ST
BSP files for STM32H747I-Discovery Copy from ST Cube delivery
Dependents: DISCO_H747I_LCD_demo DISCO_H747I_AUDIO_demo
Diff: STM32H747I-Discovery/stm32h747i_discovery_audio.c
- Revision:
- 0:146cf26a9bbb
- Child:
- 1:9716849a8de8
--- /dev/null Thu Jan 01 00:00:00 1970 +0000
+++ b/STM32H747I-Discovery/stm32h747i_discovery_audio.c Wed Sep 25 13:37:39 2019 +0200
@@ -0,0 +1,1627 @@
+/**
+ ******************************************************************************
+ * @file stm32h747i_discovery_audio.c
+ * @author MCD Application Team
+ * @brief This file provides the Audio driver for the STM32H747I-DISCOVERY
+ * board.
+ @verbatim
+ How To use this driver:
+ -----------------------
+ + This driver supports STM32H7xx devices on STM32H747I-DISCOVERY (MB1248) Discovery boards.
+ + Call the function BSP_AUDIO_OUT_Init(
+ OutputDevice: physical output mode (OUTPUT_DEVICE_SPEAKER,
+ OUTPUT_DEVICE_HEADPHONE or OUTPUT_DEVICE_BOTH)
+ Volume : Initial volume to be set (0 is min (mute), 100 is max (100%)
+ AudioFreq : Audio frequency in Hz (8000, 16000, 22500, 32000...)
+ this parameter is relative to the audio file/stream type.
+ )
+ This function configures all the hardware required for the audio application (codec, I2C, SAI,
+ GPIOs, DMA and interrupt if needed). This function returns AUDIO_OK if configuration is OK.
+ If the returned value is different from AUDIO_OK or the function is stuck then the communication with
+ the codec has failed (try to un-plug the power or reset device in this case).
+ - OUTPUT_DEVICE_SPEAKER : only speaker will be set as output for the audio stream.
+ - OUTPUT_DEVICE_HEADPHONE: only headphones will be set as output for the audio stream.
+ - OUTPUT_DEVICE_BOTH : both Speaker and Headphone are used as outputs for the audio stream
+ at the same time.
+ Note. On STM32H747I-DISCOVERY SAI_DMA is configured in CIRCULAR mode. Due to this the application
+ does NOT need to call BSP_AUDIO_OUT_ChangeBuffer() to assure streaming.
+ + Call the function BSP_AUDIO_OUT_Play(
+ pBuffer: pointer to the audio data file address
+ Size : size of the buffer to be sent in Bytes
+ )
+ to start playing (for the first time) from the audio file/stream.
+ + Call the function BSP_AUDIO_OUT_Pause() to pause playing
+ + Call the function BSP_AUDIO_OUT_Resume() to resume playing.
+ Note. After calling BSP_AUDIO_OUT_Pause() function for pause, only BSP_AUDIO_OUT_Resume() should be called
+ for resume (it is not allowed to call BSP_AUDIO_OUT_Play() in this case).
+ Note. This function should be called only when the audio file is played or paused (not stopped).
+ + For each mode, you may need to implement the relative callback functions into your code.
+ The Callback functions are named BSP_AUDIO_OUT_XXX_CallBack() and only their prototypes are declared in
+ the stm32h747i_discovery_audio.h file. (refer to the example for more details on the callbacks implementations)
+ + To Stop playing, to modify the volume level, the frequency, the audio frame slot,
+ the device output mode the mute or the stop, use the functions: BSP_AUDIO_OUT_SetVolume(),
+ AUDIO_OUT_SetFrequency(), BSP_AUDIO_OUT_SetAudioFrameSlot(), BSP_AUDIO_OUT_SetOutputMode(),
+ BSP_AUDIO_OUT_SetMute() and BSP_AUDIO_OUT_Stop().
+
+ + Call the function BSP_AUDIO_IN_Init(
+ AudioFreq: Audio frequency in Hz (8000, 16000, 22500, 32000...)
+ this parameter is relative to the audio file/stream type.
+ BitRes: Bit resolution fixed to 16bit
+ ChnlNbr: Number of channel to be configured for the DFSDM peripheral
+ )
+ This function configures all the hardware required for the audio in application (channels,
+ Clock source for SAI PDM periphiral, GPIOs, DMA and interrupt if needed).
+ This function returns AUDIO_OK if configuration is OK.If the returned value is different from AUDIO_OK then
+ the configuration should be wrong.
+ + Call the function BSP_AUDIO_IN_AllocScratch(
+ pScratch: pointer to scratch tables
+ size: size of scratch buffer)
+ This function must be called before BSP_AUDIO_IN_RECORD() to allocate buffer scratch for each DFSDM channel
+ and its size.
+ Note: These buffers scratch are used as intermidiate buffers to collect data within final record buffer.
+ size is the total size of the four buffers scratch; If size is 512 then the size of each is 128.
+ This function must be called after BSP_AUDIO_IN_Init()
+ + Call the function BSP_AUDIO_IN_RECORD(
+ pBuf: pointer to the recorded audio data file address
+ Size: size of the buffer to be written in Bytes
+ )
+ to start recording from microphones.
+
+ + Call the function BSP_AUDIO_IN_Pause() to pause recording
+ + Call the function BSP_AUDIO_IN_Resume() to recording playing.
+ Note. After calling BSP_AUDIO_IN_Pause() function for pause, only BSP_AUDIO_IN_Resume() should be called
+ for resume (it is not allowed to call BSP_AUDIO_IN_RECORD() in this case).
+ + Call the function BSP_AUDIO_IN_Stop() to stop recording
+ + For each mode, you may need to implement the relative callback functions into your code.
+ The Callback functions are named BSP_AUDIO_IN_XXX_CallBack() and only their prototypes are declared in
+ the stm32h747i_discovery_audio.h file. (refer to the example for more details on the callbacks implementations)
+ + Call the function BSP_AUDIO_IN_SelectInterface(uint32_t Interface) to select one of the two interfaces
+ available on the STM32H747I-Discovery board: SAI or PDM. This function is to be called before BSP_AUDIO_IN_InitEx().
+ + Call the function BSP_AUDIO_IN_GetInterface() to get the current used interface.
+ + Call the function BSP_AUDIO_IN_PDMToPCM_Init(uint32_t AudioFreq, uint32_t ChnlNbrIn, uint32_t ChnlNbrOut)
+ to init PDM filters if the libPDMFilter is used for audio data filtering.
+ + Call the function BSP_AUDIO_IN_PDMToPCM(uint16_t* PDMBuf, uint16_t* PCMBuf) to filter PDM data to PCM format
+ if the libPDMFilter library is used for audio data filtering.
+
+ Driver architecture:
+ --------------------
+ + This driver provides the High Audio Layer: consists of the function API exported in the stm32h747i_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 stm32h747i_discovery_audio.c file (DFSDMx_Init(), DFSDMx_DeInit(), SAIx_Init() and SAIx_DeInit())
+
+ Known Limitations:
+ ------------------
+ 1- If the TDM Format used to play in parallel 2 audio Stream (the first Stream is configured in codec SLOT0 and second
+ Stream in SLOT1) the Pause/Resume, volume and mute feature will control the both streams.
+ 2- Parsing of audio file is not implemented (in order to determine audio file properties: Mono/Stereo, Data size,
+ File size, Audio Frequency, Audio Data header size ...). The configuration is fixed for the given audio file.
+ 3- Supports only Stereo audio streaming.
+ 4- Supports only 16-bits audio data size.
+ @endverbatim
+ ******************************************************************************
+ * @attention
+ *
+ * <h2><center>© Copyright (c) 2019 STMicroelectronics.
+ * All rights reserved.</center></h2>
+ *
+ * This software component is licensed by ST under BSD 3-Clause license,
+ * the "License"; You may not use this file except in compliance with the
+ * License. You may obtain a copy of the License at:
+ * opensource.org/licenses/BSD-3-Clause
+ *
+ ******************************************************************************
+ */
+/* Includes ------------------------------------------------------------------*/
+#include "stm32h747i_discovery_audio.h"
+
+/** @addtogroup BSP
+ * @{
+ */
+
+/** @addtogroup STM32H747I_DISCOVERY
+ * @{
+ */
+
+/** @defgroup STM32H747I_DISCOVERY_AUDIO STM32H747I_DISCOVERY_AUDIO
+ * @brief This file includes the low layer driver for wm8994 Audio Codec
+ * available on STM32H747I-DISCOVERY discovery board(MB1248).
+ * @{
+ */
+
+/** @defgroup STM32H747I_DISCOVERY_AUDIO_Private_Variables Private Variables
+ * @{
+ */
+/* PLAY */
+AUDIO_DrvTypeDef *audio_drv;
+SAI_HandleTypeDef haudio_out_sai;
+SAI_HandleTypeDef haudio_in_sai;
+
+/* RECORD */
+AUDIOIN_ContextTypeDef hAudioIn;
+
+
+
+/* Audio in Volume value */
+__IO uint16_t AudioInVolume = DEFAULT_AUDIO_IN_VOLUME;
+
+/* PDM filters params */
+PDM_Filter_Handler_t PDM_FilterHandler[2];
+PDM_Filter_Config_t PDM_FilterConfig[2];
+
+/**
+ * @}
+ */
+
+/** @defgroup STM32H747I_DISCOVERY_AUDIO_Private_Function_Prototypes Private FunctionPrototypes
+ * @{
+ */
+static void SAIx_Out_Init(uint32_t SaiOutMode, uint32_t SlotActive, uint32_t AudioFreq);
+static void SAIx_Out_DeInit(SAI_HandleTypeDef *hsai);
+static void SAIx_In_MspInit(SAI_HandleTypeDef *hsai, void *Params);
+static void SAIx_In_MspDeInit(SAI_HandleTypeDef *hsai, void *Params);
+static void SAIx_In_Init(uint32_t SaiInMode, uint32_t SlotActive, uint32_t AudioFreq);
+static void SAIx_In_DeInit(SAI_HandleTypeDef *hsai);
+
+/**
+ * @}
+ */
+
+/** @defgroup STM32H747I_DISCOVERY_AUDIO_OUT_Exported_Functions OUT Exported Functions
+ * @{
+ */
+
+/**
+ * @brief Configures the audio Out peripheral.
