Special announcement from Ack! Industries

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csown

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Special announcement from Ack! Industries
« on: 15 Jan 2005, 12:35 am »
Friends and customers,

        We are pleased to announce the release of dAck! 2.0!  Details and
pricing information are available at the following URL:

        http://ack-industries.com/doc/2.0_release.pdf

Sincerely,

Christopher S. Own
  ______________________________________
 Ack! Industries | ack@ack-industries.com
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JLM

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« Reply #1 on: 15 Jan 2005, 10:09 pm »
Chris,

Sounds wonderful.  Thanks for listening to customers.  As interested as I am with having equipment modded, this is better with significant upgrades from the manufacturer at a reasonable price.

CEC transports are mentioned again.  They have a very good reputation, but with limited availablity why voice to them?  Will my Sony DVP-S7700 be a synergistic match?  Whats your opinion of modding it?

My speakers are Bob Brines FTA-2000s (single driver full range transmission line design using a Fostex F200A driver).  Would you recommend the upgrade?  http://www.geocities.com/rbrines1/

Will you still offer the original (as there will be a price gap between the two after the introductory period)?

Keep up the good work!

TIA for your reply.

doug s.

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« Reply #2 on: 16 Jan 2005, 06:00 am »
hi chris,

coupla questions -

please explain the "ultra-resolution" option more.  you say most systems wont need it, inquire for compatibility.  does this mean it will actually perform *worse* unless ya need it?

and, re: the hi-woltage option.  text sez that it increases output to 2v, but specs say 2v is standard, & option increases voltage to 4v...

thanks,

doug s.

csown

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« Reply #3 on: 16 Jan 2005, 06:42 pm »
Hi guys,

Thanks for your questions guys.  We actually did not use the C.E.C. to voice the original dAck! 1.0.  That was done with a series of different transports and later with the C.E.C. during refinements.  The performance of a transport depends on two major factors, RMS jitter and the jitter spectrum that defines it.  While it is easy to measure RMS, the spectrum is the much more important variable.  The C.E.C. is spectacular in this regard.  We have found that the dAck!'s performance scales directly with datastream quality - if you feed it a poor datastream you will hear the source, if you supply an excellent datastream you will get spectacular results.  We use the C.E.C. as a reference and recommend it because it exhibits excellent synergy with the dAck! but of course there are several other transports that perform excellently as well.  Also do not downplay the significance of the digital cable in this analysis as it affects the spectrum also.

An interesting article on this was written by Robert Harley in the early 1990's (link here http://www.stereophile.com/reference/368/index.html).  On page 4 is a listing of several very good and also very poor transports - the only metric here is RMS but on the following pages some of the spectra are shown.  Some of the biggest raves from customers (who have never seen this article, BTW) come from the ones who own products with the lowest figures shown.  I have personally heard the dAck! with the CEC, the PS Audio, the Proceed, the Delta, and Meridian products and they have been consistently good (in different ways).  With poorer transports such as cheap DVD players, the results will vary tremendously and on average bad; the Tosh SD3950 is an exception but its soundstaging performance and microdetail is not great.  For entry level I would really suggest the use of the Delta which offers a good base level of performance from the dAck! at a *very* affordable used price (note: CAL CD players are poor transports).  I have been very impressed by Meridian's digital engineering and they are a very widely available player, albeit quite expensive.

I have not had experience with your Sony but I think some customers have tried it.  You might want to ask around on the forums to see if there is cumulative experience with it.  I am not a big proponent of transport mods because they are often hit or miss - the jitter signature is usually well defined by the transport mechanism and there is not always a systematic way of improving it especially because of the big synergy issue with transport, cable, and DAC.  It is sometimes very worth it, other times not but they're usually non-refundable :).

Regarding your loudspeakers, I have never heard them and speaker design is not my expertise so I can't advise you there, sorry.  If the designer thinks the upgrade is good, and you like the sound, it is probably worth doing if you have the cash lying around.

We are not immediately discontinuing the 1.0; they will continue selling until supplies run out.  We are intending to return to our roots by offering the DIY community populated and kit 1.0 PCB's that they can use in their own projects (2.0 will only be sold as the complete converter system).  These PCBs will be available in February sometime.


