/** | |
****************************************************************************** | |
* @file stm32h745i_discovery_audio.c | |
* @author MCD Application Team | |
* @brief This file provides the Audio driver for the STM32H745I-DISCOVERY | |
* board. | |
@verbatim | |
How To use this driver: | |
----------------------- | |
+ This driver supports STM32H7xx devices on STM32H745I-DISCOVERY (MB1248) Discovery boards. | |
+ Call the function BSP_AUDIO_OUT_Init( | |
OutputDevice: physical output mode (OUTPUT_DEVICE_SPEAKER, | |
OUTPUT_DEVICE_HEADPHONE or OUTPUT_DEVICE_BOTH) | |
Volume : Initial volume to be set (0 is min (mute), 100 is max (100%) | |
AudioFreq : Audio frequency in Hz (8000, 16000, 22500, 32000...) | |
this parameter is relative to the audio file/stream type. | |
) | |
This function configures all the hardware required for the audio application (codec, I2C, SAI, | |
GPIOs, DMA and interrupt if needed). This function returns AUDIO_OK if configuration is OK. | |
If the returned value is different from AUDIO_OK or the function is stuck then the communication with | |
the codec has failed (try to un-plug the power or reset device in this case). | |
- OUTPUT_DEVICE_SPEAKER : only speaker will be set as output for the audio stream. | |
- OUTPUT_DEVICE_HEADPHONE: only headphones will be set as output for the audio stream. | |
- OUTPUT_DEVICE_BOTH : both Speaker and Headphone are used as outputs for the audio stream | |
at the same time. | |
Note. On STM32H745I-DISCOVERY SAI_DMA is configured in CIRCULAR mode. Due to this the application | |
does NOT need to call BSP_AUDIO_OUT_ChangeBuffer() to assure streaming. | |
+ Call the function BSP_AUDIO_OUT_Play( | |
pBuffer: pointer to the audio data file address | |
Size : size of the buffer to be sent in Bytes | |
) | |
to start playing (for the first time) from the audio file/stream. | |
+ Call the function BSP_AUDIO_OUT_Pause() to pause playing | |
+ Call the function BSP_AUDIO_OUT_Resume() to resume playing. | |
Note. After calling BSP_AUDIO_OUT_Pause() function for pause, only BSP_AUDIO_OUT_Resume() should be called | |
for resume (it is not allowed to call BSP_AUDIO_OUT_Play() in this case). | |
Note. This function should be called only when the audio file is played or paused (not stopped). | |
+ For each mode, you may need to implement the relative callback functions into your code. | |
The Callback functions are named BSP_AUDIO_OUT_XXX_CallBack() and only their prototypes are declared in | |
the stm32h745i_discovery_audio.h file. (refer to the example for more details on the callbacks implementations) | |
+ To Stop playing, to modify the volume level, the frequency, the audio frame slot, | |
the device output mode the mute or the stop, use the functions: BSP_AUDIO_OUT_SetVolume(), | |
AUDIO_OUT_SetFrequency(), BSP_AUDIO_OUT_SetAudioFrameSlot(), BSP_AUDIO_OUT_SetOutputMode(), | |
BSP_AUDIO_OUT_SetMute() and BSP_AUDIO_OUT_Stop(). | |
+ Call the function BSP_AUDIO_IN_Init( | |
AudioFreq: Audio frequency in Hz (8000, 16000, 22500, 32000...) | |
this parameter is relative to the audio file/stream type. | |
BitRes: Bit resolution fixed to 16bit | |
ChnlNbr: Number of channel to be configured for the DFSDM peripheral | |
) | |
This function configures all the hardware required for the audio in application (channels, | |
Clock source for SAI PDM periphiral, GPIOs, DMA and interrupt if needed). | |
This function returns AUDIO_OK if configuration is OK.If the returned value is different from AUDIO_OK then | |
the configuration should be wrong. | |
+ Call the function BSP_AUDIO_IN_AllocScratch( | |
pScratch: pointer to scratch tables | |
size: size of scratch buffer) | |
This function must be called before BSP_AUDIO_IN_RECORD() to allocate buffer scratch for each DFSDM channel | |
and its size. | |
Note: These buffers scratch are used as intermidiate buffers to collect data within final record buffer. | |
size is the total size of the four buffers scratch; If size is 512 then the size of each is 128. | |
This function must be called after BSP_AUDIO_IN_Init() | |
+ Call the function BSP_AUDIO_IN_RECORD( | |
pBuf: pointer to the recorded audio data file address | |
Size: size of the buffer to be written in Bytes | |
) | |
to start recording from microphones. | |
+ Call the function BSP_AUDIO_IN_Pause() to pause recording | |
+ Call the function BSP_AUDIO_IN_Resume() to recording playing. | |
Note. After calling BSP_AUDIO_IN_Pause() function for pause, only BSP_AUDIO_IN_Resume() should be called | |
for resume (it is not allowed to call BSP_AUDIO_IN_RECORD() in this case). | |
+ Call the function BSP_AUDIO_IN_Stop() to stop recording | |
+ For each mode, you may need to implement the relative callback functions into your code. | |
The Callback functions are named BSP_AUDIO_IN_XXX_CallBack() and only their prototypes are declared in | |
the stm32h745i_discovery_audio.h file. (refer to the example for more details on the callbacks implementations) | |
+ Call the function BSP_AUDIO_IN_SelectInterface(uint32_t Interface) to select one of the two interfaces | |
available on the STM32H745I-Discovery board: SAI or PDM. This function is to be called before BSP_AUDIO_IN_InitEx(). | |
+ Call the function BSP_AUDIO_IN_GetInterface() to get the current used interface. | |
+ Call the function BSP_AUDIO_IN_PDMToPCM_Init(uint32_t AudioFreq, uint32_t ChnlNbrIn, uint32_t ChnlNbrOut) | |
to init PDM filters if the libPDMFilter is used for audio data filtering. | |
+ Call the function BSP_AUDIO_IN_PDMToPCM(uint16_t* PDMBuf, uint16_t* PCMBuf) to filter PDM data to PCM format | |
if the libPDMFilter library is used for audio data filtering. | |
Driver architecture: | |
-------------------- | |
+ This driver provides the High Audio Layer: consists of the function API exported in the stm32h745i_discovery_audio.h file | |
(BSP_AUDIO_OUT_Init(), BSP_AUDIO_OUT_Play() ...) | |
+ This driver provide also the Media Access Layer (MAL): which consists of functions allowing to access the media containing/ | |
providing the audio file/stream. These functions are also included as local functions into | |
the stm32h745i_discovery_audio.c file (DFSDMx_Init(), DFSDMx_DeInit(), SAIx_Init() and SAIx_DeInit()) | |
Known Limitations: | |
------------------ | |
1- If the TDM Format used to play in parallel 2 audio Stream (the first Stream is configured in codec SLOT0 and second | |
Stream in SLOT1) the Pause/Resume, volume and mute feature will control the both streams. | |
2- Parsing of audio file is not implemented (in order to determine audio file properties: Mono/Stereo, Data size, | |
File size, Audio Frequency, Audio Data header size ...). The configuration is fixed for the given audio file. | |
3- Supports only Stereo audio streaming. | |
4- Supports only 16-bits audio data size. | |
@endverbatim | |
****************************************************************************** | |
* @attention | |
* | |
* <h2><center>© Copyright (c) 2019 STMicroelectronics. | |
* All rights reserved.</center></h2> | |
