/* * 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 #include #include #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 "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) { }