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#361
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I found the holy grail that explains audiophile beliefs
Paul Stamler wrote:
"bill" wrote in message t... As I just mentioned in another post on this what happens if you sample your 20KHz sine wave with a 40KHz sample rate that just happens to sample the zero crossings and not the peaks??? I think you know. Which is why you don't do that. Period. Congratulations; you've just discovered aliasing. Peace, Paul Sorry again, I am not talking anti-aliasing. Get out some paper and drink some coffee, since it appears this concept will take you some time. -- Bill (Sleepless biker) Baka |
#362
Posted to rec.audio.pro
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I found the holy grail that explains audiophile beliefs
Laurence Payne wrote:
On Wed, 25 Oct 2006 17:59:52 GMT, bill wrote: As I just mentioned in another post on this what happens if you sample your 20KHz sine wave with a 40KHz sample rate that just happens to sample the zero crossings and not the peaks??? I think you know. Yeah. We "know". But, in this case, I'm told (by people who know their stuff) that we're wrong. NYquist isn't very intuitive. But it works. It works because nobody can really hear the distortion caused at that high a frequency, and a pure sine wave used for testing is not what a musical instrument may produce at that frequency. A lower frequency example is a drum, which is a decaying oscillation, so just think about building up a good representation of that with only 2.2 samples per sine. Just when you think you have it the amplitude has changed. Why is something so simple going over some people's heads? -- Bill (Sleepless biker) Baka |
#363
Posted to rec.audio.pro
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I found the holy grail that explains audiophile beliefs
APR wrote:
It is amusing how many posters here have to attack the person all the time. It is surely a sign of something amiss when there has to be personal attacks rather the discussion of the relevant points. I think we all know very well qualified people who don't demonstrate the level of common sense and intelligence the qualifications would suggest, and it appears to me that that hypothesis is reinforced by the posters regularly resorting to personal attack. Of topic here, but I have had guys with PhD's working for me that were useless because they could not think for themselves and certainly could not think 'outside the box' they were taught not to question. 8 years of college crams you full of 'facts' but not to question those 'facts'. You might as well program a computer with all those 'facts' if there will not be any creative thought to verify them. -- Bill (Sleepless biker) Baka |
#364
Posted to rec.audio.pro
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I found the holy grail that explains audiophile beliefs
David Morgan (MAMS) wrote:
"APR" wrote in message It is amusing how many posters here have to attack the person all the time. No one who is *truthfully* regular attacks anyone unless it's political in nature. The rest of it is people just passing through who think they know it all anyway. Or, in my case, have experienced some of the shortcomings of relying totally on a theory that applies to one thing but not to what I am working on. -- Bill (Sleepless biker) Baka |
#365
Posted to rec.audio.pro
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I found the holy grail that explains audiophile beliefs
William Sommerwerck wrote:
If "the scientific method" were the definition of science, then the best scientists would be those who where the best at running scientific experiments. But despite the need to carefully desingn, run, and interpret experiments, the best scientists are those with the best insight, the best imaginations. The first hypothesis has to come from -- and it doesn't come from experiment. It comes from imagination. AMEN!!! -- Bill (Sleepless biker) Baka |
#366
Posted to rec.audio.pro
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I found the holy grail that explains audiophile beliefs
Chevdo wrote:
In article , says... If "the scientific method" were the definition of science, then the best scientists would be those who where the best at running scientific experiments. But despite the need to carefully desingn, run, and interpret experiments, the best scientists are those with the best insight, the best imaginations. The first hypothesis has to come from -- and it doesn't come from experiment. It comes from imagination. Doesn't matter whether those people are practicing science or not, you're talking about superior people, not superior 'scientists'. Science without the scientific method is 'woo-woo', aka 'flummery'. It's how we got crap like chiropracty and homeopathy that we for some reason now can't get rid of. The fact that a genius needs to be in charge of the lab janitors to generate exception results, and that even the lab janitors have to be cognitively superior to, say, the fader janitors who call themselves 'audio engineers', and that even those fader janitors have to be cognitively superior to the original mop & broom janitors, is superfulous. However, if geniuses simply relied on their geniusness, rather than