diff options
| -rw-r--r-- | atmel-samd/common-hal/audiobusio/PDMIn.c | 153 | ||||
| -rw-r--r-- | shared-bindings/audiobusio/PDMIn.c | 53 |
2 files changed, 129 insertions, 77 deletions
diff --git a/atmel-samd/common-hal/audiobusio/PDMIn.c b/atmel-samd/common-hal/audiobusio/PDMIn.c index db671f213..8263ac7d2 100644 --- a/atmel-samd/common-hal/audiobusio/PDMIn.c +++ b/atmel-samd/common-hal/audiobusio/PDMIn.c @@ -26,6 +26,7 @@ #include <stdint.h> #include <string.h> +#include <math.h> #include "py/gc.h" #include "py/mperrno.h" @@ -42,6 +43,12 @@ #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; @@ -96,8 +103,8 @@ void common_hal_audiobusio_pdmin_construct(audiobusio_pdmin_obj_t* self, mp_raise_RuntimeError("Unable to allocate audio DMA block counter."); } - if (!(bit_depth == 16 || bit_depth == 8) || !mono || oversample != 64) { - mp_raise_NotImplementedError("Only 8 or 16 bit mono with 64 oversample is supported."); + 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. @@ -112,12 +119,17 @@ void common_hal_audiobusio_pdmin_construct(audiobusio_pdmin_obj_t* self, config_clock_unit.clock.mck_out_enable = false; config_clock_unit.clock.sck_src = I2S_SERIAL_CLOCK_SOURCE_MCKDIV; - config_clock_unit.clock.sck_div = 8000000 / frequency / oversample; - self->frequency = 8000000 / config_clock_unit.clock.sck_div / oversample; + 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_1; + config_clock_unit.frame.data_delay = I2S_DATA_DELAY_0; config_clock_unit.frame.frame_sync.width = I2S_FRAME_SYNC_WIDTH_SLOT; @@ -141,6 +153,10 @@ void common_hal_audiobusio_pdmin_construct(audiobusio_pdmin_obj_t* self, 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; } @@ -154,6 +170,8 @@ void common_hal_audiobusio_pdmin_deinit(audiobusio_pdmin_obj_t* 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); @@ -213,74 +231,82 @@ static void setup_dma(audiobusio_pdmin_obj_t* self, uint32_t length, void start_dma(audiobusio_pdmin_obj_t* self) { dma_start_transfer_job(&audio_dma); tc_start_counter(MP_STATE_VM(audiodma_block_counter)); - i2s_clock_unit_enable(&self->i2s_instance, self->clock_unit); - i2s_serializer_enable(&self->i2s_instance, self->serializer); I2S->DATA[1].reg = I2S->DATA[1].reg; } void stop_dma(audiobusio_pdmin_obj_t* self) { - // Turn off the I2S clock and serializer. Peripheral is still enabled. - i2s_serializer_disable(&self->i2s_instance, self->serializer); - i2s_clock_unit_disable(&self->i2s_instance, self->clock_unit); - - // Shutdown the DMA + // Shutdown the DMA: serializer keeps running. tc_stop_counter(MP_STATE_VM(audiodma_block_counter)); dma_abort_job(&audio_dma); } -static const uint16_t sinc_filter[64] = { - 0, 1, 6, 16, 29, 49, 75, 108, - 149, 200, 261, 334, 418, 514, 622, 742, - 872, 1012, 1161, 1315, 1472, 1631, 1787, 1938, - 2081, 2212, 2329, 2429, 2509, 2568, 2604, 2616, - 2604, 2568, 2509, 2429, 2329, 2212, 2081, 1938, - 1787, 1631, 1472, 1315, 1161, 1012, 872, 742, - 622, 514, 418, 334, 261, 200, 149, 108, - 75, 49, 29, 16, 6, 1, 0, 0 +// 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 sample = 0; - for (uint8_t i = 0; i < 4; i++) { - uint16_t pdm = pdm_samples[i] & 0xffff; - for (uint8_t j = 0; j < 16; j++) { - if ((pdm & 0x8000) != 0) { - sample += sinc_filter[i * 16 + j]; + 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++; + } + pdm_sample <<= 1; } - pdm <<= 1; - } + ) } - return sample; + 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 length) { - // Write the wave file header. - - // We allocate two 256 byte buffers on the stack to use for double buffering. - // Our oversample rate is 64 (bits) so each buffer produces 32 samples. - // TODO(tannewt): Can the compiler optimize better if we fix the size of - // these buffers? - uint8_t samples_per_buffer = 32; + 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. - uint8_t words_per_sample = self->bytes_per_sample / 2; - uint8_t words_per_buffer = samples_per_buffer * words_per_sample; + 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, length, &second_descriptor, words_per_buffer, + 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 total_bytes = 0; + uint32_t values_output = 0; - uint64_t start_ticks = ticks_ms; - while (total_bytes < length) { + uint32_t remaining_samples_needed = output_buffer_length; + while (values_output < output_buffer_length) { // Wait for the next buffer to fill while (tc_get_count_value(MP_STATE_VM(audiodma_block_counter)) == buffers_processed) { #ifdef MICROPY_VM_HOOK_LOOP @@ -290,40 +316,45 @@ uint32_t common_hal_audiobusio_pdmin_record_to_buffer(audiobusio_pdmin_obj_t* se if (tc_get_count_value(MP_STATE_VM(audiodma_block_counter)) != (buffers_processed + 1)) { break; } - // Throw away the first ~10ms of data because thats during mic start up. - if (ticks_ms - start_ticks < 10) { - buffers_processed++; - continue; - } - uint32_t* buffer = first_buffer; + + // The mic is running all the time, so we don't need to wait the usual 10msec or 100msec + // for it to start up. + + 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 last buffer - int32_t samples_gathered = descriptor->BTCNT.reg / words_per_sample; - for (uint16_t i = 0; i < samples_gathered; i++) { + 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) { - ((uint8_t*) output_buffer)[total_bytes] = filter_sample(buffer + i * words_per_sample) >> 8; - total_bytes += 1; - } else if (self->bit_depth == 16) { - output_buffer[total_bytes / 2] = filter_sample(buffer + i * words_per_sample); - total_bytes += 2; + // Truncate to 8 bits. + ((uint8_t*) output_buffer)[values_output] = value >> 8; + } else { + output_buffer[values_output] = value; } + values_output++; } buffers_processed++; - if (length - total_bytes < samples_per_buffer) { - descriptor->BTCNT.reg = (length - total_bytes) * words_per_sample; - descriptor->DSTADDR.reg = ((uint32_t) buffer) + (length - total_bytes) * self->bytes_per_sample; + // See if we need to transfer less than a full buffer for the remaining needed samples. + remaining_samples_needed = output_buffer_length - values_output; + if (remaining_samples_needed > 0 && remaining_samples_needed < samples_per_buffer) { + descriptor->BTCNT.reg = remaining_samples_needed; + descriptor->DSTADDR.reg = ((uint32_t) buffer) + remaining_samples_needed * words_per_sample; descriptor->DESCADDR.reg = 0; } } stop_dma(self); - return total_bytes; + return values_output; } void common_hal_audiobusio_pdmin_record_to_file(audiobusio_pdmin_obj_t* self, uint8_t* buffer, uint32_t length) { diff --git a/shared-bindings/audiobusio/PDMIn.c b/shared-bindings/audiobusio/PDMIn.c index e13e09d64..c71f44679 100644 --- a/shared-bindings/audiobusio/PDMIn.c +++ b/shared-bindings/audiobusio/PDMIn.c @@ -28,6 +28,7 @@ #include "lib/utils/context_manager_helpers.h" #include "py/binary.h" +#include "py/mphal.h" #include "py/objproperty.h" #include "py/runtime.h" #include "shared-bindings/microcontroller/Pin.h" @@ -41,7 +42,7 @@ //| //| PDMIn can be used to record an input audio signal on a given set of pins. //| -//| .. class:: PDMIn(clock_pin, data_pin, \*, frequency=8000, bit_depth=8, mono=True, oversample=64) +//| .. class:: PDMIn(clock_pin, data_pin, \*, frequency=16000, bit_depth=8, mono=True, oversample=64, startup_delay=0.11) //| //| Create a PDMIn object associated with the given pins. This allows you to //| record audio signals from the given pins. Individual ports may put further @@ -49,11 +50,16 @@ //| //| :param ~microcontroller.Pin clock_pin: The pin to output the clock to //| :param ~microcontroller.Pin data_pin: The pin to read the data from -//| :param int frequency: Target frequency of the resulting samples. Check `frequency` for real value. +//| :param int frequency: Target frequency of the resulting samples. Check `frequency` for actual value. +//| Minimum frequency is about 16000 Hz. //| :param int bit_depth: Final number of bits per sample. Must be divisible by 8 //| :param bool mono: True when capturing a single channel of audio, captures two channels otherwise //| :param int oversample: Number of single bit samples to decimate into a final sample. Must be divisible by 8 +//| :param float startup_delay: seconds to wait after starting microphone clock +//| to allow microphone to turn on. Most require only 0.01s; some require 0.1s. Longer is safer. +//| Must be in range 0.0-1.0 seconds. //| + //| Record 8-bit unsigned samples to buffer:: //| //| import audiobusio @@ -79,15 +85,19 @@ //| mic.record(b, len(b)) //| STATIC mp_obj_t audiobusio_pdmin_make_new(const