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/**
******************************************************************************
* @file stm32412g_discovery_audio.c
* @author MCD Application Team
* @brief This file provides the Audio driver for the STM32412G-DISCOVERY board.
******************************************************************************
* @attention
*
* Copyright (c) 2017 STMicroelectronics.
* All rights reserved.
*
* This software is licensed under terms that can be found in the LICENSE file
* in the root directory of this software component.
* If no LICENSE file comes with this software, it is provided AS-IS.
*
******************************************************************************
*/
/*==============================================================================
User NOTES
How To use this driver:
-----------------------
+ This driver supports STM32F4xx devices on STM32412G-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, I2S,
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.
+ 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 AUDIO_OUT_XXX_CallBack() and only their prototypes are declared in
the stm32412g_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, use the functions: BSP_AUDIO_OUT_SetVolume(),
AUDIO_OUT_SetFrequency(), BSP_AUDIO_OUT_SetOutputMode(), BSP_AUDIO_OUT_SetMute() and BSP_AUDIO_OUT_Stop().
+ The driver API and the callback functions are at the end of the stm32412g_discovery_audio.h file.
Driver architecture:
--------------------
+ This driver provides the High Audio Layer: consists of the function API exported in the stm32412g_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 stm32412g_discovery_audio.c file (I2Sx_Out_Init(), I2Sx_Out_DeInit(), I2Sx_In_Init() and I2Sx_In_DeInit())
Known Limitations:
------------------
1- If the TDM Format used to play in parallel 2 audio Stream (the first Stream is configured in codec SLOT0 and second
Stream in SLOT1) the Pause/Resume, volume and mute feature will control the both streams.
2- Parsing of audio file is not implemented (in order to determine audio file properties: Mono/Stereo, Data size,
File size, Audio Frequency, Audio Data header size ...). The configuration is fixed for the given audio file.
3- Supports only Stereo audio streaming.
4- Supports only 16-bits audio data size.
==============================================================================*/
/* Includes ------------------------------------------------------------------*/
#include "stm32412g_discovery_audio.h"
/** @addtogroup BSP
* @{
*/
/** @addtogroup STM32412G_DISCOVERY
* @{
*/
/** @defgroup STM32412G_DISCOVERY_AUDIO STM32412G-DISCOVERY AUDIO
* @brief This file includes the low layer driver for wm8994 Audio Codec
* available on STM32412G-DISCOVERY board(MB1209).
* @{
*/
/** @defgroup STM32412G_DISCOVERY_AUDIO_Private_Types STM32412G Discovery Audio Private Types
* @{
*/
/**
* @}
*/
/** @defgroup STM32412G_DISCOVERY_AUDIO_Private_Defines STM32412G Discovery Audio Private Defines
* @{
*/
/**
* @}
*/
/** @defgroup STM32412G_DISCOVERY_AUDIO_Private_Macros STM32412G Discovery Audio Private macros
* @{
*/
#define DFSDM_OVER_SAMPLING(__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 : 25 \
#define DFSDM_CLOCK_DIVIDER(__FREQUENCY__) \
(__FREQUENCY__ == AUDIO_FREQUENCY_8K) ? 24 \
: (__FREQUENCY__ == AUDIO_FREQUENCY_11K) ? 48 \
: (__FREQUENCY__ == AUDIO_FREQUENCY_16K) ? 24 \
: (__FREQUENCY__ == AUDIO_FREQUENCY_22K) ? 48 \
: (__FREQUENCY__ == AUDIO_FREQUENCY_32K) ? 24 \
: (__FREQUENCY__ == AUDIO_FREQUENCY_44K) ? 48 \
: (__FREQUENCY__ == AUDIO_FREQUENCY_48K) ? 32 : 72 \
#define DFSDM_FILTER_ORDER(__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_SINC4_ORDER \
#define DFSDM_MIC_BIT_SHIFT(__FREQUENCY__) \
(__FREQUENCY__ == AUDIO_FREQUENCY_8K) ? 5 \
: (__FREQUENCY__ == AUDIO_FREQUENCY_11K) ? 4 \
: (__FREQUENCY__ == AUDIO_FREQUENCY_16K) ? 2 \
: (__FREQUENCY__ == AUDIO_FREQUENCY_22K) ? 2 \
: (__FREQUENCY__ == AUDIO_FREQUENCY_32K) ? 5 \
: (__FREQUENCY__ == AUDIO_FREQUENCY_44K) ? 6 \
: (__FREQUENCY__ == AUDIO_FREQUENCY_48K) ? 2 : 0 \
/* Saturate the record PCM sample */
#define SaturaLH(N, L, H) (((N)<(L))?(L):(((N)>(H))?(H):(N)))
/**
* @}
*/
/** @defgroup STM32412G_DISCOVERY_AUDIO_Private_Variables STM32412G Discovery Audio Private Variables
* @{
*/
AUDIO_DrvTypeDef *audio_drv;
I2S_HandleTypeDef haudio_i2s; /* for Audio_OUT and Audio_IN_analog mic */
I2S_HandleTypeDef haudio_in_i2sext; /* for Analog mic with full duplex mode */
AUDIOIN_ContextTypeDef hAudioIn;
DFSDM_Channel_HandleTypeDef hAudioInDfsdmChannel[DFSDM_MIC_NUMBER]; /* 2 DFSDM channel handle used for all microphones */
DFSDM_Filter_HandleTypeDef hAudioInDfsdmFilter[DFSDM_MIC_NUMBER]; /* 2 DFSDM filter handle */
DMA_HandleTypeDef hDmaDfsdm[DFSDM_MIC_NUMBER]; /* 2 DMA handle used for DFSDM regular conversions */
/* Buffers for right and left samples */
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};
/* Application Buffer Trigger */
__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;
/**
* @}
*/
/** @defgroup STM32412G_DISCOVERY_AUDIO_Private_Function_Prototypes STM32412G Discovery Audio Private Prototypes
* @{
*/
static void I2Sx_In_Init(uint32_t AudioFreq);
static void I2Sx_In_DeInit(void);
static void I2Sx_In_MspInit(void);
static void I2Sx_In_MspDeInit(void);
static void I2Sx_Out_Init(uint32_t AudioFreq);
static void I2Sx_Out_DeInit(void);
static uint8_t DFSDMx_DeInit(void);
static void DFSDMx_ChannelMspInit(void);
static void DFSDMx_ChannelMspDeInit(void);
static void DFSDMx_FilterMspInit(void);
static void DFSDMx_FilterMspDeInit(void);
/**
* @}
*/
/** @defgroup STM32412G_DISCOVERY_AUDIO_OUT_Private_Functions STM32412G Discovery Audio Out Private Functions
* @{
*/
/**
* @brief Configures the audio peripherals.
* @param OutputDevice: OUTPUT_DEVICE_SPEAKER, OUTPUT_DEVICE_HEADPHONE,
* or OUTPUT_DEVICE_BOTH.
* @param Volume: Initial volume level (from 0 (Mute) to 100 (Max))
* @param AudioFreq: Audio frequency used to play the audio stream.
* @note The I2S PLL input clock must be done in the user application.
* @retval AUDIO_OK if correct communication, else wrong communication
*/
uint8_t BSP_AUDIO_OUT_Init(uint16_t OutputDevice, uint8_t Volume, uint32_t AudioFreq)
{
uint8_t ret = AUDIO_ERROR;
uint32_t deviceid = 0x00;
uint16_t buffer_fake[16] = {0x00};
I2Sx_Out_DeInit();
AUDIO_IO_DeInit();
/* PLL clock is set depending on the AudioFreq (44.1 kHz vs 48kHz groups) */
BSP_AUDIO_OUT_ClockConfig(&haudio_i2s, AudioFreq, NULL);
/* Configure the I2S peripheral */
haudio_i2s.Instance = AUDIO_OUT_I2Sx;
if(HAL_I2S_GetState(&haudio_i2s) == HAL_I2S_STATE_RESET)
{
/* Initialize the I2S Msp: this __weak function can be rewritten by the application */
BSP_AUDIO_OUT_MspInit(&haudio_i2s, NULL);
}
I2Sx_Out_Init(AudioFreq);
AUDIO_IO_Init();
/* 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)
{
/* Send fake I2S data in order to generate MCLK needed by WM8994 to set its registers
* MCLK is generated only when a data stream is sent on I2S */
HAL_I2S_Transmit_DMA(&haudio_i2s, buffer_fake, 16);
/* Initialize the codec internal registers */
audio_drv->Init(AUDIO_I2C_ADDRESS, OutputDevice, Volume, AudioFreq);
/* Stop sending fake I2S data */
HAL_I2S_DMAStop(&haudio_i2s);
}
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_I2S_Transmit_DMA(&haudio_i2s, pBuffer, DMA_MAX(Size / AUDIODATA_SIZE));
return AUDIO_OK;
}
}
/**
* @brief Sends n-Bytes on the I2S interface.
* @param pData: pointer on data address
* @param Size: number of data to be written
*/
void BSP_AUDIO_OUT_ChangeBuffer(uint16_t *pData, uint16_t Size)
{
HAL_I2S_Transmit_DMA(&haudio_i2s, pData, Size);
}
/**
* @brief This function Pauses the audio file stream. In case
* of using DMA, the DMA Pause feature is used.
* @note When calling BSP_AUDIO_OUT_Pause() function for pause, only
* BSP_AUDIO_OUT_Resume() function should be called for resume (use of BSP_AUDIO_OUT_Play()
* function for resume could lead to unexpected behavior).
* @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_I2S_DMAPause(&haudio_i2s);
/* Return AUDIO_OK when all operations are correctly done */
return AUDIO_OK;
}
}
/**
* @brief This function Resumes the audio file stream.
