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Benchmark DAC1 now available with USB - Page 26

post #376 of 3034
Little-Endian,

It seems perfectly obvious to me that the giraffe is causing the data read errors. I'm no zoologist or anything, but he looks quite suspect to me

Elias
post #377 of 3034
@Elias

Haha, you're funny man!

Oh yes, I learned that many things can change the integrity of a S/PDIF stream. Even a nice giraffe has to be checked twice for meaning no harm. Unfornuately the ears are not very suited for headphones - that a pity, their hearing capability is probably far beyond of ours.

Ah, I've sent you a PM for the serial number ...
post #378 of 3034
mr. gwinn,

i am intrested in what improvement the upgraded output drivers provide. if the non usb unit had an impedance mismatch with an amplifier what would i be hearing to let me know there was a problem?

thanks,
music_man
post #379 of 3034
I have two questions.

Quote:
Originally Posted by EliasGwinn View Post
The Benchmark DAC1 USB runs in synchronous mode. The reason for this is that it lets the host (kmixer) operate at the original sample-rate of the audio being played at all times. If the kmixer is not forced to do any sample-rate conversion, it can maintain bit-transparent operation.
How is the time base for streaming determined in Windows? Is it from the PC's own frequency generator? Or is it like in some Linux drivers, where the software virtualizes the sound hardware buffer, and thus time base is derived from the sound hardware's own clock (the latter allows the hardware to be master)?

My second question is regarding this statement from the manual:
Quote:
Instead the converter oversampling-ratio is varied with extremely high precision to achieve the proper phase relationship to the reference clock.
How is that different from what an ASRC chip does?

Thanks.

PS
Quote:
This means you can drive 255 ft of cable @ 32 pf per foot
My cables are 4 pF per foot.
post #380 of 3034
Hey folks,

We are in the middle of some serious office re-arrangements....moving this person's office here, and this person's office there, and engineering test benches over there, etc, etc etc. The best part of all of this is I am able to put together a brand new listening room!!

Needless to say, I won't be able to answer your questions before the weekend, but I promise I'll be back on Monday or Tuesday and we'll pick it up then.

Thanks!
Elias
post #381 of 3034
Elias,

I was wondering if you could comment on Benchmark's decision not to include a power switch on the DAC1 and if it is indeed recommended to leave the unit on 24/7.

Thanks!
post #382 of 3034
Quote:
Originally Posted by The Monkey View Post
Elias,

I was wondering if you could comment on Benchmark's decision not to include a power switch on the DAC1 and if it is indeed recommended to leave the unit on 24/7.

Thanks!
I would assume that would be the case, since there is no power switch.. My VHP-1/XDAC3/X10-3 were the same way.. NP.
post #383 of 3034
i hope mr. gwinn gets back soon. i cannot drive the krell with the standard dac1. then i saw it says it needs a low impedance xlr connection. i don't know about high current though. i don't want to fry anything. the standard dac1 is all distorted at the +4db at 0dbfs that the krell takes for input. that would be the 20db pad. i don't want to order the usb unit if i it won't work either.

music_man
post #384 of 3034
Hey folks, hope everyone had a good weekend!

I will try to catch up on this thread throughout the day.

Music_Man, can you please describe your problem in detail, including the entire signal chain in question and what problems you are noticing?

Thanks,
Elias
post #385 of 3034
sony scd-1, into the benchmark with s/pdif, into krell kav400xi with xlr's. it is not actually distorted. the sound is much to "thin". it just lacks any "weight"(compared to the xlr's right out of the sony). plus hardly any bass. it is loud enough. also the left level is louder than the right level. i made another post about that but you can answer it here if you like. i know how to use a test tone and meter but i cannot do that at the moment. any other way to get the two sides even?

thank you,
music_man
post #386 of 3034
Quote:
Originally Posted by music_man View Post
mr. gwinn,

i am intrested in what improvement the upgraded output drivers provide. if the non usb unit had an impedance mismatch with an amplifier what would i be hearing to let me know there was a problem?

thanks,
music_man
Music_man,

If the amplifier has low input impedance, the output drivers of the DAC1-Classic will suffer from distortion because their is not enough current limiting. Also, as with any balanced input, low-impedance will cause the common-mode rejection to suffer (EMI, ground loops, and other noise). If the amplifier has high input capacitance, the band-width will suffer (high-frequency roll-off).

