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Merge pull request #7169 from blues/pdmin

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STM32L4 PDMIn support
This commit is contained in:
Dan Halbert 2022-11-09 14:19:51 -05:00 committed by GitHub
commit e6936decac
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GPG Key ID: 4AEE18F83AFDEB23
17 changed files with 1648 additions and 47 deletions
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10
locale/circuitpython.pot
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@ -3698,10 +3698,20 @@ msgid "offset out of bounds"
msgstr ""
#: ports/nrf/common-hal/audiobusio/PDMIn.c
#: ports/stm/common-hal/audiobusio/PDMIn.c
msgid "only bit_depth=16 is supported"
msgstr ""
#: ports/stm/common-hal/audiobusio/PDMIn.c
msgid "only mono is supported"
msgstr ""
#: ports/stm/common-hal/audiobusio/PDMIn.c
msgid "only oversample=64 is supported"
msgstr ""
#: ports/nrf/common-hal/audiobusio/PDMIn.c
#: ports/stm/common-hal/audiobusio/PDMIn.c
msgid "only sample_rate=16000 is supported"
msgstr ""

11
ports/stm/Makefile
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@ -234,6 +234,17 @@ SRC_C += \
peripherals/stm32$(MCU_SERIES_LOWER)/$(MCU_VARIANT_LOWER)/periph.c \
packages/$(MCU_PACKAGE).c
ifneq ($(CIRCUITPY_AUDIOBUSIO_PDMIN),0)
SRC_C += \
common-hal/audiobusio/MEMS_Audio.c \
common-hal/audiobusio/MEMS_Audio_ll_stm32l4.c \
common-hal/audiobusio/OpenPDMFilter.c
SRC_STM32 += \
$(HAL_DIR)/Src/stm32$(MCU_SERIES_LOWER)xx_hal_sai.c
endif
ifneq ($(CIRCUITPY_USB),0)
SRC_C += lib/tinyusb/src/portable/st/synopsys/dcd_synopsys.c
endif

75
ports/stm/boards/swan_r5/mpconfigboard.mk
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@ -15,61 +15,42 @@ LD_DEFAULT = boards/STM32L4R5_default.ld
LD_BOOT = boards/STM32L4R5_boot.ld
UF2_OFFSET = 0x8010000
UF2_BOOTLOADER ?= 1
CIRCUITPY_BUILD_EXTENSIONS = bin,uf2
# Turn all of the below off while trying to get the thing to run
# These modules are implemented in ports/<port>/common-hal:
# Typically the first module to create
CIRCUITPY_MICROCONTROLLER = 1
CIRCUITPY_ALARM = 1
# Typically the second module to create
CIRCUITPY_DIGITALIO = 1
# Other modules:
CIRCUITPY_OS = 1
CIRCUITPY_STORAGE = 1
CIRCUITPY_USB_MSC = 1
CIRCUITPY_UDB_CDC = 1
CIRCUITPY_USB_VENDOR = 1
CIRCUITPY_NVM = 0
CIRCUITPY_ANALOGIO = 1
CIRCUITPY_AUDIOBUSIO = 1
CIRCUITPY_AUDIOBUSIO_I2SOUT = 0
CIRCUITPY_AUDIOBUSIO_PDMIN = 1
CIRCUITPY_AUDIOPWMIO = 1
CIRCUITPY_BITBANGIO = 1
CIRCUITPY_BLEIO = 0
CIRCUITPY_BLEIO_HCI = 0
CIRCUITPY_BUSDEVICE = 0
CIRCUITPY_BUSIO = 1
CIRCUITPY_CANIO = 0
CIRCUITPY_DIGITALIO = 1
CIRCUITPY_DISPLAYIO = 1
CIRCUITPY_ENABLE_MPY_NATIVE = 1
CIRCUITPY_I2CTARGET = 0
CIRCUITPY_KEYPAD = 1
CIRCUITPY_MICROCONTROLLER = 1
CIRCUITPY_NEOPIXEL_WRITE = 0
CIRCUITPY_NVM = 0
CIRCUITPY_OS = 1
CIRCUITPY_PIXELBUF = 0
CIRCUITPY_PULSEIO = 1
CIRCUITPY_PWMIO = 1
CIRCUITPY_AUDIOPWMIO = 1
CIRCUITPY_CANIO = 0
CIRCUITPY_AUDIOBUSIO = 0
CIRCUITPY_I2CTARGET = 0
# Requires SPI, PulseIO (stub ok):
CIRCUITPY_DISPLAYIO = 0
# These modules are implemented in shared-module/ - they can be included in
# any port once their prerequisites in common-hal are complete.
# Requires DigitalIO:
CIRCUITPY_BITBANGIO = 1
# Requires neopixel_write or SPI (dotstar)
CIRCUITPY_PIXELBUF = 0
# Requires OS
CIRCUITPY_RANDOM = 1
# Requires Microcontroller
CIRCUITPY_TOUCHIO = 1
# Requires USB
CIRCUITPY_USB_HID = 0
CIRCUITPY_USB_MIDI = 0
# Does nothing without I2C
CIRCUITPY_REQUIRE_I2C_PULLUPS = 0
# No requirements, but takes extra flash
CIRCUITPY_ULAB = 1
# requires SPI
CIRCUITPY_SDCARDIO = 0
CIRCUITPY_BLEIO_HCI = 0
CIRCUITPY_BLEIO = 0
CIRCUITPY_BUSDEVICE = 0
CIRCUITPY_KEYPAD = 1
CIRCUITPY_RGBMATRIX = 0
CIRCUITPY_RTC = 1
CIRCUITPY_BUILD_EXTENSIONS = bin,uf2
CIRCUITPY_SDCARDIO = 0
CIRCUITPY_STORAGE = 1
CIRCUITPY_TOUCHIO = 1
CIRCUITPY_UDB_CDC = 1
CIRCUITPY_ULAB = 1
CIRCUITPY_USB_HID = 0
CIRCUITPY_USB_MIDI = 0
CIRCUITPY_USB_MSC = 1
CIRCUITPY_USB_VENDOR = 1

3
ports/stm/boards/swan_r5/pins.c
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@ -129,5 +129,8 @@ STATIC const mp_rom_map_elem_t board_module_globals_table[] = {
{ MP_ROM_QSTR(MP_QSTR_I2C), MP_ROM_PTR(&board_i2c_obj) },
{ MP_ROM_QSTR(MP_QSTR_SPI), MP_ROM_PTR(&board_spi_obj) },
{ MP_ROM_QSTR(MP_QSTR_UART), MP_ROM_PTR(&board_uart_obj) },
{ MP_ROM_QSTR(MP_QSTR_MICROPHONE_CLOCK), MP_ROM_PTR(&pin_PA03) },
{ MP_ROM_QSTR(MP_QSTR_MICROPHONE_DATA), MP_ROM_PTR(&pin_PC03) },
};
MP_DEFINE_CONST_DICT(board_module_globals, board_module_globals_table);

