stm32h743i_eval_audio.c

Go to the documentation of this file.

/**
  ******************************************************************************
  * @file    stm32h743i_eval_audio.c
  * @author  MCD Application Team
  * @version V1.0.0
  * @date    21-April-2017
  * @brief   This file provides the Audio driver for the STM32H743I-EVAL
  *          board.
  @verbatim
  How To use this driver:
  -----------------------
   + This driver supports STM32H7xx devices on STM32H743I-EVAL (MB1246) Evaluation 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 or the MFX 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 STM32H743I-EVAL 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 stm32h743i_eval_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 (DFSDM filters and channels,
      Clock source for DFSDM 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.
      Note: On STM32H743I-EVAL, two DFSDM Channel/Filters are configured and their DMA streams are configured
            in CIRCULAR mode.
   + 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 stm32h743i_eval_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 three interfaces
     available on the STM32H743I-EVAL board: SAI, DFSDM 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 stm32h743i_eval_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 stm32h743i_eval_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>&copy; COPYRIGHT(c) 2017 STMicroelectronics</center></h2>
  *
  * Redistribution and use in source and binary forms, with or without modification,
  * are permitted provided that the following conditions are met:
  *   1. Redistributions of source code must retain the above copyright notice,
  *      this list of conditions and the following disclaimer.
  *   2. Redistributions in binary form must reproduce the above copyright notice,
  *      this list of conditions and the following disclaimer in the documentation
  *      and/or other materials provided with the distribution.
  *   3. Neither the name of STMicroelectronics nor the names of its contributors
  *      may be used to endorse or promote products derived from this software
  *      without specific prior written permission.
  *
  * THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS"
  * AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
  * IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
  * DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE
  * FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
  * DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR
  * SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER
  * CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY,
  * OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
  * OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
  *
  ******************************************************************************
  */
/* Includes ------------------------------------------------------------------*/
#include "stm32h743i_eval_audio.h"

/** @addtogroup BSP
  * @{
  */

/** @addtogroup STM32H743I_EVAL
  * @{
  */

/** @addtogroup STM32H743I_EVAL_AUDIO STM32H743I EVAL board BSP AUDIO
  * @brief    This file includes the low layer driver for wm8994 Audio Codec
  *           available on STM32H743I-EVAL board (MB1246).
  * @{
  */

/** @defgroup STM32H743I_EVAL_AUDIO_Private_Types AUDIO Private Types
  * @{
  */
/**
  * @}
  */

/** @defgroup STM32H743I_EVAL_AUDIO_Private_Defines AUDIO Private Defines
  * @{
  */
/**
  * @}
  */

/** @defgroup STM32H743I_EVAL_AUDIO_Private_Macros AUDIO Private Macros
  * @{
  */
#define DFSDM_OVER_SAMPLING(__FREQUENCY__) \
        (__FREQUENCY__ == AUDIO_FREQUENCY_8K)  ? 256 \
      : (__FREQUENCY__ == AUDIO_FREQUENCY_11K) ? 256 \
      : (__FREQUENCY__ == AUDIO_FREQUENCY_16K) ? 128 \
      : (__FREQUENCY__ == AUDIO_FREQUENCY_22K) ? 128 \
      : (__FREQUENCY__ == AUDIO_FREQUENCY_32K) ? 64 \
      : (__FREQUENCY__ == AUDIO_FREQUENCY_44K) ? 64  \
      : (__FREQUENCY__ == AUDIO_FREQUENCY_48K) ? 40 : 20  \

#define DFSDM_CLOCK_DIVIDER(__FREQUENCY__) \
        (__FREQUENCY__ == AUDIO_FREQUENCY_8K)  ? 24 \
      : (__FREQUENCY__ == AUDIO_FREQUENCY_11K) ? 4 \
      : (__FREQUENCY__ == AUDIO_FREQUENCY_16K) ? 24 \
      : (__FREQUENCY__ == AUDIO_FREQUENCY_22K) ? 4 \
      : (__FREQUENCY__ == AUDIO_FREQUENCY_32K) ? 24 \
      : (__FREQUENCY__ == AUDIO_FREQUENCY_44K) ? 4  \
      : (__FREQUENCY__ == AUDIO_FREQUENCY_48K) ? 25 : 25  \

#define DFSDM_FILTER_ORDER(__FREQUENCY__) \
        (__FREQUENCY__ == AUDIO_FREQUENCY_8K)  ? DFSDM_FILTER_SINC3_ORDER \
      : (__FREQUENCY__ == AUDIO_FREQUENCY_11K) ? DFSDM_FILTER_SINC3_ORDER \
      : (__FREQUENCY__ == AUDIO_FREQUENCY_16K) ? DFSDM_FILTER_SINC3_ORDER \
      : (__FREQUENCY__ == AUDIO_FREQUENCY_22K) ? DFSDM_FILTER_SINC3_ORDER \
      : (__FREQUENCY__ == AUDIO_FREQUENCY_32K) ? DFSDM_FILTER_SINC4_ORDER \
      : (__FREQUENCY__ == AUDIO_FREQUENCY_44K) ? DFSDM_FILTER_SINC3_ORDER  \
      : (__FREQUENCY__ == AUDIO_FREQUENCY_48K) ? DFSDM_FILTER_SINC3_ORDER : DFSDM_FILTER_SINC5_ORDER  \

#define DFSDM_MIC_BIT_SHIFT(__FREQUENCY__) \
        (__FREQUENCY__ == AUDIO_FREQUENCY_8K)  ? 8 \
      : (__FREQUENCY__ == AUDIO_FREQUENCY_11K) ? 8 \
      : (__FREQUENCY__ == AUDIO_FREQUENCY_16K) ? 3 \
      : (__FREQUENCY__ == AUDIO_FREQUENCY_22K) ? 4 \
      : (__FREQUENCY__ == AUDIO_FREQUENCY_32K) ? 7 \
      : (__FREQUENCY__ == AUDIO_FREQUENCY_44K) ? 2  \
      : (__FREQUENCY__ == AUDIO_FREQUENCY_48K) ? 0 : 4  \