* | |
* This software component is licensed by ST under BSD 3-Clause license, | |
* the "License"; You may not use this file except in compliance with the | |
* License. You may obtain a copy of the License at: | |
* opensource.org/licenses/BSD-3-Clause | |
* | |
****************************************************************************** | |
*/ | |
/* Includes ------------------------------------------------------------------*/ | |
#include "stm32h745i_discovery_audio.h" | |
/** @addtogroup BSP | |
* @{ | |
*/ | |
/** @addtogroup STM32H745I_DISCOVERY | |
* @{ | |
*/ | |
/** @defgroup STM32H745I_DISCOVERY_AUDIO STM32H745I_DISCOVERY_AUDIO | |
* @brief This file includes the low layer driver for wm8994 Audio Codec | |
* available on STM32H745I-DISCOVERY discovery board(MB1248). | |
* @{ | |
*/ | |
/** @defgroup STM32H745I_DISCOVERY_AUDIO_Private_Variables Private Variables | |
* @{ | |
*/ | |
/* PLAY */ | |
AUDIO_DrvTypeDef *audio_drv; | |
SAI_HandleTypeDef haudio_out_sai; | |
SAI_HandleTypeDef haudio_in_sai; | |
/* RECORD */ | |
AUDIOIN_ContextTypeDef hAudioIn; | |
/* Audio in Volume value */ | |
__IO uint16_t AudioInVolume = DEFAULT_AUDIO_IN_VOLUME; | |
/* PDM filters params */ | |
PDM_Filter_Handler_t PDM_FilterHandler[2]; | |
PDM_Filter_Config_t PDM_FilterConfig[2]; | |
/** | |
* @} | |
*/ | |
/** @defgroup STM32H745I_DISCOVERY_AUDIO_Private_Function_Prototypes OUT Private Function Prototypes | |
* @{ | |
*/ | |
static void SAIx_Out_Init(uint32_t SaiOutMode, uint32_t SlotActive, uint32_t AudioFreq); | |
static void SAIx_Out_DeInit(SAI_HandleTypeDef *hsai); | |
/** | |
* @} | |
*/ | |
/** @defgroup STM32H745I_DISCOVERY_AUDIO_IN_Private_Function_Prototypes IN Private Function Prototypes | |
* @{ | |
*/ | |
static void SAIx_In_MspInit(SAI_HandleTypeDef *hsai, void *Params); | |
static void SAIx_In_MspDeInit(SAI_HandleTypeDef *hsai, void *Params); | |
static void SAIx_In_Init(uint32_t SaiInMode, uint32_t SlotActive, uint32_t AudioFreq); | |
static void SAIx_In_DeInit(SAI_HandleTypeDef *hsai); | |
/** | |
* @} | |
*/ | |
/** @defgroup STM32H745I_DISCOVERY_AUDIO_OUT_Exported_Functions OUT Exported Functions | |
* @{ | |
*/ | |
/** | |
* @brief Configures the audio Out peripheral. | |
* @param OutputDevice: OUTPUT_DEVICE_SPEAKER, OUTPUT_DEVICE_HEADPHONE, | |
* or OUTPUT_DEVICE_BOTH. | |
* @param Volume: Initial volume level (from 0 (Mute) to 100 (Max)) | |
* @param AudioFreq: Audio frequency used to play the audio stream. | |
* @retval AUDIO_OK if correct communication, else wrong communication | |
*/ | |
uint8_t BSP_AUDIO_OUT_Init(uint16_t OutputDevice, uint8_t Volume, uint32_t AudioFreq) | |
{ | |
uint8_t ret = AUDIO_ERROR; | |
uint32_t deviceid = 0x00; | |
uint32_t slot_active; | |
/* Initialize SAI2 sub_block A as MASTER TX */ | |
haudio_out_sai.Instance = AUDIO_OUT_SAIx; | |
/* Disable SAI */ | |
SAIx_Out_DeInit(&haudio_out_sai); | |
/* PLL clock is set depending by the AudioFreq (44.1khz vs 48khz groups) */ | |
BSP_AUDIO_OUT_ClockConfig(&haudio_out_sai, AudioFreq, NULL); | |
/* SAI data transfer preparation: | |
Prepare the Media to be used for the audio transfer from memory to SAI peripheral */ | |
if(HAL_SAI_GetState(&haudio_out_sai) == HAL_SAI_STATE_RESET) | |
{ | |
/* Init the SAI MSP: this __weak function can be redefined by the application*/ | |
BSP_AUDIO_OUT_MspInit(&haudio_out_sai, NULL); | |
} | |
/* Init SAI as master RX output */ | |
slot_active = CODEC_AUDIOFRAME_SLOT_0123; | |
SAIx_Out_Init(SAI_MODEMASTER_TX, slot_active, AudioFreq); | |
/* wm8994 codec initialization */ | |
deviceid = wm8994_drv.ReadID(AUDIO_I2C_ADDRESS); | |
if((deviceid) == WM8994_ID) | |
{ | |
/* Reset the Codec Registers */ | |
wm8994_drv.Reset(AUDIO_I2C_ADDRESS); | |
/* Initialize the audio driver structure */ | |
audio_drv = &wm8994_drv; | |
ret = AUDIO_OK; | |
} | |
else | |
{ | |
ret = AUDIO_ERROR; | |
} | |
if(ret == AUDIO_OK) | |
{ | |
/* Initialize the codec internal registers */ | |
audio_drv->Init(AUDIO_I2C_ADDRESS, OutputDevice, Volume, AudioFreq); | |
} | |
return ret; | |
} | |
/** | |
* @brief Starts playing audio stream from a data buffer for a determined size. | |
* @param pBuffer: Pointer to the buffer | |
* @param Size: Number of audio data BYTES. | |
* @retval AUDIO_OK if correct communication, else wrong communication | |
*/ | |
uint8_t BSP_AUDIO_OUT_Play(uint16_t* pBuffer, uint32_t Size) | |
{ | |
/* Call the audio Codec Play function */ | |
if(audio_drv->Play(AUDIO_I2C_ADDRESS, pBuffer, Size) != 0) | |
{ | |
return AUDIO_ERROR; | |
} | |
else | |
{ | |
/* Update the Media layer and enable it for play */ | |
HAL_SAI_Transmit_DMA(&haudio_out_sai, (uint8_t*) pBuffer, DMA_MAX(Size / AUDIODATA_SIZE)); | |
return AUDIO_OK; | |
} | |
} | |
/** | |
* @brief Sends n-Bytes on the SAI interface. | |
* @param pData: pointer on data address | |
* @param Size: number of data to be written | |
* @retval None | |
*/ | |
void BSP_AUDIO_OUT_ChangeBuffer(uint16_t *pData, uint16_t Size) | |
{ | |
HAL_SAI_Transmit_DMA(&haudio_out_sai, (uint8_t*) pData, Size); | |
} | |
/** | |
* @brief This function Pauses the audio file stream. In case | |
* of using DMA, the DMA Pause feature is used. | |
* @warning When calling BSP_AUDIO_OUT_Pause() function for pause, only | |
* BSP_AUDIO_OUT_Resume() function should be called for resume (use of BSP_AUDIO_OUT_Play() | |
* function for resume could lead to unexpected behaviour). | |
* @retval AUDIO_OK if correct communication, else wrong communication | |
*/ | |
uint8_t BSP_AUDIO_OUT_Pause(void) | |
{ | |
/* Call the Audio Codec Pause/Resume function */ | |
if(audio_drv->Pause(AUDIO_I2C_ADDRESS) != 0) | |
{ | |
return AUDIO_ERROR; | |
} | |
else | |
{ | |
/* Call the Media layer pause function */ | |
HAL_SAI_DMAPause(&haudio_out_sai); | |
/* Return AUDIO_OK when all operations are correctly done */ | |
return AUDIO_OK; | |
} | |
} | |
/** | |
* @brief Resumes the audio file stream. | |
* @warning When calling BSP_AUDIO_OUT_Pause() function for pause, only | |
* BSP_AUDIO_OUT_Resume() function should be called for resume (use of BSP_AUDIO_OUT_Play() | |
* function for resume could lead to unexpected behaviour). | |
* @retval AUDIO_OK if correct communication, else wrong communication | |
*/ | |
uint8_t BSP_AUDIO_OUT_Resume(void) | |
{ | |
/* Call the Audio Codec Pause/Resume function */ | |
if(audio_drv->Resume(AUDIO_I2C_ADDRESS) != 0) | |
{ | |
return AUDIO_ERROR; | |
} | |
else | |
{ | |
/* Call the Media layer pause/resume function */ | |
HAL_SAI_DMAResume(&haudio_out_sai); | |
/* Return AUDIO_OK when all operations are correctly done */ | |
return AUDIO_OK; | |
} | |
} | |
/** | |
* @brief Stops audio playing and Power down the Audio Codec. | |
* @param Option: could be one of the following parameters | |
* - CODEC_PDWN_SW: for software power off (by writing registers). | |
* Then no need to reconfigure the Codec after power on. | |
* - CODEC_PDWN_HW: completely shut down the codec (physically). | |
* Then need to reconfigure the Codec after power on. | |