1
ports/stm/common-hal/audiobusio/I2SOut.c Normal file
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@ -0,0 +1 @@
// Although IS2Out is not enabled on the STM32L4 family, this file is still required for the build to pass

1
ports/stm/common-hal/audiobusio/I2SOut.h Normal file
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@ -0,0 +1 @@
// Although IS2Out is not enabled on the STM32L4 family, this file is still required for the build to pass

96
ports/stm/common-hal/audiobusio/MEMS_Audio.c Normal file
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@ -0,0 +1,96 @@
/*
* 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 <assert.h>
#include <stm32l4xx_hal.h>
#include "MEMS_Audio.h"
#include "MEMS_Audio_ll.h"
static void default_pcm_data_available(MemsAudio *audio, pcm_sample_t *pcmSamples, size_t pcmLength) {
}
/**
* @brief Initializes the MemsAudio instance. Only one instance can be initialized and used at a given time.
*
* @param audio
* @return mems_audio_err_t
*/
mems_audio_err_t mems_audio_init(MemsAudio *audio) {
if (!audio->pcm_data_available) {
audio->pcm_data_available = default_pcm_data_available;
}
return mems_audio_ll_init(audio);
}
/**
* @brief Uninitializes the MemsAudio instance.
*
* @param audio
* @return mems_audio_err_t
*/
mems_audio_err_t mems_audio_uninit(MemsAudio *audio) {
return mems_audio_ll_uninit(audio);
}
/**
* @brief Asynchronously records audio.
*
* @param audio
* @param pdmBuffer
* @param pdmBufferLength
* @return mems_audio_err_t
*/
mems_audio_err_t mems_audio_record(MemsAudio *audio) {
return mems_audio_ll_record(audio);
}
/**
* @brief Pause recording audio.
*/
mems_audio_err_t mems_audio_pause(MemsAudio *audio) {
return mems_audio_ll_pause(audio);
}
/**
* @brief Resume recording audio.
*
* @param audio
* @return mems_audio_err_t
*/
mems_audio_err_t mems_audio_resume(MemsAudio *audio) {
return mems_audio_ll_resume(audio);
}
/**
* @brief Stop recording audio and
*
* @param audio
* @return mems_audio_err_t
*/
mems_audio_err_t mems_audio_stop(MemsAudio *audio) {
return mems_audio_ll_stop(audio);
}

156
ports/stm/common-hal/audiobusio/MEMS_Audio.h Normal file
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@ -0,0 +1,156 @@
/*
* 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.
*/
#ifndef _MEMS_AUDIO_H_
#define _MEMS_AUDIO_H_
#include <stdint.h>
#include <stddef.h>
#ifdef __cplusplus
extern "C" {
#endif
/**
* @brief How many milliseconds of audio can fit in the audio buffer(s).
* Interrupts for recieved data fire at half this duration / twice the frequency.
*/
#ifndef MEMS_AUDIO_MS_BUFFER
#define MEMS_AUDIO_MS_BUFFER (1)
#endif
/**
* @brief The number of bits per sample of the PCM output
*/
#define PCM_OUT_RESOLUTION 16
/**
* @brief The output frequency of PCM samples in Hz.
*/
#define PCM_OUT_SAMPLING_FREQUENCY 16000
/**
* @brief type for describing error conditions.
*/
typedef int32_t mems_audio_err_t;
/**
* @brief The datatype that holds an output PCM sample.
*/
typedef int16_t pcm_sample_t;
_Static_assert(PCM_OUT_RESOLUTION==16, "Output PCM resolution must be 16-bits");
typedef enum {
MEMS_AUDIO_OK = 0,
MEMS_AUDIO_ERROR_ALREADY_INITIALIZED = -1,
MEMS_AUDIO_ERROR_NOT_INITIALIZED = -2
} mems_audio_err_enum_t;
#define IS_MEMS_AUDIO_ERROR(e) (e)
#define CHECK_MEMS_AUDIO_ERROR(e) { if (IS_MEMS_AUDIO_ERROR(e)) return e; }
#define CHECK_MEMS_AUDIO_INITIALIZED(x) { if (!x) return MEMS_AUDIO_ERROR_NOT_INITIALIZED; }
typedef struct MemsAudio_t MemsAudio;
/**
* @brief Callback informing that PCM samples are available for processing.
*/
typedef void (*pcm_data_available_t)(MemsAudio* audio, pcm_sample_t* pcmSamples, size_t pcmLength);
/**
* @brief MemsAudio manages the filter, buffers and callbacks used to capture PDM audio samples and convert to PCM.
*
*/
typedef struct MemsAudio_t {
/**
* @brief The buffer to store PCM audio samples
*/
volatile pcm_sample_t* volatile pcmOutputBuffer;
/**
* @brief The length of the PCM buffer. SHould be at least MEMS_AUDIO_PCM_BUFFER_LENGTH
*/
volatile size_t pcmOutputBufferLength;
/**
* @brief Optional callback for when PCM data is available.
*/
pcm_data_available_t pcm_data_available;
void* audioImpl;
void* userData;
} MemsAudio;
mems_audio_err_t mems_audio_init(MemsAudio* audio);
/**
* @brief Uninitializes the MemsAudio instance.
*
* @param audio
* @return mems_audio_err_t
*/
mems_audio_err_t mems_audio_uninit(MemsAudio* audio);
/**
* @brief Asynchronously records audio.
*
* @param audio
* @param pdmBuffer
* @param pdmBufferLength
* @return mems_audio_err_t
*/
mems_audio_err_t mems_audio_record(MemsAudio* audio);
/**
* @brief Pause recording audio.
*/
mems_audio_err_t mems_audio_pause(MemsAudio* audio);
/**
* @brief Resume recording audio.
*
* @param audio
* @return mems_audio_err_t
*/
mems_audio_err_t mems_audio_resume(MemsAudio* audio);
/**
* @brief Stop recording audio and
*
* @param audio
* @return mems_audio_err_t
*/
mems_audio_err_t mems_audio_stop(MemsAudio* audio);
#ifdef __cplusplus
}
#endif
#endif // _MEMS_AUDIO_H_

75
ports/stm/common-hal/audiobusio/MEMS_Audio_ll.h Normal file
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@ -0,0 +1,75 @@
/*
* 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.
*/
#ifndef _MEMS_AUDIO_LL_H_
#define _MEMS_AUDIO_LL_H_
#include "MEMS_Audio.h"
#ifdef __cplusplus
extern "C" {
#endif
mems_audio_err_t mems_audio_ll_init(MemsAudio *audio);
mems_audio_err_t mems_audio_ll_uninit(MemsAudio *audio);
/**
* @brief Asynchronously records audio.
*
* @param audio
* @param pdmBuffer
* @param pdmBufferLength
* @return mems_audio_err_t
*/
mems_audio_err_t mems_audio_ll_record(MemsAudio *audio);
/**
* @brief Pause recording audio.
*/
mems_audio_err_t mems_audio_ll_pause(MemsAudio *audio);
/**
* @brief Resume recording audio.
*
* @param audio
* @return mems_audio_err_t
*/
mems_audio_err_t mems_audio_ll_resume(MemsAudio *audio);
/**
* @brief Stop recording audio and
*
* @param audio
* @return mems_audio_err_t
*/
mems_audio_err_t mems_audio_ll_stop(MemsAudio *audio);
#ifdef __cplusplus
}
#endif
#endif // _MEMS_AUDIO_LL_H_

