Google Git
Sign in
pigweed / third_party / github / FreeRTOS / FreeRTOS-Kernel / 9c0c37ab9b56f2c90085b78df2f58b5833e473db / . / FreeRTOS / Demo / CORTEX_M7_M4_AMP_STM32H745I_Discovery_IAR / ST_code / BSP / STM32H745I-Discovery / stm32h745i_discovery_audio.c
blob: 016cf5c8b7e70351ed2ab5a68acf9c9b02213a95 [file] [log] [blame]
/**
******************************************************************************
* @file stm32h745i_discovery_audio.c
* @author MCD Application Team
* @brief This file provides the Audio driver for the STM32H745I-DISCOVERY
* board.
@verbatim
How To use this driver:
-----------------------
+ This driver supports STM32H7xx devices on STM32H745I-DISCOVERY (MB1248) Discovery boards.
+ Call the function BSP_AUDIO_OUT_Init(
OutputDevice: physical output mode (OUTPUT_DEVICE_SPEAKER,
OUTPUT_DEVICE_HEADPHONE or OUTPUT_DEVICE_BOTH)
Volume : Initial volume to be set (0 is min (mute), 100 is max (100%)
AudioFreq : Audio frequency in Hz (8000, 16000, 22500, 32000...)
this parameter is relative to the audio file/stream type.
)
This function configures all the hardware required for the audio application (codec, I2C, SAI,
GPIOs, DMA and interrupt if needed). This function returns AUDIO_OK if configuration is OK.
If the returned value is different from AUDIO_OK or the function is stuck then the communication with
the codec has failed (try to un-plug the power or reset device in this case).
- OUTPUT_DEVICE_SPEAKER : only speaker will be set as output for the audio stream.
- OUTPUT_DEVICE_HEADPHONE: only headphones will be set as output for the audio stream.
- OUTPUT_DEVICE_BOTH : both Speaker and Headphone are used as outputs for the audio stream
at the same time.
Note. On STM32H745I-DISCOVERY SAI_DMA is configured in CIRCULAR mode. Due to this the application
does NOT need to call BSP_AUDIO_OUT_ChangeBuffer() to assure streaming.
+ Call the function BSP_AUDIO_OUT_Play(
pBuffer: pointer to the audio data file address
Size : size of the buffer to be sent in Bytes
)
to start playing (for the first time) from the audio file/stream.
+ Call the function BSP_AUDIO_OUT_Pause() to pause playing
+ Call the function BSP_AUDIO_OUT_Resume() to resume playing.
Note. After calling BSP_AUDIO_OUT_Pause() function for pause, only BSP_AUDIO_OUT_Resume() should be called
for resume (it is not allowed to call BSP_AUDIO_OUT_Play() in this case).
Note. This function should be called only when the audio file is played or paused (not stopped).
+ For each mode, you may need to implement the relative callback functions into your code.
The Callback functions are named BSP_AUDIO_OUT_XXX_CallBack() and only their prototypes are declared in
the stm32h745i_discovery_audio.h file. (refer to the example for more details on the callbacks implementations)
+ To Stop playing, to modify the volume level, the frequency, the audio frame slot,
the device output mode the mute or the stop, use the functions: BSP_AUDIO_OUT_SetVolume(),
AUDIO_OUT_SetFrequency(), BSP_AUDIO_OUT_SetAudioFrameSlot(), BSP_AUDIO_OUT_SetOutputMode(),
BSP_AUDIO_OUT_SetMute() and BSP_AUDIO_OUT_Stop().
+ Call the function BSP_AUDIO_IN_Init(
AudioFreq: Audio frequency in Hz (8000, 16000, 22500, 32000...)
this parameter is relative to the audio file/stream type.
BitRes: Bit resolution fixed to 16bit
ChnlNbr: Number of channel to be configured for the DFSDM peripheral
)
This function configures all the hardware required for the audio in application (channels,
Clock source for SAI PDM periphiral, GPIOs, DMA and interrupt if needed).
This function returns AUDIO_OK if configuration is OK.If the returned value is different from AUDIO_OK then
the configuration should be wrong.
+ Call the function BSP_AUDIO_IN_AllocScratch(
pScratch: pointer to scratch tables
size: size of scratch buffer)
This function must be called before BSP_AUDIO_IN_RECORD() to allocate buffer scratch for each DFSDM channel
and its size.
Note: These buffers scratch are used as intermidiate buffers to collect data within final record buffer.
size is the total size of the four buffers scratch; If size is 512 then the size of each is 128.
This function must be called after BSP_AUDIO_IN_Init()
+ Call the function BSP_AUDIO_IN_RECORD(
pBuf: pointer to the recorded audio data file address
Size: size of the buffer to be written in Bytes
)
to start recording from microphones.
+ Call the function BSP_AUDIO_IN_Pause() to pause recording
+ Call the function BSP_AUDIO_IN_Resume() to recording playing.
Note. After calling BSP_AUDIO_IN_Pause() function for pause, only BSP_AUDIO_IN_Resume() should be called
for resume (it is not allowed to call BSP_AUDIO_IN_RECORD() in this case).
+ Call the function BSP_AUDIO_IN_Stop() to stop recording
+ For each mode, you may need to implement the relative callback functions into your code.
The Callback functions are named BSP_AUDIO_IN_XXX_CallBack() and only their prototypes are declared in
the stm32h745i_discovery_audio.h file. (refer to the example for more details on the callbacks implementations)
+ Call the function BSP_AUDIO_IN_SelectInterface(uint32_t Interface) to select one of the two interfaces
available on the STM32H745I-Discovery board: SAI or PDM. This function is to be called before BSP_AUDIO_IN_InitEx().
+ Call the function BSP_AUDIO_IN_GetInterface() to get the current used interface.
+ Call the function BSP_AUDIO_IN_PDMToPCM_Init(uint32_t AudioFreq, uint32_t ChnlNbrIn, uint32_t ChnlNbrOut)
to init PDM filters if the libPDMFilter is used for audio data filtering.
+ Call the function BSP_AUDIO_IN_PDMToPCM(uint16_t* PDMBuf, uint16_t* PCMBuf) to filter PDM data to PCM format
if the libPDMFilter library is used for audio data filtering.
Driver architecture:
--------------------
+ This driver provides the High Audio Layer: consists of the function API exported in the stm32h745i_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 stm32h745i_discovery_audio.c file (DFSDMx_Init(), DFSDMx_DeInit(), SAIx_Init() and SAIx_DeInit())
Known Limitations:
------------------
1- If the TDM Format used to play in parallel 2 audio Stream (the first Stream is configured in codec SLOT0 and second
Stream in SLOT1) the Pause/Resume, volume and mute feature will control the both streams.
2- Parsing of audio file is not implemented (in order to determine audio file properties: Mono/Stereo, Data size,
File size, Audio Frequency, Audio Data header size ...). The configuration is fixed for the given audio file.
3- Supports only Stereo audio streaming.
4- Supports only 16-bits audio data size.
@endverbatim
******************************************************************************
* @attention
*
* <h2><center>&copy; Copyright (c) 2019 STMicroelectronics.
* All rights reserved.</center></h2>
*
* This software component is licensed by ST under BSD 3-Clause license,
* the "License"; You may not use this file except in compliance with the
* License. You may obtain a copy of the License at:
* opensource.org/licenses/BSD-3-Clause
*
******************************************************************************
*/
/* Includes ------------------------------------------------------------------*/
#include "stm32h745i_discovery_audio.h"
/** @addtogroup BSP
* @{
*/
/** @addtogroup STM32H745I_DISCOVERY
* @{
*/
/** @defgroup STM32H745I_DISCOVERY_AUDIO STM32H745I_DISCOVERY_AUDIO
* @brief This file includes the low layer driver for wm8994 Audio Codec
* available on STM32H745I-DISCOVERY discovery board(MB1248).
* @{
*/
/** @defgroup STM32H745I_DISCOVERY_AUDIO_Private_Variables Private Variables
* @{
*/
/* PLAY */
AUDIO_DrvTypeDef *audio_drv;
SAI_HandleTypeDef haudio_out_sai;
SAI_HandleTypeDef haudio_in_sai;
/* RECORD */
AUDIOIN_ContextTypeDef hAudioIn;
/* Audio in Volume value */
__IO uint16_t AudioInVolume = DEFAULT_AUDIO_IN_VOLUME;
/* PDM filters params */
PDM_Filter_Handler_t PDM_FilterHandler[2];
PDM_Filter_Config_t PDM_FilterConfig[2];
/**
* @}
*/
/** @defgroup STM32H745I_DISCOVERY_AUDIO_Private_Function_Prototypes OUT Private Function Prototypes
* @{
*/
static void SAIx_Out_Init(uint32_t SaiOutMode, uint32_t SlotActive, uint32_t AudioFreq);
static void SAIx_Out_DeInit(SAI_HandleTypeDef *hsai);
/**
* @}
*/
/** @defgroup STM32H745I_DISCOVERY_AUDIO_IN_Private_Function_Prototypes IN Private Function Prototypes
* @{
*/
static void SAIx_In_MspInit(SAI_HandleTypeDef *hsai, void *Params);
static void SAIx_In_MspDeInit(SAI_HandleTypeDef *hsai, void *Params);
static void SAIx_In_Init(uint32_t SaiInMode, uint32_t SlotActive, uint32_t AudioFreq);
static void SAIx_In_DeInit(SAI_HandleTypeDef *hsai);
/**
* @}
*/
/** @defgroup STM32H745I_DISCOVERY_AUDIO_OUT_Exported_Functions OUT Exported Functions
* @{
*/
/**
* @brief Configures the audio Out peripheral.
