/**
  ******************************************************************************
  * @file    stm32f769i_discovery_audio.c
  * @author  MCD Application Team
  * @brief   This file provides the Audio driver for the STM32F769I-DISCOVERY  
  *          board.
  @verbatim
  How To use this driver:
  -----------------------
   + This driver supports STM32F7xx devices on STM32F769I-DISCOVERY (MB1225) 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 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 STM32F769I-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 stm32f769i_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 (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 STM32F769I-DISCOVERY, four 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 stm32f769i_discovery_audio.h file. (refer to the example for more details on the callbacks implementations)

  Driver architecture:
  --------------------
   + This driver provides the High Audio Layer: consists of the function API exported in the stm32f769i_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 stm32f769i_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>&copy; COPYRIGHT(c) 2016 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.
  *
  ******************************************************************************
  */

/* Dependencies
- stm32f769i_discovery.c
- stm32f7xx_hal_sai.c
- stm32f7xx_hal_dfsdm.c
- stm32f7xx_hal_dma.c
- stm32f7xx_hal_gpio.c
- stm32f7xx_hal_cortex.c
- stm32f7xx_hal_rcc_ex.h
- wm8994.c
EndDependencies */

/* Includes ------------------------------------------------------------------*/
#include "stm32f769i_discovery_audio.h"

/** @addtogroup BSP
  * @{
  */

/** @addtogroup STM32F769I_DISCOVERY
  * @{
  */ 
  
/** @defgroup STM32F769I_DISCOVERY_AUDIO STM32F769I_DISCOVERY AUDIO
  * @brief This file includes the low layer driver for wm8994 Audio Codec
  *        available on STM32F769I-DISCOVERY discoveryuation board(MB1225).
  * @{
  */ 

/** @defgroup STM32F769I_DISCOVERY_AUDIO_Private_Types STM32F769I_DISCOVERY_AUDIO Private Types
  * @{
  */ 
typedef struct
{
  uint16_t      *pRecBuf;       /* Pointer to record user buffer */
  uint32_t      RecSize;        /* Size to record in mono, double size to record in stereo */
}AUDIOIN_TypeDef;

/**
  * @}
  */ 
  
/** @defgroup STM32F769I_DISCOVERY_AUDIO_Private_Defines STM32F769I_DISCOVERY_AUDIO Private Defines
  * @{
  */
/**
  * @}
  */ 

/** @defgroup STM32F769I_DISCOVERY_AUDIO_Private_Macros STM32F769I_DISCOVERY_AUDIO Private Macros
  * @{
  */
/*### RECORD ###*/
#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_RIGHT_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) ? 0  \
      : (__FREQUENCY__ == AUDIO_FREQUENCY_48K) ? 0 : 4  \

/* Saturate the record PCM sample */
#define SaturaLH(N, L, H) (((N)<(L))?(L):(((N)>(H))?(H):(N)))
/**
  * @}
  */ 
  
/** @defgroup STM32F769I_DISCOVERY_AUDIO_Private_Variables STM32F769I_DISCOVERY_AUDIO Private Variables
  * @{
  */
/* PLAY */
AUDIO_DrvTypeDef                *audio_drv;
SAI_HandleTypeDef               haudio_out_sai;
SAI_HandleTypeDef               haudio_in_sai;

/* RECORD */
AUDIOIN_TypeDef                 hAudioIn;

DFSDM_Channel_HandleTypeDef     hAudioInTopLeftChannel;
DFSDM_Channel_HandleTypeDef     hAudioInTopRightChannel;
DFSDM_Filter_HandleTypeDef      hAudioInTopLeftFilter;
DFSDM_Filter_HandleTypeDef      hAudioInTopRightFilter;
DMA_HandleTypeDef               hDmaTopLeft;
DMA_HandleTypeDef               hDmaTopRight;

DFSDM_Channel_HandleTypeDef     hAudioInButtomLeftChannel;
DFSDM_Channel_HandleTypeDef     hAudioInButtomRightChannel;
DFSDM_Filter_HandleTypeDef      hAudioInButtomLeftFilter;
DFSDM_Filter_HandleTypeDef      hAudioInButtomRightFilter;
DMA_HandleTypeDef               hDmaButtomLeft;
DMA_HandleTypeDef               hDmaButtomRight;

/* Buffers for right and left samples */
static int32_t                  *pScratchBuff[2*DEFAULT_AUDIO_IN_CHANNEL_NBR];
static __IO int32_t             ScratchSize;
/* Cannel number to be used: 2 channels by default */
static uint8_t                  AudioIn_ChannelNumber = DEFAULT_AUDIO_IN_CHANNEL_NBR;
/* Input device to be used: digital microphones by default */
static uint16_t                 AudioIn_Device = INPUT_DEVICE_DIGITAL_MIC; 

/* Buffers status flags */
static uint32_t                DmaTopLeftRecHalfCplt  = 0;
static uint32_t                DmaTopLeftRecCplt      = 0;
static uint32_t                DmaTopRightRecHalfCplt = 0;
static uint32_t                DmaTopRightRecCplt     = 0;
static uint32_t                DmaButtomLeftRecHalfCplt  = 0;
static uint32_t                DmaButtomLeftRecCplt      = 0;
static uint32_t                DmaButtomRightRecHalfCplt = 0;
static uint32_t                DmaButtomRightRecCplt     = 0;

/* Application Buffer Trigger */
static __IO uint32_t           AppBuffTrigger          = 0;
static __IO uint32_t           AppBuffHalf             = 0;

/**
  * @}
  */ 

/** @defgroup STM32F769I_DISCOVERY_AUDIO_Private_Function_Prototypes STM32F769I_DISCOVERY_AUDIO Private Function Prototypes
  * @{
  */
static void SAIx_Out_Init(uint32_t AudioFreq);
static void SAIx_Out_DeInit(void);
static void SAI_AUDIO_IN_MspInit(SAI_HandleTypeDef *hsai, void *Params);
static void SAI_AUDIO_IN_MspDeInit(SAI_HandleTypeDef *hsai, void *Params);
static void SAIx_In_Init(uint32_t AudioFreq);
static void SAIx_In_DeInit(void);
static void    DFSDMx_ChannelMspInit(void);
static void    DFSDMx_FilterMspInit(void);
static void    DFSDMx_ChannelMspDeInit(void);
static void    DFSDMx_FilterMspDeInit(void);
static uint8_t DFSDMx_Init(uint32_t AudioFreq);
static uint8_t DFSDMx_DeInit(void);

/**
  * @}
  */ 

/** @defgroup STM32F769I_DISCOVERY_AUDIO_out_Private_Functions STM32F769I_DISCOVERY_AUDIO_Out Private Functions
  * @{
  */ 

/**
  * @brief  Configures the audio peripherals.
  * @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;

  /* Disable SAI */
  SAIx_Out_DeInit();

  /* 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 */
  haudio_out_sai.Instance = AUDIO_OUT_SAIx;
  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);
  }
  SAIx_Out_Init(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, (uint16_t *)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.
  * @note    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.  
  * @note    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();
  /* DeInit the SAI MSP : this __weak function can be rewritten by the application */
  BSP_AUDIO_OUT_MspDeInit(&haudio_out_sai, NULL);
}

/**
  * @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 stm32f769i_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 stm32f769i_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();
  }
}

/**
  * @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  
  * @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();
  
  
  /* Enable GPIO clock */
  AUDIO_OUT_SAIx_MCLK_ENABLE();
  AUDIO_OUT_SAIx_SD_FS_CLK_ENABLE();
  
  /* CODEC_SAI pins configuration: FS, SCK, MCK 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_HIGH;
  gpio_init_structure.Alternate = AUDIO_OUT_SAIx_AF;
  HAL_GPIO_Init(AUDIO_OUT_SAIx_SD_FS_SCK_GPIO_PORT, &gpio_init_structure);
  