+ * @param OutputDevice: OUTPUT_DEVICE_SPEAKER, OUTPUT_DEVICE_HEADPHONE,
+ * or OUTPUT_DEVICE_BOTH.
+ * @param Volume: Initial volume level (from 0 (Mute) to 100 (Max))
+ * @param AudioFreq: Audio frequency used to play the audio stream.
+ * @retval AUDIO_OK if correct communication, else wrong communication
+ */
+uint8_t BSP_AUDIO_OUT_Init(uint16_t OutputDevice, uint8_t Volume, uint32_t AudioFreq)
+{
+ uint8_t ret = AUDIO_ERROR;
+ uint32_t deviceid = 0x00;
+ uint32_t slot_active;
+
+ /* Initialize SAI1 sub_block A as MASTER TX */
+ haudio_out_sai.Instance = AUDIO_OUT_SAIx;
+
+ /* Disable SAI */
+ SAIx_Out_DeInit(&haudio_out_sai);
+
+ /* PLL clock is set depending by the AudioFreq (44.1khz vs 48khz groups) */
+ BSP_AUDIO_OUT_ClockConfig(&haudio_out_sai, AudioFreq, NULL);
+
+ /* SAI data transfer preparation:
+ Prepare the Media to be used for the audio transfer from memory to SAI peripheral */
+
+ if(HAL_SAI_GetState(&haudio_out_sai) == HAL_SAI_STATE_RESET)
+ {
+ /* Init the SAI MSP: this __weak function can be redefined by the application*/
+ BSP_AUDIO_OUT_MspInit(&haudio_out_sai, NULL);
+ }
+
+ /* Init SAI as master RX output */
+ slot_active = CODEC_AUDIOFRAME_SLOT_0123;
+ SAIx_Out_Init(SAI_MODEMASTER_TX, slot_active, AudioFreq);
+
+ /* wm8994 codec initialization */
+ deviceid = wm8994_drv.ReadID(AUDIO_I2C_ADDRESS);
+
+ if((deviceid) == WM8994_ID)
+ {
+ /* Reset the Codec Registers */
+ wm8994_drv.Reset(AUDIO_I2C_ADDRESS);
+ /* Initialize the audio driver structure */
+ audio_drv = &wm8994_drv;
+ ret = AUDIO_OK;
+ }
+ else
+ {
+ ret = AUDIO_ERROR;
+ }
+
+ if(ret == AUDIO_OK)
+ {
+ /* Initialize the codec internal registers */
+ audio_drv->Init(AUDIO_I2C_ADDRESS, OutputDevice, Volume, AudioFreq);
+ }
+
+ return ret;
+}
+
+/**
+ * @brief Starts playing audio stream from a data buffer for a determined size.
+ * @param pBuffer: Pointer to the buffer
+ * @param Size: Number of audio data BYTES.
+ * @retval AUDIO_OK if correct communication, else wrong communication
+ */
+uint8_t BSP_AUDIO_OUT_Play(uint16_t* pBuffer, uint32_t Size)
+{
+ /* Call the audio Codec Play function */
+ if(audio_drv->Play(AUDIO_I2C_ADDRESS, pBuffer, Size) != 0)
+ {
+ return AUDIO_ERROR;
+ }
+ else
+ {
+ /* Update the Media layer and enable it for play */
+ HAL_SAI_Transmit_DMA(&haudio_out_sai, (uint8_t*) pBuffer, DMA_MAX(Size / AUDIODATA_SIZE));
+
+ return AUDIO_OK;
+ }
+}
+
+/**
+ * @brief Sends n-Bytes on the SAI interface.
+ * @param pData: pointer on data address
+ * @param Size: number of data to be written
+ * @retval None
+ */
+void BSP_AUDIO_OUT_ChangeBuffer(uint16_t *pData, uint16_t Size)
+{
+ HAL_SAI_Transmit_DMA(&haudio_out_sai, (uint8_t*) pData, Size);
+}
+
+/**
+ * @brief This function Pauses the audio file stream. In case
+ * of using DMA, the DMA Pause feature is used.
+ * @warning When calling BSP_AUDIO_OUT_Pause() function for pause, only
+ * BSP_AUDIO_OUT_Resume() function should be called for resume (use of BSP_AUDIO_OUT_Play()
+ * function for resume could lead to unexpected behaviour).
+ * @retval AUDIO_OK if correct communication, else wrong communication
+ */
+uint8_t BSP_AUDIO_OUT_Pause(void)
+{
+ /* Call the Audio Codec Pause/Resume function */
+ if(audio_drv->Pause(AUDIO_I2C_ADDRESS) != 0)
+ {
+ return AUDIO_ERROR;
+ }
+ else
+ {
+ /* Call the Media layer pause function */
+ HAL_SAI_DMAPause(&haudio_out_sai);
+
+ /* Return AUDIO_OK when all operations are correctly done */
+ return AUDIO_OK;
+ }
+}
+
+/**
+ * @brief Resumes the audio file stream.
+ * @warning When calling BSP_AUDIO_OUT_Pause() function for pause, only
+ * BSP_AUDIO_OUT_Resume() function should be called for resume (use of BSP_AUDIO_OUT_Play()
+ * function for resume could lead to unexpected behaviour).
+ * @retval AUDIO_OK if correct communication, else wrong communication
+ */
+uint8_t BSP_AUDIO_OUT_Resume(void)
+{
+ /* Call the Audio Codec Pause/Resume function */
+ if(audio_drv->Resume(AUDIO_I2C_ADDRESS) != 0)
+ {
+ return AUDIO_ERROR;
+ }
+ else
+ {
+ /* Call the Media layer pause/resume function */
+ HAL_SAI_DMAResume(&haudio_out_sai);
+
+ /* Return AUDIO_OK when all operations are correctly done */
+ return AUDIO_OK;
+ }
+}
+
+/**
+ * @brief Stops audio playing and Power down the Audio Codec.
+ * @param Option: could be one of the following parameters
+ * - CODEC_PDWN_SW: for software power off (by writing registers).
+ * Then no need to reconfigure the Codec after power on.
+ * - CODEC_PDWN_HW: completely shut down the codec (physically).
+ * Then need to reconfigure the Codec after power on.
+ * @retval AUDIO_OK if correct communication, else wrong communication
+ */
+uint8_t BSP_AUDIO_OUT_Stop(uint32_t Option)
+{
+ /* Call the Media layer stop function */
+ HAL_SAI_DMAStop(&haudio_out_sai);
+
+ /* Call Audio Codec Stop function */
+ if(audio_drv->Stop(AUDIO_I2C_ADDRESS, Option) != 0)
+ {
+ return AUDIO_ERROR;
+ }
+ else
+ {
+ if(Option == CODEC_PDWN_HW)
+ {
+ /* Wait at least 100us */
+ HAL_Delay(1);
+ }
+ /* Return AUDIO_OK when all operations are correctly done */
+ return AUDIO_OK;
+ }
+}
+
+/**
+ * @brief Controls the current audio volume level.
+ * @param Volume: Volume level to be set in percentage from 0% to 100% (0 for
+ * Mute and 100 for Max volume level).
+ * @retval AUDIO_OK if correct communication, else wrong communication
+ */
+uint8_t BSP_AUDIO_OUT_SetVolume(uint8_t Volume)
+{
+ /* Call the codec volume control function with converted volume value */
+ if(audio_drv->SetVolume(AUDIO_I2C_ADDRESS, Volume) != 0)
+ {
+ return AUDIO_ERROR;
+ }
+ else
+ {
+ /* Return AUDIO_OK when all operations are correctly done */
+ return AUDIO_OK;
+ }
+}
+
+/**
+ * @brief Enables or disables the MUTE mode by software
+ * @param Cmd: Could be AUDIO_MUTE_ON to mute sound or AUDIO_MUTE_OFF to
+ * unmute the codec and restore previous volume level.
+ * @retval AUDIO_OK if correct communication, else wrong communication
+ */
+uint8_t BSP_AUDIO_OUT_SetMute(uint32_t Cmd)
+{
+ /* Call the Codec Mute function */
+ if(audio_drv->SetMute(AUDIO_I2C_ADDRESS, Cmd) != 0)
+ {
+ return AUDIO_ERROR;
+ }
+ else
+ {
+ /* Return AUDIO_OK when all operations are correctly done */
+ return AUDIO_OK;
+ }
+}
+
+/**
+ * @brief Switch dynamically (while audio file is played) the output target
+ * (speaker or headphone).
+ * @param Output: The audio output target: OUTPUT_DEVICE_SPEAKER,
+ * OUTPUT_DEVICE_HEADPHONE or OUTPUT_DEVICE_BOTH
+ * @retval AUDIO_OK if correct communication, else wrong communication
+ */
+uint8_t BSP_AUDIO_OUT_SetOutputMode(uint8_t Output)
+{
+ /* Call the Codec output device function */
+ if(audio_drv->SetOutputMode(AUDIO_I2C_ADDRESS, Output) != 0)
+ {
+ return AUDIO_ERROR;
+ }
+ else
+ {
+ /* Return AUDIO_OK when all operations are correctly done */
+ return AUDIO_OK;
+ }
+}
+
+/**
+ * @brief Updates the audio frequency.
+ * @param AudioFreq: Audio frequency used to play the audio stream.
+ * @note This API should be called after the BSP_AUDIO_OUT_Init() to adjust the
+ * audio frequency.