* @param OutputDevice: OUTPUT_DEVICE_SPEAKER, OUTPUT_DEVICE_HEADPHONE,
* or OUTPUT_DEVICE_BOTH.
* @param Volume: Initial volume level (from 0 (Mute) to 100 (Max))
* @param AudioFreq: Audio frequency used to play the audio stream.
* @retval AUDIO_OK if correct communication, else wrong communication
*/
uint8_t BSP_AUDIO_OUT_Init(uint16_t OutputDevice, uint8_t Volume, uint32_t AudioFreq)
{
uint8_t ret = AUDIO_ERROR;
uint32_t deviceid = 0x00;
uint32_t slot_active;
/* Initialize SAI2 sub_block A as MASTER TX */
haudio_out_sai.Instance = AUDIO_OUT_SAIx;
/* Disable SAI */
SAIx_Out_DeInit(&haudio_out_sai);
/* PLL clock is set depending by the AudioFreq (44.1khz vs 48khz groups) */
BSP_AUDIO_OUT_ClockConfig(&haudio_out_sai, AudioFreq, NULL);
/* SAI data transfer preparation:
Prepare the Media to be used for the audio transfer from memory to SAI peripheral */
if(HAL_SAI_GetState(&haudio_out_sai) == HAL_SAI_STATE_RESET)
{
/* Init the SAI MSP: this __weak function can be redefined by the application*/
BSP_AUDIO_OUT_MspInit(&haudio_out_sai, NULL);
}
/* Init SAI as master RX output */
slot_active = CODEC_AUDIOFRAME_SLOT_0123;
SAIx_Out_Init(SAI_MODEMASTER_TX, slot_active, AudioFreq);
/* wm8994 codec initialization */
deviceid = wm8994_drv.ReadID(AUDIO_I2C_ADDRESS);
if((deviceid) == WM8994_ID)
{
/* Reset the Codec Registers */
wm8994_drv.Reset(AUDIO_I2C_ADDRESS);
/* Initialize the audio driver structure */
audio_drv = &wm8994_drv;
ret = AUDIO_OK;
}
else
{
ret = AUDIO_ERROR;
}
if(ret == AUDIO_OK)
{
/* Initialize the codec internal registers */
audio_drv->Init(AUDIO_I2C_ADDRESS, OutputDevice, Volume, AudioFreq);
}
return ret;
}
/**
* @brief Starts playing audio stream from a data buffer for a determined size.
* @param pBuffer: Pointer to the buffer
* @param Size: Number of audio data BYTES.
* @retval AUDIO_OK if correct communication, else wrong communication
*/
uint8_t BSP_AUDIO_OUT_Play(uint16_t* pBuffer, uint32_t Size)
{
/* Call the audio Codec Play function */
if(audio_drv->Play(AUDIO_I2C_ADDRESS, pBuffer, Size) != 0)
{
return AUDIO_ERROR;
}
else
{
/* Update the Media layer and enable it for play */
HAL_SAI_Transmit_DMA(&haudio_out_sai, (uint8_t*) pBuffer, DMA_MAX(Size / AUDIODATA_SIZE));
return AUDIO_OK;
}
}
/**
* @brief Sends n-Bytes on the SAI interface.
* @param pData: pointer on data address
* @param Size: number of data to be written
* @retval None
*/
void BSP_AUDIO_OUT_ChangeBuffer(uint16_t *pData, uint16_t Size)
{
HAL_SAI_Transmit_DMA(&haudio_out_sai, (uint8_t*) pData, Size);
}
/**
* @brief This function Pauses the audio file stream. In case
* of using DMA, the DMA Pause feature is used.
* @warning When calling BSP_AUDIO_OUT_Pause() function for pause, only
* BSP_AUDIO_OUT_Resume() function should be called for resume (use of BSP_AUDIO_OUT_Play()
* function for resume could lead to unexpected behaviour).
* @retval AUDIO_OK if correct communication, else wrong communication
*/
uint8_t BSP_AUDIO_OUT_Pause(void)
{
/* Call the Audio Codec Pause/Resume function */
if(audio_drv->Pause(AUDIO_I2C_ADDRESS) != 0)
{
return AUDIO_ERROR;
}
else
{
/* Call the Media layer pause function */
HAL_SAI_DMAPause(&haudio_out_sai);
/* Return AUDIO_OK when all operations are correctly done */
return AUDIO_OK;
}
}
/**
* @brief Resumes the audio file stream.
* @warning When calling BSP_AUDIO_OUT_Pause() function for pause, only
* BSP_AUDIO_OUT_Resume() function should be called for resume (use of BSP_AUDIO_OUT_Play()
* function for resume could lead to unexpected behaviour).
* @retval AUDIO_OK if correct communication, else wrong communication
*/
uint8_t BSP_AUDIO_OUT_Resume(void)
{
/* Call the Audio Codec Pause/Resume function */
if(audio_drv->Resume(AUDIO_I2C_ADDRESS) != 0)
{
return AUDIO_ERROR;
}
else
{
/* Call the Media layer pause/resume function */
HAL_SAI_DMAResume(&haudio_out_sai);
/* Return AUDIO_OK when all operations are correctly done */
return AUDIO_OK;
}
}
/**
* @brief Stops audio playing and Power down the Audio Codec.
* @param Option: could be one of the following parameters
* - CODEC_PDWN_SW: for software power off (by writing registers).
* Then no need to reconfigure the Codec after power on.
* - CODEC_PDWN_HW: completely shut down the codec (physically).
* Then need to reconfigure the Codec after power on.
* @retval AUDIO_OK if correct communication, else wrong communication
*/
uint8_t BSP_AUDIO_OUT_Stop(uint32_t Option)
{
/* Call the Media layer stop function */
HAL_SAI_DMAStop(&haudio_out_sai);
/* Call Audio Codec Stop function */
if(audio_drv->Stop(AUDIO_I2C_ADDRESS, Option) != 0)
{
return AUDIO_ERROR;
}
else
{
if(Option == CODEC_PDWN_HW)
{
/* Wait at least 100us */
HAL_Delay(1);
}
/* Return AUDIO_OK when all operations are correctly done */
return AUDIO_OK;
}
}
/**
* @brief Controls the current audio volume level.
* @param Volume: Volume level to be set in percentage from 0% to 100% (0 for
* Mute and 100 for Max volume level).
* @retval AUDIO_OK if correct communication, else wrong communication
*/
uint8_t BSP_AUDIO_OUT_SetVolume(uint8_t Volume)
{
/* Call the codec volume control function with converted volume value */
if(audio_drv->SetVolume(AUDIO_I2C_ADDRESS, Volume) != 0)
{
return AUDIO_ERROR;
}
else
{
/* Return AUDIO_OK when all operations are correctly done */
return AUDIO_OK;
}
}
/**
* @brief Enables or disables the MUTE mode by software
* @param Cmd: Could be AUDIO_MUTE_ON to mute sound or AUDIO_MUTE_OFF to
* unmute the codec and restore previous volume level.
* @retval AUDIO_OK if correct communication, else wrong communication
*/
uint8_t BSP_AUDIO_OUT_SetMute(uint32_t Cmd)
{
/* Call the Codec Mute function */
if(audio_drv->SetMute(AUDIO_I2C_ADDRESS, Cmd) != 0)
{
return AUDIO_ERROR;
}
else
{
/* Return AUDIO_OK when all operations are correctly done */
return AUDIO_OK;
}
}
/**
* @brief Switch dynamically (while audio file is played) the output target
* (speaker or headphone).
* @param Output: The audio output target: OUTPUT_DEVICE_SPEAKER,
* OUTPUT_DEVICE_HEADPHONE or OUTPUT_DEVICE_BOTH
* @retval AUDIO_OK if correct communication, else wrong communication
*/
uint8_t BSP_AUDIO_OUT_SetOutputMode(uint8_t Output)
{
/* Call the Codec output device function */
if(audio_drv->SetOutputMode(AUDIO_I2C_ADDRESS, Output) != 0)
{
return AUDIO_ERROR;
}
else
{
/* Return AUDIO_OK when all operations are correctly done */
return AUDIO_OK;
}
}
/**
* @brief Updates the audio frequency.
