Merge pull request #955 from openmv/audio_nonblocking

Make audio module non-blocking.
This commit is contained in:
Ibrahim Abd Elkader 2020-11-08 22:40:36 +02:00 committed by GitHub
commit 6e3a083621
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GPG Key ID: 4AEE18F83AFDEB23
5 changed files with 156 additions and 184 deletions

@ -1 +1 @@
Subproject commit d2880e8d39c41ba163ef1cc274578b47d7b1124a
Subproject commit a0d393a0627a7fd4825a3ae8f14c7eb5e2a70de5

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@ -70,6 +70,7 @@
#include "py_fir.h"
#include "py_tv.h"
#include "py_imu.h"
#include "py_audio.h"
#include "framebuffer.h"
@ -732,7 +733,9 @@ soft_reset:
can_deinit_all();
#endif
pyb_thread_deinit();
#if MICROPY_PY_AUDIO
py_audio_deinit();
#endif
first_soft_reset = false;
goto soft_reset;
}

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@ -9,6 +9,7 @@
* Audio Python module.
*/
#include <mp.h>
#include "py_audio.h"
#include "systick.h"
#include "py_assert.h"
#include "py_helper.h"
@ -28,7 +29,10 @@ static CRC_HandleTypeDef hcrc;
static SAI_HandleTypeDef hsai;
static DMA_HandleTypeDef hdma_sai_rx;
static int n_channels = AUDIO_SAI_NBR_CHANNELS;
static mp_obj_array_t *g_pcmbuf = NULL;
static mp_obj_t g_audio_callback = mp_const_none;
static int g_channels = AUDIO_SAI_NBR_CHANNELS;
static PDM_Filter_Handler_t PDM_FilterHandler[2];
static PDM_Filter_Config_t PDM_FilterConfig[2];
@ -41,6 +45,9 @@ static volatile uint32_t xfer_status = 0;
// BDMA can only access D3 SRAM4 memory.
uint8_t PDM_BUFFER[PDM_BUFFER_SIZE] __attribute__ ((aligned (32))) __attribute__((section(".d3_sram_buffer")));
// Pendsv dispatch callback.
static void audio_pendsv_callback(void);
void AUDIO_SAI_DMA_IRQHandler(void)
{
HAL_DMA_IRQHandler(hsai.hdmarx);
@ -50,26 +57,62 @@ void HAL_SAI_RxHalfCpltCallback(SAI_HandleTypeDef *hsai)
{
xfer_status |= DMA_XFER_HALF;
SCB_InvalidateDCache_by_Addr((uint32_t *)(&PDM_BUFFER[0]), PDM_BUFFER_SIZE / 2);
if (g_audio_callback != mp_const_none) {
pendsv_schedule_dispatch(PENDSV_DISPATCH_AUDIO, audio_pendsv_callback);
}
}
void HAL_SAI_RxCpltCallback(SAI_HandleTypeDef *hsai)
{
xfer_status |= DMA_XFER_FULL;
SCB_InvalidateDCache_by_Addr((uint32_t *)(&PDM_BUFFER[PDM_BUFFER_SIZE / 2]), PDM_BUFFER_SIZE / 2);
if (g_audio_callback != mp_const_none) {
pendsv_schedule_dispatch(PENDSV_DISPATCH_AUDIO, audio_pendsv_callback);
}
}
static uint32_t get_decimation_factor(uint32_t decimation)
{
switch (decimation) {
case 16: return PDM_FILTER_DEC_FACTOR_16;
case 24: return PDM_FILTER_DEC_FACTOR_24;
case 32: return PDM_FILTER_DEC_FACTOR_32;
case 48: return PDM_FILTER_DEC_FACTOR_48;
case 64: return PDM_FILTER_DEC_FACTOR_64;
case 80: return PDM_FILTER_DEC_FACTOR_80;
case 128: return PDM_FILTER_DEC_FACTOR_128;
default: return 0;
}
}
static mp_obj_t py_audio_init(uint n_args, const mp_obj_t *args, mp_map_t *kw_args)
{
n_channels = py_helper_keyword_int(n_args, args, 0, kw_args, MP_OBJ_NEW_QSTR(MP_QSTR_channels), 2);
// Read Args.
