Benchmark DAC1 now available with USB - Page 87

A newbie's question here: How do I hook this DAC1 through XLR to a hd650? I have the cables but I've read that you need to set it to "variable" mode? Does this allow the front pot to control the volume once I'm using XLR? Also I've heard that you have to open the DAC1 up to play with the db setting for optimal sound? Is this true and necessary?

Yes, you have to set it to variable mode in order to enable volume pot adjustment of the xlr outputs. You can try it with just a normal connection to a preamp, the loudness should vary with pot setting. Once you confirm that , turn the pot to zero and plug you headphones in.
The attenuation jumpers inside also affect the XLR volume, start with -20dB setting and see if the pot regulation is comfortable. If it is too quiet change it to -10dB or to 0 for no attenuation. Let us know if you like the sound better than from the normal headphone output. I am curious myself.

Hello dspargo,

It's interesting that you ask this, as it is the topic of our next "Feedback" newsletter.

Poo's instructions are correct. To connect the XLR outputs to RCA inputs, you must have a cable that has a female XLR connector with pin-3 floating (not connected) and a male RCA on the other end.

If you buy this cable, double check to make sure pin-3 is floating...MAKE SURE IT IS NOT GROUNDED!! There is a possibility of damaging the DAC1 if you use an XLR output with pin-3 grounded for extended periods of time.

There are a few places you can buy these cables. We sell them (part#: 500-06902-000 and 500-06906-000 are 2' and 6' versions, respectively). But wherever you buy them, check and double-check to make sure pin-3 is floating.

Thanks,
Elias

There are no differences between the optical output and USB output, provided that all settings are correct.

This is an interesting observation that a lot of our customers have. The sound of the DAC1 does not change, but instead your perception changes. Many feel the DAC1 is bright when they first listen, but this is because it doesn't have all the distortion (jitter, etc) that typically 'beef-up' the mids and low-mids...things that usually cause long-term listening fatigue. Consequently, once accustomed to the DAC1's transparency, the listener deeply appreciates the clarity of the sound. The true musical nature of the recording can finally speak for itself, unimpeded.

That was the long answer to your question, but the "softer" perception is very common. The entire playback system (the DAC1 + you're ears) have achieved their 'break-in' period. ;-)


It seems you have all the correct settings. With these settings, iTunes and VLC should perform equally.



Yes.

Yes, but it is one of many things they are interested in. Overall system performance is the main thing. Therefore, the biggest problem in CoreAudio (sample-rate conversion), may not see an upgrade for some time, because higher quality conversion will take more system resources.

Soon, hopefully.

Great!! And as always, feel free to ask as many questions as needed...

Thanks,
Elias

Hello roadtonowhere08,

Without being familiar with these other sound cards, I would guess that you are correct in your assumption. That is, the other cards' outputs are configured for the typical +4 dBu output, which is how the DAC1 is with the attenuators set at 0 dB.

However, if you change the attenuators from -20 to 0 dB, and also turn foobar's volume down by 20 dB, you will have the exact same signal level from the DAC1's outputs as you did before you changed those things. In other words, your adding 20 and subtracting 20 all at the same time.

If you were not clipping your amp before the changes, you will not be clipping after the changes...you'll be right where you started. However, you will have a slightly compromised signal-to-noise ratio because the digital signal has been attenuated so much. In other words, you will reducing the signal level, keeping the same amount of noise, and then amplifying all of it. This may or may not make a difference in your system, depending on the s-to-n ratio of the other gear etc.

The ideal configuration has the digital volumes as close to 0 dB (unity gain) as possible, and the attenuation is done in the analog domain. This way, the noise gets attenuated as much as the signal.

I hope this long-winded explanation makes sense...please let me know if you'd like me to clarify any of this...

Thanks,
Elias

Hello KarateKid,

First of all, we recommend not changing the attenuation jumpers unless you determine that the output level needs to be adjusted.

Second, you should definitely use the 'Variable' mode to control the volume. Start with the volume all the way down, and slowly increase to determine the ideal volume. The goal is to have comfortable listening levels when the volume control is between an 11 o'clock and 3 o'clock position. If comfortable listening levels are outside of this region, then the attenuation jumpers should be adjusted.

Also, we do not recommend driving headphones directly from the the XLR outputs. You won't damage anything, but there will be significant distortion (non-linear frequency response). There are several reasons why it is advantageous to avoid this mode of operation (and we can discuss these, if you want...there are several pages within this thread based on this exact topic). At first, you may enjoy the sound of XLR-driven headphones, but you also may experience listening fatigue after listening in this mode for extended periods.

The HPA2 headphone amp that is built into the DAC1 is ideal for driving headphones. This is the same headphone-amp used by recording and mastering engineers in some of the biggest studios in the world (not to mention the broadcast engineers at ABC, CBS, NPR, etc....).

Thanks,
Elias

Joijwall, Elias;

Actually USB would be much better than the Toslink output on the Macbook. The toslink is not really that well done unless you want to physically isolate the DAC1 from the computer.

We did some testing with the Prism dScope III a couple of months ago with the Macbook and jitter was really high.

As Elias has pointed out much of that will be filtered out by the DAC1 but in most cases I have found that the more there is at the beginning the more that is going to get through in the end.

Better off to use the USB I would think in this case.

Thanks
Gordon

Assuming one has an appropriate D/A-converter, the DC-isolation is actually the big privileg of the Toslink connection - and a bit more facinating as well, of course.

That's quite possible.

From what I'm aware of, the jitter actually isn't filtered out at all but the oversampling ratio is varied instead to prevent buffer underruns or overflows (Elias may correct me here if I'm wrong since he has to know the concrete implementation).

If the datasheets from Benchmark are correct - and I want to believe they are - then they have measured the effects of jitter by far more detailed than anyone is able to hear hence if they are telling the truth, there is nothing but the data which can change the sound quality.

@Elias: Since the output of the DAC1 is muted incredibly fast when no S/PDIF-signal is present anymore, I suppose that the used buffer has to be comparatively small. I'd be interested to know the exact size of it.

Makes perfect sense. I will keep everything as is, as what you said makes sense. The digital signal is less attenuated, from what I have read the -20dB jumpers make no difference in sound, and the pots on the amp are almost at the noon position which improves tracking from what I have read. Since the run of cable from the DAC to the amp is only 2 feet, the decrease in signal coming from the DAC will have little to no EMI interference. Does this check out to you?

Yeah well I just thought I could see what the fuss regarding XLR is all about. I mean isn't XLR directly off the DAC1 USB better than the HP-2 amp (the national opamp in the XLR vs the TI in the hp2)? Shouldn't it be better or good enough to use until I can get a balanced amp?

Edit: Wrong info.

infinitesymphony,

Doesn't Elias mean that XLR output in the dac1 USB is 4562?

Gordon,

This may be the case with some DAC's, but it really will not make a difference to the DAC1 / USB / PRE. Even with >12 UI of jitter (a ridiculous, astronomical amount), the DAC1 does not see any change in performance.

In fact, we created an interesting test where we connected the XLR/AES input of the DAC1 via 1000 feet of CAT-5 cable wired to XLR connectors. Even with this (insane) scenario, the DAC1 suffered no performance loss whatsoever.

Thanks,
Elias

I will get back to you on this (I want to make sure I only post correct information)....

Thanks,
Elias

I looked back through the thread, and I think you're right. Sorry about that.

But, I'm assuming that LM4562 is also used for the headphone amplifier portion, since it was mentioned that LM4562 would be superior to NE5532 for low-impedance headphones.