/* Saturate the record PCM sample */
#define SaturaLH(N, L, H) (((N)<(L))?(L):(((N)>(H))?(H):(N)))

/**
  * @}
  */

/** @defgroup STM32H743I_EVAL_AUDIO_Private_Variables AUDIO Private Variables
  * @{
  */
/* PLAY */
AUDIO_DrvTypeDef                *audio_drv;
SAI_HandleTypeDef               haudio_out_sai;
SAI_HandleTypeDef               haudio_in_sai;

/* RECORD */
AUDIOIN_ContextTypeDef          hAudioIn;

DFSDM_Channel_HandleTypeDef     hAudioInDfsdmChannel[DFSDM_MIC_NUMBER];  /* 2 DFSDM channel handle used for all microphones */
DFSDM_Filter_HandleTypeDef      hAudioInDfsdmFilter[DFSDM_MIC_NUMBER];   /* 2 DFSDM filter handle */
DMA_HandleTypeDef               hDmaDfsdm[DFSDM_MIC_NUMBER];             /* 2 DMA handle used for DFSDM regular conversions */

/* Buffers for right and left samples */
int32_t                         *pScratchBuff[DEFAULT_AUDIO_IN_CHANNEL_NBR];
int32_t                         ScratchSize;

uint32_t                        DmaRecHalfBuffCplt[DFSDM_MIC_NUMBER]  = {0};
uint32_t                        DmaRecBuffCplt[DFSDM_MIC_NUMBER]  = {0};


/* Application Buffer Trigger */
__IO uint32_t                   AppBuffTrigger          = 0;
__IO uint32_t                   AppBuffHalf             = 0;
__IO uint32_t                   MicBuff[DFSDM_MIC_NUMBER] = {0};
__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 STM32H743I_EVAL_AUDIO_OUT_Private_Function_Prototypes AUDIO 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 STM32H743I_EVAL_AUDIO_IN_Private_Function_Prototypes AUDIO 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);
static uint8_t DFSDMx_DeInit(void);
static void DFSDMx_ChannelMspInit(void);
static void DFSDMx_ChannelMspDeInit(void);
static void DFSDMx_FilterMspInit(void);
static void DFSDMx_FilterMspDeInit(void);

/**
  * @}
  */

/** @addtogroup STM32H743I_EVAL_AUDIO_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 SAI1 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
  *         This parameter can be any value of @ref CODEC_AUDIO_FRAME_SLOT_TDM
  * @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 pin if the PDM input interface is not activated
    (since if PDM input interface is activated the MCLK pin is configured in BSP_AUDIO_IN_MspInit) */
  if(hAudioIn.Interface != AUDIO_IN_INTERFACE_PDM && (haudio_in_sai.State != HAL_SAI_STATE_READY))
  {
    /* 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_SAI1;
    rcc_ex_clk_init_struct.Sai1ClockSelection = RCC_SAI1CLKSOURCE_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_SAI1;
    rcc_ex_clk_init_struct.Sai1ClockSelection = RCC_SAI1CLKSOURCE_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);
  }
}
/**
  * @}
  */

/** @addtogroup STM32H743I_EVAL_AUDIO_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 stm32h743i_eval_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 stm32h743i_eval_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);
}

/**
  * @}
  */

/** @addtogroup STM32H743I_EVAL_AUDIO_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_DFSDM);
  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 mic_enabled = 0;
  uint32_t slot_active;
  uint32_t i = 0;

  /* Store the audio record context */
  hAudioIn.Frequency     = AudioFreq;
  hAudioIn.BitResolution = BitRes;
  hAudioIn.InputDevice = InputDevice;
  hAudioIn.ChannelNbr = ChnlNbr;

  /* Store the total number of microphones enabled */
  for(i = 0; i < DFSDM_MIC_NUMBER; i ++)
  {
    if(((hAudioIn.InputDevice >> i) & INPUT_DEVICE_DIGITAL_MIC1) == INPUT_DEVICE_DIGITAL_MIC1)
    {
      mic_enabled++;
    }
  }

  if(hAudioIn.InputDevice == INPUT_DEVICE_DIGITAL_MIC)
  {
    if(hAudioIn.Interface == AUDIO_IN_INTERFACE_DFSDM)
    {
      if(hAudioIn.ChannelNbr != mic_enabled)
      {
        return AUDIO_ERROR;
      }
      else
      {
        /* 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 for analog mic */

        /* Init the DFSDM MSP: this __weak function can be redefined by the application*/
        BSP_AUDIO_IN_MspInit();

        /* Default configuration of DFSDM filters and channels */
        ret = BSP_AUDIO_IN_ConfigDigitalMic(hAudioIn.InputDevice, NULL);
      }
    }
    else if(hAudioIn.Interface == AUDIO_IN_INTERFACE_SAI)
    {
      /* Initialize SAI1 block B as SLAVE RX synchrounous with SAI1 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 :
       *   - SAI1_block_B in slave RX mode synchronous from SAI1_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 SAI1 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 :
       *   - SAI1_block_A in master RX mode
       */
      slot_active = CODEC_AUDIOFRAME_SLOT_0;
      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 default configuration of the Digital Filter for Sigma-Delta Modulators interface (DFSDM).
  * @param  InputDevice: The microphone to be configured. Can be INPUT_DEVICE_DIGITAL_MIC1..INPUT_DEVICE_DIGITAL_MIC5
  * @note   Channel output Clock Divider and Filter Oversampling are calculated as follow:
  *         - Clock_Divider = CLK(input DFSDM)/CLK(micro) with
  *           1MHZ < CLK(micro) < 3.2MHZ (TYP 2.4MHZ for MP34DT01TR)
  *         - Oversampling = CLK(input DFSDM)/(Clock_Divider * AudioFreq)
  * @retval AUDIO_OK if correct communication, else wrong communication
  */
uint8_t BSP_AUDIO_IN_ConfigMicDefault(uint32_t InputDevice)
{
  uint32_t i = 0, mic_init[DFSDM_MIC_NUMBER] = {0};
  uint32_t filter_ch = 0, mic_num = 0;

