Direct Stream Phono Preamp with an FPGA?

Can we have a Direct Stream Phono Preamp with an FPGA??? that would be supercool. Is there an analog equivalent for FPGA? or maybe convert the phono signal to digital, and apply RIAA digitally? don’t know… I just like to have a top end evolving phono preamp, just the the DSD.

You don’t want to do that. The best sounding phono stage is not going to have any digital conversion or processing, and no amount of software updates is going to compensate for the loss in quality.


The alternative view is that using great digital like the discontinued NuWave Phono Converter with a native DSD feed via I2S to the DS DAC is an excellent way to get the best out of your vinyl collection without all the downsides of that much analog gain and distortion.

Also you can record that DSD stream and play back with perfect consistency at any time in the future.

It doesn’t need to be “processed”. Just reproduced with care and precision.


If you record DSD direct with analog RIAA, then sure. It WILL change the sound a little, but probably still sound good. But for digital EQ you need PCM, and the sound will suffer. There is no reason for any this with real time phono playback, only for digital archiving.

There’s no reason why analog EQ would be any less destructive than a sufficiently high resolution PCM EQ. The same mathematical transform can be expressed numerically as is implemented in electronic component form. It’s just a question of precision. Most such processing is done in at least 32 bit resolution, many times it’s 64 bit before dithering back to the practically sufficient 24 bit form. The DS DAC can play back PCM with equal aplomb, using the exact same logic and hardware it uses for DSD input.

I think the biggest concern is the quality of the digital low-pass filters used to generate the initial PCM data during A-D conversion. Use good filters there, maintain the PCM in reasonably high resolution, feed to DS DAC, job done.

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I’ve sat in mastering sessions with some of the absolute best pro AD/DA converters in the world (Merging, Pacific Microsonics, Lavry, Prism, Burl), and used the most high quality digital equalizers. There is literally no PCM conversion chain, or digital mastering chain that is lossless, not even played via the DS DAC. A friend recently had a project transferred from 1/2" tape to an $80,000+ MSB analogue to digital converter, and played back on my own DSS it still doesn’t sound like the tape playback head. DSD is much closer to the original analogue, but you have to convert to PCM to process it, and even at 352khz in the Pyramix DAW, this is destructive.

Analogue EQ usually has a phase shift, which isn’t ideal, but in my world they almost always sound better than their digital counterparts.

Anyway, far off topic…I have a strong opinion about this because of personal experience. Carry on…


Might be on topic to wonder whether the work that Ted’s doing has a chance of closing that apparent gap.

Thanks for your comments – your real world experience in that context certainly outweighs mine and it’s good to get a reality check like that. Do you have any theories as to why you hear what you hear, given that the mathematics behind electrical circuits (eg RIAA EQ) are very well understood and in principle apply equally to a digital context?

You are in line with what I read from leading mastering engineers. The difference between analog and digital compression seems even bigger than between analog and digital EQ.

I think it’s just really hard to implement this stuff perfectly in practice, and there are no perfect PCM FIR filters or linearity. Even at 24 bit 352khz, PCM conversion isn’t on par with a direct analogue source or with DSD128 and higher.

Most digital equalizers will leave some subtle artifacts, although the best are probably more transparent than the digital conversion itself. I don’t totally understand why. It could be DSP limitations. There are some really good hardware digital equalizers made by Weiss, and they are 3 rack spaces high and go for $9000 a pop (way overkill for RIAA).

The bit about digital compression is true. Most mastering guys I know or have worked with do the basic EQ and compression with analogue hardware (often custom) in the main analogue pass/capture, and after that use digital tools to fix specific ornery problems and for peak limiting. With an ideal recording where no digital fixes or peak limiting is needed (very rare in these loud days), there may be no digital processing at all.

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Independent of perceptions of quality, most vinyl aficionados want a purely analog signal path; digital is philosophically anathema.


I have heard experts whose ears I trust swear by Pure Vinyl software, which does RIAA digitally. They say digital RIAA trumps anything with capacitors and other components in the circuit. As I haven’t listened myself, I can only go by what they say. BUT, to me it makes sense.

No, if the digital compensation is via software, you do not use PCM, at least in Pure Vinyl, unless things have changed in the last three years. I did extensive research on the Pure Vinyl, but the costs needed to implement it at the time was too great, so I haven’t kept up on it.

I agree that digital processing at 352.8k doesn’t cut it. If I were using an FPGA to do a digital RIAA filtering (or any other filtering) it would be at a much higher sampling rate than 352.8k. Analog RIAA’s will mess with the frequency and phase response more than a good digital filter because there are no perfect analog components and those non-perfections can’t help but add ripples to the frequency response, etc. But most of the time the digital processing is done at too low of a sample rate. IMO you need at least the sample rate of double rate DSD: 5.6448MHz and higher than 64 bits of precision.

Don’t get me wrong, I wouldn’t want to do any digital near the preamp proper, and I don’t know how to make a cheap enough A/D for the kind of product we are talking about so I don’t see it happening.

Perhaps I misunderstand you but their web site clearly states that they use 24/192k (which is PCM) also they mention doing 64 bit IIR filters which is PCM processing, and they talk about their dithering which is also necessary at some point in PCM processing.

As I stated, I haven’t followed them in 2-3 years. Back then they were stating, or at least strongly implying, that they were not using PCM, as were reviewers. In the interim, maybe their process has changed (or their informational transparency ??). It isn’t the first, and i’m sure won’t be the last time that what I have read might be misleading, or even blatantly a lie. So, I manfully admit I was mistaken, and retract what I have stated. And I am sure that down the road I might have to do it again. Side note: I just had to return to Amazon an SACD that wasn’t an SACS, even though they said it was. They thought, apparently, that if it was DSD remastered, that that meant is was an SACD. We are at the mercy of those who write online descriptions, I guess.

You aren’t the only one who had that impression. I had to look because I didn’t think you could do the required IIR filtering without at least an implicit conversion to PCM. IMO if you are doing anything in the digital domain doing RIAA processing in the digital domain is the way to go.

Looks like there real potential from the discussions, at first I thought it’s just me getting over excited about the new DSD release and wanted something like that for my analog chain… :slight_smile: