Why Do People Come To Computer Audiophile To Display Their Contempt For Audiophiles?

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2 hours ago, mansr said:

No, not with components of reasonable quality in a competent design.

 

I take it you are saying that all competently designed amps of reasonable quality sound identical. That could be your own experience -- only you know what you hear. My example was with what most people would consider a "competent" designer who uses reasonable quality, and so despite differences in measurement between the amps, they might each sound the same to you. Conversely you can't know what everyone else hears or doesn't hear.

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20 minutes ago, mansr said:

That is not what I said. I said that if two amps sound differently, it is not due to imperfections in a resistor.

Thanks for the clarification. Your post wasn't clear that your were limiting the discussion to resistors. I generally agree with that statement with the caveat that noise can be substantially higher with 0201 SMD resistors (as well as thick film) ... my original intention was that the "signature" of any electronic circuit depends not only on the schematic but the non-linearities in its components. All of this forms the frequency/voltage/phase spectrum that would be used to characterize this signature. 

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1 minute ago, mansr said:

And I still disagree with this. Provided good quality components are used, these imperfections will not result in audible differences. Output coupling capacitors might be an exception.

transistors have different technologies with vastly different specifications. they are not at all interchangeable in a schematic. are you suggesting that a quality BJT vs quality JFET transistor necessarily "sound the same"? Surely the schematic matters?

57 minutes ago, mansr said:

That is not what I said. I said that if two amps sound differently, it is not due to imperfections in a resistor.

please explain then, are you suggesting that if two amps sound differently, then it is due to incompetence of design? or use of "poor quality" components?

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2 hours ago, mansr said:

Are you being deliberately obtuse? Of course radically different designs can sound differently, and I never said otherwise. I did say, repeatedly, that if they do, it is because of the different designs, not because of self-noise or other minute imperfections in passive components. A design that amplifies such effects to audible levels is broken.

 

I am trying to give you the benefit of the doubt.  In this thread, the specific "sound" I was discussing was quoted from Nelson Pass own writing regarding the "sound" of the M2. He mentioned the transformer. Yes a passive component. Transformer nonlinearity is well known: http://jensen-transformers.com/wp-content/uploads/2014/09/Audio-Transformers-Chapter.pdf (section 1.3.1)

The nonlinearity of transformers varies from brand to brand and model to model. This is well known. The M2 has a transformer. The schematic is published. Feel free to build it with different brands of transformers to discover for yourself whether you can hear differences. (the parts are not expensive)

 

If you are talking about "something else" then you are quoting me out of context. If you consider either the design or designer incompetent or "broken" then  the phrase "never in doubt, often wrong" applies.

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1 hour ago, esldude said:

However, the M2 uses an input autoformer type transformer for the 14 db voltage gain in that amp.  It feeds MOSFET followers.  Simple design.

There is oft stated desire against negative feedback. The designer states that the measured distortion is good enough that negative feedback is not necessary... I highlight this series because all designs are tradeoffs and here are examples with circuits where design tradeoffs can be listened to.

 

Now the J2 (from the online manual), here the very newly introduced Silicon Carbide power JFET is discussed -- note that he states this measures "astonishingly well":

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With a device that behaves a little like a tube, it is natural to consider popular tube amplifier design topologies. Single-Ended Class A tube amplifiers are not very powerful, and their measurement numbers are not exceptional, but there is no arguing that they have strong musical appeal to much of the audiophile population. The J2 amplifier uses a classic JFET differential input stage followed by a single power JFET transistor. This power JFET is biased by another JFET in what is known in tube circles as a “mu follower”.

Here is a slightly simplified schematic of the J2 circuit. It could be a classic tube amp, except that the P channel input JFETs would have to be fabricated from anti-matter. The single-ended Class A output stage is “second harmonic” in character, and it uses about half the feedback of a comparable MOSFET circuit but with half the distortion and twice the bandwidth.

 

The J2 power amplifier is simple, clean, and measures astonishingly well. It achieves a sound which is warm and relaxed, combining precision and detail without sterility. With a pair of sensibly efficient loudspeakers, it will give you a toe-tapping experience that other solid state amplifiers do not. The design is extremely reliable and will never need adjustment.

 

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Quite a few people have asked me for a regular sort of amplifier, you know the kind you plug like any other, with some voltage gain and a real damping factor.
Amplifiers that have low distortion and noise, and will drive a 4 ohm load.

The last time people asked for that they got the Aleph J, which satisfied most of those requirements. Single-ended Class A, the Aleph J is an easy-going design which is happy driving 8 ohm loads with a warm, relaxed presentation.

By way of contrast, I present the F5 (taa-daa!), a push-pull Class A amplifier, utilizing JFETs and MOSFETs in a very simple two stage complementary circuit – a little bit like a complementary version of the Aleph J. But like all the other First Watt amps so far – this one is different.

...

In many ways, it’s an ordinary topology - the basic circuit is found in numerous preamp circuits and the odd power amplifier (Check out the Profet amp from Selectronics). But the F5 is the product of numerous decisions that set it apart.

It has very wide bandwidth, DC to > 1 MHz.

No capacitors anywhere in the circuit. (except in the power supply, of course!)

It has a high input impedance – 100 Kohms, and a high damping factor (~40)

The distortion is very low, between .001% and .005% at 1 watt.

It will drive low impedances.

It’s very quiet, about 60 microvolts or so.

Did I mention that it sounds terrific?

I preset this in contrast, for given design decisions (let's say transparency?)

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Why am I going on and on about this?

 

Precisely for the topic as stated. I give an example of FirstWatt as exemplary of a designer that "knows his stuff", is unquestionably an expert in his field and is respectful of the goals of his customers i.e. audiophiles

 

I present this to highlight the dichotomy between "blind" measurement and listening. I am hardly anti-measurement, but I think at the end of the day knowledge of the schematic is critical to understanding the measurements.

 

For "subjectivists" here are a series of different designs that can be listened to. For "objectivists", here are different schematics that can be analyzed. I have suggested that it is easier to just listen  That's what I do when going over schematics  

 

We should try to look at things from the "other guy's" perspective (I didn't bring up the schematics initially because that's not where we started but am happy to discuss...). Rather than directing what people should like, we can observe what people do like and try to explain that. There doesn't need to be value judgment.

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1 hour ago, Ralf11 said:

Peter, if you are not doing a blinded listening test, then knowledge of any spec. is clearly a source of confirmation bias, just like a groovey lookin' front panel.  There's no way around that.

 

It does not mean one should ignore specs.

I find that if I listen to:

while building a circuit, the circuit sounds much  better

 

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1 hour ago, esldude said:

Is noise modulation audible?  At some level yes.  In fact if the modulation is considerable it will show up in the dynamic range test.

 

Consider measuring not only the noise floor but close-in noise. Most frequently folks quote the noise floor at some ridiculously low level (e.g. -150 db) but that's the floor. The 1/f component will often greatly increase the close-in noise (theoretically approaching infinite as frequency approaches zero) but look at the corner frequency... see article I just posted from Analog Designs .. very readable at least until the math starts  

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1 hour ago, plissken said:

You can apply any post processing you want. You not add lost fidelity. Regardless if the loss is in choice of mastering equipment or the choices of the mastering or mixing engineer.

 

That depends entirely on what you mean by "lost fidelity" exactly. In many cases a deconvolution does exactly that, restores fidelity. If by "lost fidelity" you mean "lost information" then no, but that's an important reason to "oversample" which allows introduction of error at a level less than the LSB.

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12 hours ago, plissken said:

 

You can apply any post processing you want. You not add lost fidelity. Regardless if the loss is in choice of mastering equipment or the choices of the mastering or mixing engineer.

 

This is stemming from someone asking what hardware was 'transparent'. I answered that question.

 

11 hours ago, jabbr said:

 

That depends entirely on what you mean by "lost fidelity" exactly. In many cases a deconvolution does exactly that, restores fidelity. If by "lost fidelity" you mean "lost information" then no, but that's an important reason to "oversample" which allows introduction of error at a level less than the LSB.

 

1 hour ago, plissken said:

 

Oversample a 192kbps mp3 all you like. 

MP3 lossy compression is obviously lossy. We aren't presumably dealing with MP3 in the context of this forum although the principles are important for audio in general.

 

Using a real (for us) situation like CD Redbook, 16/44.1 if the mastering engineer applies an equalization profile its possible it could run into the headroom available in 16 bits, and information might be lost.  Instead if the file is supplied as 24 bits there might be enough headroom that the equalization is reversible by applying an "invert" equalization (just the opposite of the original eq) and then no information will be lost. This would be termed a "reversible" operation. If the mastering equalization is a very simple convolution, the reverse equalization would be a very simple deconvolution, and in this case reversible. So back to my original question: do you consider "loss of fidelity" only irreversible convolutions or do you consider reversible convolutions (where the deconvolution restores full fidelity)?

 

The functional analysis convolution-deconvolution had been developed precisely to "restore" signals that have been modified.

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6 minutes ago, plissken said:

 

I understand inverse filters. It's still not restoring information that isn't there. 

Deconvolution is a generalization of "inverse filter" and in the same way that a bandwidth limited analog signal can be entirely represented in 16/44.1 and then reconstructed, a deconvolution can restore a signal that has been altered. Back to the question: do you know the information isn't there? That is the crux of the question. If you are sure the information isn't there then it can't be re-created. Like refocussing an image to remove "blur" seemingly lost information can be restored. It really depends.

 

OK you added that clarification -- yes "mastering grade" equipment would tend to be transparent, and I assume works at high bitrate, depth for the reasons I've outlined. 

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4 minutes ago, plissken said:

 

Don't get off on tangent that I didn't bring into the conversation.

 

AGAIN: 

 

Question by other: What equipment constitutes transparent playback

 

Answer by me: Probably most mastering grade audio equipment

Agreed ... I think we all are guilty of responding to short quips out of context, as well as writing that is too brief to convey the context or intent...

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59 minutes ago, pkane2001 said:

For deconvolution to recover the 'original' signal, you need a very precise model of the corrupting signal. In most real-world cases, this is very hard to derive.

The room correction that @Jud just mentioned is an example of a deconvolution that is measured rather than estimated.

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18 minutes ago, pkane2001 said:

 

I'm not aware of any DSP that uses deconvolution to apply EQ. Convolution -- yes. Deconvolution is a complex mathematical process that doesn't have a simple solution. Convolution is much, much easier in comparison.

 

Convolution and deconvolution are a pair and map to division in the Fourier domain.  Consider the system as the convolution and the deconvolution is a Fourier domain divisor that results in 1. In room correction, the deconvolution is simply 1/impulse response. It is really as simple as dividing the Fourier domain signal by the measured impulse response.

 

To clarify deconvolution is simply 1/convolution in the Fourier domain. By measuring an impulse response, the deconvolution is thus measured.

 

Of course software like HQPlayer applies this transform in the time domain in realtime in SDM so @Miska could provide the math  

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2 hours ago, pkane2001 said:

 

As I said, not so simple in the real world, in the presence of noise and other distortions that are also convolved with the signal.

Life is implementation  dependent

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5 hours ago, AJ Soundfield said:

And extremely potent imagination.

 

When @fas42 arrived on our starship a few weeks ago, followed by you, it reminds me of the Star Trek episode: http://www.imdb.com/title/tt0708435/ ... you are the "Commissioner" ... are you "winning?"