What's wrong with Redbook, really?

Soundproof

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Which is all very well and good, but it's still a lot more than 0.1 dB. And I am led to believe that many (most?) high-end speaker manufacturers do match speaker pairs to about that same tolerance, but that 0.1 dB is close to impossible with physically large mechanical transducers like loudspeakers ( at our current level of engineering). And we're only discussing on-axis frequency response, just one of many measurable qualities contributing to the sound of a given loudspeaker.

I was responding to your "1dB" in quotes, and agree with you that 0.1dB would be almost miraculous - and probably also pointless, as placing the speakers just a little bit out of proportion to the listener would result in a greater differential than 0.1dB, and most speakers and listening positions are out of proportion.

"We" can distinguish down to 0.15dB, with confidence, though some training may be required depending upon what the signal consists of.
 
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Phelonious Ponk

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Tim, even you must have realised that more than a couple of the others here are reading from the same songbook as myself: they just prefer to keep their heads down for safety's sake most of the time, unlike me. It's nothing more than persistence that's got me to where I am today, and I just feel it would be a shame not to plant a few more seeds, as they say. It's good to see Myles pushing hard from another angle, and Tom Danley has thrown his bits in, pointing out that what actually occurs physically in the ear organ bears almost no relationship to the sound image that is contructed by the brain. You are quite "obsessed" with the body being a terribly mechanical device that can only deal with the sensory input in a very simplistic way, the truth is far more than that ...

Frank

There are many tweakers in audio. And there are those who believe in the power of the mind to transform what tickles the ear; count me among the latter. There have even been many who have built a powerful cause/effect relationship between the two: I tweak, therefore I hear.

But there is only one you, Frank, and you're ours. :)

Tim
 

rbbert

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A normal passive has a tolerance of +/- 3 dB.
Now play a sine sweep.
Do you hear differences in loudness?
Probably not.

+/- 3 dB difference in response between two individual "identical" speakers? I think you're talking about something else.
 

rbbert

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I was responding to your "1dB" in quotes, and agree with you that 0.1dB would be almost miraculous - and probably also pointless, as placing the speakers just a little bit out of proportion to the listener would result in a greater differential than 0.1dB, and most speakers and listening positions are out of proportion.

"We" can distinguish down to 0.15dB, with confidence, though some training may be required depending upon what the signal consists of.

I also think it's almost impossible to compare two pairs of speakers in this way for the reasons you mention, but my response was to the original assertion that two speakers that "measure the same" would also sound the same, and pointing out that getting 2 different speaker models (not 2 different examples of the same speaker) to "measure the same" is probably impossible.
 

microstrip

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I also think it's almost impossible to compare two pairs of speakers in this way for the reasons you mention, but my response was to the original assertion that two speakers that "measure the same" would also sound the same, and pointing out that getting 2 different speaker models (not 2 different examples of the same speaker) to "measure the same" is probably impossible.

I would like to make a suggestion, but still do not understand what you call "measure the same" .Unless your question is purely rhetorical, you have to clear specify what measurements and in what conditions.
Suppose that using acoustic null methods you could fine tune speakers to measure the same residual of less than .1 dB. at 1/3 octave pink noise. Would it satisfy your requirement?
 

rbbert

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Not me, the other Rob. My position is similar to yours in this case.
 

Robh3606

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Aug 24, 2010
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I would like to make a suggestion, but still do not understand what you call "measure the same"


If you are talking to this Rob this is what I mean. That's a pair of compression drivers tested on a horn without any network. As you can see they are very closely matched. They are well within a dB over the entire range they would be used in the intended system. You do not want to use 1/3 octave smoothing you would want 1/6-1/12. The 1/3 smooths things out too much. Yes it looks great but it's hard to see past the smoothing to whats actually going on.

Robh:)
 

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Lee

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Feb 3, 2011
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Good post Myles.

