Small form factor, budget DACs?


I'm trying to restore the musicality to my system, piece by piece. A few years ago my Jolida JD-602A CD player finally died and I've never really found a good replacement. I think really I've been mourning the loss and lacked the funds to get something of equal quality (since it was sort of a giant killer).

So, what can I get for < $400? Used is fine, but it has to be a compact form factor - I don't have room for another full-sized component. I think the 1/2 size form factor that Channel Islands, Musical Fidelity and Creek use is about as big as I could go.

24/96 is a plus since I have a bit of DVD-A stuff but not a necessity. I don't really have an opinion for or against oversampling, or regarding filterless DACs.

Here are the DACs that have popped up in my search so far:

$175 - Lite Audio filterless DAC
$250-400 - Ack! Dac
$200? - Creek OBH-14 - I'd have gotten one by now but I have yet to see one pop up on the used market. Probably a good sign.
$300-400? - Musical Fidelity X-24K - older DAC (circa 2000), but it looks nice and let's me stay with the appealing X-component form factor (I have an X-ACT and X-LPS now). Maybe a little overpriced - I can't help but think that for that money I could get something better
$400-600 - Channel Islands DAC - undoubtedly the best DAC on the list, but also the most expensive, so it would take the longest for me to save up the coinage.

Anything I'm missing from the list?
hudsonhawk

Showing 12 responses by bombaywalla

well, I was also looking for an inexpensive but good sounding DAC & a fellow who is local was selling a Scott Nixon Saru DAC+. Well, it cost me less than the DAC-Ah & so I took it for an audition.
I have it for a day now so I've played just a few CDs thru it.
I *think* I understand what Gmood1 is talking about - since I have the Wadia 861 to compare against.
Gmood1,
this particular DAC is the DAC+ version implying that it comes w/ a 4Amp AC power supply. The power supply is heavier + bigger than the DAC itself! It's got a fixed power cord. This DAC is NOT bass-shy - plenty of good quality bass & very good sound given its diminutive size.

Hey Spencer,
if I compare it'll be against the 861. I have NOT done that just yet - too many other domestic commitments + I'm being a bit lazy too (!). In & off itself, the DAC has a very good sound to it. Hard to describe what I'm hearing but if I had to use a few adjectives I'd say "natural"- the sound seems to be unhyped & "accurate"-badly recorded CDs still sound like sh** thru it.
I'll share more info as & when it becomes available. Thanks!
Undertow,
just to clarify - the non-oversampling DAC that I have is a Scott Nixon Saru DAC+. It has nothing to do w/ the DAC-AH that Hudsonhawk is speaking about.
Spencer, Gmood1 & anyone who might be interested:
I finally got my lazy ass to make a comparison between my Wadia 861SE with GNSC Reference mods & my Scott Nixon Saru DAC+. A Goliath & David sort of comparison as the 2 units are from the opposite end of the spectrum. Not an entirely fair comparison, I agree, but these are the only 2 DACs I have on hand & so it had to suffice. So, that's the disclaimer, FYI.
Executive summary: the Wadia was overall better sounding (surprise?) but the Saru DAC+ was very, very, very good in & of itself. Too damn surprisingly good for its price point.

