File content as of revision 1:7046ce26b7ed:

/* Copyright (c) 2010-2016 mbed.org, MIT License
*
* Permission is hereby granted, free of charge, to any person obtaining a copy of this software
* and associated documentation files (the "Software"), to deal in the Software without
* restriction, including without limitation the rights to use, copy, modify, merge, publish,
* distribute, sublicense, and/or sell copies of the Software, and to permit persons to whom the
* Software is furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in all copies or
* substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR IMPLIED, INCLUDING
* BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND
* NONINFRINGEMENT. IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM,
* DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
*/

#ifndef __AUDIO_DISCO_F746NG_H
#define __AUDIO_DISCO_F746NG_H

#ifdef TARGET_DISCO_F746NG

#include "mbed.h"
#include "stm32746g_discovery_audio.h"

/*
  This class drives the uSD card driver mounted on DISCO_F746NG board.

  Usage:

  #include "mbed.h"
  #include "AUDIO_DISCO_F746NG.h"

  AUDIO_DISCO_F746NG audio;

  int main()
  {
      audio.IN_OUT_Init(INPUT_DEVICE_DIGITAL_MICROPHONE_2, OUTPUT_DEVICE_HEADPHONE, 90, DEFAULT_AUDIO_IN_FREQ):
      audio.IN_Record((uint16_t*)AUDIO_BUFFER_IN, AUDIO_BLOCK_SIZE);
      audio.OUT_SetAudioFrameSlot(CODEC_AUDIOFRAME_SLOT_02);
      audio.OUT_Play((uint16_t*)AUDIO_BUFFER_OUT, AUDIO_BLOCK_SIZE * 2);
      while 1 {}

  }
*/
class AUDIO_DISCO_F746NG
{

public:
    //! Constructor
    AUDIO_DISCO_F746NG();

    //! Destructor
    ~AUDIO_DISCO_F746NG();

    /**
      * @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 OUT_Init(uint16_t OutputDevice, uint8_t Volume, uint32_t AudioFreq);

    /**
      * @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 in BYTES unit.
      *         In memory, first element is for left channel, second element is for right channel
      * @retval AUDIO_OK if correct communication, else wrong communication
      */
    uint8_t OUT_Play(uint16_t* pBuffer, uint32_t Size);

    /**
      * @brief  Sends n-Bytes on the SAI interface.
      * @param  pData: pointer on data address
      * @param  Size: number of data to be written
      * @retval None
      */
    void OUT_ChangeBuffer(uint16_t *pData, uint16_t Size);

    /**
      * @brief  This function Pauses the audio file stream. In case
      *         of using DMA, the DMA Pause feature is used.
      * @note When calling OUT_Pause(); function for pause, only
      *          OUT_Resume(); function should be called for resume (use of BSP_AUDIO_OUT_Play()
      *          function for resume could lead to unexpected behaviour);.
      * @retval AUDIO_OK if correct communication, else wrong communication
      */
    uint8_t OUT_Pause(void);

    /**
      * @brief  This function  Resumes the audio file stream.
      * @note When calling OUT_Pause(); function for pause, only
      *          OUT_Resume(); function should be called for resume (use of BSP_AUDIO_OUT_Play()
      *          function for resume could lead to unexpected behaviour);.
      * @retval AUDIO_OK if correct communication, else wrong communication
      */
    uint8_t OUT_Resume(void);

    /**
      * @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 OUT_Stop(uint32_t Option);

    /**
      * @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 OUT_SetVolume(uint8_t Volume);

    /**
      * @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 OUT_SetMute(uint32_t Cmd);

    /**
      * @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 OUT_SetOutputMode(uint8_t Output);

    /**
      * @brief  Updates the audio frequency.
      * @param  AudioFreq: Audio frequency used to play the audio stream.
      * @note   This API should be called after the OUT_Init(); to adjust the
      *         audio frequency.
      * @retval None
      */
    void OUT_SetFrequency(uint32_t AudioFreq);

