Benchmark DAC1 now available with USB

[little-endian]

Originally Posted by EliasGwinn

The DAC1 achieves 21-bits of signal-to-noise ratio.

The DAC1 can accurately resolve the 24th-bit (although it will be below the noise floor).

Interesting statement, Elias.
I already asked myself why Benchmark doesn't list the dynamic range of the DAC1. Although this value is often used equally to the signal to noise ratio, some seem to distinguish between them. For example, according to the mastering engineer Bob Katz, one can hear details below the noise level, thus the dynamik range can be greater than the SNR, especially in conjunction with dithering (as far as I remember he gave 91 dB SNR and ~ 116 dB dynamik range for properly dithered 16 bit material).

Now it would be interesting to know how great the dynamic range (!) of the DAC1 actually is. If it should be really able to resolve the 24th bit, it would have to exceed 140 dB. Is this the case?

I'm confused also why more than 20 Bit of wordlength are used, at all if no converter is actually able to reach such a huge SNR and dynamic range. Many devices don't even match 20 bit performance (by pure math).

I'm sure you can clarify this. Again this would be worth an own thread.

little-endian

[Crowbar]

Originally Posted by little-endian

I'm confused also why more than 20 Bit of wordlength are used, at all if no converter is actually able to reach such a huge SNR and dynamic range. Many devices don't even match 20 bit performance (by pure math).

24 bits is more than the dynamic range of the human ear. The reason for long words is most types of DSP processing effectively reduces the resolution, even if the DSP internally uses more bits. You can easily see that with an image editing program. Take a look at the histogram of an image, then apply some global processing such as equalization, and take a look at the histogram again. It will no longer be smooth and continuous; the effective quantization is worse.
192 kHz sampling rates also don't make sense from an audibility standpoint, but they could be useful to get increased effective resolution by dithering, and so it goes to the same point as the increased wordlength.

[CanMad]

My apologies to everyone, I didn't mean to thread cr@p.

It would seem however that a transformer based output stage might not be the most accurately measuring. Therefore I understand that this would be undesirable in a device such as the Benchmark which is striving to achieve the most accurate performance.

However this is from the Townshend web page for peoples info. Make of it what you will :
Audio Amplifier
Normal practice is to use integrated circuit operational amplifiers in the audio signal path. Unfortunately, there are serious problems, even with the “Best” audio grade devices. The first problems are over-complication. These devices may contain up to 1000 transistors and resistors every one of which has the capacity to loose a minute amount of fidelity. Secondly, the myriad resistors are not perfectly linear at very low voltages. The result is a component which has slight veiling and “grunge” distortion. Our simple discrete component fully class A operational amplifier simply doesn’t suffer from this.
Even the best mix of dual monolithic junction field effect transistors and bipolar junction transistors in a single ended pure class A configuration with optimum global feedback for the highest linearity and lowest distortion was still not good enough, so now the gain is provided by our unique EDCT wired step up transformer and unity gain buffer to eliminate all high order harmonics to bring the ultimate in fidelity. THD and IMD measured at better than -120dB.

I'd still be interested if Benchmark considered a discrete output stage though?

[Crowbar]

Originally Posted by CanMad

Not to mention all the VERY positive reviews?

Reviews weren't based on blind tests, so they are irrelevant. Those people are hearing little but their psychological bias.

Maybe the problems you mention are why they sound so good (I suspect probably not).

Some people like the sound of coloration. Why do you think Grado headphones are popular?

Anyway the question was more about a discrete output stage avoiding op-amps than specifically a transformer based one.

I prefer discrete stages myself, and I have no problem with that. But the suggestion to add a distortion-producer a.k.a. transformer had to be addressed.

I do realise that there are very good op-amps these days, but I don't think the Benchmark uses the latest and greatest (correct me if I'm wrong Elias).

I think the new LM4562 opamp used in the latest version is very nice. It's already very linear before any feedback, which is not commonly seen in opamps.

When are you starting your own hi-fi company

Is that a challenge? One doesn't need to sell commercial equipment to exercise in electronics. I'm a long time DIYer and I've no doubt I and a number of other DIYers around this and other forums can build a DAC that beats even Lavry and MSB stuff for much less of the cost. As a DIYer I have none of the economic considerations of a company that has to maximize profit.

as you obviously know more about engineering than Charles Hansen and Max Townshend?

