Senior Design: Sound Monitor
Audio Demo with DISCO Board, takes control samples, waits for user input, samples regularly.
Dependencies: CMSIS_DSP_401 STM32L4xx_HAL_Driver mbed-src_DISO_AUDIO_DEMO
Diff: stm32l476g_discovery_audio.c
- Revision:
- 0:3eee9435dd17
diff -r 000000000000 -r 3eee9435dd17 stm32l476g_discovery_audio.c
--- /dev/null Thu Jan 01 00:00:00 1970 +0000
+++ b/stm32l476g_discovery_audio.c Sun Dec 13 19:12:11 2015 +0000
@@ -0,0 +1,1071 @@
+/**
+ ******************************************************************************
+ * @file stm32l476g_discovery_audio.c
+ * @author MCD Application Team
+ * @version V1.0.1
+ * @date 16-September-2015
+ * @brief This file provides a set of functions needed to manage the
+ * Audio driver for the STM32L476G-Discovery board.
+ ******************************************************************************
+ * @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.
+ *
+ ******************************************************************************
+ */
+
+/*==============================================================================
+ User NOTES
+
+1. How To use this driver:
+--------------------------
+ + This driver supports STM32L4xx devices on STM32L476G-Discovery (MB1184) Discovery boards.
+ a) to play an audio file (all functions names start by BSP_AUDIO_OUT_xxx)
+ b) to record an audio file through MP34DT01TR, ST MEMS (all functions names start by BSP_AUDIO_IN_xxx)
+
+a) PLAY A FILE:
+==============
+ + 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 audio codec has failed.
+ - 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.
+
+ + Call the function BSP_AUDIO_OUT_RegisterCallbacks to register user callbacks
+ required to manage audio data streaming towards the audio codec (ErrorCallback(),
+ HalfTransfer_CallBack() and TransferComplete_CallBack()).
+
+ + Call the function BSP_AUDIO_OUT_Play() to start audio playback (for the first time).
+ + Call the function BSP_AUDIO_OUT_Pause() to pause audio playabck
+ + Call the function BSP_AUDIO_OUT_Resume() to resume audio playback.
+ 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).
+ + Call the function BSP_AUDIO_OUT_Stop() to stop audio playback.
+ + To modify the volume level, the sampling frequency, the device output mode,
+ the mute status or the audio configuration or the stop, use the functions: BSP_AUDIO_OUT_SetVolume(),
+ AUDIO_OUT_SetFrequency(), BSP_AUDIO_OUT_SetOutputMode(), BSP_AUDIO_OUT_SetMute()and
+ BSP_AUDIO_OUT_ChangeAudioConfig().
+
+Driver architecture:
+--------------------
+ + This driver provides the audio layer high level API: it consists in functions
+ exported in the stm32l476g_discovery_audio.h file (e.g. BSP_AUDIO_OUT_Init(),
+ BSP_AUDIO_OUT_Play(), ...).
+ + This driver also includes the Media Access Layer (MAL): it consists in
+ functions allowing to access setup the audio devices. These functions
+ are included as local functions into the stm32l476g_discovery_audio.c file
+ (e.g. AUDIO_SAIx_Init()).
+
+Known Limitations:
+------------------
+ 1- Communication with the audio codec (through I2C) may be corrupted if it is interrupted by some
+ user interrupt routines (in this case, interrupts could be disabled just before the start of
+ communication then re-enabled when it is over). Note that this communication is only done at
+ the configuration phase (BSP_AUDIO_OUT_Init() or BSP_AUDIO_OUT_Stop()) and when Volume control modification is
+ performed (BSP_AUDIO_OUT_SetVolume() or BSP_AUDIO_OUT_SetMute()or BSP_AUDIO_OUT_SetOutputMode()).
+ When the audio data is played, no communication is required with the audio codec.
+ 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 16-bits audio data size.
+
+b) RECORD A FILE:
+================
+ + Call the function BSP_AUDIO_IN_Init(
+ AudioFreq: Audio frequency in Hz (8000, 16000, 22500, 32000 ...)
+ )
+ This function configures all the hardware required for the audio application (DFSDM,
+ GPIOs, DMA and interrupt if needed). This function returns AUDIO_OK if the
+ configuration completes successfully.
+
+ + Call the function BSP_AUDIO_IN_RegisterCallbacks to register user callbacks
+ used to stream audio data toward the record buffer (ErrorCallback(),
+ HalfTransfer_CallBack() and TransferComplete_CallBack()).
+
+ + Call the function BSP_AUDIO_IN_Record(
+ pbuf Main buffer pointer for the recorded data storing
+ size Current size of the recorded buffer
+ )
+ to start recording from the microphone.
+
+ + Call the function AUDIO_IN_STOP() to stop recording
+==============================================================================*/
+
+/* Includes ------------------------------------------------------------------*/
+#include <string.h>
+#include "stm32l476g_discovery_audio.h"
+
+/** @addtogroup BSP
+ * @{
+ */
+
+/** @addtogroup STM32L476G_DISCOVERY
+ * @{
+ */
+
+/** @defgroup STM32L476G_DISCOVERY_AUDIO STM32L476G-DISCOVERY AUDIO
+ * @brief This file includes the low layer driver for cs43l22 Audio Codec
+ * available on STM32L476G-Discovery board(MB1184).