* @retval AUDIO_OK if correct communication, else wrong communication | |
*/ | |
uint8_t BSP_AUDIO_OUT_Stop(uint32_t Option) | |
{ | |
/* Call the Media layer stop function */ | |
HAL_SAI_DMAStop(&haudio_out_sai); | |
/* Call Audio Codec Stop function */ | |
if(audio_drv->Stop(AUDIO_I2C_ADDRESS, Option) != 0) | |
{ | |
return AUDIO_ERROR; | |
} | |
else | |
{ | |
if(Option == CODEC_PDWN_HW) | |
{ | |
/* Wait at least 100us */ | |
HAL_Delay(1); | |
} | |
/* Return AUDIO_OK when all operations are correctly done */ | |
return AUDIO_OK; | |
} | |
} | |
/** | |
* @brief Controls the current audio volume level. | |
* @param Volume: Volume level to be set in percentage from 0% to 100% (0 for | |
* Mute and 100 for Max volume level). | |
* @retval AUDIO_OK if correct communication, else wrong communication | |
*/ | |
uint8_t BSP_AUDIO_OUT_SetVolume(uint8_t Volume) | |
{ | |
/* Call the codec volume control function with converted volume value */ | |
if(audio_drv->SetVolume(AUDIO_I2C_ADDRESS, Volume) != 0) | |
{ | |
return AUDIO_ERROR; | |
} | |
else | |
{ | |
/* Return AUDIO_OK when all operations are correctly done */ | |
return AUDIO_OK; | |
} | |
} | |
/** | |
* @brief Enables or disables the MUTE mode by software | |
* @param Cmd: Could be AUDIO_MUTE_ON to mute sound or AUDIO_MUTE_OFF to | |
* unmute the codec and restore previous volume level. | |
* @retval AUDIO_OK if correct communication, else wrong communication | |
*/ | |
uint8_t BSP_AUDIO_OUT_SetMute(uint32_t Cmd) | |
{ | |
/* Call the Codec Mute function */ | |
if(audio_drv->SetMute(AUDIO_I2C_ADDRESS, Cmd) != 0) | |
{ | |
return AUDIO_ERROR; | |
} | |
else | |
{ | |
/* Return AUDIO_OK when all operations are correctly done */ | |
return AUDIO_OK; | |
} | |
} | |
/** | |
* @brief Switch dynamically (while audio file is played) the output target | |
* (speaker or headphone). | |
* @param Output: The audio output target: OUTPUT_DEVICE_SPEAKER, | |
* OUTPUT_DEVICE_HEADPHONE or OUTPUT_DEVICE_BOTH | |
* @retval AUDIO_OK if correct communication, else wrong communication | |
*/ | |
uint8_t BSP_AUDIO_OUT_SetOutputMode(uint8_t Output) | |
{ | |
/* Call the Codec output device function */ | |
if(audio_drv->SetOutputMode(AUDIO_I2C_ADDRESS, Output) != 0) | |
{ | |
return AUDIO_ERROR; | |
} | |
else | |
{ | |
/* Return AUDIO_OK when all operations are correctly done */ | |
return AUDIO_OK; | |
} | |
} | |
/** | |
* @brief Updates the audio frequency. | |
* @param AudioFreq: Audio frequency used to play the audio stream. | |
* @note This API should be called after the BSP_AUDIO_OUT_Init() to adjust the | |
* audio frequency. | |
* @retval None | |
*/ | |
void BSP_AUDIO_OUT_SetFrequency(uint32_t AudioFreq) | |
{ | |
/* PLL clock is set depending by the AudioFreq (44.1khz vs 48khz groups) */ | |
BSP_AUDIO_OUT_ClockConfig(&haudio_out_sai, AudioFreq, NULL); | |
/* Disable SAI peripheral to allow access to SAI internal registers */ | |
__HAL_SAI_DISABLE(&haudio_out_sai); | |
/* Update the SAI audio frequency configuration */ | |
haudio_out_sai.Init.AudioFrequency = AudioFreq; | |
HAL_SAI_Init(&haudio_out_sai); | |
/* Enable SAI peripheral to generate MCLK */ | |
__HAL_SAI_ENABLE(&haudio_out_sai); | |
} | |
/** | |
* @brief Updates the Audio frame slot configuration. | |
* @param AudioFrameSlot: specifies the audio Frame slot | |
* @note This API should be called after the BSP_AUDIO_OUT_Init() to adjust the | |
* audio frame slot. | |
* @retval None | |
*/ | |
void BSP_AUDIO_OUT_SetAudioFrameSlot(uint32_t AudioFrameSlot) | |
{ | |
/* Disable SAI peripheral to allow access to SAI internal registers */ | |
__HAL_SAI_DISABLE(&haudio_out_sai); | |
/* Update the SAI audio frame slot configuration */ | |
haudio_out_sai.SlotInit.SlotActive = AudioFrameSlot; | |
HAL_SAI_Init(&haudio_out_sai); | |
/* Enable SAI peripheral to generate MCLK */ | |
__HAL_SAI_ENABLE(&haudio_out_sai); | |
} | |
/** | |
* @brief De-initializes the audio out peripheral. | |
* @retval None | |
*/ | |
void BSP_AUDIO_OUT_DeInit(void) | |
{ | |
SAIx_Out_DeInit(&haudio_out_sai); | |
/* DeInit the SAI MSP : this __weak function can be rewritten by the application */ | |
BSP_AUDIO_OUT_MspDeInit(&haudio_out_sai, NULL); | |
} | |
/** | |
* @brief Manages the DMA full Transfer complete event. | |
* @retval None | |
*/ | |
__weak void BSP_AUDIO_OUT_TransferComplete_CallBack(void) | |
{ | |
} | |
/** | |
* @brief Manages the DMA Half Transfer complete event. | |
* @retval None | |
*/ | |
__weak void BSP_AUDIO_OUT_HalfTransfer_CallBack(void) | |
{ | |
} | |
/** | |
* @brief Manages the DMA FIFO error event. | |
* @retval None | |
*/ | |
__weak void BSP_AUDIO_OUT_Error_CallBack(void) | |
{ | |
} | |
/** | |
* @brief Initializes BSP_AUDIO_OUT MSP. | |
* @param hsai: SAI handle | |
* @param Params: pointer on additional configuration parameters, can be NULL. | |
* @retval None | |
*/ | |
__weak void BSP_AUDIO_OUT_MspInit(SAI_HandleTypeDef *hsai, void *Params) | |
{ | |
static DMA_HandleTypeDef hdma_sai_tx; | |
GPIO_InitTypeDef gpio_init_structure; | |
/* Enable SAI clock */ | |
AUDIO_OUT_SAIx_CLK_ENABLE(); | |
/* CODEC_SAI pins configuration: FS, SCK and SD pins */ | |
/* Enable FS, SCK and SD clocks */ | |
AUDIO_OUT_SAIx_SD_FS_CLK_ENABLE(); | |
/* Enable FS, SCK and SD pins */ | |
gpio_init_structure.Pin = AUDIO_OUT_SAIx_FS_PIN | AUDIO_OUT_SAIx_SCK_PIN | AUDIO_OUT_SAIx_SD_PIN; | |
gpio_init_structure.Mode = GPIO_MODE_AF_PP; | |
gpio_init_structure.Pull = GPIO_NOPULL; | |
gpio_init_structure.Speed = GPIO_SPEED_FREQ_VERY_HIGH; | |
gpio_init_structure.Alternate = AUDIO_OUT_SAIx_AF; | |
HAL_GPIO_Init(AUDIO_OUT_SAIx_SD_FS_SCK_GPIO_PORT, &gpio_init_structure); | |
/* Enable MCLK clock */ | |
AUDIO_OUT_SAIx_MCLK_ENABLE(); | |
/* Enable MCLK pin */ | |
gpio_init_structure.Pin = AUDIO_OUT_SAIx_MCLK_PIN; | |
HAL_GPIO_Init(AUDIO_OUT_SAIx_MCLK_GPIO_PORT, &gpio_init_structure); | |
/* Enable the DMA clock */ | |
AUDIO_OUT_SAIx_DMAx_CLK_ENABLE(); | |
if(hsai->Instance == AUDIO_OUT_SAIx) | |
{ | |
/* Configure the hdma_saiTx handle parameters */ | |
hdma_sai_tx.Init.Request = AUDIO_OUT_SAIx_DMAx_REQUEST; | |
hdma_sai_tx.Init.Direction = DMA_MEMORY_TO_PERIPH; | |
hdma_sai_tx.Init.PeriphInc = DMA_PINC_DISABLE; | |
hdma_sai_tx.Init.MemInc = DMA_MINC_ENABLE; | |
hdma_sai_tx.Init.PeriphDataAlignment = AUDIO_OUT_SAIx_DMAx_PERIPH_DATA_SIZE; | |
hdma_sai_tx.Init.MemDataAlignment = AUDIO_OUT_SAIx_DMAx_MEM_DATA_SIZE; | |
hdma_sai_tx.Init.Mode = DMA_CIRCULAR; | |
hdma_sai_tx.Init.Priority = DMA_PRIORITY_HIGH; | |
hdma_sai_tx.Init.FIFOMode = DMA_FIFOMODE_ENABLE; | |
hdma_sai_tx.Init.FIFOThreshold = DMA_FIFO_THRESHOLD_FULL; | |
hdma_sai_tx.Init.MemBurst = DMA_MBURST_SINGLE; | |
hdma_sai_tx.Init.PeriphBurst = DMA_PBURST_SINGLE; | |
hdma_sai_tx.Instance = AUDIO_OUT_SAIx_DMAx_STREAM; | |
/* Associate the DMA handle */ | |
__HAL_LINKDMA(hsai, hdmatx, hdma_sai_tx); | |
/* Deinitialize the Stream for new transfer */ | |
HAL_DMA_DeInit(&hdma_sai_tx); | |
/* Configure the DMA Stream */ | |
HAL_DMA_Init(&hdma_sai_tx); | |
} | |
/* SAI DMA IRQ Channel configuration */ | |
