Modifying Denon PMA-50 with a digital S/PDIF loopback

[mk2]

I chose my Denon PMA-50 some years ago. I'm a digital audio person, so I liked that it drives the speaker voltage directly from PCM audio ("DDFA" design, like a high voltage DAC, avoiding low voltage analogue audio entirely.) It also got a decent review here.

I connect a Behringer DEQ2496 processor to it (S/PDIF coax or optical). Sounds great and does its task of room correction, but it's not a very practical device for home hifi; it is fiddly selecting sources, and doesn't have the range of features (bluetooth, USB) of the Denon.

I mused that it might be possible to loop PCM audio out of the Denon, via the processor, and back into the Denon. All the usabilty of the Denon but with DSP, for a very elegant home hifi solution.

Has anyone tried anything like this?

A Denon service manual is available. The block diagram on page 43 shows:

* a PCM9211 chip (source selection, and ADC for analogue sources); into
* EPCS4SI8N FPGA processes the PCM audio signal (EQ perhaps?); into
* CSRA6601 digital modulator (drives the analogue speakers etc.)

The data sheet for the PCM9211 states that inputs and outputs are S/PDIF. That implies to me that intermediate PCM audio at each step inside the Denon is S/PDIF (not some bare PCM format)

I couldn't quite believe that it may be within reach; by breaking a track and soldering onto some choice connections, perhaps. But I may be vastly underestimating the task at hand?

PCM could be looped through the Behringer and back in for the modulator, or just out to the processor and onward to a separate (more powerful) digital amplifier.

 

[AnalogSteph]

I couldn't quite believe that it may be within reach; by breaking a track and soldering onto some choice connections, perhaps. But I may be vastly underestimating the task at hand?

Not vastly, just a bit. But as long as you can also find a pair of Toslink transmitter and receiver, a 3.3 V supply to power them inside the device and ideally a 3-pin jumper block to select between straight through and external loop (alongside maybe some small-fry passives for voltage decoupling and whatnot), this looks like a promising project.

I found some promo materials on the Zetex ZXZCM800 - before it became the CSR6601 - including this info:

Digital audio inputs:
I²S and TDM interfaces support up to 32 bit resolution
and sample rates from 32kHz to 192kHz.

That's the kind of stuff accepted by audio codecs.

So it looks like you'll only be able to do a processor loop with raw, unprocessed audio as it comes out of the PCM9211, and I woudn't bet on all sources being available (e.g. USB audio input).

 

[mk2]

I'm pleased to hear generally positive noises, though I appreciate the mountain to climb here!

Sorry if my terminology is wrong, I assume Toslink is always optical. But wasn't considering that because I assumed coax would be easier (though I imagine it equally needs some op-amp.)

If I'm understanding what you're saying it's that the input to the CSR6601 may not be S/PDIF but some other PCM format.

I've been studying page 51 of the service manual this evening, but I'm still a bit lost on how the FPGA fits in. If I take the block diagram (page 43) as gospel then it implies this FPGA has no other control or inputs; so that means it can't be implement the volume control or EQ.

How likely the FPGA exists solely to convert the format S/PDIF to I2S to connect these two other chips? I don't yet know the relationship of these two formats or how they differ.

I woudn't bet on all sources being available (e.g. USB audio input)

Interesting, why do you say this? The block diagram suggests the PMC9211 is positioned at the end of all the input source (USB, DSD etc.) processing and unify to one PCM stream. That's ideal if it's the case, as long as the sample rate is not too high for the Behringer. I need to look for the volume control (when I'm less tired!)