Good RedBook CD transports..

what DAC do you use ? wats your budget ?

Pure transports are very expensive and if you are not in the mood to spend so much, any sturdy DVDP should be a good transport...(Pioneer/Toshiba/denon et al)

Since you already have a DAC the best DVDp you can afford will be the best deal for you as that will also do proper DVD duties. and most DVDs have a robust transport mechanism which you can further enhannce by damping its body.

Arjun,
I am using a Chord DAC64. I dont mind getting a good transport if there is a genuine sonic advantage. Tried many DVD players and CD players and noticed that DVD players were performing better than most of the CD players and the toslink is better than SPDIF with cheaper DVD players.

Anil

the fact about Toslink is pretty true.. on the good transports the SPDIF has a lower jitter. If you can I will suggest getting either a Monarchy audio DIP or a Sonic Frontiers Jitterbug. both of them reclock and minimise jitter and hence you can use a DVDp as a transport. (Basically reduces the impact of a Transport unless u are tlking of megabucks)

I have the Older DIP and use it on an entry level pioneer DVDp for cd duties and also along with an Airport express via toslink for some digital streaming. Since the DIP has a AES/EBU digital out i use that to connect to my DAC. I find it excellent for its price (Lesser than a fancy IC ) especially if u can get one used

YOu already have a very detailed DAC.. you might find the DIP much more easily available..furthermore it has an upsampling feature as well.
http://www.soundstage.com/planethifi/planethifi200203.htm

Arjun Thanks for the valuable inputs. It is difficult to get Monarchy products here but let me try. This DAC is said to have a large buffer (2sec and 4 sec) and should be immune to high frequency jitter which in real life is far from true ;).

ani schrieb:

Arjun Thanks for the valuable inputs. It is difficult to get Monarchy products here but let me try. This DAC is said to have a large buffer (2sec and 4 sec) and should be immune to high frequency jitter which in real life is far from true ;).

From what I have understood in this subject Buffer in the DAC does not really help jitter. cause of jitter can be multiple . most often it is due to the masterclock itself and sometimes due to the cable as well whose capacitance/inductance can cause some bits to go wrong due to threshold levels.
These kinds of jitter needs other forms of correction than buffering.

Just a few weeks ago, I had got home, to audiotion :

1. A CDC Belt driven transport. This is a HUGE ( 20 Kg ) beast, which provides an analog characteristic to the sound.

Luxorious feel and operation, is going for only Rs 30 K ( that too possibly negotiable. ) New price was US $ 4,400.

Also on offer was a Theta DSPro DAC for maybe Rs 70 K...
( That too, I guess negotiable ? )

I did not keep them, so I guess that they are still availble.

If you are interested, PM me and I will give you the seller's Contact details.

Both transport & DAC were in Mumbai.

Cheers

Arj schrieb:

ani schrieb: Arjun Thanks for the valuable inputs. It is difficult to get Monarchy products here but let me try. This DAC is said to have a large buffer (2sec and 4 sec) and should be immune to high frequency jitter which in real life is far from true ;). From what I have understood in this subject Buffer in the DAC does not really help jitter. cause of jitter can be multiple . most often it is due to the masterclock itself and sometimes due to the cable as well whose capacitance/inductance can cause some bits to go wrong due to threshold levels. These kinds of jitter needs other forms of correction than buffering.

you probably knew that i'd step in here!!

The 2s-4s buffer in the DAC should help reduce the jitter considerably if the Chord DAC has its own master clock. I don't know anything about Chord products but I'm assuming that it has its own master clock otherwise, having the buffer is meaningless.
The master clock on the DAC board should be very low jitter. Again, if it's not, no point wasting money on components just for the sake of having them there!
The 2s-4s buffer acts like a FIFO (first in, first out) from which the DAC takes its data @ the master clock rate. Here, using the FIFO in what is called asynchronous rate conversion, the DAC can convert the 44.1khz CD data to any rate it wants to: 48k, 88.2k, 96k, etc. One option, of course, is to leave the rate at 44.1khz. Rate conversion or not depends on the designer's philosophy.
Many top-end CDPs do the same thing except that many do this in just 1 box. Here Chord has implemented a re-sampler in his DAC thereby allowing you to totally skip the Monarchy DIP or GW Labs resampler.

Causes of jitter are several - correct. The C & L of the interconnect does mess w/ the signal's rise & fall times, which, in turn, does adversely affect jitter but the buffer driver at the CDP output end has the more dominant impact on the jitter than the cable.

The worse option would have been for the Chord DAC to recover the clock from the data stream & lock onto this jittery clock via some digital PLL scheme. This would have been the clock fed to the DAC & then, sonic performance would have gone to Hell on a straight down-hill road!

bombaywalla schrieb:

you probably knew that i'd step in here!!