* @note When calling BSP_AUDIO_OUT_Pause() function for pause, only
* BSP_AUDIO_OUT_Resume() function should be called for resume (use of BSP_AUDIO_OUT_Play()
* function for resume could lead to unexpected behavior).
* @retval AUDIO_OK if correct communication, else wrong communication
*/
uint8_t BSP_AUDIO_OUT_Resume(void)
{
/* Call the Media layer pause/resume function */
/* DMA stream resumed before accessing WM8994 register as WM8994 needs the MCLK to be generated to access its registers
* MCLK is generated only when a data stream is sent on I2S */
HAL_I2S_DMAResume(&haudio_i2s);
/* Call the Audio Codec Pause/Resume function */
if(audio_drv->Resume(AUDIO_I2C_ADDRESS) != 0)
{
return AUDIO_ERROR;
}
else
{
/* 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_I2S_DMAStop(&haudio_i2s);
/* 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.
*/
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_i2s, AudioFreq, NULL);
/* Disable I2S peripheral to allow access to I2S internal registers */
__HAL_I2S_DISABLE(&haudio_i2s);
/* Update the I2S audio frequency configuration */
haudio_i2s.Init.AudioFreq = AudioFreq;
HAL_I2S_Init(&haudio_i2s);
/* Enable I2S peripheral to generate MCLK */
__HAL_I2S_ENABLE(&haudio_i2s);
}
/**
* @brief Deinit the audio peripherals.
*/
void BSP_AUDIO_OUT_DeInit(void)
{
I2Sx_Out_DeInit();
/* DeInit the I2S MSP : this __weak function can be rewritten by the application */
BSP_AUDIO_OUT_MspDeInit(&haudio_i2s, NULL);
}
/**
* @brief Tx Transfer completed callbacks.
* @param hi2s: I2S handle
*/
void HAL_I2S_TxCpltCallback(I2S_HandleTypeDef *hi2s)
{
/* Manage the remaining file size and new address offset: This function
should be coded by user (its prototype is already declared in stm32412g_discovery_audio.h) */
BSP_AUDIO_OUT_TransferComplete_CallBack();
}
/**
* @brief Tx Half Transfer completed callbacks.
* @param hi2s: I2S handle
*/
void HAL_I2S_TxHalfCpltCallback(I2S_HandleTypeDef *hi2s)
{
/* Manage the remaining file size and new address offset: This function
should be coded by user (its prototype is already declared in stm32412g_discovery_audio.h) */
BSP_AUDIO_OUT_HalfTransfer_CallBack();
}
/**
* @brief I2S error callbacks.
* @param hi2s: I2S handle
*/
void HAL_I2S_ErrorCallback(I2S_HandleTypeDef *hi2s)
{
BSP_AUDIO_OUT_Error_CallBack();
}
/**
* @brief Manages the DMA full Transfer complete event.
*/
__weak void BSP_AUDIO_OUT_TransferComplete_CallBack(void)
{
}
/**
* @brief Manages the DMA Half Transfer complete event.
*/
__weak void BSP_AUDIO_OUT_HalfTransfer_CallBack(void)
{
}
/**
* @brief Manages the DMA FIFO error event.
*/
__weak void BSP_AUDIO_OUT_Error_CallBack(void)
{
}
/**
* @brief Initializes BSP_AUDIO_OUT MSP.
* @param hi2s: I2S handle
* @param Params : pointer on additional configuration parameters, can be NULL.
*/
__weak void BSP_AUDIO_OUT_MspInit(I2S_HandleTypeDef *hi2s, void *Params)
{
static DMA_HandleTypeDef hdma_i2s_tx;
GPIO_InitTypeDef gpio_init_structure;
/* Enable I2S clock */
AUDIO_OUT_I2Sx_CLK_ENABLE();
/* Enable MCK, SCK, WS, SD and CODEC_INT GPIO clock */
AUDIO_OUT_I2Sx_MCK_GPIO_CLK_ENABLE();
AUDIO_OUT_I2Sx_SCK_GPIO_CLK_ENABLE();
AUDIO_OUT_I2Sx_SD_GPIO_CLK_ENABLE();
AUDIO_OUT_I2Sx_WS_GPIO_CLK_ENABLE();
/* CODEC_I2S pins configuration: MCK, SCK, WS and SD pins */
gpio_init_structure.Pin = AUDIO_OUT_I2Sx_MCK_PIN;
gpio_init_structure.Mode = GPIO_MODE_AF_PP;
gpio_init_structure.Pull = GPIO_NOPULL;
gpio_init_structure.Speed = GPIO_SPEED_FAST;
gpio_init_structure.Alternate = AUDIO_OUT_I2Sx_MCK_AF;
HAL_GPIO_Init(AUDIO_OUT_I2Sx_MCK_GPIO_PORT, &gpio_init_structure);
gpio_init_structure.Pin = AUDIO_OUT_I2Sx_SCK_PIN;
gpio_init_structure.Alternate = AUDIO_OUT_I2Sx_SCK_AF;
HAL_GPIO_Init(AUDIO_OUT_I2Sx_SCK_GPIO_PORT, &gpio_init_structure);
gpio_init_structure.Pin = AUDIO_OUT_I2Sx_WS_PIN;
gpio_init_structure.Alternate = AUDIO_OUT_I2Sx_WS_AF;
HAL_GPIO_Init(AUDIO_OUT_I2Sx_WS_GPIO_PORT, &gpio_init_structure);
gpio_init_structure.Pin = AUDIO_OUT_I2Sx_SD_PIN;
gpio_init_structure.Alternate = AUDIO_OUT_I2Sx_SD_AF;
HAL_GPIO_Init(AUDIO_OUT_I2Sx_SD_GPIO_PORT, &gpio_init_structure);
/* Enable the DMA clock */
AUDIO_OUT_I2Sx_DMAx_CLK_ENABLE();
if(hi2s->Instance == AUDIO_OUT_I2Sx)
{
/* Configure the hdma_i2s_rx handle parameters */
hdma_i2s_tx.Init.Channel = AUDIO_OUT_I2Sx_DMAx_CHANNEL;
hdma_i2s_tx.Init.Direction = DMA_MEMORY_TO_PERIPH;
hdma_i2s_tx.Init.PeriphInc = DMA_PINC_DISABLE;
hdma_i2s_tx.Init.MemInc = DMA_MINC_ENABLE;
hdma_i2s_tx.Init.PeriphDataAlignment = AUDIO_OUT_I2Sx_DMAx_PERIPH_DATA_SIZE;
hdma_i2s_tx.Init.MemDataAlignment = AUDIO_OUT_I2Sx_DMAx_MEM_DATA_SIZE;
hdma_i2s_tx.Init.Mode = DMA_CIRCULAR;
hdma_i2s_tx.Init.Priority = DMA_PRIORITY_HIGH;
hdma_i2s_tx.Init.FIFOMode = DMA_FIFOMODE_DISABLE;
hdma_i2s_tx.Init.FIFOThreshold = DMA_FIFO_THRESHOLD_FULL;
hdma_i2s_tx.Init.MemBurst = DMA_MBURST_SINGLE;
hdma_i2s_tx.Init.PeriphBurst = DMA_MBURST_SINGLE;
hdma_i2s_tx.Instance = AUDIO_OUT_I2Sx_DMAx_STREAM;
/* Associate the DMA handle */
__HAL_LINKDMA(hi2s, hdmatx, hdma_i2s_tx);
/* Deinitialize the Stream for new transfer */
HAL_DMA_DeInit(&hdma_i2s_tx);
/* Configure the DMA Stream */
HAL_DMA_Init(&hdma_i2s_tx);
}
/* Enable and set I2Sx Interrupt to a lower priority */
HAL_NVIC_SetPriority(SPI3_IRQn, 0x0F, 0x00);
HAL_NVIC_EnableIRQ(SPI3_IRQn);
/* I2S DMA IRQ Channel configuration */
HAL_NVIC_SetPriority(AUDIO_OUT_I2Sx_DMAx_IRQ, AUDIO_OUT_IRQ_PREPRIO, 0);
HAL_NVIC_EnableIRQ(AUDIO_OUT_I2Sx_DMAx_IRQ);
}
/**
* @brief Deinitializes I2S MSP.
* @param Params : pointer on additional configuration parameters, can be NULL.
* @param hi2s: I2S handle
*/
__weak void BSP_AUDIO_OUT_MspDeInit(I2S_HandleTypeDef *hi2s, void *Params)
{
GPIO_InitTypeDef gpio_init_structure;
/* I2S DMA IRQ Channel deactivation */
HAL_NVIC_DisableIRQ(AUDIO_OUT_I2Sx_DMAx_IRQ);
if(hi2s->Instance == AUDIO_OUT_I2Sx)
{
/* Deinitialize the DMA stream */
HAL_DMA_DeInit(hi2s->hdmatx);
}
/* Disable I2S peripheral */
__HAL_I2S_DISABLE(hi2s);
/* Deactives CODEC_I2S pins MCK, SCK, WS and SD by putting them in input mode */
gpio_init_structure.Pin = AUDIO_OUT_I2Sx_MCK_PIN;
HAL_GPIO_DeInit(AUDIO_OUT_I2Sx_MCK_GPIO_PORT, gpio_init_structure.Pin);
gpio_init_structure.Pin = AUDIO_OUT_I2Sx_SCK_PIN;
HAL_GPIO_DeInit(AUDIO_OUT_I2Sx_SCK_GPIO_PORT, gpio_init_structure.Pin);
gpio_init_structure.Pin = AUDIO_OUT_I2Sx_WS_PIN;
HAL_GPIO_DeInit(AUDIO_OUT_I2Sx_WS_GPIO_PORT, gpio_init_structure.Pin);
gpio_init_structure.Pin = AUDIO_OUT_I2Sx_SD_PIN;
HAL_GPIO_DeInit(AUDIO_OUT_I2Sx_SD_GPIO_PORT, gpio_init_structure.Pin);
/* Disable I2S clock */
AUDIO_OUT_I2Sx_CLK_DISABLE();
/* GPIO pins clock and DMA clock can be shut down in the application
by surcharging this __weak function */
}
/**
* @brief Clock Config.