Doug: The ultra-rez option is something I'm not too actively pushing because it depends on your system and your endurance for break-in.  The stock Auricap capacitors used for coupling take 100-120hr to run in but they don't change a great deal over this break-in period.  The teflon caps used in the ultra-rez are strange beasts - they sound great at first, then take several steps backward for a couple hundred hours, then they make a 180° turnaround and finally settle in to offer great balanced sound.  The bottom line here is that you won't know if you really like them until after our guarantee period so you have to be patient because they are essentially a non-returnable item.  They can always be removed, of course, but at additional cost.

The reason for soliciting inquiry is because not all systems will need it.  The 2.0 has enhanced high frequency interpretability so it will already be a departure from the original 1.0 sound... I think it may be enough for most 1.0 owners making the switch to 2.0.  With the hi-rez option you will hear enhanced depth in the highs and a new midrange palpability and involvement, but some systems (especially ones with poor transports, see above) can emphasize too much of a good thing and knock off the overall balance.  I think they will be most useful in systems with midrange emphasis such as SETs with horns.  As always, you have to try to know.  It might be advantageous to try the basic 2.0 first and then add them later if you want more.  Incidentally, this option will be just as good, if not better, of an upgrade for 1.0 dAck!s.

Regarding output level, the original dAck! was set at 0.7VRMS (2V peak-to-peak).  This was the standard for phono preamplifiers and is what most volume potentiometers are designed for.  They sound more linear, have less noise, and have better channel matching at higher settings than lower ones.  The high-output option yields 2VRMS (~5V peak-to-peak, 4 is a typo, sorry), which is the standard for digital electronics nowadays.  It can help systems with passive volume controls that can use the extra 7dB to play louder.  But in most systems the best performance and least noise generated at the volume pot will be found with the standard 0.7VRMS.

-Chris.

giorgino1

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« Reply #4 on: 17 Jan 2005, 08:57 pm »
Hi

I have in the past tried to find a phone number on your wesite to make enquiries about your product - unfortunately, I've been unsuccessful. Would be grateful if you can provide a number I can contact for technical questions.

Regards

George

PS - I have emailed you in the past but did not get a response - and so I bought a Scott Nixon DAC and then a Musical Fidelity TriVista 21 instead :D  However, I will consider your product in the future if I can talk to someone.

csown

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« Reply #5 on: 17 Jan 2005, 10:02 pm »
Hello George,

Thanks for contacting me again and for maintaining a modicum of interest in Ack! even after failed attempts at contact.  My response statistics are better than 99% by 1 week, but sometimes e-mails don't get past my spam filter or they get lost (usually rare) when I am inundated.  I do things other than Ack! by day so I cannot answer calls; usually I will call you back if you give me a number unless I have a lot of research-related deadlines.  E-mail tends to be more reliable but I do apologize for missing your mail.  

I have PM-ed my phone number to you if you would like to try me again.

Best,

-Chris

Horizons

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« Reply #6 on: 17 Jan 2005, 11:00 pm »
Chris:

Does the model 2 still use the 1545A DAC and Burr Brown opamps?

csown

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« Reply #7 on: 18 Jan 2005, 06:13 am »
The 2.0 still uses the 1545A, which is the refined version of the 1545 (both now discontinued but we have ample supply).  We use it because it's an excellent performer and has extremely low power consumption, perfect for this application.  Its performance is more like the classic sophisticated 1541 than the 1543 which is a rawer sounding boogie chip.  The output stage has been redesigned in the 2.0, however, to yield much enhanced immediacy yet it keeps a smooth, liquid midrange that is easy to enjoy.  The active device in the transconductance stage is by Analog Devices, the very low noise and very refined 8610.

-Chris

doug s.