HAL_NVIC_SetPriority(AUDIO_OUT_SAIx_DMAx_IRQ, AUDIO_OUT_IRQ_PREPRIO, 0); | |
HAL_NVIC_EnableIRQ(AUDIO_OUT_SAIx_DMAx_IRQ); | |
} | |
/** | |
* @brief Deinitializes SAI MSP. | |
* @param hsai: SAI handle | |
* @param Params: pointer on additional configuration parameters, can be NULL. | |
* @retval None | |
*/ | |
__weak void BSP_AUDIO_OUT_MspDeInit(SAI_HandleTypeDef *hsai, void *Params) | |
{ | |
GPIO_InitTypeDef gpio_init_structure; | |
/* SAI DMA IRQ Channel deactivation */ | |
HAL_NVIC_DisableIRQ(AUDIO_OUT_SAIx_DMAx_IRQ); | |
if(hsai->Instance == AUDIO_OUT_SAIx) | |
{ | |
/* Deinitialize the DMA stream */ | |
HAL_DMA_DeInit(hsai->hdmatx); | |
} | |
/* Disable SAI peripheral */ | |
__HAL_SAI_DISABLE(hsai); | |
/* Deactivates CODEC_SAI pins FS, SCK, MCK and SD by putting them in input mode */ | |
gpio_init_structure.Pin = AUDIO_OUT_SAIx_FS_PIN | AUDIO_OUT_SAIx_SCK_PIN | AUDIO_OUT_SAIx_SD_PIN; | |
HAL_GPIO_DeInit(AUDIO_OUT_SAIx_SD_FS_SCK_GPIO_PORT, gpio_init_structure.Pin); | |
gpio_init_structure.Pin = AUDIO_OUT_SAIx_MCLK_PIN; | |
HAL_GPIO_DeInit(AUDIO_OUT_SAIx_MCLK_GPIO_PORT, gpio_init_structure.Pin); | |
/* Disable SAI clock */ | |
AUDIO_OUT_SAIx_CLK_DISABLE(); | |
/* GPIO pins clock and DMA clock can be shut down in the applic | |
by surcharging this __weak function */ | |
} | |
/** | |
* @brief Clock Config. | |
* @param hsai: might be required to set audio peripheral predivider if any. | |
* @param AudioFreq: Audio frequency used to play the audio stream. | |
* @param Params: pointer on additional configuration parameters, can be NULL. | |
* @note This API is called by BSP_AUDIO_OUT_Init() and BSP_AUDIO_OUT_SetFrequency() | |
* Being __weak it can be overwritten by the application | |
* @retval None | |
*/ | |
__weak void BSP_AUDIO_OUT_ClockConfig(SAI_HandleTypeDef *hsai, uint32_t AudioFreq, void *Params) | |
{ | |
RCC_PeriphCLKInitTypeDef rcc_ex_clk_init_struct; | |
HAL_RCCEx_GetPeriphCLKConfig(&rcc_ex_clk_init_struct); | |
/* Set the PLL configuration according to the audio frequency */ | |
if((AudioFreq == AUDIO_FREQUENCY_11K) || (AudioFreq == AUDIO_FREQUENCY_22K) || (AudioFreq == AUDIO_FREQUENCY_44K)) | |
{ | |
/* SAI clock config: | |
PLL2_VCO Input = HSE_VALUE/PLL2M = 1 Mhz | |
PLL2_VCO Output = PLL2_VCO Input * PLL2N = 429 Mhz | |
SAI_CLK_x = PLL2_VCO Output/PLL2P = 429/38 = 11.289 Mhz */ | |
rcc_ex_clk_init_struct.PeriphClockSelection = RCC_PERIPHCLK_SAI2; | |
rcc_ex_clk_init_struct.Sai23ClockSelection = RCC_SAI2CLKSOURCE_PLL2; | |
rcc_ex_clk_init_struct.PLL2.PLL2P = 38; | |
rcc_ex_clk_init_struct.PLL2.PLL2Q = 1; | |
rcc_ex_clk_init_struct.PLL2.PLL2R = 1; | |
rcc_ex_clk_init_struct.PLL2.PLL2N = 429; | |
rcc_ex_clk_init_struct.PLL2.PLL2M = 25; | |
HAL_RCCEx_PeriphCLKConfig(&rcc_ex_clk_init_struct); | |
} | |
else /* AUDIO_FREQUENCY_8K, AUDIO_FREQUENCY_16K, AUDIO_FREQUENCY_48K, AUDIO_FREQUENCY_96K */ | |
{ | |
/* SAI clock config: | |
PLL2_VCO Input = HSE_VALUE/PLL2M = 1 Mhz | |
PLL2_VCO Output = PLL2_VCO Input * PLL2N = 344 Mhz | |
SAI_CLK_x = PLL2_VCO Output/PLL2P = 344/7 = 49.142 Mhz */ | |
rcc_ex_clk_init_struct.PeriphClockSelection = RCC_PERIPHCLK_SAI2; | |
rcc_ex_clk_init_struct.Sai23ClockSelection = RCC_SAI2CLKSOURCE_PLL2; | |
rcc_ex_clk_init_struct.PLL2.PLL2P = 7; | |
rcc_ex_clk_init_struct.PLL2.PLL2Q = 1; | |
rcc_ex_clk_init_struct.PLL2.PLL2R = 1; | |
rcc_ex_clk_init_struct.PLL2.PLL2N = 344; | |
rcc_ex_clk_init_struct.PLL2.PLL2M = 25; | |
HAL_RCCEx_PeriphCLKConfig(&rcc_ex_clk_init_struct); | |
} | |
} | |
/** | |
* @} | |
*/ | |
/** @defgroup STM32H745I_DISCOVERY_AUDIO_OUT_Private_Functions OUT Private Functions | |
* @{ | |
*/ | |
/******************************************************************************* | |
HAL Callbacks | |
*******************************************************************************/ | |
/** | |
* @brief Tx Transfer completed callbacks. | |
* @param hsai: SAI handle | |
* @retval None | |
*/ | |
void HAL_SAI_TxCpltCallback(SAI_HandleTypeDef *hsai) | |
{ | |
/* Manage the remaining file size and new address offset: This function | |
should be coded by user (its prototype is already declared in stm32h745i_discovery_audio.h) */ | |
BSP_AUDIO_OUT_TransferComplete_CallBack(); | |
} | |
/** | |
* @brief Tx Half Transfer completed callbacks. | |
* @param hsai: SAI handle | |
* @retval None | |
*/ | |
void HAL_SAI_TxHalfCpltCallback(SAI_HandleTypeDef *hsai) | |
{ | |
/* Manage the remaining file size and new address offset: This function | |
should be coded by user (its prototype is already declared in stm32h745i_discovery_audio.h) */ | |
BSP_AUDIO_OUT_HalfTransfer_CallBack(); | |
} | |
/** | |
* @brief SAI error callbacks. | |
* @param hsai: SAI handle | |
* @retval None | |
*/ | |
void HAL_SAI_ErrorCallback(SAI_HandleTypeDef *hsai) | |
{ | |
if(hsai->Instance == AUDIO_OUT_SAIx) | |
{ | |
BSP_AUDIO_OUT_Error_CallBack(); | |
} | |
else | |
{ | |
BSP_AUDIO_IN_Error_CallBack(); | |
} | |
} | |
/******************************************************************************* | |
Static Functions | |
*******************************************************************************/ | |
/** | |
* @brief Initializes the Audio Codec audio interface (SAI). | |
* @param SaiOutMode: Audio mode to be configured for the SAI peripheral. | |
* @param SlotActive: Audio active slot to be configured for the SAI peripheral. | |
* @param AudioFreq: Audio frequency to be configured for the SAI peripheral. | |
* @note The default SlotActive configuration is set to CODEC_AUDIOFRAME_SLOT_0123 | |
* and user can update this configuration using | |
* @retval None | |
*/ | |
static void SAIx_Out_Init(uint32_t SaiOutMode, uint32_t SlotActive, uint32_t AudioFreq) | |
{ | |
/* Disable SAI peripheral to allow access to SAI internal registers */ | |
__HAL_SAI_DISABLE(&haudio_out_sai); | |
/* Configure SAI_Block_x | |
LSBFirst: Disabled | |
DataSize: 16 */ | |
haudio_out_sai.Init.MonoStereoMode = SAI_STEREOMODE; | |
haudio_out_sai.Init.AudioFrequency = AudioFreq; | |
haudio_out_sai.Init.AudioMode = SaiOutMode; | |
haudio_out_sai.Init.NoDivider = SAI_MASTERDIVIDER_ENABLE; | |
haudio_out_sai.Init.Protocol = SAI_FREE_PROTOCOL; | |
haudio_out_sai.Init.DataSize = SAI_DATASIZE_16; | |
haudio_out_sai.Init.FirstBit = SAI_FIRSTBIT_MSB; | |
haudio_out_sai.Init.ClockStrobing = SAI_CLOCKSTROBING_RISINGEDGE; | |
haudio_out_sai.Init.Synchro = SAI_ASYNCHRONOUS; | |
haudio_out_sai.Init.OutputDrive = SAI_OUTPUTDRIVE_ENABLE; | |
haudio_out_sai.Init.FIFOThreshold = SAI_FIFOTHRESHOLD_1QF; | |
haudio_out_sai.Init.SynchroExt = SAI_SYNCEXT_DISABLE; | |
haudio_out_sai.Init.CompandingMode = SAI_NOCOMPANDING; | |
haudio_out_sai.Init.TriState = SAI_OUTPUT_NOTRELEASED; | |
haudio_out_sai.Init.Mckdiv = 0; | |
haudio_out_sai.Init.MckOverSampling = SAI_MCK_OVERSAMPLING_DISABLE; | |
haudio_out_sai.Init.PdmInit.Activation = DISABLE; | |
haudio_out_sai.Init.PdmInit.ClockEnable = 0; | |
haudio_out_sai.Init.PdmInit.MicPairsNbr = 0; | |
/* Configure SAI_Block_x Frame | |
Frame Length: 64 | |
Frame active Length: 32 | |
