You make a really good point here, but also miss one or two other critical ones.
FPGAs aren’t by themselves a replacement for a DAC chip. (OK technically you probably could generate analog audio out of an FPGA but it probably wouldn’t sound any better than the cheapest of chip DACs.) The FPGAs are there to provide complex, high speed logical operations and in particular to perform sample rate conversion, digital low pass filtering, digital attenuation, and particularly in the case of Ted’s designs as well as this newest Teac it will also perform the modulation and noise shaping necessary to generate a 1-bit DSD output stream. The FPGA is a digital-to-digital converter in both the PS Audio and the Teac products. Yes they are competing in purely algorithmic terms to determine the better-sounding bits to output given the same PCM or DSD input, but they are also competing in terms of implementation expertise to minimise noise and jitter generated by the FPGA in the process. That noise and jitter doesn’t change the bits in the output stream but it does influence the result of the actual D-to-A conversion that follows.
The deeper innovation in Ted’s design relates to clocking, and based on what I see from the Teac layout I suspect he’ll keep his edge on this one. First there was the decision to optimise the clock not for long-term stability but for peak-to-peak consistency. Our ears can handle smooth frequency drift over a period of several seconds but we are insanely sensitive to frequency instability down at the microsecond-to-picosecond levels. Second is to have a single clock that’s an integer multiple of the output sample rate, placed immediately adjacent to the actual output switches, and from which every other clock in the system is derived – this really lowers the floor for the level of output jitter that can be achieved. And lastly I’ll mention the way that the DS DAC observes the incoming data rate with a degree of tolerance and gently adjusts its own master clock to match the average pace rather than locking to and being driven by the incoming signal’s clock and whatever jitter it contains. These design features contribute massively to the quality of the DS DAC’s output and I can tell from the dual mono circuit layout that Teac is not following particularly closely in Ted’s footsteps here.
Then we have the actual D-to-A conversion from 1’s and 0’s in a MHz-scale bitstream to a line level analogue audio waveform with sub-50kHz signal bandwidth. You can get a sense for how much this matters from the discussion of the DS Sr vs DS Jr vs TSS designs vs DS Sr mods with upgraded transformers vs the changes we’ve been told are coming in the DS Mk II. You need ultra-stable power supplies. You need switching devices with sufficient speed and consistency. You might have two, four, eight of them per phase per channel to increase SNR. Then there’s the analog low-pass-filter circuit, which is where the transformers come in on the DS Sr and above. If your switches and filter are too low level you might have to then run through some kind of line driver (which is what the Jr and the Teac do).
Although they both may be using FPGA to generate a DSD bitstream and then doing a direct analog conversion with some kind of low-pass filtering, the Teac product and the Ted Smith products are focused on very different priorities and a direct price/feature comparison is almost meaningless. Just about every “feature” on the Teac spec sheet degrades the ultimate sound quality that comes out of the device. “Dual mono” has some benefit in terms of power supply but it moves the D-to-A conversion some distance from the clock source and requires all these point-to-point wires that will radiate noise throughout the chassis. Bringing all that Ethernet connectivity and CPU-based sophistication inside the box also means more noise.
The TSS DAC is an all-out effort on sound quality where the most important and expensive changes are what happens downstream from the FPGA. Put the master clock, the output switches and the analog low-pass filter in a completely separate box with uncompromising power supply design, and keep all the noisy digital pieces several feet or even tens of meters away at the other end of a non-conductive optical fibre. Yes the FPGA will be more capable and we’ll get some benefit from that, but it’s far from the most significant change.
The DirectStream MkII is halfway between DS Sr and TSS. More optimised power supplies, better output transformers (yes @tedsmith I’m pretending this is a done deal), more isolation from noise borne on digital audio connectors, the eviction of noisy ethernet things, a quieter and better isolated display and front panel controller, the ability to actually power down unused inputs like USB, switchable ground lifts all over the place. Yes it also gets the newer FPGA and will benefit from Ted’s experience in lowering the noise and jitter generated within the FPGA itself, but compared to everything else… that’s the least interesting thing about it.
The Teac product is interesting, and will probably make many customers very happy, but those are very different customers from the ones that PS Audio is targeting here. Just because it has an FPGA doesn’t mean it’s even close to being in the same category.