I agree with you - transferfunction is at the root of managing fidelity - and the most important aspect of management or alignment has been found with the phase elements...

Speakers are the real phasy noise makers if you will...
All the other components are of much less concern... but they still count as there timing qualities are average with other componets within the interconect network...

 

Quote:
Originally Posted by dogbaker
__________________________________________________ _

The rate of electron flow isn't constant nor is equal to that of photons in a vacuum (AKA Light)... This isn’t my opinion it is scientific fact...

I could cite the math – I also have the gear to make such measurements with!

Wow, where to begin. Though you may have access to a network analyzer and are able to measure the differences in cables, it is just plain irresponsible to use this as "evidence" in the audio realm.

Kevin, I have much more than a basic network tester – I have a cutting edge audio suite that is worth a nicely equipped Mercedes... I will provide you with a list of hardware and software if you require...

Transmission line parameters are largely useless in the audio world as the frequency is not high enough to make any difference. While things like skin effect and LCR losses are measurable in the audible band, unless the cables are ridiculously long or light no one could hear the loss.

Kevin, they have been proven to be ojectively measureable, the data that you and everyone else to date on this forum have cited is from pre-1995... Much has been refined since then with the advent of TSD, FFT, MLS, and the alike...

To say that the speaker cable's contribution to overall phase shift is "profound" is laughable.

Be careful you know what they say – he who laughs first ...

What type of phase shifts are you referring to – I am not talking about a fixed FRQ impulse – I am talking about impedance based delays that are discrete to frequency, I plot them every day, I will send you a plot – just send me an email address. At the same time I will have the plot include the timing shift per frequency... For any device , wire – speakers – amps – preamps... just ask and provide email address...

While the dielectric constant of speaker wire may not support transmission at light speed it is plenty fast enough to make path differences of over a hundred feet negligible in terms of phase.

Again, I think that you’re talking about a fix FRQ impulse measurement – but even at this level, everyone know agrees that lengths greater than 4’ feet start to create audible issues – just look at proliferation of receivers with impulse analysis and correct built in, but these no not offset the over time tilt as frequencies rise... Again, I have never said that they are always audible, but they’re measurable and that they contribute to over timing qualities of the entire system. More importantly, speaker wires can and do have a more profound effect over Qtc of the speakers to which they are interconnected... I can cite the math... and Qtc has a direct relationship to frequency response – this is well document by may – T/S to name one! Would you like to see the math? Just Ask....

The resistive losses may be significant but to say that the phase shift is perceptible is a joke.

Again watch the joking – laughing thing! And again, I didn’t say that they were always audible, please re-read me earlier text.

Furthermore phase, group delay and other colourations "in the pico range" are swamped by the total system THD+N.

Okay – on this one it is completely clear that you have very little experience with digital measurement equipment... Using a digital analyser that utilizes FFT technology, we have easily broken down a sample and determined what is signal and what is noise down to the Pico level – analog systems cannot – I have both as do most audio system designers do today... In fact, I can also determine the nature of the noise, using auto algorithms from Tektronix... they automatically compare them against a data base of known noise forms... This is very cool stuff and the stuff that I have been using for 10-years...

So please see this as an opportunity to learn a little more – I measure DA’s on a regular basis using this technology and EIA defined methodologies...

We are now narrowing in on new and amazing truths about human hearing and as a result are developing new audio technologies/products to cater to them... We have been mining Pico signals for years now...

Your posts contain so much inflated evidence, exaggerated statements, and obvious elitism that I don't know why I even bothered to respond.

It isn’t elitism, it’s enlightenment and nothing more... and deep down you probaly suspect that i might be correct, but you don't like my delivery - and I don't blame you!

Clearly I should be thankful you are here to school us.

You’re welcome!

BTW: The garden hose analogy was not mine.

I was able to figure that out... so no shame for you!

 

Nor was I. I was simply pointing out that your analogy was valid and could still be used if one wanted to get into a more indepth discussion. Your contribution was appreciated on my end.