the scientific method, we'd be absolutely NOWHERE, as we were for thousands of years until we finally realized that anything not proven by scientific method is WORTHLESS. It was the Catholic church who stifled invention and burned people at the stake for heresy in questioning 'God'. That put us back an easy 1,000 years. There were plenty of geniuses who were around before the scientific method, and they unfortunately didn't get very far. Church again. Read about what happened to some of the inventors back around Galileo's time. If Aristotle had been around in the 16th century, he'd probably rival or exceed Newton, and if either of them had been in the 20th century they'd probably rival or exceed Einstein. Aristotle and Pythagoras were true geniuses before their time but the Roman empire (conquer, rape, and burn) set things back a bit, not to mention all the warring for no good reason. Ever wonder what was in the library of Alexandria that has been lost forever? Einstein had the benefit of the access to the entirety of Newton, Aristotle, and many other geniuses at his disposal, without which even he wouldn't have got very far. The point is, research was totally unreliable before the scientific method. Now it is far more reliable (though not totally, as evidenced by the amount of flummery that still gets published in peer-reviewed science journals), and the various political machinations to shape and direct research by pharmacutical companies, and over-zealous and/or ideologically biased academics. Let's see. There's primitives in office, like Bush declaring stem cell research is immoral and against God's wishes. Ring a bell? -- Bill (Sleepless biker) Baka |
#367
Posted to rec.audio.pro
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I found the holy grail that explains audiophile beliefs
On Wed, 25 Oct 2006 18:43:54 -0700, "William Sommerwerck"
wrote: If "the scientific method" were the definition of science, then the best scientists would be those who where the best at running scientific experiments. But despite the need to carefully desingn, run, and interpret experiments, the best scientists are those with the best insight, the best imaginations. The first hypothesis has to come from -- and it doesn't come from experiment. It comes from imagination. Science is defined by the scientific method. If you want to re-define it, we can't stop you. But you will only cause confusion. In the scientific method, the hypothesis comes from observation. I guess we can stretch that to include unconscious observation and correlation. |
#368
Posted to rec.audio.pro
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I found the holy grail that explains audiophile beliefs
The first hypothesis has to come from somewhere -- and
it doesn't come from experiment. It comes from imagination. I don't quite know how to evaluate your use of the term "best." Could you quantify that a bit so we're all on the same page? To me, this is all very obvious. But... Who are the greatest scientists? The ones who perform the best experiments? Or the ones who do the most-creative thinking? |
#369
Posted to rec.audio.pro
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I found the holy grail that explains audiophile beliefs
Why can't you see that the imagination and intuition you speak of are
integral to the method which defines science both at the level of theory and at the level of experiment? Because they aren't. The experiment without a hypothesis is blind... What you don't seem to understand is that such experiments can be useful to learn about something you don't understand. It's precisely because people refuse to perform such experiments that we don't have reliable, accurate listening tests. About 20 years ago, the late, lamented Hewlett-Packard ran a series of ads that went something like... "An engineer asks 'How can I?'. A scientist asks 'What if...?' We never stop asking 'What if...?'. More than anything, science is about asking good questions. |
#370
Posted to rec.audio.pro
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I found the holy grail that explains audiophile beliefs
The experiment without a hypothesis is blind
That is the WRONG term. a 'blind' experiment is a BETTER experiment. A 'double-blind' experiment is even better. 'blind' experiments are those in which the experimenters are not able to influence the outcome by restricting their awareness of the constituents of the experiment. It does NOT mean that a hypothesis is not being tested. You've quite misunderstood what he meant. |
#371
Posted to rec.audio.pro
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I found the holy grail that explains audiophile beliefs
William Sommerwerck wrote: The first hypothesis has to come from somewhere -- and it doesn't come from experiment. It comes from imagination. I don't quite know how to evaluate your use of the term "best." Could you quantify that a bit so we're all on the same page? To me, this is all very obvious. But... Who are the greatest scientists? The ones who perform the best experiments? Or the ones who do the most-creative thinking? I don't quite know how to evaluate your use of the term "best." Could you quantify that a bit so we're all on the same page? Later... Ron Capik -- Will the circle be unbroken by and by... |