* @param AudioFreq: Audio frequency used to play the audio stream.
* @note This API should be called after the BSP_AUDIO_OUT_Init() to adjust the
* audio frequency.
* @retval None
*/
void BSP_AUDIO_OUT_SetFrequency(uint32_t AudioFreq)
{
/* PLL clock is set depending by the AudioFreq (44.1khz vs 48khz groups) */
BSP_AUDIO_OUT_ClockConfig(&haudio_out_sai, AudioFreq, NULL);
/* Disable SAI peripheral to allow access to SAI internal registers */
__HAL_SAI_DISABLE(&haudio_out_sai);
/* Update the SAI audio frequency configuration */
haudio_out_sai.Init.AudioFrequency = AudioFreq;
HAL_SAI_Init(&haudio_out_sai);
/* Enable SAI peripheral to generate MCLK */
__HAL_SAI_ENABLE(&haudio_out_sai);
}
/**
* @brief Updates the Audio frame slot configuration.
* @param AudioFrameSlot: specifies the audio Frame slot
* @note This API should be called after the BSP_AUDIO_OUT_Init() to adjust the
* audio frame slot.
* @retval None
*/
void BSP_AUDIO_OUT_SetAudioFrameSlot(uint32_t AudioFrameSlot)
{
/* Disable SAI peripheral to allow access to SAI internal registers */
__HAL_SAI_DISABLE(&haudio_out_sai);
/* Update the SAI audio frame slot configuration */
haudio_out_sai.SlotInit.SlotActive = AudioFrameSlot;
HAL_SAI_Init(&haudio_out_sai);
/* Enable SAI peripheral to generate MCLK */
__HAL_SAI_ENABLE(&haudio_out_sai);
}
/**
* @brief De-initializes the audio out peripheral.
* @retval None
*/
void BSP_AUDIO_OUT_DeInit(void)
{
SAIx_Out_DeInit(&haudio_out_sai);
/* DeInit the SAI MSP : this __weak function can be rewritten by the application */
BSP_AUDIO_OUT_MspDeInit(&haudio_out_sai, NULL);
}
/**
* @brief Manages the DMA full Transfer complete event.
* @retval None
*/
__weak void BSP_AUDIO_OUT_TransferComplete_CallBack(void)
{
}
/**
* @brief Manages the DMA Half Transfer complete event.
* @retval None
*/
__weak void BSP_AUDIO_OUT_HalfTransfer_CallBack(void)
{
}
/**
* @brief Manages the DMA FIFO error event.
* @retval None
*/
__weak void BSP_AUDIO_OUT_Error_CallBack(void)
{
}
/**
* @brief Initializes BSP_AUDIO_OUT MSP.
* @param hsai: SAI handle
* @param Params: pointer on additional configuration parameters, can be NULL.
* @retval None
*/
__weak void BSP_AUDIO_OUT_MspInit(SAI_HandleTypeDef *hsai, void *Params)
{
static DMA_HandleTypeDef hdma_sai_tx;
GPIO_InitTypeDef gpio_init_structure;
/* Enable SAI clock */
AUDIO_OUT_SAIx_CLK_ENABLE();
/* CODEC_SAI pins configuration: FS, SCK and SD pins */
/* Enable FS, SCK and SD clocks */
AUDIO_OUT_SAIx_SD_FS_CLK_ENABLE();
/* Enable FS, SCK and SD pins */
gpio_init_structure.Pin = AUDIO_OUT_SAIx_FS_PIN | AUDIO_OUT_SAIx_SCK_PIN | AUDIO_OUT_SAIx_SD_PIN;
gpio_init_structure.Mode = GPIO_MODE_AF_PP;
gpio_init_structure.Pull = GPIO_NOPULL;
gpio_init_structure.Speed = GPIO_SPEED_FREQ_VERY_HIGH;
gpio_init_structure.Alternate = AUDIO_OUT_SAIx_AF;
HAL_GPIO_Init(AUDIO_OUT_SAIx_SD_FS_SCK_GPIO_PORT, &gpio_init_structure);
/* Enable MCLK clock */
AUDIO_OUT_SAIx_MCLK_ENABLE();
/* Enable MCLK pin */
gpio_init_structure.Pin = AUDIO_OUT_SAIx_MCLK_PIN;
HAL_GPIO_Init(AUDIO_OUT_SAIx_MCLK_GPIO_PORT, &gpio_init_structure);
/* Enable the DMA clock */
AUDIO_OUT_SAIx_DMAx_CLK_ENABLE();
if(hsai->Instance == AUDIO_OUT_SAIx)
{
/* Configure the hdma_saiTx handle parameters */
hdma_sai_tx.Init.Request = AUDIO_OUT_SAIx_DMAx_REQUEST;
hdma_sai_tx.Init.Direction = DMA_MEMORY_TO_PERIPH;
hdma_sai_tx.Init.PeriphInc = DMA_PINC_DISABLE;
hdma_sai_tx.Init.MemInc = DMA_MINC_ENABLE;
hdma_sai_tx.Init.PeriphDataAlignment = AUDIO_OUT_SAIx_DMAx_PERIPH_DATA_SIZE;
hdma_sai_tx.Init.MemDataAlignment = AUDIO_OUT_SAIx_DMAx_MEM_DATA_SIZE;
hdma_sai_tx.Init.Mode = DMA_CIRCULAR;
hdma_sai_tx.Init.Priority = DMA_PRIORITY_HIGH;
hdma_sai_tx.Init.FIFOMode = DMA_FIFOMODE_ENABLE;
hdma_sai_tx.Init.FIFOThreshold = DMA_FIFO_THRESHOLD_FULL;
hdma_sai_tx.Init.MemBurst = DMA_MBURST_SINGLE;
hdma_sai_tx.Init.PeriphBurst = DMA_PBURST_SINGLE;
hdma_sai_tx.Instance = AUDIO_OUT_SAIx_DMAx_STREAM;
/* Associate the DMA handle */
__HAL_LINKDMA(hsai, hdmatx, hdma_sai_tx);
/* Deinitialize the Stream for new transfer */
HAL_DMA_DeInit(&hdma_sai_tx);
/* Configure the DMA Stream */
HAL_DMA_Init(&hdma_sai_tx);
}
/* SAI DMA IRQ Channel configuration */
HAL_NVIC_SetPriority(AUDIO_OUT_SAIx_DMAx_IRQ, AUDIO_OUT_IRQ_PREPRIO, 0);
HAL_NVIC_EnableIRQ(AUDIO_OUT_SAIx_DMAx_IRQ);
}
/**
* @brief Deinitializes SAI MSP.
* @param hsai: SAI handle
* @param Params: pointer on additional configuration parameters, can be NULL.
* @retval None
*/
__weak void BSP_AUDIO_OUT_MspDeInit(SAI_HandleTypeDef *hsai, void *Params)
{
GPIO_InitTypeDef gpio_init_structure;
/* SAI DMA IRQ Channel deactivation */
HAL_NVIC_DisableIRQ(AUDIO_OUT_SAIx_DMAx_IRQ);
if(hsai->Instance == AUDIO_OUT_SAIx)
{
/* Deinitialize the DMA stream */
HAL_DMA_DeInit(hsai->hdmatx);
}
/* Disable SAI peripheral */
__HAL_SAI_DISABLE(hsai);
/* Deactivates CODEC_SAI pins FS, SCK, MCK and SD by putting them in input mode */
gpio_init_structure.Pin = AUDIO_OUT_SAIx_FS_PIN | AUDIO_OUT_SAIx_SCK_PIN | AUDIO_OUT_SAIx_SD_PIN;
HAL_GPIO_DeInit(AUDIO_OUT_SAIx_SD_FS_SCK_GPIO_PORT, gpio_init_structure.Pin);
gpio_init_structure.Pin = AUDIO_OUT_SAIx_MCLK_PIN;
HAL_GPIO_DeInit(AUDIO_OUT_SAIx_MCLK_GPIO_PORT, gpio_init_structure.Pin);
/* Disable SAI clock */
AUDIO_OUT_SAIx_CLK_DISABLE();
/* GPIO pins clock and DMA clock can be shut down in the applic
by surcharging this __weak function */
}
/**
* @brief Clock Config.
* @param hsai: might be required to set audio peripheral predivider if any.
* @param AudioFreq: Audio frequency used to play the audio stream.
* @param Params: pointer on additional configuration parameters, can be NULL.
