[freeflier]

Quote:

Originally Posted by music_man great "revised" review, ferbose! you also must realise that the distortion you hear is neither the dac1 or the cs. it is the k701. as good as they are, they have a major shortcoming. certain transients turn them into $5 headphones for a quick moment sometimes. i have a certain live recording of paul simon's "50 ways to leave your lover" that simply destroys the k701's. whereas, pretty much any other headphones or loudspeakers can pull it off. even though the distortion is in the recording. music_man

Hmmm,
I really don't consider the K701s to be the source of the problem. They are certainly completely unforgiving but they don't "amplify" distortion relative to the overall sonic picture any more than they "amplify/reveal" all the subtle details of the recording. In my opinion they are just doing their job with no excuses and certainly don't "fall apart" under certain conditions as you've suggested (at least not in my experience). That's not to say they are perfect but I have trouble shot the problems I described very carefully and the AKGs aren't the source. I can back that statement up with some very simple trouble shooting techniques.

ie: 90% of the occassional "distortion" I perceive is completely eliminated when I plug into the Dac1 (and is usually reavealed to be a difficult to reproduce complex harmonic). Actualy I'm oversimplifying, what can sound like distortion can be complex harmonics associated with some instruments (particularly piano) but just as often is is result of mics that have been placed too close which pic up unwanted ringing/ resonances. The Dac1/k701 combination invariably and clearly reveals the problem for what it is. The CS and other lesser digital just makes an undistinguishable mess (though with good power and an 88 hz/ 24 bit data stream this happens less often with the CS). The few times I continue to get distortion throught the Dac1 I also get it through my Thiel 2.4 speakers (which are VERY neutral, but naturaly not quite as resolving as good headphones ). In other words I have yet to hear distortion on the AKGs that I can't hear through my speaker system or with my Shure E500 IEMs. It's just a little more obvious with the AKGs (as is everything). Also the fact that I hear occcassional distrortion regardless of source or transducer doesn't surprise me considering some of the garbage that gets past some engineers. ie the distortion that starts at 1:20 of track 4 of Sade's "Lovers Rock" is outrageous as is the entire distorted track 10 of Dido's "No Angel". (both are audible on my cheapo car system made up of a Panasonic DVD player and Infinity coax speakers). I think we are simply at the classic dilema: Do you really want to know what's on the recording or not. If not , the only way around it is with subtractive coloration. My solution to the problem is not "forgiving" speakers or headphones, it is to get the very best source processing that I can and build an audio chain that adds as little of it's own garbage as possible. I call it "EXTREME CLINICAL"!

 

[Ferbose]

Quote:

Originally Posted by freeflier Ferbose, I don't know if I've brought this up before but there is a common misconception out there regarding the Dac1 "direct" out. There is no way to bypass the volume pot in the Dac1. If you use the "variable" output you go through a volume pot controled at the front of the dac by the large volume knob. If you use the "direct" out you go through a duplicate pair of the EXACT same pots that are controled by the trimmers at the back of the Dac. The only difference is that the fixed out is meant to be set (calibrated) for an optimum output level for a given application and left at that setting. That is why the only way to access the trimmers is with a screwdriver (you can verify this with Benchmark). (You do realize that the size of the big volume control knob at the front of the Dac1 compared to the tiny screw type trimmer controls is completely irrelevant to the actual electronics?) Even though the EXACT same components are in both signal paths some people still claim there is a difference. For the most part I think this difference is largely imagined but I admit that, even knowing what I know, I also found that the "direct" out sounds "slightly" cleaner. In discussions over at Audio Asylum the concensus amoungst people who know how the dac is configured is that the signal path is routed slightly differently in the "direct" out and is thus less susceptable to noise. But be clear about this, there is no way to bypass the volume pots in the Dac1. When you throw the CS into the chain you are actually going through two sets of volume controls (the Dac1's and the CS). I seriously doubt that you are going to improve on tiny differences in the Dac1 one's output (which results from a "slightly" different signal path routing) by adding an extra set of cables , connectors and a bunch of passive components in the CS. At best you are going to get the origonal sound you would have gotten out of the Dac1 alone. Be carefull that you aren't misinterpreting the subtractive/additive colorations (that are an inevitable result of an extra set of cables connectors and passive components) for "improvement". This is an inherant danger in subjective sound quality evaluation. Ps As I noted on the origonal thread, I don't find the "cue" path better. Maybe a tad "smoother" but to my ears that's a result of the subtractive coloration of an extra volume pot. There is NO WAY that the CUE path could be better.

I know that DAC1's fixed and variable outputs both go through volume pots. One is a helipot and the other is the volume knob, but when I use fixed output I set helipot at no attenuation. When I use variable output I use volume knob to attenuate the signal. If I don't want to attenuate I will use fixed output. So the two outputs have the same circuit topology, but they are generally used differently, and therefore can sound different. And why do I want to attenuate? Because DAC1's output voltage is so high that my amps can't really handle it. Therefore I use CS to attenuate instead of DAC1, and to me it is better to use CS to attenuate, although the difference might be ever so slight.

