Simple audio in-out code for the ADSP-BF706 EZ-KIT

I have an audio in-out program in C for the ADSP-BF706 EZ-KIT mini, about 80 lines in length. It's simple and very easy to understand. It's also completely self-contained - it doesn't use any of the header files that the "TalkThrough_BF706Mini.c" program uses (supplied with the kit). It includes a basic TWI driver, SPORT0 set up and configuration routine for the on board codec (ADAU1761). With some simple modifications it can be used for filtering, both FIR and IIR. The file is attached. All welcome to use free.


  • Hi Patrick,

    Thanks for the assembly code for clock configuration!

    I think delay loops after TWI transmit are not save enough.
    This is my proposal for ADAU1761 configuration:

    void write_CODEC_reg(int8_t codec_address, int16_t reg_address, uint8_t reg_data)
        *pREG_TWI0_CTL = 0x8c;                  // Set prescale and enable TWI
        *pREG_TWI0_CLKDIV = 0x3232;             // Set duty cycle
        *pREG_TWI0_MSTRADDR = codec_address;    // Address of CODEC
        *pREG_TWI0_TXDATA8 = reg_address >> 8;  // Address of register to set, MSB
        *pREG_TWI0_MSTRCTL = 0xc1;              // Send three bytes and enable transmit
        while((*pREG_TWI0_MSTRCTL >> 6) != 2);  // Wait until first byte is transmitted
        *pREG_TWI0_TXDATA8 = reg_address;       // Address of register to set, LSB
        while((*pREG_TWI0_MSTRCTL >> 6) != 1);  // Wait until second byte is transmitted
        *pREG_TWI0_TXDATA8 = reg_data;          // Data to write
        while (!(*pREG_TWI0_ISTAT & BITM_TWI_ISTAT_MCOMP));

    void configure_CODEC()
        write_CODEC_reg(0x38, 0x4000, 0x01);    // Enable master clock, disable PLL
        write_CODEC_reg(0x38, 0x400a, 0x0b);    // Set left line-in gain to 0 dB
        write_CODEC_reg(0x38, 0x400c, 0x0b);    // Set right line-in gain to 0 dB
        write_CODEC_reg(0x38, 0x4015, 0x01);    // Set serial port master mode
        write_CODEC_reg(0x38, 0x4017, 0x00);    // Set CODEC default sample rate, 48 kHz
        write_CODEC_reg(0x38, 0x4019, 0x63);    // Set ADC to on, both channels
        write_CODEC_reg(0x38, 0x401c, 0x21);    // Enable left channel mixer
        write_CODEC_reg(0x38, 0x401e, 0x41);    // Enable right channel mixer
        write_CODEC_reg(0x38, 0x4023, 0xe7);    // Set left headphone volume to 0 dB
        write_CODEC_reg(0x38, 0x4024, 0xe7);    // Set right headphone volume to 0 dB
        write_CODEC_reg(0x38, 0x4029, 0x03);    // Turn on power, both channels
        write_CODEC_reg(0x38, 0x402a, 0x03);    // Set both DACs on
        write_CODEC_reg(0x38, 0x40f2, 0x01);    // DAC gets L, R input from serial port
        write_CODEC_reg(0x38, 0x40f3, 0x01);    // ADC sends L, R input to serial port
        write_CODEC_reg(0x38, 0x40f9, 0x7f);    // Enable all clocks
        write_CODEC_reg(0x38, 0x40fa, 0x03);    // Enable all clocks

  • Ah yes. I have only tried release mode in assembly - no compiler optimisation.

  • These are the fully tested versions of the sample-by-sample
    and block-processing examples for the BF706 EZ-KIT Mini.
    In the ADC control register, Address 0x4019, Bit 6 (ADCPOL)
    is active to avoid negative (inverted) polarity of the input signal.
    The block-processing version can be used for FFT based
    algorithms like overlap add and overlap save.
    The block size can be changed in config.h
    The software overhead is < 5%. In release mode more the
    95% of the processor cycles can be used for audio processing.
  • I have added an UART DMA control application to facilitate @PatrickG and @UweS excellent audio examples...The discussion is here : 

    BF706 UART DMA Terminal for Audio and FIR filter Control 

  • Hello Patrick,
    the DMA block in-out is issue is finally solved (I hope). After several months of frustration
    Mike Smith BF706 2D DMA Audio Loopback  proposed a solution:
    2D-DMA for double-buffering. According to my tests this works fine for very large blocks or
    even sample by sample. One DMA-interrupt per block is generated and the audio processing
    can be done in the RX_interrupt_handler.

    I would not have solved this without your posts!
    Best regards