* @param hi2s: 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
*/
__weak void BSP_AUDIO_OUT_ClockConfig(I2S_HandleTypeDef *hi2s, uint32_t AudioFreq, void *Params)
{
RCC_PeriphCLKInitTypeDef rcc_ex_clk_init_struct;
HAL_RCCEx_GetPeriphCLKConfig(&rcc_ex_clk_init_struct);
/* Set the PLL configuration according to the audio frequency */
if((AudioFreq == AUDIO_FREQUENCY_11K) || (AudioFreq == AUDIO_FREQUENCY_22K) || (AudioFreq == AUDIO_FREQUENCY_44K))
{
/* Configure PLLI2S prescalers */
rcc_ex_clk_init_struct.PeriphClockSelection = (RCC_PERIPHCLK_I2S_APB1 | RCC_PERIPHCLK_PLLI2S);
rcc_ex_clk_init_struct.I2sApb1ClockSelection = RCC_I2SAPB1CLKSOURCE_PLLI2S;
rcc_ex_clk_init_struct.PLLI2SSelection = RCC_PLLI2SCLKSOURCE_PLLSRC;
rcc_ex_clk_init_struct.PLLI2S.PLLI2SM = 8;
rcc_ex_clk_init_struct.PLLI2S.PLLI2SN = 271;
rcc_ex_clk_init_struct.PLLI2S.PLLI2SR = 2;
HAL_RCCEx_PeriphCLKConfig(&rcc_ex_clk_init_struct);
}
else if(AudioFreq == AUDIO_FREQUENCY_96K) /* AUDIO_FREQUENCY_96K */
{
/* I2S clock config */
rcc_ex_clk_init_struct.PeriphClockSelection = (RCC_PERIPHCLK_I2S_APB1 | RCC_PERIPHCLK_PLLI2S);
rcc_ex_clk_init_struct.I2sApb1ClockSelection = RCC_I2SAPB1CLKSOURCE_PLLI2S;
rcc_ex_clk_init_struct.PLLI2SSelection = RCC_PLLI2SCLKSOURCE_PLLSRC;
rcc_ex_clk_init_struct.PLLI2S.PLLI2SM = 8;
rcc_ex_clk_init_struct.PLLI2S.PLLI2SN = 344;
rcc_ex_clk_init_struct.PLLI2S.PLLI2SR = 2;
HAL_RCCEx_PeriphCLKConfig(&rcc_ex_clk_init_struct);
}
else /* AUDIO_FREQUENCY_8K, AUDIO_FREQUENCY_16K, AUDIO_FREQUENCY_48K */
{
/* I2S clock config
PLLI2S_VCO: VCO_344M
I2S_CLK(first level) = PLLI2S_VCO/PLLI2SR = 344/7 = 49.142 Mhz
I2S_CLK_x = I2S_CLK(first level)/PLLI2SDIVR = 49.142/1 = 49.142 Mhz */
rcc_ex_clk_init_struct.PeriphClockSelection = RCC_PERIPHCLK_I2S_APB1 | RCC_PERIPHCLK_PLLI2S;
rcc_ex_clk_init_struct.I2sApb1ClockSelection = RCC_I2SAPB1CLKSOURCE_PLLI2S;
rcc_ex_clk_init_struct.PLLI2SSelection = RCC_PLLI2SCLKSOURCE_PLLSRC;
rcc_ex_clk_init_struct.PLLI2S.PLLI2SM = 8;
rcc_ex_clk_init_struct.PLLI2S.PLLI2SN = 344;
rcc_ex_clk_init_struct.PLLI2S.PLLI2SR = 7;
HAL_RCCEx_PeriphCLKConfig(&rcc_ex_clk_init_struct);
}
}
/*******************************************************************************
Static Functions
*******************************************************************************/
/**
* @brief Initializes the Audio Codec audio interface (I2S)
* @note This function assumes that the I2S input clock
* is already configured and ready to be used.
* @param AudioFreq: Audio frequency to be configured for the I2S peripheral.
*/
static void I2Sx_Out_Init(uint32_t AudioFreq)
{
/* Initialize the hAudioInI2s Instance parameter */
haudio_i2s.Instance = AUDIO_OUT_I2Sx;
/* Disable I2S block */
__HAL_I2S_DISABLE(&haudio_i2s);
/* I2S peripheral configuration */
haudio_i2s.Init.AudioFreq = AudioFreq;
haudio_i2s.Init.ClockSource = I2S_CLOCK_PLL;
haudio_i2s.Init.CPOL = I2S_CPOL_LOW;
haudio_i2s.Init.DataFormat = I2S_DATAFORMAT_16B;
haudio_i2s.Init.MCLKOutput = I2S_MCLKOUTPUT_ENABLE;
haudio_i2s.Init.Mode = I2S_MODE_MASTER_TX;
haudio_i2s.Init.Standard = I2S_STANDARD_PHILIPS;
haudio_i2s.Init.FullDuplexMode = I2S_FULLDUPLEXMODE_DISABLE;
/* Init the I2S */
HAL_I2S_Init(&haudio_i2s);
/* Enable I2S block */
__HAL_I2S_ENABLE(&haudio_i2s);
}
/**
* @brief Deinitializes the Audio Codec audio interface (I2S).
*/
static void I2Sx_Out_DeInit(void)
{
/* Initialize the hAudioInI2s Instance parameter */
haudio_i2s.Instance = AUDIO_OUT_I2Sx;
/* Disable I2S block */
__HAL_I2S_DISABLE(&haudio_i2s);
/* DeInit the I2S */
HAL_I2S_DeInit(&haudio_i2s);
}
/**
* @}
*/
/** @defgroup STM32412G_DISCOVERY_AUDIO_IN_Private_Functions STM32412G Discovery Audio In Private functions
* @{
*/
/**
* @brief Initializes wave recording.
* @note This function assumes that the I2S input clock
* is already configured and ready to be used.
* @param AudioFreq: Audio frequency to be configured for the I2S peripheral.
* @param BitRes: Audio bit resolustion.
* @param ChnlNbr: Audio channel number.
* @retval AUDIO_OK if correct communication, else wrong communication
*/
uint8_t BSP_AUDIO_IN_Init(uint32_t AudioFreq, uint32_t BitRes, uint32_t ChnlNbr)
{
return BSP_AUDIO_IN_InitEx(INPUT_DEVICE_DIGITAL_MIC, AudioFreq, BitRes, ChnlNbr);
}
/**
* @brief Initializes wave recording.
* @param InputDevice: input device digital or analog
* @param AudioFreq: Audio frequency to be configured for the I2S peripheral.
* @param BitRes: Audio bit resolution.
* @param ChnlNbr: Audio channel number.
* @retval AUDIO_OK if correct communication, else wrong communication
*/
uint8_t BSP_AUDIO_IN_InitEx(uint32_t InputDevice, uint32_t AudioFreq, uint32_t BitRes, uint32_t ChnlNbr)
{
uint32_t ret = AUDIO_ERROR;
uint32_t i = 0;
uint32_t mic_enabled = 0;
uint16_t buffer_fake[16] = {0x00};
/* Store the audio record context */
hAudioIn.Frequency = AudioFreq;
hAudioIn.BitResolution = BitRes;
hAudioIn.InputDevice = InputDevice;
hAudioIn.ChannelNbr = ChnlNbr;
/* Store the total number of microphones enabled */
for(i = 0; i < DFSDM_MIC_NUMBER; i ++)
{
if(((hAudioIn.InputDevice >> i) & INPUT_DEVICE_DIGITAL_MIC1) == INPUT_DEVICE_DIGITAL_MIC1)
{
mic_enabled++;
}
}
if (InputDevice == INPUT_DEVICE_ANALOG_MIC)
{
InputDevice = INPUT_DEVICE_INPUT_LINE_1;
/* INPUT_DEVICE_ANALOG_MIC */
/* Disable I2S */
I2Sx_In_DeInit();
/* 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 for analog mic */
/* I2S data transfer preparation:
Prepare the Media to be used for the audio transfer from I2S peripheral to memory */
haudio_i2s.Instance = AUDIO_IN_I2Sx;
if(HAL_I2S_GetState(&haudio_i2s) == HAL_I2S_STATE_RESET)
{
BSP_AUDIO_OUT_MspInit(&haudio_i2s, NULL); /* Initialize GPIOs for SPI3 Master signals */
/* Init the I2S MSP: this __weak function can be redefined by the application*/
BSP_AUDIO_IN_MspInit(NULL);
}
/* Configure I2S */
I2Sx_In_Init(AudioFreq);
AUDIO_IO_Init();
/* wm8994 codec initialization */
if((wm8994_drv.ReadID(AUDIO_I2C_ADDRESS)) == 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)
{
/* Receive fake I2S data in order to generate MCLK needed by WM8994 to set its registers */
HAL_I2S_Receive_DMA(&haudio_i2s, buffer_fake, 16);
/* Initialize the codec internal registers */
audio_drv->Init(AUDIO_I2C_ADDRESS, (OUTPUT_DEVICE_HEADPHONE|InputDevice), 100, AudioFreq);
/* Stop receiving fake I2S data */
HAL_I2S_DMAStop(&haudio_i2s);
}
}
else
{
if(hAudioIn.ChannelNbr != mic_enabled)
{
return AUDIO_ERROR;
}
else
{
/* 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 for analog mic */
/* Init the DFSDM MSP: this __weak function can be redefined by the application*/
BSP_AUDIO_IN_MspInit(NULL);
/* Default configuration of DFSDM filters and channels */
ret = BSP_AUDIO_IN_ConfigDigitalMic(hAudioIn.InputDevice, NULL);
}
}
/* Return AUDIO_OK when all operations are correctly done */
return ret;
}
/**
* @brief DeInitializes the audio peripheral.