The setup is: Wadia 861SE with GNSC Reference mods fed into my CAT SL1 Mk3 preamp using Amperex white label 7308 USN-CEP & Ei 12AX7 in the line stage. Analog cable is TARA Labs Master Gen 2. Stock Sony DVP-S7000 transport & Scott Nixon Saru DAC+ with 4Amp power supply. Digital cable is XLO ER-6. Analog cable is Groneberg TS Premium.
I played a few CDs: Art Pepper "Winter Moon", Dee Dee Bridgewater "DEar Ella" & Reinhart Fendrich "All the Best". So, the comparison is not extensive.
Saru DAC+: a little brighter than the Wadia. Could not have noticed this if I didn't have the Wadia on hand. Excellent dynamics. Very natural sound. Plenty of bass but the Wadia has more. Really nice midrange.
Wadia: superbly natural. No hi-end glare & very smooth. Each instrument in the track is dilineated in its own space & its easy to follow all the instruments. The Saru DAC+ also reproduces all the instruments but they are not as easy to follow. The bass is tight & superbly controlled thereby making it easy to follow the bass tracks. This is not as easy in the Saru DAC+. The music just flows from the Wadia w/ great ease - no track seems to ruffle its feathers. In the Saru DAC+ I did get a feeling that it was "scurrying around" a bit. The DAC kept pace w/ the music & did a fantastic job but if it were a human being, it would have been a bit out of breath. Not so w/ the Wadia.
The PRaT factor of both DACs was really very good. In this short evaluation I could not find any serious faults in the Saru DAC+. The Wadia was much better. Was it as much better as the price difference? Certainly not! I also wonder if my DVP-S7000 had some power supply mods + some internal chassis damping would the results have been even closer? Probably!
So, this is a testimony to the progress that has been made in I.C.s that support the audio market. A few of them put together judiciously in a diminutive package can seriously challenge a Goliath like the Wadia 861.
Undertow,
Thanks for making the suggestion. I did think of what you wrote & decided not to go ahead because (1) the digital inputs have been disabled per Wadia's recommendation. They suggested that if the unit is being used as a single-box player, the performance is little better w/ the digital inputs disabled. I have to get this 60lb monster off the shelf, completely disassemble the top & bottom plates of the chassis to get to the DIP switches on the bottom to re-enable the digital inputs. A royal PITA! & (2) the Wadia running into the Saru DAC+ is likely to sound quite compromised since I will not be able to make use of the Clocklink feature (that now exists internally between the internal DAC & transport). From whatever I have read from other Wadia users, this Clocklink feature is of paramount importance in making the Wadia separates sounding superb. This is understandable - Wadia correctly implemented the clocking scheme by using the DAC clock to lock the transport. When one uses Wadia separates, all Wadia gear comes w/ the Clocklink feature & one is able to turn it on from a menu pick. The Saru DAC+ provides no such feature. Thus, the Wadia transport+Saru DAC+ combination is going to sound way inferior to the Wadia 1-box thereby leading to an incorrect conclusion.
(If there is any Wadia user reading this, let me know if I'm wrong. Thanks!)
>> 04-02-06: Hudsonhawk
>> It's funny, I made a similar note when listening that
>> Bombaywalla did while listening to his Nixon DAC - bad
>> CD's sound *bad*. I've got two theories about this - I
>> wonder if either these CD's have digital noise that's
>> actually in the master, or these DACs are particularly
>> jitter-prone.
good to read that someone else also experienced the same on badly recorded/pressed CDs! when I was writing my response post to Undertow, exactly the same thought crossed my mind. These non oversampling DACs use a Crystal Semi 8412 or 8414 I.C. that uses a PLL to extract & lock onto the CDP clock. This is much different than most other hi-end systems that lock the CDP to the DAC clock, which is a more stable/less jitter clock. If these badly recorded CDs must have lots of noise in them, it *could* push the PLL to edges of its lock range, which could have the same effect as high jitter.

>> 04-02-06: Islandflyfisher
>> I think itÂ’s all about error correction. A CD transport
>> can only do so much. Near perfect CD error correction
>> for today PC is a simple process for them.
sorry to burst your bubble, dude, but it's not all about error correction!
In another A'gon thread we've have been thru this - even the most economical CDPs & DVD players have good enough laser pick-ups systems to read the CD w/ near-100% accuracy. The error correction you are talking about are CRC & Reed-Solomon type error correction codes used to correct bits as they are read off the CD. Error correction is generally not used elsewhere in the CD/DVD player.

What is probably happening in your case (w/ the PC server) is that you are lending credence to Hudsonhawk's theory of these non OS DACs being jitter prone. It is well-known fact that ripping a CD to one's hard-drive before burning it to CD-R or playing it back on one's stereo is a good method to reduce jitter - the hard-drive dumps the data to a FIFO & the sound-card reads it using a much more stable clock.
>> 04-02-06: Gmood1
>> Bombaywalla,
>> .......I'm thinking that some of the differences you
>> were hearing had a lot to do with the output impedance
>> of the Wadia (51 ohms) verses the SN DAC ( guessing
>> maybe 3000 ohms).