  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.Channel             = AUDIO_OUT_SAIx_DMAx_CHANNEL;
    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  Initializes SAI Audio IN MSP.
  * @param  hsai: SAI handle
  * @param  Params  
  * @retval None
  */
static void SAI_AUDIO_IN_MspInit(SAI_HandleTypeDef *hsai, void *Params)
{
  static DMA_HandleTypeDef hdma_sai_rx;
  GPIO_InitTypeDef  gpio_init_structure;  

  /* 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_FAST;
  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_FAST;
  HAL_GPIO_Init(AUDIO_IN_INT_GPIO_PORT, &gpio_init_structure);

  /* Enable the DMA clock */
  AUDIO_IN_SAIx_DMAx_CLK_ENABLE();
    
  if(hsai->Instance == AUDIO_IN_SAIx)
  {
    /* Configure the hdma_sai_rx handle parameters */
    hdma_sai_rx.Init.Channel             = AUDIO_IN_SAIx_DMAx_CHANNEL;
    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  
  * @retval None
  */
static void SAI_AUDIO_IN_MspDeInit(SAI_HandleTypeDef *hsai, void *Params)
{
  GPIO_InitTypeDef  gpio_init_structure;
  
  /* 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  Deinitializes SAI MSP.
  * @param  hsai: SAI handle
  * @param  Params  
  * @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  
  * @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))
  {
    /* Configure PLLSAI prescalers */
    /* PLLSAI_VCO: VCO_429M 
    SAI_CLK(first level) = PLLSAI_VCO/PLLSAIQ = 429/2 = 214.5 Mhz
    SAI_CLK_x = SAI_CLK(first level)/PLLSAIDIVQ = 214.5/19 = 11.289 Mhz */ 
    rcc_ex_clk_init_struct.PeriphClockSelection = RCC_PERIPHCLK_SAI1;
    rcc_ex_clk_init_struct.Sai1ClockSelection = RCC_SAI1CLKSOURCE_PLLI2S;
    rcc_ex_clk_init_struct.PLLI2S.PLLI2SN = 429;
    rcc_ex_clk_init_struct.PLLI2S.PLLI2SQ = 2;
    rcc_ex_clk_init_struct.PLLI2SDivQ = 19;
    
    HAL_RCCEx_PeriphCLKConfig(&rcc_ex_clk_init_struct);
    
  }
  else /* AUDIO_FREQUENCY_8K, AUDIO_FREQUENCY_16K, AUDIO_FREQUENCY_48K, AUDIO_FREQUENCY_96K */
  {
    /* SAI clock config 
    PLLSAI_VCO: VCO_344M 
    SAI_CLK(first level) = PLLSAI_VCO/PLLSAIQ = 344/7 = 49.142 Mhz 
    SAI_CLK_x = SAI_CLK(first level)/PLLSAIDIVQ = 49.142/1 = 49.142 Mhz */  
    rcc_ex_clk_init_struct.PeriphClockSelection = RCC_PERIPHCLK_SAI1;
    rcc_ex_clk_init_struct.Sai1ClockSelection = RCC_SAI1CLKSOURCE_PLLI2S;
    rcc_ex_clk_init_struct.PLLI2S.PLLI2SN = 344; 
    rcc_ex_clk_init_struct.PLLI2S.PLLI2SQ = 7; 
    rcc_ex_clk_init_struct.PLLI2SDivQ = 1;      
    
    HAL_RCCEx_PeriphCLKConfig(&rcc_ex_clk_init_struct);
  }
}

/*******************************************************************************
                            Static Functions
*******************************************************************************/

/**
  * @brief  Initializes the Audio Codec audio interface (SAI).
  * @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 AudioFreq)
{
  /* Initialize the haudio_out_sai Instance parameter */
  haudio_out_sai.Instance = AUDIO_OUT_SAIx;
  
  /* 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 = SAI_MODEMASTER_TX;
  haudio_out_sai.Init.NoDivider = SAI_MASTERDIVIDER_ENABLED;
  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_ENABLED;
  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;
    
  /* 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 = CODEC_AUDIOFRAME_SLOT_0123;

  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(void)
{
  /* Initialize the haudio_out_sai Instance parameter */
  haudio_out_sai.Instance = AUDIO_OUT_SAIx;

  /* Disable SAI peripheral */
  __HAL_SAI_DISABLE(&haudio_out_sai);

  HAL_SAI_DeInit(&haudio_out_sai);
}

/**
  * @brief  Initializes the Audio Codec audio interface (SAI).
  * @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_In_Init(uint32_t AudioFreq)
{
  /* Initialize SAI1 block A in MASTER TX */
  /* Initialize the haudio_out_sai Instance parameter */
  haudio_out_sai.Instance = AUDIO_OUT_SAIx;

  /* 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      = SAI_MODEMASTER_RX;
  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_FALLINGEDGE;
  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;  

  /* 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       = 64;
  haudio_out_sai.FrameInit.ActiveFrameLength = 32;
  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     = CODEC_AUDIOFRAME_SLOT_02;

  HAL_SAI_Init(&haudio_out_sai);

  /* Initialize SAI1 block B in SLAVE RX synchronous from SAI1 block A */
  /* Initialize the haudio_in_sai Instance parameter */
  haudio_in_sai.Instance = AUDIO_IN_SAIx;
  
  /* 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      = SAI_MODESLAVE_RX;
  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;

  /* 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       = 64;
  haudio_in_sai.FrameInit.ActiveFrameLength = 32;
  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     = CODEC_AUDIOFRAME_SLOT_02;

  HAL_SAI_Init(&haudio_in_sai);

  /* Enable SAI peripheral */
  __HAL_SAI_ENABLE(&haudio_in_sai);

  /* Enable SAI peripheral to generate MCLK */
  __HAL_SAI_ENABLE(&haudio_out_sai);
}

/**
  * @brief  Deinitializes the output Audio Codec audio interface (SAI).
  * @retval None
  */
static void SAIx_In_DeInit(void)
{
  /* Initialize the haudio_in_sai Instance parameter */
  haudio_in_sai.Instance = AUDIO_IN_SAIx;
  haudio_out_sai.Instance = AUDIO_OUT_SAIx;
  /* Disable SAI peripheral */
  __HAL_SAI_DISABLE(&haudio_in_sai);

  HAL_SAI_DeInit(&haudio_in_sai);
  HAL_SAI_DeInit(&haudio_out_sai);
}

/**
  * @}
  */

/** @defgroup STM32F769I_DISCOVERY_AUDIO_In_Private_Functions STM32F769I_DISCOVERY_AUDIO_In Private 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)
{ 
  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_ERROR;
  AudioIn_Device = InputDevice;
  
  if(InputDevice == INPUT_DEVICE_DIGITAL_MIC)
  {
    AudioIn_ChannelNumber = ChnlNbr;
    /* PLL clock is set depending by the AudioFreq (44.1khz vs 48khz groups) */ 
    BSP_AUDIO_IN_ClockConfig(&hAudioInTopLeftFilter, AudioFreq, NULL);
    
    /* Init the SAI MSP: this __weak function can be redefined by the application*/
    BSP_AUDIO_IN_MspInit();
    
    /* Initializes DFSDM peripheral */
    DFSDMx_Init(AudioFreq);
    ret = AUDIO_OK;
  }
  else
  {
    /* Disable SAI */
    SAIx_In_DeInit();
    
    /* PLL clock is set depending by the AudioFreq (44.1khz vs 48khz groups) */ 
    BSP_AUDIO_OUT_ClockConfig(&haudio_in_sai, AudioFreq, NULL);
    
    haudio_in_sai.Instance = AUDIO_IN_SAIx;
    if(HAL_SAI_GetState(&haudio_in_sai) == HAL_SAI_STATE_RESET)
    {    
    BSP_AUDIO_OUT_MspInit(&haudio_in_sai, NULL);
    BSP_AUDIO_IN_MspInit();
    }