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« Reply #8 on: 18 Jan 2005, 02:20 pm »
Quote from: csown
...Regarding output level, the original dAck! was set at 0.7VRMS (2V peak-to-peak). This was the standard for phono preamplifiers and is what most volume potentiometers are designed for. They sound more linear, have less noise, and have better channel matching at higher settings than lower ones. The high-output option yields 2VRMS (~5V peak-to-peak, 4 is a typo, sorry), which is the standard for digital electronics nowadays. It can help systems with passive volume controls that can use the extra 7dB to play louder. But in most systems the best performance and least noise generated at the volume pot will be found with the standard 0.7VRMS. ...

hi chris,

so, if i am understanding this correctly, your hi-output iteration is in fact the home-audio industry standard, & your standard has considerably lower output than what's considered normal?  yust wanting to make sure i know which to choose, if i decide to bite...

regards,

doug s.

csown

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« Reply #9 on: 18 Jan 2005, 03:17 pm »
Doug,

Home audio is actually largely standard-less, but in general sources nowadays have been popping up with higher output because the noise levels are very high in digital audio. However, a lot of high-end audio has some catching up to do - the vast majority of preamplifiers have an older industry standard of relatively high gain because vinyl sources had lower output. Thus about 90% of products out there now sound loud by the 9 o'clock setting and the signal is attenuated by a great deal - what that means is that the noise problem is now shifted inside the preamplifier where *hopefully* everything is shielded and no digital signals are present to couple noise into the squashed-down signals inside the preamp.

With the dAck!, we are able to think outside the box. Because of non-oversampling and the battery supply, our noise figures are lower by more than a factor of ten over high-end power supplies (in some cases almost a factor of 100) so decreasing the output level by about a factor of two does not hurt anything from a noise standpoint. It does gain a few things because a) the standard dAck! is at the same signal level as most phonostages so the system will be operating in a similar regime between digital and analog sources, b) a standard volume pot performs best at higher settings (past 12 o'clock), and c) there is less possibility for noise pickup inside the preamp. The net result is that in the best case you will get about one bit of extra resolution out of your DAC with the low output version. So as long as your system is loud enough, there will be better power transfer and the signal does not get that small anywhere in the system where it would be sensitive to noise.

Now the one case where this doesn't apply is with passive preamplifiers, which would prefer higher output because there's nothing anything to buffer the (usually quite low) signal after the volume control. This means that there is less gain in the system overall, the oomph of an active preamp is missing in favor of enhanced transparency, and because of high output Z are more susceptible to noise on the way to power amp. So in order to get better results and allow these passive preamp systems to play louder, we have the high output version. It is also a tad less sensitive to cable type. Additionally, if you have a digital only system, this might be the way to go because it's the industry standard output level for digital devices and your system can operate in a consistent regime at all times.

-Chris

(edited to correct a typo)

doug s.

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« Reply #10 on: 18 Jan 2005, 03:22 pm »
Quote from: csown
Doug,

Home audio is actually largely standard-less, but in general sources nowadays have been popping up with higher output because the noise levels are very high in digital audio. However, a lot of high-end audio has some catching up to do - the vast majority of preamplifiers have an older industry standard of relatively high gain because vinyl sources had lower output. Thus about 90% of products out there now sound loud by the 9 o'clock setting and the signal is attenuated by a great deal - what that  ...

thanks for the clarification, chris.  i want the unit w/the switch (completely transparent, of course!) for the two different gain options!   :mrgreen:

regards,

doug s.

JohnnyLightOn

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« Reply #11 on: 19 Jan 2005, 12:49 am »
A jitter question:  

My transport is a hard disk on a PC with no noise reduction techniques inplemented, then transmitted via a wired Squeezebox to my living room.  Software on my computer (SlimServer) converts the data from FLAC to PCM and sends it out the computer, through my router, and to my Squeezebox.  I listen to rock and other music which may not be recorded as well as some classical and jazz, and I am attracted to the dAck! 2.0 because I suspect my recordings will be more listenable with it.

Certain DACs (ex: Bel Canto) have a built-in jitter-reduction stage.  If I used the dAck 2.0, would jitter from my transport system degrade the sound?  Your site suggests a Monarchy Audio jitter reducer for use with the 1.0 dAck!.   Would you still recommend such a jitter reducer for the 2.0 for someone with my setup?   Thanks.  :)

csown

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« Reply #12 on: 19 Jan 2005, 07:46 pm »
Johnny,

The input of the dAck! 2.0 is fully isolated so the ugly noise from your PC will not get through, just the data bits plus whatever clock jitter there is (including the noise-induced stuff that's can't be removed up until the dAck!'s input). The dAck! has jitter reduction circuitry in in the form of a tight PLL at the receiver. Nevertheless, we want to be honest with our customers about it - it does have some dependence on the datastream quality, hence the dAck! is fairly sensitivity to transport quality.