*/
void BSP_AUDIO_IN_DeInit(void)
{
if(hAudioIn.InputDevice != INPUT_DEVICE_ANALOG_MIC)
{
/* MSP filters/channels initialization */
BSP_AUDIO_IN_MspDeInit(NULL);
DFSDMx_DeInit();
}
else
{
I2Sx_In_DeInit();
}
}
/**
* @brief Initializes default configuration of the Digital Filter for Sigma-Delta Modulators interface (DFSDM).
* @param InputDevice: The microphone to be configured. Can be INPUT_DEVICE_DIGITAL_MIC1..INPUT_DEVICE_DIGITAL_MIC5
* @note Channel output Clock Divider and Filter Oversampling are calculated as follow:
* - Clock_Divider = CLK(input DFSDM)/CLK(micro) with
* 1MHZ < CLK(micro) < 3.2MHZ (TYP 2.4MHZ for MP34DT01TR)
* - Oversampling = CLK(input DFSDM)/(Clock_Divider * AudioFreq)
* @retval AUDIO_OK if correct communication, else wrong communication
*/
uint8_t BSP_AUDIO_IN_ConfigMicDefault(uint32_t InputDevice)
{
uint32_t i = 0, mic_init[DFSDM_MIC_NUMBER] = {0};
uint32_t filter_ch = 0, mic_num = 0;
DFSDM_Filter_TypeDef* FilterInstnace[DFSDM_MIC_NUMBER] = {AUDIO_DFSDMx_MIC1_FILTER, AUDIO_DFSDMx_MIC2_FILTER};
DFSDM_Channel_TypeDef* ChannelInstnace[DFSDM_MIC_NUMBER] = {AUDIO_DFSDMx_MIC1_CHANNEL, AUDIO_DFSDMx_MIC2_CHANNEL};
uint32_t DigitalMicPins[DFSDM_MIC_NUMBER] = {DFSDM_CHANNEL_SAME_CHANNEL_PINS, DFSDM_CHANNEL_FOLLOWING_CHANNEL_PINS};
uint32_t DigitalMicType[DFSDM_MIC_NUMBER] = {DFSDM_CHANNEL_SPI_RISING, DFSDM_CHANNEL_SPI_FALLING};
uint32_t Channel4Filter[DFSDM_MIC_NUMBER] = {AUDIO_DFSDMx_MIC1_CHANNEL_FOR_FILTER, AUDIO_DFSDMx_MIC2_CHANNEL_FOR_FILTER};
for(i = 0; i < hAudioIn.ChannelNbr; i++)
{
if(((hAudioIn.InputDevice & INPUT_DEVICE_DIGITAL_MIC1) == INPUT_DEVICE_DIGITAL_MIC1) && (mic_init[POS_VAL(INPUT_DEVICE_DIGITAL_MIC1)] != 1))
{
mic_num = POS_VAL(INPUT_DEVICE_DIGITAL_MIC1);
}
else if(((hAudioIn.InputDevice & INPUT_DEVICE_DIGITAL_MIC2) == INPUT_DEVICE_DIGITAL_MIC2) && (mic_init[POS_VAL(INPUT_DEVICE_DIGITAL_MIC2)] != 1))
{
mic_num = POS_VAL(INPUT_DEVICE_DIGITAL_MIC2);
}
mic_init[mic_num] = 1;
HAL_DFSDM_FilterDeInit(&hAudioInDfsdmFilter[mic_num]);
/* MIC filters initialization */
__HAL_DFSDM_FILTER_RESET_HANDLE_STATE(&hAudioInDfsdmFilter[mic_num]);
hAudioInDfsdmFilter[mic_num].Instance = FilterInstnace[mic_num];
hAudioInDfsdmFilter[mic_num].Init.RegularParam.Trigger = DFSDM_FILTER_SW_TRIGGER;
hAudioInDfsdmFilter[mic_num].Init.RegularParam.FastMode = ENABLE;
hAudioInDfsdmFilter[mic_num].Init.RegularParam.DmaMode = ENABLE;
hAudioInDfsdmFilter[mic_num].Init.InjectedParam.Trigger = DFSDM_FILTER_SW_TRIGGER;
hAudioInDfsdmFilter[mic_num].Init.InjectedParam.ScanMode = DISABLE;
hAudioInDfsdmFilter[mic_num].Init.InjectedParam.DmaMode = DISABLE;
hAudioInDfsdmFilter[mic_num].Init.InjectedParam.ExtTrigger = DFSDM_FILTER_EXT_TRIG_TIM8_TRGO;
hAudioInDfsdmFilter[mic_num].Init.InjectedParam.ExtTriggerEdge = DFSDM_FILTER_EXT_TRIG_BOTH_EDGES;
hAudioInDfsdmFilter[mic_num].Init.FilterParam.SincOrder = DFSDM_FILTER_ORDER(hAudioIn.Frequency);
hAudioInDfsdmFilter[mic_num].Init.FilterParam.Oversampling = DFSDM_OVER_SAMPLING(hAudioIn.Frequency);
hAudioInDfsdmFilter[mic_num].Init.FilterParam.IntOversampling = 1;
if(HAL_OK != HAL_DFSDM_FilterInit(&hAudioInDfsdmFilter[mic_num]))
{
return AUDIO_ERROR;
}
HAL_DFSDM_ChannelDeInit(&hAudioInDfsdmChannel[mic_num]);
/* MIC channels initialization */
__HAL_DFSDM_CHANNEL_RESET_HANDLE_STATE(&hAudioInDfsdmChannel[mic_num]);
hAudioInDfsdmChannel[mic_num].Init.OutputClock.Activation = ENABLE;
hAudioInDfsdmChannel[mic_num].Init.OutputClock.Selection = DFSDM_CHANNEL_OUTPUT_CLOCK_AUDIO;
hAudioInDfsdmChannel[mic_num].Init.OutputClock.Divider = DFSDM_CLOCK_DIVIDER(hAudioIn.Frequency);
hAudioInDfsdmChannel[mic_num].Init.Input.Multiplexer = DFSDM_CHANNEL_EXTERNAL_INPUTS;
hAudioInDfsdmChannel[mic_num].Init.Input.DataPacking = DFSDM_CHANNEL_STANDARD_MODE;
hAudioInDfsdmChannel[mic_num].Init.SerialInterface.SpiClock = DFSDM_CHANNEL_SPI_CLOCK_INTERNAL;
hAudioInDfsdmChannel[mic_num].Init.Awd.FilterOrder = DFSDM_CHANNEL_SINC1_ORDER;
hAudioInDfsdmChannel[mic_num].Init.Awd.Oversampling = 10;
hAudioInDfsdmChannel[mic_num].Init.Offset = 0;
hAudioInDfsdmChannel[mic_num].Init.RightBitShift = DFSDM_MIC_BIT_SHIFT(hAudioIn.Frequency);
hAudioInDfsdmChannel[mic_num].Instance = ChannelInstnace[mic_num];
hAudioInDfsdmChannel[mic_num].Init.Input.Pins = DigitalMicPins[mic_num];
hAudioInDfsdmChannel[mic_num].Init.SerialInterface.Type = DigitalMicType[mic_num];
if(HAL_OK != HAL_DFSDM_ChannelInit(&hAudioInDfsdmChannel[mic_num]))
{
return AUDIO_ERROR;
}
filter_ch = Channel4Filter[mic_num];
/* Configure injected channel */
if(HAL_OK != HAL_DFSDM_FilterConfigRegChannel(&hAudioInDfsdmFilter[mic_num], filter_ch, DFSDM_CONTINUOUS_CONV_ON))
{
return AUDIO_ERROR;
}
}
return AUDIO_OK;
}
/**
* @brief Initializes the Digital Filter for Sigma-Delta Modulators interface (DFSDM).
* @param InputDevice: The microphone to be configured. Can be INPUT_DEVICE_DIGITAL_MIC1..INPUT_DEVICE_DIGITAL_MIC5
* @param Params : pointer on additional configuration parameters, can be NULL.
* @retval AUDIO_OK if correct communication, else wrong communication
*/
__weak uint8_t BSP_AUDIO_IN_ConfigDigitalMic(uint32_t InputDevice, void *Params)
{
/* Prevent unused argument(s) compilation warning */
UNUSED(Params);
/* Default configuration of DFSDM filters and channels */
return(BSP_AUDIO_IN_ConfigMicDefault(InputDevice));
/* Note: This function can be called at application level and default configuration
can be overwritten to fit user's need */
}
/**
* @brief Allocate channel buffer scratch
* @param pScratch : pointer to scratch tables.