  DFSDM_Filter_TypeDef* FilterInstnace[DFSDM_MIC_NUMBER] = {AUDIO_DFSDMx_MIC1_FILTER, AUDIO_DFSDMx_MIC2_FILTER};
  DFSDM_Channel_TypeDef* ChannelInstnace[DFSDM_MIC_NUMBER] = {AUDIO_DFSDMx_MIC1_CHANNEL, AUDIO_DFSDMx_MIC2_CHANNEL};
  uint32_t DigitalMicPins[DFSDM_MIC_NUMBER] = {DFSDM_CHANNEL_SAME_CHANNEL_PINS, DFSDM_CHANNEL_FOLLOWING_CHANNEL_PINS};
  uint32_t DigitalMicType[DFSDM_MIC_NUMBER] = {DFSDM_CHANNEL_SPI_RISING, DFSDM_CHANNEL_SPI_FALLING};
  uint32_t Channel4Filter[DFSDM_MIC_NUMBER] = {AUDIO_DFSDMx_MIC1_CHANNEL_FOR_FILTER, AUDIO_DFSDMx_MIC2_CHANNEL_FOR_FILTER};

  for(i = 0; i < hAudioIn.ChannelNbr; i++)
  {
    if(((hAudioIn.InputDevice & INPUT_DEVICE_DIGITAL_MIC1) == INPUT_DEVICE_DIGITAL_MIC1) && (mic_init[POS_VAL(INPUT_DEVICE_DIGITAL_MIC1)] != 1))
    {
      mic_num = POS_VAL(INPUT_DEVICE_DIGITAL_MIC1);
    }
    else if(((hAudioIn.InputDevice & INPUT_DEVICE_DIGITAL_MIC2) == INPUT_DEVICE_DIGITAL_MIC2) && (mic_init[POS_VAL(INPUT_DEVICE_DIGITAL_MIC2)] != 1))
    {
      mic_num = POS_VAL(INPUT_DEVICE_DIGITAL_MIC2);
    }

    mic_init[mic_num] = 1;

    HAL_DFSDM_ChannelDeInit(&hAudioInDfsdmChannel[mic_num]);
    /* MIC channels initialization */
    __HAL_DFSDM_CHANNEL_RESET_HANDLE_STATE(&hAudioInDfsdmChannel[mic_num]);
    hAudioInDfsdmChannel[mic_num].Init.OutputClock.Activation   = ENABLE;
    hAudioInDfsdmChannel[mic_num].Init.OutputClock.Selection    = DFSDM_CHANNEL_OUTPUT_CLOCK_AUDIO;
    hAudioInDfsdmChannel[mic_num].Init.OutputClock.Divider      = DFSDM_CLOCK_DIVIDER(hAudioIn.Frequency);
    hAudioInDfsdmChannel[mic_num].Init.Input.Multiplexer        = DFSDM_CHANNEL_EXTERNAL_INPUTS;
    hAudioInDfsdmChannel[mic_num].Init.Input.DataPacking        = DFSDM_CHANNEL_STANDARD_MODE;
    hAudioInDfsdmChannel[mic_num].Init.SerialInterface.SpiClock = DFSDM_CHANNEL_SPI_CLOCK_INTERNAL;
    hAudioInDfsdmChannel[mic_num].Init.Awd.FilterOrder          = DFSDM_CHANNEL_FASTSINC_ORDER;
    hAudioInDfsdmChannel[mic_num].Init.Awd.Oversampling         = 10;
    hAudioInDfsdmChannel[mic_num].Init.Offset                   = 0;
    hAudioInDfsdmChannel[mic_num].Init.RightBitShift            = DFSDM_MIC_BIT_SHIFT(hAudioIn.Frequency);
    hAudioInDfsdmChannel[mic_num].Instance                      = ChannelInstnace[mic_num];
    hAudioInDfsdmChannel[mic_num].Init.Input.Pins               = DigitalMicPins[mic_num];
    hAudioInDfsdmChannel[mic_num].Init.SerialInterface.Type     = DigitalMicType[mic_num];

    if(HAL_OK != HAL_DFSDM_ChannelInit(&hAudioInDfsdmChannel[mic_num]))
    {
      return AUDIO_ERROR;
    }

    HAL_DFSDM_FilterDeInit(&hAudioInDfsdmFilter[mic_num]);
    /* MIC filters  initialization */
    __HAL_DFSDM_FILTER_RESET_HANDLE_STATE(&hAudioInDfsdmFilter[mic_num]);
    hAudioInDfsdmFilter[mic_num].Instance                          = FilterInstnace[mic_num];
    hAudioInDfsdmFilter[mic_num].Init.RegularParam.Trigger         = DFSDM_FILTER_SW_TRIGGER;
    hAudioInDfsdmFilter[mic_num].Init.RegularParam.FastMode        = ENABLE;
    hAudioInDfsdmFilter[mic_num].Init.RegularParam.DmaMode         = ENABLE;
    hAudioInDfsdmFilter[mic_num].Init.InjectedParam.Trigger        = DFSDM_FILTER_SW_TRIGGER;
    hAudioInDfsdmFilter[mic_num].Init.InjectedParam.ScanMode       = ENABLE;
    hAudioInDfsdmFilter[mic_num].Init.InjectedParam.DmaMode        = DISABLE;
    hAudioInDfsdmFilter[mic_num].Init.InjectedParam.ExtTrigger     = DFSDM_FILTER_EXT_TRIG_TIM1_TRGO;
    hAudioInDfsdmFilter[mic_num].Init.InjectedParam.ExtTriggerEdge = DFSDM_FILTER_EXT_TRIG_RISING_EDGE;
    hAudioInDfsdmFilter[mic_num].Init.FilterParam.SincOrder        = DFSDM_FILTER_ORDER(hAudioIn.Frequency);
    hAudioInDfsdmFilter[mic_num].Init.FilterParam.Oversampling     = DFSDM_OVER_SAMPLING(hAudioIn.Frequency);
    hAudioInDfsdmFilter[mic_num].Init.FilterParam.IntOversampling  = 1;

