[EliasGwinn]

Quote:

Originally Posted by pcf /img/forum/go_quote.gif Hi Elias, I have also been under the impression that the performance of DAC1 was less dependent on the transports used than others. That is based on our phone conversations and info from this thread. I also own two of the DVD players on your transport test list. (Oppo and Pioneer). I did my own listening tests comparing those two budget players as transport to some cd transports that costs ten times more. They seemed to sound the same to my ears when they were hooked up to the DAC1. But are you telling us now that different transports affect the performance of DAC1 more than we were led to believe? I thought that was one of the strong selling point of DAC1. I am still very happy with my DAC1pre though. Keep up the good work. Paul

No, what I'm saying is that different transports may or may not mangle the data. Almost all transports work perfectly at redbook (44/16), although errors may result because of the condition of the CD itself and any dust particles that are in the optical system. Each transport has its own method of dealing with a read-errors, and the data may be modified as a result.

Best,
Elias

 

[pcf]

Quote:

Originally Posted by EliasGwinn /img/forum/go_quote.gif No, what I'm saying is that different transports may or may not mangle the data. Almost all transports work perfectly at redbook (44/16), although errors may result because of the condition of the CD itself and any dust particles that are in the optical system. Each transport has its own method of dealing with a read-errors, and the data may be modified as a result. Best, Elias

Thanks for the quick reply and clarification!

Paul

 

[G-U-E-S-T]

[That's weird - an earlier post I made above, disappeared overnight! So I will post it again.]

Elias, I'm curious please: You've mentioned earlier that the ideal volume-dial position on the DAC1 is between approximately 12 and 3 o'clock, and that attenuating below this can compromise system SNR. But, so long as it doesn't overload the input of an amplifier of course: What about going beyond 3 o'clock on the DAC1 dial (i.e. into the DAC1's up to 10db "positive-gain" region) - why please is doing that, considered less than ideal?

 

[urbo73]

I would like to also understand this better. Why exactly is the 11-3 or 12-3 range the most ideal, or best? What happens if I'm under 11 or over 3 let's say?

 

[EliasGwinn]

Quote:

Originally Posted by G-U-E-S-T /img/forum/go_quote.gif Elias, I'm curious please: You've mentioned earlier that the ideal volume-dial position on the DAC1 is between approximately 12 and 3 o'clock, and that attenuating below this can compromise system SNR. But, so long as it doesn't overload the input of an amplifier of course: What about going beyond 3 o'clock on the DAC1 dial (i.e. into the DAC1's up to 10db "positive-gain" region) - why please is doing that, considered less than ideal?

Quote:

Originally Posted by urbo73 /img/forum/go_quote.gif I would like to also understand this better. Why exactly is the 11-3 or 12-3 range the most ideal, or best? What happens if I'm under 11 or over 3 let's say?

The ideal range of the DAC1 volume control (10:30-ish to 3ish) is where you will acheive the most accurate left-right balance. When you adjust a stereo potentiometer, you are actually moving two wipers across two elements (one for left and one for right). The 'middle' of the elements acheive pin-point precision, whereas the extreme ends of the elements are less likely to match each other quite so exactly.

G-U-E-S-T,

The SNR won't be affected by the pot position, which is one of the great benefits of using an analog volume control. Whatever noise is going into the pot will be attenuated or amplified equally w/ respect to the signal. Therefore, the signal-to-noise ratio will always remain consistent through that circuit.

Perhaps you are thinking about our conversation with high-power amps, which will limit the amount of signal you can send from the DAC1 before it becomes too loud. This will affect SNR. Although, someone made a great point (I believe in this thread) that its more a function of input gain/sensitivity. The ideal amp will allow you to send high levels from the DAC1 without it becoming too loud for your listening preference.

All the best,
Elias

 

[G-U-E-S-T]

Quote:

Originally Posted by EliasGwinn /img/forum/go_quote.gif ...The SNR won't be affected by the pot position, which is one of the great benefits of using an analog volume control. Whatever noise is going into the pot will be attenuated or amplified equally w/ respect to the signal. Therefore, the signal-to-noise ratio will always remain consistent through that circuit. Perhaps you are thinking about our conversation with high-power amps, which will limit the amount of signal you can send from the DAC1 before it becomes too loud. This will affect SNR. Although, someone made a great point (I believe in this thread) that its more a function of input gain/sensitivity. The ideal amp will allow you to send high levels from the DAC1 without it becoming too loud for your listening preference. All the best, Elias

Hi Elias,

I apologize, you are right - I was in fact thinking of the high-power amp conversation. Actually I think I've (hopefully) learned several things along the way here:

1) For best channel-balance performance, stay between approx. 11 and 3 o'clock on the volume-dial;

