Library to control Silicon Labs SI570 10 MHZ TO 1.4 GHZ I2C PROGRAMMABLE XO/VCXO.

Dependencies:   mbed

Fork of SI570 by Gerrit Polder

Revision:
1:1556bcaaf759
--- /dev/null	Thu Jan 01 00:00:00 1970 +0000
+++ b/BSP_F746NG/stm32746g_discovery_audio.c	Sun Mar 27 06:55:59 2016 +0000
@@ -0,0 +1,1425 @@
+/**
+  ******************************************************************************
+  * @file    stm32746g_discovery_audio.c
+  * @author  MCD Application Team
+  * @version V1.0.0
+  * @date    25-June-2015
+  * @brief   This file provides the Audio driver for the STM32746G-Discovery board.
+  @verbatim
+    How To use this driver:
+    -----------------------
+       + This driver supports STM32F7xx devices on STM32746G-Discovery (MB1191) board.
+       + 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 STM32746G-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_DISCOVERY_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 AUDIO_OUT_XXX_CallBack() and only their prototypes are declared in 
+          the stm32746g_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().
+       + The driver API and the callback functions are at the end of the stm32746g_discovery_audio.h file.
+     
+    Driver architecture:
+    --------------------
+       + This driver provides the High Audio Layer: consists of the function API exported in the stm32746g_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 stm32746g_discovery_audio_codec.c file (SAIx_Out_Init() and SAIx_Out_DeInit(), SAIx_In_Init() and SAIx_In_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) 2015 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 "stm32746g_discovery_audio.h"
+
+/** @addtogroup BSP
+  * @{
+  */
+
+/** @addtogroup STM32746G_DISCOVERY
+  * @{
+  */ 
+  
+/** @defgroup STM32746G_DISCOVERY_AUDIO STM32746G_DISCOVERY AUDIO
+  * @brief This file includes the low layer driver for wm8994 Audio Codec
+  *        available on STM32746G-Discovery board(MB1191).
+  * @{
+  */ 
+
+/** @defgroup STM32746G_DISCOVERY_AUDIO_Private_Types STM32746G_DISCOVERY AUDIO Private Types
+  * @{
+  */ 
+/**
+  * @}
+  */ 
+  
+/** @defgroup STM32746G_DISCOVERY_AUDIO_Private_Defines STM32746G_DISCOVERY AUDIO Private Defines
+  * @{
+  */
+/**
+  * @}
+  */ 
+
+/** @defgroup STM32746G_DISCOVERY_AUDIO_Private_Macros STM32746G_DISCOVERY AUDIO Private Macros
+  * @{
+  */
+/**
+  * @}
+  */ 
+  
+/** @defgroup STM32746G_DISCOVERY_AUDIO_Private_Variables STM32746G_DISCOVERY AUDIO Private Variables
+  * @{
+  */
+AUDIO_DrvTypeDef          *audio_drv;
+SAI_HandleTypeDef         haudio_out_sai={0};
+SAI_HandleTypeDef         haudio_in_sai={0};
+TIM_HandleTypeDef         haudio_tim;
+
+uint16_t __IO AudioInVolume = DEFAULT_AUDIO_IN_VOLUME;
+    
+/**
+  * @}
+  */ 
+
+/** @defgroup STM32746G_DISCOVERY_AUDIO_Private_Function_Prototypes STM32746G_DISCOVERY AUDIO Private Function Prototypes
+  * @{
+  */
+static void AUDIO_IN_INT_IRQHandler(void);
+static void AUDIO_IN_SAIx_DMAx_IRQHandler(void);
+static void AUDIO_OUT_SAIx_DMAx_IRQHandler(void);
+static void SAIx_Out_Init(uint32_t AudioFreq);
+static void SAIx_Out_DeInit(void);
+static void SAIx_In_Init(uint32_t SaiOutMode, uint32_t SlotActive, uint32_t AudioFreq);
+static void SAIx_In_DeInit(void);
+/**
+  * @}
+  */ 
+
+/** @defgroup STM32746G_DISCOVERY_AUDIO_OUT_Exported_Functions STM32746G_DISCOVERY AUDIO Out Exported 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.
