コード例 #1
0
ファイル: taskAK5394A.c プロジェクト: lofturj/sdr-widget
/*! \brief The PDCA interrupt handler.
 *
 * The handler reload the PDCA settings with the correct address and size using the reload register.
 * The interrupt will happen when the reload counter reaches 0
 */
__attribute__((__interrupt__)) static void spk_pdca_int_handler(void)
{
			if (spk_buffer_out == 0)
			  {
				// Set PDCA channel reload values with address where data to load are stored, and size of the data block to load.
			   pdca_reload_channel(PDCA_CHANNEL_SSC_TX, (void *)spk_buffer_1, SPK_BUFFER_SIZE);
			   spk_buffer_out = 1;
			  }
			  else
			  {
				pdca_reload_channel(PDCA_CHANNEL_SSC_TX, (void *)spk_buffer_0, SPK_BUFFER_SIZE);
				spk_buffer_out = 0;
			  }
}
コード例 #2
0
ファイル: taskAK5394A.c プロジェクト: lofturj/sdr-widget
/*! \brief The PDCA interrupt handler.
 *
 * The handler reload the PDCA settings with the correct address and size using the reload register.
 * The interrupt will happen when the reload counter reaches 0
 */
__attribute__((__interrupt__)) static void pdca_int_handler(void)
{
			if (audio_buffer_in == 0)
			  {
				// Set PDCA channel reload values with address where data to load are stored, and size of the data block to load.
			   pdca_reload_channel(PDCA_CHANNEL_SSC_RX, (void *)audio_buffer_1, AUDIO_BUFFER_SIZE);
			   audio_buffer_in = 1;
			  }
			  else
			  {
				pdca_reload_channel(PDCA_CHANNEL_SSC_RX, (void *)audio_buffer_0, AUDIO_BUFFER_SIZE);
				audio_buffer_in = 0;
			  }

}
コード例 #3
0
ファイル: pevc_example1.c プロジェクト: kerichsen/asf
__interrupt
#endif
static void pdca_int_handler(void)
{
  u32PdcaIsr = pdca_channel->isr;
  if( u32PdcaIsr & (1<<AVR32_PDCA_ISR0_TRC_OFFSET) )
  {
    // Count the number of Transfer Complete interrupts.
    pdca_reload_channel(PDCA_CHANNEL_USART, (void *)aDataTransfered, sizeof( aDataTransfered ));
    print_dbg("\r\n");
  }
}
コード例 #4
0
ファイル: tpa6130.c プロジェクト: InSoonPark/asf
/*! \brief Flushes the sample buffer being output to the ABDAC.
 */
void tpa6130_dac_flush(void)
{
  pdca_disable_interrupt_transfer_complete(TPA6130_ABDAC_PDCA_CHANNEL);
  pdca_disable_interrupt_reload_counter_zero(TPA6130_ABDAC_PDCA_CHANNEL);

  /*TODO Do we really want to wait here? Or do we just don't care when
   * the buffer is empty/flushed */
  //while(!pdca_get_transfer_status(TPA6130_ABDAC_PDCA_CHANNEL) &
  //  PDCA_TRANSFER_COMPLETE);
  pdca_disable       (TPA6130_ABDAC_PDCA_CHANNEL       );
  pdca_load_channel  (TPA6130_ABDAC_PDCA_CHANNEL,0x0, 0);
  pdca_reload_channel(TPA6130_ABDAC_PDCA_CHANNEL,0x0, 0);
  pdca_enable        (TPA6130_ABDAC_PDCA_CHANNEL       );
}
コード例 #5
0
ファイル: tlv320aic23b.c プロジェクト: Mazetti/asf
bool aic23b_adc_input(void *sample_buffer, size_t sample_length)
{
  if (!(pdca_get_transfer_status(AIC23B_SSC_RX_PDCA_CHANNEL) &
        PDCA_TRANSFER_COUNTER_RELOAD_IS_ZERO))
    return false;

  if (sample_length)
  {
    pdca_reload_channel(AIC23B_SSC_RX_PDCA_CHANNEL, sample_buffer, sample_length * 2);
    pdca_get_reload_size(AIC23B_SSC_RX_PDCA_CHANNEL);