g_channels = py_helper_keyword_int(n_args, args, 0, kw_args, MP_OBJ_NEW_QSTR(MP_QSTR_channels), 2);
uint32_t frequency = py_helper_keyword_int(n_args, args, 1, kw_args, MP_OBJ_NEW_QSTR(MP_QSTR_frequency), 16000);
int gain_db = py_helper_keyword_int(n_args, args, 2, kw_args, MP_OBJ_NEW_QSTR(MP_QSTR_gain_db), 24);
float highpass = py_helper_keyword_float(n_args, args, 3, kw_args, MP_OBJ_NEW_QSTR(MP_QSTR_highpass), 0.9883f);
if (n_channels != 1 && n_channels != 2) {
// Sanity checks
if (frequency < 16000 || frequency > 128000) {
RAISE_OS_EXCEPTION("Invalid frequency!");
}
if (g_channels != 1 && g_channels != 2) {
RAISE_OS_EXCEPTION("Invalid number of channels! Expected 1 or 2.");
}
uint32_t decimation_factor = AUDIO_SAI_FREQKHZ / (frequency / 1000);
uint32_t decimation_factor_const = get_decimation_factor(decimation_factor);
if (decimation_factor_const == 0) {
RAISE_OS_EXCEPTION("This frequency is not supported!");
}
uint32_t samples_per_channel = (PDM_BUFFER_SIZE * 8) / (decimation_factor * g_channels * 2); // Half a transfer
hsai.Instance = AUDIO_SAI;
hsai.Init.Protocol = SAI_FREE_PROTOCOL;
hsai.Init.AudioMode = SAI_MODEMASTER_RX;
hsai.Init.DataSize = (n_channels == 1) ? SAI_DATASIZE_8 : SAI_DATASIZE_16;
hsai.Init.DataSize = (g_channels == 1) ? SAI_DATASIZE_8 : SAI_DATASIZE_16;
hsai.Init.FirstBit = SAI_FIRSTBIT_LSB;
hsai.Init.ClockStrobing = SAI_CLOCKSTROBING_RISINGEDGE;
hsai.Init.Synchro = SAI_ASYNCHRONOUS;
@ -78,7 +121,7 @@ static mp_obj_t py_audio_init(uint n_args, const mp_obj_t *args, mp_map_t *kw_ar
hsai.Init.FIFOThreshold = SAI_FIFOTHRESHOLD_1QF;
hsai.Init.SynchroExt = SAI_SYNCEXT_DISABLE;
hsai.Init.AudioFrequency = SAI_AUDIO_FREQUENCY_MCKDIV;
hsai.Init.MonoStereoMode = (n_channels == 1) ? SAI_MONOMODE: SAI_STEREOMODE;
hsai.Init.MonoStereoMode = (g_channels == 1) ? SAI_MONOMODE: SAI_STEREOMODE;
hsai.Init.CompandingMode = SAI_NOCOMPANDING;
hsai.Init.TriState = SAI_OUTPUT_RELEASED;
@ -101,8 +144,8 @@ static mp_obj_t py_audio_init(uint n_args, const mp_obj_t *args, mp_map_t *kw_ar
hsai.SlotInit.FirstBitOffset = 0;
hsai.SlotInit.SlotSize = SAI_SLOTSIZE_DATASIZE;
hsai.SlotInit.SlotNumber = (n_channels == 1) ? 2 : 1;
hsai.SlotInit.SlotActive = (n_channels == 1) ? (SAI_SLOTACTIVE_0 | SAI_SLOTACTIVE_1) : SAI_SLOTACTIVE_0;
hsai.SlotInit.SlotNumber = (g_channels == 1) ? 2 : 1;
hsai.SlotInit.SlotActive = (g_channels == 1) ? (SAI_SLOTACTIVE_0 | SAI_SLOTACTIVE_1) : SAI_SLOTACTIVE_0;
// Initialize the SAI
HAL_SAI_DeInit(&hsai);
