stm32h743i_eval_audio.c File Reference

This file provides the Audio driver for the STM32H743I-EVAL board. More...

#include "stm32h743i_eval_audio.h"

Go to the source code of this file.

Defines
#define	DFSDM_OVER_SAMPLING(__FREQUENCY__)
#define	DFSDM_CLOCK_DIVIDER(__FREQUENCY__)
#define	DFSDM_FILTER_ORDER(__FREQUENCY__)
#define	DFSDM_MIC_BIT_SHIFT(__FREQUENCY__)
#define	SaturaLH(N, L, H)   (((N)<(L))?(L):(((N)>(H))?(H):(N)))
Functions
static void	SAIx_Out_Init (uint32_t SaiOutMode, uint32_t SlotActive, uint32_t AudioFreq)
	Initializes the Audio Codec audio interface (SAI).
static void	SAIx_Out_DeInit (SAI_HandleTypeDef *hsai)
	Deinitializes the Audio Codec audio interface (SAI).
static void	SAIx_In_MspInit (SAI_HandleTypeDef *hsai, void *Params)
	Initializes SAI Audio IN MSP.
static void	SAIx_In_MspDeInit (SAI_HandleTypeDef *hsai, void *Params)
	De-Initializes SAI Audio IN MSP.
static void	SAIx_In_Init (uint32_t SaiInMode, uint32_t SlotActive, uint32_t AudioFreq)
	Initializes the Audio Codec audio interface (SAI).
static void	SAIx_In_DeInit (SAI_HandleTypeDef *hsai)
	Deinitializes the output Audio Codec audio interface (SAI).
static uint8_t	DFSDMx_DeInit (void)
	De-initializes the Digital Filter for Sigma-Delta Modulators interface (DFSDM).
static void	DFSDMx_ChannelMspInit (void)
	Initializes the DFSDM channel MSP.
static void	DFSDMx_ChannelMspDeInit (void)
	DeInitializes the DFSDM channel MSP.
static void	DFSDMx_FilterMspInit (void)
	Initializes the DFSDM filter MSP.
static void	DFSDMx_FilterMspDeInit (void)
	DeInitializes the DFSDM filter MSP.
uint8_t	BSP_AUDIO_OUT_Init (uint16_t OutputDevice, uint8_t Volume, uint32_t AudioFreq)
	Configures the audio Out peripheral.
uint8_t	BSP_AUDIO_OUT_Play (uint16_t *pBuffer, uint32_t Size)
	Starts playing audio stream from a data buffer for a determined size.
void	BSP_AUDIO_OUT_ChangeBuffer (uint16_t *pData, uint16_t Size)
	Sends n-Bytes on the SAI interface.
uint8_t	BSP_AUDIO_OUT_Pause (void)
	This function Pauses the audio file stream.
uint8_t	BSP_AUDIO_OUT_Resume (void)
	Resumes the audio file stream.
uint8_t	BSP_AUDIO_OUT_Stop (uint32_t Option)
	Stops audio playing and Power down the Audio Codec.
uint8_t	BSP_AUDIO_OUT_SetVolume (uint8_t Volume)
	Controls the current audio volume level.
uint8_t	BSP_AUDIO_OUT_SetMute (uint32_t Cmd)
	Enables or disables the MUTE mode by software.
uint8_t	BSP_AUDIO_OUT_SetOutputMode (uint8_t Output)
	Switch dynamically (while audio file is played) the output target (speaker or headphone).