    SAIx_In_Init(AudioFreq);
    
    if((wm8994_drv.ReadID(AUDIO_I2C_ADDRESS)) == 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, 100, AudioFreq);
    }    
  }
  
  /* Return AUDIO_OK when all operations are correctly done */
  return ret;
}

/**
  * @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 / AudioIn_ChannelNumber);
  
  /* copy scratch pointers */
  for (idx = 0; idx < AudioIn_ChannelNumber; 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 AudioIn_ChannelNumber;
}

/**
  * @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 (AudioIn_Device == INPUT_DEVICE_DIGITAL_MIC)
  {
    hAudioIn.pRecBuf = pbuf;
    hAudioIn.RecSize = size;
    /* Reset Application Buffer Trigger */
    AppBuffTrigger = 0;
    AppBuffHalf = 0;
    
    if(AudioIn_ChannelNumber > 2)
    {
      /* Call the Media layer start function for buttom right channel */
      if(HAL_OK != HAL_DFSDM_FilterRegularStart_DMA(&hAudioInButtomRightFilter, pScratchBuff[2], ScratchSize))
      {
        return AUDIO_ERROR;
      } 
      
      /* Call the Media layer start function for buttom left channel */
      if(HAL_OK != HAL_DFSDM_FilterRegularStart_DMA(&hAudioInButtomLeftFilter, pScratchBuff[3], ScratchSize))
      {
        return AUDIO_ERROR;
      }
    }

    /* Call the Media layer start function for top right channel */
    if(HAL_OK != HAL_DFSDM_FilterRegularStart_DMA(&hAudioInTopRightFilter, pScratchBuff[0], ScratchSize))
    {
      return AUDIO_ERROR;
    }
    
    /* Call the Media layer start function for top left channel */
    if(HAL_OK != HAL_DFSDM_FilterRegularStart_DMA(&hAudioInTopLeftFilter, pScratchBuff[1], 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 (AudioIn_Device == INPUT_DEVICE_DIGITAL_MIC)
  { 
    AppBuffTrigger = 0;
    AppBuffHalf    = 0;
    
    if(AudioIn_ChannelNumber > 2)
    {    
      /* Call the Media layer stop function for buttom right channel */
      if(HAL_OK != HAL_DFSDM_FilterRegularStop_DMA(&hAudioInButtomRightFilter))
      {
        return AUDIO_ERROR;
      }
      
      /* Call the Media layer stop function for buttom left channel */
      if(HAL_OK != HAL_DFSDM_FilterRegularStop_DMA(&hAudioInButtomLeftFilter))
      {
        return AUDIO_ERROR;
      }
    }
    
    /* Call the Media layer stop function for top right channel */
    if(HAL_OK != HAL_DFSDM_FilterRegularStop_DMA(&hAudioInTopRightFilter))
    {
      return AUDIO_ERROR;
    }
    
    /* Call the Media layer stop function for top left channel */
    if(HAL_OK != HAL_DFSDM_FilterRegularStop_DMA(&hAudioInTopLeftFilter))
    {
      return AUDIO_ERROR;
    }
  }
  else
  {
    /* Call the Media layer stop function */
    HAL_SAI_DMAStop(&haudio_in_sai);
    
    /* Call Audio Codec Stop function */
    if(audio_drv->Stop(AUDIO_I2C_ADDRESS, CODEC_PDWN_HW) != 0)
    {
      return AUDIO_ERROR;
    }
    else
    {
      /* Wait at least 100us */
      HAL_Delay(1);
      
      /* Return AUDIO_OK when all operations are correctly done */
      return AUDIO_OK;
    }  
  }
  /* 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(AudioIn_ChannelNumber > 2)
  { 
    /* Call the Media layer stop function */
    if(HAL_OK != HAL_DFSDM_FilterRegularStop_DMA(&hAudioInButtomRightFilter))
    {
      return AUDIO_ERROR;
    }
    
    /* Call the Media layer stop function */
    if(HAL_OK != HAL_DFSDM_FilterRegularStop_DMA(&hAudioInButtomLeftFilter))
    {
      return AUDIO_ERROR;
    }
  }  
  /* Call the Media layer stop function */
  if(HAL_OK != HAL_DFSDM_FilterRegularStop_DMA(&hAudioInTopRightFilter))
  {
    return AUDIO_ERROR;
  }
  
  /* Call the Media layer stop function */
  if(HAL_OK != HAL_DFSDM_FilterRegularStop_DMA(&hAudioInTopLeftFilter))
  {
    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(AudioIn_ChannelNumber > 2)
  {   
    /* Call the Media layer start function for buttom right channel */
    if(HAL_OK != HAL_DFSDM_FilterRegularStart_DMA(&hAudioInButtomRightFilter, pScratchBuff[2], ScratchSize))
    {
      return AUDIO_ERROR;
    }
    
    /* Call the Media layer start function for buttom left channel */
    if(HAL_OK != HAL_DFSDM_FilterRegularStart_DMA(&hAudioInButtomLeftFilter, pScratchBuff[3], ScratchSize))
    {
      return AUDIO_ERROR;
    }
  } 
  /* Call the Media layer start function for top right channel */
  if(HAL_OK != HAL_DFSDM_FilterRegularStart_DMA(&hAudioInTopRightFilter, pScratchBuff[0], ScratchSize))
  {
    return AUDIO_ERROR;
  }
  
  /* Call the Media layer start function for top left channel */
  if(HAL_OK != HAL_DFSDM_FilterRegularStart_DMA(&hAudioInTopLeftFilter, pScratchBuff[1], ScratchSize))
  {
    return AUDIO_ERROR;
  }
  
  /* 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)
{
  BSP_AUDIO_IN_MspDeInit();
  
  if(AudioIn_Device == INPUT_DEVICE_DIGITAL_MIC)
  {
    DFSDMx_DeInit();
  }
  else
  {
    SAIx_In_DeInit();
  }
}

/**
  * @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 = 0;
  
  if(hdfsdm_filter == &hAudioInTopLeftFilter)
  {
    DmaTopLeftRecCplt = 1;
  }
  else if(hdfsdm_filter == &hAudioInTopRightFilter)
  {
    DmaTopRightRecCplt = 1;
  }
  else if(hdfsdm_filter == &hAudioInButtomLeftFilter)
  {
    DmaButtomLeftRecCplt = 1;
  }
  else
  {
    DmaButtomRightRecCplt = 1;
  }
  
  if(AudioIn_ChannelNumber > 2)
  {
    if((DmaTopLeftRecCplt == 1) && (DmaTopRightRecCplt == 1) && (DmaButtomLeftRecCplt == 1) && (DmaButtomRightRecCplt == 1))
    {
      for(index = (ScratchSize/2) ; index < ScratchSize; index++)
      {
        hAudioIn.pRecBuf[AppBuffTrigger]     = (uint16_t)(SaturaLH((pScratchBuff[1][index] >> 8), -32760, 32760));
        hAudioIn.pRecBuf[AppBuffTrigger + 1] = (uint16_t)(SaturaLH((pScratchBuff[0][index] >> 8), -32760, 32760));       
        hAudioIn.pRecBuf[AppBuffTrigger + 2] = (uint16_t)(SaturaLH((pScratchBuff[3][index] >> 8), -32760, 32760));
        hAudioIn.pRecBuf[AppBuffTrigger + 3] = (uint16_t)(SaturaLH((pScratchBuff[2][index] >> 8), -32760, 32760));      
        AppBuffTrigger +=4;
      }
      DmaTopLeftRecCplt  = 0;
      DmaTopRightRecCplt = 0;
      DmaButtomLeftRecCplt  = 0;
      DmaButtomRightRecCplt = 0;     
    }
  }
  else
  {
    if((DmaTopLeftRecCplt == 1) && (DmaTopRightRecCplt == 1))
    {    
      for(index = (ScratchSize/2) ; index < ScratchSize; index++)
      {
        hAudioIn.pRecBuf[AppBuffTrigger]     = (uint16_t)(SaturaLH((pScratchBuff[1][index] >> 8), -32760, 32760));
        hAudioIn.pRecBuf[AppBuffTrigger + 1] = (uint16_t)(SaturaLH((pScratchBuff[0][index] >> 8), -32760, 32760));
        AppBuffTrigger +=2;
      }
      DmaTopLeftRecCplt  = 0;
      DmaTopRightRecCplt = 0;  
    }
  }
  
