Nirvana Jay
patch for F746 demo
Fork of
BSP_DISCO_F746NG
by 
ST
Diff: stm32746g_discovery_audio.c
- Revision:
- 1:e8fac4061a5b
--- /dev/null Thu Jan 01 00:00:00 1970 +0000
+++ b/stm32746g_discovery_audio.c Mon Nov 02 23:35:17 2015 +0000
@@ -0,0 +1,1361 @@
+/**
+ ******************************************************************************
+ * @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>© 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 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);
+ }
+
+ /* SAI DMA IRQ Channel configuration */
+ HAL_NVIC_SetPriority(AUDIO_OUT_SAIx_DMAx_IRQ, AUDIO_OUT_IRQ_PREPRIO, 0);
+ HAL_NVIC_EnableIRQ(AUDIO_OUT_SAIx_DMAx_IRQ);
+}
+
+/**
+ * @brief Deinitializes SAI MSP.
+ * @param hsai: SAI handle
+ * @param Params
+ * @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 */
+ 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 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);
+}
+
+
+/**
+ * @}
+ */
+
+/**
+ * @}
+ */
+
+/**
+ * @}
+ */
+
+/**
+ * @}
+ */
+
+/************************ (C) COPYRIGHT STMicroelectronics *****END OF FILE****/
+