#372
Posted to rec.audio.pro
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I found the holy grail that explains audiophile beliefs
bill "just use comcast", "you know the drill for no spam bots." wrote:
It works because nobody can really hear the distortion caused at that high a frequency, and a pure sine wave used for testing is not what a musical instrument may produce at that frequency. A lower frequency example is a drum, which is a decaying oscillation, so just think about building up a good representation of that with only 2.2 samples per sine. Just when you think you have it the amplitude has changed. Why is something so simple going over some people's heads? Because it doesn't exist. It ONLY takes 2 samples per wave to reconstruct the original signal. THAT is what Nyquist says. It does not matter what the phase is, it does not matter what the frequency is. 2 samples per cycle will do it. 2.2 is more than enough. If you don't believe it, try it with a typical modern A/D and D/A box, a scope and a signal generator. What goes in is what comes out, right up close to the filter cutoff. --scott -- "C'est un Nagra. C'est suisse, et tres, tres precis." |
#373
Posted to rec.audio.pro
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I found the holy grail that explains audiophile beliefs
It ONLY takes 2 samples per wave to reconstruct
the original signal. THAT is what Nyquist says. Strictly speaking, it's 2 samples, over multiple cycles, because the sampling rate has to be than twice the highest frequency to be sampled. This makes sense if you understand the mathematical transformation between the time and frequency domains. |
#374
Posted to rec.audio.pro
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I found the holy grail that explains audiophile beliefs
Scott Dorsey wrote:
bill "just use comcast", "you know the drill for no spam bots." wrote: It works because nobody can really hear the distortion caused at that high a frequency, and a pure sine wave used for testing is not what a musical instrument may produce at that frequency. A lower frequency example is a drum, which is a decaying oscillation, so just think about building up a good representation of that with only 2.2 samples per sine. Just when you think you have it the amplitude has changed. Why is something so simple going over some people's heads? Because it doesn't exist. It ONLY takes 2 samples per wave to reconstruct the original signal. THAT is what Nyquist says. It does not matter what the phase is, it does not matter what the frequency is. 2 samples per cycle will do it. 2.2 is more than enough. If you don't believe it, try it with a typical modern A/D and D/A box, a scope and a signal generator. What goes in is what comes out, right up close to the filter cutoff. --scott I don't give a flying crap what Nyquist says because it is aimed at steady state sine waves at that kind of a sample rate. If you can't understand that, then I am arguing a point with a rock. Didn't anybody ever teach you to think for yourself? -- Bill (Sleepless biker) Baka |
#375
Posted to rec.audio.pro
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I found the holy grail that explains audiophile beliefs
But despite the need to carefully desingn, run, and interpret
experiments, the best scientists are those with the best insight, the best imaginations. The first hypothesis has to come from -- and it doesn't come from experiment. It comes from imagination. This directly contradicts what you wrote a couple days ago, when you said that good scientists begin their research without a hypothesis. I don't follow you. |
#376
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I found the holy grail that explains audiophile beliefs
Chevdo wrote:
that anything not proven by scientific method is WORTHLESS. Chevy, any high school science student knows that the scientific method doesn't "prove" anything other than what tested variables don't support a given hypothesis. It is a process of elimination, which you and Gutter Butt ought to both appreciate. Know your terms, THEN go out and preach. Love the "Brave New Worldly" intellectual dominance hierarchy you set up in your sermon, though. Wonder where you fit in... Never mind. Chewy |
#377
Posted to rec.audio.pro
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I found the holy grail that explains audiophile beliefs
David Morgan (MAMS) wrote:
"APR" wrote in message It is amusing how many posters here have to attack the person all the time. No one who is *truthfully* regular attacks anyone unless it's political in nature. The rest of it is people just passing through who think they know it all anyway. Totally not true, Dave. There are several regulars on here who attack people personally. Once a difference of opinion, taste or technique develops, the norm in this group is attack. This is usually followed by questioning the person's credentials, experience and then moving into statements about their ears or brains not working properly and ending in them either being called a Nazi or worse (!) a Republican :-) And heaven forbid if you don't use a tape machine :-) |
#378
Posted to rec.audio.pro
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I found the holy grail that explains audiophile beliefs
