/** ****************************************************************************** * @file stm324x9i_eval_audio.c * @author MCD Application Team * @brief This file provides the Audio driver for the STM324x9I-EVAL evaluation 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 STM324x9I-EVAL (MB1045) Evaluation boards. + Call the function BSP_AUDIO_OUT_Init( OutputDevice: physical output mode (OUTPUT_DEVICE_SPEAKER, OUTPUT_DEVICE_HEADPHONE or OUTPUT_DEVICE_BOTH) Volume : Initial volume to be set (0 is min (mute), 100 is max (100%) AudioFreq : Audio frequency in Hz (8000, 16000, 22500, 32000...) this parameter is relative to the audio file/stream type. ) This function configures all the hardware required for the audio application (codec, I2C, SAI, GPIOs, DMA and interrupt if needed). This function returns AUDIO_OK if configuration is OK. If the returned value is different from AUDIO_OK or the function is stuck then the communication with the codec or the IOExpander 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_EVAL_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 stm324x9i_eval_audio.h file. (refer to the example for more details on the callbacks implementations) + To Stop playing, to modify the volume level, the frequency, the audio frame slot, the device output mode the mute or the stop, use the functions: BSP_AUDIO_OUT_SetVolume(), AUDIO_OUT_SetFrequency(), BSP_AUDIO_OUT_SetAudioFrameSlot(), BSP_AUDIO_OUT_SetOutputMode(), BSP_AUDIO_OUT_SetMute() and BSP_AUDIO_OUT_Stop(). + The driver API and the callback functions are at the end of the stm324x9i_eval_audio.h file. Driver architecture: -------------------- + This driver provide the High Audio Layer: consists of the function API exported in the stm324x9i_eval_audio.h file (BSP_AUDIO_OUT_Init(), BSP_AUDIO_OUT_Play() ...) + This driver provide also the Media Access Layer (MAL): which consists of functions allowing to access the media containing/ providing the audio file/stream. Known Limitations: ------------------ 1- If the TDM Format used to paly 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 "stm324x9i_eval_audio.h" /** @addtogroup BSP * @{ *//* ... */ /** @addtogroup STM324x9I_EVAL * @{ *//* ... */ /** @defgroup STM324x9I_EVAL_AUDIO STM324x9I EVAL AUDIO * @brief This file includes the low layer driver for wm8994 Audio Codec * available on STM324x9I-EVAL evaluation board(MB1045). * @{ *//* ... */ /** @defgroup STM324x9I_EVAL_AUDIO_Private_Variables STM324x9I EVAL AUDIO Private Variables * @{ *//* ... */ AUDIO_DrvTypeDef *audio_drv; SAI_HandleTypeDef haudio_out_sai; I2S_HandleTypeDef haudio_in_i2s; TIM_HandleTypeDef haudio_tim; /* PDM filters params */ PDM_Filter_Handler_t PDM_FilterHandler[2]; PDM_Filter_Config_t PDM_FilterConfig[2]; uint8_t Channel_Demux[128] = { 0x00, 0x01, 0x00, 0x01, 0x02, 0x03, 0x02, 0x03, 0x00, 0x01, 0x00, 0x01, 0x02, 0x03, 0x02, 0x03, 0x04, 0x05, 0x04, 0x05, 0x06, 0x07, 0x06, 0x07, 0x04, 0x05, 0x04, 0x05, 0x06, 0x07, 0x06, 0x07, 0x00, 0x01, 0x00, 0x01, 0x02, 0x03, 0x02, 0x03, 0x00, 0x01, 0x00, 0x01, 0x02, 0x03, 0x02, 0x03, 0x04, 0x05, 0x04, 0x05, 0x06, 0x07, 0x06, 0x07, 0x04, 0x05, 0x04, 0x05, 0x06, 0x07, 0x06, 0x07, 0x08, 0x09, 0x08, 0x09, 0x0a, 0x0b, 0x0a, 0x0b, 0x08, 0x09, 0x08, 0x09, 0x0a, 0x0b, 0x0a, 0x0b, 0x0c, 0x0d, 0x0c, 0x0d, 0x0e, 0x0f, 0x0e, 0x0f, 0x0c, 0x0d, 0x0c, 0x0d, 0x0e, 0x0f, 0x0e, 0x0f, 0x08, 0x09, 0x08, 0x09, 0x0a, 0x0b, 0x0a, 0x0b, 0x08, 0x09, 0x08, 0x09, 0x0a, 0x0b, 0x0a, 0x0b, 0x0c, 0x0d, 0x0c, 0x0d, 0x0e, 0x0f, 0x0e, 0x0f, 0x0c, 0x0d, 0x0c, 0x0d, 0x0e, 0x0f, 0x0e, 0x0f ...