    if (aic23b_output_params.callback_opt & AUDIO_ADC_OUT_OF_SAMPLE_CB)
      pdca_enable_interrupt_transfer_complete(AIC23B_SSC_RX_PDCA_CHANNEL);
    if (aic23b_output_params.callback_opt & AUDIO_ADC_RELOAD_CB)
      pdca_enable_interrupt_reload_counter_zero(AIC23B_SSC_RX_PDCA_CHANNEL);
  }

  return true;
}
コード例 #6
0
ファイル: tlv320aic23b.c プロジェクト: Mazetti/asf
void aic23b_dac_start(uint32_t sample_rate_hz,
                      uint8_t num_channels,
                      uint8_t bits_per_sample,
                      bool swap_channels,
                      void (*callback)(uint32_t arg),
                      uint32_t callback_opt,
                      uint32_t pba_hz)
{
#if AIC23B_CTRL_INTERFACE == AIC23B_CTRL_INTERFACE_SPI
  static const spi_options_t AIC23B_SPI_OPTIONS =
  {
    .reg          = AIC23B_SPI_NPCS,
    .baudrate     = AIC23B_SPI_MASTER_SPEED,
    .bits         = AIC23B_CTRL_SIZE,
    .spck_delay   = 0,
    .trans_delay  = 0,
    .stay_act     = 0,
    .spi_mode     = 3,
    .modfdis      = 1
  };
  spi_setupChipReg(AIC23B_SPI, &AIC23B_SPI_OPTIONS, pba_hz);
#endif

  aic23b_dac_stop();

  gpio_enable_module(AIC23B_SSC_DAC_GPIO_MAP,
                     sizeof(AIC23B_SSC_DAC_GPIO_MAP) / sizeof(AIC23B_SSC_DAC_GPIO_MAP[0]));

  aic23b_pdc_t pdc;
  pdc.data  = AIC23B_DEFAULT(AIC23B_PDC);
  pdc.off   = 0;
  pdc.clk   = 0;
  pdc.osc   = 0;
  pdc.out   = 0;
  pdc.dac   = 0;
  pdc.adc   = 1;
  pdc.mic   = 1;
  pdc.line  = 1;
  aic23b_set_power_down_state(pdc);

  aic23b_dac_setup(sample_rate_hz,
                   num_channels,
                   bits_per_sample,
                   swap_channels,
                   callback,
                   callback_opt,
                   pba_hz);

  aic23b_aapc_t aapc;
  aapc.data = AIC23B_DEFAULT(AIC23B_AAPC);
  aapc.ste  = 0;
  aapc.dac  = 1;
  aapc.byp  = 0;
  aapc.micm = 1;
  aapc.micb = 0;
  aic23b_set_analog_audio_path(aapc);

  aic23b_dapc_t dapc;
  dapc.data   = AIC23B_DEFAULT(AIC23B_DAPC);
  dapc.dacm   = 0;
  dapc.deemp  = AIC23B_DAPC_DEEMP_NONE;
  dapc.adchp  = 1;
  aic23b_set_digital_audio_path(dapc);

  // set an acceptable start volume
  aic23b_set_headphone_volume(AIC23B_LEFT_CHANNEL | AIC23B_RIGHT_CHANNEL,
                              -30,
                              true);

  aic23b_activate_dig_audio(true);

  INTC_register_interrupt(&aic23b_ssc_tx_pdca_int_handler,
                          AIC23B_SSC_TX_PDCA_IRQ,
                          AIC23B_SSC_TX_PDCA_INT_LEVEL);
}


void aic23b_dac_setup(uint32_t sample_rate_hz,
                      uint8_t num_channels,
                      uint8_t bits_per_sample,
                      bool swap_channels,
                      void (*callback)(uint32_t arg),
                      uint32_t callback_opt,
                      uint32_t pba_hz)
{
#if defined(AIC23B_DAC_USE_RX_CLOCK) && AIC23B_DAC_USE_RX_CLOCK == true
  #if defined(AIC23B_DAC_RX_CLOCK_SET_CALLBACK)
    AIC23B_DAC_RX_CLOCK_SET_CALLBACK(2 * sample_rate_hz *
                                     ((bits_per_sample <= 16) ? 16 :
                                     (bits_per_sample <= 20) ? 20 :
                                     (bits_per_sample <= 24) ? 24 :
                                     32));
  #endif
  ssc_i2s_init(AIC23B_SSC,
               sample_rate_hz,
               bits_per_sample,
               (bits_per_sample <= 16) ? 16 :
               (bits_per_sample <= 20) ? 20 :
               (bits_per_sample <= 24) ? 24 :
                                         32,
               SSC_I2S_MODE_STEREO_OUT_EXT_CLK,
               pba_hz);
#else
  ssc_i2s_init(AIC23B_SSC,
               sample_rate_hz,
               bits_per_sample,
               (bits_per_sample <= 16) ? 16 :
               (bits_per_sample <= 20) ? 20 :
               (bits_per_sample <= 24) ? 24 :
                                         32,
               SSC_I2S_MODE_STEREO_OUT,
               pba_hz);
#endif

  pdca_channel_options_t aic23b_ssc_pdca_options =
  {
    .addr           = NULL,
    .size           = 0,
    .r_addr         = NULL,
    .r_size         = 0,
    .pid            = AIC23B_SSC_TX_PDCA_PID,
    .transfer_size  = (bits_per_sample <=  8) ? PDCA_TRANSFER_SIZE_BYTE      :
                      (bits_per_sample <= 16) ? PDCA_TRANSFER_SIZE_HALF_WORD :
                                                PDCA_TRANSFER_SIZE_WORD
  };
  pdca_init_channel(AIC23B_SSC_TX_PDCA_CHANNEL, &aic23b_ssc_pdca_options);
  pdca_enable(AIC23B_SSC_TX_PDCA_CHANNEL);

#if !defined(AIC23B_DAC_USE_RX_CLOCK) || AIC23B_DAC_USE_RX_CLOCK == false || \
    !defined(AIC23B_DAC_RX_CLOCK_SET_CALLBACK)
  // Set DAC frequency
  aic23b_configure_freq(AIC23B_MCLK_HZ, sample_rate_hz);
#endif

  aic23b_daif_t daif;
  daif.data   = AIC23B_DEFAULT(AIC23B_DAIF);
  daif.ms     = AIC23B_DAIF_MS_SLAVE;
  daif.lrswap = swap_channels;
  daif.lrp    = 0;
  daif.iwl    = (bits_per_sample <= 16) ? AIC23B_DAIF_IWL_16 :
                (bits_per_sample <= 20) ? AIC23B_DAIF_IWL_20 :
                (bits_per_sample <= 24) ? AIC23B_DAIF_IWL_24 :
                                          AIC23B_DAIF_IWL_32;
  daif.fmt    = AIC23B_DAIF_FMT_I2S;
  aic23b_write_reg(AIC23B_DAIF, daif.data);

  aic23b_output_params.num_channels             = num_channels;
  aic23b_output_params.callback                 = callback;
  aic23b_output_params.callback_opt             = callback_opt;
}
#endif

bool aic23b_dac_output(void *sample_buffer, size_t sample_length)
{
  bool global_interrupt_enabled;

  if (!(pdca_get_transfer_status(AIC23B_SSC_TX_PDCA_CHANNEL) &
        PDCA_TRANSFER_COUNTER_RELOAD_IS_ZERO))
    return false;

  if (sample_length)
  {
    if (aic23b_output_params.num_channels == 1)
    {
      int16_t *s16_sample_buffer = sample_buffer;
      int i;

      for (i = sample_length - 1; i >= 0; i--)
      {
        s16_sample_buffer[2 * i + 1] =
        s16_sample_buffer[2 * i]     = s16_sample_buffer[i];
      }
    }

