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1373 lines
48 KiB
C
1373 lines
48 KiB
C
/**
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******************************************************************************
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* @file stm32746g_discovery_audio.c
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* @author MCD Application Team
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* @version V2.0.0
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* @date 30-December-2016
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* @brief This file provides the Audio driver for the STM32746G-Discovery board.
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@verbatim
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How To use this driver:
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-----------------------
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+ This driver supports STM32F7xx devices on STM32746G-Discovery (MB1191) board.
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+ Call the function BSP_AUDIO_OUT_Init(
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OutputDevice: physical output mode (OUTPUT_DEVICE_SPEAKER,
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OUTPUT_DEVICE_HEADPHONE or OUTPUT_DEVICE_BOTH)
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Volume : Initial volume to be set (0 is min (mute), 100 is max (100%)
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AudioFreq : Audio frequency in Hz (8000, 16000, 22500, 32000...)
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this parameter is relative to the audio file/stream type.
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)
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This function configures all the hardware required for the audio application (codec, I2C, SAI,
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GPIOs, DMA and interrupt if needed). This function returns AUDIO_OK if configuration is OK.
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If the returned value is different from AUDIO_OK or the function is stuck then the communication with
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the codec or the MFX has failed (try to un-plug the power or reset device in this case).
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- OUTPUT_DEVICE_SPEAKER : only speaker will be set as output for the audio stream.
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- OUTPUT_DEVICE_HEADPHONE: only headphones will be set as output for the audio stream.
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- OUTPUT_DEVICE_BOTH : both Speaker and Headphone are used as outputs for the audio stream
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at the same time.
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Note. On STM32746G-Discovery SAI_DMA is configured in CIRCULAR mode. Due to this the application
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does NOT need to call BSP_AUDIO_OUT_ChangeBuffer() to assure streaming.
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+ Call the function BSP_DISCOVERY_AUDIO_OUT_Play(
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pBuffer: pointer to the audio data file address
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Size : size of the buffer to be sent in Bytes
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)
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to start playing (for the first time) from the audio file/stream.
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+ Call the function BSP_AUDIO_OUT_Pause() to pause playing
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+ Call the function BSP_AUDIO_OUT_Resume() to resume playing.
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Note. After calling BSP_AUDIO_OUT_Pause() function for pause, only BSP_AUDIO_OUT_Resume() should be called
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for resume (it is not allowed to call BSP_AUDIO_OUT_Play() in this case).
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Note. This function should be called only when the audio file is played or paused (not stopped).
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+ For each mode, you may need to implement the relative callback functions into your code.
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The Callback functions are named AUDIO_OUT_XXX_CallBack() and only their prototypes are declared in
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the stm32746g_discovery_audio.h file. (refer to the example for more details on the callbacks implementations)
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+ To Stop playing, to modify the volume level, the frequency, the audio frame slot,
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the device output mode the mute or the stop, use the functions: BSP_AUDIO_OUT_SetVolume(),
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AUDIO_OUT_SetFrequency(), BSP_AUDIO_OUT_SetAudioFrameSlot(), BSP_AUDIO_OUT_SetOutputMode(),
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BSP_AUDIO_OUT_SetMute() and BSP_AUDIO_OUT_Stop().
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+ The driver API and the callback functions are at the end of the stm32746g_discovery_audio.h file.
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Driver architecture:
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--------------------
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+ This driver provides the High Audio Layer: consists of the function API exported in the stm32746g_discovery_audio.h file
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(BSP_AUDIO_OUT_Init(), BSP_AUDIO_OUT_Play() ...)
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+ This driver provide also the Media Access Layer (MAL): which consists of functions allowing to access the media containing/
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providing the audio file/stream. These functions are also included as local functions into
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the stm32746g_discovery_audio_codec.c file (SAIx_Out_Init() and SAIx_Out_DeInit(), SAIx_In_Init() and SAIx_In_DeInit())
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Known Limitations:
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------------------
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1- If the TDM Format used to play in parallel 2 audio Stream (the first Stream is configured in codec SLOT0 and second
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Stream in SLOT1) the Pause/Resume, volume and mute feature will control the both streams.
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2- Parsing of audio file is not implemented (in order to determine audio file properties: Mono/Stereo, Data size,
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File size, Audio Frequency, Audio Data header size ...). The configuration is fixed for the given audio file.
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3- Supports only Stereo audio streaming.
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4- Supports only 16-bits audio data size.
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@endverbatim
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******************************************************************************
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* @attention
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*
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* <h2><center>© COPYRIGHT(c) 2016 STMicroelectronics</center></h2>
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*
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* Redistribution and use in source and binary forms, with or without modification,
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* are permitted provided that the following conditions are met:
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* 1. Redistributions of source code must retain the above copyright notice,
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* this list of conditions and the following disclaimer.
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* 2. Redistributions in binary form must reproduce the above copyright notice,
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* this list of conditions and the following disclaimer in the documentation
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* and/or other materials provided with the distribution.