I definitely did not but am very glad that u did

that was very interesting info for me.. and yes it does make sense. if you do have a masterclock with a buffer it definitely should be an effective antijitter device.

but if the SPDIF does not sound as good as the toslink.. then there is something else jittering there as the toslink jitter figures are much higher than a good SPDIF

and while we are on that topic, heres a good read
http://www.scalatech.co.uk/papers/aes93.pdf

Arj schrieb:

but if the SPDIF does not sound as good as the toslink.. then there is something else jittering there as the toslink jitter figures are much higher than a good SPDIF and while we are on that topic, heres a good read http://www.scalatech.co.uk/papers/aes93.pdf

the typical issue w/ SPDIF is the way that the cable is made it does not have the bandwidth to carry a digital (hi freq) signal. Dig signals are hig freq 'cuz a square wave can be considered to be an addition of several sine waves - the fundamental sine wave + ALL its harmonics.
The SPDIF cable typically attenuates the upper harmonics of the digital signal which adversely alters the phase of the music hence it ends up sounding bad.
Toslink cables themselves have better/higher bandwidth, which is translating to better sound.

The cheapest cable that can carry a digital signal is a BNC 'cuz it is a coaxial cable. These sort of cables are used routinely for cable TV & have 20MHz bandwidth or maybe even higher.

what one can do is to buy a RCA-to-BNC adapter + buy a BNC cable & use this BNC between transport & DAC.

Also, it comes as no surprise that a DVD player the better transport. This is well-known on this side of the pond. DVD players use a finer laser pick-up since the DVD disk itself has 4-5X more storage. Also, the DVD spins at 1000-1600 RPM whereas the CD spins at 360RPM. Thus, the DVD laser mechanical assembly is better quality than a CD's unless you buy TEAC's VRDS-based transports. All this xlates to better sound from a DVD transport.

Bombaywalla
I can tell you that I have heard the pioneer DV366 DVD player being used as a transport and it did its duty very well.

manek.

Thanks to all. Regarding Chord DAC64 I would like to share some technical details with you.

DAC64 has its own clock and the buffer is used is asynchronously, but you need some sort of long term sync with the input data stream.

This arrangement of large buffer with seperate clock removes much of the high freq jitter but the low freq ones that can still pass through the buffer.

They have used an unique filter called Watts Transient Aligned which is implemented on 3 FPGA chips. something like what WADIA does except in chord the D/A chip is not a stsndard of the shelf one but a Pulse Array one.

for more of it http://www.chordelectronics.co.uk/products_detail.asp?id=33

99% of what I have written above are way above my knowledge of DSP technology but all I can say is that it is a DAC which suits my taste, very dynamic and not a hint of treble glare but with all the hi freq energy intact. If I may use the much abused description "analogue like" in fact it is more like playing the new generation Lp's.

Regarding SPDIF (BNC) input on DAC64 some CDps couldnt lock with it when connected using BNC - RCA cable of reputed make such as Nordost and Professional ones like Neutrik/Belden etc. Surprisingly the same CDp works (very hard sounding) with the cheap thin RCA analogue cable that is thrown along as freebees. If you read what is given below my obseration should lead to the inferior RF shielding of Nordost and Pro cable !!!

This is info from a site said to be by the designer of Chord DAC64


DAC64 is not compatible with most transports. It does not lock to the incoming signal. However with a Toslink connection it works fine with all transports I've used so far. With coaxial after a long pause I have to switch it off and on to get it to lock again. Not so with Toslink, which somehow seems to sound better. I understand you have changed the jitter window from 20nS to 45nS by inserting a reformulated EPROM. But I have compared both models and the tighter the window the better the sound (provided you have a compatible transport). Why?
Rob answers:
The jitter window was improved by improving the design of the receiver FPGA, making it capable of extracting data with very large jitter. The jitter window was more than doubled, and I am now close to tolerating 50 per cent min bit cycle time - the theoretical limit. In performing this update, the DAC and filter FPGA programs are identical - only the extraction circuits. So it is difficult to see how this could significantly change the sound. I have not had a lot of experience with different transports, but I know people have had different experiences. The problem is the RF noise injected into the DAC from the transport, this can effect the sound, making it brighter and more up-front with noisy sources.The solution is (as you state)to use the optical which does not suffer from this effect. In normal situations optical generates so much jitter, that this benefit is masked by the degradation of the jitter. But with the RAM buffer, source jitter is eliminated, so you only have benefits in using optical. Trouble is, most audiophiles have such a prejudice against optical, they won't even try it. The reason I give 1 and 4 seconds is the only time you get that is guaranteed 8 seconds of silence is when you are changing discs. So you can have 72 minutes of playing a CD where the pointers are completely separate. If the clock on the transport is low quality and not accurate, the 1 second delay can drift to so that there is no delay - then suddenly 8 seconds! So we give the option for poor accuracy transport clocks. The vast majority of hi-end transports are accurate enough for 1 second

Here is the link to Q&A wihttp://www.hificlu...uivo&subseccao=DACth Rob Watts

ani, you may not be very wrong in your preference for toslink

a lengthy comparison below, but you could skip all that for the conclusion
http://www.6moons.com/audioreviews/toslink/toslink_2.html