+ * @retval None
+ */
+void BSP_AUDIO_OUT_SetFrequency(uint32_t AudioFreq)
+{
+ /* PLL clock is set depending by the AudioFreq (44.1khz vs 48khz groups) */
+ BSP_AUDIO_OUT_ClockConfig(&haudio_out_sai, AudioFreq, NULL);
+
+ /* Disable SAI peripheral to allow access to SAI internal registers */
+ __HAL_SAI_DISABLE(&haudio_out_sai);
+
+ /* Update the SAI audio frequency configuration */
+ haudio_out_sai.Init.AudioFrequency = AudioFreq;
+ HAL_SAI_Init(&haudio_out_sai);
+
+ /* Enable SAI peripheral to generate MCLK */
+ __HAL_SAI_ENABLE(&haudio_out_sai);
+}
+
+/**
+ * @brief Updates the Audio frame slot configuration.
+ * @param AudioFrameSlot: specifies the audio Frame slot
+ * @note This API should be called after the BSP_AUDIO_OUT_Init() to adjust the
+ * audio frame slot.
+ * @retval None
+ */
+void BSP_AUDIO_OUT_SetAudioFrameSlot(uint32_t AudioFrameSlot)
+{
+ /* Disable SAI peripheral to allow access to SAI internal registers */
+ __HAL_SAI_DISABLE(&haudio_out_sai);
+
+ /* Update the SAI audio frame slot configuration */
+ haudio_out_sai.SlotInit.SlotActive = AudioFrameSlot;
+ HAL_SAI_Init(&haudio_out_sai);
+
+ /* Enable SAI peripheral to generate MCLK */
+ __HAL_SAI_ENABLE(&haudio_out_sai);
+}
+
+/**
+ * @brief De-initializes the audio out peripheral.
+ * @retval None
+ */
+void BSP_AUDIO_OUT_DeInit(void)
+{
+ SAIx_Out_DeInit(&haudio_out_sai);
+ /* DeInit the SAI MSP : this __weak function can be rewritten by the application */
+ BSP_AUDIO_OUT_MspDeInit(&haudio_out_sai, NULL);
+}
+
+/**
+ * @brief Manages the DMA full Transfer complete event.
+ * @retval None
+ */
+__weak void BSP_AUDIO_OUT_TransferComplete_CallBack(void)
+{
+}
+
+/**
+ * @brief Manages the DMA Half Transfer complete event.
+ * @retval None
+ */
+__weak void BSP_AUDIO_OUT_HalfTransfer_CallBack(void)
+{
+}
+
+/**
+ * @brief Manages the DMA FIFO error event.
+ * @retval None
+ */
+__weak void BSP_AUDIO_OUT_Error_CallBack(void)
+{
+}
+
+/**
+ * @brief Initializes BSP_AUDIO_OUT MSP.
+ * @param hsai: SAI handle
+ * @param Params: pointer on additional configuration parameters, can be NULL.
+ * @retval None
+ */
+__weak void BSP_AUDIO_OUT_MspInit(SAI_HandleTypeDef *hsai, void *Params)
+{
+ static DMA_HandleTypeDef hdma_sai_tx;
+ GPIO_InitTypeDef gpio_init_structure;
+
+ /* Enable SAI clock */
+ AUDIO_OUT_SAIx_CLK_ENABLE();
+
+ /* CODEC_SAI pins configuration: FS, SCK and SD pins */
+ /* Enable FS, SCK and SD clocks */
+ AUDIO_OUT_SAIx_SD_FS_CLK_ENABLE();
+ /* Enable FS, SCK and SD pins */
+ gpio_init_structure.Pin = AUDIO_OUT_SAIx_FS_PIN | AUDIO_OUT_SAIx_SCK_PIN | AUDIO_OUT_SAIx_SD_PIN;
+ gpio_init_structure.Mode = GPIO_MODE_AF_PP;
+ gpio_init_structure.Pull = GPIO_NOPULL;
+ gpio_init_structure.Speed = GPIO_SPEED_FREQ_VERY_HIGH;
+ gpio_init_structure.Alternate = AUDIO_OUT_SAIx_AF;
+ HAL_GPIO_Init(AUDIO_OUT_SAIx_SD_FS_SCK_GPIO_PORT, &gpio_init_structure);
+
+ /* Enable MCLK clock */
+ AUDIO_OUT_SAIx_MCLK_ENABLE();
+ /* Enable MCLK pin */
+ gpio_init_structure.Pin = AUDIO_OUT_SAIx_MCLK_PIN;
+ HAL_GPIO_Init(AUDIO_OUT_SAIx_MCLK_GPIO_PORT, &gpio_init_structure);
+
+ /* Enable the DMA clock */
+ AUDIO_OUT_SAIx_DMAx_CLK_ENABLE();
+
+ if(hsai->Instance == AUDIO_OUT_SAIx)
+ {
+ /* Configure the hdma_saiTx handle parameters */
+ hdma_sai_tx.Init.Request = AUDIO_OUT_SAIx_DMAx_REQUEST;
+ hdma_sai_tx.Init.Direction = DMA_MEMORY_TO_PERIPH;
+ hdma_sai_tx.Init.PeriphInc = DMA_PINC_DISABLE;
+ hdma_sai_tx.Init.MemInc = DMA_MINC_ENABLE;
+ hdma_sai_tx.Init.PeriphDataAlignment = AUDIO_OUT_SAIx_DMAx_PERIPH_DATA_SIZE;
+ hdma_sai_tx.Init.MemDataAlignment = AUDIO_OUT_SAIx_DMAx_MEM_DATA_SIZE;
+ hdma_sai_tx.Init.Mode = DMA_CIRCULAR;
+ hdma_sai_tx.Init.Priority = DMA_PRIORITY_HIGH;
+ hdma_sai_tx.Init.FIFOMode = DMA_FIFOMODE_ENABLE;
+ hdma_sai_tx.Init.FIFOThreshold = DMA_FIFO_THRESHOLD_FULL;
+ hdma_sai_tx.Init.MemBurst = DMA_MBURST_SINGLE;
+ hdma_sai_tx.Init.PeriphBurst = DMA_PBURST_SINGLE;
+
+ hdma_sai_tx.Instance = AUDIO_OUT_SAIx_DMAx_STREAM;
+
+ /* Associate the DMA handle */
+ __HAL_LINKDMA(hsai, hdmatx, hdma_sai_tx);
+
+ /* Deinitialize the Stream for new transfer */
+ HAL_DMA_DeInit(&hdma_sai_tx);
+
+ /* Configure the DMA Stream */
+ HAL_DMA_Init(&hdma_sai_tx);
+ }
+
+ /* SAI DMA IRQ Channel configuration */
+ HAL_NVIC_SetPriority(AUDIO_OUT_SAIx_DMAx_IRQ, AUDIO_OUT_IRQ_PREPRIO, 0);
+ HAL_NVIC_EnableIRQ(AUDIO_OUT_SAIx_DMAx_IRQ);
+}
+
+/**
+ * @brief Deinitializes SAI MSP.
+ * @param hsai: SAI handle
+ * @param Params: pointer on additional configuration parameters, can be NULL.
+ * @retval None
+ */
+__weak void BSP_AUDIO_OUT_MspDeInit(SAI_HandleTypeDef *hsai, void *Params)
+{
+ GPIO_InitTypeDef gpio_init_structure;
+
+ /* SAI DMA IRQ Channel deactivation */
+ HAL_NVIC_DisableIRQ(AUDIO_OUT_SAIx_DMAx_IRQ);
+
+ if(hsai->Instance == AUDIO_OUT_SAIx)
+ {
+ /* Deinitialize the DMA stream */
+ HAL_DMA_DeInit(hsai->hdmatx);
+ }
+
+ /* Disable SAI peripheral */
+ __HAL_SAI_DISABLE(hsai);
+
+ /* Deactivates CODEC_SAI pins FS, SCK, MCK and SD by putting them in input mode */
+ gpio_init_structure.Pin = AUDIO_OUT_SAIx_FS_PIN | AUDIO_OUT_SAIx_SCK_PIN | AUDIO_OUT_SAIx_SD_PIN;
+ HAL_GPIO_DeInit(AUDIO_OUT_SAIx_SD_FS_SCK_GPIO_PORT, gpio_init_structure.Pin);
+
+ gpio_init_structure.Pin = AUDIO_OUT_SAIx_MCLK_PIN;
+ HAL_GPIO_DeInit(AUDIO_OUT_SAIx_MCLK_GPIO_PORT, gpio_init_structure.Pin);
+
+ /* Disable SAI clock */
+ AUDIO_OUT_SAIx_CLK_DISABLE();
+
+ /* GPIO pins clock and DMA clock can be shut down in the applic
+ by surcharging this __weak function */
+}
+
+/**
+ * @brief Clock Config.
+ * @param hsai: might be required to set audio peripheral predivider if any.
+ * @param AudioFreq: Audio frequency used to play the audio stream.
+ * @param Params: pointer on additional configuration parameters, can be NULL.