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* 3. Neither the name of STMicroelectronics nor the names of its contributors
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* may be used to endorse or promote products derived from this software
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* without specific prior written permission.
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*
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* THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS"
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* AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
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* IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
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* DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE LIABLE
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* FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
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* DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR
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* SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER
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* CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY,
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* OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
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* OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
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*
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******************************************************************************
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*/
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/* Includes ------------------------------------------------------------------*/
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#include "stm32746g_discovery_audio.h"
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/** @addtogroup BSP
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* @{
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*/
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/** @addtogroup STM32746G_DISCOVERY
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* @{
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*/
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/** @defgroup STM32746G_DISCOVERY_AUDIO STM32746G_DISCOVERY AUDIO
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* @brief This file includes the low layer driver for wm8994 Audio Codec
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* available on STM32746G-Discovery board(MB1191).
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* @{
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*/
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/** @defgroup STM32746G_DISCOVERY_AUDIO_Private_Types STM32746G_DISCOVERY AUDIO Private Types
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* @{
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*/
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/**
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* @}
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*/
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/** @defgroup STM32746G_DISCOVERY_AUDIO_Private_Defines STM32746G_DISCOVERY AUDIO Private Defines
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* @{
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*/
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/**
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* @}
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*/
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/** @defgroup STM32746G_DISCOVERY_AUDIO_Private_Macros STM32746G_DISCOVERY AUDIO Private Macros
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* @{
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*/
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/**
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* @}
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*/
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/** @defgroup STM32746G_DISCOVERY_AUDIO_Private_Variables STM32746G_DISCOVERY AUDIO Private Variables
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* @{
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*/
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AUDIO_DrvTypeDef *audio_drv;
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SAI_HandleTypeDef haudio_out_sai={0};
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SAI_HandleTypeDef haudio_in_sai={0};
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TIM_HandleTypeDef haudio_tim;
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uint16_t __IO AudioInVolume = DEFAULT_AUDIO_IN_VOLUME;
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/**
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* @}
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*/
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/** @defgroup STM32746G_DISCOVERY_AUDIO_Private_Function_Prototypes STM32746G_DISCOVERY AUDIO Private Function Prototypes
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* @{
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*/
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static void SAIx_Out_Init(uint32_t AudioFreq);
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static void SAIx_Out_DeInit(void);
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static void SAIx_In_Init(uint32_t SaiOutMode, uint32_t SlotActive, uint32_t AudioFreq);
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static void SAIx_In_DeInit(void);
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/**
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* @}
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*/
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/** @defgroup STM32746G_DISCOVERY_AUDIO_OUT_Exported_Functions STM32746G_DISCOVERY AUDIO Out Exported Functions
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* @{
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*/
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/**
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* @brief Configures the audio peripherals.
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* @param OutputDevice: OUTPUT_DEVICE_SPEAKER, OUTPUT_DEVICE_HEADPHONE,
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* or OUTPUT_DEVICE_BOTH.
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* @param Volume: Initial volume level (from 0 (Mute) to 100 (Max))
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* @param AudioFreq: Audio frequency used to play the audio stream.
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* @note The I2S PLL input clock must be done in the user application.
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* @retval AUDIO_OK if correct communication, else wrong communication
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*/
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uint8_t BSP_AUDIO_OUT_Init(uint16_t OutputDevice, uint8_t Volume, uint32_t AudioFreq)
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{
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uint8_t ret = AUDIO_ERROR;
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uint32_t deviceid = 0x00;
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/* Disable SAI */
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SAIx_Out_DeInit();
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/* PLL clock is set depending on the AudioFreq (44.1khz vs 48khz groups) */
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BSP_AUDIO_OUT_ClockConfig(&haudio_out_sai, AudioFreq, NULL);
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/* SAI data transfer preparation:
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Prepare the Media to be used for the audio transfer from memory to SAI peripheral */
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haudio_out_sai.Instance = AUDIO_OUT_SAIx;
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if(HAL_SAI_GetState(&haudio_out_sai) == HAL_SAI_STATE_RESET)
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{
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/* Init the SAI MSP: this __weak function can be redefined by the application*/
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BSP_AUDIO_OUT_MspInit(&haudio_out_sai, NULL);
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}
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SAIx_Out_Init(AudioFreq);
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/* wm8994 codec initialization */
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deviceid = wm8994_drv.ReadID(AUDIO_I2C_ADDRESS);
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if((deviceid) == WM8994_ID)
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{
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/* Reset the Codec Registers */
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wm8994_drv.Reset(AUDIO_I2C_ADDRESS);
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/* Initialize the audio driver structure */
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audio_drv = &wm8994_drv;
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ret = AUDIO_OK;
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}
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else
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{
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ret = AUDIO_ERROR;
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}
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if(ret == AUDIO_OK)
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{
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/* Initialize the codec internal registers */
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audio_drv->Init(AUDIO_I2C_ADDRESS, OutputDevice, Volume, AudioFreq);
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}
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return ret;
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}
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/**
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* @brief Starts playing audio stream from a data buffer for a determined size.