@ -119,8 +162,8 @@ static mp_obj_t py_audio_init(uint n_args, const mp_obj_t *args, mp_map_t *kw_ar
hdma_sai_rx.Init.Direction = DMA_PERIPH_TO_MEMORY;
hdma_sai_rx.Init.PeriphInc = DMA_PINC_DISABLE;
hdma_sai_rx.Init.MemInc = DMA_MINC_ENABLE;
hdma_sai_rx.Init.PeriphDataAlignment = (n_channels == 1) ? DMA_PDATAALIGN_BYTE : DMA_PDATAALIGN_HALFWORD;
hdma_sai_rx.Init.MemDataAlignment = (n_channels == 1) ? DMA_MDATAALIGN_BYTE : DMA_MDATAALIGN_HALFWORD;
hdma_sai_rx.Init.PeriphDataAlignment = (g_channels == 1) ? DMA_PDATAALIGN_BYTE : DMA_PDATAALIGN_HALFWORD;
hdma_sai_rx.Init.MemDataAlignment = (g_channels == 1) ? DMA_MDATAALIGN_BYTE : DMA_MDATAALIGN_HALFWORD;
hdma_sai_rx.Init.Mode = DMA_CIRCULAR;
hdma_sai_rx.Init.Priority = DMA_PRIORITY_HIGH;
hdma_sai_rx.Init.FIFOMode = DMA_FIFOMODE_ENABLE;
@ -151,9 +194,54 @@ static mp_obj_t py_audio_init(uint n_args, const mp_obj_t *args, mp_map_t *kw_ar
}
__HAL_CRC_DR_RESET(&hcrc);
// Configure PDM filters
for (int i=0; i<g_channels; i++) {
PDM_FilterHandler[i].bit_order = PDM_FILTER_BIT_ORDER_MSB;
PDM_FilterHandler[i].endianness = PDM_FILTER_ENDIANNESS_LE;
PDM_FilterHandler[i].high_pass_tap = (uint32_t) (highpass * 2147483647U); // coff * (2^31-1)
PDM_FilterHandler[i].out_ptr_channels = g_channels;
PDM_FilterHandler[i].in_ptr_channels = g_channels;
PDM_Filter_Init(&PDM_FilterHandler[i]);
PDM_FilterConfig[i].mic_gain = gain_db;
PDM_FilterConfig[i].output_samples_number = samples_per_channel;
PDM_FilterConfig[i].decimation_factor = decimation_factor_const;
PDM_Filter_setConfig(&PDM_FilterHandler[i], &PDM_FilterConfig[i]);
}
// Allocate global PCM buffer.
g_pcmbuf = mp_obj_new_bytearray_by_ref(
samples_per_channel * g_channels * sizeof(int16_t),
m_new(int16_t, samples_per_channel * g_channels));
return mp_const_none;
}
void py_audio_deinit()
{
// Stop SAI DMA.
if (hdma_sai_rx.Instance != NULL) {
HAL_SAI_DMAStop(&hsai);
}
// Disable IRQs
HAL_NVIC_DisableIRQ(AUDIO_SAI_DMA_IRQ);
if (hsai.Instance != NULL) {
HAL_SAI_DeInit(&hsai);
hsai.Instance = NULL;
}
if (hdma_sai_rx.Instance != NULL) {
HAL_DMA_DeInit(&hdma_sai_rx);
hdma_sai_rx.Instance = NULL;
}
g_channels = 0;
g_pcmbuf = NULL;
g_audio_callback = mp_const_none;
}
static mp_obj_t py_audio_read_pdm(mp_obj_t buf_in)
{
mp_buffer_info_t pdmbuf;
@ -163,7 +251,7 @@ static mp_obj_t py_audio_read_pdm(mp_obj_t buf_in)
// Note: samples are copied as bytes for 1 and 2 channels.
uint32_t n_samples = pdmbuf.len;
if (typesize != n_channels) {
if (typesize != g_channels) {
// Make sure the buffer type matches the number of channels.