Both Van Den Hul Optocoupler and WireWorld SuperNova III+ hold their own against far more expensive non-optical (blind?) RCA-S/PDIF cables. The differences that do exist don't proclaim one or the other the victor based on clear superiority. They simply operate inside system compatibilty effects and those of personal preferences. And true, "bad" Toslink does most certainly exist. If it's bad, it's clearly bad. What makes it so? Seeing that both VDH and WireWorld eschew plastic conductors in favor of glass fiber, one could conjecture that the former is not appropriate for High-End applications. Until, that is, one were to come across a superior Toslink that did use plastic conductors. It'd throw that theory out the window for yet another example of how a precise understanding of cause-and-effect relationships in audio remains -- mostly -- vexingly elusive. What's far more relevant and important to consider? Simply this. Don't base your purchase decisions for a new CD/DVD player, stand-alone transport or DAC on the kind of socketry it carries on its rear panel. Toslink jacks are not automatically inferior. In fact, based on Audio Asylum commentary of listeners who have performed their own comparisons, some clearly prefer glass-fiber Toslink. Perhaps that's due to creating a true stop gap against interference from ground problems. Secondly: If your current or dream player offers a choice of either hookup, don't limit your search for the right cable to the single-ended standard. You could obtain as good or better for considerably less by exploring the Toslink route. If your local dealer won't play, contact Robert Stein at the Cable Company for some sample loaners. Just don't call him if you're not truly in the market and just want to kick his tires. He'd give me a hard time for suggesting it. Rightfully so. The man's running a business, not a lonely heart's club for bored audiophiles. Lastly, if you had Toslink and RCA sockets, you could use both outputs to feed two highly resolved systems from one transport/player. The choices are dizzying. And good choices they are, too. Life's peachy in affordable audio land.

Thanks to all, this weekend I will try out a few DVD players at home, a few Denon's 1920, 2910 & 3910 are the ones I could get for trials.

Hope that one of them will serve my needs.

Anil

So bombaywalla are yu saying that buffer+good master clock will eliminate much of jitter.

One more thing how does coaxial and digital output compare wrt the jitter i mean i have both on my DVP (pioneer elite)

When i was in US i happen to see a DAC called "Benchmark" they use a technology called ultralock http://www.benchmarkmedia.com/appnotes-d/whyultralock.html
can someone explain if it is worth ?
Also can i connect an airport express to a good dac and eiminate jitter thus? or is it a function of the digital output device?

sanathan schrieb:

Also can i connect an airport express to a good dac and eiminate jitter thus? or is it a function of the digital output device?

You definitely can. only problem is that the AE to DAC is a Toslink with a 3.5 jack which is perhaps not the best to eliminate jitter but still it works very well.

I use that as my main source

sanathan schrieb:

So bombaywalla are yu saying that buffer+good master clock will eliminate much of jitter.

yes, buffer & master clock in the DAC chassis.

sanathan schrieb:

One more thing how does coaxial and digital output compare wrt the jitter i mean i have both on my DVP (pioneer elite)

IF I understand this question the comparison is between the coaxial & the RCA/SPDIF digital output, correct?
The coaxial or BNC is superior.

sanathan schrieb:

When i was in US i happen to see a DAC called "Benchmark" they use a technology called ultralock http://www.benchmarkmedia.com/appnotes-d/whyultralock.html can someone explain if it is worth ? Also can i connect an airport express to a good dac and eiminate jitter thus? or is it a function of the digital output device?

The Benchmark DACs have gotten quite a good rep here. UltraLock is a marketing term for what the Chord DAC is doing: buffer + master clock in the DAC chassis.
Wadia calls this "ClockLink". Yet another marketing term!

For the Airport Express: use a DAC like the Benchmark or the Chord 64 i.e. one that has its own clock & does not derive the clock from the CD data.
Next question is: is the format in the Airport Express MP3? If yes, why waste your time w/ removing jitter? The MP3 data format is a lossy data format & you have already lost much of the music thru this compression format. With a damn good DAC all you'll hear is just how crappy MP3 sounds! You really want to do this?

ani schrieb:

This arrangement of large buffer with seperate clock removes much of the high freq jitter but the low freq ones that can still pass through the buffer.

what do you mean by "low freq ones"???
It appears that the concept of jitter is not clear.

Jitter & phase noise are equivalent terms - jitter is used in the digital domain & phase noise in the analog domain. They are the same thing & there is even an equation to xlate between jitter & phase noise.
Typically you see phase noise for a crystal or TCXO (temp compensated crystal osc) & the plot looks like a Christmas tree. That link to the Benchmark website posted by "sanathan" showed a nice pix of a phase noise plot. Phase noise can be minimized but not eliminated. When a TCXO or XO with phase noise is used to create a digital clock, the digital clock has jitter on it. Jitter can be simply explained as the uncertainty of when the clock will change from logic 1 to logic 0 or vice-versa.
Jitter is typically specified in 3 ways: (1) periodic jitter where each period of the clock is measured & the change in the period is noted as jitter, (2) cycle-cycle jitter where difference in each adjacent cycle of the clock is noted & (3) Total interval error jitter where the deviation of the clock transition from ideal is noted. This method of spec'ing jitter is not practical as we rarely know what the ideal point in time is for the transition. Anyway, I cited it for completeness.