    if(HAL_OK != HAL_DFSDM_FilterInit(&hAudioInDfsdmFilter[mic_num]))
    {
      return AUDIO_ERROR;
    }

    filter_ch = Channel4Filter[mic_num];
    /* Configure injected channel */
    if(HAL_OK != HAL_DFSDM_FilterConfigRegChannel(&hAudioInDfsdmFilter[mic_num], filter_ch, DFSDM_CONTINUOUS_CONV_ON))
    {
      return AUDIO_ERROR;
    }
  }

  return AUDIO_OK;
}

/**
  * @brief  Initializes the Digital Filter for Sigma-Delta Modulators interface (DFSDM).
  * @param  InputDevice: The microphone to be configured. Can be INPUT_DEVICE_DIGITAL_MIC1..INPUT_DEVICE_DIGITAL_MIC5
  * @param  Params: pointer on additional configuration parameters, can be NULL.
  * @retval AUDIO_OK if correct communication, else wrong communication
  */
__weak uint8_t BSP_AUDIO_IN_ConfigDigitalMic(uint32_t InputDevice, void *Params)
{
  /* Prevent unused argument(s) compilation warning */
  UNUSED(Params);

  /* Default configuration of DFSDM filters and channels */
  return(BSP_AUDIO_IN_ConfigMicDefault(InputDevice));
  /* Note: This function can be called at application level and default configuration
           can be ovewritten to fit user's need */
}

/**
  * @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 :
    *   - SAI1_block_A in master TX mode
    *   - SAI1_block_B in slave RX mode synchronous from SAI1_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
  *                       AUDIO_IN_INTERFACE_DFSDM
  * @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  Allocate channel buffer scratch
  * @param  pScratch : pointer to scratch tables.
  * @param  size of scratch buffer
  */
uint8_t BSP_AUDIO_IN_AllocScratch (int32_t *pScratch, uint32_t size)
{
  uint32_t idx;

  ScratchSize = (size / hAudioIn.ChannelNbr);

  /* copy scratch pointers */
  for (idx = 0; idx < hAudioIn.ChannelNbr; idx++)
  {
    pScratchBuff[idx] = (int32_t *)(pScratch + idx * ScratchSize);
  }
  /* Return AUDIO_OK */
  return AUDIO_OK;
}

/**
  * @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)
{
  if (hAudioIn.Interface == AUDIO_IN_INTERFACE_DFSDM)
  {
    hAudioIn.pRecBuf = pBuf;
    hAudioIn.RecSize = size;
    /* Reset Application Buffer Trigger */
    AppBuffTrigger = 0;
    AppBuffHalf = 0;

    /* Call the Media layer start function for MIC1 channel */
    if(HAL_OK != HAL_DFSDM_FilterRegularStart_DMA(&hAudioInDfsdmFilter[POS_VAL(INPUT_DEVICE_DIGITAL_MIC1)], pScratchBuff[POS_VAL(INPUT_DEVICE_DIGITAL_MIC1)], ScratchSize))
    {
      return AUDIO_ERROR;
    }

    /* Call the Media layer start function for MIC2 channel */
    if(HAL_OK != HAL_DFSDM_FilterRegularStart_DMA(&hAudioInDfsdmFilter[POS_VAL(INPUT_DEVICE_DIGITAL_MIC2)], pScratchBuff[POS_VAL(INPUT_DEVICE_DIGITAL_MIC2)], ScratchSize))
    {
      return AUDIO_ERROR;
    }
  }
  else
  {
    /* 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)
{
  if (hAudioIn.Interface == AUDIO_IN_INTERFACE_DFSDM)
  {
    AppBuffTrigger = 0;
    AppBuffHalf    = 0;

    /* Call the Media layer stop function for MIC1 channel */
    if(AUDIO_OK != BSP_AUDIO_IN_PauseEx(INPUT_DEVICE_DIGITAL_MIC1))
    {
      return AUDIO_ERROR;
    }

    /* Call the Media layer stop function for MIC2 channel */
    if(AUDIO_OK != BSP_AUDIO_IN_PauseEx(INPUT_DEVICE_DIGITAL_MIC2))
    {
      return AUDIO_ERROR;
    }
  }
  else
  {
    /* 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  Stops audio recording.
  * @param  InputDevice: Microphone to be stopped. Can be INPUT_DEVICE_DIGITAL_MIC1 or INPUT_DEVICE_DIGITAL_MIC2.
  * @retval AUDIO_OK if correct communication, else wrong communication
  */
uint8_t BSP_AUDIO_IN_StopEx(uint32_t InputDevice)
{
  if((InputDevice < INPUT_DEVICE_DIGITAL_MIC1) || (InputDevice > INPUT_DEVICE_DIGITAL_MIC2))
  {
    return AUDIO_ERROR;
  }
  else
  {
    BSP_AUDIO_IN_PauseEx(InputDevice);
  }