    // The PDCA is not able to synchronize its start of transfer with the SSC
    // start of period, so this has to be done by polling the TF pin.
    // Not doing so may result in channels being swapped randomly.
    if ((global_interrupt_enabled = Is_global_interrupt_enabled()))
      Disable_global_interrupt();
    if (pdca_get_transfer_status(AIC23B_SSC_TX_PDCA_CHANNEL) &
        PDCA_TRANSFER_COMPLETE)
    {
      while (gpio_get_pin_value(AIC23B_SSC_TX_FRAME_SYNC_PIN));
      while (!gpio_get_pin_value(AIC23B_SSC_TX_FRAME_SYNC_PIN));
    }
    pdca_reload_channel(AIC23B_SSC_TX_PDCA_CHANNEL, sample_buffer, sample_length * 2);
    pdca_get_reload_size(AIC23B_SSC_TX_PDCA_CHANNEL);
    if (global_interrupt_enabled)
      Enable_global_interrupt();

    if (aic23b_output_params.callback_opt & AUDIO_DAC_OUT_OF_SAMPLE_CB)
      pdca_enable_interrupt_transfer_complete(AIC23B_SSC_TX_PDCA_CHANNEL);
    if (aic23b_output_params.callback_opt & AUDIO_DAC_RELOAD_CB)
      pdca_enable_interrupt_reload_counter_zero(AIC23B_SSC_TX_PDCA_CHANNEL);
  }
  return true;
}
コード例 #7
0
/**
 * \brief Set a given number of pixels to the same color
 *
 * Use this function to write a certain number of pixels to the same color
 * within a set limit.
 *
 * Limits have to be set prior to calling this function, e.g.:
 * \code
 * ili9341_set_top_left_limit(0, 0);
 * ili9341_set_bottom_right_limit(320, 240);
 * ...
 * \endcode
 *
 * \param color The color to write to the display
 * \param count The number of pixels to write with this color
 */
void ili9341_duplicate_pixel(const ili9341_color_t color, uint32_t count)
{
	/* Sanity check to make sure that the pixel count is not zero */
	Assert(count > 0);

	ili9341_send_command(ILI9341_CMD_MEMORY_WRITE);

#if defined(ILI9341_DMA_ENABLED)
	ili9341_color_t chunk_buf[ILI9341_DMA_CHUNK_SIZE];
	uint32_t chunk_len;

#  if SAM
	Pdc *SPI_DMA = spi_get_pdc_base(CONF_ILI9341_SPI);
	pdc_packet_t spi_pdc_data;

	pdc_enable_transfer(SPI_DMA, PERIPH_PTCR_TXTEN);
	spi_pdc_data.ul_addr = (uint32_t)chunk_buf;
#  elif UC3
	pdca_set_transfer_size(CONF_ILI9341_PDCA_CHANNEL,
			PDCA_TRANSFER_SIZE_BYTE);
	pdca_set_peripheral_select(CONF_ILI9341_PDCA_CHANNEL,
			CONF_ILI9341_PDCA_PID);
#  endif

	for (uint32_t i = 0; i < ILI9341_DMA_CHUNK_SIZE; i++) {
		chunk_buf[i] = le16_to_cpu(color);
	}

	while (count)
	{
		chunk_len = min(ILI9341_DMA_CHUNK_SIZE, count);

		ili9341_wait_for_send_done();

#  if SAM
		spi_pdc_data.ul_size = (uint32_t)sizeof(ili9341_color_t) * chunk_len;
		pdc_tx_init(SPI_DMA, NULL, &spi_pdc_data);
#  elif UC3
		pdca_reload_channel(CONF_ILI9341_PDCA_CHANNEL, chunk_buf,
				(uint32_t)sizeof(ili9341_color_t) * chunk_len);
		pdca_enable(CONF_ILI9341_PDCA_CHANNEL);
#  endif

		count -= chunk_len;
	}

	ili9341_wait_for_send_done();
	ili9341_deselect_chip();