+  * @note   The I2S PLL input clock must be done in the user application.  
+  * @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 on 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 in BYTES unit.
+  *         In memory, first element is for left channel, second element is for right channel
+  * @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.
+  * @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  This function  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
+  *         This parameter can be one of the following values
+  *            @arg CODEC_AUDIOFRAME_SLOT_0123
+  *            @arg CODEC_AUDIOFRAME_SLOT_02
+  *            @arg CODEC_AUDIOFRAME_SLOT_13
+  * @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  Deinit the audio peripherals.
+  * @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 stm32746g_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 stm32746g_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)
+{
+  HAL_SAI_StateTypeDef audio_out_state;
+  HAL_SAI_StateTypeDef audio_in_state;
+
+  audio_out_state = HAL_SAI_GetState(&haudio_out_sai);
+  audio_in_state = HAL_SAI_GetState(&haudio_in_sai);
+
+  /* Determines if it is an audio out or audio in error */
+  if ((audio_out_state == HAL_SAI_STATE_BUSY) || (audio_out_state == HAL_SAI_STATE_BUSY_TX)
+   || (audio_out_state == HAL_SAI_STATE_TIMEOUT) || (audio_out_state == HAL_SAI_STATE_ERROR))
+  {
+    BSP_AUDIO_OUT_Error_CallBack();
+  }
+
+  if ((audio_in_state == HAL_SAI_STATE_BUSY) || (audio_in_state == HAL_SAI_STATE_BUSY_RX)
+   || (audio_in_state == HAL_SAI_STATE_TIMEOUT) || (audio_in_state == HAL_SAI_STATE_ERROR))
+  {
+    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_SCK_SD_ENABLE();
+  AUDIO_OUT_SAIx_FS_ENABLE();
+  /* CODEC_SAI pins configuration: FS, SCK, MCK and SD pins ------------------*/
+  gpio_init_structure.Pin = AUDIO_OUT_SAIx_FS_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_FS_SD_MCLK_AF;
+  HAL_GPIO_Init(AUDIO_OUT_SAIx_FS_GPIO_PORT, &gpio_init_structure);
+
+  gpio_init_structure.Pin = AUDIO_OUT_SAIx_SCK_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_SCK_AF;
+  HAL_GPIO_Init(AUDIO_OUT_SAIx_SCK_SD_GPIO_PORT, &gpio_init_structure);
+
+  gpio_init_structure.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_FS_SD_MCLK_AF;
+  HAL_GPIO_Init(AUDIO_OUT_SAIx_SCK_SD_GPIO_PORT, &gpio_init_structure);
+
+  gpio_init_structure.Pin = AUDIO_OUT_SAIx_MCLK_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_FS_SD_MCLK_AF;
+  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);      
+  }
+#if ( __MBED__ == 1)
+    // Enable interrupt
+    IRQn_Type irqn = (IRQn_Type)(AUDIO_OUT_SAIx_DMAx_IRQ);
+    NVIC_ClearPendingIRQ(irqn);
+    NVIC_DisableIRQ(irqn);
+    NVIC_SetPriority(irqn, AUDIO_OUT_IRQ_PREPRIO);
+    NVIC_SetVector(irqn, (uint32_t)AUDIO_OUT_SAIx_DMAx_IRQHandler);
+    NVIC_EnableIRQ(irqn);
+
+#else
+  /* 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); 
+#endif
+}
+
+/**
+  * @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);  
+
+    /* Deactives CODEC_SAI pins FS, SCK, MCK and SD by putting them in input mode */
+    gpio_init_structure.Pin = AUDIO_OUT_SAIx_FS_PIN;
+    HAL_GPIO_DeInit(AUDIO_OUT_SAIx_FS_GPIO_PORT, gpio_init_structure.Pin);
+
+    gpio_init_structure.Pin = AUDIO_OUT_SAIx_SCK_PIN;