  /* 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 stop function */
    if(HAL_OK != HAL_DFSDM_FilterRegularStop_DMA(&hAudioInDfsdmFilter[POS_VAL(INPUT_DEVICE_DIGITAL_MIC1)]))
    {
      return AUDIO_ERROR;
    }

    /* Call the Media layer stop function */
    if(HAL_OK != HAL_DFSDM_FilterRegularStop_DMA(&hAudioInDfsdmFilter[POS_VAL(INPUT_DEVICE_DIGITAL_MIC2)]))
    {
      return AUDIO_ERROR;
    }
  }
  /* Return AUDIO_OK when all operations are correctly done */
  return AUDIO_OK;
}

/**
  * @brief  Pauses the audio file stream.
  * @param  InputDevice: Microphone to be paused. Can be INPUT_DEVICE_DIGITAL_MIC1 or INPUT_DEVICE_DIGITAL_MIC2.
  * @retval AUDIO_OK if correct communication, else wrong communication
  */
uint8_t BSP_AUDIO_IN_PauseEx(uint32_t InputDevice)
{
  if((InputDevice < INPUT_DEVICE_DIGITAL_MIC1) || (InputDevice > INPUT_DEVICE_DIGITAL_MIC2))
  {
    return AUDIO_ERROR;
  }
  else
  {
    /* Call the Media layer stop function */
    if(HAL_OK != HAL_DFSDM_FilterRegularStop_DMA(&hAudioInDfsdmFilter[POS_VAL(InputDevice)]))
    {
      return AUDIO_ERROR;
    }
  }
  /* 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 start function for MIC2 channel */
    if(HAL_OK != HAL_DFSDM_FilterRegularStart_DMA(&hAudioInDfsdmFilter[POS_VAL(INPUT_DEVICE_DIGITAL_MIC2)], pScratchBuff[POS_VAL(INPUT_DEVICE_DIGITAL_MIC2)], ScratchSize))
    {
      return AUDIO_ERROR;
    }

    /* Call the Media layer start function for MIC1 channel */
    if(HAL_OK != HAL_DFSDM_FilterRegularStart_DMA(&hAudioInDfsdmFilter[POS_VAL(INPUT_DEVICE_DIGITAL_MIC1)], pScratchBuff[POS_VAL(INPUT_DEVICE_DIGITAL_MIC1)], ScratchSize))
    {
      return AUDIO_ERROR;
    }
  }
  /* Return AUDIO_OK when all operations are correctly done */
  return AUDIO_OK;
}

/**
  * @brief  Resumes the audio file stream.
  * @param  pBuf: Main buffer pointer for the recorded data storing
  * @param  InputDevice: Microphone to be paused. Can be INPUT_DEVICE_DIGITAL_MIC1 or INPUT_DEVICE_DIGITAL_MIC2.
  * @retval AUDIO_OK if correct communication, else wrong communication
  */
uint8_t BSP_AUDIO_IN_ResumeEx(uint32_t *pBuf, uint32_t InputDevice)
{
  if((InputDevice < INPUT_DEVICE_DIGITAL_MIC1) || (InputDevice > INPUT_DEVICE_DIGITAL_MIC2))
  {
    return AUDIO_ERROR;
  }
  else
  {
    /* Call the Media layer stop function */
    if(HAL_OK != HAL_DFSDM_FilterRegularStart_DMA(&hAudioInDfsdmFilter[POS_VAL(InputDevice)], (int32_t*)pBuf[MicBuff[POS_VAL(InputDevice)]], hAudioIn.RecSize))
    {
      return AUDIO_ERROR;
    }
  }
  /* 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)
{
  if(hAudioIn.Interface == AUDIO_IN_INTERFACE_DFSDM)
  {
    DFSDMx_DeInit();
  }
  else
  {
    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 = 0;
    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)
{
  if (hAudioIn.Interface == AUDIO_IN_INTERFACE_DFSDM)
  {
    /* MSP channels initialization */
    DFSDMx_ChannelMspInit();

    /* MSP filters initialization */
    DFSDMx_FilterMspInit();
  }
  else
  {
    SAIx_In_MspInit(&haudio_in_sai, NULL);
  }
}

/**
  * @brief  DeInitialize BSP_AUDIO_IN MSP.
  * @retval None
  */
__weak void BSP_AUDIO_IN_MspDeInit(void)
{
  if (hAudioIn.Interface == AUDIO_IN_INTERFACE_DFSDM)
  {
    /* MSP channels initialization */
    DFSDMx_ChannelMspDeInit();
    /* MSP filters initialization */
    DFSDMx_FilterMspDeInit();
  }
  else
  {
    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_SAI1;
    rcc_ex_clk_init_struct.Sai1ClockSelection = RCC_SAI1CLKSOURCE_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_SAI1;
    rcc_ex_clk_init_struct.Sai1ClockSelection = RCC_SAI1CLKSOURCE_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);
  }
}
/**
  * @}
  */


