The DIY'rs Cookbook

Dec 2, 2015 at 1:59 PM Post #151 of 1,974
 
Which one did you get? Schiit's Ragnarok is being marketed for both "some" IEMs and desktop speakers. It should drive the HE-6 with watts to spare. And apparently should work just fine with efficient headphones like Grados.

The ragnarok is pure class A up to about 4.2watts per channel I believe, then it switches to AB. It is a beast, but the volume control was not nearly as nice feeling or adjustable as the Audio-Gd Master11's implemenation of a super exponential 100 step relay controlled volume. Similar, but not linear, more steps, smoother, no feedback issues etc etc.
 
 
I've tested some very sensitive IEM's on it with no issues. The volume control is quite nice.

16000MW /  25 ohm 

8000MW /  50 ohm 

 4000MW  /  100 ohm

1300MW / 300 ohm

650 MW / 600 ohm
 
 
1ohm headphone output and 19V output (balanced)
I opted to get the +6db gain option, for a total of +22db gain. tons of headroom, even for an HE-6 the standard +16db is plenty.
 
 
beastly? some are running speakers from it that are >95 db efficient. direct from the dual 3pin XLR's from the front.  
 
I was the second person to order the master11 from audio-gd. I just got it at the last meet when Atomic Bob grabbed that headphone cable from me @ the library meet a few months back.  New stuff, but somewhat like a Master 9 and Master 7 together in one. 
 
Dec 2, 2015 at 4:12 PM Post #152 of 1,974
  The issue I have with using power to decide suitability of matching amp to electro-acoustic driver is that it doesn't sufficiently describe the necessary parameters. One driver may require high voltage and low current to perform optimally. Another may want the opposite, low voltage and high current. Yet both might have the same power required. I will attach a graph in another post demonstrating the needs of various popular headphones. Using the published nominal impedance and sensitivity for the HE-6 would require 50 mW to achieve 100 dB SPL average with 500 mW for 110 dB SPL peaks, assuming a 10 dB crest factor in the recording. But what the HE-6 really wants is 14Vpp with 100mA peak to hit the 110 dB SPL without strain.

 
<edit> I forgot to answer the question. I think the PSIII would be marginal at best for the HE-6 unless one listens at lower sound levels. My experience at meets is most listening to planars also listen VERY LOUD. Thus the discrepancy in adequacy of a given amp for a given listener. I probably could use the PSIII for an HE-6. I listen at 75 ~ 85 dB SPL in a quiet environment. Most others would find the PSIII inadequate and even the Asgard2 inadequate to achieve 110 dB SPL avg with peaks to 120 dB that I observe them listening. I built a SPL meter dummy head to quickly verify where listeners set their auditioning sound levels in my acoustic lab. It is based on the work of Joop Nijenhuis Headphone sound level meter.


Thank you for the explanations and the formulas. Very helpful.
 
I think PSIII is unlikely to drive HE-6 with ease, and Garage1217 markets Project Ember II instead for such heavy duty. Given the price point, this seems like a very attractive proposition. In the Schiit stable, Ragnarok seems to fit the bill, but it gets to the $1.7K price point. The advantage of Ragnarok is that it could also be used with speakers like the KEF LS50.
 
Dec 2, 2015 at 4:53 PM Post #153 of 1,974
Thank you for the explanations and the formulas. Very helpful.

I think PSIII is unlikely to drive HE-6 with ease, and Garage1217 markets Project Ember II instead for such heavy duty. Given the price point, this seems like a very attractive proposition. In the Schiit stable, Ragnarok seems to fit the bill, but it gets to the $1.7K price point. The advantage of Ragnarok is that it could also be used with speakers like the KEF LS50.


There is a guy powering his kef LS50's with the master11 who is on the headfi thread for the m11. Just fine. I got my m11 for 1.7K and it has a pretty good DAC with it too :) so maybe it's not perfect since it's not made in the USA and some will turn their nose up at chi-fi. But I know kingwa takes care of his employees considering it's a Chinese company. I own two audio-gd products. master11 and nfb28(2015 edition) and used to own the dac19 10th aniv.(sweet Dac before the big upgrade to m11) my next Dac may be a Rockna Wavedream Signature (dreaming :)) if I was rich.