+ * @{
+ */
+
+/* Private typedef -----------------------------------------------------------*/
+/** @defgroup STM32L476G_DISCOVERY_AUDIO_Private_Types Private Types
+ * @{
+ */
+typedef struct
+{
+
+ Audio_CallbackTypeDef CbError; /* pointer to the callback function invoked when ... */
+ Audio_CallbackTypeDef CbHalfTransfer; /* pointer to the callback function invoked when ... */
+ Audio_CallbackTypeDef CbTransferComplete; /* pointer to the callback function invoked when ... */
+} AUDIO_OUT_TypeDef;
+
+typedef struct
+{
+ DFSDM_Channel_HandleTypeDef hDfsdmLeftChannel; /* DFSDM channel handle used for left channel */
+ DMA_HandleTypeDef hDmaDfsdmLeft; /* DMA handle used for DFSDM regular conversions on left channel */
+ int32_t * LeftRecBuff; /* Buffers for left samples */
+ uint32_t Frequency; /* Record Frequency */
+ uint32_t BitResolution; /* Record bit resolution */
+ uint32_t ChannelNbr; /* Record Channel Number */
+ uint16_t * pRecBuf; /* Pointer to record user buffer */
+ uint32_t RecSize; /* Size to record in mono, double size to record in stereo */
+ Audio_CallbackTypeDef CbError; /* pointer to the callback function invoked when a DMA transfer fails */
+ Audio_CallbackTypeDef CbHalfTransfer; /* pointer to the callback function invoked when half of the DMA transfer is completed */
+ Audio_CallbackTypeDef CbTransferComplete; /* pointer to the callback function invoked when the DMA transfer is completed */
+} AUDIO_IN_TypeDef;
+
+/**
+ * @}
+ */
+
+/* Private defines ------------------------------------------------------------*/
+/** @defgroup STM32L476G_DISCOVERY_AUDIO_Private_Constants Private Constants
+ * @{
+ */
+/**
+ * @}
+ */
+
+/* Private macros ------------------------------------------------------------*/
+/** @defgroup STM32L476G_DISCOVERY_AUDIO_Private_Macros Private Macros
+ * @{
+ */
+/*### PLAY ###*/
+/* SCK(kHz) = SAI_CK_x/(SAIClockDivider*2*256) */
+#define SAIClockDivider(__FREQUENCY__) \
+ (__FREQUENCY__ == AUDIO_FREQUENCY_8K) ? 12 \
+ : (__FREQUENCY__ == AUDIO_FREQUENCY_11K) ? 2 \
+ : (__FREQUENCY__ == AUDIO_FREQUENCY_16K) ? 6 \
+ : (__FREQUENCY__ == AUDIO_FREQUENCY_22K) ? 1 \
+ : (__FREQUENCY__ == AUDIO_FREQUENCY_32K) ? 3 \
+ : (__FREQUENCY__ == AUDIO_FREQUENCY_44K) ? 0 \
+ : (__FREQUENCY__ == AUDIO_FREQUENCY_48K) ? 2 : 1 \
+
+/*### RECORD ###*/
+#define DFSDMOverSampling(__FREQUENCY__) \
+ (__FREQUENCY__ == AUDIO_FREQUENCY_8K) ? 256 \
+ : (__FREQUENCY__ == AUDIO_FREQUENCY_11K) ? 256 \
+ : (__FREQUENCY__ == AUDIO_FREQUENCY_16K) ? 128 \
+ : (__FREQUENCY__ == AUDIO_FREQUENCY_22K) ? 128 \
+ : (__FREQUENCY__ == AUDIO_FREQUENCY_32K) ? 64 \
+ : (__FREQUENCY__ == AUDIO_FREQUENCY_44K) ? 64 \
+ : (__FREQUENCY__ == AUDIO_FREQUENCY_48K) ? 32 : 16 \
+
+#define DFSDMClockDivider(__FREQUENCY__) \
+ (__FREQUENCY__ == AUDIO_FREQUENCY_8K) ? 24 \
+ : (__FREQUENCY__ == AUDIO_FREQUENCY_11K) ? 4 \
+ : (__FREQUENCY__ == AUDIO_FREQUENCY_16K) ? 24 \
+ : (__FREQUENCY__ == AUDIO_FREQUENCY_22K) ? 4 \
+ : (__FREQUENCY__ == AUDIO_FREQUENCY_32K) ? 24 \
+ : (__FREQUENCY__ == AUDIO_FREQUENCY_44K) ? 4 \
+ : (__FREQUENCY__ == AUDIO_FREQUENCY_48K) ? 32 : 32 \
+
+#define DFSDMFilterOrder(__FREQUENCY__) \
+ (__FREQUENCY__ == AUDIO_FREQUENCY_8K) ? DFSDM_FILTER_SINC3_ORDER \
+ : (__FREQUENCY__ == AUDIO_FREQUENCY_11K) ? DFSDM_FILTER_SINC3_ORDER \
+ : (__FREQUENCY__ == AUDIO_FREQUENCY_16K) ? DFSDM_FILTER_SINC3_ORDER \
+ : (__FREQUENCY__ == AUDIO_FREQUENCY_22K) ? DFSDM_FILTER_SINC3_ORDER \
+ : (__FREQUENCY__ == AUDIO_FREQUENCY_32K) ? DFSDM_FILTER_SINC4_ORDER \
+ : (__FREQUENCY__ == AUDIO_FREQUENCY_44K) ? DFSDM_FILTER_SINC4_ORDER \
+ : (__FREQUENCY__ == AUDIO_FREQUENCY_48K) ? DFSDM_FILTER_SINC4_ORDER : DFSDM_FILTER_SINC5_ORDER \
+
+#define DFSDMRightBitShift(__FREQUENCY__) \
+ (__FREQUENCY__ == AUDIO_FREQUENCY_8K) ? 2 \
+ : (__FREQUENCY__ == AUDIO_FREQUENCY_11K) ? 3 \
+ : (__FREQUENCY__ == AUDIO_FREQUENCY_16K) ? 3 \
+ : (__FREQUENCY__ == AUDIO_FREQUENCY_22K) ? 0 \
+ : (__FREQUENCY__ == AUDIO_FREQUENCY_32K) ? 3 \
+ : (__FREQUENCY__ == AUDIO_FREQUENCY_44K) ? 3 \
+ : (__FREQUENCY__ == AUDIO_FREQUENCY_48K) ? 7 : 0 \
+
+/* Saturate the record PCM sample */
+#define SaturaLH(N, L, H) (((N)<(L))?(L):(((N)>(H))?(H):(N)))
+
+/**
+ * @}
+ */
+
+/* Private variables ---------------------------------------------------------*/
+/** @defgroup STM32L476G_DISCOVERY_AUDIO_Private_Variables Private Variables
+ * @{
+ */
+/* Audio output context information */
+static AUDIO_OUT_TypeDef hAudioOut;
+
+/* Audio input context information */
+static AUDIO_IN_TypeDef hAudioIn;
+
+/* SAI DMA handle */
+static DMA_HandleTypeDef hDmaSai;
+/**
+ * @}
+ */
+
+/* Exported variables ---------------------------------------------------------*/
+/** @defgroup STM32L476G_DISCOVERY_AUDIO_Exported_Variables Exported Variables
+ * @{
+ */
+/* SAIx handle */
+SAI_HandleTypeDef BSP_AUDIO_hSai;
+
+/* DFSDM filter handle */
+DFSDM_Filter_HandleTypeDef BSP_AUDIO_hDfsdmLeftFilter;
+/**
+ * @}
+ */
+
+/* Private function prototypes -----------------------------------------------*/
+/** @defgroup STM32L476G_DISCOVERY_AUDIO_Private_Functions Private Functions
+ * @{
+ */
+static void AUDIO_CODEC_Reset(void);
+static uint8_t AUDIO_SAIx_Init(uint32_t AudioFreq);
+static uint8_t AUDIO_SAIx_DeInit(void);
+static uint8_t AUDIO_DFSDMx_Init(uint32_t AudioFreq);
+static uint8_t AUDIO_DFSDMx_DeInit(void);
+static uint8_t AUDIO_SAIPLLConfig(uint32_t AudioFreq);
+/**
+ * @}
+ */
+
+/* Exported functions --------------------------------------------------------*/
+/** @addtogroup STM32L476G_DISCOVERY_AUDIO_Exported_Functions
+ * @{
+ */
+
+/**
+ * @brief Configures the audio codec related 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.ion.