bill "just use comcast", "you know the drill for no spam bots." wrote:
I don't give a flying crap what Nyquist says because it is aimed at steady state sine waves at that kind of a sample rate. Every signal can be represented as the sum of steady state sine waves. Fourier showed this in the 17th century. The Shannon Sampling Theorem therefore is valid for any arbitrary waveform once it's proven for a single sine wave, AS LONG AS that waveform is bandlimited. Discussion of this can be found in Shannon's original BSTJ article. If you can't understand that, then I am arguing a point with a rock. Didn't anybody ever teach you to think for yourself? I understand it, but it's not relevant. If you think it IS relevant, you haven't done the math. I strongly recommend a good introduction to DSP which will explain the Sampling Theorem. --scott -- "C'est un Nagra. C'est suisse, et tres, tres precis." |
#379
Posted to rec.audio.pro
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I found the holy grail that explains audiophile beliefs
Scott Dorsey wrote:
bill "just use comcast", "you know the drill for no spam bots." wrote: I don't give a flying crap what Nyquist says because it is aimed at steady state sine waves at that kind of a sample rate. Every signal can be represented as the sum of steady state sine waves. Fourier showed this in the 17th century. The Shannon Sampling Theorem therefore is valid for any arbitrary waveform once it's proven for a single sine wave, AS LONG AS that waveform is bandlimited. Discussion of this can be found in Shannon's original BSTJ article. Here's a pointer to Shannon's paper: http://cm.bell-labs.com/cm/ms/what/s...day/paper.html Bob Lucky gave a great talk at the Shannon Day conference addressing the impact of latency on communication theory as applied to modern media. But that's way, way OT. Also, there was a long sampling theory discussion in rec.audio.tech a few months ago, [ Subject: 10 metres audio cable going into PC = too long? ] that may be enlightening. Later... Ron Capik -- |
#380
Posted to rec.audio.pro
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I found the holy grail that explains audiophile beliefs
Scott Dorsey wrote:
bill "just use comcast", "you know the drill for no spam bots." wrote: I don't give a flying crap what Nyquist says because it is aimed at steady state sine waves at that kind of a sample rate. Every signal can be represented as the sum of steady state sine waves. Every STEADY state signal. Fourier showed this in the 17th century. I don't think he was concerned with audio. His math works nicely with sine waves that have enough repetition to build up an approximation. He had no clue that there would be computers doing things he never even dreamed of. The Shannon Sampling Theorem therefore is valid for any arbitrary waveform once it's proven for a single sine wave, AS LONG AS that waveform is bandlimited. Discussion of this can be found in Shannon's original BSTJ article. Duh! An arbitrary waveform can be a 5KHz sine wave with 'up to' 20KHz components superimposed on it. The 5 KHz will be properly digitized but the 20KHz parts won't be. If you can't understand that, then I am arguing a point with a rock. Didn't anybody ever teach you to think for yourself? I understand it, but it's not relevant. If you think it IS relevant, you haven't done the math. I strongly recommend a good introduction to DSP which will explain the Sampling Theorem. --scott Stubborn, aren't you. I'm an engineer and I did the math (Fourier and LaPlace)in the early 70's, so I understand the sampling theorem AND it's limitations. Find a book that actually mentions the limitations instead of pretending the math is foolproof and any exceptions can be compensated for by filtering. Anti-aliasing filters are not for waveform purity but to make sure the converter isn't fooled into creating a totally bogus waveform. Play with a digital oscilloscope sometime and you will see how bad they can screw up a simple sine wave at the wrong sweep and sample rate. They make a good case for me keeping my personal analog scope to do sanity checks, although I know from experience what a mis-sampled waveform looks like. You need some humbling experience, not my typing back and forth. Someone else can teach you, hopefully before you embarrass yourself by preaching in a meeting of more experienced peers. I'm done with you. -- Bill (Sleepless biker) Baka |
#381
Posted to rec.audio.pro
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I found the holy grail that explains audiophile beliefs
On Thu, 26 Oct 2006 04:18:38 -0700, "William Sommerwerck"
wrote: What you don't seem to understand is that such experiments can be useful to learn about something you don't understand. It's precisely because people refuse to perform such experiments that we don't have reliable, accurate listening tests. "Experiment" has a particular meaning in the context of the Scientific Method. While messing around, or as a side-effect of another experiment, you may notice something interesting and form a hypothesis. Then you design an experiment to test it. You're merely demonstrating that you don't understand the terminology. |
#382
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I found the holy grail that explains audiophile beliefs