So, there is no high freq jitter & low freq jitter. It's just JITTER!

The way that the Chord 64 implements its DAC, the buffer + precision clock should eliminate all effects of jitter.

The sound of jitter? Given that you have tweaked your system to have a low noise floor on a consistent basis (& this is difficult esp since the AC power quality varies a lot) then, high jitter will cause the noise floor to rise thus the overall S/N will degrade. Low level details in the music will be masked.
Also, the fundamental freq (of the music notes) will smear in freq. This will modulate the harmonics of these fundamental notes. If these harmonics were buried in the noise, then, they might rear their head. Thus, THD can also be degraded due to high jitter. Due to the freq smearing, voices, fundamental instrument notes will sound un-focused.

Don't go back & start listening to your systems & remark that jitter is the cause of smearing!! The effect is very subtle & is hard to identify/pin-point w/ certainty. Jitter can be identified but the system needs to be kick-ass overall & so many other things like vibration, AC power supply quality, digital cable effects can have more hearable effects before it is pinned to jitter.

Excellent post Bombaywalla...

Would appreciate more detail..more technical stuff....a link would do too..

Regarding MP3s, I quite agree with Bombaywallas comments on jitter reduction relevance with them

Again it is not that I personally would suggest an MP3 over the lossless variety, butt surprisingly I have found the 256kbps+ MP3s to retain soundstaging as well as treble/Bass(Bass checked with a SPL meter remains at same levels)

But to me the difference was in the "Richness factor" in music getting reduced with an MP3. This is especially tru in pieces with lots of intstruments. Hence it usually works well with most Rock/pop and some vocals.

I, unfortunatley am unable to fanthom what it is or whether it is just Psycho acoustics since i find it difficult to administer a blind test to myself without "Peeking". But if you do a lossy compression you are definitely losing something.

Only reason IMHO could be due to some of the Harmonics being removed in MP3 leaving mostly fundamentals ????? anyone knows about the technology of the compression ?

HI Bombatwalla that answers my questions completely such a nice post , All my music (about 40 Gb :-)) is in apple lossless replicated over two high speed external hard discs which can be connected to my laptop :-) , so thats why i am thinking of airport express, the damn thing is that good DACS are almost always costly :-( (for me)

Arj schrieb:

Only reason IMHO could be due to some of the Harmonics being removed in MP3 leaving mostly fundamentals ????? anyone knows about the technology of the compression ?

Hmmm...had taken a course,Wavelet Transforms, which broached on the subject of image and video compression(Mpeg-1 and 2)...Also had some on how 2 channel is mixed but nothing much other than that the two critical frequencies for recording being mentioned..something like 500Hz and 3500hz....will have to refer to my textbook which is back in Bangalore for that..

Nice but long read here
http://en.wikipedia.org/wiki/Mp3

but an interesting point is that phsychoacoustics is one of the Basis for its encoding.. now that is interesting !

As a teaser

MP3 is a compression format. It provides a representation of pulse-code modulation-encoded (PCM) audio data in a much smaller size by discarding portions that are considered less important to human hearing (similar to JPEG, a lossy compression for images). A number of techniques are employed in MP3 to determine which portions of the audio can be discarded, including psychoacoustics. MP3 audio can be compressed with different bit rates, providing a range of tradeoffs between data size and sound quality.

Wiki is good but I don't rely on the data too much as there is no governing body to autheticate the information and anyone can edit..so at the point of time I read some nut would've posted crap am led to beleive it's true.:( .maybe later experts might revert back the original info..

My appeal to all members please do not post links from Wiki as from the time you've read the content there and post the link here and by the time we read the same some cranky nuts would've edited the topic for good or bad and it dilutes the whole purpose

Dear Bombaywalla,
In theory the buffer and reclocking should be independent of the input clock frequency because the data is clocked out with reference to the accurate local Xtal ocillator. In practice the buffer reclocking can be implemented only with some kind of link to the input clock, which is usually done by synchronising the local clock to the long term averaged freq of the input signal clock.

This results in the pass through of jitter that is having periodicity longer than that of the long term avg used to sync the master clock. Since chord has gone in for a very large memory they can go in for a very long time period for averaging and hence the eliminate most of the jitter. Still long term shift in freq of input clock will cause the reaclocking freq to change and that is JITTER.

Through SPIDF Chord was not able to eliminate jitter and they put blame on RFI generated by the source. I take it as idiosyncrasies of highend audio designers !!! but I like that aluminum box

ani schrieb:

Dear Bombaywalla, In theory the buffer and reclocking should be independent of the input clock frequency because the data is clocked out with reference to the accurate local Xtal ocillator. In practice the buffer reclocking can be implemented only with some kind of link to the input clock, which is usually done by synchronising the local clock to the long term averaged freq of the input signal clock.

