[Daryl]

This is all fairly rude, isn't it? 

I guess you must feel you are on pretty solid ground, Daryl.

I agree.
Maybe there is a REASON JohnR has so many posts.

Yes Russell and Lin even I thought so,

I will admit that I have some sharp edged words for certain commercial know-it-all's who mislead others while fattening their wallets but mostly I just like to talk about the details.

Notice your comments toward me earlier this thread (Russell) that I did not get too excited about.

Thats generally how I see myself as pretty level headed except for the odd conspiracy theory here and there.

The real question is was it appropriate.

Take a look again at Johns post...

Wrong again. Bzzzt.

We all know Dan is someone very knowledgeable in this area.

Yet John swats him like fly without a single word from him enlightening the rest of us as to where or how Dan is wrong or what the true secret to the universe is that only John seems to know.

Now I am also someone who is well versed in this stuff (according to me anyway) and I can see that Dan is correct and has contirbuted immensely to thread desparately in need of his contribution (it was beginning to look more like a pagan ritual).

As I often do when someone jumps onto the seen and starts stirring the pot I clicked on Johns name and viewed his last posts to see if he has said anything intelligent recently.

Lin said "maybe there is a REASON John R has so many posts" but I came up dry (I didn't read all 3600 though) and could not find the reason John is able so slap Dan down so easily.

Thats all.

[opnly bafld]

We all know Dan is someone very knowledgeable in this area.

As I often do when someone jumps on the seen and starts stirring the pot I clicked on Johns name and viewed his last posts to see if he has said anything intellegent recently.

Lin said "maybe there is a REASON John R has so many posts" but I came up dry (I didn't read all 3600 though) and could not find the reason John is able so slap Dan down so easily.

Just because someone is knowledgeable doesn't mean they have all the answers and if someone else is not able to explain their stance scientifically that they are automatically wrong. In case you missed it science has been wrong more than right during mankind's history. (these are general statements not aimed specifically at anything in this thread)

The reason JohnR has a lot of posts is because he has been here since day 1, not because he is a self proclaimed expert on everything out to save all the incompetent masses.

Lin

[Daryl]

Were all wrong sooner or later.

My point is do you dismiss Dan like a fool without a word?

The reason JohnR has a lot of posts is because he has been here since day 1, not because he is a self proclaimed expert on everything out to save all the incompetent masses.

Yes I just recieved a PM informing me that John founded AC.

Gilda Radner would have said "Oh.....Nevermind"

Even I radical as I am might have charged in with one less gun blazing and one less flag flying.

I joined AC when the MAD board was down but I like the layout here better so way to go John.

Peace

[rajacat]

Maybe audio systems should be custom designed according to the clients particular room characteristics and the unique hearing of the customer. The client should undergo a complete hearing exam by a licensed audiologist to determine their hearing ability and then the system should be developed to fit like a glove. Under these conditions maximum synergy would be easier to attain.

Raj

[opnly bafld]

Were all wrong sooner or later.

My point is do you dismiss Dan like a fool without a word?

Agree with the first point.

To answer your ?,   YES, after so much of this    you finally do this

Lin

[JohnR]

Daryl: slew rate is a measure of how fast a device can change its output signal. If the signal it tries to produce has at any time a first derivative greater than the slew rate, then distortion is the result. Forget about sine waves - think about more complex signals. It seems to me a reasonable hypothesis that one component A may cause a following component B to distort this way, whereas a second component A' may not. And as bpape points out, simply dismissing component B as badly designed may be somewhat missing the point.

I thought that the slew rate comment by Karsten was an interesting one, and suggested that one could measure the combination of components rather than the individual pieces as a way to look at "synergy." There are other interactions that could show up this way, such as between driver Q and the high output impedance of SETs. I was surprised that Kevin missed the hint there. What about amplifier behavior on a transient overload condition -- how does the combination of the amplifier circuit with the speaker affect this?

Wouldn't those be interesting topics to explore?

As for Dan Banquer, I feel obliged to respond, since you've made a big deal out of it. In my opinion, Dan's veneer of so-called knowledge is paper-thin, and he covers for it with his arrogant and condescending manner and attitude. I and others have become very tired of it over the years. I am mostly able to ignore him these days; not as well as I perhaps should, obviously. Don't mistake him as a role model for how to behave on this forum -- to glibly assert one's superiority in technical matters, yet provide little guidance or be willing to engage in a shared learning experience. In other words, you should consider applying what you have said to me to your own posts instead.

JohnR

[shep]

It's Sunday and sunny out, the mood lifts. Is this synergy? From memory and paraphrasing shamelessly; Synergy is a fortuitous coming together of disperate elements that form a whole that is greater than the sum of their parts. It's conceptual, appreciable but probably not measurable in any meaningfull, repeatable way. Is it important to quantify our experience in this way? Obviously for some it is. Audio excellence is a wil-o-wisp; now you hear it now you don't. You can head-butt over this till you're in a coma but you wouldn't change it. For the record (and I may be wrong) this term was first used in audio officially in TAS before some of you were into solid food. Harry Pearson, bless his soul was probably the one, like so much else. Yes he was a primadonna, arrogant and impossibly stubborn but he gave us a whole legacy of discriptions and ways of talking about our passion/hobby, without which we would be much the poorer.
(We were of the same generation and exchanged occasional pleasantries, hence my digression)
Some of our experiences can only be explained in the vaguest and most "illogical" of ways. Unless there are some major break-throughs, this is going to hold true for a very long time to come. As with many other postings that cause people to ride around on their high horses until they get exausted or fall off (and thus far this is a mild example), I cannot for the life of me understand why the very mention of the subjective/unquantifiable aspects of our hobby get some people so upset and ready to go to war! Do I care if some would think me a fool for believing that I hear my opamps running in, my cables burning in, etc.? Hardly. There are enough kindred spirits in attendance so I can enjoy my folly in jolly good company!
with which Shep saunters out of the arena to enjoy a spring afternoon.

[Dan Banquer]

Daryl: slew rate is a measure of how fast a device can change its output signal. If the signal it tries to produce has at any time a first derivative greater than the slew rate, then distortion is the result. Forget about sine waves - think about more complex signals. It seems to me a reasonable hypothesis that one component A may cause a following component B to distort this way, whereas a second component A' may not. And as bpape points out, simply dismissing component B as badly designed may be somewhat missing the point.

I thought that the slew rate comment by Karsten was an interesting one, and suggested that one could measure the combination of components rather than the individual pieces as a way to look at "synergy." There are other interactions that could show up this way, such as between driver Q and the high output impedance of SETs. I was surprised that Kevin missed the hint there. What about amplifier behavior on a transient overload condition -- how does the combination of the amplifier circuit with the speaker affect this?

Wouldn't those be interesting topics to explore?

As for Dan Banquer, I feel obliged to respond, since you've made a big deal out of it. In my opinion, Dan's veneer of so-called knowledge is paper-thin, and he covers for it with his arrogant and condescending manner and attitude. I and others have become very tired of it over the years. I am mostly able to ignore him these days; not as well as I perhaps should, obviously. Don't mistake him as a role model for how to behave on this forum -- to glibly assert one's superiority in technical matters, yet provide little guidance or be willing to engage in a shared learning experience. In other words, you should consider applying what you have said to me to your own posts instead.

JohnR

This has all been explored, written about, over and over again. If you are refering to the TIM arguement, I will not even bother to respond further.
Below are two samples of "unwillingness" to share in my learning experience that I have posted in Lab forum: I'll let you judge for yourselves about my glibness. There are also a number of articles including DIY stuff over at Audioholics.