* @note This API is called by BSP_AUDIO_OUT_Init() and BSP_AUDIO_OUT_SetFrequency()
* Being __weak it can be overwritten by the application
* @retval None
*/
__weak void BSP_AUDIO_OUT_ClockConfig(SAI_HandleTypeDef *hsai, uint32_t AudioFreq, void *Params)
{
RCC_PeriphCLKInitTypeDef rcc_ex_clk_init_struct;
HAL_RCCEx_GetPeriphCLKConfig(&rcc_ex_clk_init_struct);
/* Set the PLL configuration according to the audio frequency */
if((AudioFreq == AUDIO_FREQUENCY_11K) || (AudioFreq == AUDIO_FREQUENCY_22K) || (AudioFreq == AUDIO_FREQUENCY_44K))
{
/* SAI clock config:
PLL2_VCO Input = HSE_VALUE/PLL2M = 1 Mhz
PLL2_VCO Output = PLL2_VCO Input * PLL2N = 429 Mhz
SAI_CLK_x = PLL2_VCO Output/PLL2P = 429/38 = 11.289 Mhz */
rcc_ex_clk_init_struct.PeriphClockSelection = RCC_PERIPHCLK_SAI2;
rcc_ex_clk_init_struct.Sai23ClockSelection = RCC_SAI2CLKSOURCE_PLL2;
rcc_ex_clk_init_struct.PLL2.PLL2P = 38;
rcc_ex_clk_init_struct.PLL2.PLL2Q = 1;
rcc_ex_clk_init_struct.PLL2.PLL2R = 1;
rcc_ex_clk_init_struct.PLL2.PLL2N = 429;
rcc_ex_clk_init_struct.PLL2.PLL2M = 25;
HAL_RCCEx_PeriphCLKConfig(&rcc_ex_clk_init_struct);
}
else /* AUDIO_FREQUENCY_8K, AUDIO_FREQUENCY_16K, AUDIO_FREQUENCY_48K, AUDIO_FREQUENCY_96K */
{
/* SAI clock config:
PLL2_VCO Input = HSE_VALUE/PLL2M = 1 Mhz
PLL2_VCO Output = PLL2_VCO Input * PLL2N = 344 Mhz
SAI_CLK_x = PLL2_VCO Output/PLL2P = 344/7 = 49.142 Mhz */
rcc_ex_clk_init_struct.PeriphClockSelection = RCC_PERIPHCLK_SAI2;
rcc_ex_clk_init_struct.Sai23ClockSelection = RCC_SAI2CLKSOURCE_PLL2;
rcc_ex_clk_init_struct.PLL2.PLL2P = 7;
rcc_ex_clk_init_struct.PLL2.PLL2Q = 1;
rcc_ex_clk_init_struct.PLL2.PLL2R = 1;
rcc_ex_clk_init_struct.PLL2.PLL2N = 344;
rcc_ex_clk_init_struct.PLL2.PLL2M = 25;
HAL_RCCEx_PeriphCLKConfig(&rcc_ex_clk_init_struct);
}
}
/**
* @}
*/
/** @defgroup STM32H745I_DISCOVERY_AUDIO_OUT_Private_Functions OUT Private Functions
* @{
*/
/*******************************************************************************
HAL Callbacks
*******************************************************************************/
/**
* @brief Tx Transfer completed callbacks.
* @param hsai: SAI handle
* @retval None
*/
void HAL_SAI_TxCpltCallback(SAI_HandleTypeDef *hsai)
{
/* Manage the remaining file size and new address offset: This function
should be coded by user (its prototype is already declared in stm32h745i_discovery_audio.h) */
BSP_AUDIO_OUT_TransferComplete_CallBack();
}
/**
* @brief Tx Half Transfer completed callbacks.
* @param hsai: SAI handle
* @retval None
*/
void HAL_SAI_TxHalfCpltCallback(SAI_HandleTypeDef *hsai)
{
/* Manage the remaining file size and new address offset: This function
should be coded by user (its prototype is already declared in stm32h745i_discovery_audio.h) */
BSP_AUDIO_OUT_HalfTransfer_CallBack();
}
/**
* @brief SAI error callbacks.
* @param hsai: SAI handle
* @retval None
*/
void HAL_SAI_ErrorCallback(SAI_HandleTypeDef *hsai)
{
if(hsai->Instance == AUDIO_OUT_SAIx)
{
BSP_AUDIO_OUT_Error_CallBack();
}
else
{
BSP_AUDIO_IN_Error_CallBack();
}
}
/*******************************************************************************
Static Functions
*******************************************************************************/
/**
* @brief Initializes the Audio Codec audio interface (SAI).
* @param SaiOutMode: Audio mode to be configured for the SAI peripheral.
* @param SlotActive: Audio active slot to be configured for the SAI peripheral.
* @param AudioFreq: Audio frequency to be configured for the SAI peripheral.
* @note The default SlotActive configuration is set to CODEC_AUDIOFRAME_SLOT_0123
* and user can update this configuration using
* @retval None
*/
static void SAIx_Out_Init(uint32_t SaiOutMode, uint32_t SlotActive, uint32_t AudioFreq)
{
/* Disable SAI peripheral to allow access to SAI internal registers */
__HAL_SAI_DISABLE(&haudio_out_sai);
/* Configure SAI_Block_x
LSBFirst: Disabled
DataSize: 16 */
haudio_out_sai.Init.MonoStereoMode = SAI_STEREOMODE;
haudio_out_sai.Init.AudioFrequency = AudioFreq;
haudio_out_sai.Init.AudioMode = SaiOutMode;
haudio_out_sai.Init.NoDivider = SAI_MASTERDIVIDER_ENABLE;
haudio_out_sai.Init.Protocol = SAI_FREE_PROTOCOL;
haudio_out_sai.Init.DataSize = SAI_DATASIZE_16;
haudio_out_sai.Init.FirstBit = SAI_FIRSTBIT_MSB;
haudio_out_sai.Init.ClockStrobing = SAI_CLOCKSTROBING_RISINGEDGE;
haudio_out_sai.Init.Synchro = SAI_ASYNCHRONOUS;
haudio_out_sai.Init.OutputDrive = SAI_OUTPUTDRIVE_ENABLE;
haudio_out_sai.Init.FIFOThreshold = SAI_FIFOTHRESHOLD_1QF;
haudio_out_sai.Init.SynchroExt = SAI_SYNCEXT_DISABLE;
haudio_out_sai.Init.CompandingMode = SAI_NOCOMPANDING;
haudio_out_sai.Init.TriState = SAI_OUTPUT_NOTRELEASED;
haudio_out_sai.Init.Mckdiv = 0;
haudio_out_sai.Init.MckOverSampling = SAI_MCK_OVERSAMPLING_DISABLE;
haudio_out_sai.Init.PdmInit.Activation = DISABLE;
haudio_out_sai.Init.PdmInit.ClockEnable = 0;
haudio_out_sai.Init.PdmInit.MicPairsNbr = 0;
/* Configure SAI_Block_x Frame
Frame Length: 64
Frame active Length: 32
FS Definition: Start frame + Channel Side identification
FS Polarity: FS active Low
FS Offset: FS asserted one bit before the first bit of slot 0 */
haudio_out_sai.FrameInit.FrameLength = 128;
haudio_out_sai.FrameInit.ActiveFrameLength = 64;
haudio_out_sai.FrameInit.FSDefinition = SAI_FS_CHANNEL_IDENTIFICATION;
haudio_out_sai.FrameInit.FSPolarity = SAI_FS_ACTIVE_LOW;
haudio_out_sai.FrameInit.FSOffset = SAI_FS_BEFOREFIRSTBIT;
/* Configure SAI Block_x Slot
Slot First Bit Offset: 0
Slot Size : 16
Slot Number: 4
Slot Active: All slot actives */
haudio_out_sai.SlotInit.FirstBitOffset = 0;
haudio_out_sai.SlotInit.SlotSize = SAI_SLOTSIZE_DATASIZE;
haudio_out_sai.SlotInit.SlotNumber = 4;
haudio_out_sai.SlotInit.SlotActive = SlotActive;
HAL_SAI_Init(&haudio_out_sai);
/* Enable SAI peripheral to generate MCLK */
__HAL_SAI_ENABLE(&haudio_out_sai);
}
/**
* @brief Deinitializes the Audio Codec audio interface (SAI).
* @retval None
*/
static void SAIx_Out_DeInit(SAI_HandleTypeDef *hsai)
{
/* Disable SAI peripheral */
__HAL_SAI_DISABLE(hsai);
HAL_SAI_DeInit(hsai);
}
/**
* @}
*/
/** @defgroup STM32H745I_DISCOVERY_AUDIO_IN_Exported_Functions IN Exported Functions
* @{
*/
/**
* @brief Initialize wave recording.
* @param AudioFreq: Audio frequency to be configured for the DFSDM peripheral.
