Benchmark DAC1 now available with USB

[peanuthead]

Sorry if this question has been asked before, but how do you connect the DAC1 to a headphone amp - through calibrated or variable output? Is one better than the other?
Also, do you use the volume pot on the amp or the DAC1 for adjusting headphone volume? Thanks!

 

[EliasGwinn]

Hi all, I'm sorry I haven't posted in a while.

I want to reply to all of these today...hopefully I'll have a chance to.

All the best,
Elias

 

[EliasGwinn]

Quote:

Originally Posted by peanuthead /img/forum/go_quote.gif Sorry if this question has been asked before, but how do you connect the DAC1 to a headphone amp - through calibrated or variable output? Is one better than the other? Also, do you use the volume pot on the amp or the DAC1 for adjusting headphone volume? Thanks!

Typically you would want to send the highest amplitude signal that the next device can handle. In other words, if the headphone amp can take a maximum of 2 volt input, you would want to configure the source (the DAC1 USB) to send a signal up to, but no higher then 2 volts.

However, there are exceptions, such as if the other volume control is passive. Passive volume controls present a major pit-fall because their impedance is always changing, which will change the response of the system based on volume control. In this case, you'd be better off leaving the passive volume control all the way open and use the DAC1's volume control.

You'll want to speak with the manufacturer of the headphone amp to determine the maximum input level and volume control implementation. Let me know what they say.

Thanks,
Elias

 

[peanuthead]

So how would I go about setting the DAC1 to output at 2 volts? And would this be through the calibrated (fixed?) or variable mode?

If I was going to connect the DAC1 to my Headamp GS-1 amp, how would you suggest I do it? Thanks.


Quote:

Originally Posted by EliasGwinn /img/forum/go_quote.gif Typically you would want to send the highest amplitude signal that the next device can handle. In other words, if the headphone amp can take a maximum of 2 volt input, you would want to configure the source (the DAC1 USB) to send a signal up to, but no higher then 2 volts. However, there are exceptions, such as if the other volume control is passive. Passive volume controls present a major pit-fall because their impedance is always changing, which will change the response of the system based on volume control. In this case, you'd be better off leaving the passive volume control all the way open and use the DAC1's volume control. You'll want to speak with the manufacturer of the headphone amp to determine the maximum input level and volume control implementation. Let me know what they say. Thanks, Elias

 

[EliasGwinn]

Quote:

Originally Posted by peanuthead /img/forum/go_quote.gif So how would I go about setting the DAC1 to output at 2 volts? And would this be through the calibrated (fixed?) or variable mode? If I was going to connect the DAC1 to my Headamp GS-1 amp, how would you suggest I do it? Thanks.

Contact the manufacturer and find out the maximum input level and volume control implementation (passive? IC? digital? gain circuit?). When you know these things, we can determine the proper way to connect everything.

All the best,
Elias

 

[EliasGwinn]

Quote:

Originally Posted by G-U-E-S-T /img/forum/go_quote.gif I will definitely have to try those Focals - thanks for the reference! May I ask again please: With active speakers, does the previous advice that one's regular listening levels should reach up to around 3:00 on the DAC1 volume dial, still hold?

Yes...between 11 o'clock and 3 o'clock is the ideal.

All the best,
Elias

 

[EliasGwinn]

Quote:

Originally Posted by Quaddy /img/forum/go_quote.gif elias, kind of a trivial question... after the dac1 pre has been powered off (unpluged from mains) and then with reconnection to the power it defaults to the analog input (shown via the blue led on front panel) is there any way to change this default?, is there a jumper or similar, ideally would prefer it to come onto my main source which is USB thanks

The DAC1 PRE remembers the input that was being used when it was powered down for 10-20 minutes. After that, it will default to the analog input.

Best,
Elias

 

[hpz]

Hi Elias,

I have just purchased a DAC1 USB and i am using them with IEMs via the headphone outputs. Is there anyway to lower the gain even further than level -10db?

I'm only using about 4-5 ticks up (8 o clock) and its already loud enough, but the sound is imbalanced and unclear.

Anyway to reduce the gain even more?

Any help would be appreciated
Thanks

 

[little-endian]

Hi Elias,

after some time again I want to revive two questions I had before and add another one:


1. DAC1 inside view

Weren't you going to investigate why the inside views disappeared on Benchmark's web site?


2. Volume control

Here and in the feedback newsletter you explained how and why DSPs reduce the SNR aka dynamic range compared to volume pots.

I don't doubt that a second however it is still not totally clear to me why this gives any drawback in real-life practice:

You said a DSP controls the values before the D/A conversion. Lower values mean a lower SNR by definition. A volume pot might lower the noise as well, keeping the SNR by definition, isn't it? But if one listens at lower levels for comfortable listening, reaching ~ 70 dB(A) at maximum for instance, don't my ears limit the SNR / dynamic range as well? Aren't the ears comparable to the DAC - lower "values" (or sound pressures) increase the "wasted" headroom? From hearing threshold level to maximum there might be just 40-50 dB left (due to environment noise). Lowering the volume, wouldn't the dynamic range decrease due to hearing capability anyway - regardless of the technique used to lower the volume?