Gmood1, I'm having a hard time believing this. Off the top of my head, I don't know what the input impedance of my preamp is, but I think that your guess of 50K is pretty damn good one. I'll have to look in the user's manual where it is stated.
AFAIK, if the input impedance to the next stage is 10X higher than that of the prev stage, the input imp gets defined by the prev/driving stage (in this case the DAC output). Thus, both 50 Ohms & 3K Ohms are small enough for an input imp of 50K.
The SN Saru DAC+ uses Burr-Brown OPA627 buffers. I briefly looked at the TDA1543 DAC spec page & if I read it correctly, it's a current o/p DAC. So, these OPA627 buffers must be doing a dual job of current->voltage conversion + buffering. There has simply got to be feedback around these OPA627 opamps (in the wcs, it's being operated as a unity gain buffer) in which case, the opamp's buffer o/p impedance gets divided by the OPA627's open loop gain. This DC gain is usually very high implying that the (closed loop) o/p impedance must be very small (less than 1 Ohm).
A long way of saying that I don't believe that o/p imp has anything to do w/ the sound difference.

>> The buffer which I believe your player has built in,
>> gives more presence and makes the musical lines easier
>> to follow.
This makes sense - the TDA1543 DAC does not have the capability to drive the interconnect cable + preamp input in terms of creating enough voltage swing at the preamp input. It just wasn't designed for that! Hence, the need for a buffer. The component values in feedback network for the buffer need to be carefully selected so that they do not load the TDA1543 o/p. Additionally, overall thermal noise from resistors also needs to be considered.

>> I also believe this is one of the reasons many love PC
>> audio. Some of the sound cards used have an output
>> impedance of 50 Ohms.
I don't know much about PC sound cards. Somebody w/ more experience can confirm or not whether the o/p impedance is 50 Ohms or not.

However, unless I see a good reason to contradict, I believe that Hudsonhawk is on the right track w/ his hypothesis of the sound diff - the clock jitter.
As I wrote in my prev post - the Crystal Semi 8412/8414 locks onto the recovered clock embedded in the digital data stream using an on-chip digital PLL. The o/p clock from the 8412/8414 cannot be any cleaner (jitter-wise) than what is fed into it. Hence, the clock to the TDA1543 sample & hold ckt is a jittery clock (esp for badly recorded CDs). This will certainly create D->A errors resulting in "digital" sound. The more I think about this issue, the more I'm convinced that this is the issue. If there is someone out there that thinks I'm wrong, please correct me.

One thing that could be done to alleviate this issue (& higher-priced DACs like Audio Note, etc might be doing) is to create a very low jitter clock ref for the DAC (say, using the Tent XO module or something similar). It can be 44.1KHz or 48KHz or 88.2KHz or 96KHz. Then, using the Crystal Semi 8412/8414 to lock onto the embedded clock in the data stream, dump the incoming data into a FIFO at the CD transport clock rate. Then, using the low-jitter DAC clock, clock the data out from the FIFO into the DAC. This separates the CD transport noisy & jittery clock from the DAC clock. The sound o/p must improve dramatically.

Look at a sound card - I think that you'll see a clock/crystal on that PCB! it is clocking the data into its buffers from the PC hard-drive using that clock & non OS DAC is locking onto that clean clock. Hence, the sound o/p is much better. Bet you, that's what happening!

Gmood1,
yes, 10K input imp of the next stage vs. the driving stage is not written in stone. Usually the rule of thumb is 5X-8X. However, I've found that is not high enough & that 10X works 99% of the time successfully.

OK, from your 2nd lengthier post I think I understand better where you are coming from. Let me see if I can summarize: Hudsonhawk & I were musing why these non OS DACs based on TDA1543 DACs sounded (really) bad on bad recordings. He & I were hypothesizing that it's the lousy jitter performance from the recovered clock.

NOTE: both Hudsonhawk & I have non OS DACs that use a Burr-Brown opamp buffer that drives the RCA outputs. From the spec sheet of this opamp, it has very low harmonic distortion over 20Hz-20KHz + it can drive large capacitative loads - we are talking 5nF & still have a gain-bandwidth product greater than 1MHz + it has very good settle time. So, it seems that this buffer does the job of the BVAudio SR10 & similar after-mkt buffers. Thus, I see very little advantage in further attaching an after-mkt buffer. If there is an improvement in these 2 specific non OS DACs, I surmise that it is most likely to be very little. I wish I had one on hand to give your theory a try (I would be double buffering).