    /**
      * @brief  Updates the Audio frame slot configuration.
      * @param  AudioFrameSlot: specifies the audio Frame slot
      *         This parameter can be one of the following values
      *            @arg CODEC_AUDIOFRAME_SLOT_0123
      *            @arg CODEC_AUDIOFRAME_SLOT_02
      *            @arg CODEC_AUDIOFRAME_SLOT_13
      * @note   This API should be called after the OUT_Init(); to adjust the
      *         audio frame slot.
      * @retval None
      */
    void OUT_SetAudioFrameSlot(uint32_t AudioFrameSlot);

    /**
      * @brief  Deinit the audio peripherals.
      * @retval None
      */
    void OUT_DeInit(void);

    /**
      * @brief  Tx Transfer completed callbacks.
      * @param  hsai: SAI handle
      * @retval None
      */
    void HAL_SAI_TxCpltCallback(SAI_HandleTypeDef *hsai);

    /**
      * @brief  Tx Half Transfer completed callbacks.
      * @param  hsai: SAI handle
      * @retval None
      */
    void HAL_SAI_TxHalfCpltCallback(SAI_HandleTypeDef *hsai);

    /**
      * @brief  SAI error callbacks.
      * @param  hsai: SAI handle
      * @retval None
      */
    void HAL_SAI_ErrorCallback(SAI_HandleTypeDef *hsai);

    /**
      * @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);

    /**
      * @brief  Deinitializes the output Audio Codec audio interface (SAI);.
      * @retval None
      */
    static void SAIx_Out_DeInit(void);

    /**
      * @brief  Initializes wave recording.
      * @param  InputDevice: INPUT_DEVICE_DIGITAL_MICROPHONE_2 or INPUT_DEVICE_INPUT_LINE_1
      * @param  Volume: Initial volume level (in range 0(Mute);..80(+0dB)..100(+17.625dB))
      * @param  AudioFreq: Audio frequency to be configured for the SAI peripheral.
      * @retval AUDIO_OK if correct communication, else wrong communication
      */
    uint8_t IN_Init(uint16_t InputDevice, uint8_t Volume, uint32_t AudioFreq);

    /**
      * @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 IN_OUT_Init(uint16_t InputDevice, uint16_t OutputDevice, uint32_t AudioFreq, uint32_t BitRes, uint32_t ChnlNbr);

    /**
      * @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 OUT_Play function
      * @retval AUDIO_OK if correct communication, else wrong communication
      */
    uint8_t  IN_Record(uint16_t* pbuf, uint32_t size);

    /**
      * @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 IN_Stop(uint32_t Option);

    /**
      * @brief  Pauses the audio file stream.
      * @retval AUDIO_OK if correct communication, else wrong communication
      */
    uint8_t IN_Pause(void);

    /**
      * @brief  Resumes the audio file stream.
      * @retval AUDIO_OK if correct communication, else wrong communication
      */
    uint8_t IN_Resume(void);

    /**
      * @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 IN_SetVolume(uint8_t Volume);

    /**
      * @brief  Deinit the audio IN peripherals.
      * @retval None
      */
    void IN_DeInit(void);

    /**
      * @brief  Rx Transfer completed callbacks.
      * @param  hsai: SAI handle
      * @retval None
      */
    void HAL_SAI_RxCpltCallback(SAI_HandleTypeDef *hsai);

    /**
      * @brief  Rx Half Transfer completed callbacks.
      * @param  hsai: SAI handle
      * @retval None
      */
    void HAL_SAI_RxHalfCpltCallback(SAI_HandleTypeDef *hsai);

    /**
      * @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);

    /**
      * @brief  Deinitializes the output Audio Codec audio interface (SAI);.
      * @retval None
      */
    static void SAIx_In_DeInit(void);

private:

};

#else
#error "This class must be used with DISCO_F746NG board only."
#endif // TARGET_DISCO_F746NG

#endif