386
ports/stm/common-hal/audiobusio/MEMS_Audio_ll_stm32l4.c Normal file
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@ -0,0 +1,386 @@
/*
* 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 <stm32l4xx_hal.h>
#include "MEMS_Audio_ll_stm32l4.h"
#include "MEMS_Audio.h"
/**
* @brief The implementation is a singleton.
*
*/
MemsAudio_STM32L4SAIPDM* volatile audioImpl;
static mems_audio_err_t MX_DMA_Init(void);
static mems_audio_err_t MX_DMA_Uninit(void);
static mems_audio_err_t MX_SAI1_Init(void);
#define CHECK_HAL_ERROR(x, e) \
{ \
if ((x) != HAL_OK) \
return e; \
}
/**
* @brief Checks the HAL return code and returns from a void function on error. The
* error is saved to lastError.
*/
#define CHECK_HAL_ERROR_VOID(x, e) \
{ \
if ((x) != HAL_OK) { \
audioImpl->lastError = e; \
return; \
} \
}
#define CHECK_MEMS_AUDIO_ERROR_LAST() \
{ \
if (audioImpl->lastError != MEMS_AUDIO_OK) \
return audioImpl->lastError; \
}
static bool default_pdm_data_available(MemsAudio_STM32L4SAIPDM* audio, pdm_sample_t* pdmSamples, size_t count)
{
return true;
}
int filter_pdm(MemsAudio_STM32L4SAIPDM* impl, pdm_sample_t* input, pcm_sample_t* output)
{
if (impl->filter.Decimation==64) {
Open_PDM_Filter_64(input, output, 1, &impl->filter);
}
else {
Open_PDM_Filter_128(input, output, 1, &impl->filter);
}
return impl->filter.nSamples;
}
static void mems_audio_init_filter(MemsAudio_STM32L4SAIPDM *impl)
{
TPDMFilter_InitStruct* filter = &impl->filter;
filter->Fs = PCM_OUT_SAMPLING_FREQUENCY;
filter->nSamples = MEMS_AUDIO_PCM_BUFFER_LENGTH;
filter->LP_HZ = PCM_OUT_SAMPLING_FREQUENCY / 2; // The Nyquist frequency
filter->HP_HZ = 10; // high pass to remove DC offset
filter->In_MicChannels = 1;
filter->Out_MicChannels = 1;
filter->Decimation = PDM_IN_DECIMATION_FACTOR;
Open_PDM_Filter_Init(filter);
}
volatile unsigned ignore_dma_count;
/**
* @brief Converts PDM samples
*
* @param pdmBuffer The buffer holding the PDM samples
* @param pdmBufferLength The number of samples available
*/
void pdm2pcm(uint8_t *pdmBuffer, size_t pdmBufferLength)
{
MemsAudio_STM32L4SAIPDM *impl = audioImpl;
if (impl)
{
bool convert = impl->discard_dma || impl->pdm_data_available(impl, pdmBuffer, pdmBufferLength);
if (convert)
{
MemsAudio* audio = impl->audio;
filter_pdm(impl, pdmBuffer, (pcm_sample_t*)audio->pcmOutputBuffer);
if (!impl->discard_dma)
audio->pcm_data_available(audio, (pcm_sample_t*)audio->pcmOutputBuffer, impl->filter.nSamples);
else
impl->discard_dma--;
}
}
}
/**
* @brief Initialize the PDM interface ready to begin capture.
* @retval
*/
mems_audio_err_t mems_audio_ll_init(MemsAudio *audio)
{
mems_audio_init_filter(audioImpl);
if (!audioImpl->pdm_data_available) {
audioImpl->pdm_data_available = &default_pdm_data_available;
}
CHECK_MEMS_AUDIO_ERROR(MX_DMA_Init());
CHECK_MEMS_AUDIO_ERROR(MX_SAI1_Init());
return MEMS_AUDIO_OK;
}
mems_audio_err_t uninit(void) {
if (audioImpl) {
MemsAudio_STM32L4SAIPDM* impl = audioImpl;
audioImpl = NULL;
mems_audio_ll_stop(impl->audio);
CHECK_HAL_ERROR(HAL_SAI_DeInit(&impl->hSAI_BlockA1), MEMS_AUDIO_ERROR_SAI_DEINIT);
CHECK_MEMS_AUDIO_ERROR(MX_DMA_Uninit());
}
return MEMS_AUDIO_OK;
}
/**
* @brief Uninitialize low level PDM capture
*/
mems_audio_err_t mems_audio_ll_uninit(MemsAudio *audio)
{
if (audioImpl->audio == audio) {
uninit();
}
return MEMS_AUDIO_OK;
}
mems_audio_err_t mems_audio_ll_record(MemsAudio *audio)
{
audioImpl->discard_dma = (100/MEMS_AUDIO_MS_BUFFER)+1;
CHECK_HAL_ERROR(HAL_SAI_Receive_DMA(&audioImpl->hSAI_BlockA1, audioImpl->pdmBuffer, audioImpl->pdmBufferLength),
MEMS_AUDIO_ERROR_DMA_START);
return MEMS_AUDIO_OK;
}
mems_audio_err_t mems_audio_ll_stop(MemsAudio *audio)
{
CHECK_HAL_ERROR(HAL_SAI_DMAStop(&audioImpl->hSAI_BlockA1), MEMS_AUDIO_ERROR_DMA_STOP);
return MEMS_AUDIO_OK;
}
mems_audio_err_t mems_audio_ll_pause(MemsAudio *audio)
{
CHECK_HAL_ERROR(HAL_SAI_DMAPause(&audioImpl->hSAI_BlockA1), MEMS_AUDIO_ERROR_DMA_PAUSE);
return MEMS_AUDIO_OK;
}
mems_audio_err_t mems_audio_ll_resume(MemsAudio *audio)
{
CHECK_HAL_ERROR(HAL_SAI_DMAResume(&audioImpl->hSAI_BlockA1), MEMS_AUDIO_ERROR_DMA_RESUME);
return MEMS_AUDIO_OK;
}
/**
* @brief SAI1 Initialization Function
* @param None
* @retval None
*/
static mems_audio_err_t MX_SAI1_Init(void)
{
__HAL_RCC_GPIOA_CLK_ENABLE();
__HAL_RCC_GPIOC_CLK_ENABLE();
SAI_HandleTypeDef hSAI_BlockA1 = {0};
MemsAudio_STM32L4SAIPDM* impl = audioImpl;
CHECK_MEMS_AUDIO_INITIALIZED(impl);
hSAI_BlockA1.Instance = SAI1_Block_A;
hSAI_BlockA1.Init.Protocol = SAI_FREE_PROTOCOL;
hSAI_BlockA1.Init.AudioMode = SAI_MODEMASTER_RX;
/* The PDM interface provides 8 1-bit samples at a time */
hSAI_BlockA1.Init.DataSize = SAI_DATASIZE_8;
hSAI_BlockA1.Init.FirstBit = SAI_FIRSTBIT_MSB;
hSAI_BlockA1.Init.ClockStrobing = SAI_CLOCKSTROBING_FALLINGEDGE;
hSAI_BlockA1.Init.Synchro = SAI_ASYNCHRONOUS; /* asynchronous - not chained to other SAI blocks */
hSAI_BlockA1.Init.OutputDrive = SAI_OUTPUTDRIVE_DISABLE; /* Not driving the primary SAI clock */
hSAI_BlockA1.Init.NoDivider = SAI_MASTERDIVIDER_DISABLE;
hSAI_BlockA1.Init.MckOverSampling = SAI_MCK_OVERSAMPLING_DISABLE;
hSAI_BlockA1.Init.FIFOThreshold = SAI_FIFOTHRESHOLD_FULL;
hSAI_BlockA1.Init.MonoStereoMode = SAI_MONOMODE; /* PDM is intrinsicly stereo, sampling on */