You're embarassing yourself with an argumentum ad verecundiam here. I recommend taking a basic critical thinking course at your nearest college.

But if you're going to go this route, I suggest you research the enormous engineering effort that has gone towards eliminating transformers from the audio signal path, with the numerous OTL tube amp designs from a variety of companies, and multiple improvements of the technology over time such as Rozenblit's Transcendent OTL, and culminating in Berning's ingenious ZOTL circuit. Speaking of Berning, he has a great visual demonstration of the evils of transformer coupling; take a look at Figure 3 to see the distortion and hysteresis caused by a transformer in the signal path: http://www.davidberning.com/Transfer%20Char.htm

Thanks for focusing on the one part of the question that you thought was bad.....like a politician!

Why focus on the other parts, if I didn't see anything wrong with them? Attention should be paid where there are improvements to be made.

[lowmagnet]

Guys, please cut it out. This is about the only thread I subscribe to because I'm tired of everything going downhill, and you two and your personal issues aren't helping any.

[tszyn]

Thanks to all for this very informative thread! I have posted a number of questions to the "computers-as-source components" forum in the hopes of taking care of a few loose ends in the whole KMixer discussion.

http://www.head-fi.org/forums/f11/computer-speaker-suggestions-239333-post2962741/

I would greatly appreciate it if Elias or other experts who participated in this thread would provide their input.

Thank you,

Tomasz

[little-endian]

@Crowbar

Of course I understand that it makes sense to user higher wordlengths and sample rates to have more reserves for processing audio material (rounding errors, headroom, etc.) but it doesn't explain its usage for pure digital to analog conversation for the endusers. I accept higher sample rates, used internally to simplify filtering, etc. but I don't see it for the wordlengths. Besides that, the "joke" is, that higher sample rates can be reproduced on the analog side, at least (although not hearable) but this is not the case for huge wordlengths. As far as I know, these 144 dB, 24 bit - sources theoretically provide, can't be really reproduced due to the noise of the signal path (resistance, etc.).

Correct me if I'm wrong. :-)

[Crowbar]

Yes. You'd have to use bulk metal foil resistors and very low noise transistors, maybe JFETs, etc. Well, JFETs are cheap, but the bulk foils... last time I checked Vishay was charging several dollars a piece

But 144 dB is not needed. That's more than the ratio of threshold of audibility to hearing damage. Even to accomodate large transients, you shouldn't need more than 15-20 dB less than that figure. More attention should be spent on the analog stages to preserve as much as possible of what the D/A gives you. Not to mention that a speaker unless electrostatic or plasma, will usually have distortion on the order of a percent.

[little-endian]

Of course, from the point of view in regard to the human ear's dynamic range, I want to claim that even 16 bit resolution is more than enough. No single track makes use of it thus the only benefit is the great SNR as far as I can see. One demo track (chinese drums) from a Burmester CD uses actually 40 dB of dynamic range which is already enormous! So I'm just interested if there is even one single device which reaches a signal to noise ratio of more than 144 dB. Otherwise the whole 24 bit fuss (for DACs) seems to be quite ridiculous to me.

How low the dynamic range is allowed to be while still being able to deliver great music dynamic shows the LP - some people even prefer it to digital sources (which seems to be completely unfounded since no test I'm aware of compared the direct vinyl source versus one cascaded through a potent ADC and DAC, but just different masterings). I bet, no one would pass a blind test!

[Crowbar]

Originally Posted by little-endian

No single track makes use of it

That is false. I have a number of 24 bit 96 kHz recordings on DVD (not DVD-Audio) that have a dynamic range exceeding 16 bits. Despite the non-nil noise floor in some, as was already explained above, dynamic range beyond the noise floor can still make an audible difference.
When you say a CD uses 40 dB, you must be looking at the power envelope variation between quietest and loudest passages, whereas the texture of an instrument is made up of the myriad harmonics that cover a much larger range. Even without this, just the power envelope in a classical orchestra typically does 100 dB, with potentially even larger transients.
Just because most recordings are not of quality is not a reason not to support the ones that are, as few as they may be. Plus, if the playback gear support is there, the number of high resolution recordings will increase if people would bother to buy quality stuff.