As for Cue vs Main signal pathway in CS, I thought Cue should sound worse because of an extra volume pot, but it does not. An extra pot also means an extra stage of amplification, which could increase the voltage or current gain. It is not impossible that the extra amplification stage could provide a better control of headphone's transducers. It is just my speculation.

 

[freeflier]

Quote:

Originally Posted by Ferbose . And why do I want to attenuate? Because DAC1's output voltage is so high that my amps can't really handle it.

Absolutely!!!!!!!!!!!!
That's one of the reasons I like the Dac1. Let's face it, most digital sources are too "hot" and only require attenuation. I use the trim pots to attenuate the output so that I am at the maximum comfortable listening level with my Tact room correction processor/preamp at 0db. I then control my volume over a narrow 10db range of completely transparent digital attenuation from the Tact (remotely). (Note that the Tact has 32 bits of internal resolution and the Dac1 is about 20 bits (in practical terms) meaning 24 db of attenuation can be done without harm. Since the room correction filters employ 12 db of correction I figure I can get away with about 10db of volume control safely). One of the reasons I may consider using the CS in my main system will be because I can do essentially the same thing as I do with the Dac1. ie Tune the output to the appropriate level so that I can make the best use of the Tact "preamp".

 

[Ferbose]

Quote:

Originally Posted by freeflier Note that the Tact has 32 bits of internal resolution and the Dac1 is about 20 bits (in practical terms) meaning 24 db of attenuation can be done without harm. Since the room correction filters employ 12 db of correction I figure I can get away with about 10db of volume control safely). One of the reasons I may consider using the CS in my main system will be because I can do essentially the same thing as I do with the Dac1. ie Tune the output to the appropriate level so that I can make the best use of the Tact "preamp".

This is off-topic but your math puzzles me.
So Tact takes analog in and do A/D into something like 24 bit signal and does computation at 32-bit. Actually due to thermal and electrical noises the ADC probably self-dithers at 20 bit, not unlike a DAC. The internal computation can be 32 bit (or on new DAWs in 48-bit fixed or 64-bit floating, I think), but the real resolution is still around 20 bit. So my guess is that you want to attenuate DAC1 so that 0 dBFS signal is output at a voltage that Tact A/D converts to 0 dBFS. This will maximize the resolution in your system, I suppose.

 

[freeflier]

Quote:

Originally Posted by Ferbose This is off-topic but your math puzzles me. So Tact takes analog in and do A/D into something like 24 bit signal and does computation at 32-bit. Actually due to thermal and electrical noises the ADC probably self-dithers at 20 bit, not unlike a DAC. The internal computation can be 32 bit (or on new DAWs in 48-bit fixed or 64-bit floating, I think), but the real resolution is still around 20 bit. So my guess is that you want to attenuate DAC1 so that 0 dBFS signal is output at a voltage that Tact A/D converts to 0 dBFS. This will maximize the resolution in your system, I suppose.

Well,
First of all I don't use the AD technology that the Tact is capable of. I feed it digitatal only. Also I don't think you are getting the chain I employ. The Dac1 comes last in the digital chain and goes directly into my power amp (PC>Squeezebox>Apogee Big Ben> Tact 2.0s> Dac1> Bryston 4bsst power amp> Theil 2.4s). I think the thing that is bothering you is the fact that I'm modulating the digital signal at the Tact before I feed into the Dac1. I get away with this because the Tact outputs a claimed 24bits. You are right that the theoretical resolution does not always translate directly so I assume the worst possible scenerio. In other words the Dac 1 claiims 24 bits but has been tested at a practical 20 bits. The Tact claims 32 bits and utiilizes bit shifting algorithms for attenuation outputting a claimed 24bits. I assume it's capable of at least a practical 20 bits so, assuming a 16 bit source, that leaves me 4 bits to play with safely. ie 24 db. That means I set the Dac1 output into my amp for an appropriate maximum SPL level when the Tact is at full 0db output. The Tact output is realative to a fully modulated source and has no relationship to the output voltage. I don't quite understand the different db reference standards used and the dbfs measurement used in the Dac1 specs is (I beleive) referenced to output voltage (I'm very vague on this but I think this scale is used to allow studio gear to be matched to a standard reference). Anyway the issue is moot in my application. All I need is to know is when I'm outputing a fully moduted signal from the Tact and how to match the outputs of the Dac1 (with a good meter). I don't even pay attention to the actual output voltage. I just set dac 1 with the volume control, take a read off my meter and then set the fixed outputs to this level.

 

[freeflier]

Oops,
posted in error. I wish there was a way to delete posts here.