+ * @note This API is called by BSP_AUDIO_OUT_Init() and BSP_AUDIO_OUT_SetFrequency()
+ * Being __weak it can be overwritten by the application
+ * @retval None
+ */
+__weak void BSP_AUDIO_OUT_ClockConfig(SAI_HandleTypeDef *hsai, uint32_t AudioFreq, void *Params)
+{
+ RCC_PeriphCLKInitTypeDef rcc_ex_clk_init_struct;
+
+ HAL_RCCEx_GetPeriphCLKConfig(&rcc_ex_clk_init_struct);
+
+ /* Set the PLL configuration according to the audio frequency */
+ if((AudioFreq == AUDIO_FREQUENCY_11K) || (AudioFreq == AUDIO_FREQUENCY_22K) || (AudioFreq == AUDIO_FREQUENCY_44K))
+ {
+ /* SAI clock config:
+ PLL2_VCO Input = HSE_VALUE/PLL2M = 1 Mhz
+ PLL2_VCO Output = PLL2_VCO Input * PLL2N = 429 Mhz
+ SAI_CLK_x = PLL2_VCO Output/PLL2P = 429/38 = 11.289 Mhz */
+ rcc_ex_clk_init_struct.PeriphClockSelection = RCC_PERIPHCLK_SAI1;
+ rcc_ex_clk_init_struct.Sai1ClockSelection = RCC_SAI1CLKSOURCE_PLL2;
+ rcc_ex_clk_init_struct.PLL2.PLL2P = 38;
+ rcc_ex_clk_init_struct.PLL2.PLL2Q = 1;
+ rcc_ex_clk_init_struct.PLL2.PLL2R = 1;
+ rcc_ex_clk_init_struct.PLL2.PLL2N = 429;
+ rcc_ex_clk_init_struct.PLL2.PLL2M = 25;
+ HAL_RCCEx_PeriphCLKConfig(&rcc_ex_clk_init_struct);
+ }
+ else /* AUDIO_FREQUENCY_8K, AUDIO_FREQUENCY_16K, AUDIO_FREQUENCY_48K, AUDIO_FREQUENCY_96K */
+ {
+ /* SAI clock config:
+ PLL2_VCO Input = HSE_VALUE/PLL2M = 1 Mhz
+ PLL2_VCO Output = PLL2_VCO Input * PLL2N = 344 Mhz
+ SAI_CLK_x = PLL2_VCO Output/PLL2P = 344/7 = 49.142 Mhz */
+ rcc_ex_clk_init_struct.PeriphClockSelection = RCC_PERIPHCLK_SAI1;
+ rcc_ex_clk_init_struct.Sai1ClockSelection = RCC_SAI1CLKSOURCE_PLL2;
+ rcc_ex_clk_init_struct.PLL2.PLL2P = 7;
+ rcc_ex_clk_init_struct.PLL2.PLL2Q = 1;
+ rcc_ex_clk_init_struct.PLL2.PLL2R = 1;
+ rcc_ex_clk_init_struct.PLL2.PLL2N = 344;
+ rcc_ex_clk_init_struct.PLL2.PLL2M = 25;
+ HAL_RCCEx_PeriphCLKConfig(&rcc_ex_clk_init_struct);
+ }
+}
+/**
+ * @}
+ */
+
+/** @defgroup STM32H747I_DISCOVERY_AUDIO_OUT_Private_Functions OUT Private Functions
+ * @{
+ */
+
+/*******************************************************************************
+ HAL Callbacks
+*******************************************************************************/
+/**
+ * @brief Tx Transfer completed callbacks.
+ * @param hsai: SAI handle
+ * @retval None
+ */
+void HAL_SAI_TxCpltCallback(SAI_HandleTypeDef *hsai)
+{
+ /* Manage the remaining file size and new address offset: This function
+ should be coded by user (its prototype is already declared in stm32h747i_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 stm32h747i_discovery_audio.h) */
+ BSP_AUDIO_OUT_HalfTransfer_CallBack();
+}
+
+/**
+ * @brief SAI error callbacks.
+ * @param hsai: SAI handle
+ * @retval None
+ */
+void HAL_SAI_ErrorCallback(SAI_HandleTypeDef *hsai)
+{
+ if(hsai->Instance == AUDIO_OUT_SAIx)
+ {
+ BSP_AUDIO_OUT_Error_CallBack();
+ }
+ else
+ {
+ BSP_AUDIO_IN_Error_CallBack();
+ }
+}
+
+/*******************************************************************************
+ Static Functions
+*******************************************************************************/
+
+/**
+ * @brief Initializes the Audio Codec audio interface (SAI).
+ * @param SaiOutMode: Audio mode to be configured for the SAI peripheral.
+ * @param SlotActive: Audio active slot to be configured for the SAI peripheral.
+ * @param AudioFreq: Audio frequency to be configured for the SAI peripheral.
+ * @note The default SlotActive configuration is set to CODEC_AUDIOFRAME_SLOT_0123
+ * and user can update this configuration using
+ * @retval None
+ */
+static void SAIx_Out_Init(uint32_t SaiOutMode, uint32_t SlotActive, uint32_t AudioFreq)
+{
+ /* Disable SAI peripheral to allow access to SAI internal registers */
+ __HAL_SAI_DISABLE(&haudio_out_sai);
+
+ /* Configure SAI_Block_x
+ LSBFirst: Disabled
+ DataSize: 16 */
+ haudio_out_sai.Init.MonoStereoMode = SAI_STEREOMODE;
+ haudio_out_sai.Init.AudioFrequency = AudioFreq;
+ haudio_out_sai.Init.AudioMode = SaiOutMode;
+ haudio_out_sai.Init.NoDivider = SAI_MASTERDIVIDER_ENABLE;
+ haudio_out_sai.Init.Protocol = SAI_FREE_PROTOCOL;
+ haudio_out_sai.Init.DataSize = SAI_DATASIZE_16;
+ haudio_out_sai.Init.FirstBit = SAI_FIRSTBIT_MSB;
+ haudio_out_sai.Init.ClockStrobing = SAI_CLOCKSTROBING_RISINGEDGE;
+ haudio_out_sai.Init.Synchro = SAI_ASYNCHRONOUS;
+ haudio_out_sai.Init.OutputDrive = SAI_OUTPUTDRIVE_ENABLE;
+ haudio_out_sai.Init.FIFOThreshold = SAI_FIFOTHRESHOLD_1QF;
+ haudio_out_sai.Init.SynchroExt = SAI_SYNCEXT_DISABLE;
+ haudio_out_sai.Init.CompandingMode = SAI_NOCOMPANDING;
+ haudio_out_sai.Init.TriState = SAI_OUTPUT_NOTRELEASED;
+ haudio_out_sai.Init.Mckdiv = 0;
+ haudio_out_sai.Init.MckOverSampling = SAI_MCK_OVERSAMPLING_DISABLE;
+ haudio_out_sai.Init.PdmInit.Activation = DISABLE;
+ haudio_out_sai.Init.PdmInit.ClockEnable = 0;
+ haudio_out_sai.Init.PdmInit.MicPairsNbr = 0;
+
+ /* Configure SAI_Block_x Frame
+ Frame Length: 64
+ Frame active Length: 32
+ FS Definition: Start frame + Channel Side identification
+ FS Polarity: FS active Low
+ FS Offset: FS asserted one bit before the first bit of slot 0 */
+ haudio_out_sai.FrameInit.FrameLength = 128;
+ haudio_out_sai.FrameInit.ActiveFrameLength = 64;
+ haudio_out_sai.FrameInit.FSDefinition = SAI_FS_CHANNEL_IDENTIFICATION;
+ haudio_out_sai.FrameInit.FSPolarity = SAI_FS_ACTIVE_LOW;
+ haudio_out_sai.FrameInit.FSOffset = SAI_FS_BEFOREFIRSTBIT;
+
+ /* Configure SAI Block_x Slot
+ Slot First Bit Offset: 0
+ Slot Size : 16
+ Slot Number: 4
+ Slot Active: All slot actives */
+ haudio_out_sai.SlotInit.FirstBitOffset = 0;
+ haudio_out_sai.SlotInit.SlotSize = SAI_SLOTSIZE_DATASIZE;
+ haudio_out_sai.SlotInit.SlotNumber = 4;
+ haudio_out_sai.SlotInit.SlotActive = SlotActive;
+ HAL_SAI_Init(&haudio_out_sai);
+
+ /* Enable SAI peripheral to generate MCLK */
+ __HAL_SAI_ENABLE(&haudio_out_sai);
+}
+
+/**
+ * @brief Deinitializes the Audio Codec audio interface (SAI).
+ * @retval None
+ */
+static void SAIx_Out_DeInit(SAI_HandleTypeDef *hsai)
+{
+ /* Disable SAI peripheral */
+ __HAL_SAI_DISABLE(hsai);
+
+ HAL_SAI_DeInit(hsai);
+}
+
+/**
+ * @}
+ */
+
+/** @defgroup STM32H747I_DISCOVERY_AUDIO_IN_Exported_Functions IN Exported Functions
+ * @{
+ */
+
+/**
+ * @brief Initialize wave recording.
+ * @param AudioFreq: Audio frequency to be configured for the DFSDM peripheral.
+ * @param BitRes: Audio frequency to be configured for the DFSDM peripheral.
+ * @param ChnlNbr: Audio frequency to be configured for the DFSDM peripheral.
+ * @retval AUDIO_OK if correct communication, else wrong communication
+ */
+uint8_t BSP_AUDIO_IN_Init(uint32_t AudioFreq, uint32_t BitRes, uint32_t ChnlNbr)
+{
+ /* Set audio in interface to default one */
+ BSP_AUDIO_IN_SelectInterface(AUDIO_IN_INTERFACE_PDM);
+ return BSP_AUDIO_IN_InitEx(INPUT_DEVICE_DIGITAL_MIC, AudioFreq, BitRes, ChnlNbr);
+}
+
+/**
+ * @brief Initialize wave recording.
+ * @param InputDevice: INPUT_DEVICE_DIGITAL_MIC or INPUT_DEVICE_ANALOG_MIC.
+ * @param AudioFreq: Audio frequency to be configured.
+ * @param BitRes: Audio bit resolution to be configured..
+ * @param ChnlNbr: Number of channel to be configured.