I will only offer up my experience in doing audiophile classical recordings. On numerous occasions we have split the mic feed and recorded a live concert at 16/44.1 with one feed (mostly for musicians) and 24/176 (for our downloads). There is much better soundstage and truer instrument timbre on the hirez recording. And we have repeated this dozens of time with a wide variety of groups from guitars to strings to solo piano.

Hirez is the real deal and a solid advancement in sound quality.

First, no we can't agree. It is far from a fact that we currently have all the measurements necessary to describe how we perceive sound and music or measure equipment. If you don't think so, look at the evolving work of the one Keith O. Johnson, who is far from happy with the current set of measurments used to evaluate audio gear.

Two, have you or haven't you read the book This is Our Brain on Music by Daniel Levitan? In fact, I feel this book is a MUST READ for any audiophile and engineer! Levitan, who is a musician, producer and neurophysiologist, has done new studies that disprove many accepted facts about how our brain perceives music, interprets it and processes it.

Third, you are really not understanding the concept of complex biological/systems or unpredictable synergies. These synergies are not in the equipment but in the sound and its perception. Nothing happens as they say in a vacuum. Take drugs for instance. There is no drug that specifically inhibits DNA synthesis. The drug always has a multitude of effects on other systems including RNA and protein synthesis and other cellular pathways too.

This is from the book Neurophysiological Basis of Movement by the Dr. Mark Latash, an outstanding neuroscientist from Russia who is now a Professor at PSU. From Chapter 1 of his book:

1. Function of a neural structure can be derived from properties of its elements (neural cells or neurons) and their connections. So, when researchers accumulate enough information about the structure of the central nervous system (the brain and the spinal cord), its function will become obvious. This approach is commonly called reductionism because it attempts to "reduce" the function of a complex system [my note: and certainly our hearing mechanism and audio system fit that bill] to the properties of its elements. Sometimes another fancy expression is used with respect to this approach: ascending determinism.
2. Function of a complex system cannot be understood on the basis of its structure and the properties of its elements. Understanding a complex system requires a special set of notions that cannot be derived from simply looking at the elements and their connections. So no matter how much information a scientist obtains about the elements of a complex system, he or she will not understand its function without forgetting about the elements, at least temporarily, and looking at the system as a whole. This approach is called the "complex system approach" and my heart belongs to it.

If one accepts the idea that complex systems should be studied in a different way than by accumulating information about their elements, first it is necessary to realize the general properties of a complex system and to introduce an appropriate, meaningful language (set of notions). Note that systems of seemingly different complexity may be described with the same sets of notions. For example, our planetary system consists of zillions of atoms. However, the Bohr planetary model for just one atom is qualitatively rather similar to the solar planetary system. Behavior of a motionless heavy rock may be much more simple and predictable than the behavior of an electron on one of the atomic orbits within the rock. This means that a complex system does not necessarily imply a complicated description of its behavior (emphasis author). Most of the objects surrounding us can be described with fewer parameters and simpler laws than an atomic nucleus. This simplicity of its description is in fact the most important advantage of the complex system approach (author emphasis).


So when I'm talking about unpredictable synergies, I'm referring to the reductionist approach where everything is broken down to its basic elements eg say like in the human genome project where the goal was mapping every gene in our DNA and then ID'ng their function would explain everything we needed to know about all disease. But again, this approach ignores that all these genes can interact to produce other unpredictable results.
 

fas42

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Where's Frank, still sleeping? :b
Aaargh, Bob, you caught me out!!

My one comment will be to repeat something that's obviously p!ssing off a lot of people here, that resampling, upsampling, offline can make all the difference in the world. I would like to see an experiment done where a recording feed is split between 44.1 and, go for it, 384 "native" encoders. Then take the 44.1 version, resample with the best algorithm up to 384, and then run the very best, "objectivist", DBT between the natural and the "fake" 384 version. My suspicion is that it would rattle a few cages ...

Frank
 

Lee

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Feb 3, 2011
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Aaargh, Bob, you caught me out!!