hSAI_BlockA1.Init.CompandingMode = SAI_NOCOMPANDING;
hSAI_BlockA1.Init.PdmInit.Activation = ENABLE; /* Enable PDM interface in the SAI */
hSAI_BlockA1.Init.PdmInit.MicPairsNbr = 1; /* 1 pair - 2 mics */
hSAI_BlockA1.Init.PdmInit.ClockEnable = SAI_PDM_CLOCK1_ENABLE;
hSAI_BlockA1.FrameInit.FrameLength = 16;
hSAI_BlockA1.FrameInit.ActiveFrameLength = 1;
hSAI_BlockA1.FrameInit.FSDefinition = SAI_FS_STARTFRAME; /* FS is not really used */
hSAI_BlockA1.FrameInit.FSPolarity = SAI_FS_ACTIVE_HIGH;
hSAI_BlockA1.FrameInit.FSOffset = SAI_FS_FIRSTBIT;
hSAI_BlockA1.SlotInit.FirstBitOffset = 0;
hSAI_BlockA1.SlotInit.SlotSize = SAI_SLOTSIZE_DATASIZE;
hSAI_BlockA1.SlotInit.SlotNumber = 2;
hSAI_BlockA1.SlotInit.SlotActive = 0x0001;
impl->hSAI_BlockA1 = hSAI_BlockA1;
CHECK_HAL_ERROR(HAL_SAI_Init(&impl->hSAI_BlockA1), MEMS_AUDIO_ERROR_SAI_INIT);
CHECK_MEMS_AUDIO_ERROR_LAST();
return MEMS_AUDIO_OK;
}
#define MEMS_AUDIO_DMA_IRQn DMA1_Channel6_IRQn
#define MEMS_AUDIO_DMA_CHANNEL DMA1_Channel6
#define MEMS_AUDIO_DMA_PRIORITY 6
#define DMA_HANDLER DMA1_Channel6_IRQHandler
void HAL_SAI_MspInit(SAI_HandleTypeDef *hsai)
{
GPIO_InitTypeDef GPIO_InitStruct = {0};
RCC_PeriphCLKInitTypeDef PeriphClkInit = {0};
/* SAI1 */
MemsAudio_STM32L4SAIPDM* impl = audioImpl;
if (hsai->Instance == SAI1_Block_A && impl)
{
PeriphClkInit.PeriphClockSelection = RCC_PERIPHCLK_SAI1;
PeriphClkInit.Sai1ClockSelection = RCC_SAI1CLKSOURCE_PLLSAI1;
PeriphClkInit.PLLSAI1.PLLSAI1Source = RCC_PLLSOURCE_MSI;
PeriphClkInit.PLLSAI1.PLLSAI1M = MEMS_AUDIO_CLOCK_PLLM;
PeriphClkInit.PLLSAI1.PLLSAI1N = MEMS_AUDIO_CLOCK_PLLN;
PeriphClkInit.PLLSAI1.PLLSAI1P = MEMS_AUDIO_CLOCK_PLLP;
PeriphClkInit.PLLSAI1.PLLSAI1Q = RCC_PLLQ_DIV2;
PeriphClkInit.PLLSAI1.PLLSAI1R = RCC_PLLR_DIV2;
PeriphClkInit.PLLSAI1.PLLSAI1ClockOut = RCC_PLLSAI1_SAI1CLK;
CHECK_HAL_ERROR_VOID(HAL_RCCEx_PeriphCLKConfig(&PeriphClkInit), MEMS_AUDIO_ERROR_SAI_CLOCK);
if (impl->SAI1_client == 0)
{
__HAL_RCC_SAI1_CLK_ENABLE();
}
impl->SAI1_client++;
/**SAI1_A_Block_A GPIO Configuration
PC3 ------> SAI1_D1
PA3 ------> SAI1_CK1
*/
GPIO_InitStruct.Pin = GPIO_PIN_3;
GPIO_InitStruct.Mode = GPIO_MODE_AF_PP;
GPIO_InitStruct.Pull = GPIO_NOPULL;
GPIO_InitStruct.Speed = GPIO_SPEED_FREQ_LOW;
GPIO_InitStruct.Alternate = GPIO_AF3_SAI1;
HAL_GPIO_Init(GPIOC, &GPIO_InitStruct);
GPIO_InitStruct.Pin = GPIO_PIN_3;
GPIO_InitStruct.Mode = GPIO_MODE_AF_PP;
GPIO_InitStruct.Pull = GPIO_NOPULL;
GPIO_InitStruct.Speed = GPIO_SPEED_FREQ_LOW;
GPIO_InitStruct.Alternate = GPIO_AF3_SAI1;
HAL_GPIO_Init(GPIOA, &GPIO_InitStruct);
/* Peripheral DMA init*/
DMA_HandleTypeDef hdma_sai1_a = {0};
hdma_sai1_a.Instance = MEMS_AUDIO_DMA_CHANNEL;
hdma_sai1_a.Init.Request = DMA_REQUEST_SAI1_A;
hdma_sai1_a.Init.Direction = DMA_PERIPH_TO_MEMORY;
hdma_sai1_a.Init.PeriphInc = DMA_PINC_DISABLE;
hdma_sai1_a.Init.MemInc = DMA_MINC_ENABLE;
hdma_sai1_a.Init.PeriphDataAlignment = DMA_PDATAALIGN_BYTE;
hdma_sai1_a.Init.MemDataAlignment = DMA_MDATAALIGN_BYTE;
hdma_sai1_a.Init.Mode = DMA_CIRCULAR;
hdma_sai1_a.Init.Priority = DMA_PRIORITY_HIGH;
impl->hdma_sai1_a = hdma_sai1_a;
CHECK_HAL_ERROR_VOID(HAL_DMA_Init(&impl->hdma_sai1_a), MEMS_AUDIO_ERROR_SAI_DMA_INIT);
/* Several peripheral DMA handle pointers point to the same DMA handle.
Be aware that there is only one channel to perform all the requested DMAs. */
__HAL_LINKDMA(hsai, hdmarx, impl->hdma_sai1_a);
__HAL_LINKDMA(hsai, hdmatx, impl->hdma_sai1_a);
}
}
void HAL_SAI_MspDeInit(SAI_HandleTypeDef *hsai)
{
/* SAI1 */
MemsAudio_STM32L4SAIPDM* impl = audioImpl;
if (hsai->Instance == SAI1_Block_A && impl)
{
impl->SAI1_client--;
if (impl->SAI1_client == 0)
{
/* Peripheral clock disable */
__HAL_RCC_SAI1_CLK_DISABLE();
}
/**SAI1_A_Block_A GPIO Configuration
PC3 ------> SAI1_D1
PA3 ------> SAI1_CK1
*/
HAL_GPIO_DeInit(GPIOC, GPIO_PIN_3);
HAL_GPIO_DeInit(GPIOA, GPIO_PIN_3);
/* SAI1 DMA Deinit */
HAL_DMA_DeInit(hsai->hdmarx);
HAL_DMA_DeInit(hsai->hdmatx);
}
}
/**
* @brief Initialize the DMA peripheral
*
*/
static mems_audio_err_t MX_DMA_Init(void)
{
/* DMA controller clock enable */
__HAL_RCC_DMAMUX1_CLK_ENABLE();
__HAL_RCC_DMA1_CLK_ENABLE();
/* DMA interrupt init */
/* DMA1_Channel1_IRQn interrupt configuration */
HAL_NVIC_SetPriority(MEMS_AUDIO_DMA_IRQn, MEMS_AUDIO_DMA_PRIORITY, 0);
HAL_NVIC_EnableIRQ(MEMS_AUDIO_DMA_IRQn);
return MEMS_AUDIO_OK;
}
static mems_audio_err_t MX_DMA_Uninit(void)
{
HAL_NVIC_DisableIRQ(MEMS_AUDIO_DMA_IRQn);
return MEMS_AUDIO_OK;
}
/**
* @brief Global handler for the DMA interrupt. Forwards to the HAL for further processing.
*
*/
void DMA_HANDLER(void)
{
HAL_DMA_IRQHandler(&audioImpl->hdma_sai1_a);
}
/**
* @brief Converts PDM samples in the upper half of the PDM buffer.
*
* @param hSai
*/
void HAL_SAI_RxHalfCpltCallback(SAI_HandleTypeDef *hSai)
{
(void)hSai;
pdm2pcm(audioImpl->pdmBuffer, audioImpl->pdmBufferLength>>1);
}
/**
* @brief Converts PDM samples in the upper half of the PDM buffer.
*
* @param hSai
*/
void HAL_SAI_RxCpltCallback(SAI_HandleTypeDef *hSai)
{
(void)hSai;
pdm2pcm(audioImpl->pdmBuffer+(audioImpl->pdmBufferLength>>1), audioImpl->pdmBufferLength>>1);
}
mems_audio_err_t mems_audio_init_stm32l4_sai_pdm(MemsAudio* audio, MemsAudio_STM32L4SAIPDM* impl)
{
uninit();
audioImpl = impl;
impl->audio = audio;
return mems_audio_init(audio);
}