+ * @retval AUDIO_OK if correct communication, else wrong communication
+ */
+uint8_t BSP_AUDIO_IN_InitEx(uint16_t InputDevice, uint32_t AudioFreq, uint32_t BitRes, uint32_t ChnlNbr)
+{
+ uint8_t ret = AUDIO_OK;
+ uint32_t slot_active;
+
+ /* Store the audio record context */
+ hAudioIn.Frequency = AudioFreq;
+ hAudioIn.BitResolution = BitRes;
+ hAudioIn.InputDevice = InputDevice;
+ hAudioIn.ChannelNbr = ChnlNbr;
+
+ if(hAudioIn.InputDevice == INPUT_DEVICE_DIGITAL_MIC)
+ {
+ if(hAudioIn.Interface == AUDIO_IN_INTERFACE_SAI)
+ {
+ /* Initialize SAI1 block B as SLAVE RX synchrounous with SAI1 block A */
+ haudio_in_sai.Instance = AUDIO_IN_SAIx;
+
+ /* Disable SAI */
+ SAIx_In_DeInit(&haudio_in_sai);
+
+ /* PLL clock is set depending on the AudioFreq (44.1khz vs 48khz groups) */
+ BSP_AUDIO_IN_ClockConfig(AudioFreq, NULL); /* Clock config is shared between AUDIO IN and OUT */
+
+ /* SAI data transfer preparation:
+ Prepare the Media to be used for the audio transfer from SAI peripheral to memory */
+ if(HAL_SAI_GetState(&haudio_in_sai) == HAL_SAI_STATE_RESET)
+ {
+ /* Init the SAI MSP: this __weak function can be redefined by the application*/
+ BSP_AUDIO_IN_MspInit();
+ }
+
+ /* Configure SAI in master mode :
+ * - SAI1_block_B in slave RX mode synchronous from SAI1_block_A
+ */
+ slot_active = CODEC_AUDIOFRAME_SLOT_13;
+ SAIx_In_Init(SAI_MODESLAVE_RX, slot_active, AudioFreq);
+ }
+ else if(hAudioIn.Interface == AUDIO_IN_INTERFACE_PDM)
+ {
+ /* Initialize SAI1 block A as MASTER RX */
+ haudio_in_sai.Instance = AUDIO_IN_SAI_PDMx;
+
+ /* Disable SAI */
+ SAIx_In_DeInit(&haudio_in_sai);
+
+ /* PLL clock is set depending on the AudioFreq (44.1khz vs 48khz groups) */
+ BSP_AUDIO_IN_ClockConfig(AudioFreq, NULL);
+
+ /* SAI data transfer preparation:
+ Prepare the Media to be used for the audio transfer from SAI peripheral to memory */
+ /* Initialize the haudio_in_sai Instance parameter */
+
+ if(HAL_SAI_GetState(&haudio_in_sai) == HAL_SAI_STATE_RESET)
+ {
+ /* Init the SAI MSP: this __weak function can be redefined by the application*/
+ BSP_AUDIO_IN_MspInit();
+ }
+
+ /* Configure SAI in master mode :
+ * - SAI1_block_A in master RX mode
+ */
+ slot_active = CODEC_AUDIOFRAME_SLOT_0;
+ SAIx_In_Init(SAI_MODEMASTER_RX, slot_active, AudioFreq);
+
+ if(BSP_AUDIO_IN_PDMToPCM_Init(AudioFreq, hAudioIn.ChannelNbr, hAudioIn.ChannelNbr) != AUDIO_OK)
+ {
+ ret = AUDIO_ERROR;
+ }
+ }
+ else
+ {
+ ret = AUDIO_ERROR;
+ }
+ }
+ else
+ {
+ /* Analog Input */
+ ret = AUDIO_ERROR;
+ }
+
+ /* Return AUDIO_OK when all operations are correctly done */
+ return ret;
+}
+
+
+/**
+ * @brief Initializes wave recording and playback in parallel.
+ * @param InputDevice: INPUT_DEVICE_DIGITAL_MICROPHONE_2
+ * @param OutputDevice: OUTPUT_DEVICE_SPEAKER, OUTPUT_DEVICE_HEADPHONE,
+ * or OUTPUT_DEVICE_BOTH.
+ * @param AudioFreq: Audio frequency to be configured for the SAI peripheral.
+ * @param BitRes: Audio frequency to be configured.
+ * @param ChnlNbr: Channel number.
+ * @retval AUDIO_OK if correct communication, else wrong communication
+ */
+uint8_t BSP_AUDIO_IN_OUT_Init(uint32_t InputDevice, uint32_t OutputDevice, uint32_t AudioFreq, uint32_t BitRes, uint32_t ChnlNbr)
+{
+ uint32_t slot_active;
+ uint32_t deviceid = 0, ret = AUDIO_OK;
+
+ /* Store the audio record context */
+ hAudioIn.Frequency = AudioFreq;
+ hAudioIn.BitResolution = BitRes;
+ hAudioIn.InputDevice = InputDevice;
+ hAudioIn.ChannelNbr = ChnlNbr;
+
+ /* Input device is Digital MIC2 and Codec interface is SAI */
+ if (hAudioIn.InputDevice == INPUT_DEVICE_DIGITAL_MICROPHONE_2)
+ {
+ haudio_in_sai.Instance = AUDIO_IN_SAIx;
+ haudio_out_sai.Instance = AUDIO_OUT_SAIx;
+
+ /* PLL clock is set depending on the AudioFreq (44.1khz vs 48khz groups) */
+ BSP_AUDIO_OUT_ClockConfig(&haudio_in_sai, AudioFreq, NULL);
+ /* SAI data transfer preparation:
+ Prepare the Media to be used for the audio transfer from SAI peripheral to memory */
+ if(HAL_SAI_GetState(&haudio_in_sai) == HAL_SAI_STATE_RESET)
+ {
+ /* Init the SAI MSP: this __weak function can be redefined by the application*/
+ BSP_AUDIO_IN_MspInit();
+ }
+
+ /* SAI data transfer preparation:
+ Prepare the Media to be used for the audio transfer from memory to SAI peripheral */
+ if(HAL_SAI_GetState(&haudio_out_sai) == HAL_SAI_STATE_RESET)
+ {
+ /* Init the SAI MSP: this __weak function can be redefined by the application*/
+ BSP_AUDIO_OUT_MspInit(&haudio_out_sai, NULL);
+ }
+
+ /* Configure SAI in master TX mode :
+ * - SAI1_block_A in master TX mode
+ * - SAI1_block_B in slave RX mode synchronous from SAI1_block_A
+ */
+ slot_active = CODEC_AUDIOFRAME_SLOT_13;
+ SAIx_In_Init(SAI_MODESLAVE_RX, slot_active, AudioFreq);
+
+ slot_active = CODEC_AUDIOFRAME_SLOT_02;
+ SAIx_Out_Init(SAI_MODEMASTER_TX, slot_active, AudioFreq);
+
+ /* wm8994 codec initialization */
+ deviceid = wm8994_drv.ReadID(AUDIO_I2C_ADDRESS);
+
+ if((deviceid) == WM8994_ID)
+ {
+ /* Reset the Codec Registers */
+ wm8994_drv.Reset(AUDIO_I2C_ADDRESS);
+ /* Initialize the audio driver structure */
+ audio_drv = &wm8994_drv;
+ ret = AUDIO_OK;
+ }
+ else
+ {
+ ret = AUDIO_ERROR;
+ }
+
+ if(ret == AUDIO_OK)
+ {
+ /* Initialize the codec internal registers */
+ audio_drv->Init(AUDIO_I2C_ADDRESS, InputDevice|OutputDevice, 90, AudioFreq);
+ }
+ }
+ else
+ {
+ ret = AUDIO_ERROR;
+ }
+
+ /* Return AUDIO_OK when all operations are correctly done */
+ return ret;
+}
+
+/**
+ * @brief Link digital mic to specified source
+ * @param Interface : Audio In interface for Digital mic. It can be:
+ * AUDIO_IN_INTERFACE_SAI
+ * AUDIO_IN_INTERFACE_PDM
+ * @retval None
+ */
+void BSP_AUDIO_IN_SelectInterface(uint32_t Interface)
+{
+ hAudioIn.Interface = Interface;
+}
+
+/**
+ * @brief Get digital mic interface
+ * @retval Digital mic interface.
+ */
+uint32_t BSP_AUDIO_IN_GetInterface(void)
+{
+ return (hAudioIn.Interface);
+}
+
+/**
+ * @brief Return audio in channel number
+ * @retval Number of channel
+ */
+uint8_t BSP_AUDIO_IN_GetChannelNumber(void)
+{
+ return hAudioIn.ChannelNbr;
+}
+
+/**
+ * @brief Start audio recording.
+ * @param pBuf: Main buffer pointer for the recorded data storing
+ * @param size: Current size of the recorded buffer
+ * @retval AUDIO_OK if correct communication, else wrong communication
+ */
+uint8_t BSP_AUDIO_IN_Record(uint16_t *pBuf, uint32_t size)
+{
+ /* Start the process receive DMA */
+ if(HAL_OK != HAL_SAI_Receive_DMA(&haudio_in_sai, (uint8_t*)pBuf, size))
+ {
+ return AUDIO_ERROR;
+ }
+
+ /* Return AUDIO_OK when all operations are correctly done */
+ return AUDIO_OK;
+}
+
+/**
+ * @brief Stop audio recording.
+ * @retval AUDIO_OK if correct communication, else wrong communication
+ */
+uint8_t BSP_AUDIO_IN_Stop(void)
+{
+ /* Call the Media layer stop function */
+ HAL_SAI_DMAStop(&haudio_in_sai);
+
+ /* Return AUDIO_OK when all operations are correctly done */
+ return AUDIO_OK;
+}
+
+/**
+ * @brief Pause the audio file stream.
+ * @retval AUDIO_OK if correct communication, else wrong communication
+ */
+uint8_t BSP_AUDIO_IN_Pause(void)
+{
+ if (hAudioIn.InputDevice == INPUT_DEVICE_ANALOG_MIC)
+ {
+ return AUDIO_ERROR;
+ }
+ else
+ {
+ /* Call the Media layer pause function */
+ HAL_SAI_DMAPause(&haudio_in_sai);
+ }
+
+ /* Return AUDIO_OK when all operations are correctly done */
+ return AUDIO_OK;
+}
+
+/**
+ * @brief Resume the audio file stream.