Pretty much every converter on the market has some form of jitter reduction circuitry except the ones built into to the CDP's. Jitter reducers are not made equal, and as many have seen (e.g., Benchmark, Monarchy, Bel Canto, etc...), proclaiming that the product is immune to jitter does not mean that it is truly insensitive to it. Benchmark's internal filter technique is one of the more agressive solutions out there, but there are others that are quite good as well. The Monarchy is quite useful, and gives improvement up to a point; driving the Monarchy with a C.E.C. TL-1X is actually detrimental to the dAck!'s performance however. Nevertheless, with the Monarchy most sources are made much more palatable. I'd predict that it'd do the same for the Squeezebox though inside the Squeezebox is definitely less brutal environment than a PC. You'll have to try it in your system to be sure; I've not had direct experience with the Squeezebox in my system so it might turn out to be just fine.

What'd be really neat would be mods of the Squeezebox to improve its digital datastream performance. It's an amazing product from an ergonomics standpoint.

-Chris

OBF

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« Reply #13 on: 19 Jan 2005, 09:30 pm »
Just so I understand the conversation, are we assuming the squeezebox is adding its own jitter that wasn't present from the computer's hard drive output?  Or is it related to converting from one format to another?

I'm asking because I thought one of the advantages to PC storage is less jitter and I'm wondering if an intermediary like the squeezebox takes away some of that advantage compared to using a high end soundcard???

JohnnyLightOn

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« Reply #14 on: 19 Jan 2005, 09:54 pm »
Thanks  for the detailed reply, Chris.  I think I will choose my DAC first on its own merits, and then see if a jitter reducer is necessary after I have it in my system.

OBF, I don't understand jitter, but I assume the Squeezebox unit itself just passes through the digital signal to its digital outs.  But for all I know that can add jitter.

Here's a link to an interesting post on the Squeezebox mailing list: http://lists.slimdevices.com/archives/discuss/attachments/20040930/77c7fb50/attachment.html.  The formatting is messed up, but if you can sort through it (I cut and pasted it into my word processor) it talks about how the poster benefitted from a jitter reducer for his Squeezebox.

csown

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« Reply #15 on: 20 Jan 2005, 02:53 am »
OBF and Johnny:

Jitter is a very complex and sometimes very scary phenomenon that isn't completely understood.  No clock is perfect - all are jittery to some degree, and transistors can only switch so fast, hence no datastream is perfectly timed because they must derive from *some* real clock in the system.  Once you accept this then you realize that we are simply comparing the amount of jitter between devices and there is no such thing as a jitter-free product.   Some important things to take away from this discussion are these key points:

1. The term "bits is bits" is both true and false.  

It is true because pretty much all equipment is bit perfect; most gear will give you a bit perfect copy of the source media unless it's resampled like in a lot of PC's (I'm pretty sure the squeezebox is bit transparent to source data).  The term bits is bits is false because in S/PDIF we are also encoding the clock within the datastream.  Engineers generally don't talk about bit errors anymore, we just care about how good the clock is.

2. Computer sources are bit perfect but are only as good as a) their digital output driver, b) the reference clock that provides the encapsulated clock signal, and c) the system's noise performance.

Because of this, saying "HDD sources" are jitter free and necessarily better than CD sources, etc. is as incorrect as saying the earth is flat :).  Everything is jittery to a degree and it can be additive or even compounding as the signal travels through the system.  It can also be squashed by a jitter reducer (or the jitter reducer can add more jitter, ironically).  This is also why sometimes you get worse sound if you "upgrade" to an outboard DAC from a CDP - here, its digital output is too poor.  (Incidentally, some CDP manufacturers intentionally screw up the digital output to force you to stick with their one-box players; talk about false advertising!)