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* @param pBuffer: Pointer to the buffer
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* @param Size: Number of audio data in BYTES unit.
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* In memory, first element is for left channel, second element is for right channel
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* @retval AUDIO_OK if correct communication, else wrong communication
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*/
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uint8_t BSP_AUDIO_OUT_Play(uint16_t* pBuffer, uint32_t Size)
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{
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/* Call the audio Codec Play function */
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if(audio_drv->Play(AUDIO_I2C_ADDRESS, pBuffer, Size) != 0)
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{
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return AUDIO_ERROR;
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}
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else
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{
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/* Update the Media layer and enable it for play */
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HAL_SAI_Transmit_DMA(&haudio_out_sai, (uint8_t*) pBuffer, DMA_MAX(Size / AUDIODATA_SIZE));
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return AUDIO_OK;
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}
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}
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/**
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* @brief Sends n-Bytes on the SAI interface.
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* @param pData: pointer on data address
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* @param Size: number of data to be written
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* @retval None
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*/
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void BSP_AUDIO_OUT_ChangeBuffer(uint16_t *pData, uint16_t Size)
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{
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HAL_SAI_Transmit_DMA(&haudio_out_sai, (uint8_t*) pData, Size);
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}
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/**
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* @brief This function Pauses the audio file stream. In case
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* of using DMA, the DMA Pause feature is used.
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* @note When calling BSP_AUDIO_OUT_Pause() function for pause, only
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* BSP_AUDIO_OUT_Resume() function should be called for resume (use of BSP_AUDIO_OUT_Play()
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* function for resume could lead to unexpected behaviour).
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* @retval AUDIO_OK if correct communication, else wrong communication
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*/
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uint8_t BSP_AUDIO_OUT_Pause(void)
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{
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/* Call the Audio Codec Pause/Resume function */
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if(audio_drv->Pause(AUDIO_I2C_ADDRESS) != 0)
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{
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return AUDIO_ERROR;
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}
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else
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{
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/* Call the Media layer pause function */
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HAL_SAI_DMAPause(&haudio_out_sai);
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/* Return AUDIO_OK when all operations are correctly done */
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return AUDIO_OK;
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}
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}
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/**
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* @brief This function Resumes the audio file stream.
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* @note When calling BSP_AUDIO_OUT_Pause() function for pause, only
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* BSP_AUDIO_OUT_Resume() function should be called for resume (use of BSP_AUDIO_OUT_Play()
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* function for resume could lead to unexpected behaviour).
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* @retval AUDIO_OK if correct communication, else wrong communication
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*/
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uint8_t BSP_AUDIO_OUT_Resume(void)
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{
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/* Call the Audio Codec Pause/Resume function */
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if(audio_drv->Resume(AUDIO_I2C_ADDRESS) != 0)
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{
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return AUDIO_ERROR;
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}
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else
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{
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/* Call the Media layer pause/resume function */
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HAL_SAI_DMAResume(&haudio_out_sai);
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/* Return AUDIO_OK when all operations are correctly done */
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return AUDIO_OK;
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}
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}
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/**
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* @brief Stops audio playing and Power down the Audio Codec.
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* @param Option: could be one of the following parameters
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* - CODEC_PDWN_SW: for software power off (by writing registers).
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* Then no need to reconfigure the Codec after power on.
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* - CODEC_PDWN_HW: completely shut down the codec (physically).
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* Then need to reconfigure the Codec after power on.
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* @retval AUDIO_OK if correct communication, else wrong communication
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*/
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uint8_t BSP_AUDIO_OUT_Stop(uint32_t Option)
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{
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/* Call the Media layer stop function */
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HAL_SAI_DMAStop(&haudio_out_sai);
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/* Call Audio Codec Stop function */
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if(audio_drv->Stop(AUDIO_I2C_ADDRESS, Option) != 0)
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{
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return AUDIO_ERROR;
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}
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else
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{
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if(Option == CODEC_PDWN_HW)
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{
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/* Wait at least 100us */
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HAL_Delay(1);
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}
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/* Return AUDIO_OK when all operations are correctly done */
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return AUDIO_OK;
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}
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}
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/**
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* @brief Controls the current audio volume level.
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* @param Volume: Volume level to be set in percentage from 0% to 100% (0 for
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* Mute and 100 for Max volume level).
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* @retval AUDIO_OK if correct communication, else wrong communication
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*/
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uint8_t BSP_AUDIO_OUT_SetVolume(uint8_t Volume)
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{
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/* Call the codec volume control function with converted volume value */
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if(audio_drv->SetVolume(AUDIO_I2C_ADDRESS, Volume) != 0)
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{
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return AUDIO_ERROR;
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}
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else
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{
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/* Return AUDIO_OK when all operations are correctly done */
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return AUDIO_OK;
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}
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}
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/**
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* @brief Enables or disables the MUTE mode by software
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* @param Cmd: Could be AUDIO_MUTE_ON to mute sound or AUDIO_MUTE_OFF to
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* unmute the codec and restore previous volume level.