+ * @retval AUDIO_OK if correct communication, else wrong communication
+ */
+uint8_t BSP_AUDIO_IN_Resume(void)
+{
+ if (hAudioIn.InputDevice == INPUT_DEVICE_ANALOG_MIC)
+ {
+ return AUDIO_ERROR;
+ }
+ else
+ {
+ /* Call the Media layer resume function */
+ HAL_SAI_DMAResume(&haudio_in_sai);
+ }
+
+ /* Return AUDIO_OK when all operations are correctly done */
+ return AUDIO_OK;
+}
+
+/**
+ * @brief Controls the audio in volume level.
+ * @param Volume: Volume level to be set in percentage from 0% to 100% (0 for
+ * Mute and 100 for Max volume level).
+ * @retval AUDIO_OK if correct communication, else wrong communication
+ */
+uint8_t BSP_AUDIO_IN_SetVolume(uint8_t Volume)
+{
+ /* Set the Global variable AudioInVolume */
+ AudioInVolume = Volume;
+
+ /* Return AUDIO_OK when all operations are correctly done */
+ return AUDIO_OK;
+}
+
+/**
+ * @brief Deinit the audio IN peripherals.
+ * @retval None
+ */
+void BSP_AUDIO_IN_DeInit(void)
+{
+ SAIx_In_DeInit(&haudio_in_sai);
+
+ BSP_AUDIO_IN_MspDeInit();
+}
+
+/**
+* @brief Initialize the PDM library.
+* @param AudioFreq: Audio sampling frequency
+* @param ChnlNbrIn: Number of input audio channels in the PDM buffer
+* @param ChnlNbrOut: Number of desired output audio channels in the resulting PCM buffer
+* @retval None
+*/
+uint8_t BSP_AUDIO_IN_PDMToPCM_Init(uint32_t AudioFreq, uint32_t ChnlNbrIn, uint32_t ChnlNbrOut)
+{
+ uint32_t index = 0;
+
+ /* Enable CRC peripheral to unlock the PDM library */
+ __HAL_RCC_CRC_CLK_ENABLE();
+
+ for(index = 0; index < ChnlNbrIn; index++)
+ {
+ /* Init PDM filters */
+ PDM_FilterHandler[index].bit_order = PDM_FILTER_BIT_ORDER_MSB;
+ PDM_FilterHandler[index].endianness = PDM_FILTER_ENDIANNESS_LE;
+ PDM_FilterHandler[index].high_pass_tap = 2122358088;
+ PDM_FilterHandler[index].out_ptr_channels = ChnlNbrOut;
+ PDM_FilterHandler[index].in_ptr_channels = ChnlNbrIn;
+ PDM_Filter_Init((PDM_Filter_Handler_t *)(&PDM_FilterHandler[index]));
+
+ /* PDM lib config phase */
+ PDM_FilterConfig[index].output_samples_number = AudioFreq/1000;
+ PDM_FilterConfig[index].mic_gain = 24;
+ PDM_FilterConfig[index].decimation_factor = PDM_FILTER_DEC_FACTOR_64;
+ PDM_Filter_setConfig((PDM_Filter_Handler_t *)&PDM_FilterHandler[index], &PDM_FilterConfig[index]);
+ }
+
+ return AUDIO_OK;
+}
+
+
+/**
+* @brief Converts audio format from PDM to PCM.
+
+* @param PDMBuf: Pointer to PDM buffer data
+* @param PCMBuf: Pointer to PCM buffer data
+* @retval AUDIO_OK in case of success, AUDIO_ERROR otherwise
+*/
+uint8_t BSP_AUDIO_IN_PDMToPCM(uint16_t *PDMBuf, uint16_t *PCMBuf)
+{
+ uint32_t index = 0;
+
+ for(index = 0; index < hAudioIn.ChannelNbr; index++)
+ {
+ PDM_Filter(&((uint8_t*)(PDMBuf))[index], (uint16_t*)&(PCMBuf[index]), &PDM_FilterHandler[index]);
+ }
+
+ return AUDIO_OK;
+}
+
+/**
+ * @brief User callback when record buffer is filled.
+ * @retval None
+ */
+__weak void BSP_AUDIO_IN_TransferComplete_CallBack(void)
+{
+ /* This function should be implemented by the user application.
+ It is called into this driver when the current buffer is filled
+ to prepare the next buffer pointer and its size. */
+}
+
+/**
+ * @brief Manages the DMA Half Transfer complete event.
+ * @retval None
+ */
+__weak void BSP_AUDIO_IN_HalfTransfer_CallBack(void)
+{
+ /* This function should be implemented by the user application.
+ It is called into this driver when the current buffer is filled
+ to prepare the next buffer pointer and its size. */
+}
+
+/**
+ * @brief User callback when record buffer is filled.
+ * @param InputDevice: INPUT_DEVICE_DIGITAL_MIC1 or INPUT_DEVICE_DIGITAL_MIC2
+ */
+__weak void BSP_AUDIO_IN_TransferComplete_CallBackEx(uint32_t InputDevice)
+{
+ /* This function should be implemented by the user application.
+ It is called into this driver when the current buffer is filled
+ to prepare the next buffer pointer and its size. */
+}
+
+/**
+ * @brief User callback when record buffer is filled.
+ * @param InputDevice: INPUT_DEVICE_DIGITAL_MIC1 or INPUT_DEVICE_DIGITAL_MIC2
+ */
+__weak void BSP_AUDIO_IN_HalfTransfer_CallBackEx(uint32_t InputDevice)
+{
+ /* This function should be implemented by the user application.
+ It is called into this driver when the current buffer is filled
+ to prepare the next buffer pointer and its size. */
+}
+
+/**
+ * @brief Audio IN Error callback function.
+ * @retval None
+ */
+__weak void BSP_AUDIO_IN_Error_CallBack(void)
+{
+ /* This function is called when an Interrupt due to transfer error on or peripheral
+ error occurs. */
+}
+
+/**
+ * @brief Initialize BSP_AUDIO_IN MSP.
+ * @retval None
+ */
+__weak void BSP_AUDIO_IN_MspInit(void)
+{
+ SAIx_In_MspInit(&haudio_in_sai, NULL);
+}
+
+/**
+ * @brief DeInitialize BSP_AUDIO_IN MSP.
+ * @retval None
+ */
+__weak void BSP_AUDIO_IN_MspDeInit(void)
+{
+ SAIx_In_MspDeInit(&haudio_in_sai, NULL);
+}
+
+/**
+ * @brief Clock Config.
+ * @param AudioFreq: Audio frequency used to play the audio stream.
+ * @param Params: pointer on additional configuration parameters, can be NULL.
+ * @note This API is called by BSP_AUDIO_IN_Init()
+ * Being __weak it can be overwritten by the application
+ * @retval None
+ */
+__weak void BSP_AUDIO_IN_ClockConfig(uint32_t AudioFreq, void *Params)
+{
+ RCC_PeriphCLKInitTypeDef rcc_ex_clk_init_struct;
+
+ HAL_RCCEx_GetPeriphCLKConfig(&rcc_ex_clk_init_struct);
+
+ /* Set the PLL configuration according to the audio frequency */
+ if((AudioFreq == AUDIO_FREQUENCY_11K) || (AudioFreq == AUDIO_FREQUENCY_22K) || (AudioFreq == AUDIO_FREQUENCY_44K))
+ {
+ /* SAI clock config:
+ PLL2_VCO Input = HSE_VALUE/PLL2M = 1 Mhz
+ PLL2_VCO Output = PLL2_VCO Input * PLL2N = 429 Mhz
+ SAI_CLK_x = PLL2_VCO Output/PLL2P = 429/38 = 11.289 Mhz */
+ rcc_ex_clk_init_struct.PeriphClockSelection = RCC_PERIPHCLK_SAI1;
+ rcc_ex_clk_init_struct.Sai1ClockSelection = RCC_SAI1CLKSOURCE_PLL2;
+ rcc_ex_clk_init_struct.PLL2.PLL2P = 38;
+ rcc_ex_clk_init_struct.PLL2.PLL2Q = 1;
+ rcc_ex_clk_init_struct.PLL2.PLL2R = 1;
+ rcc_ex_clk_init_struct.PLL2.PLL2N = 429;
+ rcc_ex_clk_init_struct.PLL2.PLL2M = 25;
+ if (hAudioIn.Interface == AUDIO_IN_INTERFACE_PDM)
+ {
+ rcc_ex_clk_init_struct.PeriphClockSelection = RCC_PERIPHCLK_SAI4A;
+ rcc_ex_clk_init_struct.Sai4AClockSelection = RCC_SAI4ACLKSOURCE_PLL2;
+ }
+ HAL_RCCEx_PeriphCLKConfig(&rcc_ex_clk_init_struct);
+
+ }
+ else /* AUDIO_FREQUENCY_8K, AUDIO_FREQUENCY_16K, AUDIO_FREQUENCY_32K, AUDIO_FREQUENCY_48K, AUDIO_FREQUENCY_96K */
+ {
+ /* SAI clock config:
+ PLL2_VCO Input = HSE_VALUE/PLL2M = 1 Mhz
+ PLL2_VCO Output = PLL2_VCO Input * PLL2N = 344 Mhz
+ SAI_CLK_x = PLL2_VCO Output/PLL2P = 344/7 = 49.142 Mhz */
+ rcc_ex_clk_init_struct.PeriphClockSelection = RCC_PERIPHCLK_SAI1;
+ rcc_ex_clk_init_struct.Sai1ClockSelection = RCC_SAI1CLKSOURCE_PLL2;
+ rcc_ex_clk_init_struct.PLL2.PLL2P = 7;
+ rcc_ex_clk_init_struct.PLL2.PLL2Q = 1;
+ rcc_ex_clk_init_struct.PLL2.PLL2R = 1;
+ rcc_ex_clk_init_struct.PLL2.PLL2N = 344;
+ rcc_ex_clk_init_struct.PLL2.PLL2M = 25;
+ if (hAudioIn.Interface == AUDIO_IN_INTERFACE_PDM)
+ {
+ rcc_ex_clk_init_struct.PeriphClockSelection = RCC_PERIPHCLK_SAI4A;
+ rcc_ex_clk_init_struct.Sai4AClockSelection = RCC_SAI4ACLKSOURCE_PLL2;
+ }
+ HAL_RCCEx_PeriphCLKConfig(&rcc_ex_clk_init_struct);
+ }
+}
+/**
+ * @}
+ */
+
+
+/** @defgroup STM32H747I_DISCOVERY_AUDIO_IN_Private_Functions IN Private Functions
+ * @{
+ */
+
+/*******************************************************************************
+ HAL Callbacks
+*******************************************************************************/
+
+/**
+ * @brief Half reception complete callback.