}; uint16_t __IO AudioInVolume = DEFAULT_AUDIO_IN_VOLUME; /** * @} *//* ... */ /** @defgroup STM324x9I_EVAL_AUDIO_Private_Function_Prototypes STM324x9I EVAL AUDIO Private Function Prototypes * @{ *//* ... */ static void SAIx_Init(uint32_t AudioFreq); static void SAIx_DeInit(void); static void I2Sx_Init(uint32_t AudioFreq); static void I2Sx_DeInit(void); static void TIMx_IC_MspInit(TIM_HandleTypeDef *htim); static void TIMx_IC_MspDeInit(TIM_HandleTypeDef *htim); static void TIMx_Init(void); static void TIMx_DeInit(void); static void PDMDecoder_Init(uint32_t AudioFreq, uint32_t ChnlNbrIn, uint32_t ChnlNbrOut); /** * @} *//* ... */ /** @defgroup STM324x9I_EVAL_AUDIO_out_Private_Functions STM324x9I EVAL 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; /* Disable SAI */ SAIx_DeInit(); /* 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 *//* ... */ haudio_out_sai.Instance = AUDIO_SAIx; 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); }if (HAL_SAI_GetState(&haudio_out_sai) == HAL_SAI_STATE_RESET) { ... } SAIx_Init(AudioFreq); /* wm8994 codec initialization */ if((wm8994_drv.ReadID(AUDIO_I2C_ADDRESS)) == WM8994_ID) { /* Initialize the audio driver structure */ audio_drv = &wm8994_drv; ret = AUDIO_OK; }if ((wm8994_drv.ReadID(AUDIO_I2C_ADDRESS)) == WM8994_ID) { ... } else { ret = AUDIO_ERROR; }else { ... } if(ret == AUDIO_OK) { /* Initialize the codec internal registers */ audio_drv->Init(AUDIO_I2C_ADDRESS, OutputDevice, Volume, AudioFreq); }if (ret == AUDIO_OK) { ... } return ret; }{ ... } /** * @brief De-initialize the audio peripherals. * @retval None *//* ... */ void BSP_AUDIO_OUT_DeInit(void) { SAIx_DeInit(); /* DeInit the SAI MSP : this __weak function can be rewritten by the application */ BSP_AUDIO_OUT_MspDeInit(&haudio_out_sai, NULL); }{ ... } /** * @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; }if (audio_drv->Play(AUDIO_I2C_ADDRESS, pBuffer, Size) != 0) { ... } 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; }else { ... } }{ ... } /** * @brief Sends n-Bytes on the SAI 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_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 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; }if (audio_drv->Pause(AUDIO_I2C_ADDRESS) != 0) { ... } 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; }else { ... } }{ ... } /** * @brief This function 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 behavior). * @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; }if (audio_drv->Resume(AUDIO_I2C_ADDRESS) != 0) { ... } 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; }else { ... } }{ ... } /** * @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; }if (audio_drv->Stop(AUDIO_I2C_ADDRESS, Option) != 0) { ... } else { if(Option == CODEC_PDWN_HW) { /* Wait at least 100us */ HAL_Delay(1); }if (Option == CODEC_PDWN_HW) { ... } /* Return AUDIO_OK when all operations are correctly done */ return AUDIO_OK; }else { ... } }{ ... } /** * @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; }if (audio_drv->SetVolume(AUDIO_I2C_ADDRESS, Volume) != 0) { ... } else { /* Return AUDIO_OK when all operations are correctly done */ return AUDIO_OK; }else { ... } }{ ... } /** * @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; }if (audio_drv->SetMute(AUDIO_I2C_ADDRESS, Cmd) != 0) { ... } else { /* Return AUDIO_OK when all operations are correctly done */ return AUDIO_OK; }else { ... } }{ ... } /** * @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; }if (audio_drv->SetOutputMode(AUDIO_I2C_ADDRESS, Output) != 0) { ... } else { /* Return AUDIO_OK when all operations are correctly done */ return AUDIO_OK; }else { ... } }{ ... } /** * @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_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. *//* ... */ 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 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_ExCLKInitStruct; HAL_RCCEx_GetPeriphCLKConfig(&RCC_ExCLKInitStruct); /* Set the PLL configuration according to the audio frequency */ if((AudioFreq == AUDIO_FREQUENCY_11K) || (AudioFreq == AUDIO_FREQUENCY_22K) || (AudioFreq == AUDIO_FREQUENCY_44K)) { /* Configure PLLSAI prescalers */ /* PLLI2S_VCO: VCO_429M SAI_CLK(first level) = PLLI2S_VCO/PLLI2SQ = 429/2 = 214.5 Mhz SAI_CLK_x = SAI_CLK(first level)/PLLI2SDIVQ = 214.5/19 = 11.289 Mhz *//* ... */ RCC_ExCLKInitStruct.PeriphClockSelection = RCC_PERIPHCLK_SAI_PLLI2S; RCC_ExCLKInitStruct.PLLI2S.PLLI2SN = 429; RCC_ExCLKInitStruct.PLLI2S.PLLI2SQ = 2; RCC_ExCLKInitStruct.PLLI2SDivQ = 19; HAL_RCCEx_PeriphCLKConfig(&RCC_ExCLKInitStruct); }if ((AudioFreq == AUDIO_FREQUENCY_11K) || (AudioFreq == AUDIO_FREQUENCY_22K) || (AudioFreq == AUDIO_FREQUENCY_44K)) { ... } else /* AUDIO_FREQUENCY_8K, AUDIO_FREQUENCY_16K, AUDIO_FREQUENCY_48K), AUDIO_FREQUENCY_96K */ { /* SAI clock config PLLI2S_VCO: VCO_344M SAI_CLK(first level) = PLLI2S_VCO/PLLI2SQ = 344/7 = 49.142 Mhz SAI_CLK_x = SAI_CLK(first level)/PLLI2SDIVQ = 49.142/1 = 49.142 Mhz *//* ... */ RCC_ExCLKInitStruct.PeriphClockSelection = RCC_PERIPHCLK_SAI_PLLI2S; RCC_ExCLKInitStruct.PLLI2S.PLLI2SN = 344; RCC_ExCLKInitStruct.PLLI2S.PLLI2SQ = 7; RCC_ExCLKInitStruct.PLLI2SDivQ = 1; HAL_RCCEx_PeriphCLKConfig(&RCC_ExCLKInitStruct); }else { ... } }{ ... } /** * @brief Initialize 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_saiTx; GPIO_InitTypeDef GPIO_InitStruct; /* Enable SAI clock */ AUDIO_SAIx_CLK_ENABLE(); /* Enable GPIO clock */ AUDIO_SAIx_MCLK_SCK_SD_FS_ENABLE(); /* CODEC_SAI pins configuration: FS, SCK, MCK and SD pins ------------------*/ GPIO_InitStruct.Pin = AUDIO_SAIx_FS_PIN | AUDIO_SAIx_SCK_PIN | AUDIO_SAIx_SD_PIN | AUDIO_SAIx_MCK_PIN; GPIO_InitStruct.Mode = GPIO_MODE_AF_PP; GPIO_InitStruct.Pull = GPIO_NOPULL; GPIO_InitStruct.Speed = GPIO_SPEED_HIGH; GPIO_InitStruct.Alternate = AUDIO_SAIx_MCLK_SCK_SD_FS_AF; HAL_GPIO_Init(AUDIO_SAIx_MCLK_SCK_SD_FS_GPIO_PORT, &GPIO_InitStruct); /* Enable the DMA clock */ AUDIO_SAIx_DMAx_CLK_ENABLE(); if(hsai->Instance == AUDIO_SAIx) { /* Configure the hdma_saiTx handle parameters */ hdma_saiTx.Init.Channel = AUDIO_SAIx_DMAx_CHANNEL; hdma_saiTx.Init.Direction = DMA_MEMORY_TO_PERIPH; hdma_saiTx.Init.PeriphInc = DMA_PINC_DISABLE; hdma_saiTx.Init.MemInc = DMA_MINC_ENABLE; hdma_saiTx.Init.PeriphDataAlignment = AUDIO_SAIx_DMAx_PERIPH_DATA_SIZE; hdma_saiTx.Init.MemDataAlignment = AUDIO_SAIx_DMAx_MEM_DATA_SIZE; hdma_saiTx.Init.Mode = DMA_NORMAL; hdma_saiTx.Init.Priority = DMA_PRIORITY_HIGH; hdma_saiTx.Init.FIFOMode = DMA_FIFOMODE_ENABLE; hdma_saiTx.Init.FIFOThreshold = DMA_FIFO_THRESHOLD_FULL; hdma_saiTx.Init.MemBurst = DMA_MBURST_SINGLE; hdma_saiTx.Init.PeriphBurst = DMA_PBURST_SINGLE; hdma_saiTx.Instance = AUDIO_SAIx_DMAx_STREAM; /* Associate the DMA handle */ __HAL_LINKDMA(hsai, hdmatx, hdma_saiTx); /* Deinitialize the Stream for new transfer */ HAL_DMA_DeInit(&hdma_saiTx); /* Configure the DMA Stream */ HAL_DMA_Init(&hdma_saiTx); }if (hsai->Instance == AUDIO_SAIx) { ... } /* SAI DMA IRQ Channel configuration */ HAL_NVIC_SetPriority(AUDIO_SAIx_DMAx_IRQ, AUDIO_OUT_IRQ_PREPRIO, 0); HAL_NVIC_EnableIRQ(AUDIO_SAIx_DMAx_IRQ); }{ ... } /** * @brief Deinitialize SAI MSP. * @param hsai: SAI handle * @param Params : pointer on additional configuration parameters, can be NULL. *//* ... */ __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_SAIx_DMAx_IRQ); if(hsai->Instance == AUDIO_SAIx) { /* Deinitialize the DMA stream */ HAL_DMA_DeInit(hsai->hdmatx); }if (hsai->Instance == AUDIO_SAIx) { ... } /* 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_SAIx_FS_PIN | AUDIO_SAIx_SCK_PIN | AUDIO_SAIx_SD_PIN | AUDIO_SAIx_MCK_PIN; HAL_GPIO_DeInit(AUDIO_SAIx_MCLK_SCK_SD_FS_GPIO_PORT, gpio_init_structure.Pin); /* Disable SAI clock */ AUDIO_SAIx_CLK_DISABLE(); /* GPIO pins clock and DMA clock can be shut down in the applic by surcharging this __weak function *//* ... */ }{ ... } /** * @brief Tx Transfer completed callbacks. * @param hsai: SAI handle *//* ... */ 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 stm324x9i_eval_audio.h) *//* ... */ BSP_AUDIO_OUT_TransferComplete_CallBack(); }{ ... } /** * @brief Tx Half Transfer completed callbacks. * @param hsai: SAI handle *//* ... */ 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 stm324x9i_eval_audio.h) *//* ... */ BSP_AUDIO_OUT_HalfTransfer_CallBack(); }{ ... } /** * @brief SAI error callbacks. * @param hsai: SAI handle *//* ... */ void HAL_SAI_ErrorCallback(SAI_HandleTypeDef *hsai) { 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) { }{ ... } /******************************************************************************* Static Functions *******************************************************************************//* ... */ /** * @brief Initializes the Audio Codec audio interface (SAI). * @param AudioFreq: Audio frequency to be configured for the SAI peripheral. * @note The default SlotActive configuration is set to CODEC_AUDIOFRAME_SLOT_0123 * and user can update this configuration using *//* ... */ static void SAIx_Init(uint32_t AudioFreq) { /* Initialize the haudio_out_sai Instance parameter */ haudio_out_sai.Instance = AUDIO_SAIx; /* 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.AudioFrequency = AudioFreq; haudio_out_sai.Init.ClockSource = SAI_CLKSOURCE_PLLI2S; haudio_out_sai.Init.AudioMode = SAI_MODEMASTER_TX; haudio_out_sai.Init.NoDivider = SAI_MASTERDIVIDER_ENABLED; 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_FALLINGEDGE; haudio_out_sai.Init.Synchro = SAI_ASYNCHRONOUS; haudio_out_sai.Init.OutputDrive = SAI_OUTPUTDRIVE_ENABLED; haudio_out_sai.Init.FIFOThreshold = SAI_FIFOTHRESHOLD_1QF; /* 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 = 64; haudio_out_sai.FrameInit.ActiveFrameLength = 32; 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 = CODEC_AUDIOFRAME_SLOT_0123; HAL_SAI_Init(&haudio_out_sai); /* Enable SAI peripheral to generate MCLK */ __HAL_SAI_ENABLE(&haudio_out_sai); }{ ... } /** * @brief Deinitialize the Audio Codec audio interface (SAI). *//* ... */ static void SAIx_DeInit(void) { /* Initialize the haudio_out_sai Instance parameter */ haudio_out_sai.Instance = AUDIO_SAIx; /* Disable SAI peripheral */ __HAL_SAI_DISABLE(&haudio_out_sai); HAL_SAI_DeInit(&haudio_out_sai); }{ ... } /** * @} *//* ... */ /** @defgroup STM324x9I_EVAL_AUDIO_IN_Private_Functions STM324x9I EVAL AUDIO IN Private Functions * @{ *//* ... */ /** * @brief Initializes wave recording. * @note This function assumes that the I2S input clock (through PLL_R in * Devices RevA/Z and through dedicated PLLI2S_R in