* @param BitRes: Audio frequency to be configured for the DFSDM peripheral.
* @param ChnlNbr: Audio frequency to be configured for the DFSDM peripheral.
* @retval AUDIO_OK if correct communication, else wrong communication
*/
uint8_t BSP_AUDIO_IN_Init(uint32_t AudioFreq, uint32_t BitRes, uint32_t ChnlNbr)
{
/* Set audio in interface to default one */
BSP_AUDIO_IN_SelectInterface(AUDIO_IN_INTERFACE_PDM);
return BSP_AUDIO_IN_InitEx(INPUT_DEVICE_DIGITAL_MIC, AudioFreq, BitRes, ChnlNbr);
}
/**
* @brief Initialize wave recording.
* @param InputDevice: INPUT_DEVICE_DIGITAL_MIC or INPUT_DEVICE_ANALOG_MIC.
* @param AudioFreq: Audio frequency to be configured.
* @param BitRes: Audio bit resolution to be configured..
* @param ChnlNbr: Number of channel to be configured.
* @retval AUDIO_OK if correct communication, else wrong communication
*/
uint8_t BSP_AUDIO_IN_InitEx(uint16_t InputDevice, uint32_t AudioFreq, uint32_t BitRes, uint32_t ChnlNbr)
{
uint8_t ret = AUDIO_OK;
uint32_t slot_active;
/* Store the audio record context */
hAudioIn.Frequency = AudioFreq;
hAudioIn.BitResolution = BitRes;
hAudioIn.InputDevice = InputDevice;
hAudioIn.ChannelNbr = ChnlNbr;
if(hAudioIn.InputDevice == INPUT_DEVICE_DIGITAL_MIC)
{
if(hAudioIn.Interface == AUDIO_IN_INTERFACE_SAI)
{
/* Initialize SAI2 block B as SLAVE RX synchrounous with SAI2 block A */
haudio_in_sai.Instance = AUDIO_IN_SAIx;
/* Disable SAI */
SAIx_In_DeInit(&haudio_in_sai);
/* PLL clock is set depending on the AudioFreq (44.1khz vs 48khz groups) */
BSP_AUDIO_IN_ClockConfig(AudioFreq, NULL); /* Clock config is shared between AUDIO IN and OUT */
/* SAI data transfer preparation:
Prepare the Media to be used for the audio transfer from SAI peripheral to memory */
if(HAL_SAI_GetState(&haudio_in_sai) == HAL_SAI_STATE_RESET)
{
/* Init the SAI MSP: this __weak function can be redefined by the application*/
BSP_AUDIO_IN_MspInit();
}
/* Configure SAI in master mode :
* - SAI2_block_B in slave RX mode synchronous from SAI2_block_A
*/
slot_active = CODEC_AUDIOFRAME_SLOT_13;
SAIx_In_Init(SAI_MODESLAVE_RX, slot_active, AudioFreq);
}
else if(hAudioIn.Interface == AUDIO_IN_INTERFACE_PDM)
{
/* Initialize SAI2 block A as MASTER RX */
haudio_in_sai.Instance = AUDIO_IN_SAI_PDMx;
/* Disable SAI */
SAIx_In_DeInit(&haudio_in_sai);
/* PLL clock is set depending on the AudioFreq (44.1khz vs 48khz groups) */
BSP_AUDIO_IN_ClockConfig(AudioFreq, NULL);
/* SAI data transfer preparation:
Prepare the Media to be used for the audio transfer from SAI peripheral to memory */
/* Initialize the haudio_in_sai Instance parameter */
if(HAL_SAI_GetState(&haudio_in_sai) == HAL_SAI_STATE_RESET)
{
/* Init the SAI MSP: this __weak function can be redefined by the application*/
BSP_AUDIO_IN_MspInit();
}
/* Configure SAI in master mode :
* - SAI4_block_A in master RX mode
*/
slot_active = CODEC_AUDIOFRAME_SLOT_1;
SAIx_In_Init(SAI_MODEMASTER_RX, slot_active, AudioFreq);
if(BSP_AUDIO_IN_PDMToPCM_Init(AudioFreq, hAudioIn.ChannelNbr, hAudioIn.ChannelNbr) != AUDIO_OK)
{
ret = AUDIO_ERROR;
}
}
else
{
ret = AUDIO_ERROR;
}
}
else
{
/* Analog Input */
ret = AUDIO_ERROR;
}
/* Return AUDIO_OK when all operations are correctly done */
return ret;
}
/**
* @brief Initializes wave recording and playback in parallel.
* @param InputDevice: INPUT_DEVICE_DIGITAL_MICROPHONE_2
* @param OutputDevice: OUTPUT_DEVICE_SPEAKER, OUTPUT_DEVICE_HEADPHONE,
* or OUTPUT_DEVICE_BOTH.
* @param AudioFreq: Audio frequency to be configured for the SAI peripheral.
* @param BitRes: Audio frequency to be configured.
* @param ChnlNbr: Channel number.
* @retval AUDIO_OK if correct communication, else wrong communication
*/
uint8_t BSP_AUDIO_IN_OUT_Init(uint32_t InputDevice, uint32_t OutputDevice, uint32_t AudioFreq, uint32_t BitRes, uint32_t ChnlNbr)
{
uint32_t slot_active;
uint32_t deviceid = 0, ret = AUDIO_OK;
/* Store the audio record context */
hAudioIn.Frequency = AudioFreq;
hAudioIn.BitResolution = BitRes;
hAudioIn.InputDevice = InputDevice;
hAudioIn.ChannelNbr = ChnlNbr;
/* Input device is Digital MIC2 and Codec interface is SAI */
if (hAudioIn.InputDevice == INPUT_DEVICE_DIGITAL_MICROPHONE_2)
{
haudio_in_sai.Instance = AUDIO_IN_SAIx;
haudio_out_sai.Instance = AUDIO_OUT_SAIx;
/* PLL clock is set depending on the AudioFreq (44.1khz vs 48khz groups) */
BSP_AUDIO_OUT_ClockConfig(&haudio_in_sai, AudioFreq, NULL);
/* SAI data transfer preparation:
Prepare the Media to be used for the audio transfer from SAI peripheral to memory */
if(HAL_SAI_GetState(&haudio_in_sai) == HAL_SAI_STATE_RESET)
{
/* Init the SAI MSP: this __weak function can be redefined by the application*/
BSP_AUDIO_IN_MspInit();
}
/* SAI data transfer preparation:
Prepare the Media to be used for the audio transfer from memory to SAI peripheral */
if(HAL_SAI_GetState(&haudio_out_sai) == HAL_SAI_STATE_RESET)
{
/* Init the SAI MSP: this __weak function can be redefined by the application*/
BSP_AUDIO_OUT_MspInit(&haudio_out_sai, NULL);
}
/* Configure SAI in master TX mode :
* - SAI2_block_A in master TX mode
* - SAI2_block_B in slave RX mode synchronous from SAI2_block_A
*/
slot_active = CODEC_AUDIOFRAME_SLOT_13;
SAIx_In_Init(SAI_MODESLAVE_RX, slot_active, AudioFreq);
slot_active = CODEC_AUDIOFRAME_SLOT_02;
SAIx_Out_Init(SAI_MODEMASTER_TX, slot_active, AudioFreq);
/* wm8994 codec initialization */
deviceid = wm8994_drv.ReadID(AUDIO_I2C_ADDRESS);
if((deviceid) == WM8994_ID)
{
/* Reset the Codec Registers */
wm8994_drv.Reset(AUDIO_I2C_ADDRESS);
/* Initialize the audio driver structure */
audio_drv = &wm8994_drv;
ret = AUDIO_OK;
}
else
{
ret = AUDIO_ERROR;
}
if(ret == AUDIO_OK)
{
/* Initialize the codec internal registers */
audio_drv->Init(AUDIO_I2C_ADDRESS, InputDevice|OutputDevice, 90, AudioFreq);
}
}
else
{
ret = AUDIO_ERROR;
}
/* Return AUDIO_OK when all operations are correctly done */
return ret;
}
/**
* @brief Link digital mic to specified source
* @param Interface : Audio In interface for Digital mic. It can be:
* AUDIO_IN_INTERFACE_SAI
* AUDIO_IN_INTERFACE_PDM
* @retval None
*/
void BSP_AUDIO_IN_SelectInterface(uint32_t Interface)
{
hAudioIn.Interface = Interface;
}
/**
* @brief Get digital mic interface
* @retval Digital mic interface.
*/
uint32_t BSP_AUDIO_IN_GetInterface(void)
{
return (hAudioIn.Interface);
}
/**
* @brief Return audio in channel number
* @retval Number of channel
*/
uint8_t BSP_AUDIO_IN_GetChannelNumber(void)
{
return hAudioIn.ChannelNbr;
}
/**
* @brief Start audio recording.