RAISE_OS_EXCEPTION("Buffer data type does not match the number of channels!");
}
@ -172,7 +260,7 @@ static mp_obj_t py_audio_read_pdm(mp_obj_t buf_in)
xfer_status &= DMA_XFER_NONE;
// Start DMA transfer
if (HAL_SAI_Receive_DMA(&hsai, (uint8_t*) PDM_BUFFER, PDM_BUFFER_SIZE / n_channels) != HAL_OK) {
if (HAL_SAI_Receive_DMA(&hsai, (uint8_t*) PDM_BUFFER, PDM_BUFFER_SIZE / g_channels) != HAL_OK) {
RAISE_OS_EXCEPTION("SAI DMA transfer failed!");
}
@ -207,203 +295,70 @@ static mp_obj_t py_audio_read_pdm(mp_obj_t buf_in)
return mp_const_none;
}
static uint32_t get_decimation_factor(uint32_t decimation)
static void audio_pendsv_callback(void)
{
switch (decimation) {
case 16: return PDM_FILTER_DEC_FACTOR_16;
case 24: return PDM_FILTER_DEC_FACTOR_24;
case 32: return PDM_FILTER_DEC_FACTOR_32;
case 48: return PDM_FILTER_DEC_FACTOR_48;
case 64: return PDM_FILTER_DEC_FACTOR_64;
case 80: return PDM_FILTER_DEC_FACTOR_80;
case 128: return PDM_FILTER_DEC_FACTOR_128;
default: RAISE_OS_EXCEPTION("This frequency is not supported!");
// Check for half transfer complete.
if ((xfer_status & DMA_XFER_HALF)) {
// Clear buffer state.
xfer_status &= ~(DMA_XFER_HALF);
// Convert PDM samples to PCM.
for (int i=0; i<g_channels; i++) {
PDM_Filter(&((uint8_t*)PDM_BUFFER)[i], &((int16_t*)g_pcmbuf->items)[i], &PDM_FilterHandler[i]);
}
} else if ((xfer_status & DMA_XFER_FULL)) { // Check for transfer complete.
// Clear buffer state.
xfer_status &= ~(DMA_XFER_FULL);
// Convert PDM samples to PCM.
for (int i=0; i<g_channels; i++) {
PDM_Filter(&((uint8_t*)PDM_BUFFER)[PDM_BUFFER_SIZE / 2 + i], &((int16_t*)g_pcmbuf->items)[i], &PDM_FilterHandler[i]);
}
}
// Call user callback
mp_call_function_1(g_audio_callback, MP_OBJ_FROM_PTR(g_pcmbuf));
}
static mp_obj_t py_audio_read_pcm(uint n_args, const mp_obj_t *args, mp_map_t *kw_args)
static mp_obj_t py_audio_start_streaming(mp_obj_t callback_obj)
{
mp_buffer_info_t pcmbuf;
mp_get_buffer_raise(args[0], &pcmbuf, MP_BUFFER_WRITE);
uint32_t frequency = py_helper_keyword_int(n_args, args, 1, kw_args, MP_OBJ_NEW_QSTR(MP_QSTR_frequency), 16);
int gain_db = py_helper_keyword_int(n_args, args, 2, kw_args, MP_OBJ_NEW_QSTR(MP_QSTR_gain_db), 24);
float highpass = py_helper_keyword_float(n_args, args, 3, kw_args, MP_OBJ_NEW_QSTR(MP_QSTR_highpass), 0.9883f);
g_audio_callback = callback_obj;
int16_t *output_buffer = (int16_t *) pcmbuf.buf;
uint32_t decimation_factor = AUDIO_SAI_FREQKHZ/frequency;
uint32_t output_samples = ((PDM_BUFFER_SIZE / 2) * 8) / (decimation_factor * n_channels); // Half transfer
size_t typesize = mp_binary_get_size('@', pcmbuf.typecode, NULL);
// Round down the number of samples to even buffer multiples.