Sorry for getting off topic a bit. I am very interested in learning more about amps and psu's. Certainly want to find a tube amp that can power a wide variety of cans, and do it well.
I have a pair of he1000's from Sergio right now, and a pair of hd800's from Ryan, I'm doing a bit of testing on my m11 with them. I am interested to try out the editionX hifiman's if they sound like the he1000. I like how natural the HEK's sound, but the price is insane imho. But the editionX might be possible. Both the hd800 and he1000 sound incredibly good yet also quite different. Has anyone heard the editionX's? I'm hoping they sound the same as the he1000. The he1000 is a tad more efficient than my he560's and I do like that. The hd800's are really easy to drive yet the soundstage is great yet hard to get used to. I don't like the comfort levels of the hd800's imho. Nor do I like the heft of the he1000's. I am guessing the editionX will be comfy like the he1000, yet not heavy! So it should be ideal... For me. The hd800's feel cheap as if they would break easily, yet the sound quality makes up for that. Some may not care for my opinion, but it's hard to say anything bad about these headphones. Honestly.

Would the project sunrise III be able to power the he560 to 100db maybe even 110db? I really like these headphones and figure the he1000 and editionX would be able to be powered if the the he560 can. I "might" consider an upgrade in the future. A DIY kit for the psIII sounds interesting to me. I need to just listen to one :)
 
Dec 2, 2015 at 10:04 PM Post #154 of 1,974
snip

Would the project sunrise III be able to power the he560 to 100db maybe even 110db? I really like these headphones and figure the he1000 and editionX would be able to be powered if the the he560 can. I "might" consider an upgrade in the future. A DIY kit for the psIII sounds interesting to me. I need to just listen to one :)

According to the power output vs. impedance graph of the PSIII and using the 1.5Ω output setting this amp delivers ≈ 750mW.
And according to atomicbob's graph, your he560 cans 'need' 10mW to 100mW (100dB to 110dB SPL respectively).
So reaching 100dB should be no problem.

However the sunrise can only output ≈ 5.4 volts MAX (and that rating is into no load).
They don't specify if that voltage is P-P or RMS, but the Horizon amp rates this voltage as RMS so it probably has enough headroom to reach the 110dB range.

However if you want plenty of 'extra' headroom, the sunrise probably won't provide it, nor will it power more hungry HP's to 110dB+ levels (where the peaks of the music exist).

JJ
 
Dec 2, 2015 at 10:12 PM Post #155 of 1,974
According to the power output vs. impedance graph of the PSIII and using the 1.5Ω output setting this amp delivers ≈ 750mW.
And according to atomicbob's graph, your he560 cans 'need' 10mW to 100mW (100dB to 110dB SPL respectively).
So reaching 100dB should be no problem.

However the sunrise can only output ≈ 5.4 volts MAX (and that rating is into no load).
They don't specify if that voltage is P-P or RMS, but the Horizon amp rates this voltage as RMS so it probably has enough headroom to reach the 110dB range.

However if you want plenty of 'extra' headroom, the sunrise probably won't provide it, nor will it power more hungry HP's to 110dB+ levels (where the peaks of the music exist).

JJ

 
well I have the HE1000 and HD800 now, as well as the Oppo PM3 and HE560...boy would it be nice to try the PSIII and just hear for myself. HAH.
i would imagine the HE1000 would sound slightly better on the PSIII than the HE560. The volume levels on my M11 vary slightly when I listen. probably about 6-8 steps less with the HE1000. But not really relevant I suppose. Just that the HE1000 doesnt need as much power to sound amazing. So far I prefer the HE1000 quite a bit over the HD800. Albeit i've never heard an HD800 sound this good. I've listened to them on a ROK/YGG before, ZanaDuex and I think they sound better on my setup.  I guess I have other gripes about the HD800's. But overall they are a pretty nice headphone.  Neither of the flagships are perfect though. 
 
 
I would want a PSIII for work most likely, or just a secondary listening station in the living room. I just need to find a pair of headphones I like to pair with them. I really do want to have a good affordable tube amp. :D
 
Dec 3, 2015 at 3:50 PM Post #156 of 1,974
  Here is a table comparing the Vrms, Vpp, Ima and Power to achieve 100 dB SPL average and 110 dB SPL peak for various popular headphone, based on published specifications and nominal impedances. This ignores any impedance bumps a given headphone driver may have.