+ * @param hsai: SAI handle.
+ * @retval None
+ */
+void HAL_SAI_RxHalfCpltCallback(SAI_HandleTypeDef *hsai)
+{
+ /* Manage the remaining file size and new address offset: This function should be coded by user */
+ BSP_AUDIO_IN_HalfTransfer_CallBack();
+}
+
+/**
+ * @brief Reception complete callback.
+ * @param hsai: SAI handle.
+ * @retval None
+ */
+void HAL_SAI_RxCpltCallback(SAI_HandleTypeDef *hsai)
+{
+ /* Call the record update function to get the next buffer to fill and its size (size is ignored) */
+ BSP_AUDIO_IN_TransferComplete_CallBack();
+}
+
+/*******************************************************************************
+ Static Functions
+*******************************************************************************/
+/**
+ * @brief Initializes SAI Audio IN MSP.
+ * @param hsai: SAI handle
+ * @param Params: pointer on additional configuration parameters, can be NULL.
+ * @retval None
+ */
+static void SAIx_In_MspInit(SAI_HandleTypeDef *hsai, void *Params)
+{
+ static DMA_HandleTypeDef hdma_sai_rx;
+ GPIO_InitTypeDef gpio_init_structure;
+
+ if(hsai->Instance == AUDIO_IN_SAI_PDMx)
+ {
+ /* Enable SAI clock */
+ AUDIO_IN_SAI_PDMx_CLK_ENABLE();
+
+ AUDIO_IN_SAI_PDMx_CLK_IN_ENABLE();
+ AUDIO_IN_SAI_PDMx_DATA_IN_ENABLE();
+
+ gpio_init_structure.Pin = AUDIO_IN_SAI_PDMx_CLK_IN_PIN;
+ gpio_init_structure.Mode = GPIO_MODE_AF_PP;
+ gpio_init_structure.Pull = GPIO_NOPULL;
+ gpio_init_structure.Speed = GPIO_SPEED_FREQ_HIGH;
+ gpio_init_structure.Alternate = AUDIO_IN_SAI_PDMx_DATA_CLK_AF;
+ HAL_GPIO_Init(AUDIO_IN_SAI_PDMx_CLK_IN_PORT, &gpio_init_structure);
+
+ gpio_init_structure.Pull = GPIO_PULLUP;
+ gpio_init_structure.Speed = GPIO_SPEED_FREQ_MEDIUM;
+ gpio_init_structure.Pin = AUDIO_IN_SAI_PDMx_DATA_IN_PIN;
+ HAL_GPIO_Init(AUDIO_IN_SAI_PDMx_DATA_IN_PORT, &gpio_init_structure);
+
+ AUDIO_IN_SAI_PDMx_FS_SCK_ENABLE();
+
+ /* CODEC_SAI pins configuration: FS, SCK, MCK and SD pins ------------------*/
+ gpio_init_structure.Pin = AUDIO_IN_SAI_PDMx_FS_PIN | AUDIO_IN_SAI_PDMx_SCK_PIN;
+ gpio_init_structure.Mode = GPIO_MODE_AF_PP;
+ gpio_init_structure.Pull = GPIO_NOPULL;
+ gpio_init_structure.Speed = GPIO_SPEED_FREQ_HIGH;
+ gpio_init_structure.Alternate = AUDIO_IN_SAI_PDMx_FS_SCK_AF;
+ HAL_GPIO_Init(AUDIO_IN_SAI_PDMx_FS_SCK_GPIO_PORT, &gpio_init_structure);
+
+ /* Enable the DMA clock */
+ AUDIO_IN_SAI_PDMx_DMAx_CLK_ENABLE();
+
+ /* Configure the hdma_sai_rx handle parameters */
+ hdma_sai_rx.Init.Request = AUDIO_IN_SAI_PDMx_DMAx_REQUEST;
+ hdma_sai_rx.Init.Direction = DMA_PERIPH_TO_MEMORY;
+ hdma_sai_rx.Init.PeriphInc = DMA_PINC_DISABLE;
+ hdma_sai_rx.Init.MemInc = DMA_MINC_ENABLE;
+ hdma_sai_rx.Init.PeriphDataAlignment = AUDIO_IN_SAI_PDMx_DMAx_PERIPH_DATA_SIZE;
+ hdma_sai_rx.Init.MemDataAlignment = AUDIO_IN_SAI_PDMx_DMAx_MEM_DATA_SIZE;
+ hdma_sai_rx.Init.Mode = DMA_CIRCULAR;
+ hdma_sai_rx.Init.Priority = DMA_PRIORITY_HIGH;
+ hdma_sai_rx.Init.FIFOMode = DMA_FIFOMODE_DISABLE;
+ hdma_sai_rx.Init.FIFOThreshold = DMA_FIFO_THRESHOLD_FULL;
+ hdma_sai_rx.Init.MemBurst = DMA_MBURST_SINGLE;
+ hdma_sai_rx.Init.PeriphBurst = DMA_MBURST_SINGLE;
+
+ hdma_sai_rx.Instance = AUDIO_IN_SAI_PDMx_DMAx_STREAM;
+
+ /* Associate the DMA handle */
+ __HAL_LINKDMA(hsai, hdmarx, hdma_sai_rx);
+
+ /* Deinitialize the Stream for new transfer */
+ HAL_DMA_DeInit(&hdma_sai_rx);
+
+ /* Configure the DMA Stream */
+ HAL_DMA_Init(&hdma_sai_rx);
+
+ /* SAI DMA IRQ Channel configuration */
+ HAL_NVIC_SetPriority(AUDIO_IN_SAI_PDMx_DMAx_IRQ, AUDIO_IN_IRQ_PREPRIO, 0);
+ HAL_NVIC_EnableIRQ(AUDIO_IN_SAI_PDMx_DMAx_IRQ);
+ }
+ else
+ {
+ /* Enable SAI clock */
+ AUDIO_IN_SAIx_CLK_ENABLE();
+
+ /* Enable SD GPIO clock */
+ AUDIO_IN_SAIx_SD_ENABLE();
+ /* CODEC_SAI pin configuration: SD pin */
+ gpio_init_structure.Pin = AUDIO_IN_SAIx_SD_PIN;
+ gpio_init_structure.Mode = GPIO_MODE_AF_PP;
+ gpio_init_structure.Pull = GPIO_NOPULL;
+ gpio_init_structure.Speed = GPIO_SPEED_FREQ_HIGH;
+ gpio_init_structure.Alternate = AUDIO_IN_SAIx_AF;
+ HAL_GPIO_Init(AUDIO_IN_SAIx_SD_GPIO_PORT, &gpio_init_structure);
+
+ /* Enable Audio INT GPIO clock */
+ AUDIO_IN_INT_GPIO_ENABLE();
+ /* Audio INT pin configuration: input */
+ gpio_init_structure.Pin = AUDIO_IN_INT_GPIO_PIN;
+ gpio_init_structure.Mode = GPIO_MODE_INPUT;
+ gpio_init_structure.Pull = GPIO_NOPULL;
+ gpio_init_structure.Speed = GPIO_SPEED_FREQ_HIGH;
+ HAL_GPIO_Init(AUDIO_IN_INT_GPIO_PORT, &gpio_init_structure);
+
+ /* Enable the DMA clock */
+ AUDIO_IN_SAIx_DMAx_CLK_ENABLE();
+
+ /* Configure the hdma_sai_rx handle parameters */
+ hdma_sai_rx.Init.Request = AUDIO_IN_SAIx_DMAx_REQUEST;
+ hdma_sai_rx.Init.Direction = DMA_PERIPH_TO_MEMORY;
+ hdma_sai_rx.Init.PeriphInc = DMA_PINC_DISABLE;
+ hdma_sai_rx.Init.MemInc = DMA_MINC_ENABLE;
+ hdma_sai_rx.Init.PeriphDataAlignment = AUDIO_IN_SAIx_DMAx_PERIPH_DATA_SIZE;
+ hdma_sai_rx.Init.MemDataAlignment = AUDIO_IN_SAIx_DMAx_MEM_DATA_SIZE;
+ hdma_sai_rx.Init.Mode = DMA_CIRCULAR;
+ hdma_sai_rx.Init.Priority = DMA_PRIORITY_HIGH;
+ hdma_sai_rx.Init.FIFOMode = DMA_FIFOMODE_DISABLE;
+ hdma_sai_rx.Init.FIFOThreshold = DMA_FIFO_THRESHOLD_FULL;
+ hdma_sai_rx.Init.MemBurst = DMA_MBURST_SINGLE;
+ hdma_sai_rx.Init.PeriphBurst = DMA_MBURST_SINGLE;
+
+ hdma_sai_rx.Instance = AUDIO_IN_SAIx_DMAx_STREAM;
+
+ /* Associate the DMA handle */
+ __HAL_LINKDMA(hsai, hdmarx, hdma_sai_rx);
+
+ /* Deinitialize the Stream for new transfer */
+ HAL_DMA_DeInit(&hdma_sai_rx);
+
+ /* Configure the DMA Stream */
+ HAL_DMA_Init(&hdma_sai_rx);
+
+ /* SAI DMA IRQ Channel configuration */
+ HAL_NVIC_SetPriority(AUDIO_IN_SAIx_DMAx_IRQ, AUDIO_IN_IRQ_PREPRIO, 0);
+ HAL_NVIC_EnableIRQ(AUDIO_IN_SAIx_DMAx_IRQ);
+
+ /* Audio INT IRQ Channel configuration */
+ HAL_NVIC_SetPriority(AUDIO_IN_INT_IRQ, AUDIO_IN_IRQ_PREPRIO, 0);
+ HAL_NVIC_EnableIRQ(AUDIO_IN_INT_IRQ);
+ }
+}
+
+/**
+ * @brief De-Initializes SAI Audio IN MSP.
