLR.reg = I2S_INTENCLR_MASK;
I2S->INTFLAG.reg = I2S_INTFLAG_MASK;
I2S->CTRLA.reg &= ~I2S_SYNCBUSY_ENABLE;
while (I2S->SYNCBUSY.reg & I2S_SYNCBUSY_ENABLE) {}
I2S->CTRLA.reg = I2S_CTRLA_SWRST;
}
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 frequency,
uint8_t bit_depth,
bool mono,
uint8_t oversample) {
self->clock_pin = clock_pin; // PA10, PA20 -> SCK0, PB11 -> SCK1
if (clock_pin == &pin_PA10
#ifdef PIN_PA20
|| clock_pin == &pin_PA20
#endif
) {
self->clock_unit = 0;
#ifdef PIN_PB11
} else if (clock_pin == &pin_PB11) {
self->clock_unit = 1;
#endif
} else {
mp_raise_ValueError("Invalid clock pin");
}
self->data_pin = data_pin; // PA07, PA19 -> SD0, PA08, PB16 -> SD1
if (data_pin == &pin_PA07 || data_pin == &pin_PA19) {
self->serializer = 0;
} else if (data_pin == &pin_PA08
#ifdef PB16
|| data_pin == &pin_PB16) {
#else
) {
#endif
self->serializer = 1;
} else {
mp_raise_ValueError("Invalid data pin");
}
claim_pin(clock_pin);
claim_pin(data_pin);
if (MP_STATE_VM(audiodma_block_counter) == NULL &&
!allocate_block_counter()) {
mp_raise_RuntimeError("Unable to allocate audio DMA block counter.");
}
if (!(bit_depth == 16 || bit_depth == 8) || !mono || oversample != OVERSAMPLING) {
mp_raise_NotImplementedError("Only 8 or 16 bit mono with " MP_STRINGIFY(OVERSAMPLING) "x oversampling is supported.");
}
// TODO(tannewt): Use the DPLL to get a more precise sampling rate.
// DFLL -> GCLK (/600 for 8khz, /300 for 16khz and /150 for 32khz) -> DPLL (*(63 + 1)) -> GCLK ( / 10) -> 512khz
i2s_init(&self->i2s_instance, I2S);
struct i2s_clock_unit_config config_clock_unit;
i2s_clock_unit_get_config_defaults(&config_clock_unit);
config_clock_unit.clock.gclk_src = GCLK_GENERATOR_3;
config_clock_unit.clock.mck_src = I2S_MASTER_CLOCK_SOURCE_GCLK;
config_clock_unit.clock.mck_out_enable = false;
config_clock_unit.clock.sck_src = I2S_SERIAL_CLOCK_SOURCE_MCKDIV;
uint32_t clock_divisor = (uint32_t) roundf( 8000000.0f / frequency / oversample);
config_clock_unit.clock.sck_div = clock_divisor;
float mic_clock_freq = 8000000.0f / clock_divisor;
self->frequency = mic_clock_freq / oversample;
if (mic_clock_freq < MIN_MIC_CLOCK || clock_divisor == 0 || clock_divisor > 255) {
mp_raise_ValueError("sampling frequency out of range");
}
config_clock_unit.frame.number_slots = 2;
config_clock_unit.frame.slot_size = I2S_SLOT_SIZE_16_BIT;
config_clock_unit.frame.data_delay = I2S_DATA_DELAY_0;
config_clock_unit.frame.frame_sync.width = I2S_FRAME_SYNC_WIDTH_SLOT;
config_clock_unit.mck_pin.enable = false;
config_clock_unit.sck_pin.enable = true;
config_clock_unit.sck_pin.gpio = self->clock_pin->pin;
// Mux is always the same.
config_clock_unit.sck_pin.mux = 6L;
config_clock_unit.fs_pin.enable = false;
i2s_clock_unit_set_config(&self->i2s_instance, self->clock_unit, &config_clock_unit);
struct i2s_serializer_config config_serializer;
i2s_serializer_get_config_defaults(&config_serializer);
config_serializer.clock_unit = self->clock_unit;
config_serializer.mode = I2S_SERIALIZER_PDM2;
config_serializer.data_size = I2S_DATA_SIZE_32BIT;
config_serializer.data_pin.gpio = self->data_pin->pin;
// Mux is always the same.