Looking at the chart it's obvious that compared to other flagships the HE6 is aeons behind in terms of driveability, yet the Hifiman is often revered as reference SQ. KEF's LS50 is also horrifyingly inefficient with 85dB/mW, but it too gives outstanding SQ given its price point.
 
Is there some engineering trade-off that makes such inefficient transducers output incredible SQ? And if so, why aren't other manufacturers making use of the trick?
 
Dec 3, 2015 at 11:39 PM Post #157 of 1,974
What and how do we know what IS ‘Better’?
or
It’s all in our heads, or is it?

Part 6 Feel the POWER, luke…

Acoustic POWER vs volume

Musicians often say our playback systems just aren’t the same as what they experience…

Why?

My answer is that the instantaneous acoustic power of an instrument IS what the musician hears.
It’s not simply frequency response or just simple dynamics, it’s the air being vibrated by a device that is designed to do just that.

Musical instruments MOVE air.

This requires POWER.
And HOW that power is distributed (how much acoustic power at this frequency and increment of time vs. how much at this other frequency and slice of time) as it vibrates the acoustic space IS what we hear. The resonances and harmonic series created by each instrument IS the ‘identity’ of that instrument, it’s ’signature’, if you will.

We use SPL (Sound Pressure Level) meters to measure and tell us how loud it is.
But WHAT we are really measuring is sound PRESSURE, and pressure is a measure of power, as in movement of ’stuff’ (the air) over a unit of time (frequency).

And how well our playback systems deliver acoustic power, power that accurately matches the original power signature, is a measure of how accurate and precise the playback system truly is.

We can attribute this to dynamics or a balanced frequency response or many other aspects, but really it comes down to acoustic power as it is applied in precise and sometimes minute amounts for the entire frequency spectrum and perhaps more importantly to ONLY apply power where it is ’supposed’ to be.

When it’s creation and delivery is not controlled enough such that it ‘spills over’ and gets smeared thru time we hear this as Listener Fatigue. This smearing of the re-created acoustic power causes much distortion and makes our acoustic experience less ‘real’ as a result.
But the single ‘biggest’ difference is in the amount (or lack) of power that is experienced and specifically how well that re-created power is aligned to its source and presented to our ears.
Which is why musicians repeatedly state ‘It isn’t the same’.

Our playback systems often lack this degree of coupling and instantaneous power delivery, in the same manner that a ‘natural’ acoustic instrument will.
IOW we can turn the volume (DRC) up but without this coupling and the proper timing of the power as it is applied to our re-created signal, all we get is LOUD, which just might be another way of describing Listener Fatigue.

Another interesting observation I made during these experiments was that as the tLFF was improved and as I could crank up the DRC to greater amounts before discomfort kicked in, I noticed the perception that the SPL had been reduced at any particular DRC setting.
IOW the volume seemed to drop at any given setting of the DRC as tLFF was improved due to a ‘Better’ alignment of the acoustical power.
Which in turn meant that the acoustical energy that was presented to me was due to a more coupled signal which resulted in the perception of a lower SPL.

I attribute this observed change in volume to a reduced amount of smearing of the acoustic power being presented to me.
IOW the dynamic peaks were ‘peakier’ and the low level dynamics were lower as well, with less acoustic power being smeared. or ‘wasted’.

And let me be clear in my use of the term “discomfort”.
I do NOT mean pain, of any sort, kind or type.
Discomfort is when I detect any form of irritation WAY before any sort of pain threshold is even approached.

So, when the inherent acoustic power that defines an instrument is optimized during its re-creation and then delivered to our ears, this coupling of the acoustic power to and with us, can become magical.
IOW when we experience such a degree of coupling to the music we like, there is this thing that can happen.

I call it a CNST (Central Nervous System Tap) and it is an experience that is simply unforgettable.
It can provide the individual a degree of ‘calibration’ with what is possible in terms of how ‘real’ a musical experience truly can be.
I’ll go more into this in a later post.

Suffice it to say, the totality of this level of musical experience is beyond words as it encompasses and immerses us IN the music.