Devices RevB/Y) * is already configured and ready to be used. * @param AudioFreq: Audio frequency to be configured for the I2S peripheral. * @param BitRes: Audio frequency to be configured for the I2S peripheral. * @param ChnlNbr: Audio frequency to be configured for the I2S 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) { /* DeInit the I2S */ I2Sx_DeInit(); /* Configure PLL clock */ BSP_AUDIO_IN_ClockConfig(&haudio_in_i2s, NULL); /* Configure the PDM library */ PDMDecoder_Init(AudioFreq, ChnlNbr, ChnlNbr); /* Configure the I2S peripheral */ haudio_in_i2s.Instance = AUDIO_I2Sx; if(HAL_I2S_GetState(&haudio_in_i2s) == HAL_I2S_STATE_RESET) { /* Initialize the I2S Msp: this __weak function can be rewritten by the application */ BSP_AUDIO_IN_MspInit(&haudio_in_i2s, NULL); }if (HAL_I2S_GetState(&haudio_in_i2s) == HAL_I2S_STATE_RESET) { ... } /* Configure the I2S peripheral */ I2Sx_Init(AudioFreq); /* Return AUDIO_OK when all operations are correctly done */ return AUDIO_OK; }{ ... } /** * @brief Deinit the audio IN peripherals. *//* ... */ void BSP_AUDIO_IN_DeInit(void) { /* DeInit the I2S */ I2Sx_DeInit(); /* DeInit the I2S MSP : this __weak function can be rewritten by the applic */ BSP_AUDIO_IN_MspDeInit(&haudio_in_i2s, NULL); /* DeInit the Timer */ TIMx_DeInit(); }{ ... } /** * @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) { uint32_t ret = AUDIO_ERROR; /* Start the process receive DMA */ HAL_I2S_Receive_DMA(&haudio_in_i2s, pbuf, size); /* Return AUDIO_OK when all operations are correctly done */ ret = AUDIO_OK; return ret; }{ ... } /** * @brief Stops audio recording. * @retval AUDIO_OK if correct communication, else wrong communication *//* ... */ uint8_t BSP_AUDIO_IN_Stop(void) { uint32_t ret = AUDIO_ERROR; /* Call the Media layer pause function */ HAL_I2S_DMAPause(&haudio_in_i2s); /* TIMx Peripheral clock disable */ AUDIO_TIMx_CLK_DISABLE(); /* Return AUDIO_OK when all operations are correctly done */ ret = AUDIO_OK; return ret; }{ ... } /** * @brief Pauses the audio file stream. * @retval AUDIO_OK if correct communication, else wrong communication *//* ... */ uint8_t BSP_AUDIO_IN_Pause(void) { /* Call the Media layer pause function */ HAL_I2S_DMAPause(&haudio_in_i2s); /* 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) { /* Call the Media layer pause/resume function */ HAL_I2S_DMAResume(&haudio_in_i2s); /* 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 Converts audio format from PDM to PCM. * @param PDMBuf: Pointer to data PDM buffer * @param PCMBuf: Pointer to data PCM buffer * @retval AUDIO_OK if correct communication, else wrong communication *//* ... */ uint8_t BSP_AUDIO_IN_PDMToPCM(uint16_t* PDMBuf, uint16_t* PCMBuf) { uint8_t AppPDM[INTERNAL_BUFF_SIZE*2]; uint8_t byte1 = 0, byte2 = 0; uint32_t index = 0; /* PDM Demux */ for(index = 0; index<INTERNAL_BUFF_SIZE/2; index++) { byte2 = (PDMBuf[index] >> 8)& 0xFF; byte1 = (PDMBuf[index] & 0xFF); AppPDM[(index*2)+1] = Channel_Demux[byte1 & CHANNEL_DEMUX_MASK] | Channel_Demux[byte2 & CHANNEL_DEMUX_MASK] << 4; AppPDM[(index*2)] = Channel_Demux[(byte1 >> 1) & CHANNEL_DEMUX_MASK] | Channel_Demux[(byte2 >> 1) & CHANNEL_DEMUX_MASK] << 4; }for (index = 0; index for(index = 0; index < DEFAULT_AUDIO_IN_CHANNEL_NBR; index++) { /* PDM to PCM filter */ PDM_Filter((uint8_t*)&AppPDM[index], (uint16_t*)&(PCMBuf[index]), &PDM_FilterHandler[index]); }for (index = 0; index < DEFAULT_AUDIO_IN_CHANNEL_NBR; index++) { ... } /* Return AUDIO_OK when all operations are correctly done */ return