#  if SAM
	pdc_disable_transfer(SPI_DMA, PERIPH_PTCR_TXTEN);
#  elif UC3
	pdca_disable(CONF_ILI9341_PDCA_CHANNEL);
#  endif
#else
	while (count--) {
		ili9341_send_byte(color);
		ili9341_send_byte(color >> 8);
	}

	ili9341_wait_for_send_done();
	ili9341_deselect_chip();
#endif
}
コード例 #8
0
/**
 * \brief Copy pixels from SRAM to the screen
 *
 * Used to copy a large quantitative of data to the screen in one go.
 *
 * Limits have to be set prior to calling this function, e.g.:
 * \code
 * ili9341_set_top_left_limit(0, 0);
 * ili9341_set_bottom_right_limit(320, 240);
 * ...
 * \endcode
 *
 * \param pixels Pointer to the pixel data
 * \param count Number of pixels to copy to the screen
 */
void ili9341_copy_pixels_to_screen(const ili9341_color_t *pixels, uint32_t count)
{
	const ili9341_color_t *pixel = pixels;

	/* Sanity check to make sure that the pixel count is not zero */
	Assert(count > 0);

	ili9341_send_command(ILI9341_CMD_MEMORY_WRITE);

#if defined(ILI9341_DMA_ENABLED)
	ili9341_color_t chunk_buf[ILI9341_DMA_CHUNK_SIZE];
	uint32_t chunk_len;

#  if SAM
	Pdc *SPI_DMA = spi_get_pdc_base(CONF_ILI9341_SPI);
	pdc_packet_t spi_pdc_data;

	pdc_enable_transfer(SPI_DMA, PERIPH_PTCR_TXTEN);
	spi_pdc_data.ul_addr = (uint32_t)chunk_buf;
#  elif UC3
	pdca_set_transfer_size(CONF_ILI9341_PDCA_CHANNEL,
			PDCA_TRANSFER_SIZE_BYTE);
	pdca_set_peripheral_select(CONF_ILI9341_PDCA_CHANNEL,
			CONF_ILI9341_PDCA_PID);
#  endif

	while (count)
	{
		/* We need to copy out the data to send in chunks into RAM, as the PDC
		 * does not allow FLASH->Peripheral transfers */
		chunk_len = min(ILI9341_DMA_CHUNK_SIZE, count);

		/* Wait for pending transfer to complete */
		ili9341_wait_for_send_done();

		for (uint32_t i = 0; i < chunk_len; i++) {
			chunk_buf[i] = le16_to_cpu(pixel[i]);
		}

#  if SAM
		spi_pdc_data.ul_size = (uint32_t)sizeof(ili9341_color_t) * chunk_len;
		pdc_tx_init(SPI_DMA, NULL, &spi_pdc_data);
#  elif UC3
		pdca_reload_channel(CONF_ILI9341_PDCA_CHANNEL, chunk_buf,
				(uint32_t)sizeof(ili9341_color_t) * chunk_len);
		pdca_enable(CONF_ILI9341_PDCA_CHANNEL);
#  endif

		pixel += chunk_len;
		count -= chunk_len;
	}

	ili9341_wait_for_send_done();
	ili9341_deselect_chip();

#  if SAM
	pdc_disable_transfer(SPI_DMA, PERIPH_PTCR_TXTEN);
#  elif UC3
	pdca_disable(CONF_ILI9341_PDCA_CHANNEL);
#  endif
#else
	while (count--) {
		ili9341_send_byte(*pixel);
		ili9341_send_byte(*pixel >> 8);

		pixel++;
	}

	ili9341_wait_for_send_done();
	ili9341_deselect_chip();
#endif
}
コード例 #9
0
ファイル: tpa6130.c プロジェクト: InSoonPark/asf
/*! \brief Sets the DACs up with new settings.
 *
 * \note The DACs must have been started beforehand.
 */
void tpa6130_dac_setup(uint32_t sample_rate_hz,
                       uint8_t num_channels,
                       uint8_t bits_per_sample,
                       bool swap_channels,
                       void (*callback)(uint32_t arg),
                       uint32_t callback_opt,
                       uint32_t pba_hz)
{
  // save input parameters to local driver data
  tpa6130_output_param.num_channels = num_channels;
  tpa6130_output_param.callback     = callback;
  tpa6130_output_param.callback_opt = callback_opt;