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* @retval AUDIO_OK if correct communication, else wrong communication
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*/
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uint8_t BSP_AUDIO_OUT_SetMute(uint32_t Cmd)
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{
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/* Call the Codec Mute function */
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if(audio_drv->SetMute(AUDIO_I2C_ADDRESS, Cmd) != 0)
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{
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return AUDIO_ERROR;
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}
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else
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{
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/* Return AUDIO_OK when all operations are correctly done */
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return AUDIO_OK;
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}
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}
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/**
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* @brief Switch dynamically (while audio file is played) the output target
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* (speaker or headphone).
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* @param Output: The audio output target: OUTPUT_DEVICE_SPEAKER,
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* OUTPUT_DEVICE_HEADPHONE or OUTPUT_DEVICE_BOTH
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* @retval AUDIO_OK if correct communication, else wrong communication
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*/
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uint8_t BSP_AUDIO_OUT_SetOutputMode(uint8_t Output)
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{
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/* Call the Codec output device function */
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if(audio_drv->SetOutputMode(AUDIO_I2C_ADDRESS, Output) != 0)
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{
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return AUDIO_ERROR;
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}
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else
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{
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/* Return AUDIO_OK when all operations are correctly done */
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return AUDIO_OK;
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}
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}
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/**
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* @brief Updates the audio frequency.
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* @param AudioFreq: Audio frequency used to play the audio stream.
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* @note This API should be called after the BSP_AUDIO_OUT_Init() to adjust the
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* audio frequency.
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* @retval None
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*/
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void BSP_AUDIO_OUT_SetFrequency(uint32_t AudioFreq)
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{
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/* PLL clock is set depending by the AudioFreq (44.1khz vs 48khz groups) */
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BSP_AUDIO_OUT_ClockConfig(&haudio_out_sai, AudioFreq, NULL);
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/* Disable SAI peripheral to allow access to SAI internal registers */
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__HAL_SAI_DISABLE(&haudio_out_sai);
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/* Update the SAI audio frequency configuration */
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haudio_out_sai.Init.AudioFrequency = AudioFreq;
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HAL_SAI_Init(&haudio_out_sai);
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/* Enable SAI peripheral to generate MCLK */
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__HAL_SAI_ENABLE(&haudio_out_sai);
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}
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/**
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* @brief Updates the Audio frame slot configuration.
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* @param AudioFrameSlot: specifies the audio Frame slot
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* This parameter can be one of the following values
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* @arg CODEC_AUDIOFRAME_SLOT_0123
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* @arg CODEC_AUDIOFRAME_SLOT_02
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* @arg CODEC_AUDIOFRAME_SLOT_13
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* @note This API should be called after the BSP_AUDIO_OUT_Init() to adjust the
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* audio frame slot.
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* @retval None
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*/
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void BSP_AUDIO_OUT_SetAudioFrameSlot(uint32_t AudioFrameSlot)
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{
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/* Disable SAI peripheral to allow access to SAI internal registers */
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__HAL_SAI_DISABLE(&haudio_out_sai);
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/* Update the SAI audio frame slot configuration */
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haudio_out_sai.SlotInit.SlotActive = AudioFrameSlot;
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HAL_SAI_Init(&haudio_out_sai);
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/* Enable SAI peripheral to generate MCLK */
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__HAL_SAI_ENABLE(&haudio_out_sai);
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}
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/**
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* @brief Deinit the audio peripherals.
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* @retval None
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*/
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void BSP_AUDIO_OUT_DeInit(void)
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{
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SAIx_Out_DeInit();
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/* DeInit the SAI MSP : this __weak function can be rewritten by the application */
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BSP_AUDIO_OUT_MspDeInit(&haudio_out_sai, NULL);
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}
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/**
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* @brief Tx Transfer completed callbacks.
