PCM in the directstream DAC -- twice converted?

For strict PCM data (e.g., CD player SPDIF entering a Directstream DAC), is the digital signal converted TWICE to analog?

That is:
CD player SPDIF —> Directstream DAC

Inside the Directstream DAC:
PCM —> d-to-a —> a-to-d (i.e, analog to DSD) —> low-pass-filter (analog out)

Is this correct?

I’ll bet a case of Scotch for NO…!!!

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No it only happens once.

I believe there is an intermediate PCM step to 352.8 before going to 20x DSD. All digital until the transformer analog filter out.

Ted is probably recovering from Axpona but he’ll likely confirm shortly.

PCM is converted to 352.8 or 384 (depending) then (along with DSD) to 56.448MHz (20 times the SACD DSD rate), then to quad rate DSD (DSD256) No analog involved except for the final output.

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Slàinte mhath…

Ted -is this the case with the Junior also? Or in that case final one is to double rate (DSD128) instead?

So, PCM to PDM is purely a mathematical conversion. I.e., values from PCM mapped (transcoded) to PDM algorithmically? Correct?
If so: Since the transcoding is done “on the fly”, is DSP power intensive? I.e., do the FPGAs doing the transcoding get warm?

The upsampling and other digital processing is identical between the DS and the DS Jr. They both go to DSD256. Earlier releases (before Huron?) used to go to DSD128.

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The FPGA code is deterministic. However sigma delta modulation is chaotic, change one bit anywhere and you the rest of the output will probably differ a lot, but still any bit changing in the input will have the exact expected effect on the output.

The FPGA is overkill for the DS and DS Jr - it doesn’t get noticeably warm in any circumstances.

FWIW Don’t worry about power use in the FPGA or try to think about saving work, power, noise with different inputs: In the FPGA everything is always running, there’s no way to save work by doing something different before the DAC. E.g. for an if-then-else both arms are always running and the conditional picks which answer is paid attention to. The PCM upsampler is running even with DSD input, etc. There’s no savings with the volume at 100 instead of 99 or 101, they are all multiplies by very random seeming numbers.

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Thx for your response.
The query about pwr consumption of DSP/FPGA was based on reports (and reviews) of certain other DACs that are process-intensive. E.g., Chord and the old Meitner DAC (from 1993).
I’m not sure how process-intensive the Directstream DAC is; hence, my further note about “on-the-fly” conversion. The Meitner’s DSP section was quite warm. And many of the now-classic Philips SAA and TDA devices could consume up to 200mA (they ran WARM).
Of course, that was a while back!

In general to avoid extra noise you should run things as slowly as you can to get the job done. To get the best sound I spread things out as much as I can. Also I’m using very few of the I/O pins which is where you can use a serious amount of current.
The DS’s FPGA does some serious processing, but I code it at a pretty low level and it runs fairly efficiently.

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One of the reasons for the original query was a potential for application in so-called direct-digital amplification (i.e., PCM to PWM in an elaborate class-D topology). This has been done by other manufs (e.g., Sharp, TacT and NAD) for about two decades.
Stereophile reviewers have usually praised these amps–sometimes quite enthusiastically.
I was wondering whether Ted’s DS designs could be adapted for “direct-digital”? I’m thinking it can’t because DSD (PDM) is not PWM.

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What an intriguing question!

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Andrus Aaslaid, Estelon / Alfred & Partner, HIGH END Kolleg 2018, 12. May 2018, HIGH END Munich

DSD modulated amplifiers

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I wonder if @tedsmith @Paul or @BHK have ever come across DSD amplification and what they think of it. Would it be something for PS Audio to experiment with? As I understand it DSD is a form of PCM (albeit restricted to 1 bit depth), presumably the same principles used to make Class D amps could be used to make a DSD variant.

Cool thanks for sharing I will add it to my watch list. I didn’t realise someone had already made DSD amps.

Interestingly, I read that Rob Watts (designer of DACs for Chord UK) invented the tech that NAD uses…

https://www.head-fi.org/threads/chord-electronics-qutest-dac-official-thread.869417/page-6#post-13967870

DSD and PCM are pretty far away from each other. DSD is a 1-bit system that’s actually PDM (Pulse Density Modulation) and class D amplifiers are using a PWM (pulse width modulation). Both are much closer to analog than PCM. I suppose a DSD based amplifier would
sound better than a PWM amplifier but both have engineering challenges.

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While I don’t disagree with anything you wrote, the only way that people can actually get their heads around what the DS DAC does is to realise that PDM (aka DSD) and PCM are deeply connected to each other. I like to describe it as DSD being a special case of PCM, because its use of a single bit sample depth gives it unique characteristics which we can exploit to great effect.

The thing that helped me realise this was Ted sharing that he uses the exact same upsampler for DSD and PCM inside the DS DAC. This works because both PCM and DSD directly describe a waveform – plot the points of amplitude versus time, join them with a continuous curve that avoids any frequencies above half the sampling rate, and that’s your analog signal as described by the digital data. They are only intrinsically different in terms of their frequency range and their noise spectrum, but they are practically very different in terms of how amenable they are to numerical processing and what kinds of circuits are needed to perform the D-to-A electrical conversion.

I would also argue that PDM and PWM have important similarities. They’re both based on the principle that you can switch a signal between a positive and a negative voltage at a high switching rate, then low-pass-filter the output to produce a high fidelity signal in the bottom portion of the signal bandwidth. The main difference is that we use PDM as a digital encoding and transmission format, with a requirement for precise amplitude and time domain precision to ensure accuracy of D-to-A conversion, while we use PWM as a transient abstraction to facilitate analog-to-analog conversion (ie amplification) including correction via analog feedback.

Re the original question, the YouTube link posted above is super helpful. I had no idea that Sharp had played with a PDM amp technology or that one existed today in those crazy powered speakers. The presenter does a good job of describing the electronic engineering challenges and downsides of attempting to do PDM at high power levels. Switching precisely between +/- outputs on the order of 70V at several megahertz with vanishingly low jitter is no easy thing to do.

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