* @param pBuf: Main buffer pointer for the recorded data storing
* @param size: Current size of the recorded buffer
* @retval AUDIO_OK if correct communication, else wrong communication
*/
uint8_t BSP_AUDIO_IN_Record(uint16_t *pBuf, uint32_t size)
{
/* Start the process receive DMA */
if(HAL_OK != HAL_SAI_Receive_DMA(&haudio_in_sai, (uint8_t*)pBuf, size))
{
return AUDIO_ERROR;
}
/* Return AUDIO_OK when all operations are correctly done */
return AUDIO_OK;
}
/**
* @brief Stop audio recording.
* @retval AUDIO_OK if correct communication, else wrong communication
*/
uint8_t BSP_AUDIO_IN_Stop(void)
{
/* Call the Media layer stop function */
HAL_SAI_DMAStop(&haudio_in_sai);
/* Return AUDIO_OK when all operations are correctly done */
return AUDIO_OK;
}
/**
* @brief Pause the audio file stream.
* @retval AUDIO_OK if correct communication, else wrong communication
*/
uint8_t BSP_AUDIO_IN_Pause(void)
{
if (hAudioIn.InputDevice == INPUT_DEVICE_ANALOG_MIC)
{
return AUDIO_ERROR;
}
else
{
/* Call the Media layer pause function */
HAL_SAI_DMAPause(&haudio_in_sai);
}
/* Return AUDIO_OK when all operations are correctly done */
return AUDIO_OK;
}
/**
* @brief Resume the audio file stream.
* @retval AUDIO_OK if correct communication, else wrong communication
*/
uint8_t BSP_AUDIO_IN_Resume(void)
{
if (hAudioIn.InputDevice == INPUT_DEVICE_ANALOG_MIC)
{
return AUDIO_ERROR;
}
else
{
/* Call the Media layer resume function */
HAL_SAI_DMAResume(&haudio_in_sai);
}
/* Return AUDIO_OK when all operations are correctly done */
return AUDIO_OK;
}
/**
* @brief Controls the audio in volume level.
* @param Volume: Volume level to be set in percentage from 0% to 100% (0 for
* Mute and 100 for Max volume level).
* @retval AUDIO_OK if correct communication, else wrong communication
*/
uint8_t BSP_AUDIO_IN_SetVolume(uint8_t Volume)
{
/* Set the Global variable AudioInVolume */
AudioInVolume = Volume;
/* Return AUDIO_OK when all operations are correctly done */
return AUDIO_OK;
}
/**
* @brief Deinit the audio IN peripherals.
* @retval None
*/
void BSP_AUDIO_IN_DeInit(void)
{
SAIx_In_DeInit(&haudio_in_sai);
BSP_AUDIO_IN_MspDeInit();
}
/**
* @brief Initialize the PDM library.
* @param AudioFreq: Audio sampling frequency
* @param ChnlNbrIn: Number of input audio channels in the PDM buffer
* @param ChnlNbrOut: Number of desired output audio channels in the resulting PCM buffer
* @retval None
*/
uint8_t BSP_AUDIO_IN_PDMToPCM_Init(uint32_t AudioFreq, uint32_t ChnlNbrIn, uint32_t ChnlNbrOut)
{
uint32_t index = 0;
/* Enable CRC peripheral to unlock the PDM library */
__HAL_RCC_CRC_CLK_ENABLE();
for(index = 0; index < ChnlNbrIn; index++)
{
/* Init PDM filters */
PDM_FilterHandler[index].bit_order = PDM_FILTER_BIT_ORDER_MSB;
PDM_FilterHandler[index].endianness = PDM_FILTER_ENDIANNESS_LE;
PDM_FilterHandler[index].high_pass_tap = 2122358088;
PDM_FilterHandler[index].out_ptr_channels = ChnlNbrOut;
PDM_FilterHandler[index].in_ptr_channels = ChnlNbrIn;
PDM_Filter_Init((PDM_Filter_Handler_t *)(&PDM_FilterHandler[index]));
/* PDM lib config phase */
PDM_FilterConfig[index].output_samples_number = AudioFreq/1000;
PDM_FilterConfig[index].mic_gain = 24;
PDM_FilterConfig[index].decimation_factor = PDM_FILTER_DEC_FACTOR_64;
PDM_Filter_setConfig((PDM_Filter_Handler_t *)&PDM_FilterHandler[index], &PDM_FilterConfig[index]);
}
return AUDIO_OK;
}
/**
* @brief Converts audio format from PDM to PCM.
* @param PDMBuf: Pointer to PDM buffer data
* @param PCMBuf: Pointer to PCM buffer data
* @retval AUDIO_OK in case of success, AUDIO_ERROR otherwise
*/
uint8_t BSP_AUDIO_IN_PDMToPCM(uint16_t *PDMBuf, uint16_t *PCMBuf)
{
uint32_t index = 0;
for(index = 0; index < hAudioIn.ChannelNbr; index++)
{
PDM_Filter(&((uint8_t*)(PDMBuf))[index], (uint16_t*)&(PCMBuf[index]), &PDM_FilterHandler[index]);
}
return AUDIO_OK;
}
/**
* @brief User callback when record buffer is filled.
* @retval None
*/
__weak void BSP_AUDIO_IN_TransferComplete_CallBack(void)
{
/* This function should be implemented by the user application.
It is called into this driver when the current buffer is filled
to prepare the next buffer pointer and its size. */
}
/**
* @brief Manages the DMA Half Transfer complete event.
* @retval None
*/
__weak void BSP_AUDIO_IN_HalfTransfer_CallBack(void)
{
/* This function should be implemented by the user application.
It is called into this driver when the current buffer is filled
to prepare the next buffer pointer and its size. */
}
/**
* @brief User callback when record buffer is filled.
* @param InputDevice: INPUT_DEVICE_DIGITAL_MIC1 or INPUT_DEVICE_DIGITAL_MIC2
*/
__weak void BSP_AUDIO_IN_TransferComplete_CallBackEx(uint32_t InputDevice)
{
/* This function should be implemented by the user application.
It is called into this driver when the current buffer is filled
to prepare the next buffer pointer and its size. */
}
/**
* @brief User callback when record buffer is filled.
* @param InputDevice: INPUT_DEVICE_DIGITAL_MIC1 or INPUT_DEVICE_DIGITAL_MIC2
*/
__weak void BSP_AUDIO_IN_HalfTransfer_CallBackEx(uint32_t InputDevice)
{
/* This function should be implemented by the user application.
It is called into this driver when the current buffer is filled
to prepare the next buffer pointer and its size. */
}
/**
* @brief Audio IN Error callback function.
* @retval None
*/
__weak void BSP_AUDIO_IN_Error_CallBack(void)
{
/* This function is called when an Interrupt due to transfer error on or peripheral
error occurs. */
}
/**
* @brief Initialize BSP_AUDIO_IN MSP.
* @retval None
*/
__weak void BSP_AUDIO_IN_MspInit(void)
{
SAIx_In_MspInit(&haudio_in_sai, NULL);
}
/**
* @brief DeInitialize BSP_AUDIO_IN MSP.
* @retval None
*/
__weak void BSP_AUDIO_IN_MspDeInit(void)
{
SAIx_In_MspDeInit(&haudio_in_sai, NULL);
}
/**
* @brief Clock Config.
* @param AudioFreq: Audio frequency used to play the audio stream.
* @param Params: pointer on additional configuration parameters, can be NULL.
