circuitpython/ports/stm/common-hal/audiobusio/PDMIn.c
Scott Shawcroft 8137e2d6d2
Switch all ports to auto-growing split heap
This simplifies allocating outside of the VM because the VM doesn't
take up all remaining memory by default.

On ESP we delegate to the IDF for allocations. For all other ports,
we use TLSF to manage an outer "port" heap. The IDF uses TLSF
internally and we use their fork for the other ports.

This also removes the dynamic C stack sizing. It wasn't often used
and is not possible with a fixed outer heap.

Fixes #8512. Fixes #7334.
2023-11-01 15:24:16 -07:00

199 lines
6.5 KiB
C

/*
* This file is part of the MicroPython project, http://micropython.org/
*
* The MIT License (MIT)
*
* Copyright (c) 2022 Matthew McGowan for Blues Inc.
*
* 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 "common-hal/audiobusio/PDMIn.h"
#include "shared-bindings/audiobusio/PDMIn.h"
#include "shared-bindings/microcontroller/Pin.h"
#include "py/runtime.h"
#include "MEMS_Audio_ll_stm32l4.h"
MemsAudio memsAudio;
MemsAudio_STM32L4SAIPDM memsAudioImpl;
pdm_sample_t pdmBuffer[MEMS_AUDIO_PDM_BUFFER_LENGTH];
audiobusio_pdmin_obj_t *instance;
static bool pdm_data_available(MemsAudio_STM32L4SAIPDM *impl, uint8_t *pdmBuffer, size_t pdmBufferLength);
// Caller validates that pins are free.
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 sample_rate,
uint8_t bit_depth,
bool mono,
uint8_t oversample) {
self->sample_rate = sample_rate;
self->mono = mono;
self->oversample = oversample;
self->recording_complete = true;
if (!mono) {
mp_raise_ValueError(MP_ERROR_TEXT("only mono is supported"));
}
if (sample_rate != 16000) {
mp_raise_ValueError(MP_ERROR_TEXT("only sample_rate=16000 is supported"));
}
if (bit_depth != 16) {
mp_raise_ValueError(MP_ERROR_TEXT("only bit_depth=16 is supported"));
}
if (oversample != 64) {
mp_raise_ValueError(MP_ERROR_TEXT("only oversample=64 is supported"));
}
// wait for the previous instance to finish.
if (instance) {
common_hal_audiobusio_pdmin_deinit(instance);
}
instance = self;
memset(&memsAudio, 0, sizeof(memsAudio));
memset(&memsAudioImpl, 0, sizeof(memsAudioImpl));
common_hal_mcu_pin_claim(clock_pin);
self->clock_pin = clock_pin;
common_hal_mcu_pin_claim(data_pin);
self->data_pin = data_pin;
self->audio = &memsAudio;
self->audio_impl = &memsAudioImpl;
self->audio_impl->pdmBuffer = pdmBuffer;
self->audio_impl->pdmBufferLength = sizeof(pdmBuffer) / sizeof(pdmBuffer[0]);
self->audio_impl->pdm_data_available = pdm_data_available;
mems_audio_init_stm32l4_sai_pdm(self->audio, self->audio_impl);
mems_audio_record(self->audio);
mems_audio_pause(self->audio);
}
bool common_hal_audiobusio_pdmin_deinited(audiobusio_pdmin_obj_t *self) {
return self->clock_pin == NULL;
}
void wait_dma_complete(audiobusio_pdmin_obj_t *self) {
while (!self->recording_complete) {
MICROPY_VM_HOOK_LOOP;
}
}
void common_hal_audiobusio_pdmin_deinit(audiobusio_pdmin_obj_t *self) {
if (instance != self) {
return;
}
instance = NULL;
if (self->audio) {
wait_dma_complete(self);
mems_audio_stop(self->audio);
mems_audio_uninit(self->audio);
self->audio = NULL;
self->audio_impl = NULL;
}
if (self->data_pin) {
common_hal_reset_pin(self->data_pin);
self->data_pin = NULL;
}
if (self->clock_pin) {
common_hal_reset_pin(self->clock_pin);
self->clock_pin = NULL;
}
}
uint8_t common_hal_audiobusio_pdmin_get_bit_depth(audiobusio_pdmin_obj_t *self) {
return 16;
}
uint32_t common_hal_audiobusio_pdmin_get_sample_rate(audiobusio_pdmin_obj_t *self) {
return 16000;
}
static bool pdm_data_available(MemsAudio_STM32L4SAIPDM *impl, uint8_t *pdmBuffer, size_t pdmBufferLength) {
// update the filter with the correct number of samples
audiobusio_pdmin_obj_t *pdmIn = (audiobusio_pdmin_obj_t *)(impl->audio->userData);
MemsAudio *audio = impl->audio;
uint32_t pcmSamplesAvailable = pdmBufferLength * 8 / PDM_IN_DECIMATION_FACTOR;
if (pcmSamplesAvailable > audio->pcmOutputBufferLength) {
pcmSamplesAvailable = audio->pcmOutputBufferLength;
}
// ensure the filter doesn't try to produce more samples than available
pdmIn->audio_impl->filter.nSamples = pcmSamplesAvailable;
return pcmSamplesAvailable > 0;
}
static void pcm_data_available(MemsAudio *audio, int16_t *pcmBuffer, size_t pcmBufferLength) {
// data is already in the output buffer
audiobusio_pdmin_obj_t *pdmIn = (audiobusio_pdmin_obj_t *)(audio->userData);
// if DMA copies more data than will fit into the output buffer, crop the length to what will fit
if (audio->pcmOutputBufferLength < pcmBufferLength) {
pcmBufferLength = audio->pcmOutputBufferLength;
}
audio->pcmOutputBuffer += pcmBufferLength;
audio->pcmOutputBufferLength -= pcmBufferLength;
if (audio->pcmOutputBufferLength == 0) {
pdmIn->recording_complete = true;
mems_audio_pause(audio);
}
}
uint32_t common_hal_audiobusio_pdmin_record_to_buffer(audiobusio_pdmin_obj_t *self,
uint16_t *output_buffer, uint32_t output_buffer_length) {
MemsAudio *audio = self->audio;
wait_dma_complete(self);
audio->pcmOutputBuffer = (int16_t *)output_buffer;
audio->pcmOutputBufferLength = output_buffer_length;
audio->pcm_data_available = pcm_data_available;
audio->userData = self; /// reference back to the PDMIn instance
self->recording_complete = false;
mems_audio_err_t err = mems_audio_resume(audio);
if (!IS_MEMS_AUDIO_ERROR(err)) {
wait_dma_complete(self);
}
mems_audio_pause(audio);
int samples_output = (int)(output_buffer_length) - audio->pcmOutputBufferLength;
// convert from signed to unsigned (min-point moves from 0 to 32k)
for (int i = 0; i < samples_output; i++) {
output_buffer[i] += 0x8000;
}
return samples_output;
}