+ * @retval BSP AUDIO status
+ * @note The SAI PLL input clock must be configure in the user application.
+ * The SAI PLL configuration done within this function assumes that
+ * the SAI PLL input clock runs at 8 MHz.
+ */
+
+/**
+ * @brief Tx Transfer completed callbacks.
+ * @param hsai: SAI handle
+ * @retval None
+ */
+void HAL_SAI_TxCpltCallback(SAI_HandleTypeDef *hsai)
+{
+ /* Invoke the registered 'TransferComplete' function (if any) */
+ if (hAudioOut.CbTransferComplete != (Audio_CallbackTypeDef)NULL)
+ {
+ hAudioOut.CbTransferComplete();
+ }
+}
+
+/**
+ * @brief Tx Half Transfer completed callbacks.
+ * @param hsai: SAI handle
+ * @retval None
+ */
+void HAL_SAI_TxHalfCpltCallback(SAI_HandleTypeDef *hsai)
+{
+ /* Invoke the registered 'HalfTransfer' callback function (if any) */
+ if (hAudioOut.CbHalfTransfer != (Audio_CallbackTypeDef)NULL)
+ {
+ hAudioOut.CbHalfTransfer();
+ }
+}
+
+/**
+ * @brief SAI error callbacks.
+ * @param hsai: SAI handle
+ * @retval None
+ */
+void HAL_SAI_ErrorCallback(SAI_HandleTypeDef *hsai)
+{
+ /* Invoke the registered 'ErrorCallback' callback function (if any) */
+ if (hAudioOut.CbError != (Audio_CallbackTypeDef)NULL)
+ {
+ hAudioOut.CbError();
+ }
+}
+
+/**
+ * @}
+ */
+
+/** @addtogroup STM32L476G_EVAL_AUDIO_Exported_Functions
+ * @{
+ */
+
+/**
+ * @brief Initializes micropone related peripherals.
+ * @note This function assumes that the SAI input clock (through PLL_M)
+ * is already configured and ready to be used.
+ * @param AudioFreq: Audio frequency to be configured for the SAI peripheral.
+ * @param BitRes: Audio frequency to be configured for the SAI peripheral.
+ * @param ChnlNbr: Audio frequency to be configured for the SAI peripheral.
+ * @retval BSP AUDIO status
+ */
+uint8_t BSP_AUDIO_IN_Init(uint32_t AudioFreq, uint32_t BitRes, uint32_t ChnlNbr)
+{
+ /* Update the audio input context */
+ hAudioIn.Frequency = AudioFreq;
+ hAudioIn.BitResolution = BitRes;
+ hAudioIn.ChannelNbr = ChnlNbr;
+ hAudioIn.CbError = (Audio_CallbackTypeDef)NULL;
+ hAudioIn.CbHalfTransfer = (Audio_CallbackTypeDef)NULL;
+ hAudioIn.CbTransferComplete = (Audio_CallbackTypeDef)NULL;
+
+ /* Configure the SAI PLL according to the requested audio frequency */
+ if (AUDIO_SAIPLLConfig(AudioFreq) != AUDIO_OK)
+ {
+ return AUDIO_ERROR;
+ }
+
+ /* Initializes the Digital Filter for Sigma-Delta Modulators interface */
+ if(AUDIO_DFSDMx_Init(AudioFreq) != AUDIO_OK)
+ {
+ return AUDIO_ERROR;
+ }
+
+ return AUDIO_OK;
+ }
+
+/**
+ * @brief De-Initializes microphone related peripherals.
+ * @retval BSP AUDIO status
+
+ */
+uint8_t BSP_AUDIO_IN_DeInit(void)
+{
+ /* De-initializes the Digital Filter for Sigma-Delta Modulators interface */
+ if (AUDIO_DFSDMx_DeInit() != AUDIO_OK)
+ {
+ return AUDIO_ERROR;
+ }
+
+ /* Reset the audio input context */
+ memset(&hAudioIn, 0, sizeof(hAudioIn));
+
+ return AUDIO_OK;
+}
+
+/**
+ * @brief Starts audio recording.
+ * @param pbuf: Main buffer pointer for the recorded data storing
+ * @param size: Current size of the recorded buffer
+ * @note The Right channel is start at first with synchro on start of Left channel
+ * @retval BSP AUDIO status
+ */
+uint8_t BSP_AUDIO_IN_Record(uint16_t* pbuf, uint32_t size)
+{
+ hAudioIn.pRecBuf = pbuf;
+ hAudioIn.RecSize = size;
+
+ /* Allocate hAudioIn.LeftRecBuff buffer */
+#if defined(BSP_AUDIO_USE_RTOS)
+ hAudioIn.LeftRecBuff = (int32_t *)k_malloc(size * sizeof(int32_t));
+#else
+ hAudioIn.LeftRecBuff = (int32_t *)malloc(size * sizeof(int32_t));
+#endif
+ if(hAudioIn.LeftRecBuff == NULL)
+ {
+ return AUDIO_ERROR;
+ }
+
+ /* Call the Media layer start function for left channel */
+ if(HAL_DFSDM_FilterRegularStart_DMA(&BSP_AUDIO_hDfsdmLeftFilter,
+ (int32_t*)hAudioIn.LeftRecBuff,
+ (hAudioIn.RecSize/DEFAULT_AUDIO_IN_CHANNEL_NBR)) != HAL_OK)
+ {
+ return AUDIO_ERROR;
+ }
+
+ return AUDIO_OK;
+}
+
+/**
+ * @brief Updates the audio frequency.
+ * @param AudioFreq: Audio frequency used to record the audio stream.