JJ


End Part 6
Next up - Jitter, it’s not just the coffee anymore.
 
Dec 9, 2015 at 2:18 PM Post #158 of 1,974

JJ Special Power Cable Update:

After having my new JJ envisioned power cables for a few months now, the enriched SQ has become the norm so I have been afraid to A/B the difference. Per JJ's request, I did just that and wow, there is no going back. In general, the feeling was somewhat like using my DAP with my BH2 amp added for more dynamics, bigger sound stage, better texturing, etc. and then trying to go directly from my DAP without the amp. It just sounds small and lifeless now. I have to turn up the volumes to painful levels to try to get the expected details so listening becomes fatiguing. It is the same thing in my desktop and full sized speaker setup.
 
A new discovery downstairs in the entertainment center with my new power cable. My 1-year old loves to use the cables going to my receiver to pull himself up and to play with the entertainment center buttons - which is constantly disabling my setup. Unknown to me, he disabled my dual subs which is usually pretty obvious. However, this time I couldn't tell until a bass head song came on that was under powered in the 10 to 20 hertz frequency range. Upon investigation, my subs were not playing, but the Maggies were moving strong air in the 40 to 60 hertz range that was very sub like. The Maggies typically roll off at about 80 hertz with most frequencies under being inaudible - hence the need to pair with dual subs. Being mid focused and with a huge diafram, they don't typically move air like a smaller traditional speaker, they create lifelike frequency resonances the simulate a real environment. Now they are moving strong air at lower frequencies. Switching the power cable out on the amp for fun, this goes away. I have to say that this is a huge improvement in the Maggie performance. When I get time to figure out the subs, I may have to dial them down to balance out the system or it may just be more fun like it is. Can't wait to figure this one out.
 
Conclusion - power cables make a difference!
 
Dec 9, 2015 at 2:58 PM Post #159 of 1,974
 

JJ Special Power Cable Update:

After having my new JJ envisioned power cables for a few months now, the enriched SQ has become the norm so I have been afraid to A/B the difference.
 
[...]
 
 
Conclusion - power cables make a difference!

Which power cables are you using?
 
On that note, I have been eyeing the offerings by Cullen Cable (http://www.cullencable.com/) as inexpensive alternatives to the eye-watering mainstream cables out there but even his modest prices are a stretch at this point with two small kids in the budget mix.  Does anyone have any experience with inexpensive offerings they can share?
 
Nedifer 
 
Dec 9, 2015 at 4:59 PM Post #160 of 1,974
 
 

JJ Special Power Cable Update:

After having my new JJ envisioned power cables for a few months now, the enriched SQ has become the norm so I have been afraid to A/B the difference.
 
[...]
 
 
Conclusion - power cables make a difference!

Which power cables are you using?
 
On that note, I have been eyeing the offerings by Cullen Cable (http://www.cullencable.com/) as inexpensive alternatives to the eye-watering mainstream cables out there but even his modest prices are a stretch at this point with two small kids in the budget mix.  Does anyone have any experience with inexpensive offerings they can share?
 
Nedifer 

These are DIY that JJ helped me with that are standard Home Depot copper - don't know the guage or ends, that are cooked and cryoed. JJ can fill us in on the specifics.
 
Dec 9, 2015 at 9:50 PM Post #161 of 1,974
These cables are made using the BottleHead ac power cable 'design', I just used better materials, (all cooper 16awg TFFN wires, and cheap chinese knock off rhodium cable connectors).
And I cryo'd them and then cooked the cables on an industrial strength cable cooker (not a cd playing thru the system).

This combination of stuff and treatment was the first truly spectacular set of cables I had with my system and they remain a reference to judge other cables by to this day.

If memory serves the cost of materials is like under $50/cable with the cryo treatment adding ≈ $40± and the cooking can be big bucks (mostly because of the cost of the cooker (≈$1K), unless you know someone who has a cooker, like me… :atsmile:


JJ
 
Dec 9, 2015 at 10:11 PM Post #162 of 1,974
Which power cables are you using?

On that note, I have been eyeing the offerings by Cullen Cable (http://www.cullencable.com/) as inexpensive alternatives to the eye-watering mainstream cables out there but even his modest prices are a stretch at this point with two small kids in the budget mix.  Does anyone have any experience with inexpensive offerings they can share?