1. Typical stereo separation at the listening position is usually somewhere around 8db to 10db or so I have read. This alone would appear to nullify further investigation of electronics and stereo separation, but over the years I have made some interesting observations that I would like to share with all of you.
In designing and building a few line level units I have observed that the coupled signal when doing stereo separation testing has been 90 degrees out of phase from the input signal. It also appears that it is capacitively coupled and decreases as the frequency drops, and typically 6 db per octave. In reducing this coupling I have also observed better noise immunity. For Example: If I use an “unshielded” wire to go from the back panel input to the board I will degrade stereo separation and at the same time, due to the high input impedance of the input I am connecting to I will have more noise due to pick up. If I use a piece of coax for the same connection I get an increase in stereo separation and lower noise pick up. It would appear that a higher level of stereo separation could well be more indicative of better noise immunity in this application. My own personal observation is that reducing the 90 degree out of phase coupling gives a better “image” along with reduced noise. At present I generally shoot for close to 70 db of separation at 20 kHz, I find audibility is very difficult above this. The above makes me ask the question; Is it stereo separation, or noise immunity, or both?
Recently, I was designing a line stage - headphone amp. In the design process I decided to add resistive coupling across the headphone outputs. This increased the “image” across the front of my head giving the “ illusion” of a wider soundstage. I repeated a similar experiment with my loudspeakers and found that resistive coupling levels (no phase difference between the coupled signal and the main signal) can be much higher than capacitively coupled signals before I was able to detect this. This makes sense if you think about it, because all that is being added is small amounts of mono to the “mix”. I should also note that all of the listening tests that I did on my own used CD only.
If anyone else has something substantive to add; feel free.

2.
GROUNDING PRACTICES IN CONSUMER AUDIO

As many of us have observed over the years grounding practices in consumer audio have for many of us been a frustrating experience. Compatibility/Synergy are consistently major issues, as well as safety, reducing ground loops, noise and interference. Designers are equally frustrated by this problem. Some of us wish it would go away and others attempt to deal with as best as we know how. There does not appear to be much of any standardization that I, and many others are aware of. Many of us who work in the electronics industry outside of audio who have observed the posts from audiophiles, designers and dealers scratch our heads in near disbelief.
The following is not only an outline for reducing these problems, but a wake up call to this end of the industry. The following is for the application to consumer unbalanced audio, and could well be a possible outline for positive discussion and direction. It is also a way to use Earth Ground to our advantage instead of a problem.
Chassis Grounding:
The following is a technique used in instrumentation for low frequency applications.
The chassis will be earth grounded via the earth ground at the three-prong outlet. The audio signal ground contained in the chassis is not connected to the chassis ground. This will require the design to be electrically isolated from the chassis ground which is easily solved by using nylon stand offs to mount pc boards and isolated bulkhead RCA jacks. (I am going to break this rule later on but bear with me for the moment). The above forces the design to be star grounded at the return of the power supply, which is always good practice to reduce noise and ground loops. However, this does leave the present configuration susceptible to interference from the inputs. This interference can be reduced by the using a simple common mode ac line filter at the AC input and using either a well shielded coax or microphone cable at the line level input. This configuration also poses an additional problem due the fact that we now have two different grounds with two different potentials. In the past I have observed this problem when using a certain brand of rotary switch for a volume control. The rotary switch was not well isolated internally and had enough of a leakage current so that noise was developed when it was used. Moving to a different vendor with higher isolation devices corrected the problem. I have not observed any problem with standard switches for on off applications or anything similar.
I have applied this technique to basic audio chain of equipment consisting of an outboard DAC, line level pre amp, and power amps using the chassis and grounding design I outlined above. The transport that I presently use is a modified consumer device and is equipped with a two-prong plug. The system also has an FM tuner and an old pre amp that is used as a phono pre amp. Both of these devices are standard consumer issue with a two-prong plug. I have observed no compatibility issues with the older style units.
As I outlined earlier I am going to break this rule at one point. The line level pre amp now has a connection from the return of the line stage pre amp power supply to the chassis of the unit. I have now connected earth ground to the analog “center point of the system.” This did not cause a ground loop at all, and to be more precise, for CD playback the inherent ground loops that are typical for unbalanced circuitry simply disappeared. The FM tuner and the old pre amp appear to be unaffected by the center point earth ground. A welcome addition was that the rotary switch that had a leakage problem because of the two different potential levels described earlier no longer had the problem due to center point earth ground.
The use of the system center point earth ground for low frequency applications has been in the textbooks for at least 30 some odd years and has been applied to other low frequency applications. Applied at this level to a simple chain of audio playback equipment. CD playback now has reduced hum and hiss to levels more akin to balanced design than unbalanced design. Playback of FM tuner and Phono pre amp remains unaffected.
One thing that has surprised me relates to the issue of low frequency applications. I was expecting to find problems with the digital portion of this playback chain. I have not found one to date but I think this needs to be investigated further when time allows.
A note to all of the tweakers who read this: I am not recommending any changes to existing designs; in fact I would discourage it.
To DIY folks: You may wish to rethink some of your present chassis/grounding schemes.
To the rest of the industry; this is a subject that not only deserves discussion but an active participation to reach acceptable standards.
The grounding system described above will not address the problem of toroidal transformers mechanically vibrating due to either DC on the AC lines or as I have observed on occasion, low frequency oscillations.
Dan Banquer
R.E. Designs
P.S. I would greatly appreciate it if the following discussion is limited to the issue of grounding only.