+ * @note This API should be called after the BSP_AUDIO_IN_Init() to adjust the
+ * audio frequency.
+ * @retval BSP AUDIO status
+ */
+uint8_t BSP_AUDIO_IN_SetFrequency(uint32_t AudioFreq)
+{
+ /* Configure the SAI PLL according to the requested audio frequency */
+ if (AUDIO_SAIPLLConfig(AudioFreq) != AUDIO_OK)
+ {
+ return AUDIO_ERROR;
+ }
+
+ /* De-initializes the Digital Filter for Sigma-Delta Modulators interface */
+ if(AUDIO_DFSDMx_DeInit() != AUDIO_OK)
+ {
+ return AUDIO_ERROR;
+ }
+
+ /* Initializes the Digital Filter for Sigma-Delta Modulators interface */
+ if(AUDIO_DFSDMx_Init(AudioFreq) != AUDIO_OK)
+ {
+ return AUDIO_ERROR;
+ }
+
+ return AUDIO_OK;
+}
+
+/**
+ * @brief Regular conversion complete callback.
+ * @note In interrupt mode, user has to read conversion value in this function
+ using HAL_DFSDM_FilterGetRegularValue.
+ * @param hdfsdm_filter : DFSDM filter handle.
+ * @retval None
+ */
+void HAL_DFSDM_FilterRegConvCpltCallback(DFSDM_Filter_HandleTypeDef *hdfsdm_filter)
+{
+ uint32_t index;
+ uint32_t recbufsize = (hAudioIn.RecSize/DEFAULT_AUDIO_IN_CHANNEL_NBR);
+
+ for(index = (recbufsize/2); index < recbufsize; index++)
+ {
+ hAudioIn.pRecBuf[index] = (uint16_t)(SaturaLH((hAudioIn.LeftRecBuff[index] >> 8), -32760, 32760));
+ }
+
+ /* Invoke the registered 'TransferComplete' function (if any) */
+ if (hAudioIn.CbTransferComplete != (Audio_CallbackTypeDef)NULL)
+ {
+ hAudioIn.CbTransferComplete();
+ }
+}
+
+/**
+ * @brief Half regular conversion complete callback.
+ * @param hdfsdm_filter : DFSDM filter handle.
+ * @retval None
+ */
+void HAL_DFSDM_FilterRegConvHalfCpltCallback(DFSDM_Filter_HandleTypeDef *hdfsdm_filter)
+{
+ uint32_t index;
+ uint32_t recbufsize = (hAudioIn.RecSize/DEFAULT_AUDIO_IN_CHANNEL_NBR);
+
+
+ for(index = 0; index < (recbufsize/2); index++)
+ {
+ hAudioIn.pRecBuf[index] = (uint16_t)(SaturaLH((hAudioIn.LeftRecBuff[index] >> 8), -32760, 32760));
+ }
+
+ /* Invoke the registered 'HalfTransfer' callback function (if any) */
+ if (hAudioIn.CbHalfTransfer != (Audio_CallbackTypeDef)NULL)
+ {
+ hAudioIn.CbHalfTransfer();
+ }
+}
+
+/**
+ * @brief Error callback.
+ * @param hdfsdm_filter : DFSDM filter handle.
+ * @retval None
+ */
+void HAL_DFSDM_FilterErrorCallback(DFSDM_Filter_HandleTypeDef *hdfsdm_filter)
+{
+ /* Invoke the registered 'ErrorCallback' callback function (if any) */
+ if (hAudioIn.CbError != (Audio_CallbackTypeDef)NULL)
+ {
+ hAudioIn.CbError();
+ }
+}
+
+/**
+ * @brief Stops audio recording.
+ * @retval BSP AUDIO status
+ */
+uint8_t BSP_AUDIO_IN_Stop(void)
+{
+ /* Call the Media layer stop function for left channel */
+ if(HAL_DFSDM_FilterRegularStop_DMA(&BSP_AUDIO_hDfsdmLeftFilter) != HAL_OK )
+ {
+ return AUDIO_ERROR;
+ }
+
+ /* Free hAudioIn.LeftRecBuff buffer */
+#if defined(BSP_AUDIO_USE_RTOS)
+ k_free((void *)hAudioIn.LeftRecBuff);
+#else
+ free((void *)hAudioIn.LeftRecBuff);
+#endif
+
+ return AUDIO_OK;
+}
+
+/**
+ * @brief Pauses the audio file stream.
+ * @retval BSP AUDIO status
+ */
+uint8_t BSP_AUDIO_IN_Pause(void)
+{
+ /* Call the Media layer stop function */
+ if(HAL_DFSDM_FilterRegularStop_DMA(&BSP_AUDIO_hDfsdmLeftFilter) != HAL_OK)
+ {
+ return AUDIO_ERROR;
+ }
+
+ return AUDIO_OK;
+}
+
+/**
+ * @brief Resumes the audio file stream.
+ * @retval BSP AUDIO status
+ */
+uint8_t BSP_AUDIO_IN_Resume(void)
+{
+ /* Call the Media layer start function for left channel */
+ if(HAL_DFSDM_FilterRegularStart_DMA(&BSP_AUDIO_hDfsdmLeftFilter,
+ (int32_t*)hAudioIn.LeftRecBuff,
+ (hAudioIn.RecSize/DEFAULT_AUDIO_IN_CHANNEL_NBR)) != HAL_OK)
+ {
+ return AUDIO_ERROR;
+ }
+
+ return AUDIO_OK;
+}
+
+/**
+ * @brief register user callback functions
+ * @param ErrorCallback: pointer to the error callback function
+ * @param HalfTransferCallback: pointer to the half transfer callback function
+ * @param TransferCompleteCallback: pointer to the transfer complete callback function
+ * @retval None
+ */
+void BSP_AUDIO_IN_RegisterCallbacks(Audio_CallbackTypeDef ErrorCallback,
+ Audio_CallbackTypeDef HalfTransferCallback,
+ Audio_CallbackTypeDef TransferCompleteCallback)
+{
+ hAudioIn.CbError = ErrorCallback;
+ hAudioIn.CbHalfTransfer = HalfTransferCallback;
+ hAudioIn.CbTransferComplete = TransferCompleteCallback;
+}
+/**
+ * @}
+ */
+
+/* private functions --------------------------------------------------------*/
+/** @addtogroup STM32L476G_DISCOVERY_AUDIO_Private_Functions
+ * @{
+ */
+/**
+ * @brief Initializes the Audio Codec audio interface (SAI).