Asked another way: Assuming I have music playing with an approximate dynamic range of 30 dB (this should be realistic for recordings beyond the loudness war; early 80s for instance) - one time with the DAC1, the other with an AVR using a DSP for volume control for instance.

In both cases I set the volume so the loudest parts reach 80 dB(A) (and the quietest 50 dB (A) respectively). What exactly will I gain in regard to SNR or dynamic range when listening to the DAC1 compared to the AVR?

Of course - by definition, the noise will be slightly higher with the AVR since the DAC's full scale isn't entirely used anymore but wouldn't that introduced error below the ~ 50 dB of the quietest parts in the music anyway - being masked so to speak?


3. AVRs

I know you avoid taking position when it comes to competitive products for good reason but what do you think in general about most AV-receivers. As far as I know almost all use DSPs to adjust the volume, aren't they? Are 32-bit DSPs common or something else?


Maybe it would be no mistake to create an own thread "SNR / dynamic range" to discuss that for all in more detail.

 

[Shadorne]

Quote:

Originally Posted by little-endian /img/forum/go_quote.gif 3. AVRs I know you avoid taking position when it comes to competitive products for good reason but what do you think in general about most AV-receivers. As far as I know almost all use DSPs to adjust the volume, aren't they? Are 32-bit DSPs common or something else? Maybe it would be no mistake to create an own thread "SNR / dynamic range" to discuss that for all in more detail.

My Anthem AVM20 uses a crystal 3310 chip (matched resistor array on a chip) - these are pretty good at volume control ( I am not saying they are by any means perfect but the specs ain't bad)

Complete Digital Volume Control
— 2 Independent Channels
— Serial Control
— 0.5 dB Step Size
Wide Adjustable Range
— -95.5 dB Attenuation
— +31.5 dB Gain
Low Distortion & Noise
— 0.001% THD+N
— 116 dB Dynamic Range
Noise Free Level Transitions
Channel-to-Channel Crosstalk Better Than
110 dB

So I guess the answer is it all depends....what kind of DSP or AVR...

 

[wavoman]

We have discussed here from time-to-time Benchmark's "UltraLock", and posters have asked if it is more than just re-clocking on the DAC side. The Benchmark man, EG, says "yes, much more".

I now have my own proof. Through some poor re-cabling and SPDIF signal splitting, I have (without meaning to) introduced a high degree of jitter into my SPDIF chain.

One of my DACs, which does re-clock for sure, can no longer reliably sync with the SPDIF signal. The manufacturer says it is the jitter I have introduced. Even though his unit re-clocks, the SPDIF receiver circuit itself cannot "lock on", the SPDIF jitter is so bad. He said "Benchmark's UltraLock can do this" ... and indeed it can. I plugged my Benchmark DAC-1 in instead, and it sync'd right away. Sounded great!

There is no mistaking this -- I plugged in one, then the other -- back and forth ... the non-Benchmark would lock-on sometimes and then suffer audio drop-outs, other times just display "No Data". But the Benchmark DAC-1 was rock solid every time!!

 

[Quaddy]

/\ good stuff, i always had the feeling and with experience of their product that the dac1 and benchmark in general really dont dabble in snake oil and fluffy vague talk.

i always have seen there products to be solid, reliable, highly practical and more heeled in the realm of professional type audio where everything needs to be as it says on the tin, not like these boutique type products where the language can get very fancy and very meaningless real quick

 

[JamieGreen]

First I have to say that I like the Benchmark DAC1 pre with my IEM (Wum3x) very much with 20 dB gain reduction. The question is if I can set one headphone jack -20dB and second left with -10dB for my another 300Ohm headphones?

Thanks

 

[wavoman]

Quote:

Originally Posted by Quaddy /img/forum/go_quote.gif ...heeled in the realm of professional type audio ...

Yea that's it exactly. It is both pro gear, and well suited for home use. Not a lot of manufacturers fit so well in to both worlds. The other DAC is extremely high-end audiophile, but simply can't tolerate the cable length and patch panels I use, at least as far as its SPDIF input module is concerned.

If I used AES/EBU everywhere, with tight-tolerance 110 ohm connectors for the SPDIF I would be fine, but like everyone I am lazy and sloppy -- I am re-using 75-ohm composite video wiring. Still, the Benchmark DAC-1 locks on no problem.

 

[little-endian]

Quote:

Originally Posted by wavoman /img/forum/go_quote.gif One of my DACs, which does re-clock for sure, can no longer reliably sync with the SPDIF signal. The manufacturer says it is the jitter I have introduced. Even though his unit re-clocks, the SPDIF receiver circuit itself cannot "lock on", the SPDIF jitter is so bad. He said "Benchmark's UltraLock can do this" ... and indeed it can. I plugged my Benchmark DAC-1 in instead, and it sync'd right away. Sounded great!

Yeah, I think that would be my forth point for Elias.

It is interesting - I recognized that my Harman/Kardon AVR for example outputs a distored audio signal if the S/PDIF is too bad while the Benchmark DAC1 plays the same source flawlessly and never distorts but mutes when in doubt.

The question is if the jitter really increases with cable length or just the attenuation.

It would be interesting to know what causes the different input sesitivity of different devices.