* @note This API is called by BSP_AUDIO_IN_Init()
* Being __weak it can be overwritten by the application
* @retval None
*/
__weak void BSP_AUDIO_IN_ClockConfig(uint32_t AudioFreq, void *Params)
{
RCC_PeriphCLKInitTypeDef rcc_ex_clk_init_struct;
HAL_RCCEx_GetPeriphCLKConfig(&rcc_ex_clk_init_struct);
/* Set the PLL configuration according to the audio frequency */
if((AudioFreq == AUDIO_FREQUENCY_11K) || (AudioFreq == AUDIO_FREQUENCY_22K) || (AudioFreq == AUDIO_FREQUENCY_44K))
{
/* SAI clock config:
PLL2_VCO Input = HSE_VALUE/PLL2M = 1 Mhz
PLL2_VCO Output = PLL2_VCO Input * PLL2N = 429 Mhz
SAI_CLK_x = PLL2_VCO Output/PLL2P = 429/38 = 11.289 Mhz */
rcc_ex_clk_init_struct.PeriphClockSelection = RCC_PERIPHCLK_SAI2;
rcc_ex_clk_init_struct.Sai23ClockSelection = RCC_SAI2CLKSOURCE_PLL2;
rcc_ex_clk_init_struct.PLL2.PLL2P = 38;
rcc_ex_clk_init_struct.PLL2.PLL2Q = 1;
rcc_ex_clk_init_struct.PLL2.PLL2R = 1;
rcc_ex_clk_init_struct.PLL2.PLL2N = 429;
rcc_ex_clk_init_struct.PLL2.PLL2M = 25;
if (hAudioIn.Interface == AUDIO_IN_INTERFACE_PDM)
{
rcc_ex_clk_init_struct.PeriphClockSelection = RCC_PERIPHCLK_SAI4A;
rcc_ex_clk_init_struct.Sai4AClockSelection = RCC_SAI4ACLKSOURCE_PLL2;
}
HAL_RCCEx_PeriphCLKConfig(&rcc_ex_clk_init_struct);
}
else /* AUDIO_FREQUENCY_8K, AUDIO_FREQUENCY_16K, AUDIO_FREQUENCY_32K, AUDIO_FREQUENCY_48K, AUDIO_FREQUENCY_96K */
{
/* SAI clock config:
PLL2_VCO Input = HSE_VALUE/PLL2M = 1 Mhz
PLL2_VCO Output = PLL2_VCO Input * PLL2N = 344 Mhz
SAI_CLK_x = PLL2_VCO Output/PLL2P = 344/7 = 49.142 Mhz */
rcc_ex_clk_init_struct.PeriphClockSelection = RCC_PERIPHCLK_SAI2;
rcc_ex_clk_init_struct.Sai23ClockSelection = RCC_SAI2CLKSOURCE_PLL2;
rcc_ex_clk_init_struct.PLL2.PLL2P = 7;
rcc_ex_clk_init_struct.PLL2.PLL2Q = 1;
rcc_ex_clk_init_struct.PLL2.PLL2R = 1;
rcc_ex_clk_init_struct.PLL2.PLL2N = 344;
rcc_ex_clk_init_struct.PLL2.PLL2M = 25;
if (hAudioIn.Interface == AUDIO_IN_INTERFACE_PDM)
{
rcc_ex_clk_init_struct.PeriphClockSelection = RCC_PERIPHCLK_SAI4A;
rcc_ex_clk_init_struct.Sai4AClockSelection = RCC_SAI4ACLKSOURCE_PLL2;
}
HAL_RCCEx_PeriphCLKConfig(&rcc_ex_clk_init_struct);
}
}
/**
* @}
*/
/** @defgroup STM32H745I_DISCOVERY_AUDIO_IN_Private_Functions IN Private Functions
* @{
*/
/*******************************************************************************
HAL Callbacks
*******************************************************************************/
/**
* @brief Half reception complete callback.
* @param hsai: SAI handle.
* @retval None
*/
void HAL_SAI_RxHalfCpltCallback(SAI_HandleTypeDef *hsai)
{
/* Manage the remaining file size and new address offset: This function should be coded by user */
BSP_AUDIO_IN_HalfTransfer_CallBack();
}
/**
* @brief Reception complete callback.
* @param hsai: SAI handle.
* @retval None
*/
void HAL_SAI_RxCpltCallback(SAI_HandleTypeDef *hsai)
{
/* Call the record update function to get the next buffer to fill and its size (size is ignored) */
BSP_AUDIO_IN_TransferComplete_CallBack();
}
/*******************************************************************************
Static Functions
*******************************************************************************/
/**
* @brief Initializes SAI Audio IN MSP.
* @param hsai: SAI handle
* @param Params: pointer on additional configuration parameters, can be NULL.
* @retval None
*/
static void SAIx_In_MspInit(SAI_HandleTypeDef *hsai, void *Params)
{
static DMA_HandleTypeDef hdma_sai_rx;
GPIO_InitTypeDef gpio_init_structure;
if(hsai->Instance == AUDIO_IN_SAI_PDMx)
{
/* Enable SAI clock */
AUDIO_IN_SAI_PDMx_CLK_ENABLE();
/* Enable PDM clock */
AUDIO_IN_SAI_PDMx_CLK_IN_ENABLE();
gpio_init_structure.Pin = AUDIO_IN_SAI_PDMx_CLK_IN_PIN;
gpio_init_structure.Mode = GPIO_MODE_AF_PP;
gpio_init_structure.Pull = GPIO_NOPULL;
gpio_init_structure.Speed = GPIO_SPEED_FREQ_HIGH;
gpio_init_structure.Alternate = AUDIO_IN_SAI_PDMx_DATA_CLK_AF;
HAL_GPIO_Init(AUDIO_IN_SAI_PDMx_CLK_IN_PORT, &gpio_init_structure);
/* Enable PDM data */
AUDIO_IN_SAI_PDMx_DATA_IN_ENABLE();
gpio_init_structure.Pull = GPIO_PULLUP;
gpio_init_structure.Speed = GPIO_SPEED_FREQ_MEDIUM;
gpio_init_structure.Pin = AUDIO_IN_SAI_PDMx_DATA_IN_PIN;
HAL_GPIO_Init(AUDIO_IN_SAI_PDMx_DATA_IN_PORT, &gpio_init_structure);
/* Enable the DMA clock */
AUDIO_IN_SAI_PDMx_DMAx_CLK_ENABLE();
/* Configure the hdma_sai_rx handle parameters */
hdma_sai_rx.Init.Request = AUDIO_IN_SAI_PDMx_DMAx_REQUEST;
hdma_sai_rx.Init.Direction = DMA_PERIPH_TO_MEMORY;
hdma_sai_rx.Init.PeriphInc = DMA_PINC_DISABLE;
hdma_sai_rx.Init.MemInc = DMA_MINC_ENABLE;
hdma_sai_rx.Init.PeriphDataAlignment = AUDIO_IN_SAI_PDMx_DMAx_PERIPH_DATA_SIZE;
hdma_sai_rx.Init.MemDataAlignment = AUDIO_IN_SAI_PDMx_DMAx_MEM_DATA_SIZE;
hdma_sai_rx.Init.Mode = DMA_CIRCULAR;
hdma_sai_rx.Init.Priority = DMA_PRIORITY_HIGH;
hdma_sai_rx.Init.FIFOMode = DMA_FIFOMODE_DISABLE;
hdma_sai_rx.Init.FIFOThreshold = DMA_FIFO_THRESHOLD_FULL;
hdma_sai_rx.Init.MemBurst = DMA_MBURST_SINGLE;
hdma_sai_rx.Init.PeriphBurst = DMA_MBURST_SINGLE;
hdma_sai_rx.Instance = AUDIO_IN_SAI_PDMx_DMAx_STREAM;
/* Associate the DMA handle */
__HAL_LINKDMA(hsai, hdmarx, hdma_sai_rx);
/* Deinitialize the Stream for new transfer */
HAL_DMA_DeInit(&hdma_sai_rx);
/* Configure the DMA Stream */
HAL_DMA_Init(&hdma_sai_rx);
/* SAI DMA IRQ Channel configuration */
HAL_NVIC_SetPriority(AUDIO_IN_SAI_PDMx_DMAx_IRQ, AUDIO_IN_IRQ_PREPRIO, 0);
HAL_NVIC_EnableIRQ(AUDIO_IN_SAI_PDMx_DMAx_IRQ);
}
else
{
/* Enable SAI clock */
AUDIO_IN_SAIx_CLK_ENABLE();
/* Enable SD GPIO clock */
AUDIO_IN_SAIx_SD_ENABLE();
/* CODEC_SAI pin configuration: SD pin */
gpio_init_structure.Pin = AUDIO_IN_SAIx_SD_PIN;
gpio_init_structure.Mode = GPIO_MODE_AF_PP;
gpio_init_structure.Pull = GPIO_NOPULL;
gpio_init_structure.Speed = GPIO_SPEED_FREQ_HIGH;
gpio_init_structure.Alternate = AUDIO_IN_SAIx_AF;
HAL_GPIO_Init(AUDIO_IN_SAIx_SD_GPIO_PORT, &gpio_init_structure);
/* Enable the DMA clock */
AUDIO_IN_SAIx_DMAx_CLK_ENABLE();
/* Configure the hdma_sai_rx handle parameters */
hdma_sai_rx.Init.Request = AUDIO_IN_SAIx_DMAx_REQUEST;
hdma_sai_rx.Init.Direction = DMA_PERIPH_TO_MEMORY;
hdma_sai_rx.Init.PeriphInc = DMA_PINC_DISABLE;
hdma_sai_rx.Init.MemInc = DMA_MINC_ENABLE;
hdma_sai_rx.Init.PeriphDataAlignment = AUDIO_IN_SAIx_DMAx_PERIPH_DATA_SIZE;
hdma_sai_rx.Init.MemDataAlignment = AUDIO_IN_SAIx_DMAx_MEM_DATA_SIZE;
hdma_sai_rx.Init.Mode = DMA_CIRCULAR;
hdma_sai_rx.Init.Priority = DMA_PRIORITY_HIGH;
hdma_sai_rx.Init.FIFOMode = DMA_FIFOMODE_DISABLE;
hdma_sai_rx.Init.FIFOThreshold = DMA_FIFO_THRESHOLD_FULL;
hdma_sai_rx.Init.MemBurst = DMA_MBURST_SINGLE;
hdma_sai_rx.Init.PeriphBurst = DMA_MBURST_SINGLE;
hdma_sai_rx.Instance = AUDIO_IN_SAIx_DMAx_STREAM;
/* Associate the DMA handle */
__HAL_LINKDMA(hsai, hdmarx, hdma_sai_rx);
/* Deinitialize the Stream for new transfer */
HAL_DMA_DeInit(&hdma_sai_rx);
/* Configure the DMA Stream */
HAL_DMA_Init(&hdma_sai_rx);
/* SAI DMA IRQ Channel configuration */
HAL_NVIC_SetPriority(AUDIO_IN_SAIx_DMAx_IRQ, AUDIO_IN_IRQ_PREPRIO, 0);
HAL_NVIC_EnableIRQ(AUDIO_IN_SAIx_DMAx_IRQ);
}
}
/**
* @brief De-Initializes SAI Audio IN MSP.