I see! That's how they have implemented their jitter reduction scheme. Altho I read what u had posted before on the Chord 64 but I didn't quite get it. Maybe the way it was written. Plus, given the price of that DAC esp here in the USA, I didn't expect them to cheap out this way!

ani schrieb:

This results in the pass through of jitter that is having periodicity longer than that of the long term avg used to sync the master clock. Since chord has gone in for a very large memory they can go in for a very long time period for averaging and hence the eliminate most of the jitter. Still long term shift in freq of input clock will cause the reaclocking freq to change and that is JITTER.

I'm afraid that I do NOT agree! IMHO, that is NOT jitter.
The way that jitter is defined (anywhere that I have read & understood it + played w/ it), it is defined as a fast occuring event. When periodic jitter is spec'd they talk about 1 period to the next period change. Never long-term average.
Long term change in the re-clocking freq is called "drift" (& not jitter).
When the Chord 64 buffer takes 1s or 4s to average the data, the output will have RMS jitter to which the DAC clock will lock. This 1s-4s buffer is really a digital filter effectively. In the analog domain, the equivalent filter would be one w/ a very low freq -3dB corner to eliminate most of the noise.
So, yes, they would have minimized jitter but it'll end up being the average/RMS jitter of the CDP/transport clock.
If the CDP/transport clock is sloppy (as most are even in hi-end units), then, the average/RMS asertained by the Chord 64 buffer will be a high number.
This means that the Chord DAC is D-->A'ing with a higher jitter clock that could have been possible if it has used its own separate master clock.
If the CDP/transport clock is very good then the DAC performance improves therefore/automatically.

ani schrieb:

Through SPIDF Chord was not able to eliminate jitter and they put blame on RFI generated by the source. I take it as idiosyncrasies of highend audio designers !!! but I like that aluminum box :)

don't look for Chord (or any other manuf, for that matter)to eliminate jitter! That's hardly possible. Even the very best crystals out there (& I would know about these as it is 1 part of my business) have some jitter. These are in the pS range - far ahead of what is needed in a CDP/transport but the point is that they, too, have jitter!
Well, a bad carpenter blames his tools!
Each implementation has its + & -. The Chord's is no diff. The key is to realize those short-comings & find a within-budget work-around.
It is indeed good that you like that AL box! That's the 1st committment to making it play better in your system.
There are number of things that you can do to minimize jitter from the CDP/transport end. They all involve soldering, I'm afraid.

Thanks for pointing out the "Drift" in my explanation
. You are absolutly right it is the drift not jitter.

Regarding minimising jitter, I would be happy to try modifications even if it needs a bit of soldering. Please inform us how it can be done.

Thanking you in advance.

Anil

ani schrieb:

Regarding minimising jitter, I would be happy to try modifications even if it needs a bit of soldering. Please inform us how it can be done. Thanking you in advance. Anil

IMHO you are giving it more relevance than it really deserves

from what i can understand from you earlier post, chord has quite a good built in mechanism for minimising it.. the risk we audiophiles run is to get too techinical that we get into the realm of technical improvements which may not really translate to any perceivable sonic difference

If it is still your concern it may be a good idea to take up Amp-Nuts idea on the transport.. the only belt drive i have heard is the CEC TL series and they are pretty solid looking beasts

ani schrieb:

Thanks for pointing out the "Drift" in my explanation . You are absolutly right it is the drift not jitter.

no problem!

ani schrieb:

Regarding minimising jitter, I would be happy to try modifications even if it needs a bit of soldering. Please inform us how it can be done. Thanking you in advance. Anil

to reduce jitter
* use a permanent marker to fill in the see-thru plastic at the center hole of the CD. this prevents the laser from scattering when it hits this region.
when scattering of the laser beam occurs, the optical sense mechanism has to work harder to decipher whether or not it is a pit (0) or not (1). related to laser beam scattering is the glitching in the transport power supply which dirties the power supply for all other circuits hanging off it. This is principal reason for using a black or green marker. Reduced transport power glitching reduces jitter from the transport. It is also the principal reason that digital power supply is separated from analog power supply in well implemented transports/CDPs/DVDs.
* for the same reason use a black or green marker along the outer-most rim of the CD. The place where your thumb would grip the CD periphery.
* Use a stat-mat of your favourite kind. Several are available in the market.
Often, if you use a stat-mat you might not need to use a green/black marker as this mat often provides this function.
* damp the entire transport mechanism WITHOUT interfering w/ its smooth operation.
Often one can use damping sheets cut to size on the outside box of the transport - the big box you see w/ a class 1 laser warning once you take off the top cover.
Along the same lines (if you are jittery doing this) you can put a heavy weight like a 12"X12" slab of slate on top of the transport chassis. This is an exterior tweak.
* damp the patra top lid of the transport. Again, the less the mechanical vibration of the transport, the better off you are from a jitter perspective.
* upgrade your power supply caps to computer grade power supplies or Blackgates or Jensens or whatever you can afford.
* upgrade the digital drive output buffer opamp. This upgrade probably has the biggest impact on reducing jitter because it is the slow rise & fall times of this buffer that degrades jitter performance. For example, one Sony DVD player I know has 25nS rise & falls times @ the digital out & the upgrade reduces it to <= 5nS.
* ensure that the output impedance is 75 ohms on the digital output. Almost all digital outputs are not. This involves soldering a 75 to gnd as the buffer load. Of course, the buffer must be able to drive this load! So, correct selection of the opamp buffer is key.
* use a BNC or XLR or glass-fibre optical cable. SPDIF cables & RCA jacks do not have the bandwidth to carry digital signals without distorting them. That's why some hi-end CDPs & transports simply do not have SPDIF digital outputs!
* Implicit from the immediately above is that one should replace the RCA digital out w/ a BNC. If you have multiple digital outputs, then you are good to go.
* According to 1 CDP/transport/DAC master modifier, having the digital cable 1.5m MIMIMUM is key to avoid reflected signals from coming back from the DAC to the digital output & messing w/ the digital output buffer. It's called reduction of VSWR. This person swears by it but I don't know how technically true this is. Some digital cables are affordable enough that an additional 0.5m is not going to break the bank.
* if you are using the CDP/DVD as a transport don't bother w/ the on-board DAC clock. it's not being used - you are using the stand-alone DAC clock (if it has one) OR are depending on the data recovered clock OR depending on the filtered/averaged data recovered clock (Chord 64) to drive the DAC.
Of course, if you are using it as a 1-box player, upgrade the on-board clock to a SuperClock, Tent XO, etc.