Ron Capik wrote:
Scott Dorsey wrote: bill "just use comcast", "you know the drill for no spam bots." wrote: I don't give a flying crap what Nyquist says because it is aimed at steady state sine waves at that kind of a sample rate. Every signal can be represented as the sum of steady state sine waves. Fourier showed this in the 17th century. The Shannon Sampling Theorem therefore is valid for any arbitrary waveform once it's proven for a single sine wave, AS LONG AS that waveform is bandlimited. Discussion of this can be found in Shannon's original BSTJ article. Here's a pointer to Shannon's paper: http://cm.bell-labs.com/cm/ms/what/s...day/paper.html Bob Lucky gave a great talk at the Shannon Day conference addressing the impact of latency on communication theory as applied to modern media. But that's way, way OT. Also, there was a long sampling theory discussion in rec.audio.tech a few months ago, [ Subject: 10 metres audio cable going into PC = too long? ] that may be enlightening. I give up on you guys on sampling theory versus reality. On the cable, 10 metres = about 33 feet which means a propagation delay of about 50 nS at the speed electricity travels in a conductor within a dielectric. 1/50nS = about 20 MHz , not KHz, speaking in terms of wavelength. I probably would have just gotten a good laugh out of that discussion. Later... Ron Capik -- -- Bill (Sleepless biker) Baka |
#383
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I found the holy grail that explains audiophile beliefs
Laurence Payne wrote:
On Thu, 26 Oct 2006 04:18:38 -0700, "William Sommerwerck" wrote: What you don't seem to understand is that such experiments can be useful to learn about something you don't understand. It's precisely because people refuse to perform such experiments that we don't have reliable, accurate listening tests. "Experiment" has a particular meaning in the context of the Scientific Method. While messing around, or as a side-effect of another experiment, you may notice something interesting and form a hypothesis. Then you design an experiment to test it. You're merely demonstrating that you don't understand the terminology. I think you are on to what I have been telling the "I was taught this in school so it is absolutely the truth no matter what you have seen." crowd. Education just doesn't seem to be teaching the 'question everything' method anymore. It's more like memorize it and never question anything. That's why we experiment. -- Bill (Sleepless biker) Baka |
#384
Posted to rec.audio.pro
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I found the holy grail that explains audiophile beliefs
bill "just use comcast", "you know the drill for no spam bots." wrote:
Scott Dorsey wrote: bill "just use comcast", "you know the drill for no spam bots." wrote: I don't give a flying crap what Nyquist says because it is aimed at steady state sine waves at that kind of a sample rate. Every signal can be represented as the sum of steady state sine waves. Every STEADY state signal. No. It does not have to be steady state. Fourier showed this in the 17th century. I don't think he was concerned with audio. His math works nicely with sine waves that have enough repetition to build up an approximation. He had no clue that there would be computers doing things he never even dreamed of. That's the wonderful thing about mathematics! It works the same way, no matter what the application is. The Shannon Sampling Theorem therefore is valid for any arbitrary waveform once it's proven for a single sine wave, AS LONG AS that waveform is bandlimited. Discussion of this can be found in Shannon's original BSTJ article. Duh! An arbitrary waveform can be a 5KHz sine wave with 'up to' 20KHz components superimposed on it. The 5 KHz will be properly digitized but the 20KHz parts won't be. They will be, just as long as the sampling rate is over 40 Ksamp/sec and the anti-aliasing filter is properly set up. Aperiodical signals? Square waves? Impulses? They can ALL be properly reproduced as long as they are bandlimited; the only distortion that occurs is the bandlimiting itself. Stubborn, aren't you. I'm an engineer and I did the math (Fourier and LaPlace)in the early 70's, so I understand the sampling theorem AND it's limitations. Find a book that actually mentions the limitations instead of pretending the math is foolproof and any exceptions can be compensated for by filtering. I don't think you did do the math because if you had read the original Shannon paper, you'd understand just how incredibly elegant the whole process is. It really is ingenious as hell. Anti-aliasing filters are not for waveform purity but to make sure the converter isn't fooled into creating a totally bogus waveform. These are the same thing. Play with a digital oscilloscope sometime and you will see how bad they can screw up a simple sine wave at the wrong sweep and sample rate. They make a good case for me keeping my personal analog scope to do sanity checks, although I know from experience what a mis-sampled waveform looks like. Right. You can also put a 10 MHz square wave into a 10 MHz analogue scope and have it look like a sine wave, because of the bandlimiting. The artifacts you see are due to bandlimiting. You need some humbling experience, not my typing back and forth. Someone else can teach you, hopefully before you embarrass yourself by preaching in a meeting of more experienced peers. I'm done with you. I strongly, strongly want to suggest you sit down with the original Shannon paper and read it, because once the idea hits you how the thing works, you will be surprised. It really is pretty amazing. --scott -- "C'est un Nagra. C'est suisse, et tres, tres precis." |