 

I was talking about a proper high frequency Scalar Network Analyzer perhaps capable of meaningful measurements to 40 GHz. They cost as much as a small house and so would be restricted to labs of corporations. I see now that your equipement is not this sophisticated and I'm not trying to be condescending. You are lucky to have a well equiped shop but I question how you obtain measurements in the pico range with affordable equipment.
This is true but the question is whether it is "profound" or whether one can even hear it. I think not and dislike the tone you have adopted since you joined the discussions. There are people on this forum who understand what you are talking about even if they do not work in the Hi-Fi industry. Some are active but many are lurkers who don't participate in the discussions.
I'm still laughing. Thanks for that I guess.
No need to provide plots I know what you are referring to. While I was not aware entirely what you were talking about at the time I still say you are overstating. I don't mean to stonewall you with subjective opinions. I truly believe that pico range time or amplitude differences are inaudible.
Everyone? There is no receiver I'm aware of (please correct me) that is capable of correcting for a few feet of wire. The time resolution is just not available, at least not in anything stretching the definition of reasonably affordable. Definitely not over frequency. While cables do contribute to many system parameters like Qtc simply using a large stranded cable with reasonable lengths will more than suffice. Exotic cables may improve measurement scores but in my opinion they aren't worth the money. You've repeated the offer to provide the math, I'm familiar with it, I don't dispute it, I'm just not interested because I don't think it's even interesting.
Yes you later recanted a bit. But you continue to exagerate and talk to other members in a manner that is elitist and condescending (no not enlightening). I'm afraid your first impression may be unrepairable.
Untrue, while I no longer work with it daily I have used high speed digital equipment that dwarfs the cost of your entire lab/workshop. I now use FFT and MLS for taking measurements at home. I dont use it in a professional capacity anymore. While you obviously have access to some nice equipment I doubt there are affordable digital systems capable of time resolution to the pico range. Not reliably anyway. I am interested in what you use to achieve it though (see I keep an open mind).
I will. You come across as someone who believes he is much more intelligent and knowledgable than any member of this forum. Why is that?
Yet you persist. I don't doubt that you are correct in the measurements, I know many of these things can be measured I just don't share your opinion that they are audible. I do dislike your delivery and suspect it will continue.
Yup.
Nor was it for the individual who posted it. It may have been overly simple but it is more than suitable for an internet forum where no one asked for an indepth discussion of transmission lines. When posting here there is no need to be condescending and tell people to learn or remain stupid. In my experience it's best to explain things the technical way using the simplest technical terms possible. The people who wish to learn the details will ask for clarification. Others may wish to remain ignorant and continue to dis-information but this can't be fixed no matter how we challenge them. Around here we only talk about what amp is ok to use with a particular subwoofer. Soon we will wear you down. You will get stupider and stupider and we will get along good.

 

See now here is a classic example of demeanor and the reaction that it receives. I don't know either dogbaker or Kevin but it appears their level of kung-fu is similar.

Kevin has been on the board some time and when he offers up a technical tid-bit it is in a way that is neither demeaning or elitist. He makes himself a part of the community rather than expecting the community to praise his presence as some sort of audio savior.

Mr. Dog - you could learn this from him.

Oh - and I don't care what math you produce (I'd even understand most of it I bet) you will not hear the difference in 3 feet of wire compared to 97 let alone 8 to 17.

 

For all your Knowledge Sir (dogbaker), you fail to realize that your delivery makes you ineffectual. You claim you want to educate, but your technique for doing so leaves your "students" wishing it was another class. Perhaps, you should take a few classes yourself or read a few books on how to become a good teacher. It would be a shame to waste all that knowledge on ears that are tuned elsewhere simply because they detest the medium.

 

Okay - Okay

I'm not that much of a hard ***, I just wanted to get everyone thinking and put them on notice that if they want some facts, then ask then listen - I am not here to be taught (but I am sure that I will learn a few things), but to share what I know with others...

So rather than us continuing in this fashion, let's settle down and clear up a few remaining miscommunications...

So come-on, 40gig tester for Audio - we are after samples between 20Hz to 100KHz... We are not interested in microwaves... Agreed...?

Secondly please stop misquoting or loosely quoting me - I haven't said that Pico latencies are audible in all devices - what I did say is that they and other small delays sum and ultimately do become audible... Now, where I have assumed we went down the wrong path, was when I stated delays in CD players as being audible and cited delays in the Pico range as contributing to these differences. Not in cables and amps etc..., but can be measured!