* @param size of scratch buffer
*/
uint8_t BSP_AUDIO_IN_AllocScratch (int32_t *pScratch, uint32_t size)
{
uint32_t idx;
ScratchSize = size / DEFAULT_AUDIO_IN_CHANNEL_NBR;
/* copy scratch pointers */
for (idx = 0; idx < DEFAULT_AUDIO_IN_CHANNEL_NBR ; idx++)
{
pScratchBuff[idx] = (int32_t *)(pScratch + idx * ScratchSize);
}
/* Return AUDIO_OK */
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
* @retval AUDIO_OK if correct communication, else wrong communication
*/
uint8_t BSP_AUDIO_IN_Record(uint16_t *pBuf, uint32_t size)
{
hAudioIn.pRecBuf = pBuf;
hAudioIn.RecSize = size;
/* Reset Application Buffer Trigger */
AppBuffTrigger = 0;
AppBuffHalf = 0;
if (hAudioIn.InputDevice == INPUT_DEVICE_DIGITAL_MIC)
{
/* Call the Media layer start function for MIC1 channel */
if(HAL_OK != HAL_DFSDM_FilterRegularStart_DMA(&hAudioInDfsdmFilter[POS_VAL(INPUT_DEVICE_DIGITAL_MIC1)], pScratchBuff[POS_VAL(INPUT_DEVICE_DIGITAL_MIC1)], ScratchSize))
{
return AUDIO_ERROR;
}
/* Call the Media layer start function for MIC2 channel */
if(HAL_OK != HAL_DFSDM_FilterRegularStart_DMA(&hAudioInDfsdmFilter[POS_VAL(INPUT_DEVICE_DIGITAL_MIC2)], pScratchBuff[POS_VAL(INPUT_DEVICE_DIGITAL_MIC2)], ScratchSize))
{
return AUDIO_ERROR;
}
}
else
{
/* Start the process to receive the DMA */
if (HAL_OK != HAL_I2SEx_TransmitReceive_DMA(&haudio_i2s, pBuf, pBuf, size))
{
return AUDIO_ERROR;
}
}
/* Return AUDIO_OK when all operations are correctly done */
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
* @retval AUDIO_OK if correct communication, else wrong communication
*/
uint8_t BSP_AUDIO_IN_RecordEx(uint32_t *pBuf, uint32_t size)
{
uint8_t ret = AUDIO_ERROR;
hAudioIn.RecSize = size;
uint32_t i = 0;
uint32_t mic_init[DFSDM_MIC_NUMBER] = {0};
if (hAudioIn.InputDevice == INPUT_DEVICE_ANALOG_MIC)
{
return ret;
}
else
{
hAudioIn.MultiBuffMode = 1;
for(i = 0; i < hAudioIn.ChannelNbr; i++)
{
if(((hAudioIn.InputDevice & INPUT_DEVICE_DIGITAL_MIC1) == INPUT_DEVICE_DIGITAL_MIC1) && (mic_init[POS_VAL(INPUT_DEVICE_DIGITAL_MIC1)] != 1))
{
/* Call the Media layer start function for MIC1 channel 1 */
if(HAL_OK != HAL_DFSDM_FilterRegularStart_DMA(&hAudioInDfsdmFilter[POS_VAL(INPUT_DEVICE_DIGITAL_MIC1)], (int32_t*)pBuf[i], size))
{
return AUDIO_ERROR;
}
MicBuff[POS_VAL(INPUT_DEVICE_DIGITAL_MIC1)] = i;
mic_init[POS_VAL(INPUT_DEVICE_DIGITAL_MIC1)] = 1;
}
else if(((hAudioIn.InputDevice & INPUT_DEVICE_DIGITAL_MIC2) == INPUT_DEVICE_DIGITAL_MIC2) && (mic_init[POS_VAL(INPUT_DEVICE_DIGITAL_MIC2)] != 1))
{
/* Call the Media layer start function for MIC2 channel 1 */
if(HAL_OK != HAL_DFSDM_FilterRegularStart_DMA(&hAudioInDfsdmFilter[POS_VAL(INPUT_DEVICE_DIGITAL_MIC2)], (int32_t*)pBuf[i], size))
{
return AUDIO_ERROR;
}
MicBuff[POS_VAL(INPUT_DEVICE_DIGITAL_MIC2)] = i;
mic_init[POS_VAL(INPUT_DEVICE_DIGITAL_MIC2)] = 1;
}
}
}
/* Return AUDIO_OK when all operations are correctly done */
return AUDIO_OK;
}
/**
* @brief Initializes the I2S MSP.
*/
static void I2Sx_In_MspInit(void)
{
static DMA_HandleTypeDef hdma_i2s_rx;
GPIO_InitTypeDef gpio_init_structure;
/* Enable I2S clock */
AUDIO_IN_I2Sx_CLK_ENABLE();
/* Enable MCK GPIO clock, needed by the codec */
AUDIO_OUT_I2Sx_MCK_GPIO_CLK_ENABLE();
/* CODEC_I2S pins configuration: MCK pins */
gpio_init_structure.Pin = AUDIO_OUT_I2Sx_MCK_PIN;
gpio_init_structure.Mode = GPIO_MODE_AF_PP;
gpio_init_structure.Pull = GPIO_NOPULL;
gpio_init_structure.Speed = GPIO_SPEED_FAST;
gpio_init_structure.Alternate = AUDIO_OUT_I2Sx_MCK_AF;
HAL_GPIO_Init(AUDIO_OUT_I2Sx_MCK_GPIO_PORT, &gpio_init_structure);
/* Enable SD GPIO clock */
AUDIO_IN_I2Sx_EXT_SD_GPIO_CLK_ENABLE();
/* CODEC_I2S pin configuration: SD pin */
gpio_init_structure.Pin = AUDIO_IN_I2Sx_EXT_SD_PIN;
gpio_init_structure.Alternate = AUDIO_IN_I2Sx_EXT_SD_AF;
HAL_GPIO_Init(AUDIO_IN_I2Sx_EXT_SD_GPIO_PORT, &gpio_init_structure);
/* Enable the DMA clock */
AUDIO_IN_I2Sx_DMAx_CLK_ENABLE();
if(haudio_i2s.Instance == AUDIO_IN_I2Sx)
{
/* Configure the hdma_i2s_rx handle parameters */
hdma_i2s_rx.Init.Channel = AUDIO_IN_I2Sx_DMAx_CHANNEL;
hdma_i2s_rx.Init.Direction = DMA_PERIPH_TO_MEMORY;
hdma_i2s_rx.Init.PeriphInc = DMA_PINC_DISABLE;
hdma_i2s_rx.Init.MemInc = DMA_MINC_ENABLE;
hdma_i2s_rx.Init.PeriphDataAlignment = AUDIO_IN_I2Sx_DMAx_PERIPH_DATA_SIZE;
hdma_i2s_rx.Init.MemDataAlignment = AUDIO_IN_I2Sx_DMAx_MEM_DATA_SIZE;
hdma_i2s_rx.Init.Mode = DMA_CIRCULAR;
hdma_i2s_rx.Init.Priority = DMA_PRIORITY_HIGH;
hdma_i2s_rx.Init.FIFOMode = DMA_FIFOMODE_DISABLE;
hdma_i2s_rx.Init.FIFOThreshold = DMA_FIFO_THRESHOLD_FULL;
hdma_i2s_rx.Init.MemBurst = DMA_MBURST_SINGLE;
hdma_i2s_rx.Init.PeriphBurst = DMA_MBURST_SINGLE;
hdma_i2s_rx.Instance = AUDIO_IN_I2Sx_DMAx_STREAM;
/* Associate the DMA handle */
__HAL_LINKDMA(&haudio_i2s, hdmarx, hdma_i2s_rx);
/* Deinitialize the Stream for new transfer */
HAL_DMA_DeInit(&hdma_i2s_rx);
/* Configure the DMA Stream */
HAL_DMA_Init(&hdma_i2s_rx);
}
/* I2S DMA IRQ Channel configuration */
HAL_NVIC_SetPriority(AUDIO_IN_I2Sx_DMAx_IRQ, AUDIO_IN_IRQ_PREPRIO, 0);
HAL_NVIC_EnableIRQ(AUDIO_IN_I2Sx_DMAx_IRQ);
}
/**
* @brief De-Initializes the I2S MSP.
*/
static void I2Sx_In_MspDeInit(void)
{
GPIO_InitTypeDef gpio_init_structure;
/* I2S DMA IRQ Channel deactivation */
HAL_NVIC_DisableIRQ(AUDIO_IN_I2Sx_DMAx_IRQ);
if(haudio_i2s.Instance == AUDIO_IN_I2Sx)
{
/* Deinitialize the DMA stream */
HAL_DMA_DeInit(haudio_i2s.hdmarx);
}
/* Disable I2S peripheral */
__HAL_I2S_DISABLE(&haudio_i2s);
/* Deactives CODEC_I2S pins MCK by putting them in input mode */
gpio_init_structure.Pin = AUDIO_OUT_I2Sx_MCK_PIN;
HAL_GPIO_DeInit(AUDIO_OUT_I2Sx_MCK_GPIO_PORT, gpio_init_structure.Pin);
gpio_init_structure.Pin = AUDIO_IN_I2Sx_EXT_SD_PIN;
HAL_GPIO_DeInit(AUDIO_IN_I2Sx_EXT_SD_GPIO_PORT, gpio_init_structure.Pin);
/* Disable I2S clock */
AUDIO_IN_I2Sx_CLK_DISABLE();
}
/**
* @brief Initializes BSP_AUDIO_IN MSP.
* @param Params : pointer on additional configuration parameters, can be NULL.
*/
__weak void BSP_AUDIO_IN_MspInit(void *Params)
{
if(hAudioIn.InputDevice == INPUT_DEVICE_ANALOG_MIC)
{
I2Sx_In_MspInit();
}
else
{
/* MSP channels initialization */
DFSDMx_ChannelMspInit();
/* MSP filters initialization */
DFSDMx_FilterMspInit();
}
}
/**
* @brief De-Initializes BSP_AUDIO_IN MSP.