+    HAL_GPIO_DeInit(AUDIO_OUT_SAIx_SCK_SD_GPIO_PORT, gpio_init_structure.Pin);
+
+    gpio_init_structure.Pin =  AUDIO_OUT_SAIx_SD_PIN;
+    HAL_GPIO_DeInit(AUDIO_OUT_SAIx_SCK_SD_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 application
+       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 PLLI2S prescalers */
+    /* PLLI2S_VCO: VCO_429M
+    I2S_CLK(first level) = PLLI2S_VCO/PLLI2SQ = 429/2 = 214.5 Mhz
+    I2S_CLK_x = I2S_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_48K), AUDIO_FREQUENCY_96K */
+  {
+    /* I2S clock config
+    PLLI2S_VCO: VCO_344M
+    I2S_CLK(first level) = PLLI2S_VCO/PLLI2SQ = 344/7 = 49.142 Mhz
+    I2S_CLK_x = I2S_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);
+  }
+}
+
+/*******************************************************************************
+                            Static Functions
+*******************************************************************************/
+
+/**
+  * @brief  Initializes the output 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.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;
+  
+  /* 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_0123;
+
+  HAL_SAI_Init(&haudio_out_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_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);
+}
+
+/**
+  * @}
+  */
+
+/** @defgroup STM32746G_DISCOVERY_AUDIO_Out_Private_Functions STM32746G_DISCOVERY_AUDIO Out Private Functions
+  * @{
+  */ 
+  
+/**
+  * @brief  Initializes wave recording.
+  * @param  InputDevice: INPUT_DEVICE_DIGITAL_MICROPHONE_2 or INPUT_DEVICE_INPUT_LINE_1
+  * @param  Volume: Initial volume level (in range 0(Mute)..80(+0dB)..100(+17.625dB))
+  * @param  AudioFreq: Audio frequency to be configured for the SAI peripheral.
+  * @retval AUDIO_OK if correct communication, else wrong communication
+  */
+uint8_t BSP_AUDIO_IN_Init(uint16_t InputDevice, uint8_t Volume, uint32_t AudioFreq)
+{
+  uint8_t ret = AUDIO_ERROR;
+  uint32_t deviceid = 0x00;
+  uint32_t slot_active;
+
+  if ((InputDevice != INPUT_DEVICE_INPUT_LINE_1) &&       /* Only INPUT_LINE_1 and MICROPHONE_2 inputs supported */
+      (InputDevice != INPUT_DEVICE_DIGITAL_MICROPHONE_2))
+  {
+    ret = AUDIO_ERROR;
+  }
+  else
+  {
+    /* Disable SAI */
+    SAIx_In_DeInit();
+
+    /* PLL clock is set depending on the AudioFreq (44.1khz vs 48khz groups) */
+    BSP_AUDIO_OUT_ClockConfig(&haudio_in_sai, 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 */
+    haudio_in_sai.Instance = AUDIO_IN_SAIx;
+    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_OUT_MspInit(&haudio_in_sai, NULL);  /* Initialize GPIOs for SAI2 block A Master signals */
+      BSP_AUDIO_IN_MspInit(&haudio_in_sai, NULL);
+    }
+
+    /* Configure SAI in master RX mode :
+     *   - SAI2_block_A in master RX mode
+     *   - SAI2_block_B in slave RX mode synchronous from SAI2_block_A
+     */
+    if (InputDevice == INPUT_DEVICE_DIGITAL_MICROPHONE_2)
+    {
+      slot_active = CODEC_AUDIOFRAME_SLOT_13;
+    }
+    else
+    {
+      slot_active = CODEC_AUDIOFRAME_SLOT_02;
+    }
+    SAIx_In_Init(SAI_MODEMASTER_RX, 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, Volume, AudioFreq);
+    }
+  }
+  return ret;
+}
+
+/**
+  * @brief  Initializes wave recording and playback in parallel.