So let me make this a little more clear - in purely digital domains, which strive to stave off a multiple in layers of inherent timing errors, DDC's, DAC's, ADC's etc included... In these environments/circuits, signal delays directly contribute to harnesses observed subjectively, in many digital audio source units... Source units with delays in the nano region or lower sound smoother and more like analog devices, there are but a few that have measure delays as minuet as Pico seconds, and as stated before they are all over $6000.00... I didn't say that Pico delays in a cable are audible... milli are...

So it is now very clear that you have very little knowledge of the vast amounts of test equipment and related technologies of which they are based, available today for measuring audio qualities - not microwaves... It is also just as clear that you have just as poor of an understanding of the availability of DSP within consumer products, here is a list of a few companies (car, home, pro & commercial) that offer units with advance DSP, which will facilitate either automatic or manual adjustment of electrical delays and physical asymmetric placement of speakers within a listening space...

Alpine, Sony, Eclipse, Rockford, Orion, Clarion, Pioneer, Audio Control, Yamaha, Denon, Onkyo, Kenwood, NAD, Rotel, Mission, Naim, Anthem, Moon, Linn, Rane, Behringer, Roland, Danley, Lexicon, ARC, Zapco, Audison, Crown, QSC, IED, Biamp, Crest, Ashley, Atlas Sound, Netstreams, Speaker Craft, Sunfire, etc, etc, etc

There are at least a hundred manufacturers using this technology within several of their varying models...

So if you are unaware of common consumer offers, there is no chance that you would have a complete understanding of esoteric lab grade offerings - you might be aware of a few, but that's it!

You sound like you might be more familiar with low amplitude, high frequency gear - am I correct. You keep mentioning gear that isn't intended to measure audio signal goodness (20Hz-20KHz), so I am getting the feeling that you’re in college, gearing up for a career in communications verticals - am I close?

So let’s tone this down - your knowledgeable - I'm knowledgeable, we sometime misunderstand what the other is attempting to say and other times we simply agree to disagree...

Fair enough - ?

Do you feel that you have worn me down enough, so that we can know move on, or do we need to go head-to-head on every point to know ones real benefit but each other’s ego's etc...???
You decide...

As for the rest of your comments, they have become stale...

 

the best part is when he tells kevin to stop laughing hahahahahahahaha

stop being a clown, people will stop laughing

 

Hi Dukk,

I feel that this overview relates to your question, so I have pasted it over from another thread...

Most people believe that at the outputs of an amp, preamp, speaker surface etc that the frequencies are being emitted all at the sample time – in other words, that an electro-mechanical or electronic device outputs: each and every frequency, at precisely the same time, which they don’t. They do so at deferent times, for each frequency, primarily because of impedance fluctuations caused by increases in frequency, within in purely electrical circuits (but not solely as we will learn – the digital domain also incurs delays). This in turn leads to a diminishment in stereo image reproductions, frequency and transient response - at the bare minimum!

So what devices are most troublesome?

No. 1: Speakers have the most errors up to 600ms at low frequencies - that’s over a half of a second
No. 2: Then preamps & amps are next in line with up to 10ms, at the bottom and top of the bandwidth
No. 3: Then Source units, when measured at their digital SPDIF stage or output – CD Players – iPods etc come in at up to 500 micro seconds in many entry level types (single bit), (up to 10Ms at analog outputs) and up to 500 nano seconds in most quality devices (multi-bit), (up to 3ms at analog outputs), some esoteric products are ultra fast, coming in with mere pico latencies (6K for one of these babies), (up to 3ms at analog outputs)...
No. 4: Then Cabling: Which is dependent on length and cable quality – in a car micro-nano latencies are common.

As a system, these sum and in fact, can create additional timing errors due to poor electrical compatibilities etc...

Digging in!

So let’s look at a source unit...I wish to keep this at a low level , so i will talk in broad terms and attempt to paint a picture of sorts...

I will attempt to reveal to you that there are in fact several time errors, which affect the quality and usability of the frequencies, outputted by electronic and electro-mechanical devices and in this case – so called digital devices...

This is a small part of the picture, but as time goes on – all aspects will be brought in.

Let's use a typical CD - Head-unit

It has a transport in which the CD is load into, from that point on, a laser assemble is guided across the CD emitting a measured wavelength of light. Diffractions from this light are collected as they reflect off the surface of the CD and are converted from light onto electrical pulse codes.

