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Please don't get sidetracked

While I admit that I think one of the biggest jokes in the high-end industry is that many so-called "circuit designers" no longer design circuits, but instead just use pre-packaged boxes of circuits - often the exact same pre-packaged boxes of circuits used in extremely low-cost components from the Far East, that misses the main point.

The main point to me is that Jim Austin tested two completely different DACs. Almost nothing about them was the same:

Analog stage:

Benchmark uses op-amps, both for I-V conversion, the audio output, and the low-pass reconstruction filter.

PS Audio uses the FPGA to send a PWM signal to (I suspect) logic gates that produce a PWM that approximates the analog signal and (as has been claimed many times about DSD) only needs a low-pass filter for decoding. My understanding is that the low-pass filters used are a pair of high-quality (Jensen?) small-signal audio transformers.

DAC stage:

Benchmark uses an ESS ES9028, which includes more options than you can shake a stick at. If you want, you can use it as the DIR (Digital Interface Receiver) to decode S/PDIF, will also auto-decode DoP and convert it to DSD, will auto-detect and decode DSD. It also includes a 32-bit internal digital volume control, ones's choice of digital filter options, an IIR-based low-pass filter for DSD, an ASRC used as a "jitter eliminator" (turned on by default, and if so normally has a fixed 100 MHz clock to run the modulators which is asynchronous to either family of digital audio sampling), the modulator used to convert to 6-bit output, and a 6-bit (64 unitary weighted current sources)

PS Audio uses a (7th order?) modulator to convert all incoming signals, either PCM or DSD to 20x the rate (was originally 10x) of DSD at presumably 1 bit (?), or 1280 x 44.1kHz = 56.448 MHz. I am unsure if this changes in the case of 48kHz based signals or not, but it is equal to 1176 x 48kHz, so one master clock would be synchronous to to both sample rates. I believe there is some type of digital volume control implemented, but have no idea if that is true or at what stage it is applied (must be before modulation to 1 bit (if that is what it does), as it is not possible to change the volume of a 1-bit signal without first converting it to multi-bit.

There are of course many differences in the power supplies, connectors, PCB material, PCB layout, and just about a zillion other design choices - each of which affects the sound quality of the final unit. And Jim Austin can hear no difference between them whatsoever.

In my experience, changing just one of these factors results in an audible difference. So that leaves us with only a few possible conclusions:

1) I simply imagine the differences I hear whenever I perform listening tests to anything, including speakers.

2) Jim has reached the point where he is unable to hear any differences between equipment built not only with different parts, but completely different circuits and completely different design principles.

3) Jim's system is of such low resolution that it masks the differences between the two different sources.

4) The result of Jim's listening test reflect a listener bias, determined to hear exactly the same thing regardless of the actual sound.

I will leave it to you to let me know if there are any other possibilities I overlooked, and what your conclusion for his "no difference" conclusion is. In the meantime there seem to be a group of "objectivists" at the Stereophile website joyfully cheering the day that one of Stereophile's own reviewers has "exposed" the high-end as a fraud, and that a $2000 Benchmark sounds every bit as good as a $6000 PS Audio.

What's next - Stereophile will become the official journal of Hydrogen Audio? That is why I found Jim Austin's review to be so depressing/disgusting.


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