+ * @param hsai: SAI handle
+ * @param Params: pointer on additional configuration parameters, can be NULL.
+ * @retval None
+ */
+static void SAIx_In_MspDeInit(SAI_HandleTypeDef *hsai, void *Params)
+{
+ GPIO_InitTypeDef gpio_init_structure;
+
+ if(hsai->Instance == AUDIO_IN_SAI_PDMx)
+ {
+ /* Deinitialize the DMA stream */
+ HAL_DMA_Abort(hsai->hdmarx);
+
+ HAL_SAI_DeInit(hsai);
+ /* Disable SAI peripheral */
+ __HAL_SAI_DISABLE(hsai);
+
+ /* Deinitialize the DMA stream */
+ HAL_DMA_DeInit(hsai->hdmarx);
+
+ gpio_init_structure.Pin = AUDIO_IN_SAI_PDMx_CLK_IN_PIN;
+ HAL_GPIO_DeInit(AUDIO_IN_SAI_PDMx_CLK_IN_PORT, gpio_init_structure.Pin);
+
+ gpio_init_structure.Pin = AUDIO_IN_SAI_PDMx_DATA_IN_PIN;
+ HAL_GPIO_DeInit(AUDIO_IN_SAI_PDMx_DATA_IN_PORT, gpio_init_structure.Pin);
+
+ /* CODEC_SAI pins configuration: FS, SCK, MCK and SD pins ------------------*/
+ gpio_init_structure.Pin = AUDIO_IN_SAI_PDMx_FS_PIN | AUDIO_IN_SAI_PDMx_SCK_PIN;
+ HAL_GPIO_DeInit(AUDIO_IN_SAI_PDMx_FS_SCK_GPIO_PORT, gpio_init_structure.Pin);
+
+ /* Disable SAI clock */
+ AUDIO_IN_SAI_PDMx_CLK_DISABLE();
+ }
+ else
+ {
+ /* SAI DMA IRQ Channel deactivation */
+ HAL_NVIC_DisableIRQ(AUDIO_IN_SAIx_DMAx_IRQ);
+
+ if(hsai->Instance == AUDIO_IN_SAIx)
+ {
+ /* Deinitialize the DMA stream */
+ HAL_DMA_DeInit(hsai->hdmatx);
+ }
+
+ /* Disable SAI peripheral */
+ __HAL_SAI_DISABLE(hsai);
+
+ /* Deactivates CODEC_SAI pin SD by putting them in input mode */
+ gpio_init_structure.Pin = AUDIO_IN_SAIx_SD_PIN;
+ HAL_GPIO_DeInit(AUDIO_IN_SAIx_SD_GPIO_PORT, gpio_init_structure.Pin);
+
+ gpio_init_structure.Pin = AUDIO_IN_INT_GPIO_PIN;
+ HAL_GPIO_DeInit(AUDIO_IN_INT_GPIO_PORT, gpio_init_structure.Pin);
+
+ /* Disable SAI clock */
+ AUDIO_IN_SAIx_CLK_DISABLE();
+ }
+}
+
+/**
+ * @brief Initializes the Audio Codec audio interface (SAI).
+ * @param SaiInMode: Audio mode to be configured for the SAI peripheral.
+ * @param SlotActive: Audio active slot to be configured for the SAI peripheral.
+ * @param AudioFreq: Audio frequency to be configured for the SAI peripheral.
+ * @retval None
+ */
+static void SAIx_In_Init(uint32_t SaiInMode, uint32_t SlotActive, uint32_t AudioFreq)
+{
+ /* Disable SAI peripheral to allow access to SAI internal registers */
+ __HAL_SAI_DISABLE(&haudio_in_sai);
+
+ /* Configure SAI_Block_x
+ LSBFirst: Disabled
+ DataSize: 16 */
+ haudio_in_sai.Init.MonoStereoMode = SAI_STEREOMODE;
+ haudio_in_sai.Init.AudioFrequency = AudioFreq;
+ haudio_in_sai.Init.AudioMode = SaiInMode;
+ haudio_in_sai.Init.NoDivider = SAI_MASTERDIVIDER_ENABLE;
+ haudio_in_sai.Init.Protocol = SAI_FREE_PROTOCOL;
+ haudio_in_sai.Init.DataSize = SAI_DATASIZE_16;
+ haudio_in_sai.Init.FirstBit = SAI_FIRSTBIT_MSB;
+ haudio_in_sai.Init.ClockStrobing = SAI_CLOCKSTROBING_RISINGEDGE;
+ haudio_in_sai.Init.Synchro = SAI_SYNCHRONOUS;
+ haudio_in_sai.Init.OutputDrive = SAI_OUTPUTDRIVE_DISABLE;
+ haudio_in_sai.Init.FIFOThreshold = SAI_FIFOTHRESHOLD_1QF;
+ haudio_in_sai.Init.SynchroExt = SAI_SYNCEXT_DISABLE;
+ haudio_in_sai.Init.CompandingMode = SAI_NOCOMPANDING;
+ haudio_in_sai.Init.TriState = SAI_OUTPUT_RELEASED;
+ haudio_in_sai.Init.Mckdiv = 0;
+ haudio_in_sai.Init.MckOverSampling = SAI_MCK_OVERSAMPLING_DISABLE;
+ haudio_in_sai.Init.PdmInit.Activation = DISABLE;
+
+ /* Configure SAI_Block_x Frame
+ Frame Length: 64
+ Frame active Length: 32
+ FS Definition: Start frame + Channel Side identification
+ FS Polarity: FS active Low
+ FS Offset: FS asserted one bit before the first bit of slot 0 */
+ haudio_in_sai.FrameInit.FrameLength = 128;
+ haudio_in_sai.FrameInit.ActiveFrameLength = 64;
+ haudio_in_sai.FrameInit.FSDefinition = SAI_FS_CHANNEL_IDENTIFICATION;
+ haudio_in_sai.FrameInit.FSPolarity = SAI_FS_ACTIVE_LOW;
+ haudio_in_sai.FrameInit.FSOffset = SAI_FS_BEFOREFIRSTBIT;
+
+ /* Configure SAI Block_x Slot
+ Slot First Bit Offset: 0
+ Slot Size : 16
+ Slot Number: 4
+ Slot Active: All slot active */
+ haudio_in_sai.SlotInit.FirstBitOffset = 0;
+ haudio_in_sai.SlotInit.SlotSize = SAI_SLOTSIZE_DATASIZE;
+ haudio_in_sai.SlotInit.SlotNumber = 4;
+ haudio_in_sai.SlotInit.SlotActive = SlotActive;
+
+ if(hAudioIn.Interface == AUDIO_IN_INTERFACE_PDM)
+ {
+ haudio_in_sai.Init.AudioFrequency = AudioFreq * 8;
+ haudio_in_sai.Init.Synchro = SAI_ASYNCHRONOUS;
+ haudio_in_sai.Init.NoDivider = SAI_MASTERDIVIDER_DISABLE;
+
+ haudio_in_sai.Init.PdmInit.Activation = ENABLE;
+ haudio_in_sai.Init.PdmInit.MicPairsNbr = 1;
+ haudio_in_sai.Init.PdmInit.ClockEnable = SAI_PDM_CLOCK1_ENABLE;
+ haudio_in_sai.Init.FirstBit = SAI_FIRSTBIT_LSB;
+ haudio_in_sai.Init.ClockStrobing = SAI_CLOCKSTROBING_FALLINGEDGE;
+
+ haudio_in_sai.FrameInit.FrameLength = 16;
+ haudio_in_sai.FrameInit.ActiveFrameLength = 1;
+ haudio_in_sai.FrameInit.FSDefinition = SAI_FS_STARTFRAME;
+ haudio_in_sai.FrameInit.FSPolarity = SAI_FS_ACTIVE_HIGH;
+ haudio_in_sai.FrameInit.FSOffset = SAI_FS_FIRSTBIT;
+
+ haudio_in_sai.SlotInit.SlotNumber = 1;
+ haudio_in_sai.SlotInit.SlotActive = SlotActive;
+ }
+
+ HAL_SAI_Init(&haudio_in_sai);
+
+ /* Enable SAI peripheral */
+ __HAL_SAI_ENABLE(&haudio_in_sai);
+}
+
+/**
+ * @brief De-initializes the output Audio Codec audio interface (SAI).
+ * @retval None
+ */
+static void SAIx_In_DeInit(SAI_HandleTypeDef *hsai)
+{
+ /* Disable SAI peripheral */
+ __HAL_SAI_DISABLE(hsai);
+
+ HAL_SAI_DeInit(hsai);
+}
+
+/**
+ * @}
+ */
+
+/**
+ * @}
+ */
+
+/**
+ * @}
+ */
+
+/**
+ * @}
+ */
+
+/************************ (C) COPYRIGHT STMicroelectronics *****END OF FILE****/