+ * @param AudioFreq: Audio frequency to be configured for the SAI peripheral.
+ * @note The default SlotActive configuration is set to CODEC_AUDIOFRAME_SLOT_0123
+ * and user can update this configuration using
+ * @retval BSP AUDIO status
+ */
+static uint8_t AUDIO_SAIx_Init(uint32_t AudioFreq)
+{
+ /* Disable SAI peripheral to allow access to SAI internal registers */
+ __HAL_SAI_DISABLE(&BSP_AUDIO_hSai);
+
+ /* Initialize the BSP_AUDIO_hSai Instance parameter */
+ BSP_AUDIO_hSai.Instance = AUDIO_SAIx;
+
+ /* Configure SAI_Block_x
+ LSBFirst: Disabled
+ DataSize: 16 */
+ BSP_AUDIO_hSai.Init.AudioMode = SAI_MODEMASTER_TX;
+ BSP_AUDIO_hSai.Init.Synchro = SAI_ASYNCHRONOUS;
+ BSP_AUDIO_hSai.Init.SynchroExt = SAI_SYNCEXT_DISABLE;
+ BSP_AUDIO_hSai.Init.OutputDrive = SAI_OUTPUTDRIVE_ENABLE;
+ BSP_AUDIO_hSai.Init.NoDivider = SAI_MASTERDIVIDER_ENABLE;
+ BSP_AUDIO_hSai.Init.FIFOThreshold = SAI_FIFOTHRESHOLD_1QF;
+ BSP_AUDIO_hSai.Init.AudioFrequency = SAI_AUDIO_FREQUENCY_MCKDIV;
+ BSP_AUDIO_hSai.Init.Mckdiv = SAIClockDivider(AudioFreq);
+ BSP_AUDIO_hSai.Init.MonoStereoMode = SAI_STEREOMODE;
+ BSP_AUDIO_hSai.Init.CompandingMode = SAI_NOCOMPANDING;
+ BSP_AUDIO_hSai.Init.TriState = SAI_OUTPUT_NOTRELEASED;
+ BSP_AUDIO_hSai.Init.Protocol = SAI_FREE_PROTOCOL;
+ BSP_AUDIO_hSai.Init.DataSize = SAI_DATASIZE_16;
+ BSP_AUDIO_hSai.Init.FirstBit = SAI_FIRSTBIT_MSB;
+ BSP_AUDIO_hSai.Init.ClockStrobing = SAI_CLOCKSTROBING_RISINGEDGE;
+
+ /* Configure SAI_Block_x Frame
+ Frame Length: 32
+ Frame active Length: 16
+ 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 */
+ BSP_AUDIO_hSai.FrameInit.FrameLength = 32;
+ BSP_AUDIO_hSai.FrameInit.ActiveFrameLength = 16;
+ BSP_AUDIO_hSai.FrameInit.FSDefinition = SAI_FS_CHANNEL_IDENTIFICATION;
+ BSP_AUDIO_hSai.FrameInit.FSPolarity = SAI_FS_ACTIVE_LOW;
+ BSP_AUDIO_hSai.FrameInit.FSOffset = SAI_FS_BEFOREFIRSTBIT;
+
+ /* Configure SAI Block_x Slot
+ Slot First Bit Offset: 0
+ Slot Size : 16
+ Slot Number: 2
+ Slot Active: Slots 0 and 1 actives */
+ BSP_AUDIO_hSai.SlotInit.FirstBitOffset = 0;
+ BSP_AUDIO_hSai.SlotInit.SlotSize = SAI_SLOTSIZE_DATASIZE;
+ BSP_AUDIO_hSai.SlotInit.SlotNumber = 2;
+ BSP_AUDIO_hSai.SlotInit.SlotActive = SAI_SLOTACTIVE_0 | SAI_SLOTACTIVE_1;
+
+ /* Initializes the SAI peripheral*/
+ if (HAL_SAI_Init(&BSP_AUDIO_hSai) != HAL_OK)
+ {
+ return AUDIO_ERROR;
+ }
+
+ /* Enable SAI peripheral to generate MCLK */
+ __HAL_SAI_ENABLE(&BSP_AUDIO_hSai);
+
+ return AUDIO_OK;
+
+}
+
+/**
+ * @brief De-initializes the Audio Codec audio interface (SAI).
+ * @retval BSP AUDIO status
+ */
+static uint8_t AUDIO_SAIx_DeInit(void)
+{
+ /* Disable the SAI audio block */
+ __HAL_SAI_DISABLE(&BSP_AUDIO_hSai);
+
+ /* De-initializes the SAI peripheral */
+ if (HAL_SAI_DeInit(&BSP_AUDIO_hSai) != HAL_OK)
+ {
+ return AUDIO_ERROR;
+ }
+
+ /* Disable SAIx PLL */
+ if (AUDIO_SAIx_PLL_DISABLE() != AUDIO_OK)
+ {
+ return AUDIO_ERROR;
+ }
+
+ return AUDIO_OK;
+}
+
+/**
+ * @brief SAI MSP Init
+ * @param hsai : pointer to a SAI_HandleTypeDef structure
+ * @retval None
+ */
+void HAL_SAI_MspInit(SAI_HandleTypeDef *hsai)
+{
+ GPIO_InitTypeDef GPIO_InitStruct;
+
+ /* Enable SAI clock */
+ AUDIO_SAIx_CLK_ENABLE();
+
+ /* Enable GPIO clock */
+ AUDIO_SAIx_MCK_SCK_SD_FS_ENABLE();
+
+ /* CODEC_SAI pins configuration: FS, SCK, MCK and SD pins ------------------*/
+ GPIO_InitStruct.Pin = AUDIO_SAIx_FS_PIN | AUDIO_SAIx_SCK_PIN | AUDIO_SAIx_SD_PIN | AUDIO_SAIx_MCK_PIN;
+ GPIO_InitStruct.Mode = GPIO_MODE_AF_PP;
+ GPIO_InitStruct.Pull = GPIO_NOPULL;
+ GPIO_InitStruct.Speed = GPIO_SPEED_FREQ_VERY_HIGH;
+ GPIO_InitStruct.Alternate = AUDIO_SAIx_MCK_SCK_SD_FS_AF;
+ HAL_GPIO_Init(AUDIO_SAIx_MCK_SCK_SD_FS_GPIO_PORT, &GPIO_InitStruct);
+
+ /* Enable the DMA clock */
+ AUDIO_SAIx_DMAx_CLK_ENABLE();
+
+ if(hsai->Instance == AUDIO_SAIx)
+ {
+ /* Configure the hDmaSai handle parameters */
+ hDmaSai.Init.Request = DMA_REQUEST_1;
+ hDmaSai.Init.Direction = DMA_MEMORY_TO_PERIPH;
+ hDmaSai.Init.PeriphInc = DMA_PINC_DISABLE;
+ hDmaSai.Init.MemInc = DMA_MINC_ENABLE;
+ hDmaSai.Init.PeriphDataAlignment = AUDIO_SAIx_DMAx_PERIPH_DATA_SIZE;
+ hDmaSai.Init.MemDataAlignment = AUDIO_SAIx_DMAx_MEM_DATA_SIZE;
+ hDmaSai.Init.Mode = DMA_NORMAL;
+ hDmaSai.Init.Priority = DMA_PRIORITY_HIGH;
+
+ hDmaSai.Instance = AUDIO_SAIx_DMAx_CHANNEL;
+
+ /* Associate the DMA handle */