config_serializer.data_pin.mux = 6L;
config_serializer.data_pin.enable = true;
i2s_serializer_set_config(&self->i2s_instance, self->serializer, &config_serializer);
i2s_enable(&self->i2s_instance);
// Run the serializer all the time. This eliminates startup delay for the microphone.
i2s_clock_unit_enable(&self->i2s_instance, self->clock_unit);
i2s_serializer_enable(&self->i2s_instance, self->serializer);
self->bytes_per_sample = oversample >> 3;
self->bit_depth = bit_depth;
}
bool common_hal_audiobusio_pdmin_deinited(audiobusio_pdmin_obj_t* self) {
return self->clock_pin == mp_const_none;
}
void common_hal_audiobusio_pdmin_deinit(audiobusio_pdmin_obj_t* self) {
if (common_hal_audiobusio_pdmin_deinited(self)) {
return;
}
i2s_disable(&self->i2s_instance);
i2s_serializer_disable(&self->i2s_instance, self->serializer);
i2s_clock_unit_disable(&self->i2s_instance, self->clock_unit);
i2s_reset(&self->i2s_instance);
reset_pin(self->clock_pin->pin);
reset_pin(self->data_pin->pin);
self->clock_pin = mp_const_none;
self->data_pin = mp_const_none;
}
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_frequency(audiobusio_pdmin_obj_t* self) {
return self->frequency;
}
static void setup_dma(audiobusio_pdmin_obj_t* self, uint32_t length,
DmacDescriptor* second_descriptor,
uint8_t words_per_buffer, uint8_t words_per_sample,
uint32_t* first_buffer, uint32_t* second_buffer) {
// Set up the DMA
struct dma_descriptor_config descriptor_config;
dma_descriptor_get_config_defaults(&descriptor_config);
descriptor_config.beat_size = DMA_BEAT_SIZE_WORD;
descriptor_config.step_selection = DMA_STEPSEL_SRC;
descriptor_config.source_address = (uint32_t)&I2S->DATA[self->serializer];
descriptor_config.src_increment_enable = false;
// Block transfer count is the number of beats per block (aka descriptor).
// In this case there are two bytes per beat so divide the length by two.
uint16_t block_transfer_count = words_per_buffer;
if (length * words_per_sample < words_per_buffer) {
block_transfer_count = length * words_per_sample;
}
descriptor_config.block_transfer_count = block_transfer_count;
descriptor_config.destination_address = ((uint32_t) first_buffer + sizeof(uint32_t) * block_transfer_count);
descriptor_config.event_output_selection = DMA_EVENT_OUTPUT_BLOCK;
descriptor_config.next_descriptor_address = 0;
if (length * words_per_sample > words_per_buffer) {
descriptor_config.next_descriptor_address = ((uint32_t)second_descriptor);
}
dma_descriptor_create(audio_dma.descriptor, &descriptor_config);
// Do we need more values than will fit in the first buffer?
// If so, set up a second buffer chained to be filled after the first buffer.
if (length * words_per_sample > words_per_buffer) {
block_transfer_count = words_per_buffer;
descriptor_config.next_descriptor_address = ((uint32_t)audio_dma.descriptor);
if (length * words_per_sample < 2 * words_per_buffer) {
// Length needed is more than one buffer but less than two.
// Subtract off the size of the first buffer, and what remains is the count we need.
block_transfer_count = length * words_per_sample - words_per_buffer;
descriptor_config.next_descriptor_address = 0;
}
descriptor_config.block_transfer_count = block_transfer_count;
descriptor_config.destination_address = ((uint32_t) second_buffer + sizeof(uint32_t) * block_transfer_count);
dma_descriptor_create(second_descriptor, &descriptor_config);
}
switch_audiodma_trigger(I2S_DMAC_ID_RX_0 + self->serializer);
}
void start_dma(audiobusio_pdmin_obj_t* self) {
dma_start_transfer_job(&audio_dma);
tc_start_counter(MP_STATE_VM(audiodma_block_counter));
I2S->DATA[1].reg = I2S->DATA[1].reg;
}
void stop_dma(audiobusio_pdmin_obj_t* self) {
// Shutdown the DMA: serializer keeps running.
tc_stop_counter(MP_STATE_VM(audiodma_block_counter));
dma_abort_job(&audio_dma);
}
// 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.