mp_obj_type_t *type, size_t n_args, size_t n_kw, const mp_obj_t *pos_args) { - enum { ARG_frequency, ARG_bit_depth, ARG_mono, ARG_oversample }; + enum { ARG_frequency, ARG_bit_depth, ARG_mono, ARG_oversample, ARG_startup_delay }; mp_map_t kw_args; mp_map_init_fixed_table(&kw_args, n_kw, pos_args + n_args); static const mp_arg_t allowed_args[] = { - { MP_QSTR_frequency, MP_ARG_INT, {.u_int = 8000} }, - { MP_QSTR_bit_depth, MP_ARG_INT, {.u_int = 8} }, - { MP_QSTR_mono, MP_ARG_BOOL,{.u_bool = true} }, - { MP_QSTR_oversample, MP_ARG_INT, {.u_int = 64} }, + { MP_QSTR_frequency, MP_ARG_KW_ONLY | MP_ARG_INT, {.u_int = 16000} }, + { MP_QSTR_bit_depth, MP_ARG_KW_ONLY | MP_ARG_INT, {.u_int = 8} }, + { MP_QSTR_mono, MP_ARG_KW_ONLY | MP_ARG_BOOL,{.u_bool = true} }, + { MP_QSTR_oversample, MP_ARG_KW_ONLY | MP_ARG_INT, {.u_int = 64} }, + { MP_QSTR_startup_delay, MP_ARG_KW_ONLY | MP_ARG_OBJ, {.u_obj = MP_OBJ_NULL} }, }; + // Default microphone startup delay is 110msecs. Have seen mics that need 100 msecs plus a bit. + static const float STARTUP_DELAY_DEFAULT = 0.110F; + mp_arg_val_t args[MP_ARRAY_SIZE(allowed_args)]; mp_arg_parse_all(n_args - 2, pos_args + 2, &kw_args, MP_ARRAY_SIZE(allowed_args), allowed_args, args); @@ -116,8 +126,18 @@ STATIC mp_obj_t audiobusio_pdmin_make_new(const mp_obj_type_t *type, size_t n_ar } bool mono = args[ARG_mono].u_bool; + float startup_delay = (args[ARG_startup_delay].u_obj == MP_OBJ_NULL) + ? STARTUP_DELAY_DEFAULT + : mp_obj_get_float(args[ARG_startup_delay].u_obj); + if (startup_delay < 0.0 || startup_delay > 1.0) { + mp_raise_ValueError("Microphone startup delay must be in range 0.0 to 1.0"); + } + common_hal_audiobusio_pdmin_construct(self, clock_pin, data_pin, frequency, - bit_depth, mono, oversample); + bit_depth, mono, oversample); + + // Wait for the microphone to start up. Some start in 10 msecs; some take as much as 100 msecs. + mp_hal_delay_ms(startup_delay * 1000); return MP_OBJ_FROM_PTR(self); } @@ -160,11 +180,14 @@ STATIC MP_DEFINE_CONST_FUN_OBJ_VAR_BETWEEN(audiobusio_pdmin___exit___obj, 4, 4, //| audio at the given rate. For internal flash, writing all 1s to the file //| before recording is recommended to speed up writes. //| +//| :return: The number of samples recorded. If this is less than `destination_length`, +//| some samples were missed due to processing time. +//| STATIC mp_obj_t audiobusio_pdmin_obj_record(mp_obj_t self_obj, mp_obj_t destination, mp_obj_t destination_length) { audiobusio_pdmin_obj_t *self = MP_OBJ_TO_PTR(self_obj); raise_error_if_deinited(common_hal_audiobusio_pdmin_deinited(self)); - if (!MP_OBJ_IS_SMALL_INT(destination_length)) { - mp_raise_TypeError("destination_length must be int"); + if (!MP_OBJ_IS_SMALL_INT(destination_length) || MP_OBJ_SMALL_INT_VALUE(destination_length) < 0) { + mp_raise_TypeError("destination_length must be an int >= 0"); } uint32_t length = MP_OBJ_SMALL_INT_VALUE(destination_length); @@ -172,8 +195,8 @@ STATIC mp_obj_t audiobusio_pdmin_obj_record(mp_obj_t self_obj, mp_obj_t destinat if (MP_OBJ_IS_TYPE(destination, &fatfs_type_fileio)) { mp_raise_NotImplementedError(""); } else if (mp_get_buffer(destination, &bufinfo, MP_BUFFER_WRITE)) { - if (bufinfo.len < length) { - mp_raise_ValueError("Target buffer cannot hold destination_length bytes."); + if (bufinfo.len / mp_binary_get_size('@', bufinfo.typecode, NULL) < length) { + mp_raise_ValueError("Destination capacity is smaller than destination_length."); } uint8_t bit_depth = common_hal_audiobusio_pdmin_get_bit_depth(self); if (bufinfo.typecode != 'H' && bit_depth == 16) { @@ -181,12 +204,10 @@ STATIC mp_obj_t audiobusio_pdmin_obj_record(mp_obj_t self_obj, mp_obj_t destinat } else if (bufinfo.typecode != 'B' && bufinfo.typecode != BYTEARRAY_TYPECODE && bit_depth == 8) { mp_raise_ValueError("destination buffer must be a bytearray or array of type 'B' for bit_depth = 8"); } - length *= bit_depth / 8; + // length is the buffer length in slots, not bytes. uint32_t length_written = common_hal_audiobusio_pdmin_record_to_buffer(self, bufinfo.buf, length); - if (length_written != length) { - mp_printf(&mp_plat_print, "length mismatch %d %d\n", length_written, length); - } + return MP_OBJ_NEW_SMALL_INT(length_written); } return mp_const_none; } |