Nedifer 
Another cable company that has my attention, and no I have not listened to their cables, is
http://www.blacksandaudio.net/power-cords/

Yes their prices are higher but they are 'treating' their cables much in the same way I do for the cables I now use.

If I were to sum up in a single concept what seems to be most important, it would be to lower the thruput resistance for the entire cable.
From the duplex receptacle thru to the IEC connector on the back of the amp, dac, etc.
IOW it's not just the cable itself but the ability of the cable to pass current as 'efficiently' as possible to the load, whatever it is.

JJ
 
Dec 11, 2015 at 5:55 AM Post #163 of 1,974
What and how do we know what IS ‘Better’?
or
It’s all in our heads, or is it?

Part 7 ‘Jitter’, it’s not just the coffee anymore.


The last post was a short one, this however is another long one, so we’re diving into the deep end of the pool again.

‘Jitter’
This is not a technical analysis of Jitter plots nor a ‘scientific’ examination of what Jitter is, nor what it isn’t.
Instead I’m going to delve into what ‘Jitter’ is and what I have come to understand its effects and influences are,
AS I HEAR THEM!

Jitter, we all know it’s ‘bad’ to have too much (any?) Jitter in our digital playback system. But is ‘real’ Jitter limited to just the digital domain exclusively?
I submit that ‘Jitter’ is more than just a digital timing ‘error’ during the sequential re-construction process, indeed I find this to be only 1/2 of the ‘total’ picture.

I submit, at least for this ‘Better” series, that a definition that what we call ‘Jitter’ is ANY resultant error (deviation) during the re-construction of (from) the ‘original’ acoustic waveform signal. This definition encompasses both ‘Domains’ (analog & digital) and ‘Dimensions’ (frequency/time & voltage/amplitude) needed to re-construct and deliver the original ‘target’ signal.
I come to this conclusion based upon lots of observation and fussing with the knobs (experimentation).
But I also submit that ‘Jitter’, is in a sense, a displacement ‘error’ away from the perfect re-construction of the original wave form.

Perhaps another term could be introduced such as smear, or re-construction deviation, etc. But for now I’m going to stick to ‘Jitter’, mostly because the audible effects of reducing any of these errors results in near identical audible improvements, regardless of which ‘Dimension’ or which ‘Domain’ is being improved. And it’s these same audible consequences which I have come to identify and define as to what ‘Jitter’ is and what we can expect as any amount of ‘Jitter’ is reduced.

All of these various sub-sets of ‘Jitter’ have a unique variation for how ‘Jitter’ is introduced into the signal we hear.
And really we as audiophools can only affect Jitter reduction in rather limited ways. Sure we can buy new pieces of equipment, but other than that all we can change is the setup of the system, which includes the cables, and the ac power we feed the equipment, and more. And fortunately these sorts of changes can result in significant sonic improvements which have sonic characteristics that match other improvements made to improve Jitter directly, which is why I have come to equate all of these improvements to ‘standard’ digital Jitter.

So Domains, Dimensions?
What?
Let me explain a bit further…
And for the sake of this "Better" series I categorize our audio systems thusly…

DOMAINS
Currently there are 2 ‘domains’ of signal re-creation methodologies in our playback systems,
analog (tape, vinyl, radio)
and digital (CD’s, DVD’s, digital files from a computer or server).
And as indicated above most don’t associate analog with‘Jitter’, since the term was introduced with DACs, but I hear the affects ‘Jitter’ has in both of these types of 'Domains'.

DIMENSIONS
The 2 ‘dimensions’ of (frequency/time) & (voltage/amplitude) (think of an FFT display and a signal trace on an ‘O’scope) are both necessary aspects of any musical signal.
The frequency & voltage pair applies more to the analog ‘Domain’, while the time & amplitude applies more to the digital ‘Domain’. But these aren’t hard and fast associations, by any means.

In the analog ‘Domain’ the signal is based upon a real time complex dynamic analog signal (voltage), with what we usually term as phase shift (frequency) anomalies due to the nature of the circuits and the physics and the properties of the materials used, and of course the setup of the equipment etc.