  /* Call Half Transfer Complete callback */
  if((AppBuffTrigger == hAudioIn.RecSize/2) && (AppBuffHalf == 0))
  {
    AppBuffHalf = 1;  
    BSP_AUDIO_IN_HalfTransfer_CallBack();
  }
  /* Call Transfer Complete callback */
  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();
  }  
}

/**
  * @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 = 0;
  
  if(hdfsdm_filter == &hAudioInTopLeftFilter)
  {
    DmaTopLeftRecHalfCplt = 1;
  }
  else if(hdfsdm_filter == &hAudioInTopRightFilter)
  {
    DmaTopRightRecHalfCplt = 1;
  }
  else if(hdfsdm_filter == &hAudioInButtomLeftFilter)
  {
    DmaButtomLeftRecHalfCplt = 1;
  }
  else
  {
    DmaButtomRightRecHalfCplt = 1;
  }
  
  if(AudioIn_ChannelNumber > 2)
  {
    if((DmaTopLeftRecHalfCplt == 1) && (DmaTopRightRecHalfCplt == 1) && (DmaButtomLeftRecHalfCplt == 1) && (DmaButtomRightRecHalfCplt == 1))
    {
      for(index = 0 ; index < ScratchSize/2; index++)
      {
        hAudioIn.pRecBuf[AppBuffTrigger]     = (uint16_t)(SaturaLH((pScratchBuff[1][index] >> 8), -32760, 32760));
        hAudioIn.pRecBuf[AppBuffTrigger + 1] = (uint16_t)(SaturaLH((pScratchBuff[0][index] >> 8), -32760, 32760)); 
        hAudioIn.pRecBuf[AppBuffTrigger + 2] = (uint16_t)(SaturaLH((pScratchBuff[3][index] >> 8), -32760, 32760));
        hAudioIn.pRecBuf[AppBuffTrigger + 3] = (uint16_t)(SaturaLH((pScratchBuff[2][index] >> 8), -32760, 32760));      
        AppBuffTrigger +=4;
      }
      DmaTopLeftRecHalfCplt  = 0;
      DmaTopRightRecHalfCplt = 0;
      DmaButtomLeftRecHalfCplt  = 0;
      DmaButtomRightRecHalfCplt = 0;     
    }
  }
  else
  {
    if((DmaTopLeftRecHalfCplt == 1) && (DmaTopRightRecHalfCplt == 1))
    {    
      for(index = 0 ; index < ScratchSize/2; index++)
      {
        hAudioIn.pRecBuf[AppBuffTrigger]     = (uint16_t)(SaturaLH((pScratchBuff[1][index] >> 8), -32760, 32760));
        hAudioIn.pRecBuf[AppBuffTrigger + 1] = (uint16_t)(SaturaLH((pScratchBuff[0][index] >> 8), -32760, 32760));
        AppBuffTrigger +=2;
      }
      DmaTopLeftRecHalfCplt  = 0;
      DmaTopRightRecHalfCplt = 0;  
    }
  }
  
  /* Call Half Transfer Complete callback */
  if((AppBuffTrigger == hAudioIn.RecSize/2) && (AppBuffHalf == 0))
  { 
    AppBuffHalf = 1;  
    BSP_AUDIO_IN_HalfTransfer_CallBack();
  }
  /* Call Transfer Complete callback */
  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();
  }  
}

/**
  * @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 (its prototype is already declared in stm32769i_discovery_audio.h) */
  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();
}

/**
  * @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  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 (AudioIn_Device == INPUT_DEVICE_DIGITAL_MIC)
  {  
  /* MSP channels initialization */
  DFSDMx_ChannelMspInit();  
  /* MSP filters initialization */
  DFSDMx_FilterMspInit();
  }
  else
  {
   SAI_AUDIO_IN_MspInit(&haudio_in_sai, NULL); 
  }
}

/**
  * @brief  DeInitialize BSP_AUDIO_IN MSP.
  * @retval None
  */
__weak void BSP_AUDIO_IN_MspDeInit(void)
{
  if (AudioIn_Device == INPUT_DEVICE_DIGITAL_MIC)
  {  
    /* MSP channels initialization */
    DFSDMx_ChannelMspDeInit();  
    /* MSP filters initialization */
    DFSDMx_FilterMspDeInit();
  }
  else
  {
    SAI_AUDIO_IN_MspDeInit(&haudio_in_sai, NULL); 
  }
}

/**
  * @brief  Clock Config.
  * @param  hdfsdm_filter: might be required to set audio peripheral predivider if any.
  * @param  AudioFreq: Audio frequency used to play the audio stream.
  * @param  Params  
  * @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(DFSDM_Filter_HandleTypeDef *hdfsdm_filter, 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))
  {
    /* Configure PLLSAI prescalers */
    /* PLLI2S_VCO: VCO_429M 
    SAI_CLK(first level) = PLLI2S_VCO/PLLI2SQ = 429/2 = 214.5 Mhz
    SAI_CLK_x = SAI_CLK(first level)/PLLI2SDIVQ = 214.5/19 = 11.289 Mhz */ 
    rcc_ex_clk_init_struct.PeriphClockSelection = RCC_PERIPHCLK_SAI2;
    rcc_ex_clk_init_struct.Sai2ClockSelection = RCC_SAI2CLKSOURCE_PLLI2S;
    rcc_ex_clk_init_struct.PLLI2S.PLLI2SN = 429;
    rcc_ex_clk_init_struct.PLLI2S.PLLI2SQ = 2;
    rcc_ex_clk_init_struct.PLLI2SDivQ = 19;
    
    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 
    PLLI2S_VCO: VCO_344M 
    SAI_CLK(first level) = PLLI2S_VCO/PLLI2SQ = 344/7 = 49.142 Mhz 
    SAI_CLK_x = SAI_CLK(first level)/PLLI2SDIVQ = 49.142/1 = 49.142 Mhz */  
    rcc_ex_clk_init_struct.PeriphClockSelection = RCC_PERIPHCLK_SAI2;
    rcc_ex_clk_init_struct.Sai2ClockSelection = RCC_SAI2CLKSOURCE_PLLI2S;
    rcc_ex_clk_init_struct.PLLI2S.PLLI2SN = 344; 
    rcc_ex_clk_init_struct.PLLI2S.PLLI2SQ = 7; 
    rcc_ex_clk_init_struct.PLLI2SDivQ = 1;   
    HAL_RCCEx_PeriphCLKConfig(&rcc_ex_clk_init_struct);
  }
  
  rcc_ex_clk_init_struct.PeriphClockSelection = RCC_PERIPHCLK_DFSDM1_AUDIO;
  rcc_ex_clk_init_struct.Dfsdm1AudioClockSelection = RCC_DFSDM1AUDIOCLKSOURCE_SAI2;
  HAL_RCCEx_PeriphCLKConfig(&rcc_ex_clk_init_struct); 
}