So let's pause here and review:

The disk is spinning, but not at a perfectly constant rate, light is pulsing at close to a perfect rate, but is traversing the gap at a constant speed. In that we have some mechanical elements integrated into our data acquisition, some very small but important time errors occur and they need to be compensated for, so a common approach is to incorporate 'buffers' which store/pause the data stream then release it to other stages (a little more on this in a bit).

These compensations and other processes take place after the photon pulses are converted to electron pulses. This conversation process introduces more time errors (as well as other distortions).

So at this juncture, we have converted photons pulses into electron pulses and we have a host of errors / distortions, but what’s primarily at the root of most of them (in the digital domain), is timing errors and of course low sample rate issues...

So let’s get back to our main flow...

Once conversion has taken place, buffering, oversampling and other digital process take place within and at various stages, but all in the digital domain, and all working towards one goal, the reproduction of a linear mathematical equivalent of an analog signal.

Now, this is where DAC’s topologies play a big part, as they each have pro’s and con’s, in terms of trades offs. Some have fewer stages which typically reduce distortions (Time and others), some have higher sample resolutions or faster thread poles, some require the implementation of analog filters post digital processing and the list goes on. As a rule though, Multi-bit topologies currently offer the best blend of compromise and overall value, in terms of dollars (yes – marketing has a say, unfortunately).

Regardless of DA topology, they all produce a myriad of distortions, with most of them rooted in sample resolution and processing time errors (but not all). The reason for this is that digital data – binary, is managed by transistors based processors, which can be viewed as electro -mechanical devices that are capable of turning on an off from a few times a second, to billions of times a second and onward (3.2 GHs CPU etc)... So in other words, it takes time to process or create digital data and that time leads to delays and in modern DAC’s they are processing many things at once and combining the results in a buffer, which in turns takes more time to organize the data, into packets (and makes errors while doing so, leading to more distortions). Once these packets are complete, they are ready to be converted into analog signals or to be transmitted as binary via a SPDIF or converted back to photon pulses and transmitted via a TOSlink output (which can add more errors).

So as you can see, there is a lot of room for error - All of the signal goodness is based primarily on sample rate – then processor rate and software/algorithm quality – and lastly circuit and board design.

I realize that I have left sample rate and circuit board element out of this overview until this point, but I am quite sure that I will be ask about them specifically, as time progress and I will address them at that time.

This is a general overview of the key contributing elements that have the most significant effect on phase response at the digital output stages, as such; the other elements are less important and have been left out off of focus within this discussion.

So let’s pause again and review:

DAC’s manage binary signals via transistors based CPU’s, which are in essences micro relays. These devices open and close controlling the flow of electrons etc. All this creates a bottle neck and takes time to process (hence the name processor). DAC’s process many layers of data at the same time and combine the results in buffers, which sort, compile and create the data packets for streaming to the final stage – digital to analog conversion or other. Most processors are running in the MHz range, which holds them down to at best, low-micro– high-nano scale frequency generation, some esoteric products use faster processors and more advances software algorithms, resulting in faster and more accurate frequency production.

Regardless of speed in the digital domain, they all must enter into a final stage of conversion from digital to analog and it is in this stage where in terms of phase they all incur shifts to a very similar degree, which is to say that most implement very similar analog circuits. Depending of what type of DA topology that the incorporated, they may also be required to additional analog filters to diminish some harsh effects of poor digital processing, which in turn can and does lead to significant phase shifts, centered around the filter frequency, increasing delays into the high micro – low milli second range, in some very entry level products.

In closing,

My point was and is that it has offend been argued that all CD Players/DAC’s should and do sound the same – but, that argument began in 1980 and has most been mostly abandon, as we have all experienced different tonal qualities (sonic signatures) from different DAC’s, as evidence by the fact that, year after year, as a myriad of audio companies release new and improved variants, we continue to trade-up. If you have the opportunity, listen to some older DAC’s within your existing system for a few hours - then put yours back in – I am highly confident that you will hear a difference – who cares if it’s an improvement or not – that would be subjective – but you should perceive a difference and I am sure you will, unless the audio system is very poor, as a whole.