#385
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I found the holy grail that explains audiophile beliefs
bill "just use comcast", "you know the drill for no spam bots." wrote:
I give up on you guys on sampling theory versus reality. Don't give up! Get the paper and read it! Really, I read it when I was an undergrad, and it took me several times through to really figure it out, but it blew my mind when I finally did. --scott -- "C'est un Nagra. C'est suisse, et tres, tres precis." |
#386
Posted to rec.audio.pro
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I found the holy grail that explains audiophile beliefs
bill wrote:
Ron Capik wrote: ...snip... Also, there was a long sampling theory discussion in rec.audio.tech a few months ago, [ Subject: 10 metres audio cable going into PC = too long? ] that may be enlightening. I give up on you guys on sampling theory versus reality. On the cable, 10 metres = about 33 feet which means a propagation delay of about 50 nS at the speed electricity travels in a conductor within a dielectric. 1/50nS = about 20 MHz , not KHz, speaking in terms of wavelength. I probably would have just gotten a good laugh out of that discussion. -- Bill (Sleepless biker) Baka The communications theory discussion was part of the OT thread drift much as this sub-thread has drifted from the topic, ...and yes, sadly some of that discussion was humorous. Later... Ron Capik -- |
#387
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I found the holy grail that explains audiophile beliefs
bill wrote:
I don't give a flying crap what Nyquist says because it is aimed at steady state sine waves at that kind of a sample rate. If you can't understand that, then I am arguing a point with a rock. Didn't anybody ever teach you to think for yourself? You shouldn't try that until you understand the fundamentals yourself. So long as you are sampling a signal with a bandwidth strictly less than twice the sampling rate you can reconstruct what was sampled _exactly_ with the use of a filter with an impulse response of a sampled sinc() function. That function is infinite in length in both directions from its center which means it gathers information from the entire signal to fill the gaps between the samples. Of course infinity isn't physically realizable but windowed sinc() filters with a finite number of coeficients can approximate that ideal reconstruction with arbitrary accuracy. This is called the sampling theorem and is one of the first things taught and proved in any digital signal processing book or first course. Your reliance on intuition gained from pictures is leading you astray. Bob -- "Things should be described as simply as possible, but no simpler." A. Einstein |
#388
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I found the holy grail that explains audiophile beliefs
Scott Dorsey wrote:
bill "just use comcast", "you know the drill for no spam bots." wrote: Scott Dorsey wrote: bill "just use comcast", "you know the drill for no spam bots." wrote: I don't give a flying crap what Nyquist says because it is aimed at steady state sine waves at that kind of a sample rate. Every signal can be represented as the sum of steady state sine waves. Every STEADY state signal. No. It does not have to be steady state. Fourier showed this in the 17th century. I don't think he was concerned with audio. His math works nicely with sine waves that have enough repetition to build up an approximation. He had no clue that there would be computers doing things he never even dreamed of. That's the wonderful thing about mathematics! It works the same way, no matter what the application is. The Shannon Sampling Theorem therefore is valid for any arbitrary waveform once it's proven for a single sine wave, AS LONG AS that waveform is bandlimited. Discussion of this can be found in Shannon's original BSTJ article. Duh! An arbitrary waveform can be a 5KHz sine wave with 'up to' 20KHz components superimposed on it. The 5 KHz will be properly digitized but the 20KHz parts won't be. They will be, just as long as the sampling rate is over 40 Ksamp/sec and the anti-aliasing filter is properly set up. Aperiodical signals? Square waves? Impulses? They can ALL be properly reproduced as long as they are bandlimited; the only distortion that occurs is the bandlimiting itself. Stubborn, aren't you. I'm an engineer and I did the math (Fourier and LaPlace)in the early 70's, so I understand the sampling theorem AND it's limitations. Find a book that actually mentions the limitations instead of pretending the math is foolproof and any exceptions can be compensated for by filtering. I don't think you did do the math because if you had read the original Shannon paper, you'd understand just how incredibly elegant the whole process is. It really is ingenious as hell. Anti-aliasing filters are not for waveform