2) The DAC1 output's signal-to-noise ratio (SNR) is always constant, regardless of its volume-dial position and/or internal attenuator-pad selection;

3) The amplifier's input-stage has its own inherent noise that gets amplified along with the input signal. So generally speaking, and obviously assuming no ground-loops or other external issues: The optimum SNR through the amp's input stage is by definition achieved by delivering the highest-level (i.e. greatest voltage, least attenuation) input signal from DAC1 to amp as possible, without ever exceeding the amps maximum input voltage (since exceeding this would overload the amp's input stage, and cause the amp to clip and distort). Thus the ideal amp for the DAC1 will have a low input sensitivity, being able to at least handle the maximum peak voltage output of the DAC1;

4) Well-built amps perform best (i.e. with lowest distortion & noise) at the top-end (i.e. between 75% and 95%) of their power-delivery capability. So the ideal amp, when receiving the highest possible DAC1 signal as described in (3) above, will also simultaneously be driving the user's speakers to their most comfortable listening levels with their preferred music type, without also having too much more additional headroom. Thus the ideal amp will have a maximum input voltage at least equal to (but at most only slightly greater than) the maximum peak voltage output of the DAC1, and will also be producing comfortable listening levels when receiving from the DAC1 these optimally highest-possible input signal levels (which also might mean that in a system with proper gain-staging, the ideal home-use amps for most users are likely of much lower power than the marketing strategies of many high-end amp manufacturers would have us believe).

Elias, please advise: Do these above points look correct to you?

 

[music_man]

Quote:

Originally Posted by Matias /img/forum/go_quote.gif There is controversy. Just yesterday we tested my DAC1 USB in a system based on Dynaudio Focus 220 speakers and Krell 400ix integrated amplifier, with an Arcam transport. Switching the stock fuses to the HiFi-Tuning.com ones I've said a couple os pages before, and switching power cables from a good one to a Cardas Golden Reference, both gave clear results: wider soundstage, heavier and faster bass transients, clearer mid range (voices). But then again, there are those who don't "believe" in cables... even though there's nothing to believe, but to be heard. Cheers, Matias

i know this is an old post. forget dbt. with pro equipment a/b is easy. if i have 5 dac1's in a rack, 4 have supplied power cables and one has any power cable you choose. i challenge you to "guess" which dac1 has the better power cable 5 times in a row. of course you cannot look behind the rack. you can listen to anything you choose for as long as you wish.

i do feel power cords make a difference in some equipment but not on the dac1. neither do cables.

i would filter my electric if it had noise in it otherwise i would simply plug the dac1 into a distribution strip as is done in the biggest most respected studios in the world.

music_man

 

[music_man]

i am assuming most people here have one dac1 which is on a table or on top of another component. with it's stock feet how do you even use an aftermarket powercord? the housing of the iec connector will lift up the back of the dac1! unless you are using isolation or some platform as well. the cardas above would certainly lift up the dac1 i'd think. anyone tried it?

edit: i see many reviews on the net that state the dac1 is very power cable dependent. i like anything that might elevate my audio experience. i have a whole drawer of super expensive power cables and i have done the test i stated above. the only thing i have noticed is that some cables messed up the sound a little. this is because cable companies use tricks in the geometry,shielding,dielectric etc to change the electron flow. nothing actually made it better. than if you get a power cable why not mod the dac1 altogether? mr. gwinn has been through that with us as well. hey remember "pet rocks"? showing my age.

i noticed another thing while i was playing with cables. the iec socket on all my dac1's are a little larger than the cord end. i can wiggle them around quite a bit. anyone else notice this?

music_man

 

[doctorcilantro]

@ ELIAS

Need your opinion on how the new ADC1 USB will play with WASAPI and Sampitude. I posted this in the Samplitude forum; my concern is that ASIO protocol may be overall more elegant (all i/o handled in Patchmix, and less-cpu intensive):

Quote:

Hi all, I have a question about recording with my 1616M. I currently use the EMU ASIO driver to record and monitor a wet signal simultaneously; input balanced into the 1616M ADC converters and push back out via coax S/PDIF to my monitors. I may have some hardware changes soon and would like to be able to do the same thing but using two separate usb devices capable of 24/96 (& 192kHz). I'm not sure of the Benchmark ADC1 USB uses async usb but my playback devices will also be USB which would utilize WASAPI. I'm not real familar with the ADC1 USB but maybe some of you have an opinion on whether or not this will work as well as the all-in-one 1616M method. I don't need low latency as I am recording vinyl with digital RIAA implemented in Sampitude, so I am listening to vinyl this way, but also recording it simultaneously. thanks & happy holidays! DC

 

[music_man]

could someone please tell me. is balanced mic wire(xlr) the appropriate cable to use with the dac1 on the analog outs and 110ohm for the aes/ebu? like vtg,belden,mogami?