Now, in your Audio Sector Premium non OS DAC that uses passive buffering, the ball-game is entirely different. As an aside, if you look @ the TDA1543 data sheet, you'll see (in Fig 1) that they have suggested the use of buffer opamps that have some bandwidth limiting (that parallel cap in the feedback). It is not the only way to "terminate" the TDA1543 output i.e. one could also use passive buffering. However, the passive buffering will rely on the TDA1543 to drive the interconnect parasitic C + the preamp input. It'll do the job (as your ears have discovered) but you know that the sound could be better (again, as you have discovered). In your particular case, the BVAudio SR10 & similar products work & show the difference since the passive buffered TDA1543 has to drive 100K & very little parasitic capacitance of the active circuitry, a much easier load. I do not think that it'll be quite the same for my SN DAC or the DAC-AH.
There is nothing magical about the 50Ohms output impedance of the BVAudio SR10. It is a standard impedance used in test & measurement equipment & by the RF engineers. It is low enough where it'll work w/ 99.9% of the equipment in the market-place no questions asked. For that matter, 600 Ohms would have worked just as well (would have been an easier load actually) as it would have been low enough to work w/ all the preamps out there.
So, you have to pay the Piper - now (opamp buffer as part of overall DAC in the same chassis) or later (use after-mkt buffer).
So, in your particular case, it appears you have 2 issues affecting the sound: insufficient drive from the Audio Sector DAC & poor jitter performance from badly recorded CDs. IMHO.
Gmood1,

>> You must ask yourself. Why use a preamp between your
>> Wadia and amplifier?
the answer is very simple: Wadia's digital gain control is the absolute pitts! Sonically, it stinks! I know I'm an owner but I will not prop up this unit just 'cuz I spent money on it. What I do is put the volume control to 100 thereby bypassing it. Now I have totally lost control over adjusting the volume. That is where my preamp comes in. Inside the Wadia they provide a set of DIP switches to adjust the buffer outputs from 4.25V all the way down to 0.25V for each channel separately. I believe that the user adjusts this output depending on the amount of voltage gain in his/her preamp.
Wadia claims to have a patented I-->V conversion technique & I wonder if these buffers are doing double duty of I-->V + buffering? I don't know enough about the Wadia design (& the DAC control PCB is multi-layered thereby making wire tracing nearly impossible) to make a useful comment.

>> There's a reason why APL,RAM and other modders
>> concentrate so much on the output stage of their
>> players. Which includes doing away with the negative
>> feedback opamps.
well, even if I try the BVAudio SR10 external buffer, there's no getting rid of the negative feedback opamps inside the SN DAC.
OK, I can understand negative feedback in the opamp-based output stage is less sonically than those outputs stages not having any.

>> Also the impedance does matter when driving long
>> interconnects or passives.
agree! How long is long tho? Not 1m interconnect! I think long would be in excess of 5m.
I would agree that output impedance is very imp when driving passives. A CDP having an output imp of 3K (as written in your orig post) would likely be an issue when driving a passive preamp. I agree!

>> I can imagine handling interconnect interactions is a
>> walk in the park for such players.
I believe that this would also be a true statement for a DAC using a Burr-Brown OPA627.
When I plug this SN DAC into my preamp, I have so much gain from this DAC that I have to crank down the volume knob atleast 3 clicks from where I listen to the Wadia. In the CAT that's quite a gain reduction.

>> I wish you were closer ..I could quickly prove your
>> hypothesis wrong.
well, my hypothesis is, for the SN Saru DAC+ which uses a strong enough output buffer than can drive my 1m interconnect + my preamp input capacitance, that badly recorded CDs sound terrible thru it 'cuz the dominant effect is poor jitter performance in the recovered clock from the input data stream. The poor jitter performance xlates to poor DAC performance as the DAC uses this jittery clock to perform a D-->A.

(Were I to additionally remove the negative feedback in the output buffer would I get even better sound? probably!
I think that in my DAC the jitter performance dominates over the negative feedback in the output buffer).
Gmood1,
I think that we have been agreeing on almost everything all along & just didn't read close enough to recognize that.
I think that re-stating what my hypothesis could have helped clear the air a bit - I was/am looking for the dominant cause for my case.

>> Maybe this is the problem you hear in your DAC or
>> transport..I really don't know. Honestly I've never
>> cared for the stock Sony 7700 players analog outputs,
>> they are quite the ear bleeders in my book.
well, I think that you might have indirectly hit upon something I wanted to mention in my prev post but it got too long: the issue, I think, lies in my stock DVP-S7000 DIGITAL output (I'm using it as a transport to the SN DAC so the analog outputs do not feature). The S7000 digital outputs have very sloppy rise & fall times, which has the tendency to aggravate jitter issues. Sending it in for a mod to improve this should give me good ROI because it is the digital data stream that the SN DAC sees as its input & it is from this data stream that it extracts a clock & lights up the "lock" LED on the front panel. If an improved digital output buffer adds minimal distortion to the data stream, the performance is bound to improve.