[JohnR]

If you are refering to the TIM arguement, I will not even bother to respond further.

See?! There it is again 

[Dan Banquer]

If you are refering to the TIM arguement, I will not even bother to respond further.

See?! There it is again 

http://www.audioholics.com/education/amplifier-technology/the-high-instantaneous-current-spec
This has been up for over a year and yes it does pertain to TIM. BTW:
Duh!
       d.b.
         

[JohnR]

Well... I think that article pretty much proves my point.

[opaqueice]

Daryl: slew rate is a measure of how fast a device can change its output signal. If the signal it tries to produce has at any time a first derivative greater than the slew rate, then distortion is the result. Forget about sine waves - think about more complex signals. It seems to me a reasonable hypothesis that one component A may cause a following component B to distort this way, whereas a second component A' may not. And as bpape points out, simply dismissing component B as badly designed may be somewhat missing the point.

What do you mean by "Forget about sine waves"?  Every signal, no matter how complex, can be decomposed into sine waves, and then the maximum time derivative of that signal is simply related to the bandwidth and amplitude, exactly as was said earlier.  Using that we can get the max time derivative for any given recording format, so we just have to design the amp to be able to handle that.   If we do an inadequate job and the time derivative exceeds the slew rate of the amp there will be distortion, but I don't see what this has to do with synergy, except in the very limited sense that it depends on the load the speakers present. 

[Daryl]

Not shure where you came from Opaqueice but that is an excellent first post.

John R and Bpape as I said already a couple of posts ago Frequency and Time domain are two views of the SAME THING (what shows up on one also shows on the other).

Any componet who's magnitude/phase vs frequency transfer function is smooth cannot have any signifigant time domain anomalies.

I know what slew rate is (you probably already knew I did).

More importantly Dan already pointed out that any decent modern componet has a slew rate far beyond any music signal it will ever see.

[bpape]

Playing and measuring a component with steady state or sweeps will show only frequency related anomolies.  Looking at pure single frequency sine waves is largely the same thing.  This says nothing about how it reacts in the time domain in terms of reacting to an input signal.  What is the delay between input and the device reacting and starting to 'ramp up'?  How long does it take for it to get from a quiescent state to full output?  How quickly does it stop?  None of those things will show up in any frequency related testing.  Nor, will any frequency related anomolies show up when testing those things. 

Please explain how a component that generates ruler flat frequency response guarantees exceptional time domain performance?  There are tons of amplifiers that have basically perfectly flat FR in the audible range.  Does this mean that they all perform the same in the time domain?  Have the same slew rate?

None of this really has to do with synergy other than trying to quantify a measurement for each individual piece that when matched, would potentially work well together.

Bryan

[Daryl]

If you are refering to the TIM arguement, I will not even bother to respond further.

See?! There it is again 

I don't know what this is about as I have not seen this thread.

What I can tell you and it may or may not be what that other thread covered is that TIM (Transient Intermodulation Distortion) is an unnecessary term.

Simple IM (Intermodulation Distortion) covers the same thing.

A transient is something that happens fast and therefore will have a wide bandwidth and more to this point a lot of high frequency energy.

Also a transient is often high energy as it is usually caused by something violent (percussion).

So what you have is simply high frequency information at high level.

A simple sine wave at full power and high frequency will cause the feedback loop of an amplifier to become unstable just the same as an impulse which renders the TIM spec. unnecessary.