thanks,
music_man

 

[JMS]

Quote:

Originally Posted by EliasGwinn /img/forum/go_quote.gif I can only guess, as I've never heard of this before. My guess is that it has something to do with the differences between the designs of the RCA and XLR input stages in the amplifier. Possibly the DC coupling or input transformer of the XLR stage is causing significant low-frequency attenuation. The XLR and RCA outputs on the DAC1 use the exact same signal source and components. All the best, Elias

Hi Elias, revisiting this issue from about a month ago, I now think it's probably an issue with the DAC1 rather than the amp. I just tried comparing the noise floors of the DAC1's balanced output and the unbalanced (with digital input set to silence). With unbalanced, I hear no static/white noise at all through the speakers, while switching to balanced, I hear a very faint amount. It doesn't seem to be the amp's own noise -- when I disconnect the XLR input from the amp altogether, I hear no noise.

Not sure if this is necessarily causing the sonic differences I hear, but it does suggest that the two outputs are not identical, at least in terms of noise floor.

 

[EliasGwinn]

Quote:

Originally Posted by G-U-E-S-T /img/forum/go_quote.gif Hi Elias, I apologize, you are right - I was in fact thinking of the high-power amp conversation. Actually I think I've (hopefully) learned several things along the way here: 1) For best channel-balance performance, stay between approx. 11 and 3 o'clock on the volume-dial; 2) The DAC1 output's signal-to-noise ratio (SNR) is always constant, regardless of its volume-dial position and/or internal attenuator-pad selection; 3) The amplifier's input-stage has its own inherent noise that gets amplified along with the input signal. So generally speaking, and obviously assuming no ground-loops or other external issues: The optimum SNR through the amp's input stage is by definition achieved by delivering the highest-level (i.e. greatest voltage, least attenuation) input signal from DAC1 to amp as possible, without ever exceeding the amps maximum input voltage (since exceeding this would overload the amp's input stage, and cause the amp to clip and distort). Thus the ideal amp for the DAC1 will have a low input sensitivity, being able to at least handle the maximum peak voltage output of the DAC1; 4) Well-built amps perform best (i.e. with lowest distortion & noise) at the top-end (i.e. between 75% and 95%) of their power-delivery capability. So the ideal amp, when receiving the highest possible DAC1 signal as described in (3) above, will also simultaneously be driving the user's speakers to their most comfortable listening levels with their preferred music type, without also having too much more additional headroom. Thus the ideal amp will have a maximum input voltage at least equal to (but at most only slightly greater than) the maximum peak voltage output of the DAC1, and will also be producing comfortable listening levels when receiving from the DAC1 these optimally highest-possible input signal levels (which also might mean that in a system with proper gain-staging, the ideal home-use amps for most users are likely of much lower power than the marketing strategies of many high-end amp manufacturers would have us believe). Elias, please advise: Do these above points look correct to you?

This is 99% correct!

The 1% that I would change is that the ideal amp will deliver a comfortable volume range when the DAC1's volume control is in its best range (11 - 3 o'clock), with 3 o'clock being the loudest the listener would ever want. Also, I would suggest erring on the side of being slightly too powerful, versus not powerful enough.

All the best,
Elias

 

[john11f]

Hi Elias,

I currently use a Dynaudio BM6A MKII w/ my DAC1 Pre connected to an iMac w/c is similar to the BM5A recommended use on the Benchmark website. I find that the bass is a bit lacking and would like to improve on the sound. Have you had any experience connecting a BM9S sub?

 

[EliasGwinn]

Quote:

Originally Posted by music_man /img/forum/go_quote.gif i know this is an old post. forget dbt. with pro equipment a/b is easy. if i have 5 dac1's in a rack, 4 have supplied power cables and one has any power cable you choose. i challenge you to "guess" which dac1 has the better power cable 5 times in a row. of course you cannot look behind the rack. you can listen to anything you choose for as long as you wish.

This is a good test, but you can do something like this without 5 DAC1's. Have your spouse, child, or friend switch the object in question (power cable, digital cable, transport, etc) without you seeing. Obviously, you need to control your own testing (make sure you can't tell which cable/device is connected), but you owe it to yourself to challenge the supposed improvements with expensive aftermarket add-ons.

Best,
Elias

 

[EliasGwinn]

Quote:

Originally Posted by doctorcilantro /img/forum/go_quote.gif @ ELIAS Need your opinion on how the new ADC1 USB will play with WASAPI and Sampitude. I posted this in the Samplitude forum; my concern is that ASIO protocol may be overall more elegant (all i/o handled in Patchmix, and less-cpu intensive):

I'm currently testing this. Keep in touch...

Best,
Elias