* @param Params : pointer on additional configuration parameters, can be NULL.
*/
__weak void BSP_AUDIO_IN_MspDeInit(void *Params)
{
if(hAudioIn.InputDevice == INPUT_DEVICE_ANALOG_MIC)
{
I2Sx_In_MspDeInit();
}
else
{
/* MSP channels initialization */
DFSDMx_ChannelMspDeInit();
/* MSP filters initialization */
DFSDMx_FilterMspDeInit();
}
}
/**
* @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_OUT_Init() and BSP_AUDIO_OUT_SetFrequency()
* Being __weak it can be overwritten by the application
* @retval AUDIO_OK if correct communication, else wrong communication
*/
__weak uint8_t 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))
{
/* Configure PLLI2S prescalers */
rcc_ex_clk_init_struct.PeriphClockSelection = (RCC_PERIPHCLK_I2S_APB1 | RCC_PERIPHCLK_DFSDM);
rcc_ex_clk_init_struct.I2sApb1ClockSelection = RCC_I2SAPB1CLKSOURCE_PLLI2S;
rcc_ex_clk_init_struct.DfsdmClockSelection = RCC_DFSDM1CLKSOURCE_APB2;
rcc_ex_clk_init_struct.PLLI2S.PLLI2SM = 8;
rcc_ex_clk_init_struct.PLLI2S.PLLI2SN = 271;
rcc_ex_clk_init_struct.PLLI2S.PLLI2SR = 2;
HAL_RCCEx_PeriphCLKConfig(&rcc_ex_clk_init_struct);
}
else if(AudioFreq == AUDIO_FREQUENCY_96K)
{
/* I2S clock config */
rcc_ex_clk_init_struct.PeriphClockSelection = (RCC_PERIPHCLK_I2S_APB1 | RCC_PERIPHCLK_DFSDM);
rcc_ex_clk_init_struct.I2sApb1ClockSelection = RCC_I2SAPB1CLKSOURCE_PLLI2S;
rcc_ex_clk_init_struct.DfsdmClockSelection = RCC_DFSDM1CLKSOURCE_APB2;
rcc_ex_clk_init_struct.PLLI2S.PLLI2SM = 8;
rcc_ex_clk_init_struct.PLLI2S.PLLI2SN = 344;
rcc_ex_clk_init_struct.PLLI2S.PLLI2SR = 2;
HAL_RCCEx_PeriphCLKConfig(&rcc_ex_clk_init_struct);
}
else /* AUDIO_FREQUENCY_8K, AUDIO_FREQUENCY_16K, AUDIO_FREQUENCY_32K, AUDIO_FREQUENCY_48K */
{
/* I2S clock config
PLLI2S_VCO: VCO_344M
I2S_CLK(first level) = PLLI2S_VCO/PLLI2SR = 344/7 = 49.142 Mhz
I2S_CLK_x = I2S_CLK(first level)/PLLI2SDIVR = 49.142/1 = 49.142 Mhz */
rcc_ex_clk_init_struct.PeriphClockSelection = (RCC_PERIPHCLK_I2S_APB1 | RCC_PERIPHCLK_DFSDM);
rcc_ex_clk_init_struct.I2sApb1ClockSelection = RCC_I2SAPB1CLKSOURCE_PLLI2S;
rcc_ex_clk_init_struct.DfsdmClockSelection = RCC_DFSDM1CLKSOURCE_APB2;
rcc_ex_clk_init_struct.PLLI2S.PLLI2SM = 8;
rcc_ex_clk_init_struct.PLLI2S.PLLI2SN = 344;
rcc_ex_clk_init_struct.PLLI2S.PLLI2SR = 7;
HAL_RCCEx_PeriphCLKConfig(&rcc_ex_clk_init_struct);
}
if(hAudioIn.InputDevice != INPUT_DEVICE_ANALOG_MIC)
{
/* I2S_APB1 selected as DFSDM audio clock source */
__HAL_RCC_DFSDM1AUDIO_CONFIG(RCC_DFSDM1AUDIOCLKSOURCE_I2SAPB1);
}
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.
*/
void HAL_DFSDM_FilterRegConvCpltCallback(DFSDM_Filter_HandleTypeDef *hdfsdm_filter)
{
uint32_t index, input_device = 0;
if(hdfsdm_filter->Instance == AUDIO_DFSDMx_MIC1_FILTER)
{
DmaRecBuffCplt[POS_VAL(INPUT_DEVICE_DIGITAL_MIC1)] = 1;
input_device = INPUT_DEVICE_DIGITAL_MIC1;
}
else if(hdfsdm_filter->Instance == AUDIO_DFSDMx_MIC2_FILTER)
{
DmaRecBuffCplt[POS_VAL(INPUT_DEVICE_DIGITAL_MIC2)] = 1;
input_device = INPUT_DEVICE_DIGITAL_MIC2;
}
if(hAudioIn.MultiBuffMode == 1)
{
BSP_AUDIO_IN_TransferComplete_CallBackEx(input_device);
}
else
{
if((DmaRecBuffCplt[POS_VAL(INPUT_DEVICE_DIGITAL_MIC2)] == 1) && (DmaRecBuffCplt[POS_VAL(INPUT_DEVICE_DIGITAL_MIC1)] == 1))
{
if(AppBuffTrigger >= hAudioIn.RecSize)
AppBuffTrigger = 0;
for(index = (ScratchSize/2) ; index < ScratchSize; index++)
{
hAudioIn.pRecBuf[AppBuffTrigger] = (uint16_t)(SaturaLH((pScratchBuff[POS_VAL(INPUT_DEVICE_DIGITAL_MIC1)][index] >> 8), -32760, 32760));
hAudioIn.pRecBuf[AppBuffTrigger + 1] = (uint16_t)(SaturaLH((pScratchBuff[POS_VAL(INPUT_DEVICE_DIGITAL_MIC2)][index] >> 8), -32760, 32760));
AppBuffTrigger += 2;
}
DmaRecBuffCplt[POS_VAL(INPUT_DEVICE_DIGITAL_MIC2)] = DmaRecBuffCplt[POS_VAL(INPUT_DEVICE_DIGITAL_MIC1)] = 0;
}
/* Update Trigger with Remaining Byte before callback if necessary */
if(AppBuffTrigger >= hAudioIn.RecSize)
{
/* Reset Application Buffer Trigger */
AppBuffTrigger = 0;
AppBuffHalf = 0;
/* Call the record update function to get the next buffer to fill and its size (size is ignored) */
BSP_AUDIO_IN_TransferComplete_CallBack();
}
else if((AppBuffTrigger >= hAudioIn.RecSize/2))
{
if(AppBuffHalf == 0)
{
AppBuffHalf = 1;
/* Manage the remaining file size and new address offset: This function
should be coded by user (its prototype is already declared in stm32l476g_eval_audio.h) */
BSP_AUDIO_IN_HalfTransfer_CallBack();
}
}
}
}
/**
* @brief Half regular conversion complete callback.
* @param hdfsdm_filter : DFSDM filter handle.
*/
void HAL_DFSDM_FilterRegConvHalfCpltCallback(DFSDM_Filter_HandleTypeDef *hdfsdm_filter)
{
uint32_t index, input_device = 0;
if(hdfsdm_filter->Instance == AUDIO_DFSDMx_MIC1_FILTER)
{
DmaRecHalfBuffCplt[POS_VAL(INPUT_DEVICE_DIGITAL_MIC1)] = 1;
input_device = INPUT_DEVICE_DIGITAL_MIC1;
}
else if(hdfsdm_filter->Instance == AUDIO_DFSDMx_MIC2_FILTER)
{
DmaRecHalfBuffCplt[POS_VAL(INPUT_DEVICE_DIGITAL_MIC2)] = 1;
input_device = INPUT_DEVICE_DIGITAL_MIC2;
}
if(hAudioIn.MultiBuffMode == 1)
{
BSP_AUDIO_IN_HalfTransfer_CallBackEx(input_device);
}
else
{
if((DmaRecHalfBuffCplt[POS_VAL(INPUT_DEVICE_DIGITAL_MIC2)] == 1) && (DmaRecHalfBuffCplt[POS_VAL(INPUT_DEVICE_DIGITAL_MIC1)] == 1))
{
if(AppBuffTrigger >= hAudioIn.RecSize)
AppBuffTrigger = 0;
for(index = 0; index < ScratchSize/2; index++)
{
hAudioIn.pRecBuf[AppBuffTrigger] = (int16_t)(SaturaLH((pScratchBuff[POS_VAL(INPUT_DEVICE_DIGITAL_MIC1)][index] >> 8), -32760, 32760));
hAudioIn.pRecBuf[AppBuffTrigger + 1] = (int16_t)(SaturaLH((pScratchBuff[POS_VAL(INPUT_DEVICE_DIGITAL_MIC2)][index] >> 8), -32760, 32760));
AppBuffTrigger += 2;
}
DmaRecHalfBuffCplt[POS_VAL(INPUT_DEVICE_DIGITAL_MIC2)] = DmaRecHalfBuffCplt[POS_VAL(INPUT_DEVICE_DIGITAL_MIC1)] = 0;
}
/* Update Trigger with Remaining Byte before callback if necessary */
if(AppBuffTrigger >= hAudioIn.RecSize)
{
/* Reset Application Buffer Trigger */
AppBuffTrigger = 0;
AppBuffHalf = 0;
/* Call the record update function to get the next buffer to fill and its size (size is ignored) */
BSP_AUDIO_IN_TransferComplete_CallBack();
}
else if((AppBuffTrigger >= hAudioIn.RecSize/2))
{
if(AppBuffHalf == 0)
{
AppBuffHalf = 1;
/* Manage the remaining file size and new address offset: This function
should be coded by user */
BSP_AUDIO_IN_HalfTransfer_CallBack();
}
}
}
}
/**
* @brief Half reception complete callback.