/** @addtogroup STM32H743I_EVAL_AUDIO_IN_Private_Functions
  * @{
  */

/*******************************************************************************
                            HAL Callbacks
*******************************************************************************/
/**
  * @brief  Regular conversion complete callback.
  * @note   In interrupt mode, user has to read conversion value in this function
            using HAL_DFSDM_FilterGetRegularValue.
  * @param  hdfsdm_filter : DFSDM filter handle.
  * @retval None
  */
void HAL_DFSDM_FilterRegConvCpltCallback(DFSDM_Filter_HandleTypeDef *hdfsdm_filter)
{
  uint32_t index, input_device = 0;

  if(hdfsdm_filter->Instance == AUDIO_DFSDMx_MIC1_FILTER)
  {
    DmaRecBuffCplt[POS_VAL(INPUT_DEVICE_DIGITAL_MIC1)] = 1;
    input_device = INPUT_DEVICE_DIGITAL_MIC1;
  }
  else if(hdfsdm_filter->Instance == AUDIO_DFSDMx_MIC2_FILTER)
  {
    DmaRecBuffCplt[POS_VAL(INPUT_DEVICE_DIGITAL_MIC2)] = 1;
    input_device = INPUT_DEVICE_DIGITAL_MIC2;
  }

  if(hAudioIn.MultiBuffMode == 1)
  {
    BSP_AUDIO_IN_TransferComplete_CallBackEx(input_device);
  }
  else
  {
    if((DmaRecBuffCplt[POS_VAL(INPUT_DEVICE_DIGITAL_MIC2)] == 1) && (DmaRecBuffCplt[POS_VAL(INPUT_DEVICE_DIGITAL_MIC1)] == 1))
    {
      if(AppBuffTrigger >= hAudioIn.RecSize)
        AppBuffTrigger = 0;

      for(index = (ScratchSize/2) ; index < ScratchSize; index++)
      {
        hAudioIn.pRecBuf[AppBuffTrigger]     = (uint16_t)(SaturaLH((pScratchBuff[POS_VAL(INPUT_DEVICE_DIGITAL_MIC1)][index] >> 8), -32760, 32760));
        hAudioIn.pRecBuf[AppBuffTrigger + 1] = (uint16_t)(SaturaLH((pScratchBuff[POS_VAL(INPUT_DEVICE_DIGITAL_MIC2)][index] >> 8), -32760, 32760));
        AppBuffTrigger += 2;
      }
      DmaRecBuffCplt[POS_VAL(INPUT_DEVICE_DIGITAL_MIC2)] = DmaRecBuffCplt[POS_VAL(INPUT_DEVICE_DIGITAL_MIC1)] = 0;
    }

    /* Update Trigger with Remaining Byte before callback if necessary */
    if(AppBuffTrigger >= hAudioIn.RecSize)
    {
      /* Reset Application Buffer Trigger */
      AppBuffTrigger = 0;
      AppBuffHalf = 0;

      /* Call the record update function to get the next buffer to fill and its size (size is ignored) */
      BSP_AUDIO_IN_TransferComplete_CallBack();
    }
    else if((AppBuffTrigger >= hAudioIn.RecSize/2))
    {
      if(AppBuffHalf == 0)
      {
        AppBuffHalf = 1;
        /* Manage the remaining file size and new address offset: This function
        should be coded by user (its prototype is already declared in stm32xx_eval_audio.h) */
        BSP_AUDIO_IN_HalfTransfer_CallBack();
      }
    }
  }
}

/**
  * @brief  Half regular conversion complete callback.
  * @param  hdfsdm_filter : DFSDM filter handle.
  * @retval None
  */
void HAL_DFSDM_FilterRegConvHalfCpltCallback(DFSDM_Filter_HandleTypeDef *hdfsdm_filter)
{
  uint32_t index, input_device = 0;

  if(hdfsdm_filter->Instance == AUDIO_DFSDMx_MIC1_FILTER)
  {
    DmaRecHalfBuffCplt[POS_VAL(INPUT_DEVICE_DIGITAL_MIC1)] = 1;
    input_device = INPUT_DEVICE_DIGITAL_MIC1;
  }
  else if(hdfsdm_filter->Instance == AUDIO_DFSDMx_MIC2_FILTER)
  {
    DmaRecHalfBuffCplt[POS_VAL(INPUT_DEVICE_DIGITAL_MIC2)] = 1;
    input_device = INPUT_DEVICE_DIGITAL_MIC2;
  }

  if(hAudioIn.MultiBuffMode == 1)
  {
    BSP_AUDIO_IN_HalfTransfer_CallBackEx(input_device);
  }
  else
  {
    if((DmaRecHalfBuffCplt[POS_VAL(INPUT_DEVICE_DIGITAL_MIC2)] == 1) && (DmaRecHalfBuffCplt[POS_VAL(INPUT_DEVICE_DIGITAL_MIC1)] == 1))
    {
      if(AppBuffTrigger >= hAudioIn.RecSize)
        AppBuffTrigger = 0;

      for(index = 0; index < ScratchSize/2; index++)
      {
        hAudioIn.pRecBuf[AppBuffTrigger]     = (uint16_t)(SaturaLH((pScratchBuff[POS_VAL(INPUT_DEVICE_DIGITAL_MIC1)][index] >> 8), -32760, 32760));
        hAudioIn.pRecBuf[AppBuffTrigger + 1] = (uint16_t)(SaturaLH((pScratchBuff[POS_VAL(INPUT_DEVICE_DIGITAL_MIC2)][index] >> 8), -32760, 32760));
        AppBuffTrigger += 2;
      }
      DmaRecHalfBuffCplt[POS_VAL(INPUT_DEVICE_DIGITAL_MIC2)] = DmaRecHalfBuffCplt[POS_VAL(INPUT_DEVICE_DIGITAL_MIC1)] = 0;
    }


    /* Update Trigger with Remaining Byte before callback if necessary */
    if(AppBuffTrigger >= hAudioIn.RecSize)
    {
      /* Reset Application Buffer Trigger */
      AppBuffTrigger = 0;
      AppBuffHalf = 0;

      /* Call the record update function to get the next buffer to fill and its size (size is ignored) */
      BSP_AUDIO_IN_TransferComplete_CallBack();
    }
    else if((AppBuffTrigger >= hAudioIn.RecSize/2))
    {
      if(AppBuffHalf == 0)
      {
        AppBuffHalf = 1;
        /* Manage the remaining file size and new address offset: This function
        should be coded by user (its prototype is already declared in stm32xx_eval_audio.h) */
        BSP_AUDIO_IN_HalfTransfer_CallBack();
      }
    }
  }
}