+  * @param  InputDevice: INPUT_DEVICE_DIGITAL_MICROPHONE_2
+  * @param  OutputDevice: OUTPUT_DEVICE_SPEAKER, OUTPUT_DEVICE_HEADPHONE,
+  *                       or OUTPUT_DEVICE_BOTH.
+  * @param  Volume: Initial volume level (in range 0(Mute)..80(+0dB)..100(+17.625dB))
+  * @param  AudioFreq: Audio frequency to be configured for the SAI peripheral.
+  * @retval AUDIO_OK if correct communication, else wrong communication
+  */
+uint8_t BSP_AUDIO_IN_OUT_Init(uint16_t InputDevice, uint16_t OutputDevice, uint8_t Volume, uint32_t AudioFreq)
+{
+  uint8_t ret = AUDIO_ERROR;
+  uint32_t deviceid = 0x00;
+  uint32_t slot_active;
+
+  if (InputDevice != INPUT_DEVICE_DIGITAL_MICROPHONE_2)  /* Only MICROPHONE_2 input supported */
+  {
+    ret = AUDIO_ERROR;
+  }
+  else
+  {
+    /* Disable SAI */
+    SAIx_In_DeInit();
+    SAIx_Out_DeInit();
+
+    /* PLL clock is set depending on the AudioFreq (44.1khz vs 48khz groups) */
+    BSP_AUDIO_OUT_ClockConfig(&haudio_in_sai, 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 */
+    haudio_in_sai.Instance = AUDIO_IN_SAIx;
+    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(&haudio_in_sai, 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);
+    }
+
+    /* Configure SAI in master mode :
+     *   - SAI2_block_A in master TX mode
+     *   - SAI2_block_B in slave RX mode synchronous from SAI2_block_A
+     */
+    if (InputDevice == INPUT_DEVICE_DIGITAL_MICROPHONE_2)
+    {
+      slot_active = CODEC_AUDIOFRAME_SLOT_13;
+    }
+    else
+    {
+      slot_active = CODEC_AUDIOFRAME_SLOT_02;
+    }
+    SAIx_In_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, Volume, AudioFreq);
+    }
+  }
+  return ret;
+}
+
+
+/**
+  * @brief  Starts audio recording.
+  * @param  pbuf: Main buffer pointer for the recorded data storing  
+  * @param  size: size of the recorded buffer in number of elements (typically number of half-words)
+  *               Be careful that it is not the same unit than BSP_AUDIO_OUT_Play function
+  * @retval AUDIO_OK if correct communication, else wrong communication
+  */
+uint8_t  BSP_AUDIO_IN_Record(uint16_t* pbuf, uint32_t size)
+{
+  uint32_t ret = AUDIO_ERROR;
+  
+  /* Start the process receive DMA */
+  HAL_SAI_Receive_DMA(&haudio_in_sai, (uint8_t*)pbuf, size);
+  
+  /* Return AUDIO_OK when all operations are correctly done */
+  ret = AUDIO_OK;
+  
+  return ret;
+}
+
+/**
+  * @brief  Stops audio recording.
+  * @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_IN_Stop(uint32_t Option)
+{
+  /* Call the Media layer stop function */
+  HAL_SAI_DMAStop(&haudio_in_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  Pauses the audio file stream.
+  * @retval AUDIO_OK if correct communication, else wrong communication
+  */
+uint8_t BSP_AUDIO_IN_Pause(void)
+{    
+  /* Call the Media layer pause function */
+  HAL_SAI_DMAPause(&haudio_in_sai);
+  /* Return AUDIO_OK when all operations are correctly done */
+  return AUDIO_OK;
+}
+
+/**
+  * @brief  Resumes the audio file stream.
+  * @retval AUDIO_OK if correct communication, else wrong communication
+  */
+uint8_t BSP_AUDIO_IN_Resume(void)
+{    
+  /* Call the Media layer pause/resume function */
+  HAL_SAI_DMAResume(&haudio_in_sai);
+  /* Return AUDIO_OK when all operations are correctly done */
+  return AUDIO_OK;
+}
+
+/**
+  * @brief  Controls the audio in volume level. 