3. Jitter affects non-intuitiveaspects of the sound.  

Jitter is a nonlinear phenomenon and is related to audio through a rather complex transformation of variables.  Instead of thinking about it in terms of highs, mids, lows, you have to start thinking about spatiality, image localization, tonal purity, and musical interpretability.  These are abstract, non-intuitive, and above all else impossible to measure (!).

4.  It is very hard to predict what changing the jitter signature does to the sound.

You can no longer say definitive things like "silver cables tend to be bright" and "A mosfet amplifier tends to sound warmer and softer than BJT's".  You have to start saying things RF engineers talk about like, "this cable preserves full bandwidth end to end so I'll probably get excellent resolution", and "changing the dielectric to X material affects this type of jitter spectrum adversely so I get soundstage collapse most of the time".

5. You can either "fix" the clock or replace it, but you can never have a perfect clock.  

This is what all this jitter rejection stuff is about.  Most DACs fix the clock provided within the datastream through the use of a PLL and low pass filter.  This is the traditional way it's done but the results can be very, very good.  Others like the Benchmark take the data sent in one clock domain and resamples it into a new clock domain.  Another technique is to buffer the data then clock it out later using a better clock.  The most extreme example of replacing a clock is what the SB does: replaces the computer's clock by sending the data over a line and buffering it, then outputting it at its leisure.  This is extreme because the playback device can tell the computer, "hey, slow down, I'm not ready to process this yet", or "give me a lot of samples, I'm running low here."  The two clocks are truly independent then, just too bad the Squeezebox has a consumer grade clock because it has a lot of potential.

These are just different ways of doing it.  By no means is one always better than another, it depends on implementation.


So, Johnny, in relation to using the SB with the dAck!, I think you have the right approach.  Keeping in mind that the dAck! is capable of an extraordinary level of performance, give it the best signal you can possibly give it and it will shine.  If you can't, it will still try to do its best with it.  If it's the right mediocre signal (not too shabby a jitter spectrum, e.g. Toshiba SD-3950), it will sound great still but of course could be better.  Because of 4. above, you won't know until you try.

Hrmmm... this message ended up being longer than I expected .  I hope it is educational - let me know if you need clarification.

-Chris

TheChairGuy

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« Reply #16 on: 20 Jan 2005, 05:01 am »
Quote from: csown
Hrmmm... this message ended up being longer than I expected . I hope it is educational - let me know if you need clarification


I hope you post every day for the next three years with long-winded  :wink: , educational stuff like that.  Although I'm darn happy with my current front end set-up, I'd put your product on my future shopping list merely because of the time and attention you so obviously lavish on to those asking for information about it.  Bravo, my man, and thank you  :thumb:

brj

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« Reply #17 on: 20 Jan 2005, 05:22 am »
I agree with TheChairGuy - incredibly informative and much appreciated post, Chris!!

(No worries about the length, it was worth it - although I had almost forgotten the original thread topic by the time I finished it! :) )

doug s.

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« Reply #18 on: 20 Jan 2005, 02:05 pm »
here's a relatively cheap solution folks may wanna try before getting something like a monarchy dip.  i have used this w/excellent results, to get an optical-only-output cdp to work w/a coax-only-input dac:

http://www.rdlnet.com/fp-spr1.htm


rated at < 200 pS jitter; awailable for <$100 from a place like fullcompass.com

doug s.

csown

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« Reply #19 on: 21 Jan 2005, 07:16 am »
Hi guys,

Thanks much for the compliments, I'm glad it's appreciated and happy to oblige - I think the educated customer is the best customer.  There are a lot of technical misconceptions in audio and it's so hard to separate fact from the market-speak (I'm a physicist by training so I tend to be really obsessive-compulsive about it).  At Ack! we try to be as straight as possible because it circumvents disappointment, I think people are in general very happy with the results they are getting and the concept of selling solely based upon word-of-mouth recommendations seems to work out well for everybody involved.  It's immensely rewarding when a customer comes back and tells me they love what the product does for them.

Doug - cool product, I'll have to look into it.  <200ps is not bad at all, though I have a hunch that is a theoretical value based upon the specs on the output drivers they use...  Worth investigating for sure.

-Chris