FS Definition: Start frame + Channel Side identification | |
FS Polarity: FS active Low | |
FS Offset: FS asserted one bit before the first bit of slot 0 */ | |
haudio_out_sai.FrameInit.FrameLength = 128; | |
haudio_out_sai.FrameInit.ActiveFrameLength = 64; | |
haudio_out_sai.FrameInit.FSDefinition = SAI_FS_CHANNEL_IDENTIFICATION; | |
haudio_out_sai.FrameInit.FSPolarity = SAI_FS_ACTIVE_LOW; | |
haudio_out_sai.FrameInit.FSOffset = SAI_FS_BEFOREFIRSTBIT; | |
/* Configure SAI Block_x Slot | |
Slot First Bit Offset: 0 | |
Slot Size : 16 | |
Slot Number: 4 | |
Slot Active: All slot actives */ | |
haudio_out_sai.SlotInit.FirstBitOffset = 0; | |
haudio_out_sai.SlotInit.SlotSize = SAI_SLOTSIZE_DATASIZE; | |
haudio_out_sai.SlotInit.SlotNumber = 4; | |
haudio_out_sai.SlotInit.SlotActive = SlotActive; | |
HAL_SAI_Init(&haudio_out_sai); | |
/* Enable SAI peripheral to generate MCLK */ | |
__HAL_SAI_ENABLE(&haudio_out_sai); | |
} | |
/** | |
* @brief Deinitializes the Audio Codec audio interface (SAI). | |
* @retval None | |
*/ | |
static void SAIx_Out_DeInit(SAI_HandleTypeDef *hsai) | |
{ | |
/* Disable SAI peripheral */ | |
__HAL_SAI_DISABLE(hsai); | |
HAL_SAI_DeInit(hsai); | |
} | |
/** | |
* @} | |
*/ | |
/** @defgroup STM32H745I_DISCOVERY_AUDIO_IN_Exported_Functions IN Exported Functions | |
* @{ | |
*/ | |
/** | |
* @brief Initialize wave recording. | |
* @param AudioFreq: Audio frequency to be configured for the DFSDM peripheral. | |
* @param BitRes: Audio frequency to be configured for the DFSDM peripheral. | |
* @param ChnlNbr: Audio frequency to be configured for the DFSDM peripheral. | |
* @retval AUDIO_OK if correct communication, else wrong communication | |
*/ | |
uint8_t BSP_AUDIO_IN_Init(uint32_t AudioFreq, uint32_t BitRes, uint32_t ChnlNbr) | |
{ | |
/* Set audio in interface to default one */ | |
BSP_AUDIO_IN_SelectInterface(AUDIO_IN_INTERFACE_PDM); | |
return BSP_AUDIO_IN_InitEx(INPUT_DEVICE_DIGITAL_MIC, AudioFreq, BitRes, ChnlNbr); | |
} | |
/** | |
* @brief Initialize wave recording. | |
* @param InputDevice: INPUT_DEVICE_DIGITAL_MIC or INPUT_DEVICE_ANALOG_MIC. | |
* @param AudioFreq: Audio frequency to be configured. | |
* @param BitRes: Audio bit resolution to be configured.. | |
* @param ChnlNbr: Number of channel to be configured. | |
* @retval AUDIO_OK if correct communication, else wrong communication | |
*/ | |
uint8_t BSP_AUDIO_IN_InitEx(uint16_t InputDevice, uint32_t AudioFreq, uint32_t BitRes, uint32_t ChnlNbr) | |
{ | |
uint8_t ret = AUDIO_OK; | |
uint32_t slot_active; | |
/* Store the audio record context */ | |
hAudioIn.Frequency = AudioFreq; | |
hAudioIn.BitResolution = BitRes; | |
hAudioIn.InputDevice = InputDevice; | |
hAudioIn.ChannelNbr = ChnlNbr; | |
if(hAudioIn.InputDevice == INPUT_DEVICE_DIGITAL_MIC) | |
{ | |
if(hAudioIn.Interface == AUDIO_IN_INTERFACE_SAI) | |
{ | |
/* Initialize SAI2 block B as SLAVE RX synchrounous with SAI2 block A */ | |
haudio_in_sai.Instance = AUDIO_IN_SAIx; | |
/* Disable SAI */ | |
SAIx_In_DeInit(&haudio_in_sai); | |
/* PLL clock is set depending on the AudioFreq (44.1khz vs 48khz groups) */ | |
BSP_AUDIO_IN_ClockConfig(AudioFreq, NULL); /* Clock config is shared between AUDIO IN and OUT */ | |
/* SAI data transfer preparation: | |
Prepare the Media to be used for the audio transfer from SAI peripheral to memory */ | |
if(HAL_SAI_GetState(&haudio_in_sai) == HAL_SAI_STATE_RESET) | |
{ | |
/* Init the SAI MSP: this __weak function can be redefined by the application*/ | |
BSP_AUDIO_IN_MspInit(); | |
} | |
/* Configure SAI in master mode : | |
* - SAI2_block_B in slave RX mode synchronous from SAI2_block_A | |
*/ | |
slot_active = CODEC_AUDIOFRAME_SLOT_13; | |
SAIx_In_Init(SAI_MODESLAVE_RX, slot_active, AudioFreq); | |
} | |
else if(hAudioIn.Interface == AUDIO_IN_INTERFACE_PDM) | |
{ | |
/* Initialize SAI2 block A as MASTER RX */ | |
haudio_in_sai.Instance = AUDIO_IN_SAI_PDMx; | |
/* Disable SAI */ | |
SAIx_In_DeInit(&haudio_in_sai); | |
/* PLL clock is set depending on the AudioFreq (44.1khz vs 48khz groups) */ | |
BSP_AUDIO_IN_ClockConfig(AudioFreq, NULL); | |
/* SAI data transfer preparation: | |
Prepare the Media to be used for the audio transfer from SAI peripheral to memory */ | |
/* Initialize the haudio_in_sai Instance parameter */ | |
if(HAL_SAI_GetState(&haudio_in_sai) == HAL_SAI_STATE_RESET) | |
{ | |
/* Init the SAI MSP: this __weak function can be redefined by the application*/ | |
BSP_AUDIO_IN_MspInit(); | |
} | |
/* Configure SAI in master mode : | |
* - SAI4_block_A in master RX mode | |
*/ | |
slot_active = CODEC_AUDIOFRAME_SLOT_1; | |
SAIx_In_Init(SAI_MODEMASTER_RX, slot_active, AudioFreq); | |
if(BSP_AUDIO_IN_PDMToPCM_Init(AudioFreq, hAudioIn.ChannelNbr, hAudioIn.ChannelNbr) != AUDIO_OK) | |
{ | |
ret = AUDIO_ERROR; | |
} | |
} | |
else | |
{ | |
ret = AUDIO_ERROR; | |
} | |
} | |
else | |
{ | |
/* Analog Input */ | |
ret = AUDIO_ERROR; | |
} | |
/* Return AUDIO_OK when all operations are correctly done */ | |
return ret; | |
} | |
/** | |
* @brief Initializes wave recording and playback in parallel. | |
* @param InputDevice: INPUT_DEVICE_DIGITAL_MICROPHONE_2 | |
* @param OutputDevice: OUTPUT_DEVICE_SPEAKER, OUTPUT_DEVICE_HEADPHONE, | |
* or OUTPUT_DEVICE_BOTH. | |
* @param AudioFreq: Audio frequency to be configured for the SAI peripheral. | |
* @param BitRes: Audio frequency to be configured. | |
* @param ChnlNbr: Channel number. | |
* @retval AUDIO_OK if correct communication, else wrong communication | |
*/ | |
uint8_t BSP_AUDIO_IN_OUT_Init(uint32_t InputDevice, uint32_t OutputDevice, uint32_t AudioFreq, uint32_t BitRes, uint32_t ChnlNbr) | |
{ | |
uint32_t slot_active; | |
uint32_t deviceid = 0, ret = AUDIO_OK; | |
/* Store the audio record context */ | |
hAudioIn.Frequency = AudioFreq; | |
hAudioIn.BitResolution = BitRes; | |
hAudioIn.InputDevice = InputDevice; | |
hAudioIn.ChannelNbr = ChnlNbr; | |
/* Input device is Digital MIC2 and Codec interface is SAI */ | |
if (hAudioIn.InputDevice == INPUT_DEVICE_DIGITAL_MICROPHONE_2) | |
{ | |
haudio_in_sai.Instance = AUDIO_IN_SAIx; | |
haudio_out_sai.Instance = AUDIO_OUT_SAIx; | |
/* PLL clock is set depending on the AudioFreq (44.1khz vs 48khz groups) */ | |
BSP_AUDIO_OUT_ClockConfig(&haudio_in_sai, AudioFreq, NULL); | |
/* SAI data transfer preparation: | |
Prepare the Media to be used for the audio transfer from SAI peripheral to memory */ | |
if(HAL_SAI_GetState(&haudio_in_sai) == HAL_SAI_STATE_RESET) | |
{ | |
/* Init the SAI MSP: this __weak function can be redefined by the application*/ | |
BSP_AUDIO_IN_MspInit(); | |
} | |
/* SAI data transfer preparation: | |
Prepare the Media to be used for the audio transfer from memory to SAI peripheral */ | |
if(HAL_SAI_GetState(&haudio_out_sai) == HAL_SAI_STATE_RESET) | |
{ | |
/* Init the SAI MSP: this __weak function can be redefined by the application*/ | |
BSP_AUDIO_OUT_MspInit(&haudio_out_sai, NULL); | |
} | |
/* Configure SAI in master TX mode : | |
* - SAI2_block_A in master TX mode | |
* - SAI2_block_B in slave RX mode synchronous from SAI2_block_A | |