/*******************************************************************************
                            Static Functions
*******************************************************************************/
/**
  * @brief  Initialize the Digital Filter for Sigma-Delta Modulators interface (DFSDM).
  * @param  AudioFreq: Audio frequency to be used to set correctly the DFSDM peripheral.
  * @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
  */
static uint8_t DFSDMx_Init(uint32_t AudioFreq)
{
  /****************************************************************************/ 
  /********************** Channels configuration  *****************************/
  /****************************************************************************/ 
  /* CHANNEL 1 configuration */
  __HAL_DFSDM_CHANNEL_RESET_HANDLE_STATE(&hAudioInTopLeftChannel);  
  hAudioInTopLeftChannel.Instance                      = DFSDM1_Channel1;  
  hAudioInTopLeftChannel.Init.OutputClock.Activation   = ENABLE;
  hAudioInTopLeftChannel.Init.OutputClock.Selection    = DFSDM_CHANNEL_OUTPUT_CLOCK_AUDIO;
  /* Set the DFSDM clock OUT audio frequency configuration */
  hAudioInTopLeftChannel.Init.OutputClock.Divider      = DFSDM_CLOCK_DIVIDER(AudioFreq);
  hAudioInTopLeftChannel.Init.Input.Multiplexer        = DFSDM_CHANNEL_EXTERNAL_INPUTS;
  hAudioInTopLeftChannel.Init.Input.DataPacking        = DFSDM_CHANNEL_STANDARD_MODE;
  hAudioInTopLeftChannel.Init.Input.Pins               = DFSDM_CHANNEL_SAME_CHANNEL_PINS;
  /* Request to sample stable data for LEFT micro on Rising edge */
  hAudioInTopLeftChannel.Init.SerialInterface.Type     = DFSDM_CHANNEL_SPI_RISING;
  hAudioInTopLeftChannel.Init.SerialInterface.SpiClock = DFSDM_CHANNEL_SPI_CLOCK_INTERNAL;
  hAudioInTopLeftChannel.Init.Awd.FilterOrder          = DFSDM_CHANNEL_FASTSINC_ORDER;
  hAudioInTopLeftChannel.Init.Awd.Oversampling         = 10;
  hAudioInTopLeftChannel.Init.Offset                   = 0;
  hAudioInTopLeftChannel.Init.RightBitShift            = DFSDM_RIGHT_BIT_SHIFT(AudioFreq);
  if(HAL_OK != HAL_DFSDM_ChannelInit(&hAudioInTopLeftChannel))
  {
    return AUDIO_ERROR;
  }
  
  /* CHANNEL 0 configuration */
  __HAL_DFSDM_CHANNEL_RESET_HANDLE_STATE(&hAudioInTopRightChannel);  
  hAudioInTopRightChannel.Instance                      = DFSDM1_Channel0;  
  hAudioInTopRightChannel.Init.OutputClock.Activation   = ENABLE;
  hAudioInTopRightChannel.Init.OutputClock.Selection    = DFSDM_CHANNEL_OUTPUT_CLOCK_AUDIO;
  /* Set the DFSDM clock OUT audio frequency configuration */
  hAudioInTopRightChannel.Init.OutputClock.Divider      = DFSDM_CLOCK_DIVIDER(AudioFreq);
  hAudioInTopRightChannel.Init.Input.Multiplexer        = DFSDM_CHANNEL_EXTERNAL_INPUTS;
  hAudioInTopRightChannel.Init.Input.DataPacking        = DFSDM_CHANNEL_STANDARD_MODE;
  hAudioInTopRightChannel.Init.Input.Pins               = DFSDM_CHANNEL_FOLLOWING_CHANNEL_PINS;
  /* Request to sample stable data for RIGHT micro on Falling edge */
  hAudioInTopRightChannel.Init.SerialInterface.Type     = DFSDM_CHANNEL_SPI_FALLING;
  hAudioInTopRightChannel.Init.SerialInterface.SpiClock = DFSDM_CHANNEL_SPI_CLOCK_INTERNAL;
  hAudioInTopRightChannel.Init.Awd.FilterOrder          = DFSDM_CHANNEL_FASTSINC_ORDER;
  hAudioInTopRightChannel.Init.Awd.Oversampling         = 10;
  hAudioInTopRightChannel.Init.Offset                   = 0;
  hAudioInTopRightChannel.Init.RightBitShift            = DFSDM_RIGHT_BIT_SHIFT(AudioFreq);
  if(HAL_OK != HAL_DFSDM_ChannelInit(&hAudioInTopRightChannel))
  {
    return AUDIO_ERROR;
  }
  
  if(AudioIn_ChannelNumber > 2)
  {  
    /* CHANNEL 5 configuration */
    __HAL_DFSDM_CHANNEL_RESET_HANDLE_STATE(&hAudioInButtomLeftChannel);  
    hAudioInButtomLeftChannel.Instance                      = DFSDM1_Channel5;  
    hAudioInButtomLeftChannel.Init.OutputClock.Activation   = ENABLE;
    hAudioInButtomLeftChannel.Init.OutputClock.Selection    = DFSDM_CHANNEL_OUTPUT_CLOCK_AUDIO;
    /* Set the DFSDM clock OUT audio frequency configuration */
    hAudioInButtomLeftChannel.Init.OutputClock.Divider      = DFSDM_CLOCK_DIVIDER(AudioFreq);
    hAudioInButtomLeftChannel.Init.Input.Multiplexer        = DFSDM_CHANNEL_EXTERNAL_INPUTS;
    hAudioInButtomLeftChannel.Init.Input.DataPacking        = DFSDM_CHANNEL_STANDARD_MODE;
    hAudioInButtomLeftChannel.Init.Input.Pins               = DFSDM_CHANNEL_SAME_CHANNEL_PINS;
    /* Request to sample stable data for LEFT micro on Rising edge */
    hAudioInButtomLeftChannel.Init.SerialInterface.Type     = DFSDM_CHANNEL_SPI_RISING;
    hAudioInButtomLeftChannel.Init.SerialInterface.SpiClock = DFSDM_CHANNEL_SPI_CLOCK_INTERNAL;
    hAudioInButtomLeftChannel.Init.Awd.FilterOrder          = DFSDM_CHANNEL_FASTSINC_ORDER;
    hAudioInButtomLeftChannel.Init.Awd.Oversampling         = 10;
    hAudioInButtomLeftChannel.Init.Offset                   = 0;
    hAudioInButtomLeftChannel.Init.RightBitShift            = DFSDM_RIGHT_BIT_SHIFT(AudioFreq);
    if(HAL_OK != HAL_DFSDM_ChannelInit(&hAudioInButtomLeftChannel))
    {
      return AUDIO_ERROR;
    }
    
    /* CHANNEL 4 configuration */
    __HAL_DFSDM_CHANNEL_RESET_HANDLE_STATE(&hAudioInButtomRightChannel);  
    hAudioInButtomRightChannel.Instance                      = DFSDM1_Channel4;  
    hAudioInButtomRightChannel.Init.OutputClock.Activation   = ENABLE;
    hAudioInButtomRightChannel.Init.OutputClock.Selection    = DFSDM_CHANNEL_OUTPUT_CLOCK_AUDIO;
    /* Set the DFSDM clock OUT audio frequency configuration */
    hAudioInButtomRightChannel.Init.OutputClock.Divider      = DFSDM_CLOCK_DIVIDER(AudioFreq);
    hAudioInButtomRightChannel.Init.Input.Multiplexer        = DFSDM_CHANNEL_EXTERNAL_INPUTS;
    hAudioInButtomRightChannel.Init.Input.DataPacking        = DFSDM_CHANNEL_STANDARD_MODE;
    hAudioInButtomRightChannel.Init.Input.Pins               = DFSDM_CHANNEL_FOLLOWING_CHANNEL_PINS;
    /* Request to sample stable data for RIGHT micro on Falling edge */
    hAudioInButtomRightChannel.Init.SerialInterface.Type     = DFSDM_CHANNEL_SPI_FALLING;
    hAudioInButtomRightChannel.Init.SerialInterface.SpiClock = DFSDM_CHANNEL_SPI_CLOCK_INTERNAL;
    hAudioInButtomRightChannel.Init.Awd.FilterOrder          = DFSDM_CHANNEL_FASTSINC_ORDER;
    hAudioInButtomRightChannel.Init.Awd.Oversampling         = 10;
    hAudioInButtomRightChannel.Init.Offset                   = 0;
    hAudioInButtomRightChannel.Init.RightBitShift            = DFSDM_RIGHT_BIT_SHIFT(AudioFreq);
    if(HAL_OK != HAL_DFSDM_ChannelInit(&hAudioInButtomRightChannel))
    {
      return AUDIO_ERROR;
    }
  }
  /****************************************************************************/ 
  /********************** Filters configuration  ******************************/
  /****************************************************************************/
  