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/*
* 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.
*/
#ifndef _MEMS_AUDIO_LL_STM32L4_H_
#define _MEMS_AUDIO_LL_STM32L4_H_
#include <stdbool.h>
#include <assert.h>
#include <stm32l4xx_hal.h>
#include "OpenPDMFilter.h"
#include "MEMS_Audio.h"
#include "MEMS_Audio_ll.h"
#ifdef __cplusplus
extern "C" {
#endif
/**
* @brief The SAI PDM interface captures 8 bits from the PDM signal.
*/
typedef uint8_t pdm_sample_t;
/**
* @brief The PDM sample frequency in kHz. (Bit samples per millisecond.)
*/
#define PDM_IN_FREQUENCY_KHZ 1024
#define PDM_IN_FREQUENCY (PDM_IN_FREQUENCY_KHZ * 1000)
/**
* @brief The number of channels of audio captured
*/
#define PDM_IN_CHANNELS 1
/**
* @brief The decimation of the PDM bit stream to produce PCM samples at the desired output rate.
*/
#define PDM_IN_DECIMATION_FACTOR 64
/**
* @brief The number of pdm samples captured per millisecond from the PDM interface.
*/
#define MEMS_AUDIO_PDM_SAMPLES_PER_MS ((PDM_IN_FREQUENCY_KHZ / (sizeof(pdm_sample_t) * 8)) * PDM_IN_CHANNELS)
/**
* @brief The size of the buffer used to hold PDM samples prior to conversion to PCM.
* Each half of the buffer generates an interrupt.
*/
#define MEMS_AUDIO_PDM_BUFFER_LENGTH (MEMS_AUDIO_PDM_SAMPLES_PER_MS * MEMS_AUDIO_MS_BUFFER * 2)
/**
* @brief The length of the PCM buffer required to hold converted samples.
*/
#define MEMS_AUDIO_PCM_BUFFER_LENGTH (PCM_OUT_SAMPLING_FREQUENCY * MEMS_AUDIO_MS_BUFFER / 1000)
/**
* Presently the internal PDM parameters and output PCM parameters are fixed for the values given here.
*/
/**
* @brief 128 point decimation did not work with the OpenPDMFilter and just produced PCM output
* approaching a square wave.
*/
_Static_assert(PDM_IN_DECIMATION_FACTOR == 64 || PDM_IN_DECIMATION_FACTOR == 128, "A decomation factor of 64 or 128 is supported at present.");
/**
* @brief The PDM bitstream frequency divided by the decimation factor should be the same as the desired output PCM frequency.
*/
_Static_assert(PDM_IN_FREQUENCY / PDM_IN_DECIMATION_FACTOR == PCM_OUT_SAMPLING_FREQUENCY, "PDM output frequency should equal the input frequency divided by the decimation factor.");
//
// SAI PDM interface clock configuration
//
#define MEMS_AUDIO_MSI_FREQUENCY (48 * 1000 * 1000)
#define MEMS_AUDIO_CLOCK_PLLM (15)
#define MEMS_AUDIO_CLOCK_PLLN (16)
#define MEMS_AUDIO_CLOCK_PLLP (RCC_PLLP_DIV25)
/**
* @brief The SAI PDM clock should be twice the desired PDM bitstream frequency
*/
_Static_assert((MEMS_AUDIO_MSI_FREQUENCY / MEMS_AUDIO_CLOCK_PLLM * MEMS_AUDIO_CLOCK_PLLN / MEMS_AUDIO_CLOCK_PLLP) == (PDM_IN_FREQUENCY_KHZ * 1000 * 2), "PDM clock should be twice the PDM sample frequency.");
typedef struct MemsAudio_STM32L4SAIPDM_t MemsAudio_STM32L4SAIPDM;
/**
* @brief Callback informing that PDM samples are available for processing.
* @param audio The MemsAudio instance
* @return `false` to skip conversion of PDM to PCM. `true` to convert the PDM samples to PCM.
*/
typedef bool (*pdm_data_available_t)(MemsAudio_STM32L4SAIPDM *audio, pdm_sample_t *pdmSamples, size_t pdmLength);
/**
* @brief Implementation details for the STM32 SAI PDM implementation.
*
*/
/**
* @brief Audio capture from a MEMS microphone on the STM32L4 using the SAI PDM interface.
*/
typedef struct MemsAudio_STM32L4SAIPDM_t {
MemsAudio *audio;
/**
* @brief The last error that happened in a void function (e.g. HAL callback)
*/
mems_audio_err_t lastError;
/**
* @brief The buffer to store PDM audio samples
*/
pdm_sample_t *pdmBuffer;
/**
* @brief The length of the PDM buffer. Should be at least MEMS_AUDIO_PDM_BUFFER_LENGTH
*/
size_t pdmBufferLength;
/**
* @brief Optional callback for when PDM data is available.
*/
pdm_data_available_t pdm_data_available;
/**
* @brief A cound of the number of PDM clients in use.
*/
uint32_t SAI1_client;
/**
* @brief The SAI peripheral handle being used for SAI A subclock 1.
*/
SAI_HandleTypeDef hSAI_BlockA1;
/**
* @brief The DMA handle to transfer SAI data from the peripheral to memory.
*/
DMA_HandleTypeDef hdma_sai1_a;
/**
* @brief An instance of the PDM filter that performs decimation, and high and low pass filtering.
* Unlike the DFSDM peripheral, the SAI PDM interface doesn't perform these operations in hardware.
*/
TPDMFilter_InitStruct filter;
/**
* @brief The number of DMA transfers to ignore after starting recording.
*/
volatile uint16_t discard_dma;
} MemsAudio_STM32L4SAIPDM;
/**
* @brief Creates a MemsAudio instance that retrieves PDM samples from SAI A block 1 via the PDM interface,
* decimates and filters these in software to produce the PCM output stream.
*
* @param audio
* @param implementation
* @return meems_audio_error_t
*/
mems_audio_err_t mems_audio_init_stm32l4_sai_pdm(MemsAudio *audio, MemsAudio_STM32L4SAIPDM *implementation);
/**
* @brief Implementation-specific error codes.
*
*/
typedef enum mems_audio_err_stm32l4_t {
MEMS_AUDIO_ERROR_SAI_DMA_INIT = 1,
MEMS_AUDIO_ERROR_SAI_CLOCK = 2,
MEMS_AUDIO_ERROR_SAI_INIT = 3,
MEMS_AUDIO_ERROR_SAI_DEINIT = 4,
MEMS_AUDIO_ERROR_DMA_START = 5,
MEMS_AUDIO_ERROR_DMA_STOP = 6,
MEMS_AUDIO_ERROR_DMA_PAUSE = 7,
MEMS_AUDIO_ERROR_DMA_RESUME = 8
} mems_audio_err_stm32l4_t;
#ifdef __cplusplus
}
#endif
#endif // _MEMS_AUDIO_LL_STM32L4_H_