+ __HAL_LINKDMA(hsai, hdmatx, hDmaSai);
+
+ /* Deinitialize the Stream for new transfer */
+ HAL_DMA_DeInit(&hDmaSai);
+
+ /* Configure the DMA Stream */
+ HAL_DMA_Init(&hDmaSai);
+ }
+
+ /* SAI DMA IRQ Channel configuration */
+ HAL_NVIC_SetPriority(AUDIO_SAIx_DMAx_IRQ, AUDIO_OUT_IRQ_PREPRIO, 0);
+ HAL_NVIC_EnableIRQ(AUDIO_SAIx_DMAx_IRQ);
+}
+
+/**
+ * @brief SAI MSP De-init
+ * @param hsai : pointer to a SAI_HandleTypeDef structure
+ * @retval None
+ */
+void HAL_SAI_MspDeInit(SAI_HandleTypeDef *hsai)
+{
+ /* Disable SAI DMA Channel IRQ */
+ HAL_NVIC_DisableIRQ(AUDIO_SAIx_DMAx_IRQ);
+
+ /* Reset the DMA Stream configuration*/
+ HAL_DMA_DeInit(&hDmaSai);
+
+ /* Disable the DMA clock */
+ AUDIO_SAIx_DMAx_CLK_DISABLE();
+
+ /* De-initialize FS, SCK, MCK and SD pins*/
+ HAL_GPIO_DeInit(AUDIO_SAIx_MCK_SCK_SD_FS_GPIO_PORT,
+ AUDIO_SAIx_FS_PIN | AUDIO_SAIx_SCK_PIN | AUDIO_SAIx_SD_PIN | AUDIO_SAIx_MCK_PIN);
+
+ /* Disable GPIO clock */
+ AUDIO_SAIx_MCK_SCK_SD_FS_DISABLE();
+
+ /* Disable SAI clock */
+ AUDIO_SAIx_CLK_DISABLE();
+}
+
+/**
+ * @brief Resets the audio codec. It restores the default configuration of the
+ * codec (this function shall be called before initializing the codec).
+ * @retval None
+ */
+
+
+/**
+ * @}
+ */
+
+/** @addtogroup STM32L476G_DISCOVERY_AUDIO_Private_Functions
+ * @{
+ */
+
+/**
+ * @brief Initializes the Digital Filter for Sigma-Delta Modulators interface (DFSDM).
+ * @param AudioFreq: Audio frequency to be used to set correctly the DFSDM peripheral.
+ * @retval BSP AUDIO status
+ */
+static uint8_t AUDIO_DFSDMx_Init(uint32_t AudioFreq)
+{
+ /*####CHANNEL 2####*/
+ hAudioIn.hDfsdmLeftChannel.Init.OutputClock.Activation = ENABLE;
+ hAudioIn.hDfsdmLeftChannel.Init.OutputClock.Selection = DFSDM_CHANNEL_OUTPUT_CLOCK_AUDIO;
+ /* Set the DFSDM clock OUT audio frequency configuration */
+ hAudioIn.hDfsdmLeftChannel.Init.OutputClock.Divider = DFSDMClockDivider(AudioFreq);
+ hAudioIn.hDfsdmLeftChannel.Init.Input.Multiplexer = DFSDM_CHANNEL_EXTERNAL_INPUTS;
+ hAudioIn.hDfsdmLeftChannel.Init.Input.DataPacking = DFSDM_CHANNEL_STANDARD_MODE;
+ hAudioIn.hDfsdmLeftChannel.Init.Input.Pins = DFSDM_CHANNEL_SAME_CHANNEL_PINS;
+ /* Request to sample stable data for LEFT micro on Rising edge */
+ hAudioIn.hDfsdmLeftChannel.Init.SerialInterface.Type = DFSDM_CHANNEL_SPI_RISING;
+ hAudioIn.hDfsdmLeftChannel.Init.SerialInterface.SpiClock = DFSDM_CHANNEL_SPI_CLOCK_INTERNAL;
+ hAudioIn.hDfsdmLeftChannel.Init.Awd.FilterOrder = DFSDM_CHANNEL_SINC1_ORDER;
+ hAudioIn.hDfsdmLeftChannel.Init.Awd.Oversampling = 10;
+ hAudioIn.hDfsdmLeftChannel.Init.Offset = 0;
+ hAudioIn.hDfsdmLeftChannel.Init.RightBitShift = DFSDMRightBitShift(AudioFreq);
+
+ hAudioIn.hDfsdmLeftChannel.Instance = DFSDM_Channel2;
+
+ /* Init the DFSDM Channel */
+ if (HAL_DFSDM_ChannelInit(&hAudioIn.hDfsdmLeftChannel) != HAL_OK)
+ {
+ return AUDIO_ERROR;
+ }
+
+ /*####FILTER 0####*/
+ BSP_AUDIO_hDfsdmLeftFilter.Init.RegularParam.Trigger = DFSDM_FILTER_SW_TRIGGER;
+ BSP_AUDIO_hDfsdmLeftFilter.Init.RegularParam.FastMode = ENABLE;
+ BSP_AUDIO_hDfsdmLeftFilter.Init.RegularParam.DmaMode = ENABLE;
+ BSP_AUDIO_hDfsdmLeftFilter.Init.InjectedParam.Trigger = DFSDM_FILTER_SW_TRIGGER;
+ BSP_AUDIO_hDfsdmLeftFilter.Init.InjectedParam.ScanMode = DISABLE;
+ BSP_AUDIO_hDfsdmLeftFilter.Init.InjectedParam.DmaMode = DISABLE;
+ BSP_AUDIO_hDfsdmLeftFilter.Init.InjectedParam.ExtTrigger = DFSDM_FILTER_EXT_TRIG_TIM8_TRGO;
+ BSP_AUDIO_hDfsdmLeftFilter.Init.InjectedParam.ExtTriggerEdge = DFSDM_FILTER_EXT_TRIG_BOTH_EDGES;
+ BSP_AUDIO_hDfsdmLeftFilter.Init.FilterParam.SincOrder = DFSDMFilterOrder(AudioFreq);
+ /* Set the DFSDM Filters Oversampling to have correct sample rate */
+ BSP_AUDIO_hDfsdmLeftFilter.Init.FilterParam.Oversampling = DFSDMOverSampling(AudioFreq);
+ BSP_AUDIO_hDfsdmLeftFilter.Init.FilterParam.IntOversampling = 1;
+
+ BSP_AUDIO_hDfsdmLeftFilter.Instance = AUDIO_DFSDMx_LEFT_FILTER;
+
+ /* Init the DFSDM Filter */
+ if (HAL_DFSDM_FilterInit(&BSP_AUDIO_hDfsdmLeftFilter) != HAL_OK)
+ {
+ return AUDIO_ERROR;
+ }
+
+ /* Configure regular channel */
+ if (HAL_DFSDM_FilterConfigRegChannel(&BSP_AUDIO_hDfsdmLeftFilter,
+ DFSDM_CHANNEL_2,
+ DFSDM_CONTINUOUS_CONV_ON) != HAL_OK)
+ {
+ return AUDIO_ERROR;
+ }
+
+ return AUDIO_OK;
+}
+
+/**
+ * @brief De-initializes the Digital Filter for Sigma-Delta Modulators interface (DFSDM).