  /* Probe for amplifier and initialize it */
  tpa6130_init();

#if defined(TPA6130_DAC_CLOCK_SET_CALLBACK)
  TPA6130_DAC_CLOCK_SET_CALLBACK(sample_rate_hz);
#else
  /* ABDAC configuration
   * The ABDAC needs the input frequency of its generic clock (bus_hz)
   * Here we use the configuration value from the conf_tpa6130.h file
   * (TPA6130_ABDAC_GCLK_INPUT_HZ).
   * The sample rate specifies the desired sample rate for the ABDAC.
   * The generic clock input must be greater than 256*sample_rate_hz
   * or the setup of the ABDAC will fail silently here.
   * TODO we could add asserts here to detect wrong settings during
   * compile time.
   */
  if(!abdac_set_dac_sample_rate(sample_rate_hz)) {
    // if it is not possible to set correctly the sample rate
    // Use default set function
    abdac_set_dac_hz(TPA6130_ABDAC, TPA6130_ABDAC_GCLK_INPUT_HZ,sample_rate_hz);
  }
#endif

  if(swap_channels)
  {
    abdac_swap_channels(TPA6130_ABDAC);
  }
  abdac_enable(TPA6130_ABDAC);

  /* PDCA setup */
  /*FIXME we use only word as transfer size for now.
   * half-word transfer size will only write to channel0
   * of the ABDAC, this can be used to implement mono */
  pdca_channel_options_t tpa6130_abdac_pdca_options =
  {
    .addr   = NULL,
    .size   = 0,
    .r_addr   = 0,
    .r_size   = 0,
    .pid    = TPA6130_ABDAC_PDCA_PID,
    .transfer_size  = PDCA_TRANSFER_SIZE_WORD
  };

  /* Initialize the PCDA for the ABDAC
   * The channel number can be set in the configuration file
   * with the define TPA6130_ABDAC_PDCA_CHANNEL.
   */
  pdca_init_channel(TPA6130_ABDAC_PDCA_CHANNEL,
    &tpa6130_abdac_pdca_options);
  /* Enable the PDCA channel. Since we did not provide any data
   * yet the channel is in idle mode */
  pdca_enable(TPA6130_ABDAC_PDCA_CHANNEL);

}

/*! \brief Outputs a sample buffer to the DACs.
 * The input requires a sample buffer that consists of words (32-bit)
 * which contain two (16-bit) samples, one for each channel.
 *
 * \note The DACs must have been started beforehand.
 */
bool tpa6130_dac_output(void *sample_buffer, size_t sample_length)
{
  //int global_interrupt_enabled;

  /*Wait until the PDCA loads the reload value to its transfer
   * counter register(TCRR=0). Then we are ready to set up a new
   * transfer */
  if(!(pdca_get_transfer_status(TPA6130_ABDAC_PDCA_CHANNEL) &
    PDCA_TRANSFER_COUNTER_RELOAD_IS_ZERO))
  {
    return false;
  }

  /* Nothing to do if we get no data. */
  if(sample_length)
  {

    /*TODO Do we really need to adjust the buffer for mono*/

    /* While reloading the PDC we do not need any active interrupt*/
    //if((global_interrupt_enabled = cpu_irq_is_enabled()))
    //  cpu_irq_disable();

    /*FIXME This assumes a stereo 16-bit sample size */
    // one sample here consists of 2x16-bit (16-bit stereo)
    pdca_reload_channel(TPA6130_ABDAC_PDCA_CHANNEL,
      sample_buffer, sample_length);

    //if(global_interrupt_enabled)
    //  cpu_irq_enable();

