[Xiode]

What's the benefit of having a dedicated source as opposed to a computer-as-source component?

Also, what's a good dedicated source for under $200 that would sound good coupled with an amp and Senn HD-555's? I sort of like the idea of having my music away from my computer.

 

[memepool]

Basically in the best case scenario that a specialist manufacturer with specific experience in building audio equipment has gone to the trouble of sourcing all the components, DVD drive, Chipset etc and played around with different combinations until they find the best formula where the sound quality is as good as it can be for the money, then packaged it in a nice case that won't look out of place in your living space.
A computer is a multitasking device full of noisy cooling fans whereas a cd player just plays CD's. Even if you spend a lot of time and money finding ultra quiet power supplies and using acoustic foam to deaden case resonances etc, at bottom you are still competeing against something which has been, in the best of breed, built from the ground up with 25 years experience just to perform this task.
Check out the Cambridge Audio range of players which can be had for around 250USD.

 

[Garbz]

aside from the noise and looks, good cdplayers provide task specific circuits. usually linear supplies, low jitter, minimalist design. There's nothing audio specific about a computer. The EMI emitted inside the case alone affects audio circuits, the ground plane is full of digital noise, and the powersupplies are eletronically and mechanically noisy and underpower the audio circuits.

So what's the benefit? It looks and sounds better

 

[jpr703]

You might want to keep an eye on Audiogon and see if you can pick up a used NAD player in that price range.

 

[jpr703]

I've never heard the Onix but it's got a pretty good rep for sound quality and modability. I believe it sells for about $300.

 

[sumone]

Quote:

Originally Posted by Garbz There's nothing audio specific about a computer. The EMI emitted inside the case alone affects audio circuits, the ground plane is full of digital noise, and the powersupplies are eletronically and mechanically noisy and underpower the audio circuits.

What if you were just using a soundcard's digital output? None of these factors would be a problem, correct? Cause doesn't bit-perfect mean it's the same bits from the hard-drive to the (outboard) DAC?

 

[shplorgh]

Quote:

Originally Posted by sumone What if you were just using a soundcard's digital output? None of these factors would be a problem, correct? Cause doesn't bit-perfect mean it's the same bits from the hard-drive to the (outboard) DAC?

There's still the issue of jitter.

 

[sumone]

Quote:

Originally Posted by hungrych There's still the issue of jitter.

But that's just on the way to the DAC. Everything before that is a 100% copy of what's on the hard drive (if say the soundcard doesn't resample or anything)? No noise or any other interference?

 

[Beauregard]

Quote:

Originally Posted by sumone But that's just on the way to the DAC. Everything before that is a 100% copy of what's on the hard drive (if say the soundcard doesn't resample or anything)? No noise or any other interference?

Simplistically, the interference, noise, etc. is the jitter in the digital output. The effect of the electronic garbage in the computer will be to cause timing errors (jitter) in the stream of electrical pulses that form the digital signal. In the worst case, these errors will be so severe that the DAC's receiver can't recognize parts of the signal; bits will be lost and one will hear dropouts, clicks, or other glitches. More commonly, the effect of jitter is manifested as artifacts in the reconstructed analog signal. This results in a a loss of low-level detail, problems with stereo imaging, and the like. If the soundcard is bit-perfect, all the music will be there - but it won't sound as good as it might from a less jittery source.

Because the output from a computer can have a lot of jitter, a jitter-resistant DAC is likely to give a noticeably higher level of sound quality than a DAC with less-than-optimal ability to handle upstream jitter before converting the digital datastream to an analog output. The current off-the-shelf DACs available at a halfway reasonable price that are virtually jitter-immune use an asynchronous sample-rate converter chip (ASRC) to oversample and reclock the incoming signal. Unlike traditional DACs that rely on clock information contained in the incoming bitstream, the resampled signal that is passed to the digital-analog conversion chip is referenced to an internal clock and is not affected by timing errors that may have existed in the incoming signal. DACs that fall in this category include the Benchmark DAC1, Bel Canto DAC2, and the new HeadRoom Max DAC.