void	BSP_AUDIO_OUT_SetFrequency (uint32_t AudioFreq)
	Updates the audio frequency.
void	BSP_AUDIO_OUT_SetAudioFrameSlot (uint32_t AudioFrameSlot)
	Updates the Audio frame slot configuration.
void	BSP_AUDIO_OUT_DeInit (void)
	De-initializes the audio out peripheral.
__weak void	BSP_AUDIO_OUT_TransferComplete_CallBack (void)
	Manages the DMA full Transfer complete event.
__weak void	BSP_AUDIO_OUT_HalfTransfer_CallBack (void)
	Manages the DMA Half Transfer complete event.
__weak void	BSP_AUDIO_OUT_Error_CallBack (void)
	Manages the DMA FIFO error event.
__weak void	BSP_AUDIO_OUT_MspInit (SAI_HandleTypeDef *hsai, void *Params)
	Initializes BSP_AUDIO_OUT MSP.
__weak void	BSP_AUDIO_OUT_MspDeInit (SAI_HandleTypeDef *hsai, void *Params)
	Deinitializes SAI MSP.
__weak void	BSP_AUDIO_OUT_ClockConfig (SAI_HandleTypeDef *hsai, uint32_t AudioFreq, void *Params)
	Clock Config.
void	HAL_SAI_TxCpltCallback (SAI_HandleTypeDef *hsai)
	Tx Transfer completed callbacks.
void	HAL_SAI_TxHalfCpltCallback (SAI_HandleTypeDef *hsai)
	Tx Half Transfer completed callbacks.
void	HAL_SAI_ErrorCallback (SAI_HandleTypeDef *hsai)
	SAI error callbacks.
uint8_t	BSP_AUDIO_IN_Init (uint32_t AudioFreq, uint32_t BitRes, uint32_t ChnlNbr)
	Initialize wave recording.
uint8_t	BSP_AUDIO_IN_InitEx (uint16_t InputDevice, uint32_t AudioFreq, uint32_t BitRes, uint32_t ChnlNbr)
	Initialize wave recording.
uint8_t	BSP_AUDIO_IN_ConfigMicDefault (uint32_t InputDevice)
	Initializes default configuration of the Digital Filter for Sigma-Delta Modulators interface (DFSDM).
__weak uint8_t	BSP_AUDIO_IN_ConfigDigitalMic (uint32_t InputDevice, void *Params)
	Initializes the Digital Filter for Sigma-Delta Modulators interface (DFSDM).
uint8_t	BSP_AUDIO_IN_OUT_Init (uint32_t InputDevice, uint32_t OutputDevice, uint32_t AudioFreq, uint32_t BitRes, uint32_t ChnlNbr)
	Initializes wave recording and playback in parallel.
void	BSP_AUDIO_IN_SelectInterface (uint32_t Interface)
	Link digital mic to specified source.
uint32_t	BSP_AUDIO_IN_GetInterface (void)
	Get digital mic interface.
uint8_t	BSP_AUDIO_IN_AllocScratch (int32_t *pScratch, uint32_t size)
	Allocate channel buffer scratch.
uint8_t	BSP_AUDIO_IN_GetChannelNumber (void)
	Return audio in channel number.
uint8_t	BSP_AUDIO_IN_Record (uint16_t *pBuf, uint32_t size)
	Start audio recording.
uint8_t	BSP_AUDIO_IN_Stop (void)
	Stop audio recording.
uint8_t	BSP_AUDIO_IN_StopEx (uint32_t InputDevice)
	Stops audio recording.
uint8_t	BSP_AUDIO_IN_Pause (void)
	Pause the audio file stream.
uint8_t	BSP_AUDIO_IN_PauseEx (uint32_t InputDevice)
