Someone kindly pointed out the eeschema library file was missing from the git repository. I did notice a few weeks back and had committed the file. I obviously did not push the update to the master branch though!. 

Which brings me to board revisions 1.1/2,0. The ADC board revision 1.1 is almost ready to release with correct power pin assignments, I am also working on a newer revisions of the DSP boards with a master SPI connection. There is a new revision of the DAC board with the mating slave SPI connector. This means the ADAU1452 should control the ADAU1966 for faster sample rates without the need for a microcontroller.

I am also revising the PSU and buffer board. This is mostly component spacing and silk screen tweaks. On all boards I have decided to switch to some lower cost regulators in digital supplies. The TPS7A45xx series are quite expensive but I will still use these in the analogue supplies due to their low noise across the audio band.. 

I have also been playing with RightMark Audio Analyser. There seems to be a problem with Intermodulation distortion+noise readings using the SPDIF input on the DSP. THD at lower frequencies is high on the SPDIF input as well. Otherwise the measurements seem to fall in the noise floor of my Xonar U7 soundcard. Some of the noise issues are from low level 50Hz and multiples (a lot actually, all the way up to 500Hz). This is a very crude test so take all results with a pinch of salt, but it is interesting anyway.

Results shown below. First result is the Xonar U7 analogue output connected to the Xonar U7 line input. Second is connecting the Xonar U7 analogue output to the DSP platform ADC analogue input, and the DAC analogue output to the Xonar U7 analogue input. Third is the Xonar U7 SPDIF output connected to the DSP board directly, and the DAC analogue output to the Xonar U7 analogue input

Finally soldered an EEPROM (well a socket actually) on one of the DSP boards and it self boots just fine. 

Here is a bad phone picture showing the DAC buffer board (single ended outputs) and the PCM4202 ADC board wired into my test chassis. Boards are stacked so the DSP and DAC are underneath.

So, the reason the PCM4202 sounded so different to the digital input is because.... I have wired left to right and right to left (or LRCLK may be inverted). Also I think the phono cable I was using was faulty as well.

Having replaced the cable, swapped the channels in SigmaStudio (which is nice!), and set the levels between the two to almost the same, the analogue and digital inputs sound pretty much similar to me. If anything I would say the digital input sounds a little more clinical. I guess that is partly down to the sound card I am using to generate the analogue audio from my laptop in the first place (a Xonar U7).

So after a little bodge on the power pins the PCM4202 lives. Using the ADAU1452 with sigma studio is nice, also my soundcard can toggle between SPDIF and Analogue out on the fly. This means I can compare direct digital into the DSP with the analogue sound into the PCM4202 and back out again through the ADAU1966. Whilst this seems pretty "pointless" it is quite useful if the intended use is as an active crossovers after existing audio equipment.

I didn't expect there to be such a big difference in sound, I'm not sure how much of an effect my sound card is having but this isn't anything subtle at all. Whatever is causing the difference both options sound fine and I could listen to either of them all day. . 

Well it would if I didn't use my lab supply and bring the rails up slowly!

The DVDD and DGND pins on the schematic are reversed, what makes it an even more annoying is that the library part is correct so no excuses really!

I've "bodged" a board which powered up fine now with 2 lifted pins/wires. Just need to test it with audio tonight. Given these boards cost $25 I will junk them and order some fixed ones.

In other news I am crunching through building/testing other boards (all release 1.0). Initially building two boxes. One will be a fixed analogue single ended inputs/outputs to play with as a crossover box. The other will be my experimentation box to try anything I can with. The DSP board with an ADAU1450 chip talks fine to SigmaStudio as well. Two out of the 3 LFCSP chips I hand soldered are alive, just one board left to test but that is missing some parts which I have ordered.

I am also working through schematic/PCB designs to release version 1.1. of all current designs with tweaks to layouts (part spacing mostly), errors fixed, and better silkscreen labeling of jumpers/power connectors. The KiCad library will be updated when these are released as I have fixed errors in PCB footprints. 

I am also thinking about making this site a little more friendly if I can. Also need to finish content on the hardware page with a little info on each board (think pictures with captions and so on).

So I have been digging into clocking on the ADAU1452 and the ADAU1966. I have previously played with some AKM DACs and generally if the ratios of MCLK/BCLK where correct it didn't matter what sample rate you fed the chip it would work just fine. This does not seem to be the case with the ADAU1966.

In stand alone mode I can only get the ADAU1966 to work at 48kHz, and if you look at the register settings this seems to be it's default mode. This does not seem to be explained anywhere clearly in the datasheet. I have tried running at 96kHz, with all the clocks correct, and audio comes out, along with lots of extra noises. To be honest it sounds good to me at 48kHz. which is good for running in TDM16 mode that is limited to 44.1/48kHz. 

If I had included a master SPI port on the DSP board then I should be able to control the DAC chip through SPI. I wasn't sure how this would work with self-booting though and was trying to keep it simple. 

So I hooked up my Gould oscilloscope (50MHz analogue bandwidth, not calibrated) and checked a few things out, the circuit looks fine from what I can see.

The picture below could be any random square wave from anything, but it is actually the BCLK signal probed right on the DAC input pin (right on the package). It looks pretty clean. I can not work out why it seems to be running half the speed it should. It seems to be 6.144MHz, I was expecting 12.288MHz. That might explain why the DAC sounded horrid running at what I thought was 48kHz. Half that is obviously 24kHz which isn't ideal for audio! I think it is a software thing (SigmaStudio) as the hardware looks and tests just fine, or I could be calculating it incorrectly..