uint32_t n_samples = (((pcmbuf.len / typesize) / output_samples) & (~1U)) * output_samples;
if (typesize != 2) {
// Make sure the buffer is 16-Bits array for PCM samples.
RAISE_OS_EXCEPTION("Wrong data type, expected 16-Bits array!");
}
// Configure PDM library
for (int i=0; i<n_channels; i++) {
PDM_FilterHandler[i].bit_order = PDM_FILTER_BIT_ORDER_MSB;
PDM_FilterHandler[i].endianness = PDM_FILTER_ENDIANNESS_LE;
PDM_FilterHandler[i].high_pass_tap = (uint32_t) (highpass * 2147483647U); // coff * (2^31-1)
PDM_FilterHandler[i].out_ptr_channels = n_channels;
PDM_FilterHandler[i].in_ptr_channels = n_channels;
PDM_Filter_Init(&PDM_FilterHandler[i]);
PDM_FilterConfig[i].mic_gain = gain_db;
PDM_FilterConfig[i].output_samples_number = output_samples;
PDM_FilterConfig[i].decimation_factor = get_decimation_factor(decimation_factor);
PDM_Filter_setConfig(&PDM_FilterHandler[i], &PDM_FilterConfig[i]);
if (!mp_obj_is_callable(g_audio_callback)) {
g_audio_callback = mp_const_none;
RAISE_OS_EXCEPTION("Invalid callback object!");
}
// Clear DMA buffer status
xfer_status &= DMA_XFER_NONE;
// Start DMA transfer
if (HAL_SAI_Receive_DMA(&hsai, (uint8_t*) PDM_BUFFER, PDM_BUFFER_SIZE / n_channels) != HAL_OK) {
if (HAL_SAI_Receive_DMA(&hsai, (uint8_t*) PDM_BUFFER, PDM_BUFFER_SIZE / g_channels) != HAL_OK) {
g_audio_callback = mp_const_none;
RAISE_OS_EXCEPTION("SAI DMA transfer failed!");
}
while (n_samples) {
uint32_t start = HAL_GetTick();
// Wait for half transfer complete.
while ((xfer_status & DMA_XFER_HALF) == 0) {
if ((HAL_GetTick() - start) >= 1000) {
HAL_SAI_DMAStop(&hsai);
RAISE_OS_EXCEPTION("SAI DMA transfer timeout!");
}
}
// Clear buffer state.
xfer_status &= ~(DMA_XFER_HALF);
// Convert PDM samples to PCM.
for (int i=0; i<n_channels; i++) {
PDM_Filter(&((uint8_t*)PDM_BUFFER)[i], &output_buffer[i], &PDM_FilterHandler[i]);
}
output_buffer += output_samples * n_channels;
// Wait for transfer complete.
while ((xfer_status & DMA_XFER_FULL) == 0) {
if ((HAL_GetTick() - start) >= 1000) {
HAL_SAI_DMAStop(&hsai);
RAISE_OS_EXCEPTION("SAI DMA transfer timeout!");
}
}
// Clear buffer state.
xfer_status &= ~(DMA_XFER_FULL);
// Convert PDM samples to PCM.
for (int i=0; i<n_channels; i++) {
PDM_Filter(&((uint8_t*)PDM_BUFFER)[PDM_BUFFER_SIZE / 2 + i], &output_buffer[i], &PDM_FilterHandler[i]);
}
output_buffer += output_samples * n_channels;
n_samples -= output_samples * n_channels * 2;
}
// Stop SAI DMA.