I didn't think that the analog output of the S7000 was an ear-bleeder as much as it was very dull/flat sounding.

BTW, I wanted to clarify that your transport having the Superclock3 has much less to do w/ the DAC jitter performance compared to the quality of your digital output. What is making the bigger diff in your case is the mod to your digital output. Don't get me wrong - the Superclock3 makes a positive diff but it is 2nd only to the superior modified digital output. The Superclock3 would have a dramatic effect if you were using the Pioneer internal DACs i.e. using it as an integrated player, 'cuz this low jitter clock would be connected to the DAC, which would have kicked its performance up a few notches (iffff the internal DACs are good quality in the 1st place). The data stream that your Audio Sector DAC sees is just a 1-0 pattern. Whether it is correct or incorrect data, the DAC doesn't care nor does it know. The job of the Superclock3 in extracting the data from the CD is complete by this time & it is the quality of the digital output that will affect the overall sound.
It is no wonder that the websites of many 3rd-party modders lists the Superclock3 mod as a separate line item & don't always push you into getting it if you are using the unit as a transport. (or, maybe they still do 'cuz they want your money anyway!!)

>> There's a reason why APL,RAM and other modders
>> concentrate so much on the output stage of their
>> players. Which includes doing away with the negative
>> feedback opamps in some cases. I noticed most of them
>> use single ended designs with no negative feedback and
>> powerful output transformers in their top
>> players.
Another point I wanted to clarify - there isn't a single electronic device on planet Earth that can work without negative feedback! Trust me on this - I'm 110% certain! I think that you meant to say that the top CDP modders use output stages with extremely localized feedback to make them sound the best. I agree w/ that.

My sis-in-law lives in ATL. If I'm visiting her, I'll email you & we can probably get together & compare my SN DAC w/ your Audio Sector?
Like-wise, my friend, all the best & enjoy the music! :-)
Undertow,
As I understand it, when mods are done that EXCLUDE the clock mod, the modifications are (1) improving the power supplies - digital & analog, (2) upgrading coupling capacitors, (3) improving the rise & fall times of the digital output to the 1-2nS range from something much higher (like 20-30nS range) & (4) ensuring that the digital output is 75Ohms so that the power/signal transfer to the DAC module is maximized.
If you do this in a DIY fashion, you do not need to mess w/ re-attaching a Superclock3 or Tent XO, etc to the CDP, which is very specific for each CDP & can be quite a pain if there is no guidance. Doing the above mentioned mods (tho' tedious) are relatively simply to do & the parts are freely available from multiple vendors (DigiKey, Newark, Percy Audio, etc).

Yes, I have also seen people add a re-clocker unit/jitter reducer like a Monarchy DIP between the digital output & the DAC input. These type of units seem to be quite cheap esp. on the used market.
I don't know how easy it is to add a re-clocker to the DAC chassis itself - maybe there is a DIY forum where they have discussed how to interface a clock module to a digital receiver like the 8414?
Adding a separate chassis re-clocker seems to be cheaper & easier to do. The sound should improve IMHO.

There is another way stated by a few members already & a growing trend :- use a PC hard-drive as your transport & a sound card to put the data out on a USB port. People like Scott Nixon & Wavelength Audio & Headroom are producing USB DACs to enable this.
Then, you do not have to deal w/ this jitter issue 'cuz you rip the CD to your hard-drive & the sound-card (which has a high quality clock + digital buffer) feeds the DAC.

If you are willing to spend money to upgrade your CDP transport, re-channel that same money into getting a plug-in sound card for your PC, a USB cable & exchanging your DAC to one having a USB input port.
>> 04-05-06: Undertow
>> I got a little lost on the Re-attaching the XO or
>> superclock,
look at this page
http://www.tentlabs.com/Support/Support.html
click on "XO 2 and 3" to read the mounting instructions to understand what I'm talking about.

>> so upgrading everything but the clock in a CD player
>> used primarily as a transport would be a good idea or
>> not without the clock?
good idea (if you don't want to go the PC route).