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* @param hsai: SAI handle
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* @retval None
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*/
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void HAL_SAI_TxCpltCallback(SAI_HandleTypeDef *hsai)
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{
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/* Manage the remaining file size and new address offset: This function
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|
should be coded by user (its prototype is already declared in stm32746g_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 stm32746g_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)
|
|
{
|
|
HAL_SAI_StateTypeDef audio_out_state;
|
|
HAL_SAI_StateTypeDef audio_in_state;
|
|
|
|
audio_out_state = HAL_SAI_GetState(&haudio_out_sai);
|
|
audio_in_state = HAL_SAI_GetState(&haudio_in_sai);
|
|
|
|
/* Determines if it is an audio out or audio in error */
|
|
if ((audio_out_state == HAL_SAI_STATE_BUSY) || (audio_out_state == HAL_SAI_STATE_BUSY_TX))
|
|
{
|
|
BSP_AUDIO_OUT_Error_CallBack();
|
|
}
|
|
|
|
if ((audio_in_state == HAL_SAI_STATE_BUSY) || (audio_in_state == HAL_SAI_STATE_BUSY_RX))
|
|
{
|
|
BSP_AUDIO_IN_Error_CallBack();
|
|
}
|
|
}
|
|
|
|
/**
|
|
* @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
|
|
* @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();
|
|
|
|
/* Enable GPIO clock */
|
|
AUDIO_OUT_SAIx_MCLK_ENABLE();
|
|
AUDIO_OUT_SAIx_SCK_SD_ENABLE();
|
|
AUDIO_OUT_SAIx_FS_ENABLE();
|
|
/* CODEC_SAI pins configuration: FS, SCK, MCK and SD pins ------------------*/
|
|
gpio_init_structure.Pin = AUDIO_OUT_SAIx_FS_PIN;
|
|
gpio_init_structure.Mode = GPIO_MODE_AF_PP;
|
|
gpio_init_structure.Pull = GPIO_NOPULL;
|
|
gpio_init_structure.Speed = GPIO_SPEED_HIGH;
|
|
gpio_init_structure.Alternate = AUDIO_OUT_SAIx_FS_SD_MCLK_AF;
|
|
HAL_GPIO_Init(AUDIO_OUT_SAIx_FS_GPIO_PORT, &gpio_init_structure);
|
|
|
|
gpio_init_structure.Pin = AUDIO_OUT_SAIx_SCK_PIN;
|
|
gpio_init_structure.Mode = GPIO_MODE_AF_PP;
|
|
gpio_init_structure.Pull = GPIO_NOPULL;
|
|
gpio_init_structure.Speed = GPIO_SPEED_HIGH;
|
|
gpio_init_structure.Alternate = AUDIO_OUT_SAIx_SCK_AF;
|
|
HAL_GPIO_Init(AUDIO_OUT_SAIx_SCK_SD_GPIO_PORT, &gpio_init_structure);
|
|
|
|
gpio_init_structure.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_HIGH;
|
|
gpio_init_structure.Alternate = AUDIO_OUT_SAIx_FS_SD_MCLK_AF;
|
|
HAL_GPIO_Init(AUDIO_OUT_SAIx_SCK_SD_GPIO_PORT, &gpio_init_structure);
|
|
|
|
gpio_init_structure.Pin = AUDIO_OUT_SAIx_MCLK_PIN;
|
|
gpio_init_structure.Mode = GPIO_MODE_AF_PP;
|
|
gpio_init_structure.Pull = GPIO_NOPULL;
|
|
gpio_init_structure.Speed = GPIO_SPEED_HIGH;
|
|
gpio_init_structure.Alternate = AUDIO_OUT_SAIx_FS_SD_MCLK_AF;
|
|
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.Channel = AUDIO_OUT_SAIx_DMAx_CHANNEL;
|
|
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
|
|
* @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);
|
|
|
|
/* Deactives CODEC_SAI pins FS, SCK, MCK and SD by putting them in input mode */
|
|
gpio_init_structure.Pin = AUDIO_OUT_SAIx_FS_PIN;
|
|
HAL_GPIO_DeInit(AUDIO_OUT_SAIx_FS_GPIO_PORT, gpio_init_structure.Pin);
|
|
|
|
gpio_init_structure.Pin = AUDIO_OUT_SAIx_SCK_PIN;
|
|
HAL_GPIO_DeInit(AUDIO_OUT_SAIx_SCK_SD_GPIO_PORT, gpio_init_structure.Pin);
|
|
|
|
gpio_init_structure.Pin = AUDIO_OUT_SAIx_SD_PIN;
|
|
HAL_GPIO_DeInit(AUDIO_OUT_SAIx_SCK_SD_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 application
|
|
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
|
|
* @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))
|
|
{
|
|
/* Configure PLLI2S prescalers */
|
|
/* PLLI2S_VCO: VCO_429M
|
|
I2S_CLK(first level) = PLLI2S_VCO/PLLI2SQ = 429/2 = 214.5 Mhz
|
|
I2S_CLK_x = I2S_CLK(first level)/PLLI2SDIVQ = 214.5/19 = 11.289 Mhz */
|
|
rcc_ex_clk_init_struct.PeriphClockSelection = RCC_PERIPHCLK_SAI2;
|
|
rcc_ex_clk_init_struct.Sai2ClockSelection = RCC_SAI2CLKSOURCE_PLLI2S;
|
|
rcc_ex_clk_init_struct.PLLI2S.PLLI2SN = 429;
|
|
rcc_ex_clk_init_struct.PLLI2S.PLLI2SQ = 2;
|
|
rcc_ex_clk_init_struct.PLLI2SDivQ = 19;
|
|
|
|
HAL_RCCEx_PeriphCLKConfig(&rcc_ex_clk_init_struct);
|
|
|
|
}
|
|
else /* AUDIO_FREQUENCY_8K, AUDIO_FREQUENCY_16K, AUDIO_FREQUENCY_48K), AUDIO_FREQUENCY_96K */
|
|
{
|
|
/* I2S clock config
|
|
PLLI2S_VCO: VCO_344M
|
|
I2S_CLK(first level) = PLLI2S_VCO/PLLI2SQ = 344/7 = 49.142 Mhz
|
|
I2S_CLK_x = I2S_CLK(first level)/PLLI2SDIVQ = 49.142/1 = 49.142 Mhz */
|
|
rcc_ex_clk_init_struct.PeriphClockSelection = RCC_PERIPHCLK_SAI2;
|
|
rcc_ex_clk_init_struct.Sai2ClockSelection = RCC_SAI2CLKSOURCE_PLLI2S;
|
|
rcc_ex_clk_init_struct.PLLI2S.PLLI2SN = 344;
|
|
rcc_ex_clk_init_struct.PLLI2S.PLLI2SQ = 7;
|
|
rcc_ex_clk_init_struct.PLLI2SDivQ = 1;
|
|
|
|
HAL_RCCEx_PeriphCLKConfig(&rcc_ex_clk_init_struct);
|
|
}
|
|
}
|
|
|
|
/*******************************************************************************
|
|
Static Functions
|
|
*******************************************************************************/
|
|
|
|
/**
|
|
* @brief Initializes the output Audio Codec audio interface (SAI).