HAL_SAI_DMAStop(&hsai);
return mp_const_none;
}
static mp_obj_t py_audio_start_streaming(uint n_args, const mp_obj_t *args, mp_map_t *kw_args)
static mp_obj_t py_audio_stop_streaming()
{
mp_obj_t callback = args[0];
mp_obj_t arg_obj = args[1];
uint32_t time = (uint32_t) (py_helper_keyword_float(n_args, args, 2, kw_args, MP_OBJ_NEW_QSTR(MP_QSTR_time), 1.0) * 1000);
uint32_t frequency = py_helper_keyword_int(n_args, args, 3, kw_args, MP_OBJ_NEW_QSTR(MP_QSTR_frequency), 16);
int gain_db = py_helper_keyword_int(n_args, args, 4, kw_args, MP_OBJ_NEW_QSTR(MP_QSTR_gain_db), 24);
float highpass = py_helper_keyword_float(n_args, args, 5, kw_args, MP_OBJ_NEW_QSTR(MP_QSTR_highpass), 0.9883f);
uint32_t decimation_factor = AUDIO_SAI_FREQKHZ/frequency;
uint32_t output_samples = ((PDM_BUFFER_SIZE / 2) * 8) / (decimation_factor * n_channels); // Half transfer
mp_obj_array_t *pcmbuf = mp_obj_new_bytearray_by_ref(
output_samples * n_channels * sizeof(int16_t), m_new(int16_t, output_samples * n_channels));
// Configure PDM library
for (int i=0; i<n_channels; i++) {
PDM_FilterHandler[i].bit_order = PDM_FILTER_BIT_ORDER_MSB;
PDM_FilterHandler[i].endianness = PDM_FILTER_ENDIANNESS_LE;
PDM_FilterHandler[i].high_pass_tap = (uint32_t) (highpass * 2147483647U); // coff * (2^31-1)
PDM_FilterHandler[i].out_ptr_channels = n_channels;
PDM_FilterHandler[i].in_ptr_channels = n_channels;
PDM_Filter_Init(&PDM_FilterHandler[i]);
PDM_FilterConfig[i].mic_gain = gain_db;
PDM_FilterConfig[i].output_samples_number = output_samples;
PDM_FilterConfig[i].decimation_factor = get_decimation_factor(decimation_factor);
PDM_Filter_setConfig(&PDM_FilterHandler[i], &PDM_FilterConfig[i]);
}
// Clear DMA buffer status
xfer_status &= DMA_XFER_NONE;
// Start DMA transfer
if (HAL_SAI_Receive_DMA(&hsai, (uint8_t*) PDM_BUFFER, PDM_BUFFER_SIZE / n_channels) != HAL_OK) {
RAISE_OS_EXCEPTION("SAI DMA transfer failed!");
}
uint32_t record_start = HAL_GetTick();
while ((HAL_GetTick() - record_start) < time) {
uint32_t start = HAL_GetTick();
// Wait for half transfer complete.
while ((xfer_status & DMA_XFER_HALF) == 0) {
if ((HAL_GetTick() - start) >= 1000) {
HAL_SAI_DMAStop(&hsai);
RAISE_OS_EXCEPTION("SAI DMA transfer timeout!");
}
}
// Clear buffer state.
xfer_status &= ~(DMA_XFER_HALF);
// Convert PDM samples to PCM.
for (int i=0; i<n_channels; i++) {
PDM_Filter(&((uint8_t*)PDM_BUFFER)[i], &((int16_t*)pcmbuf->items)[i], &PDM_FilterHandler[i]);
}
mp_call_function_2(callback, MP_OBJ_FROM_PTR(pcmbuf), arg_obj);
// Wait for transfer complete.
while ((xfer_status & DMA_XFER_FULL) == 0) {
if ((HAL_GetTick() - start) >= 1000) {
HAL_SAI_DMAStop(&hsai);
RAISE_OS_EXCEPTION("SAI DMA transfer timeout!");
}
}
// Clear buffer state.
xfer_status &= ~(DMA_XFER_FULL);
// Convert PDM samples to PCM.
for (int i=0; i<n_channels; i++) {
PDM_Filter(&((uint8_t*)PDM_BUFFER)[PDM_BUFFER_SIZE / 2 + i], &((int16_t*)pcmbuf->items)[i], &PDM_FilterHandler[i]);
}
mp_call_function_2(callback, MP_OBJ_FROM_PTR(pcmbuf), arg_obj);
}
// Stop SAI DMA.