    /*TODO enable transfer complete interrupt
     * Is it possible to move this to setup or other places?*/
    if(tpa6130_output_param.callback_opt & AUDIO_DAC_OUT_OF_SAMPLE_CB)
      pdca_enable_interrupt_transfer_complete(TPA6130_ABDAC_PDCA_CHANNEL);
    if (tpa6130_output_param.callback_opt & AUDIO_DAC_RELOAD_CB)
      pdca_enable_interrupt_reload_counter_zero(TPA6130_ABDAC_PDCA_CHANNEL);
  }
  return true;
}

bool tpa6130_dac_is_volume_muted(void)
{
  return false;
}

void tpa6130_dac_mute(bool mute)
{
  // //1st Version Mute Audio for Play/Pause
/*  int volume=tpa6130_get_volume();
  if(mute==true) {
     //Mute volume
     volume= volume|MUTE_L|MUTE_R;
  }
  else {
     //Unmute volume
     volume= volume&(~(MUTE_L|MUTE_R));

  }
 tpa6130_write_data(TPA6130_VOLUME_AND_MUTE,volume);
*/
  //2n Version Stop PDCA >> No lost of audio when pause
/*  if(mute==true) {
    pdca_disable(TPA6130_ABDAC_PDCA_CHANNEL);
  }
  else {
    pdca_enable(TPA6130_ABDAC_PDCA_CHANNEL);
  }
*/

  // 3rd Version wait until the current buffers are empty and disable the interrupts
  int8_t volume = tpa6130_get_volume();
  if (mute)
  {
    uint32_t save_dac_reload_callback_opt;

    // Mute the audio stream
    volume = volume | MUTE_L | MUTE_R;
    tpa6130_write_data(TPA6130_VOLUME_AND_MUTE, volume);
    // Disable the reload channel of the interrupt
    save_dac_reload_callback_opt = tpa6130_output_param.callback_opt;
    tpa6130_output_param.callback_opt = 0;
    // Disable the reload interruption and wait until the transfer is complete
    pdca_disable_interrupt_reload_counter_zero(TPA6130_ABDAC_PDCA_CHANNEL);
    while (!(pdca_get_transfer_status(TPA6130_ABDAC_PDCA_CHANNEL) & PDCA_TRANSFER_COMPLETE));
    // Restore the reload callback function
    tpa6130_output_param.callback_opt = save_dac_reload_callback_opt;
  }
  else
  {
    // Re-enable the interrupts
    pdca_enable_interrupt_reload_counter_zero(TPA6130_ABDAC_PDCA_CHANNEL);
    // Un-mute the audio stream
    volume = volume & (~(MUTE_L | MUTE_R));
    tpa6130_write_data(TPA6130_VOLUME_AND_MUTE, volume);
  }
}
コード例 #10
0
ファイル: ms3.c プロジェクト: avrxml/asf
void ms3_dac_setup(U32 sample_rate_hz, U8 num_channels, U8 bits_per_sample,
		bool swap_channels, void (*callback)(U32 arg), U32 callback_opt,
		U32 pba_hz)
{
	//configure clock
	if (sample_rate_hz < (8000 + 8021) / 2) {
		usb_stream_resync_frequency = 4096000;
		cs2200_freq_clk_out(_32_BITS_RATIO(usb_stream_resync_frequency, CS2200_FREF));
	}
	else if (sample_rate_hz < (8021 + 22050) / 2) {
		usb_stream_resync_frequency = 4106752;
		cs2200_freq_clk_out(_32_BITS_RATIO(usb_stream_resync_frequency, CS2200_FREF));
	}
	else if (sample_rate_hz < (22050 + 32000) / 2) {
		usb_stream_resync_frequency = 11289600;
		cs2200_freq_clk_out(_32_BITS_RATIO(usb_stream_resync_frequency, CS2200_FREF));
	}
	else if (sample_rate_hz < (32000 + 44100) / 2) {
		usb_stream_resync_frequency = 16384000;
		cs2200_freq_clk_out(_32_BITS_RATIO(usb_stream_resync_frequency, CS2200_FREF));
	}
	else if (sample_rate_hz < (44100 + 48000) / 2) {
		usb_stream_resync_frequency = 22579200;
		cs2200_freq_clk_out(_32_BITS_RATIO(usb_stream_resync_frequency, CS2200_FREF));
	}
	else if (sample_rate_hz < (48000 + 88200) / 2) {
		usb_stream_resync_frequency = 24576000;
		cs2200_freq_clk_out(_32_BITS_RATIO(usb_stream_resync_frequency, CS2200_FREF));
	}