In the analog ‘Domain’ we have seen a significant increase in the development of power supply sophistication with multiple layers and levels of regulation and control. And this technique is ‘generously’ applied to the digital circuitry used in dacs and other signal processing equipment.
But not exclusively.

And in the digital ‘Domain’ we have been hearing of and seeing developments of femto-second clocks in the push or attempts to obtain the optimal degree of precision, repeatability, and reproducibility in the time axis ‘Dimension’ of the equation.

I have heard the results of the reduction of digital Jitter made to the incoming digital stream and have ‘categorized’ these improvements in a descriptive narrative.
In the digital ‘Domain’, where timing is it’s ‘thing’, we find that as the degree of exact precision of the timing increases, the results, in complex ways, can be quite audible, but again these sonic changes/improvements are very similar to those where the power supplies ability to delivery precisely the correct voltage is also improved.

So these timing/voltage/frequency/amplitude improvements all share similar sonic characteristics when they are improved or where ‘choke points’ that are associated with them are improved.

And these very same audible changes/improvements are also brought about via, albeit more indirectly, ac power system delivery changes/improvements. And some of these changes are from ‘non-ordinary’ sources as most would view them.
IOW as ‘choke points’ that affect both the domains and dimensions are improved/removed, the acoustical net results have very similar sonic consequences.

So next we will delve a bit deeper in the signal re-creation modes AND the signal delivery modes, and look at how they overlap.

Analog signal creation
Is your phono cartridge diamond tip in THE groove? or just sorta close?
And can you tell when it ‘drops into’ THE groove?
When the diamond tip doesn’t read the groove exactly as it was created, time smearing is the result due to not being in phase and ‘correctly’ reading the original cut in the groove in ‘real time’.
A ‘proper’ read of the groove, that matches the line that was cut by the cutting head, can result in a near ‘Jitter’ free analog signal with little to no timing (phase) nor amplitude (voltage) ‘errors’, assuming the rest of the system is up to the task.

This is a tall order, to be able to ‘nail’ reading the one and only correct portion of the groove, in real time, despite the physical variables of LP thickness, or warpage, or lack of ‘flatness’ or eccentricity, etc. of every record. Not to mention the additional variables of cartridge tip to body alignment, as well as ‘ideal’ cantilever ‘sag’ and the aging of the cantilever mounting viscoelastic material, among other factors.

This difficulty, in (mis)reading the groove, translates into timing and amplitude errors with respect to transferring the physical waveform from the groove wall into an identical mirror image electrical signal. Any deviation of this waveform from the ‘original’ is ‘Jitter’ as I have described it above.

Digital signal creation
Traditional Jitter (the strictly digital kind) exists as a frequency or time dimension ‘error’.
(Yes it’s more complicated than this simplified metric, but it will suffice for this write up.)
But I submit that ‘Jitter’ exists in 2 'Dimensions' (time/frequency) & (voltage/amplitude) and extends into a 3rd element, (‘real time’) as an additional factor.

So really, what is the usual understanding of Jitter? and where does it come from?
But first lets take a quick look at digital circuits.
Digital circuits are just optimized analog circuits that get triggered to ‘go off’ (or on) as ideally as possible.
And since they are operated in a sequential fashion, the sequencing itself becomes yet another issue, at least for dac’s anyway.
So they ‘switch’ to the ‘next’ state as quickly, precisely, and repeatably as possible.
But they also need to ’stabilize’ at that ‘new’ state with equal (or better) quickness, precision, and repeatability.
So we need ‘good’ fast, clean, noise free switches (the time domain) and we need a power supply that mirrors with equal quickness, precision, and repeatability as it delivers the exact voltage (the voltage domain) thru the switch(es) which then becomes the analog signal.

Put another way, the timing variability of the master clock as it is used to trigger the switch(es) to ‘dump’ (add) the precise voltage supplied by the power supply distribution system which is used to re-create the exact amplitude at that exact time, is what is being improved to more precisely mirror the original analog signal.
To me this is why we have Jitter in the first place, as it’s a means of describing how to ‘measure’ the time/frequency deviations in this re-creation process which along with the other variable(s) play a critical role in the re-creation process.