|
|
* @param AudioFreq: Audio frequency to be configured for the SAI peripheral.
|
|
* @note The default SlotActive configuration is set to CODEC_AUDIOFRAME_SLOT_0123
|
|
* and user can update this configuration using
|
|
* @retval None
|
|
*/
|
|
static void SAIx_Out_Init(uint32_t AudioFreq)
|
|
{
|
|
/* Initialize the haudio_out_sai Instance parameter */
|
|
haudio_out_sai.Instance = AUDIO_OUT_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.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_RISINGEDGE;
|
|
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 Deinitializes the output Audio Codec audio interface (SAI).
|
|
* @retval None
|
|
*/
|
|
static void SAIx_Out_DeInit(void)
|
|
{
|
|
/* Initialize the haudio_out_sai Instance parameter */
|
|
haudio_out_sai.Instance = AUDIO_OUT_SAIx;
|
|
|
|
/* Disable SAI peripheral */
|
|
__HAL_SAI_DISABLE(&haudio_out_sai);
|
|
|
|
HAL_SAI_DeInit(&haudio_out_sai);
|
|
}
|
|
|
|
/**
|
|
* @}
|
|
*/
|
|
|
|
/** @defgroup STM32746G_DISCOVERY_AUDIO_Out_Private_Functions STM32746G_DISCOVERY_AUDIO Out Private Functions
|
|
* @{
|
|
*/
|
|
|
|
/**
|
|
* @brief Initializes wave recording.
|
|
* @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_Init(uint32_t AudioFreq, uint32_t BitRes, uint32_t ChnlNbr)
|
|
{
|
|
return BSP_AUDIO_IN_InitEx(INPUT_DEVICE_DIGITAL_MICROPHONE_2, AudioFreq, BitRes, ChnlNbr);
|
|
}
|
|
|
|
/**
|
|
* @brief Initializes wave recording.
|
|
* @param InputDevice: INPUT_DEVICE_DIGITAL_MICROPHONE_2 or INPUT_DEVICE_INPUT_LINE_1
|
|
* @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_InitEx(uint16_t InputDevice, uint32_t AudioFreq, uint32_t BitRes, uint32_t ChnlNbr)
|
|
{
|
|
uint8_t ret = AUDIO_ERROR;
|
|
uint32_t deviceid = 0x00;
|
|
uint32_t slot_active;
|
|
|
|
if ((InputDevice != INPUT_DEVICE_INPUT_LINE_1) && /* Only INPUT_LINE_1 and MICROPHONE_2 inputs supported */
|
|
(InputDevice != INPUT_DEVICE_DIGITAL_MICROPHONE_2))
|
|
{
|
|
ret = AUDIO_ERROR;
|
|
}
|
|
else
|
|
{
|
|
/* Disable SAI */
|
|
SAIx_In_DeInit();
|
|
|
|
/* PLL clock is set depending on the AudioFreq (44.1khz vs 48khz groups) */
|
|
BSP_AUDIO_OUT_ClockConfig(&haudio_in_sai, 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 */
|
|
haudio_in_sai.Instance = AUDIO_IN_SAIx;
|
|
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_OUT_MspInit(&haudio_in_sai, NULL); /* Initialize GPIOs for SAI2 block A Master signals */
|
|
BSP_AUDIO_IN_MspInit(&haudio_in_sai, NULL);
|
|
}
|
|
|
|
/* Configure SAI in master RX mode :
|
|
* - SAI2_block_A in master RX mode
|
|
* - SAI2_block_B in slave RX mode synchronous from SAI2_block_A
|
|
*/
|
|
if (InputDevice == INPUT_DEVICE_DIGITAL_MICROPHONE_2)
|
|
{
|
|
slot_active = CODEC_AUDIOFRAME_SLOT_13;
|
|
}
|
|
else
|
|
{
|
|
slot_active = CODEC_AUDIOFRAME_SLOT_02;
|
|
}
|
|
SAIx_In_Init(SAI_MODEMASTER_RX, 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, 100, AudioFreq);
|
|
}
|
|
}
|
|
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(uint16_t InputDevice, uint16_t OutputDevice, uint32_t AudioFreq, uint32_t BitRes, uint32_t ChnlNbr)
|
|
{
|
|
uint8_t ret = AUDIO_ERROR;
|
|
uint32_t deviceid = 0x00;
|
|
uint32_t slot_active;
|
|
|
|
if (InputDevice != INPUT_DEVICE_DIGITAL_MICROPHONE_2) /* Only MICROPHONE_2 input supported */
|
|
{
|
|
ret = AUDIO_ERROR;
|
|
}
|
|
else
|
|
{
|
|
/* Disable SAI */
|
|
SAIx_In_DeInit();
|
|
SAIx_Out_DeInit();
|
|
|
|
/* PLL clock is set depending on the AudioFreq (44.1khz vs 48khz groups) */
|
|
BSP_AUDIO_OUT_ClockConfig(&haudio_in_sai, 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 */
|
|
haudio_in_sai.Instance = AUDIO_IN_SAIx;
|
|
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(&haudio_in_sai, 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_OUT_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);
