Quote from: Ethan Winer on 2010-03-24 20:07:01

I honestly don't understand why people get so emotional about this stuff. It's just audio!

Because, right or wrong, they do it for a living and they care?

But errors can creep into the math because computational systems are not perfect and additional errors can be intriduced in the conversion process, which is also not perfect.

So while you may be correct in theory reality may differ. This is an ongoing problem when trying to discuss audio with mathemeticians.......

Actually the whole discussion is a series of pissing matches and word games. Fun to watch! Keep it up!

Using this approach, it is definitely not fair to say that a soundblaster "sounds better" than analog tape.

Similarly in electronics, fidelity refers to the correspondence of the output signal to the input signal, rather than sound.

I've snipped the deeply embedded quotes - you'll have to read back to get context. Basically Arny claiming that lossy codecs pass all traditional measurements, me saying they don't, but it's a silly argument because Arny hasn't defined what "traditional measurements" he's talking about, then...

Quote from: Arnold B. Krueger on 2010-03-24 14:27:13

I have. Two words: Audio Rightmark.

Audio Rightmark certainly does include measurements that reveal the noise/distortion introduced by mp3 encoders.

See here for mp3 vs wav:

http://www.jensign.com/RMAA/ZenXtra/Comparison.htm

Scroll down to THD and IMD graphs - quite revealing.

Quote from: Notat on 2010-03-22 20:36:24

Quote from: Arnold B. Krueger on 2010-03-22 19:59:35

For the most part, high fidelity is currently pretty much all about rooms and tranducers. For recording the tranducers are of course microphones, and for playback the transducers are loudspeakers, headphones, and earphones with the latter of course obviating concerns about rooms.

Through measurement, you do find that the majority of the infidelity is where you say, in traducers and acoustics.

So far so good.

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To say that problems in the electronics are insignificant in comparison is like saying a 3 kHz tone is insignificant in comparison to higher level background noise.

I don't get that at all. The harmonics and IM products that are created by transducers are basically the same as those generated by electronics except that the transducers make far more of them and start creating them at far lower levels.

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We are are very good at hearing past acoustics

So far so good.

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and through transducer imperfections.

My experiences say not at all.

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In many cases these effects/imperfections are euphonic.

My experiences say not at all.

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It is not insane for recording engineers and audio enthusiasts to pay attention to details several orders of magnitude below what you would consider to be the primary imperfections.

I don't get that at all.

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One man's imperfection is another man's character.

I'm not buying any of that, either.

What is true is that 40% nonlinearity is an organ pipe is different than 40% nonlinearity in a woofer because an organ pipe makes only one tone at a time, while a woofer makes multiple tones at the same time. Single tone = no IM. Multiple tones = IM.

HOw nonlinearity in an organ pipe is different from nonlinearity in a guitar amp is demontrated when one plays multiple notes at the same time. In the organ, multiple tones means multiple pipes, one tone per pipe. In an electric guitar, it all goes through the same woofer and reducing nonlinearity in tha woofer is of the essence. Playing just one string at a time is not uncommon on a bass guitar, which has the practical effect of reducing IM. When multiple tones are played on a bass guitar their frequencies are often far enough apart that only one of them is actually in the range of greatest nonlinearity which also reduces IM. In your Hifi, you can't count on any of those things happening, so having a reasonbly linear woofer can be very important.

As far as so-called euphony in tubed hi fi amps goes, it turns out that linear distortion due to interactions between the amps high source impedance and speaker impeadance variations is likely the most obvious audible effect.

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Remember that there's art and science in what we do. Those who have an appreciation of both are going to be the most successful.

I agree with the idea that recording is both art and science, but the room for selling distortion as art goes downhill very fast on the reproduction side.

I record and listen all of the time. I'm constantly changing the linear distortion I add on the record side, but I rarely have the need to change it on the playback side.

I'll let others who are more expert than me explain what is "left out" in lossy files. (I'll guess it's frequency response that changes dynamically.) But clearly, a delay of any type will fall under time-based errors.

--Ethan

Quote from: 2Bdecided on 2010-03-23 13:43:52

Quote from: Arnold B. Krueger on 2010-03-23 12:15:58

The first problem here is that the baseline for evaluating digital processing is not zero noise.

but you said there was zero noise. Which is untrue. Then posted five paragraphs to try to dig yourself out.

I said that there was zero added noise.  I can't believe you're holding me responsible for noise that came in the input terminals.

I agree with that in principle. We can call them categories of faults, or categories of errors.

One of the mistakes that has been made by people who misunderstand Ethan's list is to equlate categories of fault with measurements. The basic misapprehension that these critics have is there need only be one measurement to fully charaterize a given kind of fault, which is not exactly true.

In fact you can measure comon instances of all four kinds of of faults with just one measruement (e.g. multitone).  The reverse is also true - it can take more than one measurement to characterize complex faults.

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Even so, I'm not sure where reverb would fall into this.

If reverb is due to a linear process and it usually is,  then it is a form of linear distortion.  Reverb is usually the result of delaying the signal, possibly filtering it with a linear filter, and then linearlly adding it back to itself. The delay is a special case of phase shift.