these easy tweaks should help out a lot.
Of course do not expect a cheap DVD/CDP to start sounding like a Mark Levinson!

this post is to be made a sticky ! B'Walla has consolidated most info on damping into one place..

one more point.. many CDPs (Especially NADs and Marantz) can be further enhanced by
-Biasing the output to Class A
-Changing the muting circuit.

Having a schematic to the CDP/DAC really helps and most distributers/service centers will have a service manual with full schematics (usually available for a price)

I have personally tried out the Damping of transports and can vouch for its working. the simple stuff I could do was
1. cork Strips stuck to the internal sides of the casing with Caulk
2. Use blu tac to dampen the entire tray and the mechanism holding the disc.. you can even use Rope Caulk
3. Telephone directories over the cdd (I preferred it to slab.. although of granite)
4. 3 cones (Any material ..i got some brass stuff) as footers with one just below the tray, another below the transformer and the third at the point it can be made the most stable.

If you try searching the digital section of audioasylum to some modifications made on a classic and cheap toshiba dvd especially by a guy called vinny (Rossi) who later founded red wine audio you will get a lot of pointers in modding of sources as well.

Since subboss tends to get very offended, no more links being put in from my side

Thanks to Bombaywalla, Arjun and all contributors for this wonderful esay on jitter reduction and tweaks. Hope this thread will be usefull to a many in future.

Arjun,

I am not trying to give jitter undue importance or get it reduced to "zero" :). It all started with my quest to find a cheap transport for my dac assuming that the dac will take care of much of the transport related problems. Then came the incompatibilty problems, SPIDF not locking but Toslink of the same working fine and Higher quality SPIDF cable not working but a cheapie analog interconnect allows you to play music with lots of hard edges.

Then the designers comment of RFI and how toslink eliminates it. I have decided to try out some transports and keep the one that sound best to my ears and that of my wife, she doesnt know any thing about brands and best of all not aware of all this jitter

ani, my intention was only to point out thats the way we all get hooked deeper into the hobby for me personally, if it was only about listening to music this is not all that fun but moment a little bit of theory comes in there is something to it other than just enjoyment that brings an anticipation to music.
Thats why threads like this are the most interesting in terms of gaining some more insight into this addictive hobby !

I just have to keep reminding myself that unless one keeps to actually making any true sonic benefits it is very easy to get carried away. In fact my signature is a reminder to myself

But again it works differently for different folks.

BTW heard a small unknown band in a park yesterday playing unheard of music amplified via some PA system.

Was Amazing. and obviously there was no sonic soundstage and only a visuall one!

It is surprising how much of influence the absence of a room and as well as your mental state has on enjoying music

Live music unamplified or not is in a different league. Unknown music who are alive and kicking who cares of soundstage. Like you said you can see them, how much more fleshed can the soundstage be

Now you have opened another topic the room

a friend of mine was telling me today that he has come to a stage where he feels the room is the single most limiting factor !! But that will be another thread.

Arj schrieb:

this post is to be made a sticky ! B'Walla has consolidated most info on damping into one place..

Thanks & welcome! I hope that these tweaks will be useful to all of you. I have personally benefitted from them & can attest to their effectiveness.

Arj schrieb:

one more point.. many CDPs (Especially NADs and Marantz) can be further enhanced by -Biasing the output to Class A -Changing the muting circuit.

This is true - many of the Marantz units come w/ HDAM output stages that sound better in class-A. However, I'd like to issue a WARNING/HEADS-UP about doing this:
if you bias the output stage in class-A be SURE to HEAT-SINK that opamp or opamps! The theoretical efficiency of class-A is 25% => it'll burn 1W & deliver only 0.25W at the output. The remaing wattage will be dissipated as heat! If you forget to heatsink the opamp(s), you'll FRY your unit within 5 minutes of power-on! So, my conservative recommendation is NOT to do this unless you are 100% sure of what you are doing.
To bias the opamp(s) in class-A you need to understand the concept of "load line" so that you can bias the opamp @ the optimum operating point thereby providing it sufficient voltage swing in the "source current" & "sink current" directions. Otherwise, you'll increase the distortion to very much hearable limits!
When heat sinking, use thermal conductive glue & glue the heat sink on the top of the opamp (with the heat sink fins pointing upwards).

