Interfacing the AD1862 with AK4137 SCR

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I want to build an oldschool DAC using the AD1862, but of course because this IC is not I2S compatible, i need a glue logic. I also want to use an SRC because i don't want this to run in NOS mode. I choose the AK4137 IC as my SRC but while i was reading the datasheet i realized that i am unable to understand how this IC works.

What i have: I2S signal from a CM6631A, 44.1kHz - 192kHz, 16 and 24 bit data, with a 32 bit frame.

Can any of you help me to connect the DAC IC to the AK4137 with a sampling rate of 768kHz and 20 bits?
You might consider starting with a Chinese AK4137 board, of which there are a few (say, maybe 3) different styles. They are easy enough to reverse engineer if you have one in your hands to look at since the boards are only two layers.

The only thing to understand then would be the I2C register programming which is not too complicated.

Suggesting the above to help reduce the amount of help that might have to be provided by other members in this thread. If you can figure out most of the hardware, we should be able to help fill in the blanks.

Cheapest type to start with that shows pretty much everything might be one in the style of: AK4137 I2S/DSD Sample Rate Conversion Board Supports PCM/DSD Interconversion DOP | eBay the HF version from the drop down menu...
It wouldn't. Then again the AD1862 would not pass 24bit data. The point was if all you have is 16/44, then the DF1700 will do. For more, use the appropriate filter.

This is why i suggested the DF1706. It can accept data in an I2S format, anything from 44.1 till 192 and supports both 16 and 24 bit. But from what i understand i would need to modify the clocks on the fly for the DF1706 in order to work with audio ranging from 44.1 until 192.
Don't know where the clock modding idea came from but it is not correct. It works wihout any fuss.

Happy new year! :)

So you are saying that if i input a signal from 44.1kHz anywhere until 192kHz into the DF1706 and i have an external master clock coming from a low-jitter source it will be able to work at a custom system clock? The datasheet says that fs must be run at 128fS, 192fS, 256fS, 384fS, 512fS, or 768fS. If i have audio that is sampled at multiples of 44.1kHz i will need a crystal for that, but if i have audio that is sampled at 48kHz or multiples i will need another one to have the proper (128fS, 192fS, 256fS, 384fS, 512fS, or 768fS) ratios so that the DF1706 can calculate the appropriate fs from these 6 possible values.

Or am i not getting something right here?
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