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It might show up on a frequency response measurement. We could argue about frequency response vs impulse response - but that's a pointless argument.

Reverb does show up in a FR measurment, usualy as some kind of comb filtering effect.

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Far more important IMO is that this list implies an oversimplification that doesn't hold in the real world - just because the effect of any "fault" falls into one of these four categories doesn't mean there are four measurements that can catch any fault.

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Ethan doesn't say this of course - there are two specific measurements listed under the single category of distortion, for example.

This is why I list 2 different kinds of distortion, linear and nonlinear.  I further gave examples of both kinds of distortion.  There are actually two kinds of linear distoriton - amplitude modulation distoriton and frequency modulation distortion. THD and IM measure amplitude modulation nonlinear distortion, while jitter, flutter and wow measure frequency modulation nonlinear distortion.

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My point is this: we generally use measurements tailored to the specific faults we expect to find - "tailored" both in terms of revealing them, and in terms of giving us data in a domain and form that makes some sense, or reveals something useful.

That is more habit and cutom than necessity.  Our ability to analyze signals shot up rapdily when we started doing the analysis with computers.  If you study the more recent literature of audio measruements there have been a number of papers discussing nwere approaches. Papers by Gene Czerwinwski and Richard Cabot come quickly to mind.

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I really wonder if we can define a set of measurements which would catch every possible fault - both now, and in the future.

The answer is generally yes.  Old relics like THD and IM are artifacts of the days when only very simple equipment was available to generate test signals and analyze them.

A great deal can be determined with no specific test signal at all - there is readily available software that analyzes both linear and nonlinear distortion by automatically developing linear and nonlienar models of the system under test. Mathematically, this is called identification. 

Several programs measure linear transfer functions including SMAART.

The essence of Klippel's speaker distortion measurement system is the mathematical process of parameter indentification by comparing observations the operation of the real system and a model with various test signals.  The software just tunes the parametrs of the model until it works like the real system.

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Here's a practical example: put lossyWAV into a stand alone box, including a slight delay which is itself very slightly varying in a random way (i.e. an inaudible amount of flutter). What measurements will characterise that black box properly?

The random delay can be measured by the usual means for measuring FM or phase distortion. I am unfamilir with lossywave.  However I reject this line of argumentation because it is an intellectual game that sheds littls light on the problems we need to solve in the real world. 

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If we can leave the "I must be right / you must be wrong" level of argument at the door, it would be much appreciated.

Well Dr. cure yourself. You played that game a number of times in just this post. You made unfounded assertions.

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This genuinely interests me, and it's more of a challenge than people like to admit - especially when they're arguing with audiofools who want to turn it all into back magic (which it isn't). But let's have a grown up discussion please.

Well then leave the tricks, riddles, and unfounded assertions at the door.

The bottom line is that simple definitions of accuracy may themselves be inaccurate.

Quote from: Arnold B. Krueger on 2010-03-24 11:48:57

The bottom line is that simple definitions of accuracy may themselves be inaccurate.

That is an extremely interesting statement. But where does it leave us?

Quote from: John Eppstein on 2010-03-24 06:23:41

Quote from: Canar on 2010-03-22 17:55:29

Quote from: dwoz on 2010-03-22 17:13:39

Thus, while the idealized console/DAW itself is "perfect", there is NO perfect idealized model of one of the primary, key components, the A/D converter.

There are several. The big question is: which one? http://en.wikipedia.org/wiki/Analog-to-dig...#ADC_structures

Ok, I read over your Wiki layman's reference but I don't see anything there that would lead one to believe that any existing ADC is, in fact, perfect.

A thing does not have to be perfect to form a perfect idealized model of it. Science is noisy and full of error. However, generally the error behaves according to some model. It is not difficult, for example, to measure the spectrum of the noise floor of a given ADC.

My standard comment is along the lines of: A $25 SoundBlaster card has higher fidelity than the finest analog recorder in every way one could possibly assess "fidelity."

And when those on the other side question how "fidelity" is defined, wishing it were how they want it to be, I send them to Wikipedia which explains it very nicely:

True. But such "perfect, idealized models" although they may be handy for ivory tower types and internet discussions, have no actual relevance in the real world where nothing (and only nothing) is perfect.

Quote from: Arnold B. Krueger on 2010-03-23 18:17:57

Quote from: 2Bdecided on 2010-03-23 13:43:52

Quote from: Arnold B. Krueger on 2010-03-23 12:15:58

The first problem here is that the baseline for evaluating digital processing is not zero noise.

but you said there was zero noise. Which is untrue. Then posted five paragraphs to try to dig yourself out.

I said that there was zero added noise.  I can't believe you're holding me responsible for noise that came in the input terminals.

How can you say there is "zero added noise" when the dithering process deliberately adds a specific amount of noise?

Quote from: Arnold B. Krueger on 2010-03-24 11:48:57

The bottom line is that simple definitions of accuracy may themselves be inaccurate.

That is an extremely interesting statement. But where does it leave us?

Quote from: John Eppstein on 2010-03-23 22:21:55

Not all analog mixers are op-amp based.