+ * @retval BSP AUDIO status
+ */
+static uint8_t AUDIO_DFSDMx_DeInit(void)
+{
+ /* De-initializes the DFSDM filters to allow access to DFSDM internal registers */
+ if (HAL_DFSDM_FilterDeInit(&BSP_AUDIO_hDfsdmLeftFilter) != HAL_OK)
+ {
+ return AUDIO_ERROR;
+ }
+
+ /* De-initializes the DFSDM channels to allow access to DFSDM internal registers */
+ if (HAL_DFSDM_ChannelDeInit(&hAudioIn.hDfsdmLeftChannel) != HAL_OK)
+ {
+ return AUDIO_ERROR;
+ }
+
+ /* Disable DFSDM clock */
+ AUDIO_DFSDMx_CLK_DISABLE();
+
+ /* Disable SAIx PLL */
+ if (AUDIO_SAIx_PLL_DISABLE() != AUDIO_OK)
+ {
+ return AUDIO_ERROR;
+ }
+
+ /* DFSDM reset */
+ __HAL_RCC_DFSDM_FORCE_RESET();
+ __HAL_RCC_DFSDM_RELEASE_RESET();
+
+ return AUDIO_OK;
+}
+
+/**
+ * @brief Initializes the DFSDM channel MSP.
+ * @param hdfsdm_channel : DFSDM channel handle.
+ * @retval None
+ */
+void HAL_DFSDM_ChannelMspInit(DFSDM_Channel_HandleTypeDef *hdfsdm_channel)
+{
+ GPIO_InitTypeDef GPIO_InitStruct;
+
+ /* Enable DFSDM clock */
+ AUDIO_DFSDMx_CLK_ENABLE();
+
+ /* Enable GPIO clock */
+ AUDIO_DFSDMx_CKOUT_DMIC_DATIN_GPIO_CLK_ENABLE();
+
+ /* DFSDM pins configuration: DFSDM_CKOUT, DMIC_DATIN pins ------------------*/
+ GPIO_InitStruct.Pin = AUDIO_DFSDMx_CKOUT_PIN | AUDIO_DFSDMx_DMIC_DATIN_PIN;
+ GPIO_InitStruct.Mode = GPIO_MODE_AF_PP;
+ GPIO_InitStruct.Pull = GPIO_NOPULL;
+ GPIO_InitStruct.Speed = GPIO_SPEED_FREQ_VERY_HIGH;
+ GPIO_InitStruct.Alternate = AUDIO_DFSDMx_CKOUT_DMIC_DATIN_AF;
+ HAL_GPIO_Init(AUDIO_DFSDMx_CKOUT_DMIC_DATIN_GPIO_PORT, &GPIO_InitStruct);
+}
+
+/**
+ * @brief De-initializes the DFSDM channel MSP.
+ * @param hdfsdm_channel : DFSDM channel handle.
+ * @retval None
+ */
+void HAL_DFSDM_ChannelMspDeInit(DFSDM_Channel_HandleTypeDef *hdfsdm_channel)
+{
+ GPIO_InitTypeDef GPIO_InitStruct;
+
+ /* Enable GPIO clock */
+ AUDIO_DFSDMx_CKOUT_DMIC_DATIN_GPIO_CLK_ENABLE();
+
+ /* DFSDM pins configuration: DFSDM_CKOUT */
+ GPIO_InitStruct.Pin = AUDIO_DFSDMx_CKOUT_PIN;
+ GPIO_InitStruct.Mode = GPIO_MODE_OUTPUT_PP;
+ GPIO_InitStruct.Pull = GPIO_NOPULL;
+ GPIO_InitStruct.Speed = GPIO_SPEED_FREQ_LOW;
+ HAL_GPIO_Init(AUDIO_DFSDMx_CKOUT_DMIC_DATIN_GPIO_PORT, &GPIO_InitStruct);
+ HAL_GPIO_WritePin(AUDIO_DFSDMx_CKOUT_DMIC_DATIN_GPIO_PORT, AUDIO_DFSDMx_CKOUT_PIN, GPIO_PIN_RESET);
+
+
+ /* De-initialize DMIC_DATIN pin */
+ HAL_GPIO_DeInit(AUDIO_DFSDMx_CKOUT_DMIC_DATIN_GPIO_PORT, AUDIO_DFSDMx_DMIC_DATIN_PIN);
+}
+
+/**
+ * @brief Initializes the DFSDM filter MSP.
+ * @param hdfsdm_filter : DFSDM filter handle.