  /* FILTER 0 configuration */
  __HAL_DFSDM_FILTER_RESET_HANDLE_STATE(&hAudioInTopLeftFilter);
  hAudioInTopLeftFilter.Instance                          = AUDIO_DFSDMx_TOP_LEFT_FILTER;  
  hAudioInTopLeftFilter.Init.RegularParam.Trigger         = DFSDM_FILTER_SW_TRIGGER;
  hAudioInTopLeftFilter.Init.RegularParam.FastMode        = ENABLE;
  hAudioInTopLeftFilter.Init.RegularParam.DmaMode         = ENABLE;
  hAudioInTopLeftFilter.Init.InjectedParam.Trigger        = DFSDM_FILTER_SW_TRIGGER;
  hAudioInTopLeftFilter.Init.InjectedParam.ScanMode       = ENABLE;
  hAudioInTopLeftFilter.Init.InjectedParam.DmaMode        = DISABLE;
  hAudioInTopLeftFilter.Init.InjectedParam.ExtTrigger     = DFSDM_FILTER_EXT_TRIG_TIM1_TRGO;
  hAudioInTopLeftFilter.Init.InjectedParam.ExtTriggerEdge = DFSDM_FILTER_EXT_TRIG_RISING_EDGE;
  hAudioInTopLeftFilter.Init.FilterParam.SincOrder        = DFSDM_FILTER_ORDER(AudioFreq);
  /* Set the DFSDM Filters Oversampling to have correct sample rate */
  hAudioInTopLeftFilter.Init.FilterParam.Oversampling     = DFSDM_OVER_SAMPLING(AudioFreq);
  hAudioInTopLeftFilter.Init.FilterParam.IntOversampling  = 1;
  if(HAL_OK != HAL_DFSDM_FilterInit(&hAudioInTopLeftFilter))
  {
    return AUDIO_ERROR;
  }
  
  /* Configure injected channel */
  if(HAL_OK != HAL_DFSDM_FilterConfigRegChannel(&hAudioInTopLeftFilter, AUDIO_DFSDMx_TOP_LEFT_CHANNEL, DFSDM_CONTINUOUS_CONV_ON))
  {
    return AUDIO_ERROR;
  }
  
  /* FILTER 1 configuration */
  __HAL_DFSDM_FILTER_RESET_HANDLE_STATE(&hAudioInTopRightFilter);
  hAudioInTopRightFilter.Instance                          = AUDIO_DFSDMx_TOP_RIGHT_FILTER;
  hAudioInTopRightFilter.Init.RegularParam.Trigger         = DFSDM_FILTER_SYNC_TRIGGER;
  hAudioInTopRightFilter.Init.RegularParam.FastMode        = ENABLE;
  hAudioInTopRightFilter.Init.RegularParam.DmaMode         = ENABLE;
  hAudioInTopRightFilter.Init.InjectedParam.Trigger        = DFSDM_FILTER_SW_TRIGGER;
  hAudioInTopRightFilter.Init.InjectedParam.ScanMode       = DISABLE;
  hAudioInTopRightFilter.Init.InjectedParam.DmaMode        = DISABLE;
  hAudioInTopRightFilter.Init.InjectedParam.ExtTrigger     = DFSDM_FILTER_EXT_TRIG_TIM1_TRGO;
  hAudioInTopRightFilter.Init.InjectedParam.ExtTriggerEdge = DFSDM_FILTER_EXT_TRIG_RISING_EDGE;
  hAudioInTopRightFilter.Init.FilterParam.SincOrder        = DFSDM_FILTER_ORDER(AudioFreq);
  /* Set the DFSDM Filters Oversampling to have correct sample rate */
  hAudioInTopRightFilter.Init.FilterParam.Oversampling     = DFSDM_OVER_SAMPLING(AudioFreq);
  hAudioInTopRightFilter.Init.FilterParam.IntOversampling  = 1;
  if(HAL_OK != HAL_DFSDM_FilterInit(&hAudioInTopRightFilter))
  {
    return AUDIO_ERROR;
  }
  /* Configure injected channel */
  if(HAL_OK != HAL_DFSDM_FilterConfigRegChannel(&hAudioInTopRightFilter, AUDIO_DFSDMx_TOP_RIGHT_CHANNEL, DFSDM_CONTINUOUS_CONV_ON))
  {
    return AUDIO_ERROR;
  } 
  
  if(AudioIn_ChannelNumber > 2)
  {
    /* FILTER 2 configuration */
    __HAL_DFSDM_FILTER_RESET_HANDLE_STATE(&hAudioInButtomLeftFilter);
    hAudioInButtomLeftFilter.Instance                          = AUDIO_DFSDMx_BUTTOM_LEFT_FILTER;  
    hAudioInButtomLeftFilter.Init.RegularParam.Trigger         = DFSDM_FILTER_SYNC_TRIGGER;
    hAudioInButtomLeftFilter.Init.RegularParam.FastMode        = ENABLE;
    hAudioInButtomLeftFilter.Init.RegularParam.DmaMode         = ENABLE;
    hAudioInButtomLeftFilter.Init.InjectedParam.Trigger        = DFSDM_FILTER_SW_TRIGGER;
    hAudioInButtomLeftFilter.Init.InjectedParam.ScanMode       = ENABLE;
    hAudioInButtomLeftFilter.Init.InjectedParam.DmaMode        = DISABLE;
    hAudioInButtomLeftFilter.Init.InjectedParam.ExtTrigger     = DFSDM_FILTER_EXT_TRIG_TIM1_TRGO;
    hAudioInButtomLeftFilter.Init.InjectedParam.ExtTriggerEdge = DFSDM_FILTER_EXT_TRIG_RISING_EDGE;
    hAudioInButtomLeftFilter.Init.FilterParam.SincOrder        = DFSDM_FILTER_ORDER(AudioFreq);
    /* Set the DFSDM Filters Oversampling to have correct sample rate */
    hAudioInButtomLeftFilter.Init.FilterParam.Oversampling     = DFSDM_OVER_SAMPLING(AudioFreq);
    hAudioInButtomLeftFilter.Init.FilterParam.IntOversampling  = 1;
    if(HAL_OK != HAL_DFSDM_FilterInit(&hAudioInButtomLeftFilter))
    {
      return AUDIO_ERROR;
    }
    
    /* Configure injected channel */
    if(HAL_OK != HAL_DFSDM_FilterConfigRegChannel(&hAudioInButtomLeftFilter, AUDIO_DFSDMx_BUTTOM_LEFT_CHANNEL, DFSDM_CONTINUOUS_CONV_ON))
    {
      return AUDIO_ERROR;
    }
    
