ADAU1787 EVB I2S in & out non-reasonable

Hello masteraudio,

If you look at the data on a scope with the trigger triggered on the LRCLK you will see the data changing from frame to frame. In your case you will not see it changing much, just a few bits. What you are looking at is the DC offset that all ADCs will have. You do not show the LRCLK so it is difficult for me to see where the MSB is but I am guessing that in this case the offset is negative. It could be either and it really does not matter and does not suggest a problem. What I suggest you do is to turn on the High Pass Filter that is included in the ADC. You will find these registers here:

This screenshot was taken of your project and they are all off. Turn them on and you will see the data be all zeros so long as the noise floor coming in is low enough. You might want to read my post about using a scope to capture I2S signals.

(+) How to Take Meaningful Screenshots of I2S Audio Signals - Q&A - Audio - EngineerZone (analog.com)

Dave T

Hi Dave,

Here are more inforamtion.

I tied AIN0 & 1 to Ground, I2S out still have something.

If I mute ADC then I2S no data

Ares

HI Dave,

Could you help me to set the ADC2 &3 to SAI1 output in the file I offered before?

Thanks

Ares

Hello MasterAudio,

You misunderstood what I said. I guess I need to add more detail. Your test did confirm that the DC offset is not coming in from an external source. It certainly is possible to create a DC offset external to the ADC and sometimes that is done to get rid of idle tones. What I was talking about are the DC offsets that come from internal mismatches in the silicon that every ADC will have. So it is internally generated and you cannot get rid of it by shorting the input. I suppose one could try to match the offset exactly so it cancels out but that is not a practical solution and the DC offset would need to track temperature etc. The way to get rid of the DC offset is to turn on the high pass filter. The filter is set to a corner frequency that is well below 1Hz so it will not degrade the performance of the ADC.

When you muted the ADC it gets rid of it because the mute takes place in the digital section of the ADC.

Dave T

Hello MasterAudio,

There are so many options here. I will ask a few of them.

Does the serial port need to be a master or slave?

Do you need to use the ASRC?

What sampling rate do you need the ADC to use?

What sampling rate do you need for the actual serial port?

You want the ADC to directly go out to the serial port correct? You do not want to go through any of the DSP cores?

Dave T

Hi Dave,

Thanks for your help.

Does the serial port need to be a master or slave?

-> Master

Do you need to use the ASRC?

-> No

What sampling rate do you need the ADC to use?

-> 12.288MHz

What sampling rate do you need for the actual serial port?

-> 192kHz

You want the ADC to directly go out to the serial port correct?

-> Yes, I want I2S out

You do not want to go through any of the DSP cores?

-> Yes, because I use my own DSP

BR,

Ares

Hi MasterAudio,

There seems to be a bit of confusion on your desired ADC Sample Rate of 12.228MHz. The range of sample rates available for ADC are from 8kHz to 786kHz. Perhaps the 12.228MHz is in reference to the BCLK for a setup with SR - 192kHz, 32-bit data, and 2CH (I2S format)?

If that is the case, I've created an example project that accepts ADC data sampled at 192kHz, and routes to SPT0 with a SR - 192kHz which transmits I2S with ADAU1787 in Master Mode.

ADAU1787_ADC_to_SPT_192kHz.dspproj

Hopefully this example project helps clear up some of the issues you were having!

Thanks,

TM