*/ | |
slot_active = CODEC_AUDIOFRAME_SLOT_13; | |
SAIx_In_Init(SAI_MODESLAVE_RX, slot_active, AudioFreq); | |
slot_active = CODEC_AUDIOFRAME_SLOT_02; | |
SAIx_Out_Init(SAI_MODEMASTER_TX, slot_active, AudioFreq); | |
/* wm8994 codec initialization */ | |
deviceid = wm8994_drv.ReadID(AUDIO_I2C_ADDRESS); | |
if((deviceid) == WM8994_ID) | |
{ | |
/* Reset the Codec Registers */ | |
wm8994_drv.Reset(AUDIO_I2C_ADDRESS); | |
/* Initialize the audio driver structure */ | |
audio_drv = &wm8994_drv; | |
ret = AUDIO_OK; | |
} | |
else | |
{ | |
ret = AUDIO_ERROR; | |
} | |
if(ret == AUDIO_OK) | |
{ | |
/* Initialize the codec internal registers */ | |
audio_drv->Init(AUDIO_I2C_ADDRESS, InputDevice|OutputDevice, 90, AudioFreq); | |
} | |
} | |
else | |
{ | |
ret = AUDIO_ERROR; | |
} | |
/* Return AUDIO_OK when all operations are correctly done */ | |
return ret; | |
} | |
/** | |
* @brief Link digital mic to specified source | |
* @param Interface : Audio In interface for Digital mic. It can be: | |
* AUDIO_IN_INTERFACE_SAI | |
* AUDIO_IN_INTERFACE_PDM | |
* @retval None | |
*/ | |
void BSP_AUDIO_IN_SelectInterface(uint32_t Interface) | |
{ | |
hAudioIn.Interface = Interface; | |
} | |
/** | |
* @brief Get digital mic interface | |
* @retval Digital mic interface. | |
*/ | |
uint32_t BSP_AUDIO_IN_GetInterface(void) | |
{ | |
return (hAudioIn.Interface); | |
} | |
/** | |
* @brief Return audio in channel number | |
* @retval Number of channel | |
*/ | |
uint8_t BSP_AUDIO_IN_GetChannelNumber(void) | |
{ | |
return hAudioIn.ChannelNbr; | |
} | |
/** | |
* @brief Start audio recording. | |
* @param pBuf: Main buffer pointer for the recorded data storing | |
* @param size: Current size of the recorded buffer | |
* @retval AUDIO_OK if correct communication, else wrong communication | |
*/ | |
uint8_t BSP_AUDIO_IN_Record(uint16_t *pBuf, uint32_t size) | |
{ | |
/* Start the process receive DMA */ | |
if(HAL_OK != HAL_SAI_Receive_DMA(&haudio_in_sai, (uint8_t*)pBuf, size)) | |
{ | |
return AUDIO_ERROR; | |
} | |
/* Return AUDIO_OK when all operations are correctly done */ | |
return AUDIO_OK; | |
} | |
/** | |
* @brief Stop audio recording. | |
* @retval AUDIO_OK if correct communication, else wrong communication | |
*/ | |
uint8_t BSP_AUDIO_IN_Stop(void) | |
{ | |
/* Call the Media layer stop function */ | |
HAL_SAI_DMAStop(&haudio_in_sai); | |
/* Return AUDIO_OK when all operations are correctly done */ | |
return AUDIO_OK; | |
} | |
/** | |
* @brief Pause the audio file stream. | |
* @retval AUDIO_OK if correct communication, else wrong communication | |
*/ | |
uint8_t BSP_AUDIO_IN_Pause(void) | |
{ | |
if (hAudioIn.InputDevice == INPUT_DEVICE_ANALOG_MIC) | |
{ | |
return AUDIO_ERROR; | |
} | |
else | |
{ | |
/* Call the Media layer pause function */ | |
HAL_SAI_DMAPause(&haudio_in_sai); | |
} | |
/* Return AUDIO_OK when all operations are correctly done */ | |
return AUDIO_OK; | |
} | |
/** | |
* @brief Resume the audio file stream. | |
* @retval AUDIO_OK if correct communication, else wrong communication | |
*/ | |
uint8_t BSP_AUDIO_IN_Resume(void) | |
{ | |
if (hAudioIn.InputDevice == INPUT_DEVICE_ANALOG_MIC) | |
{ | |
return AUDIO_ERROR; | |
} | |
else | |
{ | |
/* Call the Media layer resume function */ | |
HAL_SAI_DMAResume(&haudio_in_sai); | |
} | |
/* Return AUDIO_OK when all operations are correctly done */ | |
return AUDIO_OK; | |
} | |
/** | |
* @brief Controls the audio in volume level. | |
* @param Volume: Volume level to be set in percentage from 0% to 100% (0 for | |
* Mute and 100 for Max volume level). | |
* @retval AUDIO_OK if correct communication, else wrong communication | |
*/ | |
uint8_t BSP_AUDIO_IN_SetVolume(uint8_t Volume) | |
{ | |
/* Set the Global variable AudioInVolume */ | |
AudioInVolume = Volume; | |
/* Return AUDIO_OK when all operations are correctly done */ | |
return AUDIO_OK; | |
} | |
/** | |
* @brief Deinit the audio IN peripherals. | |
* @retval None | |
*/ | |
void BSP_AUDIO_IN_DeInit(void) | |
{ | |
SAIx_In_DeInit(&haudio_in_sai); | |
BSP_AUDIO_IN_MspDeInit(); | |
} | |
/** | |
* @brief Initialize the PDM library. | |
* @param AudioFreq: Audio sampling frequency | |
* @param ChnlNbrIn: Number of input audio channels in the PDM buffer | |
* @param ChnlNbrOut: Number of desired output audio channels in the resulting PCM buffer | |
* @retval None | |
*/ | |
uint8_t BSP_AUDIO_IN_PDMToPCM_Init(uint32_t AudioFreq, uint32_t ChnlNbrIn, uint32_t ChnlNbrOut) | |
{ | |
uint32_t index = 0; | |
/* Enable CRC peripheral to unlock the PDM library */ | |
__HAL_RCC_CRC_CLK_ENABLE(); | |
for(index = 0; index < ChnlNbrIn; index++) | |
{ | |
/* Init PDM filters */ | |
PDM_FilterHandler[index].bit_order = PDM_FILTER_BIT_ORDER_MSB; | |
PDM_FilterHandler[index].endianness = PDM_FILTER_ENDIANNESS_LE; | |
PDM_FilterHandler[index].high_pass_tap = 2122358088; | |
PDM_FilterHandler[index].out_ptr_channels = ChnlNbrOut; | |
PDM_FilterHandler[index].in_ptr_channels = ChnlNbrIn; | |
PDM_Filter_Init((PDM_Filter_Handler_t *)(&PDM_FilterHandler[index])); | |
/* PDM lib config phase */ | |
PDM_FilterConfig[index].output_samples_number = AudioFreq/1000; | |
PDM_FilterConfig[index].mic_gain = 24; | |
PDM_FilterConfig[index].decimation_factor = PDM_FILTER_DEC_FACTOR_64; | |
PDM_Filter_setConfig((PDM_Filter_Handler_t *)&PDM_FilterHandler[index], &PDM_FilterConfig[index]); | |
} | |
return AUDIO_OK; | |
} | |
/** | |
* @brief Converts audio format from PDM to PCM. | |
* @param PDMBuf: Pointer to PDM buffer data | |
* @param PCMBuf: Pointer to PCM buffer data | |
* @retval AUDIO_OK in case of success, AUDIO_ERROR otherwise | |
*/ | |
uint8_t BSP_AUDIO_IN_PDMToPCM(uint16_t *PDMBuf, uint16_t *PCMBuf) | |
{ | |
uint32_t index = 0; | |
for(index = 0; index < hAudioIn.ChannelNbr; index++) | |
{ | |
PDM_Filter(&((uint8_t*)(PDMBuf))[index], (uint16_t*)&(PCMBuf[index]), &PDM_FilterHandler[index]); | |
} | |
return AUDIO_OK; | |
} | |
/** | |
* @brief User callback when record buffer is filled. | |
* @retval None | |
*/ | |
__weak void BSP_AUDIO_IN_TransferComplete_CallBack(void) | |
{ | |
/* This function should be implemented by the user application. | |
It is called into this driver when the current buffer is filled | |
to prepare the next buffer pointer and its size. */ | |
} | |
/** | |
* @brief Manages the DMA Half Transfer complete event. | |
* @retval None | |
*/ | |
__weak void BSP_AUDIO_IN_HalfTransfer_CallBack(void) | |
{ | |
/* This function should be implemented by the user application. | |
It is called into this driver when the current buffer is filled | |
to prepare the next buffer pointer and its size. */ | |
} | |
/** | |
* @brief User callback when record buffer is filled. | |
* @param InputDevice: INPUT_DEVICE_DIGITAL_MIC1 or INPUT_DEVICE_DIGITAL_MIC2 | |
*/ | |
__weak void BSP_AUDIO_IN_TransferComplete_CallBackEx(uint32_t InputDevice) | |
{ | |
/* This function should be implemented by the user application. | |
It is called into this driver when the current buffer is filled | |