/**
  * @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  De-initializes the Digital Filter for Sigma-Delta Modulators interface (DFSDM).
  * @retval AUDIO_OK if correct communication, else wrong communication
  */
static uint8_t DFSDMx_DeInit(void)
{
  for(uint32_t i = 0; i < DFSDM_MIC_NUMBER; i++)
  {
    if(hAudioInDfsdmFilter[i].Instance != NULL)
    {
      if(HAL_OK != HAL_DFSDM_FilterDeInit(&hAudioInDfsdmFilter[i]))
      {
        return AUDIO_ERROR;
      }
      hAudioInDfsdmFilter[i].Instance = NULL;
    }
    if(hAudioInDfsdmChannel[i].Instance != NULL)
    {
      if(HAL_OK != HAL_DFSDM_ChannelDeInit(&hAudioInDfsdmChannel[i]))
      {
        return AUDIO_ERROR;
      }
      hAudioInDfsdmChannel[i].Instance = NULL;
    }
  }
  return AUDIO_OK;
}

/**
  * @brief  Initializes the DFSDM channel MSP.
  */
static void DFSDMx_ChannelMspInit(void)
{
  GPIO_InitTypeDef  GPIO_InitStruct;

  /* Enable DFSDM clock */
  AUDIO_DFSDMx_CLK_ENABLE();

  /* Enable GPIO clock */
  AUDIO_DFSDMx_CKOUT_DMIC_GPIO_CLK_ENABLE();
  /* DFSDM pins configuration: DFSDM_CKOUT, DMIC_DATIN pins ------------------*/
  GPIO_InitStruct.Pin = AUDIO_DFSDMx_CKOUT_PIN;
  GPIO_InitStruct.Mode = GPIO_MODE_AF_PP;
  GPIO_InitStruct.Pull = GPIO_NOPULL;
  GPIO_InitStruct.Speed = GPIO_SPEED_FREQ_VERY_HIGH;
  GPIO_InitStruct.Alternate = AUDIO_DFSDMx_CKOUT_DMIC_AF;
  HAL_GPIO_Init(AUDIO_DFSDMx_CKOUT_DMIC_GPIO_PORT, &GPIO_InitStruct);

  /* MP34DT01TR microphones uses DFSDM_DATIN0 input pin */
  AUDIO_DFSDMx_DMIC_GPIO_CLK_ENABLE();
  GPIO_InitStruct.Pin = AUDIO_DFSDMx_DMIC_PIN;
  GPIO_InitStruct.Mode = GPIO_MODE_AF_PP;
  GPIO_InitStruct.Pull = GPIO_NOPULL;
  GPIO_InitStruct.Speed = GPIO_SPEED_FREQ_VERY_HIGH;
  GPIO_InitStruct.Alternate = AUDIO_DFSDMx_DMIC_AF;
  HAL_GPIO_Init(AUDIO_DFSDMx_DMIC_GPIO_PORT, &GPIO_InitStruct);

}

/**
  * @brief  DeInitializes the DFSDM channel MSP.
  */
static void DFSDMx_ChannelMspDeInit(void)
{
  GPIO_InitTypeDef  GPIO_InitStruct;

  /* DFSDM pins configuration: DFSDM_CKOUT, DMIC_DATIN pins ------------------*/
  GPIO_InitStruct.Pin = AUDIO_DFSDMx_CKOUT_PIN;
  HAL_GPIO_DeInit(AUDIO_DFSDMx_CKOUT_DMIC_GPIO_PORT, GPIO_InitStruct.Pin);

  /* MP34DT01TR microphones uses DFSDM_DATIN0 input pin */
  GPIO_InitStruct.Pin = AUDIO_DFSDMx_DMIC_PIN;
  HAL_GPIO_DeInit(AUDIO_DFSDMx_DMIC_GPIO_PORT, GPIO_InitStruct.Pin);

}

/**
  * @brief  Initializes the DFSDM filter MSP.
  */
static void DFSDMx_FilterMspInit(void)
{
  uint32_t i = 0, mic_num = 0, mic_init[DFSDM_MIC_NUMBER] = {0};
  IRQn_Type AUDIO_DFSDM_DMAx_MIC_IRQHandler[DFSDM_MIC_NUMBER] = {AUDIO_DFSDMx_DMAx_MIC1_IRQ, AUDIO_DFSDMx_DMAx_MIC2_IRQ};
  DMA_Stream_TypeDef* AUDIO_DFSDMx_DMAx_MIC_STREAM[DFSDM_MIC_NUMBER] = {AUDIO_DFSDMx_DMAx_MIC1_STREAM, AUDIO_DFSDMx_DMAx_MIC2_STREAM};
  uint32_t AUDIO_DFSDMx_DMAx_MIC_REQUEST[DFSDM_MIC_NUMBER] = {AUDIO_DFSDMx_DMAx_MIC1_REQUEST, AUDIO_DFSDMx_DMAx_MIC2_REQUEST};

  /* Enable the DMA clock */
  AUDIO_DFSDMx_DMAx_CLK_ENABLE();

  for(i = 0; i < hAudioIn.ChannelNbr; i++)
  {
    if(((hAudioIn.InputDevice & INPUT_DEVICE_DIGITAL_MIC1) == INPUT_DEVICE_DIGITAL_MIC1) && (mic_init[POS_VAL(INPUT_DEVICE_DIGITAL_MIC1)] != 1))
    {
      mic_num = POS_VAL(INPUT_DEVICE_DIGITAL_MIC1);
      mic_init[mic_num] = 1;
    }
    else if(((hAudioIn.InputDevice & INPUT_DEVICE_DIGITAL_MIC2) == INPUT_DEVICE_DIGITAL_MIC2) && (mic_init[POS_VAL(INPUT_DEVICE_DIGITAL_MIC2)] != 1))
    {
      mic_num = POS_VAL(INPUT_DEVICE_DIGITAL_MIC2);
      mic_init[mic_num] = 1;
    }