|
|
}
|
|
|
|
/* Configure SAI in master mode :
|
|
* - SAI2_block_A in master TX mode
|
|
* - SAI2_block_B in slave RX mode synchronous from SAI2_block_A
|
|
*/
|
|
if (InputDevice == INPUT_DEVICE_DIGITAL_MICROPHONE_2)
|
|
{
|
|
slot_active = CODEC_AUDIOFRAME_SLOT_13;
|
|
}
|
|
else
|
|
{
|
|
slot_active = CODEC_AUDIOFRAME_SLOT_02;
|
|
}
|
|
SAIx_In_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, 100, AudioFreq);
|
|
}
|
|
}
|
|
return ret;
|
|
}
|
|
|
|
|
|
/**
|
|
* @brief Starts audio recording.
|
|
* @param pbuf: Main buffer pointer for the recorded data storing
|
|
* @param size: size of the recorded buffer in number of elements (typically number of half-words)
|
|
* Be careful that it is not the same unit than BSP_AUDIO_OUT_Play function
|
|
* @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_SAI_Receive_DMA(&haudio_in_sai, (uint8_t*)pbuf, size);
|
|
|
|
/* Return AUDIO_OK when all operations are correctly done */
|
|
ret = AUDIO_OK;
|
|
|
|
return ret;
|
|
}
|
|
|
|
/**
|
|
* @brief Stops audio recording.
|
|
* @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_IN_Stop(uint32_t Option)
|
|
{
|
|
/* Call the Media layer stop function */
|
|
HAL_SAI_DMAStop(&haudio_in_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 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_SAI_DMAPause(&haudio_in_sai);
|
|
/* 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_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 in range 0(Mute)..80(+0dB)..100(+17.625dB)
|
|
* @retval AUDIO_OK if correct communication, else wrong communication
|
|
*/
|
|
uint8_t BSP_AUDIO_IN_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
|
|
{
|
|
/* 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();
|
|
/* DeInit the SAI MSP : this __weak function can be rewritten by the application */
|
|
BSP_AUDIO_IN_MspDeInit(&haudio_in_sai, NULL);
|
|
}
|
|
|
|
/**
|
|
* @brief Rx Transfer completed callbacks.
|
|
* @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();
|
|
}
|
|
|
|
/**
|
|
* @brief Rx Half Transfer completed callbacks.
|
|
* @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 (its prototype is already declared in stm32746g_discovery_audio.h) */
|
|
BSP_AUDIO_IN_HalfTransfer_CallBack();
|
|
}
|
|
|
|
/**
|
|
* @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 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 Initializes BSP_AUDIO_IN MSP.
|
|
* @param hsai: SAI handle
|
|
* @param Params
|
|
* @retval None
|
|
*/
|
|
__weak void BSP_AUDIO_IN_MspInit(SAI_HandleTypeDef *hsai, void *Params)
|
|
{
|
|
static DMA_HandleTypeDef hdma_sai_rx;
|
|
GPIO_InitTypeDef gpio_init_structure;
|
|
|
|
/* 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_FAST;
|
|
gpio_init_structure.Alternate = AUDIO_IN_SAIx_SD_AF;
|
|
HAL_GPIO_Init(AUDIO_IN_SAIx_SD_GPIO_PORT, &gpio_init_structure);
|
|
|
|
/* Enable Audio INT GPIO clock */
|
|
AUDIO_IN_INT_GPIO_ENABLE();
|
|
/* Audio INT pin configuration: input */
|
|
gpio_init_structure.Pin = AUDIO_IN_INT_GPIO_PIN;
|
|
gpio_init_structure.Mode = GPIO_MODE_INPUT;
|
|
gpio_init_structure.Pull = GPIO_NOPULL;
|
|
gpio_init_structure.Speed = GPIO_SPEED_FAST;
|
|
HAL_GPIO_Init(AUDIO_IN_INT_GPIO_PORT, &gpio_init_structure);
|
|
|
|
/* Enable the DMA clock */
|
|
AUDIO_IN_SAIx_DMAx_CLK_ENABLE();
|
|
|
|
if(hsai->Instance == AUDIO_IN_SAIx)
|
|
{
|
|
/* Configure the hdma_sai_rx handle parameters */
|
|
hdma_sai_rx.Init.Channel = AUDIO_IN_SAIx_DMAx_CHANNEL;
|
|
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);
|
|
|
|
/* Audio INT IRQ Channel configuration */
|
|
HAL_NVIC_SetPriority(AUDIO_IN_INT_IRQ, AUDIO_IN_IRQ_PREPRIO, 0);
|
|
HAL_NVIC_EnableIRQ(AUDIO_IN_INT_IRQ);
|
|
}
|
|
|
|
/**
|
|
* @brief DeInitializes BSP_AUDIO_IN MSP.
