SC589 mini ADAU1761 initialization / sampling set / PLL

Update:

After looking over the SAM audio start adau1761 code, i found that there should be a 0-10 second wait for the PLL to lock.  Other sources on the web for Linux implementations recommended the same.  With some additional changes, I was able to get the default SHARC core 1 loopback audio processing working.  The volume is very low, so turn up your input source and the amplifier receiving the output from the SC589 / ADAU1761.  I have yet to mess with the gain settings.

I modified the while loop as follows:

uint16_t timeout = 10000;

log_event(EVENT_INFO, "Attempting PLL lock");

while (timeout-- && ((value & 0x2) == 0)) {

    twi_read_block(&adau1761->twi, result, 6);
    value = result[5];
}

if (timeout == 0) {
    log_event(EVENT_FATAL, "PLL would not lock before 10 sec timeout");
} else {
   log_event(EVENT_INFO, "PLL lock success!");
}

Additionally, I took the exported SigmaAudio parameter files from the SAM audio starter project and replaced those at these paths:

drivers/bm_adau_driver/configurations/ss_schematics/ADAU1761_ADAU1761_8ch_i2s_master/Exported_init_files/NumBytes_ADAU1761.dat

drivers/bm_adau_driver/configurations/ss_schematics/ADAU1761_ADAU1761_8ch_i2s_master/Exported_init_files/TxBuffer_ADAU1761.dat

A verification block of code was an issue, so I commented that out (this is just after the sampling setting / PLL lock).

// Confirm that the ADAU1761 is running (a good indication that it has been initialized properly)
//uint8_t sigmadspRunning;
//adau_read_ctrl_reg(&adau1761_local, ADAU1761_REG_DSP_RUN, &sigmadspRunning);
//if (!(sigmadspRunning & 0x1)) {
// log_event(EVENT_FATAL, " The SigmaDSP core inside the ADAU1761 is not running");
//}

Also, if you are not using the Audio Project Fin, comment out the gain settings:

// Set the AUX channel gain depending on whether a Audio Project Fin is present, and which version of the board
//uint8_t currentRegVal;
// Left channel AUX gain
//adau_read_ctrl_reg(&adau1761_local, ADAU1761_REG_REC_MIXER_LEFT_1, &currentRegVal);
//currentRegVal &= 0xF8;
//currentRegVal |= auxGain;
//adau_write_ctrl_reg(&adau1761_local, ADAU1761_REG_REC_MIXER_LEFT_1, currentRegVal);

// Right channel AUX gain
//adau_read_ctrl_reg(&adau1761_local, ADAU1761_REG_REC_MIXER_RIGHT_1, &currentRegVal);
//currentRegVal &= 0xF8;
//currentRegVal |= auxGain;
//adau_write_ctrl_reg(&adau1761_local, ADAU1761_REG_REC_MIXER_RIGHT_1, currentRegVal);

Update (9/10):

The DMA's in the SHARC cores are not enabling.  So, the audio observation was not observed from within audio processing blocks.  This is likely just the codec's own loopback.

Still working to have the audio processing part work. 

Update (9/10) - 2:

DMAs enable with a while / timeout, but audio is still not being processed as expected.  The audio buffers in the processing block do not appear to be connected to anything, where the ADUA1761 is concerned.  I may raise a new question for this specific issue.  Or, it may be that I need to wait on a new framework revision.

Update:

After looking over the SAM audio start adau1761 code, i found that there should be a 0-10 second wait for the PLL to lock.  Other sources on the web for Linux implementations recommended the same.  With some additional changes, I was able to get the default SHARC core 1 loopback audio processing working.  The volume is very low, so turn up your input source and the amplifier receiving the output from the SC589 / ADAU1761.  I have yet to mess with the gain settings.

I modified the while loop as follows:

uint16_t timeout = 10000;

log_event(EVENT_INFO, "Attempting PLL lock");

while (timeout-- && ((value & 0x2) == 0)) {

    twi_read_block(&adau1761->twi, result, 6);
    value = result[5];
}

if (timeout == 0) {
    log_event(EVENT_FATAL, "PLL would not lock before 10 sec timeout");
} else {
   log_event(EVENT_INFO, "PLL lock success!");
}

Additionally, I took the exported SigmaAudio parameter files from the SAM audio starter project and replaced those at these paths:

drivers/bm_adau_driver/configurations/ss_schematics/ADAU1761_ADAU1761_8ch_i2s_master/Exported_init_files/NumBytes_ADAU1761.dat

drivers/bm_adau_driver/configurations/ss_schematics/ADAU1761_ADAU1761_8ch_i2s_master/Exported_init_files/TxBuffer_ADAU1761.dat

A verification block of code was an issue, so I commented that out (this is just after the sampling setting / PLL lock).

// Confirm that the ADAU1761 is running (a good indication that it has been initialized properly)
//uint8_t sigmadspRunning;
//adau_read_ctrl_reg(&adau1761_local, ADAU1761_REG_DSP_RUN, &sigmadspRunning);
//if (!(sigmadspRunning & 0x1)) {
// log_event(EVENT_FATAL, " The SigmaDSP core inside the ADAU1761 is not running");
//}

Also, if you are not using the Audio Project Fin, comment out the gain settings:

// Set the AUX channel gain depending on whether a Audio Project Fin is present, and which version of the board
//uint8_t currentRegVal;
// Left channel AUX gain
//adau_read_ctrl_reg(&adau1761_local, ADAU1761_REG_REC_MIXER_LEFT_1, &currentRegVal);
//currentRegVal &= 0xF8;
//currentRegVal |= auxGain;
//adau_write_ctrl_reg(&adau1761_local, ADAU1761_REG_REC_MIXER_LEFT_1, currentRegVal);

// Right channel AUX gain
//adau_read_ctrl_reg(&adau1761_local, ADAU1761_REG_REC_MIXER_RIGHT_1, &currentRegVal);
//currentRegVal &= 0xF8;
//currentRegVal |= auxGain;
//adau_write_ctrl_reg(&adau1761_local, ADAU1761_REG_REC_MIXER_RIGHT_1, currentRegVal);

Update (9/10):

The DMA's in the SHARC cores are not enabling.  So, the audio observation was not observed from within audio processing blocks.  This is likely just the codec's own loopback.

Still working to have the audio processing part work. 

Update (9/10) - 2:

DMAs enable with a while / timeout, but audio is still not being processed as expected.  The audio buffers in the processing block do not appear to be connected to anything, where the ADUA1761 is concerned.  I may raise a new question for this specific issue.  Or, it may be that I need to wait on a new framework revision.