    /* Configure the hDmaDfsdm[i] handle parameters */
    hDmaDfsdm[mic_num].Init.Request             = AUDIO_DFSDMx_DMAx_MIC_REQUEST[mic_num];
    hDmaDfsdm[mic_num].Instance                 = AUDIO_DFSDMx_DMAx_MIC_STREAM[mic_num];
    hDmaDfsdm[mic_num].Init.Direction           = DMA_PERIPH_TO_MEMORY;
    hDmaDfsdm[mic_num].Init.PeriphInc           = DMA_PINC_DISABLE;
    hDmaDfsdm[mic_num].Init.MemInc              = DMA_MINC_ENABLE;
    hDmaDfsdm[mic_num].Init.PeriphDataAlignment = AUDIO_DFSDMx_DMAx_PERIPH_DATA_SIZE;
    hDmaDfsdm[mic_num].Init.MemDataAlignment    = AUDIO_DFSDMx_DMAx_MEM_DATA_SIZE;
    hDmaDfsdm[mic_num].Init.Mode                = DMA_CIRCULAR;
    hDmaDfsdm[mic_num].Init.Priority            = DMA_PRIORITY_HIGH;
    hDmaDfsdm[mic_num].Init.FIFOMode            = DMA_FIFOMODE_DISABLE;
    hDmaDfsdm[mic_num].Init.MemBurst            = DMA_MBURST_SINGLE;
    hDmaDfsdm[mic_num].Init.PeriphBurst         = DMA_PBURST_SINGLE;
    hDmaDfsdm[mic_num].State                    = HAL_DMA_STATE_RESET;

    /* Associate the DMA handle */
    __HAL_LINKDMA(&hAudioInDfsdmFilter[mic_num], hdmaReg, hDmaDfsdm[mic_num]);

    /* Reset DMA handle state */
    __HAL_DMA_RESET_HANDLE_STATE(&hDmaDfsdm[mic_num]);

    /* Configure the DMA Channel */
    HAL_DMA_Init(&hDmaDfsdm[mic_num]);

    /* DMA IRQ Channel configuration */
    HAL_NVIC_SetPriority(AUDIO_DFSDM_DMAx_MIC_IRQHandler[mic_num], AUDIO_IN_IRQ_PREPRIO, 0);
    HAL_NVIC_EnableIRQ(AUDIO_DFSDM_DMAx_MIC_IRQHandler[mic_num]);
  }
}

/**
  * @brief  DeInitializes the DFSDM filter MSP.
  */
static void DFSDMx_FilterMspDeInit(void)
{
  /* Configure the DMA Channel */
  for(uint32_t i = 0; i < DFSDM_MIC_NUMBER; i++)
  {
    if(hDmaDfsdm[i].Instance != NULL)
    {
      HAL_DMA_DeInit(&hDmaDfsdm[i]);
    }
  }
}

/**
  * @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();

    AUDIO_IN_SAI_PDMx_CLK_IN_ENABLE();
    AUDIO_IN_SAI_PDMx_DATA_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);

    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);

    AUDIO_IN_SAI_PDMx_FS_SCK_ENABLE();

    /* CODEC_SAI pins configuration: FS, SCK, MCK and SD pins ------------------*/
    gpio_init_structure.Pin = AUDIO_IN_SAI_PDMx_FS_PIN | AUDIO_IN_SAI_PDMx_SCK_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_FS_SCK_AF;
    HAL_GPIO_Init(AUDIO_IN_SAI_PDMx_FS_SCK_GPIO_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 Audio INT GPIO clock */
    AUDIO_IN_INT_GPIO_ENABLE();
    /* Audio INT pin configuration: input */
    gpio_init_structure.Pin = AUDIO_IN_INT_GPIO_PIN;
    gpio_init_structure.Mode = GPIO_MODE_INPUT;
    gpio_init_structure.Pull = GPIO_NOPULL;
    gpio_init_structure.Speed = GPIO_SPEED_FREQ_HIGH;
    HAL_GPIO_Init(AUDIO_IN_INT_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);

    /* Audio INT IRQ Channel configuration */
    HAL_NVIC_SetPriority(AUDIO_IN_INT_IRQ, AUDIO_IN_IRQ_PREPRIO, 0);
    HAL_NVIC_EnableIRQ(AUDIO_IN_INT_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);

    /* CODEC_SAI pins configuration: FS, SCK, MCK and SD pins ------------------*/
    gpio_init_structure.Pin = AUDIO_IN_SAI_PDMx_FS_PIN | AUDIO_IN_SAI_PDMx_SCK_PIN;
    HAL_GPIO_DeInit(AUDIO_IN_SAI_PDMx_FS_SCK_GPIO_PORT, gpio_init_structure.Pin);

    /* Disable SAI clock */
    AUDIO_IN_SAI_PDMx_CLK_DISABLE();

    /* Set audio_in interface as default one */
    hAudioIn.Interface = AUDIO_IN_INTERFACE_DFSDM;
  }
  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);

    gpio_init_structure.Pin = AUDIO_IN_INT_GPIO_PIN;
    HAL_GPIO_DeInit(AUDIO_IN_INT_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 = 1;
    haudio_in_sai.Init.PdmInit.ClockEnable = SAI_PDM_CLOCK1_ENABLE;
    haudio_in_sai.Init.FirstBit            = SAI_FIRSTBIT_LSB;
    haudio_in_sai.Init.ClockStrobing       = SAI_CLOCKSTROBING_FALLINGEDGE;

    haudio_in_sai.FrameInit.FrameLength       = 16;
    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     = 1;
    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****/