AUDIO_OK; }{ ... } /** * @brief Rx Transfer completed callbacks. * @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 Rx Half Transfer completed callbacks. * @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 stm324x9i_eval_audio.h) *//* ... */ BSP_AUDIO_IN_HalfTransfer_CallBack(); }{ ... } /** * @brief I2S error callbacks. * @param hi2s: I2S handle *//* ... */ void HAL_I2S_ErrorCallback(I2S_HandleTypeDef *hi2s) { /* Manage the error generated on DMA FIFO: This function should be coded by user (its prototype is already declared in stm324x9i_eval_audio.h) *//* ... */ BSP_AUDIO_IN_Error_Callback(); }{ ... } /** * @brief Clock Config. * @param hi2s: I2S handle * @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 *//* ... */ __weak void BSP_AUDIO_IN_ClockConfig(I2S_HandleTypeDef *hi2s, void *Params) { RCC_PeriphCLKInitTypeDef RCC_ExCLKInitStruct; HAL_RCCEx_GetPeriphCLKConfig(&RCC_ExCLKInitStruct); RCC_ExCLKInitStruct.PeriphClockSelection = RCC_PERIPHCLK_I2S; RCC_ExCLKInitStruct.PLLI2S.PLLI2SN = 384; RCC_ExCLKInitStruct.PLLI2S.PLLI2SR = 2; HAL_RCCEx_PeriphCLKConfig(&RCC_ExCLKInitStruct); }{ ... } /** * @brief Initialize BSP_AUDIO_IN MSP. * @param hi2s: I2S handle * @param Params : pointer on additional configuration parameters, can be NULL. *//* ... */ __weak void BSP_AUDIO_IN_MspInit(I2S_HandleTypeDef *hi2s, void *Params) { static DMA_HandleTypeDef hdma_i2sRx; GPIO_InitTypeDef GPIO_InitStruct; /* Configure the Timer which clocks the MEMS */ /* Moved inside MSP to allow applic to redefine the TIMx_MspInit */ TIMx_Init(); /* Enable I2S clock */ AUDIO_I2Sx_CLK_ENABLE(); /* Enable SCK and SD GPIO clock */ AUDIO_I2Sx_SD_GPIO_CLK_ENABLE(); AUDIO_I2Sx_SCK_GPIO_CLK_ENABLE(); /* CODEC_I2S pins configuration: WS, SCK and SD pins */ GPIO_InitStruct.Pin = AUDIO_I2Sx_SCK_PIN; GPIO_InitStruct.Mode = GPIO_MODE_AF_PP; GPIO_InitStruct.Pull = GPIO_NOPULL; GPIO_InitStruct.Speed = GPIO_SPEED_FAST; GPIO_InitStruct.Alternate = AUDIO_I2Sx_SCK_AF; HAL_GPIO_Init(AUDIO_I2Sx_SCK_GPIO_PORT, &GPIO_InitStruct); GPIO_InitStruct.Pin = AUDIO_I2Sx_SD_PIN; GPIO_InitStruct.Alternate = AUDIO_I2Sx_SD_AF; HAL_GPIO_Init(AUDIO_I2Sx_SD_GPIO_PORT, &GPIO_InitStruct); /* Enable the DMA clock */ AUDIO_I2Sx_DMAx_CLK_ENABLE(); if(hi2s->Instance == AUDIO_I2Sx) { /* Configure the hdma_i2sRx handle parameters */ hdma_i2sRx.Init.Channel = AUDIO_I2Sx_DMAx_CHANNEL; hdma_i2sRx.Init.Direction = DMA_PERIPH_TO_MEMORY; hdma_i2sRx.Init.PeriphInc = DMA_PINC_DISABLE; hdma_i2sRx.Init.MemInc = DMA_MINC_ENABLE; hdma_i2sRx.Init.PeriphDataAlignment = AUDIO_I2Sx_DMAx_PERIPH_DATA_SIZE; hdma_i2sRx.Init.MemDataAlignment = AUDIO_I2Sx_DMAx_MEM_DATA_SIZE; hdma_i2sRx.Init.Mode = DMA_CIRCULAR; hdma_i2sRx.Init.Priority = DMA_PRIORITY_HIGH; hdma_i2sRx.Init.FIFOMode = DMA_FIFOMODE_DISABLE; hdma_i2sRx.Init.FIFOThreshold = DMA_FIFO_THRESHOLD_FULL; hdma_i2sRx.Init.MemBurst = DMA_MBURST_SINGLE; hdma_i2sRx.Init.PeriphBurst = DMA_MBURST_SINGLE; hdma_i2sRx.Instance = AUDIO_I2Sx_DMAx_STREAM; /* Associate the DMA handle */ __HAL_LINKDMA(hi2s, hdmarx, hdma_i2sRx); /* Deinitialize the Stream for new transfer */ HAL_DMA_DeInit(&hdma_i2sRx); /* Configure the DMA Stream */ HAL_DMA_Init(&hdma_i2sRx); }if (hi2s->Instance == AUDIO_I2Sx) { ... } /* I2S DMA IRQ Channel configuration */ HAL_NVIC_SetPriority(AUDIO_I2Sx_DMAx_IRQ, AUDIO_IN_IRQ_PREPRIO, 0); HAL_NVIC_EnableIRQ(AUDIO_I2Sx_DMAx_IRQ); }{ ... } /** * @brief BSP AUDIO IN MSP Init. * @param hi2s: I2S handle * @param Params : pointer on additional configuration parameters, can be NULL. *//* ... */ __weak void BSP_AUDIO_IN_MspDeInit(I2S_HandleTypeDef *hi2s, void *Params) { GPIO_InitTypeDef gpio_init_structure; /* I2S DMA IRQ Channel deactivation */ HAL_NVIC_DisableIRQ(AUDIO_I2Sx_DMAx_IRQ); if(hi2s->Instance == AUDIO_I2Sx) { /* Deinitialize the Stream for new transfer */ HAL_DMA_DeInit(hi2s->hdmarx); }if (hi2s->Instance == AUDIO_I2Sx) { ... } /* Disable I2S block */ __HAL_I2S_DISABLE(hi2s); /* Disable pins: SCK and SD pins */ gpio_init_structure.Pin = AUDIO_I2Sx_SCK_PIN; HAL_GPIO_DeInit(AUDIO_I2Sx_SCK_GPIO_PORT, gpio_init_structure.Pin); gpio_init_structure.Pin = AUDIO_I2Sx_SD_PIN; HAL_GPIO_DeInit(AUDIO_I2Sx_SD_GPIO_PORT, gpio_init_structure.Pin); /* Disable I2S clock */ AUDIO_I2Sx_CLK_DISABLE(); /* GPIO pins clock and DMA clock can be shut down in the applic by surcgarging this __weak function *//* ... */ }{ ... } /** * @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 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 Initializes 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 * Number of audio channels (1: mono; 2: stereo) *//* ... */ static void PDMDecoder_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_LSB; 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]); }for (index = 0; index < ChnlNbrIn; index++) { ... } }{ ... } /** * @brief Initializes the Audio Codec audio interface (I2S) * @note This function assumes that the I2S input clock (through PLL_R in * Devices RevA/Z and through dedicated PLLI2S_R in Devices RevB/Y) * is already configured and ready to be used. * @param AudioFreq: Audio frequency to be configured for the I2S peripheral. *//* ... */ static void I2Sx_Init(uint32_t AudioFreq) { /* Initialize the haudio_in_i2s Instance parameter */ haudio_in_i2s.Instance = AUDIO_I2Sx; /* Disable I2S block */ __HAL_I2S_DISABLE(&haudio_in_i2s); /* I2S2 peripheral configuration */ haudio_in_i2s.Init.AudioFreq = 4 * AudioFreq; haudio_in_i2s.Init.ClockSource = I2S_CLOCK_PLL; haudio_in_i2s.Init.CPOL = I2S_CPOL_HIGH; haudio_in_i2s.Init.DataFormat = I2S_DATAFORMAT_16B; haudio_in_i2s.Init.MCLKOutput = I2S_MCLKOUTPUT_DISABLE; haudio_in_i2s.Init.Mode = I2S_MODE_MASTER_RX; haudio_in_i2s.Init.Standard = I2S_STANDARD_LSB; /* Init the I2S */ HAL_I2S_Init(&haudio_in_i2s); }{ ... } /** * @brief Deinitializes the Audio Codec audio interface (I2S). *//* ... */ static void I2Sx_DeInit(void) { /* Initialize the hAudioInI2s Instance parameter */ haudio_in_i2s.Instance = AUDIO_I2Sx; /* Disable I2S block */ __HAL_I2S_DISABLE(&haudio_in_i2s); /* DeInit the I2S */ HAL_I2S_DeInit(&haudio_in_i2s); }{ ... } /** * @brief Initializes the TIM INput Capture MSP. * @param htim: TIM handle *//* ... */ static void TIMx_IC_MspInit(TIM_HandleTypeDef *htim) { GPIO_InitTypeDef GPIO_InitStruct; /* Enable peripherals and GPIO Clocks --------------------------------------*/ /* TIMx Peripheral clock enable */ AUDIO_TIMx_CLK_ENABLE(); /* Enable GPIO Channels Clock */ AUDIO_TIMx_GPIO_CLK_ENABLE(); Enable peripherals and GPIO Clocks /* Configure I/Os ----------------------------------------------------------*/ /* Common configuration for all channels */ GPIO_InitStruct.Mode = GPIO_MODE_AF_PP; GPIO_InitStruct.Pull = GPIO_NOPULL; GPIO_InitStruct.Speed = GPIO_SPEED_HIGH; GPIO_InitStruct.Alternate = AUDIO_TIMx_AF; /* Configure TIM input channel */ GPIO_InitStruct.Pin = AUDIO_TIMx_IN_GPIO_PIN; HAL_GPIO_Init(AUDIO_TIMx_GPIO, &GPIO_InitStruct); /* Configure TIM output channel */ GPIO_InitStruct.Pin = AUDIO_TIMx_OUT_GPIO_PIN; HAL_GPIO_Init(AUDIO_TIMx_GPIO, &GPIO_InitStruct); }{ ... } /** * @brief Initializes the TIM INput Capture MSP. * @param htim: TIM handle *//* ... */ static void TIMx_IC_MspDeInit(TIM_HandleTypeDef *htim) { /* Disable TIMx Peripheral clock */ AUDIO_TIMx_CLK_DISABLE(); /* GPIO pins clock and DMA clock can be shut down in the applic by surcgarging this __weak function *//* ... */ }{ ... } /** * @brief Configure TIM as a clock divider by 2. * I2S_SCK is externally connected to TIMx input channel *//* ... */ static void TIMx_Init(void) { TIM_IC_InitTypeDef sICConfig; TIM_OC_InitTypeDef sOCConfig; TIM_ClockConfigTypeDef sCLKSourceConfig; TIM_SlaveConfigTypeDef sSlaveConfig; /* Configure the TIM peripheral --------------------------------------------*/ /* Set TIMx instance */ haudio_tim.Instance = AUDIO_TIMx; /* Timer Input Capture Configuration Structure declaration */ /* Initialize TIMx peripheral as follow: + Period = 0xFFFF + Prescaler = 0 + ClockDivision = 0 + Counter direction = Up *//* ... */ haudio_tim.Init.Period = 1; haudio_tim.Init.Prescaler = 0; haudio_tim.Init.ClockDivision = 0; haudio_tim.Init.CounterMode = TIM_COUNTERMODE_UP; /* Initialize the TIMx peripheral with the structure above */ TIMx_IC_MspInit(&haudio_tim); HAL_TIM_IC_Init(&haudio_tim); Configure the TIM peripheral /* Configure the Input Capture channel -------------------------------------*/ /* Configure the Input Capture of channel 2 */ sICConfig.ICPolarity = TIM_ICPOLARITY_FALLING; sICConfig.ICSelection = TIM_ICSELECTION_DIRECTTI; sICConfig.ICPrescaler = TIM_ICPSC_DIV1; sICConfig.ICFilter = 0; HAL_TIM_IC_ConfigChannel(&haudio_tim, &sICConfig, AUDIO_TIMx_IN_CHANNEL); /* Select external clock mode 1 */ sCLKSourceConfig.ClockSource = TIM_CLOCKSOURCE_ETRMODE1; sCLKSourceConfig.ClockPolarity = TIM_CLOCKPOLARITY_NONINVERTED; sCLKSourceConfig.ClockPrescaler = TIM_CLOCKPRESCALER_DIV1; sCLKSourceConfig.ClockFilter = 0; HAL_TIM_ConfigClockSource(&haudio_tim, &sCLKSourceConfig); /* Select Input Channel as input trigger */ sSlaveConfig.InputTrigger = TIM_TS_TI1FP1; sSlaveConfig.SlaveMode = TIM_SLAVEMODE_EXTERNAL1; sSlaveConfig.TriggerPolarity = TIM_TRIGGERPOLARITY_NONINVERTED; sSlaveConfig.TriggerPrescaler = TIM_CLOCKPRESCALER_DIV1; sSlaveConfig.TriggerFilter = 0; HAL_TIM_SlaveConfigSynchronization(&haudio_tim, &sSlaveConfig); /* Output Compare PWM Mode configuration: Channel2 */ sOCConfig.OCMode = TIM_OCMODE_PWM1; sOCConfig.OCIdleState = TIM_OCIDLESTATE_SET; sOCConfig.Pulse = 1; sOCConfig.OCPolarity = TIM_OCPOLARITY_HIGH; sOCConfig.OCNPolarity = TIM_OCNPOLARITY_HIGH; sOCConfig.OCFastMode = TIM_OCFAST_DISABLE; sOCConfig.OCNIdleState = TIM_OCNIDLESTATE_SET; /* Initialize the TIM3 Channel2 with the structure above */ HAL_TIM_PWM_ConfigChannel(&haudio_tim, &sOCConfig, AUDIO_TIMx_OUT_CHANNEL); /* Start the TIM3 Channel2 */ HAL_TIM_PWM_Start(&haudio_tim, AUDIO_TIMx_OUT_CHANNEL); /* Start the TIM3 Channel1 */ HAL_TIM_IC_Start(&haudio_tim, AUDIO_TIMx_IN_CHANNEL); }{ ... } /** * @brief Configure TIM as a clock divider by 2. * I2S_SCK is externally connected to TIMx input channel *//* ... */ static void TIMx_DeInit(void) { haudio_tim.Instance = AUDIO_TIMx; /* Stop the TIM3 Channel2 */ HAL_TIM_PWM_Stop(&haudio_tim, AUDIO_TIMx_OUT_CHANNEL); /* Stop the TIM3 Channel1 */ HAL_TIM_IC_Stop(&haudio_tim, AUDIO_TIMx_IN_CHANNEL); HAL_TIM_IC_DeInit(&haudio_tim); /* Initialize the TIMx peripheral with the structure above */ TIMx_IC_MspDeInit(&haudio_tim); }{ ... } /** * @} *//* ... */ /** * @} *//* ... */ /** * @} *//* ... */ /** * @} *//* ... */ /** * @} *//* ... */