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/**
*******************************************************************************
* @file OpenPDMFilter.c
* @author CL
* @version V1.0.0
* @date 9-September-2015
* @brief Open PDM audio software decoding Library.
* This Library is used to decode and reconstruct the audio signal
* produced by ST MEMS microphone (MP45Dxxx, MP34Dxxx).
*******************************************************************************
* @attention
*
* <h2><center>&copy; COPYRIGHT 2018 STMicroelectronics</center></h2>
*
* Licensed under the Apache License, Version 2.0 (the "License");
* you may not use this file except in compliance with the License.
* You may obtain a copy of the License at
*
* http://www.apache.org/licenses/LICENSE-2.0
*
* Unless required by applicable law or agreed to in writing, software
* distributed under the License is distributed on an "AS IS" BASIS,
* WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
* See the License for the specific language governing permissions and
* limitations under the License.
*******************************************************************************
*/
/* Includes ------------------------------------------------------------------*/
#include "OpenPDMFilter.h"
/* Functions -----------------------------------------------------------------*/
#ifdef USE_LUT
int32_t filter_table_mono_64(lut_t lut, uint8_t *data, uint8_t sincn) {
return (int32_t)
lut[data[0]][0][sincn] +
lut[data[1]][1][sincn] +
lut[data[2]][2][sincn] +
lut[data[3]][3][sincn] +
lut[data[4]][4][sincn] +
lut[data[5]][5][sincn] +
lut[data[6]][6][sincn] +
lut[data[7]][7][sincn];
}
int32_t filter_table_stereo_64(lut_t lut, uint8_t *data, uint8_t sincn) {
return (int32_t)
lut[data[0]][0][sincn] +
lut[data[2]][1][sincn] +
lut[data[4]][2][sincn] +
lut[data[6]][3][sincn] +
lut[data[8]][4][sincn] +
lut[data[10]][5][sincn] +
lut[data[12]][6][sincn] +
lut[data[14]][7][sincn];
}
#if DECIMATION_MAX == 128
int32_t filter_table_mono_128(lut_t lut, uint8_t *data, uint8_t sincn) {
return (int32_t)
lut[data[0]][0][sincn] +
lut[data[1]][1][sincn] +
lut[data[2]][2][sincn] +
lut[data[3]][3][sincn] +
lut[data[4]][4][sincn] +
lut[data[5]][5][sincn] +
lut[data[6]][6][sincn] +
lut[data[7]][7][sincn] +
lut[data[8]][8][sincn] +
lut[data[9]][9][sincn] +
lut[data[10]][10][sincn] +
lut[data[11]][11][sincn] +
lut[data[12]][12][sincn] +
lut[data[13]][13][sincn] +
lut[data[14]][14][sincn] +
lut[data[15]][15][sincn];
}
int32_t filter_table_stereo_128(lut_t lut, uint8_t *data, uint8_t sincn) {
return (int32_t)
lut[data[0]][0][sincn] +
lut[data[2]][1][sincn] +
lut[data[4]][2][sincn] +
lut[data[6]][3][sincn] +
lut[data[8]][4][sincn] +
lut[data[10]][5][sincn] +
lut[data[12]][6][sincn] +
lut[data[14]][7][sincn] +
lut[data[16]][8][sincn] +
lut[data[18]][9][sincn] +
lut[data[20]][10][sincn] +
lut[data[22]][11][sincn] +
lut[data[24]][12][sincn] +
lut[data[26]][13][sincn] +
lut[data[28]][14][sincn] +
lut[data[30]][15][sincn];
}
#endif
int32_t (*filter_tables_64[2])(lut_t lut, uint8_t *data, uint8_t sincn) = {filter_table_mono_64, filter_table_stereo_64};
#if DECIMATION_MAX == 128
int32_t (*filter_tables_128[2])(lut_t lut, uint8_t *data, uint8_t sincn) = {filter_table_mono_128, filter_table_stereo_128};
#endif
#else
int32_t filter_table(uint8_t *data, uint8_t sincn, TPDMFilter_InitStruct *param) {
uint8_t c, i;
uint16_t data_index = 0;
uint32_t *coef_p = &param->coef[sincn][0];
int32_t F = 0;
uint8_t decimation = param->Decimation;
uint8_t channels = param->In_MicChannels;
for (i = 0; i < decimation; i += 8) {
c = data[data_index];
F += ((c >> 7)) * coef_p[i ] +
((c >> 6) & 0x01) * coef_p[i + 1] +
((c >> 5) & 0x01) * coef_p[i + 2] +
((c >> 4) & 0x01) * coef_p[i + 3] +
((c >> 3) & 0x01) * coef_p[i + 4] +
((c >> 2) & 0x01) * coef_p[i + 5] +
((c >> 1) & 0x01) * coef_p[i + 6] +
((c) & 0x01) * coef_p[i + 7];
data_index += channels;
}
return F;
}
#endif
void convolve(uint32_t Signal[] /* SignalLen */, unsigned short SignalLen,
uint32_t Kernel[] /* KernelLen */, unsigned short KernelLen,
uint32_t Result[] /* SignalLen + KernelLen - 1 */) {
uint16_t n;
for (n = 0; n < SignalLen + KernelLen - 1; n++)
{
unsigned short kmin, kmax, k;
Result[n] = 0;
kmin = (n >= KernelLen - 1) ? n - (KernelLen - 1) : 0;
kmax = (n < SignalLen - 1) ? n : SignalLen - 1;
for (k = kmin; k <= kmax; k++) {
Result[n] += Signal[k] * Kernel[n - k];
}
}
}
void Open_PDM_Filter_Init(TPDMFilter_InitStruct *Param) {
uint16_t i, j;
int64_t sum = 0;
uint8_t decimation = Param->Decimation;
for (i = 0; i < SINCN; i++) {
Param->Coef[i] = 0;
Param->bit[i] = 0;
}
for (i = 0; i < decimation; i++) {
Param->sinc1[i] = 1;
}
Param->OldOut = Param->OldIn = Param->OldZ = 0;
Param->LP_ALFA = (Param->LP_HZ != 0 ? (uint16_t)((float)Param->LP_HZ * 256 / (Param->LP_HZ + Param->Fs / (2 * 3.14159))) : 0);
Param->HP_ALFA = (Param->HP_HZ != 0 ? (uint16_t)((float)Param->Fs * 256 / (2 * 3.14159 * Param->HP_HZ + Param->Fs)) : 0);
Param->FilterLen = decimation * SINCN;
Param->sinc[0] = 0;
Param->sinc[decimation * SINCN - 1] = 0;
convolve(Param->sinc1, decimation, Param->sinc1, decimation, Param->sinc2);
convolve(Param->sinc2, (decimation << 1) - 1, Param->sinc1, decimation, &Param->sinc[1]);
for (j = 0; j < SINCN; j++) {
for (i = 0; i < decimation; i++) {
Param->coef[j][i] = Param->sinc[j * decimation + i];
sum += Param->sinc[j * decimation + i];
}
}
Param->sub_const = sum >> 1;
uint32_t div_const = Param->sub_const * Param->MaxVolume / 32768 / FILTER_GAIN;
Param->div_const = (div_const == 0 ? 1 : div_const);
#ifdef USE_LUT
/* Look-Up Table. */
uint16_t c, d, s;
for (s = 0; s < SINCN; s++)
{
uint32_t *coef_p = &Param->coef[s][0];
for (c = 0; c < 256; c++) {
for (d = 0; d < decimation / 8; d++) {
Param->lut[c][d][s] = ((c >> 7)) * coef_p[d * 8 ] +
((c >> 6) & 0x01) * coef_p[d * 8 + 1] +
((c >> 5) & 0x01) * coef_p[d * 8 + 2] +
((c >> 4) & 0x01) * coef_p[d * 8 + 3] +
((c >> 3) & 0x01) * coef_p[d * 8 + 4] +
((c >> 2) & 0x01) * coef_p[d * 8 + 5] +
((c >> 1) & 0x01) * coef_p[d * 8 + 6] +
((c) & 0x01) * coef_p[d * 8 + 7];
}
}
}
#endif
}
int Open_PDM_Filter_64(uint8_t *data, int16_t *dataOut, uint16_t volume, TPDMFilter_InitStruct *Param) {
uint8_t i, data_out_index;
uint8_t channels = Param->In_MicChannels;
uint8_t data_inc = ((DECIMATION_MAX >> 4) * channels);
int64_t Z, Z0, Z1, Z2;
int64_t OldOut, OldIn, OldZ;
OldOut = Param->OldOut;
OldIn = Param->OldIn;
OldZ = Param->OldZ;
#ifdef USE_LUT
uint8_t j = channels - 1;
#endif
for (i = 0, data_out_index = 0; i < Param->nSamples; i++, data_out_index += channels) {
#ifdef USE_LUT
Z0 = filter_tables_64[j](Param->lut, data, 0);
Z1 = filter_tables_64[j](Param->lut, data, 1);
Z2 = filter_tables_64[j](Param->lut, data, 2);
#else
Z0 = filter_table(data, 0, Param);
Z1 = filter_table(data, 1, Param);
Z2 = filter_table(data, 2, Param);
#endif
Z = Param->Coef[1] + Z2 - Param->sub_const;
Param->Coef[1] = Param->Coef[0] + Z1;
Param->Coef[0] = Z0;
OldOut = (Param->HP_ALFA * (OldOut + Z - OldIn)) >> 8;
OldIn = Z;
OldZ = ((256 - Param->LP_ALFA) * OldZ + Param->LP_ALFA * OldOut) >> 8;
Z = OldZ * volume;
Z = RoundDiv(Z, (Param->div_const));
Z = SaturaLH(Z, -32700, 32700);
dataOut[data_out_index] = Z;
data += data_inc;
}
Param->OldOut = OldOut;
Param->OldIn = OldIn;
Param->OldZ = OldZ;
return data_out_index;
}
#if DECIMATION_MAX == 128
int Open_PDM_Filter_128(uint8_t *data, int16_t *dataOut, uint16_t volume, TPDMFilter_InitStruct *Param) {
uint8_t i, data_out_index;
uint8_t channels = Param->In_MicChannels;
uint8_t data_inc = ((DECIMATION_MAX >> 3) * channels);
int64_t Z, Z0, Z1, Z2;
int64_t OldOut, OldIn, OldZ;
OldOut = Param->OldOut;
OldIn = Param->OldIn;
OldZ = Param->OldZ;
#ifdef USE_LUT
uint8_t j = channels - 1;
#endif
for (i = 0, data_out_index = 0; i < Param->nSamples; i++, data_out_index += channels) {
#ifdef USE_LUT
Z0 = filter_tables_128[j](Param->lut, data, 0);
Z1 = filter_tables_128[j](Param->lut, data, 1);
Z2 = filter_tables_128[j](Param->lut, data, 2);
#else
Z0 = filter_table(data, 0, Param);
Z1 = filter_table(data, 1, Param);
Z2 = filter_table(data, 2, Param);
#endif
Z = Param->Coef[1] + Z2 - Param->sub_const;
Param->Coef[1] = Param->Coef[0] + Z1;
Param->Coef[0] = Z0;
OldOut = (Param->HP_ALFA * (OldOut + Z - OldIn)) >> 8;
OldIn = Z;
OldZ = ((256 - Param->LP_ALFA) * OldZ + Param->LP_ALFA * OldOut) >> 8;
Z = OldZ * volume;
Z = RoundDiv(Z, (Param->div_const));
Z = SaturaLH(Z, -32700, 32700);
dataOut[data_out_index] = Z;
data += data_inc;
}
Param->OldOut = OldOut;
Param->OldIn = OldIn;
Param->OldZ = OldZ;
return data_out_index;
}
#endif