One more thing: I read a closing comment from "ani" saying that "It all started with my quest to find a cheap transport for my dac assuming that the dac will take care of much of the transport related problems."

To this I'd like to say: if you build a system using the philosophy stated above, you'll, FOR SURE, end up w/ a mediocre system at best!
the whole audio chain will be frought w/ problems: you'll have the DAC take up the slack of the transport, the preamp take up the slack of the DAC &/or cables, the power amp compensating for the speakers &/or speaker cables, etc, etc.
You will be layering compromise on top of compromise & will never like the overall sound. There'll always be an urge to upgrade & your friends' systems will all tend to sound better than yours in YOUR opinion!
I am NOT saying that you need to spend a megabuck on a transport.
Just try to buy the best transport within your budget.
The one thing I did emperically determine is that each budget amount has a best-in-class unit & it is just a matter of auditioning &/or discovering that unit. Sometimes it's a popular unit & often it is some under-appreciated unit relagated to the fringes of audio discussion. DO NOT always go w/ the popular vote - go w/ the one that has the fewest compromises in your budget. It involves scouring several other audio forums to do your research + as much audtioning as you can manage.

Happy Listening fellow audiophiles!

if you bias the output stage in class-A be SURE to HEAT-SINK that opamp or opamps

I am not sure how one can change the bias of an op-amp IC....

The only way that comes to mind is to take a 'dummy' resistor from the o/p amp output to the -ve supply. But that will only force the top transistor into class A mode... the lower transistor will continue in Class AB....

Or am I missing something ?

Amp_Nut schrieb:

if you bias the output stage in class-A be SURE to HEAT-SINK that opamp or opamps I am not sure how one can change the bias of an op-amp IC.... The only way that comes to mind is to take a 'dummy' resistor from the o/p amp output to the -ve supply. But that will only force the top transistor into class A mode... the lower transistor will continue in Class AB.... Or am I missing something ? :.

depends Amp_Nut. Some opamps made for the audio market come w/ bias adjusting pins on their packages that allow the user to change the output stage bias.

Ani,
do also try out the Airport Express or any of the digital wireless receivers with a Good Hard disk transport with Uncompressed tracks.. i feel they should beat any transport as if correct reading of 0s and 1s is the primary objective of a transport, nothing to beat a computer

i thought i might share this with u all..

has quite a bit of info..
lots of useful links..especially on interface jitter and the likes...

http://www.lessloss.com

arj..appreciate if u could let me know abt Airport express..anything else on the market whihc is better or comparable to the Apple airport...
this is just what i have been looking for..a perfect gift for my folks...

is it available here in India?

does teh SQ degarde with respect to the distance from the PC...?

Bombaywalla,

they claim that by making te DAC as a master clock and the transport the slave(they claim that it is how it is in pro studio) the jitter problem is minimized a lot...

How true is this?...

Savyasaachi schrieb:

arj..appreciate if u could let me know abt Airport express..anything else on the market whihc is better or comparable to the Apple airport... this is just what i have been looking for..a perfect gift for my folks... is it available here in India? does teh SQ degarde with respect to the distance from the PC...?

I ll answer points 2/3 first.

I think it should be available in the Apple store in India. or else you could see if you can buy it off the web either from singapore ( http://store.apple.c...D4&nplm=M9470FE%2FA)
Or from Ebay in Europe, from anypone who ship internationally.
Since it transmits only in digital, The sound does not degrade for upt 50 M or thru 2 walls after that the signal gets weak.

Other than that, i have not really tried it but it can also be used to create a wireless network at home or extend the range of an existing wireless network.

the other options in the genre, more expensive but with more features and simpler to setup are the
1. Slim devices Squeeze box
2. Roku soundbridge.

Both have apparently a better built in DAC and internet radio as well as easier to set up.

the AE is a pain in the you-know-what to setup from a windows machine but is small and handy and very good if you already have a DAC.

Savyasaachi schrieb:

i thought i might share this with u all.. has quite a bit of info.. lots of useful links..especially on interface jitter and the likes... http://www.lessloss.com

thanks for that link.. pretty interesting stuff.

but whatever you use as a slave the idea is to have a good clock.. whether u use a good clock in the transport and master the DAC with it, the converse or altogether a different approach of using an extenal clock as a master with the transport and dac as a slave.. something which dCS does with its Verona

Savyasaachi schrieb:

Bombaywalla, they claim that by making te DAC as a master clock and the transport the slave(they claim that it is how it is in pro studio) the jitter problem is minimized a lot... How true is this?...