+  * @param  Volume: Volume level in range 0(Mute)..80(+0dB)..100(+17.625dB)
+  * @retval AUDIO_OK if correct communication, else wrong communication
+  */
+uint8_t BSP_AUDIO_IN_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
+  {
+    /* Set the Global variable AudioInVolume  */
+    AudioInVolume = Volume;
+    /* Return AUDIO_OK when all operations are correctly done */
+    return AUDIO_OK;
+  }
+}
+
+/**
+  * @brief  Deinit the audio IN peripherals.
+  * @retval None
+  */
+void BSP_AUDIO_IN_DeInit(void)
+{
+  SAIx_In_DeInit();
+  /* DeInit the SAI MSP : this __weak function can be rewritten by the application */
+  BSP_AUDIO_IN_MspDeInit(&haudio_in_sai, NULL);
+}
+
+ /**
+  * @brief  Rx Transfer completed callbacks.
+  * @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  Rx Half Transfer completed callbacks.
+  * @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 stm32746g_discovery_audio.h) */
+  BSP_AUDIO_IN_HalfTransfer_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  Initializes BSP_AUDIO_IN MSP.
+  * @param  hsai: SAI handle
+  * @param  Params
+  * @retval None
+  */
+__weak void BSP_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_SD_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 */
+#if ( __MBED__ == 1)
+    IRQn_Type irqn = (IRQn_Type)(AUDIO_IN_SAIx_DMAx_IRQ);
+    NVIC_ClearPendingIRQ(irqn);
+    NVIC_DisableIRQ(irqn);
+    NVIC_SetPriority(irqn, AUDIO_IN_IRQ_PREPRIO);
+    NVIC_SetVector(irqn, (uint32_t)AUDIO_IN_SAIx_DMAx_IRQHandler);
+    NVIC_EnableIRQ(irqn);
+#else
+  HAL_NVIC_SetPriority(AUDIO_IN_SAIx_DMAx_IRQ, AUDIO_IN_IRQ_PREPRIO, 0);
+  HAL_NVIC_EnableIRQ(AUDIO_IN_SAIx_DMAx_IRQ);
+#endif
+
+  /* Audio INT IRQ Channel configuration */
+#if ( __MBED__ == 1)
+    irqn = (IRQn_Type)(AUDIO_IN_INT_IRQ);
+    NVIC_ClearPendingIRQ(irqn);
+    NVIC_DisableIRQ(irqn);
+    NVIC_SetPriority(irqn, AUDIO_IN_IRQ_PREPRIO);
+    NVIC_SetVector(irqn, (uint32_t)AUDIO_IN_INT_IRQHandler);
+    NVIC_EnableIRQ(irqn);
+#else
+  HAL_NVIC_SetPriority(AUDIO_IN_INT_IRQ, AUDIO_IN_IRQ_PREPRIO, 0);
+  HAL_NVIC_EnableIRQ(AUDIO_IN_INT_IRQ);
+#endif
+}
+
+/**
+  * @brief  DeInitializes BSP_AUDIO_IN MSP.
+  * @param  hsai: SAI handle
+  * @param  Params
+  * @retval None
+  */
+__weak void BSP_AUDIO_IN_MspDeInit(SAI_HandleTypeDef *hsai, void *Params)
+{
+  GPIO_InitTypeDef  gpio_init_structure;
+
+  static DMA_HandleTypeDef hdma_sai_rx;
+
+  /* SAI IN DMA IRQ Channel deactivation */
+  HAL_NVIC_DisableIRQ(AUDIO_IN_SAIx_DMAx_IRQ);
+
+  if(hsai->Instance == AUDIO_IN_SAIx)
+  {
+    /* Deinitialize the Stream for new transfer */
+    HAL_DMA_DeInit(&hdma_sai_rx);
+  }
+
+ /* Disable SAI block */
+  __HAL_SAI_DISABLE(hsai);
+
+  /* Disable pin: SD pin */
+  gpio_init_structure.Pin = AUDIO_IN_SAIx_SD_PIN;
+  HAL_GPIO_DeInit(AUDIO_IN_SAIx_SD_GPIO_PORT, gpio_init_structure.Pin);
+
+  /* Disable SAI clock */
+  AUDIO_IN_SAIx_CLK_DISABLE();
+
+  /* GPIO pins clock and DMA clock can be shut down in the application
+     by surcharging this __weak function */
+}
+
+
+/*******************************************************************************
+                            Static Functions
+*******************************************************************************/
+
+/**
+  * @brief  Initializes the input Audio Codec audio interface (SAI).