My one comment will be to repeat something that's obviously p!ssing off a lot of people here, that resampling, upsampling, offline can make all the difference in the world. I would like to see an experiment done where a recording feed is split between 44.1 and, go for it, 384 "native" encoders. Then take the 44.1 version, resample with the best algorithm up to 384, and then your run very best, "objectivist", DBT between the natural and the "fake" 384 version. My suspicion is that it would rattle a few cages ...

Frank

So Frank what is wrong with the test I am doing? The only thing we are changing is the sampling rate and word length. Same mics, cables, battery power, etc.

Also you seem to think that an algorithm is capable of filling in the difference between the real life audio event captured in 24/176 and that of redbook. That seems impossible as an algorithm's best guess will never be better than having the real data straight from the A/D.
 

NorthStar

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Aaargh, Bob, you caught me out!!

My one comment will be to repeat something that's obviously p!ssing off a lot of people here, that resampling, upsampling, offline can make all the difference in the world. I would like to see an experiment done where a recording feed is split between 44.1 and, go for it, 384 "native" encoders. Then take the 44.1 version, resample with the best algorithm up to 384, and then run the very best, "objectivist", DBT between the natural and the "fake" 384 version. My suspicion is that it would rattle a few cages ...

Frank

I'm not pissed for one single bit whatsoever Frank. :b
 

Lee

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Redbook is fine.... as long as you maintain signal integrity throughout the process. It is way harder to create a good musical CD than it is for a hi-rez file. So many things you need to take in consideration. Everything is cumulative with no margin for error (pun intended).
This is why it takes so long to produce an album on the FIM label. FIM is regarded as one of the industry's leaders in producing the best audiophile CD's. Every step along the way gets scrutinized and double checked. Unfortunately of late, Winston hasn't been satisfied with the stampers that have come out of production and has a warehouse full of rejects. He'd rather lose money than put out something sub-par.
I feel this is why most CD's produced today, sound like crap. No one is taking time through the process to make sure signal integrity is maintained. It is too easy now for someone to put their shingle out and say "I'm an engineer".

Good post Bruce. There are some very natural sounding CDs out there like these recent Nat King Cole DCCs from Steve Hoffman. However, as good as those are, or even the CCRs that Steve did, I know the hirez version would be better. In fact Steve created the CCR SACDs and they sound better than the CDs as good as those are. I also like Winston Ma's work greatly.
 

fas42

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So Frank what is wrong with the test I am doing? The only thing we are changing is the sampling rate and word length. Same mics, cables, battery power, etc.

Also you seem to think that an algorithm is capable of filling in the difference between the real life audio event captured in 24/176 and that of redbook. That seems impossible as an algorithm's best guess will never be better than having the real data straight from the A/D.
Absolutely nothing is wrong with what you are doing! What I am saying is, yes, the algorithm is capable of "filling in the gaps", in part because that is exactly what is happening anyway in the operation of the A/Ds. Yes, I know the end result sounds different, at times very different, but more and more I'm starting to realise, believe, that these differences are solely due to how the DAC behaves, how well it functions, while being fed audio data at certain rates. Now, the only way to truly verify this is to run experiments along the lines I've mentioned -- I would be just as interested as everyone else in the results, and be perfectly content to be proven wrong. But, first such tests must actually be done ...

Frank
 

Lee

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Feb 3, 2011
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Absolutely nothing is wrong with what you are doing! What I am saying is, yes, the algorithm is capable of "filling in the gaps", in part because that is exactly what is happening anyway in the operation of the A/Ds. Yes, I know the end result sounds different, at times very different, but more and more I'm starting to realise, believe, that these differences are solely due to how the DAC behaves, how well it functions, while being fed audio data at certain rates. Now, the only way to truly verify this is to run experiments along the lines I've mentioned -- I would be just as interested as everyone else in the results, and be perfectly content to be proven wrong. But, first such tests must actually be done ...

Frank

I don't think so because the hirez benefits seem to accrue over many types of converters, mic feeds, concerts, etc. My view is an algorithm can guess at what happens at t=x but knowing exactly what "x" is by capturing the value (height of waveform in PCM, change in DSD) is best.
 

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