112
ports/stm/common-hal/audiobusio/OpenPDMFilter.h Normal file
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/**
*******************************************************************************
* @file OpenPDMFilter.h
* @author CL
* @version V1.0.0
* @date 9-September-2015
* @brief Header file for Open PDM audio software decoding Library.
* This Library is used to decode and reconstruct the audio signal
* produced by ST MEMS microphone (MP45Dxxx, MP34Dxxx).
*******************************************************************************
* @attention
*
* <h2><center>&copy; COPYRIGHT 2018 STMicroelectronics</center></h2>
*
* Licensed under the Apache License, Version 2.0 (the "License");
* you may not use this file except in compliance with the License.
* You may obtain a copy of the License at
*
* http://www.apache.org/licenses/LICENSE-2.0
*
* Unless required by applicable law or agreed to in writing, software
* distributed under the License is distributed on an "AS IS" BASIS,
* WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
* See the License for the specific language governing permissions and
* limitations under the License.
*******************************************************************************
*/
/* Define to prevent recursive inclusion -------------------------------------*/
#ifndef __OPENPDMFILTER_H
#define __OPENPDMFILTER_H
#ifdef __cplusplus
extern "C" {
#endif
/* Includes ------------------------------------------------------------------*/
#include <stdint.h>
/* Definitions ---------------------------------------------------------------*/
/*
* Enable to use a Look-Up Table to improve performances while using more FLASH
* and RAM memory.
* Note: Without Look-Up Table up to stereo@16KHz configuration is supported.
*/
#define USE_LUT
#define SINCN 3
#define DECIMATION_MAX 128 // can be 128 but this didn't work for me
#define FILTER_GAIN 16
#define HTONS(A) ((((uint16_t)(A) & 0xff00) >> 8) | \
(((uint16_t)(A) & 0x00ff) << 8))
#define RoundDiv(a, b) (((a) > 0) ? (((a) + (b) / 2) / (b)) : (((a) - (b) / 2) / (b)))
#define SaturaLH(N, L, H) (((N) < (L)) ? (L) : (((N) > (H)) ? (H) : (N)))
/* Types ---------------------------------------------------------------------*/
typedef int32_t lut_t[256][DECIMATION_MAX / 8][SINCN];
typedef struct
{
/* Public */
uint16_t LP_HZ;
uint16_t HP_HZ;
uint16_t Fs;
unsigned int nSamples;
uint8_t In_MicChannels;
uint8_t Out_MicChannels;
uint8_t Decimation;
uint8_t MaxVolume;
/* Private */
uint32_t Coef[SINCN];
uint16_t FilterLen;
int64_t OldOut, OldIn, OldZ;
uint16_t LP_ALFA;
uint16_t HP_ALFA;
uint16_t bit[5];
uint16_t byte;
uint32_t div_const;
int64_t sub_const;
uint32_t sinc[DECIMATION_MAX * SINCN];
uint32_t sinc1[DECIMATION_MAX];
uint32_t sinc2[DECIMATION_MAX * 2];
uint32_t coef[SINCN][DECIMATION_MAX];
#ifdef USE_LUT
lut_t lut;
#endif
} TPDMFilter_InitStruct;
/* Exported functions ------------------------------------------------------- */
void Open_PDM_Filter_Init(TPDMFilter_InitStruct *init_struct);
int Open_PDM_Filter_64(uint8_t *data, int16_t *data_out, uint16_t mic_gain, TPDMFilter_InitStruct *init_struct);
#if DECIMATION_MAX == 128
int Open_PDM_Filter_128(uint8_t *data, int16_t *data_out, uint16_t mic_gain, TPDMFilter_InitStruct *init_struct);
#endif
#ifdef __cplusplus
}
#endif
#endif // __OPENPDMFILTER_H
/************************ (C) COPYRIGHT STMicroelectronics *****END OF FILE****/