* @param hi2s : I2S handle.
*/
void HAL_I2S_RxHalfCpltCallback(I2S_HandleTypeDef *hi2s)
{
/* Manage the remaining file size and new address offset: This function
should be coded by user (its prototype is already declared in stm32746g_discovery_audio.h) */
BSP_AUDIO_IN_HalfTransfer_CallBack();
}
/**
* @brief Reception complete callback.
* @param hi2s : I2S handle.
*/
void HAL_I2S_RxCpltCallback(I2S_HandleTypeDef *hi2s)
{
/* Call the record update function to get the next buffer to fill and its size (size is ignored) */
BSP_AUDIO_IN_TransferComplete_CallBack();
}
/**
* @brief Stops audio recording.
* @retval AUDIO_OK if correct communication, else wrong communication
*/
uint8_t BSP_AUDIO_IN_Stop(void)
{
AppBuffTrigger = 0;
AppBuffHalf = 0;
if (hAudioIn.InputDevice == INPUT_DEVICE_ANALOG_MIC)
{
/* Call the Media layer stop function */
if(HAL_OK != HAL_I2S_DMAStop(&haudio_i2s))
{
return AUDIO_ERROR;
}
/* Call Audio Codec Stop function */
if(audio_drv->Stop(AUDIO_I2C_ADDRESS, CODEC_PDWN_HW) != 0)
{
return AUDIO_ERROR;
}
/* Wait at least 100us */
HAL_Delay(1);
}
else /* InputDevice = Digital Mic */
{
/* Call the Media layer stop function for MIC1 channel */
if(AUDIO_OK != BSP_AUDIO_IN_PauseEx(INPUT_DEVICE_DIGITAL_MIC1))
{
return AUDIO_ERROR;
}
/* Call the Media layer stop function for MIC2 channel */
if(AUDIO_OK != BSP_AUDIO_IN_PauseEx(INPUT_DEVICE_DIGITAL_MIC2))
{
return AUDIO_ERROR;
}
}
/* Return AUDIO_OK when all operations are correctly done */
return AUDIO_OK;
}
/**
* @brief Stops audio recording.
* @param InputDevice: Microphone to be stopped. Can be INPUT_DEVICE_DIGITAL_MIC1 or INPUT_DEVICE_DIGITAL_MIC2.
* @retval AUDIO_OK if correct communication, else wrong communication
*/
uint8_t BSP_AUDIO_IN_StopEx(uint32_t InputDevice)
{
if((InputDevice < INPUT_DEVICE_DIGITAL_MIC1) || (InputDevice > INPUT_DEVICE_DIGITAL_MIC2))
{
return AUDIO_ERROR;
}
else
{
BSP_AUDIO_IN_PauseEx(InputDevice);
}
/* Return AUDIO_OK when all operations are correctly done */
return AUDIO_OK;
}
/**
* @brief Pauses the audio file stream.
* @retval AUDIO_OK if correct communication, else wrong communication
*/
uint8_t BSP_AUDIO_IN_Pause(void)
{
if (hAudioIn.InputDevice == INPUT_DEVICE_ANALOG_MIC)
{
return AUDIO_ERROR;
}
else
{
/* Call the Media layer stop function */
if(HAL_OK != HAL_DFSDM_FilterRegularStop_DMA(&hAudioInDfsdmFilter[POS_VAL(INPUT_DEVICE_DIGITAL_MIC1)]))
{
return AUDIO_ERROR;
}
/* Call the Media layer stop function */
if(HAL_OK != HAL_DFSDM_FilterRegularStop_DMA(&hAudioInDfsdmFilter[POS_VAL(INPUT_DEVICE_DIGITAL_MIC2)]))
{
return AUDIO_ERROR;
}
}
/* Return AUDIO_OK when all operations are correctly done */
return AUDIO_OK;
}
/**
* @brief Pauses the audio file stream.
* @param InputDevice: Microphone to be paused. Can be INPUT_DEVICE_DIGITAL_MIC1 or INPUT_DEVICE_DIGITAL_MIC2.
* @retval AUDIO_OK if correct communication, else wrong communication
*/
uint8_t BSP_AUDIO_IN_PauseEx(uint32_t InputDevice)
{
if((InputDevice < INPUT_DEVICE_DIGITAL_MIC1) || (InputDevice > INPUT_DEVICE_DIGITAL_MIC2))
{
return AUDIO_ERROR;
}
else
{
/* Call the Media layer stop function */
if(HAL_OK != HAL_DFSDM_FilterRegularStop_DMA(&hAudioInDfsdmFilter[POS_VAL(InputDevice)]))
{
return AUDIO_ERROR;
}
}
/* Return AUDIO_OK when all operations are correctly done */
return AUDIO_OK;
}
/**
* @brief Resumes the audio file stream.
* @retval AUDIO_OK if correct communication, else wrong communication
*/
uint8_t BSP_AUDIO_IN_Resume(void)
{
if (hAudioIn.InputDevice == INPUT_DEVICE_ANALOG_MIC)
{
return AUDIO_ERROR;
}
else
{
/* Call the Media layer start function for MIC2 channel */
if(HAL_OK != HAL_DFSDM_FilterRegularStart_DMA(&hAudioInDfsdmFilter[POS_VAL(INPUT_DEVICE_DIGITAL_MIC2)], pScratchBuff[POS_VAL(INPUT_DEVICE_DIGITAL_MIC2)], ScratchSize))
{
return AUDIO_ERROR;
}
/* Call the Media layer start function for MIC1 channel */
if(HAL_OK != HAL_DFSDM_FilterRegularStart_DMA(&hAudioInDfsdmFilter[POS_VAL(INPUT_DEVICE_DIGITAL_MIC1)], pScratchBuff[POS_VAL(INPUT_DEVICE_DIGITAL_MIC1)], ScratchSize))
{
return AUDIO_ERROR;
}
}
/* Return AUDIO_OK when all operations are correctly done */
return AUDIO_OK;
}
/**
* @brief Resumes the audio file stream.
* @param pBuf: Main buffer pointer for the recorded data storing
* @param InputDevice: Microphone to be paused. Can be INPUT_DEVICE_DIGITAL_MIC1 or INPUT_DEVICE_DIGITAL_MIC2.
* @retval AUDIO_OK if correct communication, else wrong communication
*/
uint8_t BSP_AUDIO_IN_ResumeEx(uint32_t *pBuf, uint32_t InputDevice)
{
if((InputDevice < INPUT_DEVICE_DIGITAL_MIC1) || (InputDevice > INPUT_DEVICE_DIGITAL_MIC2))
{
return AUDIO_ERROR;
}
else
{
/* Call the Media layer stop function */
if(HAL_OK != HAL_DFSDM_FilterRegularStart_DMA(&hAudioInDfsdmFilter[POS_VAL(InputDevice)], (int32_t*)pBuf[MicBuff[POS_VAL(InputDevice)]], hAudioIn.RecSize))
{
return AUDIO_ERROR;
}
}
/* 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 User callback when record buffer is filled.
*/
__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.
*/
__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.
*/
__weak void BSP_AUDIO_IN_Error_Callback(void)
{
/* This function is called when an Interrupt due to transfer error on or peripheral
error occurs. */
}
/**
* @}
*/
/*******************************************************************************
Static Functions
*******************************************************************************/
/**
* @brief De-initializes the Digital Filter for Sigma-Delta Modulators interface (DFSDM).
* @retval AUDIO_OK if correct communication, else wrong communication
*/
static uint8_t DFSDMx_DeInit(void)
{
for(uint32_t i = 0; i < DFSDM_MIC_NUMBER; i++)
{
if(hAudioInDfsdmFilter[i].Instance != NULL)
{
if(HAL_OK != HAL_DFSDM_FilterDeInit(&hAudioInDfsdmFilter[i]))
{
return AUDIO_ERROR;
}
hAudioInDfsdmFilter[i].Instance = NULL;
}
if(hAudioInDfsdmChannel[i].Instance != NULL)
{
if(HAL_OK != HAL_DFSDM_ChannelDeInit(&hAudioInDfsdmChannel[i]))
{
return AUDIO_ERROR;
}
hAudioInDfsdmChannel[i].Instance = NULL;
}
}
return AUDIO_OK;
}
/**
* @brief Initializes the DFSDM channel MSP.
*/
static void DFSDMx_ChannelMspInit(void)
{
GPIO_InitTypeDef GPIO_InitStruct;
/* Enable DFSDM clock */
AUDIO_DFSDMx_CLK_ENABLE();
/* Enable GPIO clock */
AUDIO_DFSDMx_CKOUT_DMIC_GPIO_CLK_ENABLE();
/* DFSDM pins configuration: DFSDM_CKOUT, DMIC_DATIN pins ------------------*/
GPIO_InitStruct.Pin = AUDIO_DFSDMx_CKOUT_PIN;
GPIO_InitStruct.Mode = GPIO_MODE_AF_PP;
GPIO_InitStruct.Pull = GPIO_NOPULL;
GPIO_InitStruct.Speed = GPIO_SPEED_HIGH;
GPIO_InitStruct.Alternate = AUDIO_DFSDMx_CKOUT_DMIC_AF;
HAL_GPIO_Init(AUDIO_DFSDMx_CKOUT_DMIC_GPIO_PORT, &GPIO_InitStruct);
/* MP34DT01TR microphones uses DFSDM_DATIN0 input pin */
AUDIO_DFSDMx_DMIC_GPIO_CLK_ENABLE();
GPIO_InitStruct.Pin = AUDIO_DFSDMx_DMIC_PIN;
GPIO_InitStruct.Mode = GPIO_MODE_AF_PP;
GPIO_InitStruct.Pull = GPIO_NOPULL;
GPIO_InitStruct.Speed = GPIO_SPEED_HIGH;
GPIO_InitStruct.Alternate = AUDIO_DFSDMx_DMIC_AF;
HAL_GPIO_Init(AUDIO_DFSDMx_DMIC_GPIO_PORT, &GPIO_InitStruct);
}
/**
* @brief DeInitializes the DFSDM channel MSP.