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_16_TIMES(X) X X X X X X X X X X X X X X X X
static uint16_t filter_sample(uint32_t pdm_samples[4]) {
uint16_t running_sum = 0;
const uint16_t *filter_ptr = sinc_filter;
for (uint8_t i = 0; i < OVERSAMPLING/16; i++) {
// The sample is 16-bits right channel in the upper two bytes and 16-bits left channel
// in the lower two bytes.
// We just ignore the upper bits
uint32_t pdm_sample = pdm_samples[i];
REPEAT_16_TIMES( {
if (pdm_sample & 0x8000) {
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) {
// We allocate two buffers on the stack to use for double buffering.
const uint8_t samples_per_buffer = SAMPLES_PER_BUFFER;
// For every word we record, we throw away 2 bytes of a phantom second channel.
const uint8_t words_per_sample = self->bytes_per_sample / 2;
const uint8_t words_per_buffer = samples_per_buffer * words_per_sample;
uint32_t first_buffer[words_per_buffer];
uint32_t second_buffer[words_per_buffer];
COMPILER_ALIGNED(16) DmacDescriptor second_descriptor;
setup_dma(self, output_buffer_length, &second_descriptor, words_per_buffer,
words_per_sample, first_buffer, second_buffer);
start_dma(self);
// Record
uint32_t buffers_processed = 0;
uint32_t values_output = 0;
uint32_t remaining_samples_needed = output_buffer_length;
while (values_output < output_buffer_length) {
// Wait for the next buffer to fill
uint32_t block_counter;
while ((block_counter = tc_get_count_value(MP_STATE_VM(audiodma_block_counter))) == buffers_processed) {
#ifdef MICROPY_VM_HOOK_LOOP
MICROPY_VM_HOOK_LOOP
#endif
}
if (block_counter != (buffers_processed + 1)) {
// Looks like we aren't keeping up. We shouldn't skip a buffer.
break;
}
// The mic is running all the time, so we don't need to wait the usual 10msec or 100msec
// for it to start up.
// Flip back and forth between processing the first and second buffers.
uint32_t *buffer = first_buffer;
DmacDescriptor* descriptor = audio_dma.descriptor;
if (buffers_processed % 2 == 1) {
buffer = second_buffer;
descriptor = &second_descriptor;
}
// Decimate and filter the buffer that was just filled.
uint32_t samples_gathered = descriptor->BTCNT.reg / words_per_sample;
// Don't run off the end of output buffer. Process only as many as needed.
uint32_t samples_to_process = min(remaining_samples_needed, samples_gathered);
for (uint32_t i = 0; i < samples_to_process; i++) {
// Call filter_sample just one place so it can be inlined.
uint16_t value = filter_sample(buffer + i * words_per_sample);
if (self->bit_depth == 8) {
// Truncate to 8 bits.
((uint8_t*) output_buffer)[values_output] = value >> 8;
} else {
output_buffer[values_output] = value;
}
values_output++;
}
buffers_processed++;
// Compute how many more samples we need, and if the last buffer is the last
// set of samples needed, adjust the DMA count to only fetch as necessary.
remaining_samples_needed = output_buffer_length - values_output;
if (remaining_samples_needed <= samples_per_buffer*2 &&
remaining_samples_needed > samples_per_buffer) {
// Adjust the DMA settings for the current buffer, which will be processed
// after the other buffer, which is now receiving samples via DMA.
// We don't adjust the DMA in progress, but the one after that.
// Timeline:
// 1. current buffer (already processed)
// 2. alternate buffer (DMA in progress)
// 3. current buffer (last set of samples needed)
// Set up to receive the last set of samples (don't include the alternate buffer, now in use).
uint32_t samples_needed_for_last_buffer = remaining_samples_needed - samples_per_buffer;
descriptor->BTCNT.reg = samples_needed_for_last_buffer * words_per_sample;
descriptor->DSTADDR.reg = ((uint32_t) buffer)
+ samples_needed_for_last_buffer * words_per_sample * sizeof(buffer[0]);
// Break chain to alternate buffer.
descriptor->DESCADDR.reg = 0;
}
}
stop_dma(self);
return values_output;
}
void common_hal_audiobusio_pdmin_record_to_file(audiobusio_pdmin_obj_t* self, uint8_t* buffer, uint32_t length) {
}
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