Best,
Beau

 

[mulveling]

Quote:

Originally Posted by Beauregard Because the output from a computer can have a lot of jitter, a jitter-resistant DAC is likely to give a noticeably higher level of sound quality than a DAC with less-than-optimal ability to handle upstream jitter before converting the digital datastream to an analog output. The current off-the-shelf DACs available at a halfway reasonable price that are virtually jitter-immune use an asynchronous sample-rate converter chip (ASRC) to oversample and reclock the incoming signal. Unlike traditional DACs that rely on clock information contained in the incoming bitstream, the resampled signal that is passed to the digital-analog conversion chip is referenced to an internal clock and is not affected by timing errors that may have existed in the incoming signal. DACs that fall in this category include the Benchmark DAC1, Bel Canto DAC2, and the new HeadRoom Max DAC. Best, Beau

Actually, DACs like the DAC2 and DAC1 still have to synchronize to the signal clock - they are effective at attenuating jitter, though. Something like the Chord DAC64, which buffers all incoming data and tosses the signal clock in favor of its local clock, is truly independant of transport/interface jitter (unless it's great enough to cause data errors) - though the consequence is that it does run a slight risk of buffer overrun/underrun over a long period of time.

 

[shplorgh]

Quote:

Originally Posted by mulveling Actually, DACs like the DAC2 and DAC1 still have to synchronize to the signal clock - they are effective at attenuating jitter, though. Something like the Chord DAC64, which buffers all incoming data and tosses the signal clock in favor of its local clock, is truly independant of transport/interface jitter (unless it's great enough to cause data errors) - though the consequence is that it does run a slight risk of buffer overrun/underrun over a long period of time.

What confuses me is that Chord still makes a seperate cd transport to go with it's DAC!

 

[Beauregard]

Quote:

Originally Posted by mulveling Actually, DACs like the DAC2 and DAC1 still have to synchronize to the signal clock - they are effective at attenuating jitter, though. Something like the Chord DAC64, which buffers all incoming data and tosses the signal clock in favor of its local clock, is truly independant of transport/interface jitter (unless it's great enough to cause data errors) - though the consequence is that it does run a slight risk of buffer overrun/underrun over a long period of time.

I didn't include the DAC64 in my discussion because, at US$3000, it didn't meet my halfway-reasonably-priced criterion

. Seriously... I don't claim sophistication in this stuff; I'm very much a layperson who's trying to understand the complexities of digital audio - simply but accurately.

After seeing your post, I had a look at the Stereophile review of the DAC64; in the introduction, Atkinson states: Quote:

I referred above to the RAM buffer. This is basically arranged as a FIFO (First-In, First Out) store. In theory, the clock accuracy with which the data are clocked into the FIFO doesn't matter, as the data are clocked out with a high-precision local crystal, which in turn should reduce jitter to vanishingly low levels. In practice, there has to be some means of locking the local clock to the long-term-averaged clock of the incoming data, which will mean low-frequency jitter might still propagate to the DAC chip. ...

(emphasis added)

My understanding is that any external DAC using an SPDIF interface needs to synchronize with the incoming clock data simply to be able to read the signal. Writeups about the Benchmark DAC1 have described the attention to PCB design required to isolate the effect of the incoming jittery clock from the resampled internally clocked signal that the ASRC sends to the DAC chip. Is there theoretically any difference in the concept of isolating the incoming clock from the internally clocked bitstream depending on whether an ASRC or buffer is used as the intermediary between the two?

Rereading my post, it seems like I was overstating to say, in regard to ASRC DACs, that "the resampled signal... is not affected by timing errors that may have existed in the original signal." But I wonder about your statement, "Something like the Chord DAC64, which buffers all incoming data and tosses the signal clock in favor of its local clock, is truly independant of transport/interface jitter..." Is there some way that the buffer-based design of the DAC64 is "truly independent" in a way that ASRC solutions aren't? Or is it perhaps more correct to say that the extent to which either successfully create new signals that are independent of transport/interface jitter depends on how well the engineers have accomplished the isolation from the input clock?

FWIW, given my limited ability to understand such, it appears that the DAC1 and DAC64 perform very similarly on the Stereophile jitter measurements.

Always learning...

Best,
Beau

 

[gaillard]

i don't understand, surely bits arn't lost when sending over usb to an externel dac correct? is this just spdif that we are talking here... because that would make more sense