If you’re unable or unwilling to attempt the aforementioned then at least ask yourself this: why if most of us have not perceived such, why have we all been gear junkies for the last 5-10-15-20-25-30 plus years... the answer, simple – if technology hasn’t in fact improved year after year, as objectively verifiable advancements in science and mathematics have claimed, then it can be for no other reason than, wait for it – we must be able to hear the difference...

So now ask yourself, where did that difference come from?

Now this is for a few readers specifically: the answer is NOT FROM OUTER SPACE, but from the group effort of Scientists and Engineers!

So what was once argued has passed because the mass major of people have in fact, been able to hear the improvements within each new variant of DAC’s (and other devices) and also that we have been able to start objectively measuring and therefore map more of the contributing: electrical, electro-mechanical and digital factors, which govern the transfer function of the data that is requisite to high quality audio reproduction.

What has been discovered is that a correlation exists that allows for the classification of DAC performance, based on their frequency phase responses. DAC’s which have mere pico latencies in the delay of frequency production (pre-conversion into analog signal) tend to produce sonic qualities closer to analog (post conversion). Furthermore, it has been discovered that delays, which occur and are greater in the mid range, but within the same scale, (pico, nano etc...) produce a sound that is laid back to most listeners. Most DAC’s produce latencies, (pre-conversion) in the micro ranges and have a poor grouping in latencies scores. In the land of electrons, micros seconds are miles and pico seconds are feet (i am not being literal)! Regardless, of the aforementioned digital qualities, the design and build quality of the analog stages are very important, and heavily influence the final outcome of the analog signal, which will next be presented to the preamp – and the story continues!

While we still don’t have the complete picture and may make a few mistakes, while we are still forced to permit some assumption(s), it has become clear that pico scale latencies are in fact, important to ultra high fidelity audio reproduction – not low!

As to the question, can we actual hear such delays – if we couldn’t, they wouldn’t be - becoming, an important consideration in ultra high fidelity audio design.

But let’s end with this – while this was in fact a low level overview, to some, it has been an esoteric one, which is to stay that it was meant for and understood by only a few...

In practical audio design, milli-seconds are all that matter and nothing more, and your head unit, cabling and even your amplification are not your biggest enemy(s), in this regards, it’s your loud speakers! So focus 99% of your energy on getting them to produce timely sound – for SQ or SPL systems, it’s all about phase!

 

I'm a little new here and I may not know as much as some people on here. I was just wondering what people thought of ribbon or planer technology. Sony did have a car component set that had amazing SQ and the tweets were ribbon technology. ( XS- HF167 )

So what are the advantages of a ribbon over a regular dome?

- High sensitivity and superb dynamics. Less than a few watts will typically yield very high spl levels. Also, above a certain frequency usually 10khz or so, a ribbon acts as a line source meaning that for every doubling of distance, you lose only -3db rather than -6db like you would with conventional point source drivers. The longer the ribbon, the lower the frequency.

- Greatly improved horizontal off-axis response resulting in excellent high frequency detail retrieval, a more open and "airy" sound, and a wider/deeper soundstage.

- Very fast decay times, which give you a smooth, uncolored, and grain free sound. The worst example of this here is your typical metal dome tweeter, which can be extraordinarily detailed but sound harsh and "metallic".

- A limited vertical dispersion. Simply put, the sound present above and below the tweeter is greatly reduced in comparison to the sound radiating directly in front, and to the sides of the tweeter. This is an advantage because it reduces in-car reflections from the floor and the underside of the dash when mounted in kickpanels.

- In the case of Aurum Cantus tweeters, replacing/reconing the ribbon is extremely cheap and easy, usually costing you only a few minutes of time and about $5-$10 USD.

Disadvantages:

- Limited vertical dispersion. While this can be an advantage, it can also be a disadvantage. The tweeter must be aimed directly at the listening position, otherwise the spl will drop off. Can be a pain to work with if there are 2 people of different height sitting in the car.

- Fragile ribbon element. Yes the ribbons are very fragile. They require steep crossovers usually 18db (3rd order) or above to protect them from overexcursion. It is also absolutely recommended that you place a capacitor inline with the tweeter to prevent them from being damaged by turn on/off noises etc.

- Size. These suckers are pretty big and deep.


P.S. Awesome article!