Thanks,
Elias
post #387 of 3034
Quote:
Originally Posted by Crowbar View Post
I have two questions.


How is the time base for streaming determined in Windows? Is it from the PC's own frequency generator? Or is it like in some Linux drivers, where the software virtualizes the sound hardware buffer, and thus time base is derived from the sound hardware's own clock (the latter allows the hardware to be master)?
For the DAC1 USB, the clocking is done natively in the computer. This mode of operation is called Asynchronous. This mode is prone to significant amounts of jitter. Thankfully, the DAC1 is immune to jitter because it utilizes a proprietary clocking system called UltraLock.

Quote:
Originally Posted by Crowbar View Post
My second question is regarding this statement from the manual:

How is that different from what an ASRC chip does?
It is, in fact, an ASRC chip that is used in the DAC1.

Thanks,
Elias
post #388 of 3034
Quote:
Originally Posted by music_man View Post
sony scd-1, into the benchmark with s/pdif, into krell kav400xi with xlr's. it is not actually distorted. the sound is much to "thin". it just lacks any "weight"(compared to the xlr's right out of the sony). plus hardly any bass. it is loud enough. also the left level is louder than the right level. i made another post about that but you can answer it here if you like. i know how to use a test tone and meter but i cannot do that at the moment. any other way to get the two sides even?

thank you,
music_man
Music_man,

I'm surprised by this. It could be that your DAC1 needs servicing, but its hard to say. Sometimes, when the DAC1 is A/B'd against other D/A sources (usually low quality), listeners sometimes say that the DAC1 seems to be "missing" low / mid range information. The reason for this is the absence of jitter artifacts. Converters that suffer from jitter distortion have an audible amount of extra low / mid-range information which is, in fact, jitter artifacts.

As for the issue with the balance between channels: Is the volume control low when you are experiencing this? The nature of a stereo pot is that they are inherently inaccurate in the first +/-25% of their range. Above that, the volume pot is extremely accurate between channels. If the volume gets too loud before you can turn the pot above the first 25%, the internal jumpers can adjust the range properly.

Please keep me informed about these issues. Also, let me know if you want us to take a look at it for you.

Thanks,
Elias
post #389 of 3034
it was the calibrated pots that were off about 1db. someone else answered how to level them with a voltmeter in my other thread. it's fixed now.

it does not seem "flat" simply in an a/b. if i listen to some other dacs they seem to have simply more "character". i know those are subjective terms.
i think it is just because the dac1 is supposed to present what is on the source with out adding anything. that is why i said i prefer to listen to other dacs for enjoyment and use the dac1 for monitoring.

that is not an insult, it is in fact a compliment. it is hard to make a product that does not add it's own sound to the signal chain. to be honest it seems the cheaper the dac the more color it adds. these are dacs with no jitter control. to some people this color is pleasent even though it means a technically inferior product. i know from a technical standpoint the dac1 is top of the line.

music_man
post #390 of 3034
Music_man,

Thank you for the kind words. I'm glad you were able to correct the balance issue as well.

It is true that, in an inexpensive design process, jitter usually is not addressed properly, if at all. The performance of these devices are usually very dependent on digital cables (length and quality), as well as source transmission.

I should say that this is not limited to inexpensive devices. We recently purchased another manufacturer's D-to-A device that has recently experienced a surge in popularity. (I won't mention the name of the manufacturer nor the device, as we refrain from publicly critiquing competitors. I will note that the device cost significantly more then the DAC1). We tested the device on the Audio Precision (AP) testing station, and we were shocked to find that this device had absolutely NO jitter attenuation!! In other words, if the AP induced the slightest amount of jitter on the digital signal, it rendered itself directly as distortion within the device. Here is a picture of the result of 0.25 UI of jitter at 44.1 kHz.



The spike in the middle represents the audio - the only signal that was meant to be heard. The spikes all around it represent jitter artifacts at frequencies ranging from 100 Hz to 9 kHz, in steps of 300 Hz (ie, 100, 400, 700, etc.). The magnitude of the jitter is a very small 0.25 UI.

Thanks,
Elias
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