	Pauses the audio file stream.
uint8_t	BSP_AUDIO_IN_Resume (void)
	Resume the audio file stream.
uint8_t	BSP_AUDIO_IN_ResumeEx (uint32_t *pBuf, uint32_t InputDevice)
	Resumes the audio file stream.
uint8_t	BSP_AUDIO_IN_SetVolume (uint8_t Volume)
	Controls the audio in volume level.
void	BSP_AUDIO_IN_DeInit (void)
	Deinit the audio IN peripherals.
uint8_t	BSP_AUDIO_IN_PDMToPCM_Init (uint32_t AudioFreq, uint32_t ChnlNbrIn, uint32_t ChnlNbrOut)
	Initialize the PDM library.
uint8_t	BSP_AUDIO_IN_PDMToPCM (uint16_t *PDMBuf, uint16_t *PCMBuf)
	Converts audio format from PDM to PCM.
__weak void	BSP_AUDIO_IN_TransferComplete_CallBack (void)
	User callback when record buffer is filled.
__weak void	BSP_AUDIO_IN_HalfTransfer_CallBack (void)
	Manages the DMA Half Transfer complete event.
__weak void	BSP_AUDIO_IN_TransferComplete_CallBackEx (uint32_t InputDevice)
	User callback when record buffer is filled.
__weak void	BSP_AUDIO_IN_HalfTransfer_CallBackEx (uint32_t InputDevice)
	User callback when record buffer is filled.
__weak void	BSP_AUDIO_IN_Error_CallBack (void)
	Audio IN Error callback function.
__weak void	BSP_AUDIO_IN_MspInit (void)
	Initialize BSP_AUDIO_IN MSP.
__weak void	BSP_AUDIO_IN_MspDeInit (void)
	DeInitialize BSP_AUDIO_IN MSP.
__weak void	BSP_AUDIO_IN_ClockConfig (uint32_t AudioFreq, void *Params)
	Clock Config.
void	HAL_DFSDM_FilterRegConvCpltCallback (DFSDM_Filter_HandleTypeDef *hdfsdm_filter)
	Regular conversion complete callback.
void	HAL_DFSDM_FilterRegConvHalfCpltCallback (DFSDM_Filter_HandleTypeDef *hdfsdm_filter)
	Half regular conversion complete callback.
void	HAL_SAI_RxHalfCpltCallback (SAI_HandleTypeDef *hsai)
	Half reception complete callback.
void	HAL_SAI_RxCpltCallback (SAI_HandleTypeDef *hsai)
	Reception complete callback.
Variables
AUDIO_DrvTypeDef *	audio_drv
SAI_HandleTypeDef	haudio_out_sai
SAI_HandleTypeDef	haudio_in_sai
AUDIOIN_ContextTypeDef	hAudioIn
DFSDM_Channel_HandleTypeDef	hAudioInDfsdmChannel [DFSDM_MIC_NUMBER]
DFSDM_Filter_HandleTypeDef	hAudioInDfsdmFilter [DFSDM_MIC_NUMBER]
DMA_HandleTypeDef	hDmaDfsdm [DFSDM_MIC_NUMBER]
int32_t *	pScratchBuff [DEFAULT_AUDIO_IN_CHANNEL_NBR]
int32_t	ScratchSize
uint32_t	DmaRecHalfBuffCplt [DFSDM_MIC_NUMBER] = {0}
uint32_t	DmaRecBuffCplt [DFSDM_MIC_NUMBER] = {0}
__IO uint32_t	AppBuffTrigger = 0
__IO uint32_t	AppBuffHalf = 0
__IO uint32_t	MicBuff [DFSDM_MIC_NUMBER] = {0}
__IO uint16_t	AudioInVolume = DEFAULT_AUDIO_IN_VOLUME
PDM_Filter_Handler_t	PDM_FilterHandler [2]
PDM_Filter_Config_t	PDM_FilterConfig [2]