And the 3rd element is to perform this re-construction process, without deviation, with this same degree of precision, repeatability and reproducibility, dynamically, in real time, continually.

And then once we have re-created this signal we need to ‘move it along’, and then ‘present’ it to our ears. This is where the wires and other ac power supply systems come into play, along with the rest of the setup of the entire system.

Analog signal delivery
Many are VERY familiar with the possibility and results of changing the sound by changing both the power and signal cables. A great many theories abound and are bandied about, with relish by some.

And in my experience cables can alter the analog signal voltages that pass the musical signal along, and so can the power cables used by the components themselves. And in many cases these changes reflect the same sonic effects heard as does reducing ‘Jitter’, which can increase the overall quality of the sound we hear.
There are many factors such as slew rate, phase shift, signal propagation down the wires (and in some cases return signal reflections), ampacity of the cables and many more widely known factors all of which can have audible effects.

Digital signal delivery
Strangely enough digital cabling does (or can) make an audible difference. And beyond just the crude, ‘does the music skip or hiccup’, digital signal transfer from one device to another can be effected by ones choice of cabling. And I know according to some, cables, especially digital signal cables shouldn’t make ANY differences.
Except that’s not what I’m hearing, but that isn’t the full extent of all of the improvements made from seemingly unlikely or ‘impossible’ sources.

IOW whether it’s improving the timing of the delivery of the digital signal to its destination, or improving the delivery of the ac power pulses to the power supply, or transferring the analog signal from an output stage to an input stage of the next device, reducing the time and amplitude ‘errors’ of these signals results in audible improvements which have remarkably similar net sonic effects.
And in some cases, the mutual beneficial interaction between associated gear can sound like analog coloration shifts, which was quite a surprise when I first encountered it.

What I am hearing in most cases due to these reductions of ‘Jitter’ are those very attributes I have already brought up and described in my previous missives.
Namely …
Small Signal Dynamics (which most are already familiar with)
C3 (Cohesion – Coherence – Coupling),
T3 (Toe Tapping Time),
HB&W (Head Bobbing & Weaving),
Spooky/Scary,
& more.

As my experiments continued, all of these attributes tightened up, came into sharper focus, produced greater resolution at each level and also influenced each other in a holistic fashion (sympathetically), all the while the musical performance continued to become all the more compelling.
 The net result was an ever increasing sense of realism, palpability, and reduction in my sense of disbelief (it became harder to nit pick).
Which is nearly the same description I used after I heard the digital Jitter reduced at the DAC.
Only more so.

It seems as though as further improvements are made, as the ‘choke points’ are eliminated, or their severity reduced such that they no longer act as a primary sound quality limiter, the cumulative benefits add up to more than just the sum of the constituent parts.
 New intonations and harmonically related aspects of each ‘voice’ ’snap’ into place, and the complete tonality of the ‘voice’ becomes much more evident.
In short there,
IS…
MOAR,
there, there.

It’s like tweaking any system.
As the roughest edge’s are smoothed, the greatest amounts of change/improvements are noted first.
As additional improvements are made, as the remaining ‘choke points’ are further reduced/eliminated, the degree of change may not be as great but the overall level of SQ keeps increasing with greater degrees of resolution and efficiency, all with reduced ‘error’.
This can have unexpected results in terms of hearing additional sonic nuances that have Never been noticed before.
IOW it’s like hearing your music anew, again.

Or put another way, as the total accumulation of ‘error’ continues to be reduced, and more of the original re-created signal is ‘allowed’ to be presented to our ears, the greater the degree of satisfaction is experienced with more of the creativity of the musical intent being heard.

Or think of it like looking thru a telescope, or microscope, or binoculars.
As the view comes more and more into focus ALL of the details become obvious and immediately apparent as to what they are (shadow?, reflection?, etc) what they belong to (what IS the parent object?), how they belong within the view of what is seen (the nature of the relationships), and more precise 3d spatial relationships come into better focus (in front or behind? bigger or smaller, by how much?), how sharp is the ‘outline’ of each object (how precise are its optical limits?)
And more…

And further, as the optics are improved even more, (as the overall system’s resolution increases) and the distortions (error’s) in the field of view (presentation) are reduced or eliminated still further, the changes to and additional available information that result from these improvements, means there is more there, there.
And what there is there, is all the more apparent and obvious.