    /* FILTER 3 configuration */
    __HAL_DFSDM_FILTER_RESET_HANDLE_STATE(&hAudioInButtomRightFilter);
    hAudioInButtomRightFilter.Instance                          = AUDIO_DFSDMx_BUTTOM_RIGHT_FILTER;
    hAudioInButtomRightFilter.Init.RegularParam.Trigger         = DFSDM_FILTER_SYNC_TRIGGER;
    hAudioInButtomRightFilter.Init.RegularParam.FastMode        = ENABLE;
    hAudioInButtomRightFilter.Init.RegularParam.DmaMode         = ENABLE;
    hAudioInButtomRightFilter.Init.InjectedParam.Trigger        = DFSDM_FILTER_SW_TRIGGER;
    hAudioInButtomRightFilter.Init.InjectedParam.ScanMode       = DISABLE;
    hAudioInButtomRightFilter.Init.InjectedParam.DmaMode        = DISABLE;
    hAudioInButtomRightFilter.Init.InjectedParam.ExtTrigger     = DFSDM_FILTER_EXT_TRIG_TIM1_TRGO;
    hAudioInButtomRightFilter.Init.InjectedParam.ExtTriggerEdge = DFSDM_FILTER_EXT_TRIG_RISING_EDGE;
    hAudioInButtomRightFilter.Init.FilterParam.SincOrder        = DFSDM_FILTER_ORDER(AudioFreq);
    /* Set the DFSDM Filters Oversampling to have correct sample rate */
    hAudioInButtomRightFilter.Init.FilterParam.Oversampling     = DFSDM_OVER_SAMPLING(AudioFreq);
    hAudioInButtomRightFilter.Init.FilterParam.IntOversampling  = 1;
    if(HAL_OK != HAL_DFSDM_FilterInit(&hAudioInButtomRightFilter))
    {
      return AUDIO_ERROR;
    }
    /* Configure injected channel */
    if(HAL_OK != HAL_DFSDM_FilterConfigRegChannel(&hAudioInButtomRightFilter, AUDIO_DFSDMx_BUTTOM_RIGHT_CHANNEL, DFSDM_CONTINUOUS_CONV_ON))
    {
      return AUDIO_ERROR;
    } 
  }
  return AUDIO_OK;
}

/**
  * @brief  De-initialize the Digital Filter for Sigma-Delta Modulators interface (DFSDM).
  * @retval AUDIO_OK if correct communication, else wrong communication
  */
static uint8_t DFSDMx_DeInit(void)
{  
  /* De-initializes the DFSDM filters to allow access to DFSDM internal registers */
  if(HAL_OK != HAL_DFSDM_FilterDeInit(&hAudioInTopLeftFilter))
  {
    return AUDIO_ERROR;
  }
  
  if(HAL_OK != HAL_DFSDM_FilterDeInit(&hAudioInTopRightFilter))
  {
    return AUDIO_ERROR;
  }
  
  /* De-initializes the DFSDM channels to allow access to DFSDM internal registers */
  if(HAL_OK != HAL_DFSDM_ChannelDeInit(&hAudioInTopLeftChannel))
  {
    return AUDIO_ERROR;
  }
  
  if(HAL_OK != HAL_DFSDM_ChannelDeInit(&hAudioInTopRightChannel))
  {
    return AUDIO_ERROR;
  }

  if(AudioIn_ChannelNumber > 2)
  {
    /* De-initializes the DFSDM filters to allow access to DFSDM internal registers */        
    if(HAL_OK != HAL_DFSDM_FilterDeInit(&hAudioInButtomLeftFilter))
    {
      return AUDIO_ERROR;
    }
    
    if(HAL_OK != HAL_DFSDM_FilterDeInit(&hAudioInButtomRightFilter))
    {
      return AUDIO_ERROR;
    }
    
    /* De-initializes the DFSDM channels to allow access to DFSDM internal registers */  
    if(HAL_OK != HAL_DFSDM_ChannelDeInit(&hAudioInButtomLeftChannel))
    {
      return AUDIO_ERROR;
    }
    
    if(HAL_OK != HAL_DFSDM_ChannelDeInit(&hAudioInButtomRightChannel))
    {
      return AUDIO_ERROR;
    }  
  }
  
  return AUDIO_OK;
}

/**
  * @brief  Initialize the DFSDM channel MSP.
  * @retval None
  */
static void DFSDMx_ChannelMspInit(void)
{
  GPIO_InitTypeDef  GPIO_InitStruct;  
  
  /* Enable DFSDM clock */
  AUDIO_DFSDMx_CLK_ENABLE();
  
  /* Enable GPIO clock */
  AUDIO_DFSDMx_DMIC_DATIN_GPIO_CLK_ENABLE();
  AUDIO_DFSDMx_CKOUT_DMIC_GPIO_CLK_ENABLE();
  
  /* DFSDM pins configuration: DFSDM_CKOUT, DMIC_DATIN1 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_AF;
  HAL_GPIO_Init(AUDIO_DFSDMx_CKOUT_DMIC_GPIO_PORT, &GPIO_InitStruct);
  
  /* DFSDM pin configuration: DMIC_DATIN1 pin --------------------------------*/
  GPIO_InitStruct.Pin = AUDIO_DFSDMx_DMIC_DATIN1_PIN;
  GPIO_InitStruct.Alternate = AUDIO_DFSDMx_DMIC_DATIN_AF;
  HAL_GPIO_Init(AUDIO_DFSDMx_DMIC_DATIN_GPIO_PORT, &GPIO_InitStruct);
  
  if(AudioIn_ChannelNumber > 2)
  {
  /* DFSDM pin configuration: DMIC_DATIN5 pin --------------------------------*/  
    GPIO_InitStruct.Pin = AUDIO_DFSDMx_DMIC_DATIN5_PIN;
    GPIO_InitStruct.Alternate = AUDIO_DFSDMx_DMIC_DATIN_AF;
    HAL_GPIO_Init(AUDIO_DFSDMx_DMIC_DATIN_GPIO_PORT, &GPIO_InitStruct);    
  }
}

/**
  * @brief  DeInitialize the DFSDM channel MSP.
  * @retval None
  */
static void DFSDMx_ChannelMspDeInit(void)
{
  GPIO_InitTypeDef  GPIO_InitStruct;  
  
  /* DFSDM pin configuration: DMIC_DATIN1 pin --------------------------------*/
  GPIO_InitStruct.Pin = AUDIO_DFSDMx_CKOUT_PIN;
  HAL_GPIO_DeInit(AUDIO_DFSDMx_CKOUT_DMIC_GPIO_PORT, GPIO_InitStruct.Pin);
  GPIO_InitStruct.Pin = AUDIO_DFSDMx_DMIC_DATIN1_PIN;
  HAL_GPIO_DeInit(AUDIO_DFSDMx_DMIC_DATIN_GPIO_PORT, GPIO_InitStruct.Pin);
  
  if(AudioIn_ChannelNumber > 2)
  {
    /* DFSDM pin configuration: DMIC_DATIN5 pin ------------------------------*/
    GPIO_InitStruct.Pin = AUDIO_DFSDMx_CKOUT_PIN;
    HAL_GPIO_DeInit(AUDIO_DFSDMx_CKOUT_DMIC_GPIO_PORT, GPIO_InitStruct.Pin);
    GPIO_InitStruct.Pin = AUDIO_DFSDMx_DMIC_DATIN5_PIN;
    HAL_GPIO_DeInit(AUDIO_DFSDMx_DMIC_DATIN_GPIO_PORT, GPIO_InitStruct.Pin);    
  }
}

/**
  * @brief  Initialize the DFSDM filter MSP.
  * @retval None
  */
static void DFSDMx_FilterMspInit(void)
{  
  /* Enable DFSDM clock */
  AUDIO_DFSDMx_CLK_ENABLE();
  
  /* Enable the DMA clock */
  AUDIO_DFSDMx_DMAx_CLK_ENABLE();
  