to prepare the next buffer pointer and its size. */ | |
} | |
/** | |
* @brief User callback when record buffer is filled. | |
* @param InputDevice: INPUT_DEVICE_DIGITAL_MIC1 or INPUT_DEVICE_DIGITAL_MIC2 | |
*/ | |
__weak void BSP_AUDIO_IN_HalfTransfer_CallBackEx(uint32_t InputDevice) | |
{ | |
/* This function should be implemented by the user application. | |
It is called into this driver when the current buffer is filled | |
to prepare the next buffer pointer and its size. */ | |
} | |
/** | |
* @brief Audio IN Error callback function. | |
* @retval None | |
*/ | |
__weak void BSP_AUDIO_IN_Error_CallBack(void) | |
{ | |
/* This function is called when an Interrupt due to transfer error on or peripheral | |
error occurs. */ | |
} | |
/** | |
* @brief Initialize BSP_AUDIO_IN MSP. | |
* @retval None | |
*/ | |
__weak void BSP_AUDIO_IN_MspInit(void) | |
{ | |
SAIx_In_MspInit(&haudio_in_sai, NULL); | |
} | |
/** | |
* @brief DeInitialize BSP_AUDIO_IN MSP. | |
* @retval None | |
*/ | |
__weak void BSP_AUDIO_IN_MspDeInit(void) | |
{ | |
SAIx_In_MspDeInit(&haudio_in_sai, NULL); | |
} | |
/** | |
* @brief Clock Config. | |
* @param AudioFreq: Audio frequency used to play the audio stream. | |
* @param Params: pointer on additional configuration parameters, can be NULL. | |
* @note This API is called by BSP_AUDIO_IN_Init() | |
* Being __weak it can be overwritten by the application | |
* @retval None | |
*/ | |
__weak void BSP_AUDIO_IN_ClockConfig(uint32_t AudioFreq, void *Params) | |
{ | |
RCC_PeriphCLKInitTypeDef rcc_ex_clk_init_struct; | |
HAL_RCCEx_GetPeriphCLKConfig(&rcc_ex_clk_init_struct); | |
/* Set the PLL configuration according to the audio frequency */ | |
if((AudioFreq == AUDIO_FREQUENCY_11K) || (AudioFreq == AUDIO_FREQUENCY_22K) || (AudioFreq == AUDIO_FREQUENCY_44K)) | |
{ | |
/* SAI clock config: | |
PLL2_VCO Input = HSE_VALUE/PLL2M = 1 Mhz | |
PLL2_VCO Output = PLL2_VCO Input * PLL2N = 429 Mhz | |
SAI_CLK_x = PLL2_VCO Output/PLL2P = 429/38 = 11.289 Mhz */ | |
rcc_ex_clk_init_struct.PeriphClockSelection = RCC_PERIPHCLK_SAI2; | |
rcc_ex_clk_init_struct.Sai23ClockSelection = RCC_SAI2CLKSOURCE_PLL2; | |
rcc_ex_clk_init_struct.PLL2.PLL2P = 38; | |
rcc_ex_clk_init_struct.PLL2.PLL2Q = 1; | |
rcc_ex_clk_init_struct.PLL2.PLL2R = 1; | |
rcc_ex_clk_init_struct.PLL2.PLL2N = 429; | |
rcc_ex_clk_init_struct.PLL2.PLL2M = 25; | |
if (hAudioIn.Interface == AUDIO_IN_INTERFACE_PDM) | |
{ | |
rcc_ex_clk_init_struct.PeriphClockSelection = RCC_PERIPHCLK_SAI4A; | |
rcc_ex_clk_init_struct.Sai4AClockSelection = RCC_SAI4ACLKSOURCE_PLL2; | |
} | |
HAL_RCCEx_PeriphCLKConfig(&rcc_ex_clk_init_struct); | |
} | |
else /* AUDIO_FREQUENCY_8K, AUDIO_FREQUENCY_16K, AUDIO_FREQUENCY_32K, AUDIO_FREQUENCY_48K, AUDIO_FREQUENCY_96K */ | |
{ | |
/* SAI clock config: | |
PLL2_VCO Input = HSE_VALUE/PLL2M = 1 Mhz | |
PLL2_VCO Output = PLL2_VCO Input * PLL2N = 344 Mhz | |
SAI_CLK_x = PLL2_VCO Output/PLL2P = 344/7 = 49.142 Mhz */ | |
rcc_ex_clk_init_struct.PeriphClockSelection = RCC_PERIPHCLK_SAI2; | |
rcc_ex_clk_init_struct.Sai23ClockSelection = RCC_SAI2CLKSOURCE_PLL2; | |
rcc_ex_clk_init_struct.PLL2.PLL2P = 7; | |
rcc_ex_clk_init_struct.PLL2.PLL2Q = 1; | |
rcc_ex_clk_init_struct.PLL2.PLL2R = 1; | |
rcc_ex_clk_init_struct.PLL2.PLL2N = 344; | |
rcc_ex_clk_init_struct.PLL2.PLL2M = 25; | |
if (hAudioIn.Interface == AUDIO_IN_INTERFACE_PDM) | |
{ | |
rcc_ex_clk_init_struct.PeriphClockSelection = RCC_PERIPHCLK_SAI4A; | |
rcc_ex_clk_init_struct.Sai4AClockSelection = RCC_SAI4ACLKSOURCE_PLL2; | |
} | |
HAL_RCCEx_PeriphCLKConfig(&rcc_ex_clk_init_struct); | |
} | |
} | |
/** | |
* @} | |
*/ | |
/** @defgroup STM32H745I_DISCOVERY_AUDIO_IN_Private_Functions IN Private Functions | |
* @{ | |
*/ | |
/******************************************************************************* | |
HAL Callbacks | |
*******************************************************************************/ | |
/** | |
* @brief Half reception complete callback. | |
* @param hsai: SAI handle. | |
* @retval None | |
*/ | |
void HAL_SAI_RxHalfCpltCallback(SAI_HandleTypeDef *hsai) | |
{ | |
/* Manage the remaining file size and new address offset: This function should be coded by user */ | |
BSP_AUDIO_IN_HalfTransfer_CallBack(); | |
} | |
/** | |
* @brief Reception complete callback. | |
* @param hsai: SAI handle. | |
* @retval None | |
*/ | |
void HAL_SAI_RxCpltCallback(SAI_HandleTypeDef *hsai) | |
{ | |
/* Call the record update function to get the next buffer to fill and its size (size is ignored) */ | |
BSP_AUDIO_IN_TransferComplete_CallBack(); | |
} | |
/******************************************************************************* | |
Static Functions | |
*******************************************************************************/ | |
/** | |
* @brief Initializes SAI Audio IN MSP. | |
* @param hsai: SAI handle | |
* @param Params: pointer on additional configuration parameters, can be NULL. | |
* @retval None | |
*/ | |
static void SAIx_In_MspInit(SAI_HandleTypeDef *hsai, void *Params) | |
{ | |
static DMA_HandleTypeDef hdma_sai_rx; | |
GPIO_InitTypeDef gpio_init_structure; | |
if(hsai->Instance == AUDIO_IN_SAI_PDMx) | |
{ | |
/* Enable SAI clock */ | |
AUDIO_IN_SAI_PDMx_CLK_ENABLE(); | |
/* Enable PDM clock */ | |
AUDIO_IN_SAI_PDMx_CLK_IN_ENABLE(); | |
gpio_init_structure.Pin = AUDIO_IN_SAI_PDMx_CLK_IN_PIN; | |
gpio_init_structure.Mode = GPIO_MODE_AF_PP; | |
gpio_init_structure.Pull = GPIO_NOPULL; | |
gpio_init_structure.Speed = GPIO_SPEED_FREQ_HIGH; | |
gpio_init_structure.Alternate = AUDIO_IN_SAI_PDMx_DATA_CLK_AF; | |
HAL_GPIO_Init(AUDIO_IN_SAI_PDMx_CLK_IN_PORT, &gpio_init_structure); | |
/* Enable PDM data */ | |
AUDIO_IN_SAI_PDMx_DATA_IN_ENABLE(); | |
gpio_init_structure.Pull = GPIO_PULLUP; | |
gpio_init_structure.Speed = GPIO_SPEED_FREQ_MEDIUM; | |
gpio_init_structure.Pin = AUDIO_IN_SAI_PDMx_DATA_IN_PIN; | |
HAL_GPIO_Init(AUDIO_IN_SAI_PDMx_DATA_IN_PORT, &gpio_init_structure); | |
/* Enable the DMA clock */ | |
AUDIO_IN_SAI_PDMx_DMAx_CLK_ENABLE(); | |
/* Configure the hdma_sai_rx handle parameters */ | |
hdma_sai_rx.Init.Request = AUDIO_IN_SAI_PDMx_DMAx_REQUEST; | |
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 = AUDIO_IN_SAI_PDMx_DMAx_PERIPH_DATA_SIZE; | |
hdma_sai_rx.Init.MemDataAlignment = AUDIO_IN_SAI_PDMx_DMAx_MEM_DATA_SIZE; | |
hdma_sai_rx.Init.Mode = DMA_CIRCULAR; | |
hdma_sai_rx.Init.Priority = DMA_PRIORITY_HIGH; | |
hdma_sai_rx.Init.FIFOMode = DMA_FIFOMODE_DISABLE; | |
hdma_sai_rx.Init.FIFOThreshold = DMA_FIFO_THRESHOLD_FULL; | |
hdma_sai_rx.Init.MemBurst = DMA_MBURST_SINGLE; | |
hdma_sai_rx.Init.PeriphBurst = DMA_MBURST_SINGLE; | |
hdma_sai_rx.Instance = AUDIO_IN_SAI_PDMx_DMAx_STREAM; | |
/* Associate the DMA handle */ | |
__HAL_LINKDMA(hsai, hdmarx, hdma_sai_rx); | |
/* Deinitialize the Stream for new transfer */ | |
HAL_DMA_DeInit(&hdma_sai_rx); | |
/* Configure the DMA Stream */ | |
HAL_DMA_Init(&hdma_sai_rx); | |
/* SAI DMA IRQ Channel configuration */ | |
HAL_NVIC_SetPriority(AUDIO_IN_SAI_PDMx_DMAx_IRQ, AUDIO_IN_IRQ_PREPRIO, 0); | |
HAL_NVIC_EnableIRQ(AUDIO_IN_SAI_PDMx_DMAx_IRQ); | |