*/
static void DFSDMx_ChannelMspDeInit(void)
{
GPIO_InitTypeDef GPIO_InitStruct;
/* DFSDM pins configuration: DFSDM_CKOUT, DMIC_DATIN pins ------------------*/
GPIO_InitStruct.Pin = AUDIO_DFSDMx_CKOUT_PIN;
HAL_GPIO_DeInit(AUDIO_DFSDMx_CKOUT_DMIC_GPIO_PORT, GPIO_InitStruct.Pin);
/* MP34DT01TR microphones uses DFSDM_DATIN0 input pin */
GPIO_InitStruct.Pin = AUDIO_DFSDMx_DMIC_PIN;
HAL_GPIO_DeInit(AUDIO_DFSDMx_DMIC_GPIO_PORT, GPIO_InitStruct.Pin);
}
/**
* @brief Initializes the DFSDM filter MSP.
*/
static void DFSDMx_FilterMspInit(void)
{
uint32_t i = 0, mic_num = 0, mic_init[DFSDM_MIC_NUMBER] = {0};
IRQn_Type AUDIO_DFSDM_DMAx_MIC_IRQHandler[DFSDM_MIC_NUMBER] = {AUDIO_DFSDMx_DMAx_MIC1_IRQ, AUDIO_DFSDMx_DMAx_MIC2_IRQ};
DMA_Stream_TypeDef* AUDIO_DFSDMx_DMAx_MIC_STREAM[DFSDM_MIC_NUMBER] = {AUDIO_DFSDMx_DMAx_MIC1_STREAM, AUDIO_DFSDMx_DMAx_MIC2_STREAM};
uint32_t AUDIO_DFSDMx_DMAx_MIC_CHANNEL[DFSDM_MIC_NUMBER] = {AUDIO_DFSDMx_DMAx_MIC1_CHANNEL, AUDIO_DFSDMx_DMAx_MIC2_CHANNEL};
/* Enable the DMA clock */
AUDIO_DFSDMx_DMAx_CLK_ENABLE();
for(i = 0; i < hAudioIn.ChannelNbr; i++)
{
if(((hAudioIn.InputDevice & INPUT_DEVICE_DIGITAL_MIC1) == INPUT_DEVICE_DIGITAL_MIC1) && (mic_init[POS_VAL(INPUT_DEVICE_DIGITAL_MIC1)] != 1))
{
mic_num = POS_VAL(INPUT_DEVICE_DIGITAL_MIC1);
mic_init[mic_num] = 1;
}
else if(((hAudioIn.InputDevice & INPUT_DEVICE_DIGITAL_MIC2) == INPUT_DEVICE_DIGITAL_MIC2) && (mic_init[POS_VAL(INPUT_DEVICE_DIGITAL_MIC2)] != 1))
{
mic_num = POS_VAL(INPUT_DEVICE_DIGITAL_MIC2);
mic_init[mic_num] = 1;
}
/* Configure the hDmaDfsdm[i] handle parameters */
hDmaDfsdm[mic_num].Init.Channel = AUDIO_DFSDMx_DMAx_MIC_CHANNEL[mic_num];
hDmaDfsdm[mic_num].Instance = AUDIO_DFSDMx_DMAx_MIC_STREAM[mic_num];
hDmaDfsdm[mic_num].Init.Direction = DMA_PERIPH_TO_MEMORY;
hDmaDfsdm[mic_num].Init.PeriphInc = DMA_PINC_DISABLE;
hDmaDfsdm[mic_num].Init.MemInc = DMA_MINC_ENABLE;
hDmaDfsdm[mic_num].Init.PeriphDataAlignment = AUDIO_DFSDMx_DMAx_PERIPH_DATA_SIZE;
hDmaDfsdm[mic_num].Init.MemDataAlignment = AUDIO_DFSDMx_DMAx_MEM_DATA_SIZE;
hDmaDfsdm[mic_num].Init.Mode = DMA_CIRCULAR;
hDmaDfsdm[mic_num].Init.Priority = DMA_PRIORITY_HIGH;
hDmaDfsdm[mic_num].Init.FIFOMode = DMA_FIFOMODE_DISABLE;
hDmaDfsdm[mic_num].Init.MemBurst = DMA_MBURST_SINGLE;
hDmaDfsdm[mic_num].Init.PeriphBurst = DMA_PBURST_SINGLE;
hDmaDfsdm[mic_num].State = HAL_DMA_STATE_RESET;
/* Associate the DMA handle */
__HAL_LINKDMA(&hAudioInDfsdmFilter[mic_num], hdmaReg, hDmaDfsdm[mic_num]);
/* Reset DMA handle state */
__HAL_DMA_RESET_HANDLE_STATE(&hDmaDfsdm[mic_num]);
/* Configure the DMA Channel */
HAL_DMA_Init(&hDmaDfsdm[mic_num]);
/* DMA IRQ Channel configuration */
HAL_NVIC_SetPriority(AUDIO_DFSDM_DMAx_MIC_IRQHandler[mic_num], AUDIO_IN_IRQ_PREPRIO, 0);
HAL_NVIC_EnableIRQ(AUDIO_DFSDM_DMAx_MIC_IRQHandler[mic_num]);
}
}
/**
* @brief DeInitializes the DFSDM filter MSP.
*/
static void DFSDMx_FilterMspDeInit(void)
{
/* Configure the DMA Channel */
for(uint32_t i = 0; i < DFSDM_MIC_NUMBER; i++)
{
if(hDmaDfsdm[i].Instance != NULL)
{
HAL_DMA_DeInit(&hDmaDfsdm[i]);
}
}
}
/**
* @brief Initializes the Audio Codec audio interface (I2S)
* @note This function assumes that the I2S input clock
* is already configured and ready to be used.
* @param AudioFreq: Audio frequency to be configured for the I2S peripheral.
*/
static void I2Sx_In_Init(uint32_t AudioFreq)
{
/* Initialize the hAudioInI2s and haudio_in_i2sext Instance parameters */
haudio_i2s.Instance = AUDIO_IN_I2Sx;
haudio_in_i2sext.Instance = I2S3ext;
/* Disable I2S block */
__HAL_I2S_DISABLE(&haudio_i2s);
__HAL_I2S_DISABLE(&haudio_in_i2sext);
/* I2S peripheral configuration */
haudio_i2s.Init.AudioFreq = AudioFreq;
haudio_i2s.Init.ClockSource = I2S_CLOCK_PLL;
haudio_i2s.Init.CPOL = I2S_CPOL_LOW;
haudio_i2s.Init.DataFormat = I2S_DATAFORMAT_16B;
haudio_i2s.Init.MCLKOutput = I2S_MCLKOUTPUT_ENABLE;
haudio_i2s.Init.Mode = I2S_MODE_MASTER_TX;
haudio_i2s.Init.Standard = I2S_STANDARD_PHILIPS;
haudio_i2s.Init.FullDuplexMode = I2S_FULLDUPLEXMODE_ENABLE;
/* Init the I2S */
HAL_I2S_Init(&haudio_i2s);
/* I2Sext peripheral configuration */
haudio_in_i2sext.Init.AudioFreq = AudioFreq;
haudio_in_i2sext.Init.ClockSource = I2S_CLOCK_PLL;
haudio_in_i2sext.Init.CPOL = I2S_CPOL_HIGH;
haudio_in_i2sext.Init.DataFormat = I2S_DATAFORMAT_16B;
haudio_in_i2sext.Init.MCLKOutput = I2S_MCLKOUTPUT_ENABLE;
haudio_in_i2sext.Init.Mode = I2S_MODE_SLAVE_RX;
haudio_in_i2sext.Init.Standard = I2S_STANDARD_PHILIPS;
/* Init the I2Sext */
HAL_I2S_Init(&haudio_in_i2sext);
/* Enable I2S block */
__HAL_I2S_ENABLE(&haudio_i2s);
__HAL_I2S_ENABLE(&haudio_in_i2sext);
}
/**
* @brief Deinitializes the Audio Codec audio interface (I2S).
*/
static void I2Sx_In_DeInit(void)
{
/* Initialize the hAudioInI2s Instance parameter */
haudio_i2s.Instance = AUDIO_IN_I2Sx;
/* Disable I2S block */
__HAL_I2S_DISABLE(&haudio_i2s);
/* DeInit the I2S */
HAL_I2S_DeInit(&haudio_i2s);
/* Initialize the hAudioInI2s Instance parameter */
haudio_in_i2sext.Instance = I2S3ext;
/* Disable I2S block */
__HAL_I2S_DISABLE(&haudio_in_i2sext);
/* DeInit the I2S */
HAL_I2S_DeInit(&haudio_in_i2sext);
}
/**
* @}
*/
/**
* @}
*/
/**
* @}
*/
/**
* @}
*/
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