  /*********** Configure DMA stream for TOP LEFT microphone *******************/
  hDmaTopLeft.Init.Direction           = DMA_PERIPH_TO_MEMORY;
  hDmaTopLeft.Init.PeriphInc           = DMA_PINC_DISABLE;
  hDmaTopLeft.Init.MemInc              = DMA_MINC_ENABLE;
  hDmaTopLeft.Init.PeriphDataAlignment = AUDIO_DFSDMx_DMAx_PERIPH_DATA_SIZE;
  hDmaTopLeft.Init.MemDataAlignment    = AUDIO_DFSDMx_DMAx_MEM_DATA_SIZE;
  hDmaTopLeft.Init.Mode                = DMA_CIRCULAR;
  hDmaTopLeft.Init.Priority            = DMA_PRIORITY_HIGH;
  hDmaTopLeft.Instance                 = AUDIO_DFSDMx_DMAx_TOP_LEFT_STREAM;
  hDmaTopLeft.Init.Channel             = AUDIO_DFSDMx_DMAx_CHANNEL; 
  
  /* Associate the DMA handle */
  __HAL_LINKDMA(&hAudioInTopLeftFilter, hdmaReg, hDmaTopLeft);
  
  /* Reset DMA handle state */
  __HAL_DMA_RESET_HANDLE_STATE(&hDmaTopLeft);
  
  /* Configure the DMA Channel */
  HAL_DMA_Init(&hDmaTopLeft);      
  
  /* DMA IRQ Channel configuration */
  HAL_NVIC_SetPriority(AUDIO_DFSDMx_DMAx_TOP_LEFT_IRQ, AUDIO_OUT_IRQ_PREPRIO, 0);
  HAL_NVIC_EnableIRQ(AUDIO_DFSDMx_DMAx_TOP_LEFT_IRQ);
  
  
  /*********** Configure DMA stream for TOP RIGHT microphone ******************/
  hDmaTopRight.Init.Direction           = DMA_PERIPH_TO_MEMORY;
  hDmaTopRight.Init.PeriphInc           = DMA_PINC_DISABLE;
  hDmaTopRight.Init.MemInc              = DMA_MINC_ENABLE;
  hDmaTopRight.Init.PeriphDataAlignment = AUDIO_DFSDMx_DMAx_PERIPH_DATA_SIZE;
  hDmaTopRight.Init.MemDataAlignment    = AUDIO_DFSDMx_DMAx_MEM_DATA_SIZE;
  hDmaTopRight.Init.Mode                = DMA_CIRCULAR;
  hDmaTopRight.Init.Priority            = DMA_PRIORITY_HIGH;  
  hDmaTopRight.Instance                 = AUDIO_DFSDMx_DMAx_TOP_RIGHT_STREAM;
  hDmaTopRight.Init.Channel             = AUDIO_DFSDMx_DMAx_CHANNEL;
  
  /* Associate the DMA handle */
  __HAL_LINKDMA(&hAudioInTopRightFilter, hdmaReg, hDmaTopRight);
  
  /* Reset DMA handle state */
  __HAL_DMA_RESET_HANDLE_STATE(&hDmaTopRight);
  
  /* Configure the DMA Channel */
  HAL_DMA_Init(&hDmaTopRight);      
  
  /* DMA IRQ Channel configuration */
  HAL_NVIC_SetPriority(AUDIO_DFSDMx_DMAx_TOP_RIGHT_IRQ, AUDIO_OUT_IRQ_PREPRIO, 0);
  HAL_NVIC_EnableIRQ(AUDIO_DFSDMx_DMAx_TOP_RIGHT_IRQ);
  
  if(AudioIn_ChannelNumber > 2)
  {  
    /*********** Configure DMA stream for BUTTOM LEFT microphone ****************/
    hDmaButtomLeft.Init.Direction           = DMA_PERIPH_TO_MEMORY;
    hDmaButtomLeft.Init.PeriphInc           = DMA_PINC_DISABLE;
    hDmaButtomLeft.Init.MemInc              = DMA_MINC_ENABLE;
    hDmaButtomLeft.Init.PeriphDataAlignment = AUDIO_DFSDMx_DMAx_PERIPH_DATA_SIZE;
    hDmaButtomLeft.Init.MemDataAlignment    = AUDIO_DFSDMx_DMAx_MEM_DATA_SIZE;
    hDmaButtomLeft.Init.Mode                = DMA_CIRCULAR;
    hDmaButtomLeft.Init.Priority            = DMA_PRIORITY_HIGH;
    hDmaButtomLeft.Instance                 = AUDIO_DFSDMx_DMAx_BUTTOM_LEFT_STREAM;
    hDmaButtomLeft.Init.Channel             = AUDIO_DFSDMx_DMAx_CHANNEL; 
    
    /* Associate the DMA handle */
    __HAL_LINKDMA(&hAudioInButtomLeftFilter, hdmaReg, hDmaButtomLeft);
    
    /* Reset DMA handle state */
    __HAL_DMA_RESET_HANDLE_STATE(&hDmaButtomLeft);
    
    /* Configure the DMA Channel */
    HAL_DMA_Init(&hDmaButtomLeft);      
    
    /* DMA IRQ Channel configuration */
    HAL_NVIC_SetPriority(AUDIO_DFSDMx_DMAx_BUTTOM_LEFT_IRQ, AUDIO_OUT_IRQ_PREPRIO, 0);
    HAL_NVIC_EnableIRQ(AUDIO_DFSDMx_DMAx_BUTTOM_LEFT_IRQ);
    
    
    /*********** Configure DMA stream for BUTTOM RIGHT microphone ***************/
    hDmaButtomRight.Init.Direction           = DMA_PERIPH_TO_MEMORY;
    hDmaButtomRight.Init.PeriphInc           = DMA_PINC_DISABLE;
    hDmaButtomRight.Init.MemInc              = DMA_MINC_ENABLE;
    hDmaButtomRight.Init.PeriphDataAlignment = AUDIO_DFSDMx_DMAx_PERIPH_DATA_SIZE;
    hDmaButtomRight.Init.MemDataAlignment    = AUDIO_DFSDMx_DMAx_MEM_DATA_SIZE;
    hDmaButtomRight.Init.Mode                = DMA_CIRCULAR;
    hDmaButtomRight.Init.Priority            = DMA_PRIORITY_HIGH;  
    hDmaButtomRight.Instance                 = AUDIO_DFSDMx_DMAx_BUTTOM_RIGHT_STREAM;
    hDmaButtomRight.Init.Channel             = AUDIO_DFSDMx_DMAx_CHANNEL;
    
    /* Associate the DMA handle */
    __HAL_LINKDMA(&hAudioInButtomRightFilter, hdmaReg, hDmaButtomRight);
    
    /* Reset DMA handle state */
    __HAL_DMA_RESET_HANDLE_STATE(&hDmaButtomRight);
    
    /* Configure the DMA Channel */
    HAL_DMA_Init(&hDmaButtomRight);      
    
    /* DMA IRQ Channel configuration */
    HAL_NVIC_SetPriority(AUDIO_DFSDMx_DMAx_BUTTOM_RIGHT_IRQ, AUDIO_OUT_IRQ_PREPRIO, 0);
    HAL_NVIC_EnableIRQ(AUDIO_DFSDMx_DMAx_BUTTOM_RIGHT_IRQ); 
  }
}

/**
  * @brief  DeInitialize the DFSDM filter MSP.
  * @retval None
  */
static void DFSDMx_FilterMspDeInit(void)
{
  /* Configure the DMA Channel */
  HAL_DMA_DeInit(&hDmaTopLeft);
  HAL_DMA_DeInit(&hDmaTopRight);
  if(AudioIn_ChannelNumber > 2)
  {
    HAL_DMA_DeInit(&hDmaButtomLeft);
    HAL_DMA_DeInit(&hDmaButtomRight);
  }  
}

/**
  * @}
  */ 
  
/**
  * @}
  */

/**
  * @}
  */

/**
  * @}
  */ 

/************************ (C) COPYRIGHT STMicroelectronics *****END OF FILE****/