} | |
else | |
{ | |
/* Enable SAI clock */ | |
AUDIO_IN_SAIx_CLK_ENABLE(); | |
/* Enable SD GPIO clock */ | |
AUDIO_IN_SAIx_SD_ENABLE(); | |
/* CODEC_SAI pin configuration: SD pin */ | |
gpio_init_structure.Pin = AUDIO_IN_SAIx_SD_PIN; | |
gpio_init_structure.Mode = GPIO_MODE_AF_PP; | |
gpio_init_structure.Pull = GPIO_NOPULL; | |
gpio_init_structure.Speed = GPIO_SPEED_FREQ_HIGH; | |
gpio_init_structure.Alternate = AUDIO_IN_SAIx_AF; | |
HAL_GPIO_Init(AUDIO_IN_SAIx_SD_GPIO_PORT, &gpio_init_structure); | |
/* Enable the DMA clock */ | |
AUDIO_IN_SAIx_DMAx_CLK_ENABLE(); | |
/* Configure the hdma_sai_rx handle parameters */ | |
hdma_sai_rx.Init.Request = AUDIO_IN_SAIx_DMAx_REQUEST; | |
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 = AUDIO_IN_SAIx_DMAx_PERIPH_DATA_SIZE; | |
hdma_sai_rx.Init.MemDataAlignment = AUDIO_IN_SAIx_DMAx_MEM_DATA_SIZE; | |
hdma_sai_rx.Init.Mode = DMA_CIRCULAR; | |
hdma_sai_rx.Init.Priority = DMA_PRIORITY_HIGH; | |
hdma_sai_rx.Init.FIFOMode = DMA_FIFOMODE_DISABLE; | |
hdma_sai_rx.Init.FIFOThreshold = DMA_FIFO_THRESHOLD_FULL; | |
hdma_sai_rx.Init.MemBurst = DMA_MBURST_SINGLE; | |
hdma_sai_rx.Init.PeriphBurst = DMA_MBURST_SINGLE; | |
hdma_sai_rx.Instance = AUDIO_IN_SAIx_DMAx_STREAM; | |
/* Associate the DMA handle */ | |
__HAL_LINKDMA(hsai, hdmarx, hdma_sai_rx); | |
/* Deinitialize the Stream for new transfer */ | |
HAL_DMA_DeInit(&hdma_sai_rx); | |
/* Configure the DMA Stream */ | |
HAL_DMA_Init(&hdma_sai_rx); | |
/* SAI DMA IRQ Channel configuration */ | |
HAL_NVIC_SetPriority(AUDIO_IN_SAIx_DMAx_IRQ, AUDIO_IN_IRQ_PREPRIO, 0); | |
HAL_NVIC_EnableIRQ(AUDIO_IN_SAIx_DMAx_IRQ); | |
} | |
} | |
/** | |
* @brief De-Initializes SAI Audio IN MSP. | |
* @param hsai: SAI handle | |
* @param Params: pointer on additional configuration parameters, can be NULL. | |
* @retval None | |
*/ | |
static void SAIx_In_MspDeInit(SAI_HandleTypeDef *hsai, void *Params) | |
{ | |
GPIO_InitTypeDef gpio_init_structure; | |
if(hsai->Instance == AUDIO_IN_SAI_PDMx) | |
{ | |
/* Deinitialize the DMA stream */ | |
HAL_DMA_Abort(hsai->hdmarx); | |
HAL_SAI_DeInit(hsai); | |
/* Disable SAI peripheral */ | |
__HAL_SAI_DISABLE(hsai); | |
/* Deinitialize the DMA stream */ | |
HAL_DMA_DeInit(hsai->hdmarx); | |
gpio_init_structure.Pin = AUDIO_IN_SAI_PDMx_CLK_IN_PIN; | |
HAL_GPIO_DeInit(AUDIO_IN_SAI_PDMx_CLK_IN_PORT, gpio_init_structure.Pin); | |
gpio_init_structure.Pin = AUDIO_IN_SAI_PDMx_DATA_IN_PIN; | |
HAL_GPIO_DeInit(AUDIO_IN_SAI_PDMx_DATA_IN_PORT, gpio_init_structure.Pin); | |
/* Disable SAI clock */ | |
AUDIO_IN_SAI_PDMx_CLK_DISABLE(); | |
} | |
else | |
{ | |
/* SAI DMA IRQ Channel deactivation */ | |
HAL_NVIC_DisableIRQ(AUDIO_IN_SAIx_DMAx_IRQ); | |
if(hsai->Instance == AUDIO_IN_SAIx) | |
{ | |
/* Deinitialize the DMA stream */ | |
HAL_DMA_DeInit(hsai->hdmatx); | |
} | |
/* Disable SAI peripheral */ | |
__HAL_SAI_DISABLE(hsai); | |
/* Deactivates CODEC_SAI pin SD by putting them in input mode */ | |
gpio_init_structure.Pin = AUDIO_IN_SAIx_SD_PIN; | |
HAL_GPIO_DeInit(AUDIO_IN_SAIx_SD_GPIO_PORT, gpio_init_structure.Pin); | |
/* Disable SAI clock */ | |
AUDIO_IN_SAIx_CLK_DISABLE(); | |
} | |
} | |
/** | |
* @brief Initializes the Audio Codec audio interface (SAI). | |
* @param SaiInMode: Audio mode to be configured for the SAI peripheral. | |
* @param SlotActive: Audio active slot to be configured for the SAI peripheral. | |
* @param AudioFreq: Audio frequency to be configured for the SAI peripheral. | |
* @retval None | |
*/ | |
static void SAIx_In_Init(uint32_t SaiInMode, uint32_t SlotActive, uint32_t AudioFreq) | |
{ | |
/* Disable SAI peripheral to allow access to SAI internal registers */ | |
__HAL_SAI_DISABLE(&haudio_in_sai); | |
/* Configure SAI_Block_x | |
LSBFirst: Disabled | |
DataSize: 16 */ | |
haudio_in_sai.Init.MonoStereoMode = SAI_STEREOMODE; | |
haudio_in_sai.Init.AudioFrequency = AudioFreq; | |
haudio_in_sai.Init.AudioMode = SaiInMode; | |
haudio_in_sai.Init.NoDivider = SAI_MASTERDIVIDER_ENABLE; | |
haudio_in_sai.Init.Protocol = SAI_FREE_PROTOCOL; | |
haudio_in_sai.Init.DataSize = SAI_DATASIZE_16; | |
haudio_in_sai.Init.FirstBit = SAI_FIRSTBIT_MSB; | |
haudio_in_sai.Init.ClockStrobing = SAI_CLOCKSTROBING_RISINGEDGE; | |
haudio_in_sai.Init.Synchro = SAI_SYNCHRONOUS; | |
haudio_in_sai.Init.OutputDrive = SAI_OUTPUTDRIVE_DISABLE; | |
haudio_in_sai.Init.FIFOThreshold = SAI_FIFOTHRESHOLD_1QF; | |
haudio_in_sai.Init.SynchroExt = SAI_SYNCEXT_DISABLE; | |
haudio_in_sai.Init.CompandingMode = SAI_NOCOMPANDING; | |
haudio_in_sai.Init.TriState = SAI_OUTPUT_RELEASED; | |
haudio_in_sai.Init.Mckdiv = 0; | |
haudio_in_sai.Init.MckOverSampling = SAI_MCK_OVERSAMPLING_DISABLE; | |
haudio_in_sai.Init.PdmInit.Activation = DISABLE; | |
/* Configure SAI_Block_x Frame | |
Frame Length: 64 | |
Frame active Length: 32 | |
FS Definition: Start frame + Channel Side identification | |
FS Polarity: FS active Low | |
FS Offset: FS asserted one bit before the first bit of slot 0 */ | |
haudio_in_sai.FrameInit.FrameLength = 128; | |
haudio_in_sai.FrameInit.ActiveFrameLength = 64; | |
haudio_in_sai.FrameInit.FSDefinition = SAI_FS_CHANNEL_IDENTIFICATION; | |
haudio_in_sai.FrameInit.FSPolarity = SAI_FS_ACTIVE_LOW; | |
haudio_in_sai.FrameInit.FSOffset = SAI_FS_BEFOREFIRSTBIT; | |
/* Configure SAI Block_x Slot | |
Slot First Bit Offset: 0 | |
Slot Size : 16 | |
Slot Number: 4 | |
Slot Active: All slot active */ | |
haudio_in_sai.SlotInit.FirstBitOffset = 0; | |
haudio_in_sai.SlotInit.SlotSize = SAI_SLOTSIZE_DATASIZE; | |
haudio_in_sai.SlotInit.SlotNumber = 4; | |
haudio_in_sai.SlotInit.SlotActive = SlotActive; | |
if(hAudioIn.Interface == AUDIO_IN_INTERFACE_PDM) | |
{ | |
haudio_in_sai.Init.AudioFrequency = AudioFreq * 8; | |
haudio_in_sai.Init.Synchro = SAI_ASYNCHRONOUS; | |
haudio_in_sai.Init.NoDivider = SAI_MASTERDIVIDER_DISABLE; | |
haudio_in_sai.Init.PdmInit.Activation = ENABLE; | |
haudio_in_sai.Init.PdmInit.MicPairsNbr = 2; | |
haudio_in_sai.Init.PdmInit.ClockEnable = SAI_PDM_CLOCK2_ENABLE; | |
haudio_in_sai.Init.FirstBit = SAI_FIRSTBIT_LSB; | |
haudio_in_sai.Init.ClockStrobing = SAI_CLOCKSTROBING_FALLINGEDGE; | |
haudio_in_sai.FrameInit.FrameLength = 32; | |
haudio_in_sai.FrameInit.ActiveFrameLength = 1; | |
haudio_in_sai.FrameInit.FSDefinition = SAI_FS_STARTFRAME; | |
haudio_in_sai.FrameInit.FSPolarity = SAI_FS_ACTIVE_HIGH; | |
haudio_in_sai.FrameInit.FSOffset = SAI_FS_FIRSTBIT; | |
haudio_in_sai.SlotInit.SlotNumber = 2; | |
haudio_in_sai.SlotInit.SlotActive = SlotActive; | |
} | |
HAL_SAI_Init(&haudio_in_sai); | |
/* Enable SAI peripheral */ | |
__HAL_SAI_ENABLE(&haudio_in_sai); | |
} | |
/** | |
* @brief Deinitializes the output Audio Codec audio interface (SAI). | |
* @retval None | |
*/ | |
static void SAIx_In_DeInit(SAI_HandleTypeDef *hsai) | |
{ | |
/* Disable SAI peripheral */ | |
__HAL_SAI_DISABLE(hsai); | |
HAL_SAI_DeInit(hsai); | |
} | |
/** | |
* @} | |
*/ | |
/** | |
* @} | |
*/ | |
/** | |
* @} | |
*/ | |
/** | |
* @} | |
*/ | |
/************************ (C) COPYRIGHT STMicroelectronics *****END OF FILE****/ |