* @param hsai: SAI handle
* @param Params: pointer on additional configuration parameters, can be NULL.
* @retval None
*/
static void SAIx_In_MspDeInit(SAI_HandleTypeDef *hsai, void *Params)
{
GPIO_InitTypeDef gpio_init_structure;
if(hsai->Instance == AUDIO_IN_SAI_PDMx)
{
/* Deinitialize the DMA stream */
HAL_DMA_Abort(hsai->hdmarx);
HAL_SAI_DeInit(hsai);
/* Disable SAI peripheral */
__HAL_SAI_DISABLE(hsai);
/* Deinitialize the DMA stream */
HAL_DMA_DeInit(hsai->hdmarx);
gpio_init_structure.Pin = AUDIO_IN_SAI_PDMx_CLK_IN_PIN;
HAL_GPIO_DeInit(AUDIO_IN_SAI_PDMx_CLK_IN_PORT, gpio_init_structure.Pin);
gpio_init_structure.Pin = AUDIO_IN_SAI_PDMx_DATA_IN_PIN;
HAL_GPIO_DeInit(AUDIO_IN_SAI_PDMx_DATA_IN_PORT, gpio_init_structure.Pin);
/* Disable SAI clock */
AUDIO_IN_SAI_PDMx_CLK_DISABLE();
}
else
{
/* SAI DMA IRQ Channel deactivation */
HAL_NVIC_DisableIRQ(AUDIO_IN_SAIx_DMAx_IRQ);
if(hsai->Instance == AUDIO_IN_SAIx)
{
/* Deinitialize the DMA stream */
HAL_DMA_DeInit(hsai->hdmatx);
}
/* Disable SAI peripheral */
__HAL_SAI_DISABLE(hsai);
/* Deactivates CODEC_SAI pin SD by putting them in input mode */
gpio_init_structure.Pin = AUDIO_IN_SAIx_SD_PIN;
HAL_GPIO_DeInit(AUDIO_IN_SAIx_SD_GPIO_PORT, gpio_init_structure.Pin);
/* Disable SAI clock */
AUDIO_IN_SAIx_CLK_DISABLE();
}
}
/**
* @brief Initializes the Audio Codec audio interface (SAI).
* @param SaiInMode: Audio mode to be configured for the SAI peripheral.
* @param SlotActive: Audio active slot to be configured for the SAI peripheral.
* @param AudioFreq: Audio frequency to be configured for the SAI peripheral.
* @retval None
*/
static void SAIx_In_Init(uint32_t SaiInMode, uint32_t SlotActive, uint32_t AudioFreq)
{
/* Disable SAI peripheral to allow access to SAI internal registers */
__HAL_SAI_DISABLE(&haudio_in_sai);
/* Configure SAI_Block_x
LSBFirst: Disabled
DataSize: 16 */
haudio_in_sai.Init.MonoStereoMode = SAI_STEREOMODE;
haudio_in_sai.Init.AudioFrequency = AudioFreq;
haudio_in_sai.Init.AudioMode = SaiInMode;
haudio_in_sai.Init.NoDivider = SAI_MASTERDIVIDER_ENABLE;
haudio_in_sai.Init.Protocol = SAI_FREE_PROTOCOL;
haudio_in_sai.Init.DataSize = SAI_DATASIZE_16;
haudio_in_sai.Init.FirstBit = SAI_FIRSTBIT_MSB;
haudio_in_sai.Init.ClockStrobing = SAI_CLOCKSTROBING_RISINGEDGE;
haudio_in_sai.Init.Synchro = SAI_SYNCHRONOUS;
haudio_in_sai.Init.OutputDrive = SAI_OUTPUTDRIVE_DISABLE;
haudio_in_sai.Init.FIFOThreshold = SAI_FIFOTHRESHOLD_1QF;
haudio_in_sai.Init.SynchroExt = SAI_SYNCEXT_DISABLE;
haudio_in_sai.Init.CompandingMode = SAI_NOCOMPANDING;
haudio_in_sai.Init.TriState = SAI_OUTPUT_RELEASED;
haudio_in_sai.Init.Mckdiv = 0;
haudio_in_sai.Init.MckOverSampling = SAI_MCK_OVERSAMPLING_DISABLE;
haudio_in_sai.Init.PdmInit.Activation = DISABLE;
/* Configure SAI_Block_x Frame
Frame Length: 64
Frame active Length: 32
FS Definition: Start frame + Channel Side identification
FS Polarity: FS active Low
FS Offset: FS asserted one bit before the first bit of slot 0 */
haudio_in_sai.FrameInit.FrameLength = 128;
haudio_in_sai.FrameInit.ActiveFrameLength = 64;
haudio_in_sai.FrameInit.FSDefinition = SAI_FS_CHANNEL_IDENTIFICATION;
haudio_in_sai.FrameInit.FSPolarity = SAI_FS_ACTIVE_LOW;
haudio_in_sai.FrameInit.FSOffset = SAI_FS_BEFOREFIRSTBIT;
/* Configure SAI Block_x Slot
Slot First Bit Offset: 0
Slot Size : 16
Slot Number: 4
Slot Active: All slot active */
haudio_in_sai.SlotInit.FirstBitOffset = 0;
haudio_in_sai.SlotInit.SlotSize = SAI_SLOTSIZE_DATASIZE;
haudio_in_sai.SlotInit.SlotNumber = 4;
haudio_in_sai.SlotInit.SlotActive = SlotActive;
if(hAudioIn.Interface == AUDIO_IN_INTERFACE_PDM)
{
haudio_in_sai.Init.AudioFrequency = AudioFreq * 8;
haudio_in_sai.Init.Synchro = SAI_ASYNCHRONOUS;
haudio_in_sai.Init.NoDivider = SAI_MASTERDIVIDER_DISABLE;
haudio_in_sai.Init.PdmInit.Activation = ENABLE;
haudio_in_sai.Init.PdmInit.MicPairsNbr = 2;
haudio_in_sai.Init.PdmInit.ClockEnable = SAI_PDM_CLOCK2_ENABLE;
haudio_in_sai.Init.FirstBit = SAI_FIRSTBIT_LSB;
haudio_in_sai.Init.ClockStrobing = SAI_CLOCKSTROBING_FALLINGEDGE;
haudio_in_sai.FrameInit.FrameLength = 32;
haudio_in_sai.FrameInit.ActiveFrameLength = 1;
haudio_in_sai.FrameInit.FSDefinition = SAI_FS_STARTFRAME;
haudio_in_sai.FrameInit.FSPolarity = SAI_FS_ACTIVE_HIGH;
haudio_in_sai.FrameInit.FSOffset = SAI_FS_FIRSTBIT;
haudio_in_sai.SlotInit.SlotNumber = 2;
haudio_in_sai.SlotInit.SlotActive = SlotActive;
}
HAL_SAI_Init(&haudio_in_sai);
/* Enable SAI peripheral */
__HAL_SAI_ENABLE(&haudio_in_sai);
}
/**
* @brief Deinitializes the output Audio Codec audio interface (SAI).
* @retval None
*/
static void SAIx_In_DeInit(SAI_HandleTypeDef *hsai)
{
/* Disable SAI peripheral */
__HAL_SAI_DISABLE(hsai);
HAL_SAI_DeInit(hsai);
}
/**
* @}
*/
/**
* @}
*/
/**
* @}
*/
/**
* @}
*/
/************************ (C) COPYRIGHT STMicroelectronics *****END OF FILE****/