100% true!
They make the DAC clock is precision clock using some low PPM xtal or even TCXO (unlikely because it's a home environment where air is well controlled but some might even use this option! Jacks up the price 'cuz a temp controlled xtal is more expensive.) & then have several outputs. One output drives the DAC & the other the transport. They need to ensure that the clock phases are synchronized for which they might use a small FIFO/buffer or some meta-stable ckt using D flip-flops to synchronize clock domains.
SOme hi-end units like Esoteric/TEAC, Wadia, Weiss Media, EMM Labs, Ensemble, Mark Levinson, Goldmund, Gryphon (& pro/studio gear, as cited by you) use this technique.

Arj schrieb:

but whatever you use as a slave the idea is to have a good clock.. whether u use a good clock in the transport and master the DAC with it, the converse or altogether a different approach of using an extenal clock as a master with the transport and dac as a slave.. something which dCS does with its Verona

correct, the imp thing is to have a good clock.

Re. the dCS stuff: the user could ruin their work by using a bad cable! In a system as hi-end as dCS, the effect of cables will/should be quite hearable.
Maybe dCS supplies their own cable/hose to connect the transport & DAC?

Bombaywalla Wrote :

depends Amp_Nut. Some opamps made for the audio market come w/ bias adjusting pins on their packages that allow the user to change the output stage bias.

Apologies for persevering on this line .... but op amps are my favs... :-)

Could you name me a couple of such op amps ?

I have been searching the net...

Thanks for your help.

Cheers

Amp_Nut schrieb:

Bombaywalla Wrote : depends Amp_Nut. Some opamps made for the audio market come w/ bias adjusting pins on their packages that allow the user to change the output stage bias. Apologies for persevering on this line .... but op amps are my favs... :-) Could you name me a couple of such op amps ? I have been searching the net... Thanks for your help. Cheers

Amp_Nut,
try this link:
http://focus.ti.com/lit/ds/symlink/ina103.pdf

look @ Fig 11. Not exactly how I wrote it but quite close.

Bombaywalla Wrote :

depends Amp_Nut. Some opamps made for the audio market come w/ bias adjusting pins on their packages that allow the user to change the output stage bias. try this link: http://focus.ti.com/lit/ds/symlink/ina103.pdf look @ Fig 11.

Thanks for the link, Checked out the datasheet in detail, particularly Fig 11.

Nah, it has Nothing to do with Increasing the IC Op Amp's output stage bias, to run in Class A, for improved sound quality.

Infact the circuit of Fig 11, in Very old ( as old as the first commercial IC Op amps like the 741 and the 301 ).

It is infact a Class AB ( or much worse ) Current booster output stage... Precisely what Audiophiles would want to avoid today...

Peter Walker often used a similar block diagram to explain his "Current Dumping " amplifier principal, used in the Quad 405 Power Amp.

In the Fig 11 Circuit, The 2 External transistors Turn on HARD when tyhe voltage accross the 100 Ohms resistor exceeds 0.67 Volts. Then it is left to the Op Amp's feedback to correct for this Gross Non liniarity. This is the Anti Thesis of today's low feedback designs.

This IC Op amp has No facility to set its Output stage bias current.

If you do know of an IC Op amp that allows external setting of its output stage bias, do let me know.

As far as I know, its not possible to Tweak Op Amp's performance this way.

One option is to run the Op Amp closer to its absolute Max Voltage rating. That increases the Output Transistor bias current, repotedly improving the sound quality. But the higher voltage means the op amp is being operated on the brink of self destruction, that is why a commercial product designer will not retort to this....

Amp_nut,
have come across this in the following links..

from what i understand this biasing is not via any pins in the opamp but via a resistor in the output to ground to ensure full cyle conduction..

http://www.zerogain.com/forum/archive/index.php/t-547.html
http://tangentsoft.net/audio/opamp-bias.html

perhaps you could comment on how possible the above is..

Hi Arj,

Thanks for the 2 links.

Yes, both links talk of the same thing, viz : Connecting a resistor from the output to the -ve supply.

This bleeds current thru the upper ( npn ) transistor, forcing it to continiously conduct current all the time into the external resistor, even when there is no output. This means the npn transistor is operating in Class A.

UNFORTUNATELY, the other half of the circuit, ie the pnp transistor is NOT in Class A !

Like a man limping with 1 good, and 1 bad leg

One of the references that you have provided, also admits this :

The reason we connect the current source to V- is that it forces the NPN transistors in the op-amp to remain active instead of the PNP ones. Biasing the output to V+ would also work, but in general PNP transistors don't behave as well as NPN ones, so we'd rather make the NPN's do the work.

This resistor trick is actually Very old, I came accross it, for the el-cheapo LM 324 Quad (4) IC opamp, 20 years ago, this 4 op amp package then sold for Rs 8, and had BAD crossover distortion ( without the external resistor).


Incidentally, I had mentioned this in my post #37 posted on May 21. 2006, 18:23

Amp_nut said : I am not sure how one can change the bias of an op-amp IC.... The only way that comes to mind is to take a 'dummy' resistor from the o/p amp output to the -ve supply. But that will only force the top transistor into class A mode... the lower transistor will continue in Class AB.... Or am I missing something ?

Cheers !