+ * @retval None
+ */
+void HAL_DFSDM_FilterMspInit(DFSDM_Filter_HandleTypeDef *hdfsdm_filter)
+{
+ /* Enable DFSDM clock */
+ AUDIO_DFSDMx_CLK_ENABLE();
+
+ /* Enable the DMA clock */
+ AUDIO_DFSDMx_DMAx_CLK_ENABLE();
+
+ /* Configure the hAudioIn.hDmaDfsdmLeft handle parameters */
+ hAudioIn.hDmaDfsdmLeft.Init.Request = DMA_REQUEST_0;
+ hAudioIn.hDmaDfsdmLeft.Init.Direction = DMA_PERIPH_TO_MEMORY;
+ hAudioIn.hDmaDfsdmLeft.Init.PeriphInc = DMA_PINC_DISABLE;
+ hAudioIn.hDmaDfsdmLeft.Init.MemInc = DMA_MINC_ENABLE;
+ hAudioIn.hDmaDfsdmLeft.Init.PeriphDataAlignment = AUDIO_DFSDMx_DMAx_PERIPH_DATA_SIZE;
+ hAudioIn.hDmaDfsdmLeft.Init.MemDataAlignment = AUDIO_DFSDMx_DMAx_MEM_DATA_SIZE;
+ hAudioIn.hDmaDfsdmLeft.Init.Mode = DMA_CIRCULAR;
+ hAudioIn.hDmaDfsdmLeft.Init.Priority = DMA_PRIORITY_HIGH;
+
+ hAudioIn.hDmaDfsdmLeft.Instance = AUDIO_DFSDMx_DMAx_LEFT_CHANNEL;
+
+ /* Associate the DMA handle */
+ __HAL_LINKDMA(hdfsdm_filter, hdmaReg, hAudioIn.hDmaDfsdmLeft);
+
+ /* Reset DMA handle state */
+ __HAL_DMA_RESET_HANDLE_STATE(&hAudioIn.hDmaDfsdmLeft);
+
+ /* Configure the DMA Channel */
+ HAL_DMA_Init(&hAudioIn.hDmaDfsdmLeft);
+
+ /* DMA IRQ Channel configuration */
+ HAL_NVIC_SetPriority(AUDIO_DFSDMx_DMAx_LEFT_IRQ, AUDIO_OUT_IRQ_PREPRIO, 0);
+ HAL_NVIC_EnableIRQ(AUDIO_DFSDMx_DMAx_LEFT_IRQ);
+ }
+
+ /**
+ * @brief De-initializes the DFSDM filter MSP.
+ * @param hdfsdm_filter : DFSDM filter handle.
+ * @retval None
+ */
+void HAL_DFSDM_FilterMspDeInit(DFSDM_Filter_HandleTypeDef *hdfsdm_filter)
+{
+ /* Disable DMA Channel IRQ */
+ HAL_NVIC_DisableIRQ(AUDIO_DFSDMx_DMAx_LEFT_IRQ);
+
+ /* De-initialize the DMA Channel */
+ HAL_DMA_DeInit(&hAudioIn.hDmaDfsdmLeft);
+
+ /* Disable the DMA clock */
+ AUDIO_DFSDMx_DMAx_CLK_DISABLE();
+}
+
+/**
+ * @brief Configures the SAI PLL clock according to the required audio frequency.
+ * @param Frequency: Audio frequency.
+ * @retval BSP AUDIO status
+ * @note The SAI PLL input clock must be configured in the user application.
+ * The SAI PLL configuration done within this function assumes that
+ * the SAI PLL input clock runs at 8 MHz.
+ */
+static uint8_t AUDIO_SAIPLLConfig(uint32_t Frequency)
+{
+ RCC_PeriphCLKInitTypeDef RCC_ExCLKInitStruct;
+
+ /* Retreive actual RCC configuration */
+ HAL_RCCEx_GetPeriphCLKConfig(&RCC_ExCLKInitStruct);
+
+ if ( (Frequency == AUDIO_FREQUENCY_11K)
+ || (Frequency == AUDIO_FREQUENCY_22K)
+ || (Frequency == AUDIO_FREQUENCY_44K) )
+ {
+ /* Configure PLLSAI prescalers */
+ /* SAI clock config
+ PLLSAI1_VCO= 8 Mhz * PLLSAI1N = 8 * 24 = VCO_192M
+ SAI_CK_x = PLLSAI1_VCO/PLLSAI1P = 192/17 = 11.294 Mhz */
+ RCC_ExCLKInitStruct.PeriphClockSelection = RCC_PERIPHCLK_SAI1;
+ RCC_ExCLKInitStruct.PLLSAI1.PLLSAI1N = 24;
+ RCC_ExCLKInitStruct.PLLSAI1.PLLSAI1P = 17;
+ RCC_ExCLKInitStruct.PLLSAI1.PLLSAI1ClockOut = RCC_PLLSAI1_SAI1CLK;
+ RCC_ExCLKInitStruct.Sai1ClockSelection = RCC_SAI1CLKSOURCE_PLLSAI1;
+ }
+ else /* AUDIO_FREQUENCY_8K, AUDIO_FREQUENCY_16K, AUDIO_FREQUENCY_48K, AUDIO_FREQUENCY_96K */
+ {
+ /* SAI clock config
+ PLLSAI1_VCO= 8 Mhz * PLLSAI1N = 8 * 43 = VCO_344M
+ SAI_CK_x = PLLSAI1_VCO/PLLSAI1P = 344/7 = 49.142 Mhz */
+ RCC_ExCLKInitStruct.PeriphClockSelection = RCC_PERIPHCLK_SAI1;
+ RCC_ExCLKInitStruct.PLLSAI1.PLLSAI1N = 43;
+ RCC_ExCLKInitStruct.PLLSAI1.PLLSAI1P = 7;
+ RCC_ExCLKInitStruct.PLLSAI1.PLLSAI1ClockOut = RCC_PLLSAI1_SAI1CLK;
+ RCC_ExCLKInitStruct.Sai1ClockSelection = RCC_SAI1CLKSOURCE_PLLSAI1;
+ }
+
+ if (HAL_RCCEx_PeriphCLKConfig(&RCC_ExCLKInitStruct) != HAL_OK)
+ {
+ return AUDIO_ERROR;
+ }
+
+ return AUDIO_OK;
+}
+
+/**
+ * @}
+ */
+
+/**
+ * @}
+ */
+
+/**
+ * @}
+ */
+
+/**
+ * @}
+ */
+
+/**
+ * @}
+ */
+
+/************************ (C) COPYRIGHT STMicroelectronics *****END OF FILE****/
+
+