199
ports/stm/common-hal/audiobusio/PDMIn.c Normal file
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/*
* 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 "supervisor/memory.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(translate("only mono is supported"));
}
if (sample_rate != 16000) {
mp_raise_ValueError(translate("only sample_rate=16000 is supported"));
}
if (bit_depth != 16) {
mp_raise_ValueError(translate("only bit_depth=16 is supported"));
}
if (oversample != 64) {
mp_raise_ValueError(translate("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;
}

56
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/*
* 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.
*/
#ifndef MICROPY_INCLUDED_STM_COMMON_HAL_AUDIOBUSIO_AUDIOOUT_H
#define MICROPY_INCLUDED_STM_COMMON_HAL_AUDIOBUSIO_AUDIOOUT_H
#include <stdint.h>
#include "py/obj.h"
#include "peripherals/pins.h"
#include "supervisor/memory.h"
typedef struct MemsAudio_t MemsAudio;
typedef struct MemsAudio_STM32L4SAIPDM_t MemsAudio_STM32L4SAIPDM;
typedef struct {
mp_obj_base_t base;
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;
supervisor_allocation *audio_allocation;
MemsAudio *audio;
MemsAudio_STM32L4SAIPDM *audio_impl;
/**
* @brief Flag to indicate from the ISR that recording is complete.
*/
volatile bool recording_complete;
} audiobusio_pdmin_obj_t;
#endif

0
ports/stm/common-hal/audiobusio/__init__.c Normal file
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3
supervisor/shared/memory.c
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@ -58,6 +58,9 @@ enum {
#endif
+ CIRCUITPY_PORT_NUM_SUPERVISOR_ALLOCATIONS
#if CIRCUITPY_AUDIOBUSIO_PDMIN
+ 1
#endif
,
CIRCUITPY_SUPERVISOR_MOVABLE_ALLOC_COUNT =