[Daryl]

Playing and measuring a component with steady state or sweeps will show only frequency related anomolies.  Looking at pure single frequency sine waves is largely the same thing.  This says nothing about how it reacts in the time domain in terms of reacting to an input signal.  What is the delay between input and the device reacting and starting to 'ramp up'?  How long does it take for it to get from a quiescent state to full output?  How quickly does it stop?  None of those things will show up in any frequency related testing.  Nor, will any frequency related anomolies show up when testing those things. 

Please explain how a component that generates ruler flat frequency response guarantees exceptional time domain performance?  There are tons of amplifiers that have basically perfectly flat FR in the audible range.  Does this mean that they all perform the same in the time domain?  Have the same slew rate?

None of this really has to do with synergy other than trying to quantify a measurement for each individual piece that when matched, would potentially work well together.

Bryan

Thankyou Bryan,

Now we are talking.

That is precisely the law that Fourrier revealed long ago.

That any time domain signal or transfer function has an equivilant frequency domain signal/transfer function (one will always directly indicate the other).

Doesn't necessarily apply to audio alone it applies to everything in the universe it is just math and NOTHING can defy it.

It is a major revelation but once you understand it you are on your way to seeing the world in a different way (so to speak).

Take a magnitude/phase vs. frequency measurement for any componet and apply a Fourrier transform and you have it's impulse response which can also be converted to any other type of response you might want like step or time/energy.

Thus any aspect of a componets linear transfer function is indicated by it's frequency response including rise time, delay ringing and so forth.

Also any signal can be represented as a magnitude/phase vs. frequency chart.

You can take your favorite song like, oh I don't know 'Oops I did it again' by Britney Spears and convert it to a frequency response chart that will completely represent the entire song including the exact timing of every drum beat and synth note.

Further if you had millions of tone generators you could set each one to the frequencies of the chart and the magnitude and phase and when you turned them all on they would play 'Oops I did it again' over and over forever (long has this been a dream of mine).

[opaqueice]

There are tons of amplifiers that have basically perfectly flat FR in the audible range.  Does this mean that they all perform the same in the time domain?  Have the same slew rate?

Yes, at least if by "perfectly flat FR" you also mean perfectly linear phase response - otherwise there will be (mostly inaudible) time-domain distortions.

Further if you had millions of tone generators you could set each one to the frequencies of the chart and the magnitude and phase and when you turned them all on they would play 'Oops I did it again' over and over forever (long has this been a dream of mine).

Dear god....

[bpape]

opaqueice,

Exactly!  What had been stated earlier was that FR and time domain are simply different views of the same thing.  To me, they're complementary and to be viewed together as you stated.  FR by itself cannot determine time domain issues.

Bryan

[Daryl]

Please explain how a component that generates ruler flat frequency response guarantees exceptional time domain performance?  There are tons of amplifiers that have basically perfectly flat FR in the audible range.  Does this mean that they all perform the same in the time domain?

Exactly!  What had been stated earlier was that FR and time domain are simply different views of the same thing.  To me, they're complementary and to be viewed together as you stated.  FR by itself cannot determine time domain issues.

In a nutshell, any signal can be represented by (costructed from) a series sinewave building blocks (componets) each will have their own frequency, phase and magnitude.

When you add them up they take the form of the signal they are meant to represent.

If you apply the amplifiers frequency response curve to each of your desired signals sine wave componets (adjust the magnitude and phase by that amount indicated by the amplifiers frequency response) and then sum them you will have your signal as reproduced by your amplifier.

Thereby frequency response does determine time domain issues.

Frequency response is a componets Linear Transfer Function, Impulse response is also that very same Linear Transfer Function not a different part of it but the same thing.

[JohnR]

There are tons of amplifiers that have basically perfectly flat FR in the audible range.  Does this mean that they all perform the same in the time domain?  Have the same slew rate?

Yes, at least if by "perfectly flat FR" you also mean perfectly linear phase response - otherwise there will be (mostly inaudible) time-domain distortions.

No, because that assumes that the system is linear, which (since we are talking about distortion) it isn't.

Before Daryl gets all excited about this statement don't forget that there are actually two things being discussed here: the time/frequency domain representations of a signal and the time/frequency domain response of a system.