JJ

End Part 7
Next up,
Getting Calibrated and CNST
 
Dec 12, 2015 at 7:48 AM Post #164 of 1,974
Digital signal delivery
Strangely enough digital cabling does (or can) make an audible difference. And beyond just the crude, ‘does the music skip or hiccup’, digital signal transfer from one device to another can be effected by ones choice of cabling. And I know according to some, cables, especially digital signal cables shouldn’t make ANY differences.
IOW whether it’s improving the timing of the delivery of the digital signal to its destination, or improving the delivery of the ac power pulses to the power supply, or transferring the analog signal from an output stage to an input stage of the next device, reducing the time and amplitude ‘errors’ of these signals results in audible improvements which have remarkably similar net sonic effects.
 


I suspect that the key to this conundrum is throughput performance: do all the bits get to the destination in time for D/A conversion, or not?
 
To use the somewhat unrelated example of television, most of us are familiar with the difficulties of analogue TV: when there are interferences along the network (e.g. a storm), the image on your TV set may get grainy, but even so the signal is almost always passing "in full" (i.e. you can always see "something" decipherable). In countries that have made the switch from analogue to digital television, however, most everyone has experienced a much more horrid experience when there are interferences along the network: the image that you are getting may not be watchable at all. When this happens the audio either mutes or becomes a cacophony of garbled, periodically reproduced strident sounds; whereas the video appears as 1/4th of the top-right screen now, 1/6th of the bottom-left screen in 2sec, 1/2 of the screen in one second further, etc., and all else in green or black. Basically, the delivery of bits (prior to the DAC!) is COMPLETELY broken down. The throughput performance is zilch for all practical purposes.
 
And it should not really matter if ALL the bits get delivered and checksumed "at some point", since some bits will be delivered late. Generally TV is bound by time constraints (as is listening to audio), and once the moment for displaying a particular frame is passed, it's passed for good. Bits that get lost for any reason and are not delivered in time do not make the cut, and at that moment the DAC must decide whether it converts and displays something (e.g. replacing undelivered bits with 0s) or nothing at all (e.g. displaying black screen until hopefully the signal improves, and then retry). Dropped packages can happen for a variety of reasons, not least for exceeding real, available bandwidth.
 
If these issues can happen in the digital domain in such dramatic fashion (with obvious conscious effects), then it is not at all inconceivable that similar yet more subtle "delivery timing" difficulties arise elsewhere in digital applications (with potentially less easily noticeable, and perhaps subconscious effects). "Bits are bits" when timing is of no urgency, as when downloading a file: if bits (packages) are delayed or lost, the user can wait until they're delivered/re-fetched and checksums performed. However waiting is not a luxury available in instantaneous video/audio applications. When timing is critical, as all of us have experienced at least once in their lives when streaming videos on the internet, bits can get lost, buffers depleted, and the instantaneous video/audio experience may pause or even break down.
 
Here's what FiiO mentions in one of their manuals:
"1. For desktop computers, connect the X3 to a USB port on the rear of the case, not the
front. This is because front USB ports on desktop computers are connected to the
motherboard via long unshielded cables. As a result, front USB ports have low power
output
, low transfer speeds and are prone to electrical interference, degrading sound
quality or causing crackling / dropouts
.

 
2.When the X3 is decoding 192k/24bit audio via USB, the required data throughput is
huge. A high quality USB cable, with low signal loss and good shielding, is required.
Please use the original USB cable provided with the X3 if possible. If you need to
purchase another cable, please do not buy generic USB cables (including cables
provided with most smartphones), but buy USB-to-micro-USB cables from a reputable
brand supporting the USB2.0 standard." (emphasis mine)

 
Hardly a discourse suggestive of "bits are bits" and "all USB cables are the same"...
 
Dec 12, 2015 at 8:45 AM Post #165 of 1,974
As I'm reading through these recent posts (and looking around at unused cables) I wondered for the first time what differences are built into interconnects that may share the same connectors but are marketed for different applications.

Are their differences between dual audio RCA interconnects and, say, those packaged in threes for composite video? Are they interchangeable back and forth, video to audio, or not at all?
 

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