|
|
* @param hsai: SAI handle
|
|
* @param Params
|
|
* @retval None
|
|
*/
|
|
__weak void BSP_AUDIO_IN_MspDeInit(SAI_HandleTypeDef *hsai, void *Params)
|
|
{
|
|
GPIO_InitTypeDef gpio_init_structure;
|
|
|
|
static DMA_HandleTypeDef hdma_sai_rx;
|
|
|
|
/* SAI IN DMA IRQ Channel deactivation */
|
|
HAL_NVIC_DisableIRQ(AUDIO_IN_SAIx_DMAx_IRQ);
|
|
|
|
if(hsai->Instance == AUDIO_IN_SAIx)
|
|
{
|
|
/* Deinitialize the Stream for new transfer */
|
|
HAL_DMA_DeInit(&hdma_sai_rx);
|
|
}
|
|
|
|
/* Disable SAI block */
|
|
__HAL_SAI_DISABLE(hsai);
|
|
|
|
/* Disable pin: SD pin */
|
|
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();
|
|
|
|
/* GPIO pins clock and DMA clock can be shut down in the application
|
|
by surcharging this __weak function */
|
|
}
|
|
|
|
|
|
/*******************************************************************************
|
|
Static Functions
|
|
*******************************************************************************/
|
|
|
|
/**
|
|
* @brief Initializes the input Audio Codec audio interface (SAI).
|
|
* @param SaiOutMode: SAI_MODEMASTER_TX (for record and playback in parallel)
|
|
* or SAI_MODEMASTER_RX (for record only).
|
|
* @param SlotActive: CODEC_AUDIOFRAME_SLOT_02 or CODEC_AUDIOFRAME_SLOT_13
|
|
* @param AudioFreq: Audio frequency to be configured for the SAI peripheral.
|
|
* @retval None
|
|
*/
|
|
static void SAIx_In_Init(uint32_t SaiOutMode, uint32_t SlotActive, uint32_t AudioFreq)
|
|
{
|
|
/* Initialize SAI2 block A in MASTER RX */
|
|
/* Initialize the haudio_out_sai Instance parameter */
|
|
haudio_out_sai.Instance = AUDIO_OUT_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.AudioMode = SaiOutMode;
|
|
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_RISINGEDGE;
|
|
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 = SlotActive;
|
|
|
|
HAL_SAI_Init(&haudio_out_sai);
|
|
|
|
/* Initialize SAI2 block B in SLAVE RX synchronous from SAI2 block A */
|
|
/* Initialize the haudio_in_sai Instance parameter */
|
|
haudio_in_sai.Instance = AUDIO_IN_SAIx;
|
|
|
|
/* 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.AudioFrequency = AudioFreq;
|
|
haudio_in_sai.Init.AudioMode = SAI_MODESLAVE_RX;
|
|
haudio_in_sai.Init.NoDivider = SAI_MASTERDIVIDER_ENABLED;
|
|
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_DISABLED;
|
|
haudio_in_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_in_sai.FrameInit.FrameLength = 64;
|
|
haudio_in_sai.FrameInit.ActiveFrameLength = 32;
|
|
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;
|
|
|
|
HAL_SAI_Init(&haudio_in_sai);
|
|
|
|
/* Enable SAI peripheral to generate MCLK */
|
|
__HAL_SAI_ENABLE(&haudio_out_sai);
|
|
|
|
/* Enable SAI peripheral */
|
|
__HAL_SAI_ENABLE(&haudio_in_sai);
|
|
}
|
|
|
|
|
|
|
|
/**
|
|
* @brief Deinitializes the output Audio Codec audio interface (SAI).
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* @retval None
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*/
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static void SAIx_In_DeInit(void)
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{
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/* Initialize the haudio_in_sai Instance parameter */
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haudio_in_sai.Instance = AUDIO_IN_SAIx;
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/* Disable SAI peripheral */
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__HAL_SAI_DISABLE(&haudio_in_sai);
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HAL_SAI_DeInit(&haudio_in_sai);
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}
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/**
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* @}
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*/
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/**
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* @}
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*/
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/**
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* @}
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*/
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/**
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* @}
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*/
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/************************ (C) COPYRIGHT STMicroelectronics *****END OF FILE****/
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