---

Detailed Description

This file provides the Audio driver for the STM32H743I-EVAL board.

Author:

MCD Application Team

Version:

V1.0.0

Date:

21-April-2017

  How To use this driver:
  -----------------------
   + This driver supports STM32H7xx devices on STM32H743I-EVAL (MB1246) Evaluation boards.
   + Call the function BSP_AUDIO_OUT_Init(
                                    OutputDevice: physical output mode (OUTPUT_DEVICE_SPEAKER,
                                                  OUTPUT_DEVICE_HEADPHONE or OUTPUT_DEVICE_BOTH)
                                    Volume      : Initial volume to be set (0 is min (mute), 100 is max (100%)
                                    AudioFreq   : Audio frequency in Hz (8000, 16000, 22500, 32000...)
                                                  this parameter is relative to the audio file/stream type.
                                   )
      This function configures all the hardware required for the audio application (codec, I2C, SAI,
      GPIOs, DMA and interrupt if needed). This function returns AUDIO_OK if configuration is OK.
      If the returned value is different from AUDIO_OK or the function is stuck then the communication with
      the codec or the MFX has failed (try to un-plug the power or reset device in this case).
      - OUTPUT_DEVICE_SPEAKER  : only speaker will be set as output for the audio stream.
      - OUTPUT_DEVICE_HEADPHONE: only headphones will be set as output for the audio stream.
      - OUTPUT_DEVICE_BOTH     : both Speaker and Headphone are used as outputs for the audio stream
                                 at the same time.
      Note. On STM32H743I-EVAL 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 stm32h743i_eval_audio.h file. (refer to the example for more details on the callbacks implementations)
   + To Stop playing, to modify the volume level, the frequency, the audio frame slot,
      the device output mode the mute or the stop, use the functions: BSP_AUDIO_OUT_SetVolume(),
      AUDIO_OUT_SetFrequency(), BSP_AUDIO_OUT_SetAudioFrameSlot(), BSP_AUDIO_OUT_SetOutputMode(),
      BSP_AUDIO_OUT_SetMute() and BSP_AUDIO_OUT_Stop().

   + Call the function BSP_AUDIO_IN_Init(
                                    AudioFreq: Audio frequency in Hz (8000, 16000, 22500, 32000...)
                                                  this parameter is relative to the audio file/stream type.
                                    BitRes: Bit resolution fixed to 16bit
                                    ChnlNbr: Number of channel to be configured for the DFSDM peripheral
                                   )
      This function configures all the hardware required for the audio in application (DFSDM filters and channels,
      Clock source for DFSDM periphiral, GPIOs, DMA and interrupt if needed).
      This function returns AUDIO_OK if configuration is OK.If the returned value is different from AUDIO_OK then
      the configuration should be wrong.
      Note: On STM32H743I-EVAL, two DFSDM Channel/Filters are configured and their DMA streams are configured
            in CIRCULAR mode.
   + Call the function BSP_AUDIO_IN_AllocScratch(
                                        pScratch: pointer to scratch tables
                                        size: size of scratch buffer)
     This function must be called before BSP_AUDIO_IN_RECORD() to allocate buffer scratch for each DFSDM channel
     and its size.
     Note: These buffers scratch are used as intermidiate buffers to collect data within final record buffer.
           size is the total size of the four buffers scratch; If size is 512 then the size of each is 128.
           This function must be called after BSP_AUDIO_IN_Init()
   + Call the function BSP_AUDIO_IN_RECORD(
                                  pBuf: pointer to the recorded audio data file address
                                  Size: size of the buffer to be written in Bytes
                                 )
      to start recording from microphones.

   + Call the function BSP_AUDIO_IN_Pause() to pause recording
   + Call the function BSP_AUDIO_IN_Resume() to recording playing.
       Note. After calling BSP_AUDIO_IN_Pause() function for pause, only BSP_AUDIO_IN_Resume() should be called
          for resume (it is not allowed to call BSP_AUDIO_IN_RECORD() in this case).
   + Call the function BSP_AUDIO_IN_Stop() to stop recording
   + For each mode, you may need to implement the relative callback functions into your code.
      The Callback functions are named BSP_AUDIO_IN_XXX_CallBack() and only their prototypes are declared in
      the stm32h743i_eval_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 three interfaces
     available on the STM32H743I-EVAL board: SAI, DFSDM 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 stm32h743i_eval_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 stm32h743i_eval_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.
  

Attention:

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Definition in file stm32h743i_eval_audio.c.