 

[manstretch]

I'm thinking of getting one of these to replace my Entech NC and Little Dot 2+ with my (coming) HD650's. I listen to mainly 256k+ bitrate mp3's... is something like this going to benefit me?

 

[freeflier]

As far as I'm concerned full rez CD (44/16) is, at best, barely adequate and worrying about the quality of Dac if you are going to feed it even worse garbage (compessed files) doesn't make a lot of sense.

 

[manstretch]

Quote:

Originally Posted by freeflier As far as I'm concerned full rez CD (44/16) is, at best, barely adequate and worrying about the quality of Dac if you are going to feed it even worse garbage (compessed files) doesn't make a lot of sense.

Well, what I meant was... relative to what I have now, would this be a noticeable improvement? I don't ALWAYS use compressed files, but even when I would be using pure redbook audio, it would be from a computer cd-rom drive. Would you still say this would be a worthless upgrade?

 

[freeflier]

Quote:

Originally Posted by manstretch but even when I would be using pure redbook audio, it would be from a computer cd-rom drive. Would you still say this would be a worthless upgrade?

I'd say this would be an improvement if you have a decent sound card with a low jitter SPDIF output. But, if you are going to run straight out of computer and really want to see some serious improvement in sound quality I would seriously consider the Dac1 (since it is realatively immune to jitter and noise). The Presonus works best only under fairly ideal conditions (which you need to have the equipment to create) while the Dac1 is much more forgiving under a broader range of situations. In addition the Dac1 is better than the even the best case scenerio of the Cental Station (in my experience).

 

[Ferbose]

Quote:

Originally Posted by manstretch Well, what I meant was... relative to what I have now, would this be a noticeable improvement? I don't ALWAYS use compressed files, but even when I would be using pure redbook audio, it would be from a computer cd-rom drive. Would you still say this would be a worthless upgrade?

If you get the CS you could compare it to Entech Number Cruncher and Little Dot. I think there is a good chance that CS can replace both. Then you can sell them off and save money. I would guess that Transit probably has OK output jitter, so connecting CS to it is not a problem, at least I tried it and it worked fine. Since you are listening to lossy compression for the most part, jitter is the least of your concerns. IMO jitter was often overlooked a few years ago but now it's a hip term that people throw around. Jitter is generally measured in its quantity but in fact it's like noise in the analog domain, and it's the quailty that matters. Just like THD tells you little about perceived distortion, total jitter measurement (expressed in picoseconds) tells you little about the sound quality. Jitter has a signal-dependent spectrum too, but its nature is poorly understood. But at least the sonic effect of jitter is always subtle.

 

[freeflier]

Quote:

Originally Posted by Ferbose IMO jitter was often overlooked a few years ago

That's true but I'm not sure what your point is. I doubt that it's a coincidence that in the last 5 years since jittter has become a widely accepted phenomenon there have been huge improvements in real world priced dacs. Also, digital coloration/distortion is not like analog distrortion/coloration which generaly manifests at the frequency extremes and is easy to detect. The fact that the differences between digital processors are harder to identify makes them no less problematic. I find the "subtle" differences in dacs to have a profound effect on listening enjoyement over the long term.

 

[mikeliao]

Next mod for your CS! Build your own power supply.

I had the same problem as a previous poster who couldn't get 220V power supply. It exists but you just can't buy them from Presonus USA. So...

Electronics shop...2 16V transformers, 1 9v (I think, I don't remember, maybe it was 5V), and a 5 pin female XLR. Wire it up and more power to you.



It sounds the same as the 110V brick on a step up transformer. But then, my building has very clean power.

 

[5ergo]

Well i really haven’t noticed the 2 threads about presonus CS. I own one too for more than a year now and here are my thoughts:

-it has a quality dac and so but i think the rest is cheap. I had 4 of those at home because every time i pushed the B (speakers output) button i heard electricity sound like “bzzzzzz”. First i though my monitors had some grounding issue but that wasn’t it. It was the CS and only 1 of those 4 didn’t have the issue. I chose the one and told the seller about it. It was really annoying sound.

-it somehow isn’t compatible with NAD C542 CD player via digital out. I tried both coax and spdif and i still get peak, the real one, i almost damaged my monitors. It works well with other CD players like Teac….

-I really miss the on/off button as I have to unplug it every night.

-I figured out with TC sound help that you need an additional component to connect a sub with mono input.

-Is has totally passive headphone out, i don’t see any headphone amp. If you turn phones vol to max it will be no louder that the source (soundcard for instance).

However i don’t have any power issues as here in Europe we use 220V only.

I have it connected to:
A pair of dynaudio BM6A
Audigy2 (via analog in) for games
Revolution 7.a (via spdif) for movies and mp3
NAD C542 CD player (via AUX but would prefer optical in)

How do you notice jitter while playing back music for instance?
Is emu 0404 DAC any near presonus dac?

 

[music_man]

5ergo, do you realise everything you just said made no sense whatsoever?
lol.

music_man