+  * @param  SaiOutMode: SAI_MODEMASTER_TX (for record and playback in parallel)
+  *                     or SAI_MODEMASTER_RX (for record only).
+  * @param  SlotActive: CODEC_AUDIOFRAME_SLOT_02 or CODEC_AUDIOFRAME_SLOT_13
+  * @param  AudioFreq: Audio frequency to be configured for the SAI peripheral.
+  * @retval None
+  */
+static void SAIx_In_Init(uint32_t SaiOutMode, uint32_t SlotActive, uint32_t AudioFreq)
+{
+  /* Initialize SAI2 block A in MASTER RX */
+  /* 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.AudioFrequency = AudioFreq;
+  haudio_out_sai.Init.AudioMode = SaiOutMode;
+  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;
+
+  /* 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 = SlotActive;
+
+  HAL_SAI_Init(&haudio_out_sai);
+
+  /* Initialize SAI2 block B in SLAVE RX synchronous from SAI2 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.AudioFrequency = AudioFreq;
+  haudio_in_sai.Init.AudioMode = SAI_MODESLAVE_RX;
+  haudio_in_sai.Init.NoDivider = SAI_MASTERDIVIDER_ENABLED;
+  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_DISABLED;
+  haudio_in_sai.Init.FIFOThreshold = SAI_FIFOTHRESHOLD_1QF;
+  
+  /* 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 = SlotActive;
+
+  HAL_SAI_Init(&haudio_in_sai);
+
+  /* Enable SAI peripheral to generate MCLK */
+  __HAL_SAI_ENABLE(&haudio_out_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(void)
+{
+  /* Initialize the haudio_in_sai Instance parameter */
+  haudio_in_sai.Instance = AUDIO_IN_SAIx;
+
+  /* Disable SAI peripheral */
+  __HAL_SAI_DISABLE(&haudio_in_sai);
+
+  HAL_SAI_DeInit(&haudio_in_sai);
+}
+
+/**
+  * @brief  This function handles External line 15_10 interrupt request.
+  * @param  None
+  * @retval None
+  */
+static void AUDIO_IN_INT_IRQHandler(void)
+{
+  /* Interrupt handler shared between SD_DETECT pin, USER_KEY button and touch screen interrupt */
+  if (__HAL_GPIO_EXTI_GET_IT(AUDIO_IN_INT_GPIO_PIN) != RESET)
+  {
+    HAL_GPIO_EXTI_IRQHandler(AUDIO_IN_INT_GPIO_PIN);   /* Audio Interrupt */
+  }
+}
+
+/**
+  * @brief This function handles DMA2 Stream 7 interrupt request.
+  * @param None
+  * @retval None
+  */
+static void AUDIO_IN_SAIx_DMAx_IRQHandler(void)
+{
+  HAL_DMA_IRQHandler(haudio_in_sai.hdmarx);
+}
+
+/**
+  * @brief  This function handles DMA2 Stream 6 interrupt request.
+  * @param  None
+  * @retval None
+  */
+static void AUDIO_OUT_SAIx_DMAx_IRQHandler(void)
+{
+  HAL_DMA_IRQHandler(haudio_out_sai.hdmatx);
+}
+
+/**
+  * @}
+  */ 
+  
+/**
+  * @}
+  */
+
+/**
+  * @}
+  */
+
+/**
+  * @}
+  */ 
+
+/************************ (C) COPYRIGHT STMicroelectronics *****END OF FILE****/
+