Agreed. However some very highly respected mixers (e.g. classic Neve) have made very heavy use of op amps.

IME, aversion to op amps traces back to the little dust-up we had in the 70s about an obsolete concept called "Slew rate distortion".  The goal posts have moved since then, and we now thow and kick very different balls.

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Many of the better ones use discrete circuitry.

Again, that's probably far more style than substance. The lowest distortion op amps around are probably ICs.  In fact purveryers of discrete op amp replacements are not always forthcoming about how their products perform vis-a-vis the best chips.

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In fact, one of the primary reasons for the popularity of much "vintage" gear is that it does NOT contain opamps.

Again, there's no logical reason for the obsession with class A amplifiers with regard to signal handling. 

We still use discrete op amps for high power levels. Most if not all modern linear (as oppossed to switchmode) power amps are basically just really big op amps.

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Any gear that operates in Class A does not, by definition, use opamps because there ain't no such thing as a "Class A opamp".

Many op amps are class AB which means that they are  class A when driving high impedance loads. Also, connecting a resistor from the output of an op amp to one of the power supply rails will force the op amp to run class A over a wider range of loads and signals.

I attribute the fascination with class A amplfiiers to a linguistic oddity - class A also means "of the highest caliber" in American English.

As an aside, for a few months this fall I was the unintentional owner of an essentially new (NOS) Pass SA4e which is allegedly a class A power amp. I've listened to it and I've had it on my test bench. I compared it to a Behringer A500 which is in many ways a pretty close comparison. I kept the A500 and sold the SA4e.

Quote from: Ethan Winer on 2010-03-23 17:57:28

Quote from: 2Bdecided on 2010-03-23 16:47:34

I'm not sure where reverb would fall into this. It might show up on a frequency response measurement. We could argue about frequency response vs impulse response - but that's a pointless argument.

I don't consider reverb effects in my four parameters because, at heart, reverb is an "external effect" that happens acoustically in enclosed spaces. Yes, it can be emulated by hardware and software devices, so you can still assess frequency response and distortion.

I was thinking about the effect you sometimes get with valve amplifiers, where the metal in the valves sings along with the music. Replace the speakers with an 8 ohm resistor, crank up the volume, put your ears (not too) close to the valves, and you can hear it. Also, if you use the amplifier normally, and tap the valves, you can hear the tapping through the speaker. These two effects together suggest to me that, under certain circumstances, some valve amps will act as little reverb chambers. I've no idea if it's audible. I suspect it could be caught by a frequency response plot, but might just fall within what most people would judge to be an "acceptable deviation" - while in practice it might not be "acceptable" because the "deviation" occurs so long after the original sound.

Actually there's an electronic reason, at least historically. Class AB and B amplifiers frequently exhibited some distortion around the crossover point in the waveform. This could be anywhere from a distinct "crossover notch" at zero crossing found in nearly all Class B designs to some very, very slight non-linearity around zero crossing in modern class AB amps. The thing is that this particular type of distortion is highly audible (being a form of nonharmonic distortion) and is much worse on low level signal than on high level signals, due to the fact that the amplitude of the distortion products are constant regardless of the amplitude of the signal.

Now in modern, well designed Class AB equipment this really isn't a problem any more, at least for most folks, but some people still believe that there's a slight, but audible difference in the distortion products of the different design classes.

Running various examples of gear on a high quality distortion analyzer yields different distortion spectra.

Whether any of the differences are attributable to this particular design factor? Well, some say yes, some say no, I say use what sounds best to you.

As to your amp, well, based on my experience with the B brand you may regret the decision when it goes up in smoke after the warranty expires.

BTW, did you recap the power supply and do the bias adjustment on the Threshold? Because if you didn't you didn't give the amp a fair evaluation.

Quote from: John Eppstein on 2010-03-24 23:50:37

True. But such "perfect, idealized models" although they may be handy for ivory tower types and internet discussions, have no actual relevance in the real world where nothing (and only nothing) is perfect.

Well, I must point out that one can also model noise and inaccuracy.

It's not like math types haven't seen this. I mean, Walter Heisenburg did a splendid job of it, and the Schroedinger Wave Equation is nothing but a model of the complex function of probability, and one that works frighteningly well to the present.

Now, my friend Greynol...you I have no use for.  You're drunk on moderating, and are a heavy-handed, imaginationless troll accusing others of membership in your clan.  If you were in my forum, you'd lose your first-lieutenant stripes right quick.  You violate your own terms of service merely by showing up in a thread, and on the internet, hypocrites are seldom rewarded.

This is not a troll post and I am not baiting.

Defining what affects audio reproduction has always been the entire point of my four parameters. I go out of my way to explain in forums (again and again and again) that I don't include intentional "euphonic" distortion in the list because that's a creative tool. As is reverb. This is why some people get so upset when I claim that a $25 SoundBlaster card has higher fidelity than the finest analog tape recorder. They immediately see red, and go on about how people prefer the sound of analog tape. And tubes. And hardware having transformers. And all the rest. But subjective preference was never my point or my intent.

--Ethan