	//configure ssc to use clock on TX_CLOCK pin
	AVR32_SSC.tcmr = (0 << AVR32_SSC_TCMR_CKO_OFFSET)
			| (1 << AVR32_SSC_TCMR_STTDLY_OFFSET)
			| (2 << AVR32_SSC_TCMR_CKS_OFFSET)
			| (7 << AVR32_SSC_TCMR_START_OFFSET)
			| (0x1f << AVR32_SSC_TCMR_PERIOD_OFFSET);

	AVR32_SSC.tfmr = (0xf << AVR32_SSC_TFMR_DATLEN_OFFSET)
			| (1 << AVR32_SSC_TFMR_MSBF_OFFSET)
			| (1 << AVR32_SSC_TFMR_FSOS_OFFSET)
			| (1 << AVR32_SSC_TFMR_FSLENHI_OFFSET)
			| (0xf << AVR32_SSC_TFMR_FSLEN_OFFSET);

	AVR32_SSC.cr = AVR32_SSC_CR_TXEN_MASK;

	//configure DMA
	pdca_channel_options_t ms3_ssc_pdca_options = {
		.addr = NULL,
		.size = 0,
		.r_addr = NULL,
		.r_size = 0,
		.pid = AVR32_PDCA_PID_SSC_TX,
		.transfer_size = (bits_per_sample <=  8)?PDCA_TRANSFER_SIZE_BYTE:
				(bits_per_sample <= 16)?PDCA_TRANSFER_SIZE_HALF_WORD:
				PDCA_TRANSFER_SIZE_WORD
	};
	pdca_init_channel(MS3_SSC_TX_PDCA_CHANNEL, &ms3_ssc_pdca_options);
	pdca_enable(MS3_SSC_TX_PDCA_CHANNEL);

	//configure audio parameters
	ms3_output_params.num_channels = num_channels;
	ms3_output_params.callback = callback;
	ms3_output_params.callback_opt = callback_opt;
}

bool ms3_dac_output(void *sample_buffer, size_t sample_length)
{
	bool global_interrupt_enabled;

	if (!(pdca_get_transfer_status(MS3_SSC_TX_PDCA_CHANNEL) &
			PDCA_TRANSFER_COUNTER_RELOAD_IS_ZERO))
		return false;

	if (sample_length) {
		if (ms3_output_params.num_channels == 1) {
			S16 *s16_sample_buffer = sample_buffer;
			int i;

			for (i = sample_length - 1; i >= 0; i--) {
				s16_sample_buffer[2 * i + 1] =
						s16_sample_buffer[2 * i] =
						s16_sample_buffer[i];
			}
		}

		// The PDCA is not able to synchronize its start of transfer with the SSC
		// start of period, so this has to be done by polling the TF pin.
		// Not doing so may result in channels being swapped randomly.
		if ((global_interrupt_enabled = Is_global_interrupt_enabled()))
			Disable_global_interrupt();

		if (pdca_get_transfer_status(MS3_SSC_TX_PDCA_CHANNEL) &
				PDCA_TRANSFER_COMPLETE) {
			while (gpio_get_pin_value(MS3_SSC_TX_FRAME_SYNC_PIN));
			while (!gpio_get_pin_value(MS3_SSC_TX_FRAME_SYNC_PIN));
		}

		pdca_reload_channel(MS3_SSC_TX_PDCA_CHANNEL, sample_buffer, sample_length * 2);
		pdca_get_reload_size(MS3_SSC_TX_PDCA_CHANNEL);
		if (global_interrupt_enabled)
			Enable_global_interrupt();

		if (ms3_output_params.callback_opt & AUDIO_DAC_OUT_OF_SAMPLE_CB)
			pdca_enable_interrupt_transfer_complete(MS3_SSC_TX_PDCA_CHANNEL);
		if (ms3_output_params.callback_opt & AUDIO_DAC_RELOAD_CB)
			pdca_enable_interrupt_reload_counter_zero(MS3_SSC_TX_PDCA_CHANNEL);
	}

	return true;
}