HAL_SAI_DMAStop(&hsai);
g_audio_callback = mp_const_none;
return mp_const_none;
}
STATIC MP_DEFINE_CONST_FUN_OBJ_KW(py_audio_init_obj, 0, py_audio_init);
STATIC MP_DEFINE_CONST_FUN_OBJ_1(py_audio_read_pdm_obj, py_audio_read_pdm);
STATIC MP_DEFINE_CONST_FUN_OBJ_KW(py_audio_read_pcm_obj, 1, py_audio_read_pcm);
STATIC MP_DEFINE_CONST_FUN_OBJ_KW(py_audio_start_streaming_obj, 2, py_audio_start_streaming);
STATIC MP_DEFINE_CONST_FUN_OBJ_1(py_audio_start_streaming_obj, py_audio_start_streaming);
STATIC MP_DEFINE_CONST_FUN_OBJ_0(py_audio_stop_streaming_obj, py_audio_stop_streaming);
static const mp_map_elem_t globals_dict_table[] = {
{ MP_OBJ_NEW_QSTR(MP_QSTR___name__), MP_OBJ_NEW_QSTR(MP_QSTR_audio) },
{ MP_OBJ_NEW_QSTR(MP_QSTR_init), (mp_obj_t)&py_audio_init_obj },
{ MP_OBJ_NEW_QSTR(MP_QSTR_read_pcm), (mp_obj_t)&py_audio_read_pcm_obj },
{ MP_OBJ_NEW_QSTR(MP_QSTR_read_pdm), (mp_obj_t)&py_audio_read_pdm_obj },
{ MP_OBJ_NEW_QSTR(MP_QSTR_start_streaming), (mp_obj_t)&py_audio_start_streaming_obj },
static const mp_rom_map_elem_t globals_dict_table[] = {
{ MP_ROM_QSTR(MP_QSTR___name__), MP_ROM_QSTR(MP_QSTR_audio) },
{ MP_ROM_QSTR(MP_QSTR_init), MP_ROM_PTR(&py_audio_init_obj) },
{ MP_ROM_QSTR(MP_QSTR_read_pdm), MP_ROM_PTR(&py_audio_read_pdm_obj) },
{ MP_ROM_QSTR(MP_QSTR_start_streaming), MP_ROM_PTR(&py_audio_start_streaming_obj)},
{ MP_ROM_QSTR(MP_QSTR_stop_streaming), MP_ROM_PTR(&py_audio_stop_streaming_obj) },
};
STATIC MP_DEFINE_CONST_DICT(globals_dict, globals_dict_table);

14
src/omv/py/py_audio.h Normal file
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@ -0,0 +1,14 @@
/*
* This file is part of the OpenMV project.
*
* Copyright (c) 2013-2019 Ibrahim Abdelkader <iabdalkader@openmv.io>
* Copyright (c) 2013-2019 Kwabena W. Agyeman <kwagyeman@openmv.io>
*
* This work is licensed under the MIT license, see the file LICENSE for details.
*
* Audio Python module.
*/
#ifndef __PY_AUDIO_H__
#define __PY_AUDIO_H__
void py_audio_deinit();
#endif // __PY_AUDIO_H__

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@ -1256,5 +1256,5 @@ Q(gain_db)
Q(highpass)
Q(frequency)
Q(read_pdm)
Q(read_pcm)
Q(start_streaming)
Q(stop_streaming)