purity but to make sure the converter isn't fooled into creating a totally bogus waveform. These are the same thing. Play with a digital oscilloscope sometime and you will see how bad they can screw up a simple sine wave at the wrong sweep and sample rate. They make a good case for me keeping my personal analog scope to do sanity checks, although I know from experience what a mis-sampled waveform looks like. Right. You can also put a 10 MHz square wave into a 10 MHz analogue scope and have it look like a sine wave, because of the bandlimiting. The artifacts you see are due to bandlimiting. You need some humbling experience, not my typing back and forth. Someone else can teach you, hopefully before you embarrass yourself by preaching in a meeting of more experienced peers. I'm done with you. I strongly, strongly want to suggest you sit down with the original Shannon paper and read it, because once the idea hits you how the thing works, you will be surprised. It really is pretty amazing. --scott I strongly advise you to sit down with some graph paper, so you feel like you are 'enginering' and draw yourself a picture. You can live with your 'blind' faith. Later. -- Bill (Sleepless biker) Baka |
#389
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I found the holy grail that explains audiophile beliefs
Ron Capik wrote:
bill wrote: Ron Capik wrote: ...snip... Also, there was a long sampling theory discussion in rec.audio.tech a few months ago, [ Subject: 10 metres audio cable going into PC = too long? ] that may be enlightening. I give up on you guys on sampling theory versus reality. On the cable, 10 metres = about 33 feet which means a propagation delay of about 50 nS at the speed electricity travels in a conductor within a dielectric. 1/50nS = about 20 MHz , not KHz, speaking in terms of wavelength. I probably would have just gotten a good laugh out of that discussion. -- Bill (Sleepless biker) Baka The communications theory discussion was part of the OT thread drift much as this sub-thread has drifted from the topic, ...and yes, sadly some of that discussion was humorous. Later... Ron Capik -- I think this was already off topic when I discovered this group about a week ago, totally by accident when somebody cross posted to the bicycle group I was on. Hate cross-posters with nothing intelligent to say. -- Bill (Sleepless biker) Baka |
#390
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I found the holy grail that explains audiophile beliefs
Bob Cain wrote:
bill wrote: I don't give a flying crap what Nyquist says because it is aimed at steady state sine waves at that kind of a sample rate. If you can't understand that, then I am arguing a point with a rock. Didn't anybody ever teach you to think for yourself? You shouldn't try that until you understand the fundamentals yourself. I do and they never failed me with R.F. and video work where there was a reasonably steady signal, unlike audio. So long as you are sampling a signal with a bandwidth strictly less than twice the sampling rate you can reconstruct what was sampled _exactly_ with the use of a filter with an impulse response of a sampled sinc() function. That function is infinite in length in both directions from its center which means it gathers information from the entire signal to fill the gaps between the samples. Of course infinity isn't physically realizable but windowed sinc() filters with a finite number of coeficients can approximate that ideal reconstruction with arbitrary accuracy. This is called the sampling theorem and is one of the first things taught and proved in any digital signal processing book or first course. Your reliance on intuition gained from pictures is leading you astray. I hate to have to keep telling you that while the math was good with a steady amplitude sine wave it does not work with a decaying sine wave, like Cymbols (sp?) clanging, unless the decay rate is slow enough to build up a number of samples. The proof of this would be to look at distortion and sidebands generated by even a 15 KHz signal. While it may be reconstructed within reason, a side by side comparison of the original analog and the digitized output will show differences, distortion, phase delay, and some sideband spikes a KHz or so on either side of the desired waveform. This depends on the sampling rate, the filter, and what you consider tolerable. If the sideband spurs are 80 dB down with a noise floor of -120 dB then they will show up on a spectrum analyzer but I don't think anybody could hear them. This can even be an artifact of a clocked switched capacitor filter, which while nice, is not perfect. Have you ever seen anything man made that was totally flawless? Bob -- Bill (Sleepless biker) Baka |
#391
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I found the holy grail that explains audiophile beliefs
bill "just use comcast", "you know the drill for no spam bots." wrote:
I strongly advise you to sit down with some graph paper, so you feel like you are 'enginering' and draw yourself a picture. You can live with your 'blind' faith. It's not blind, because: 1. I have done the math. You ought to, if only because it's really neat. If you don't like the original Shannon paper, get Stephen Harris' application note from the Crystal Semi databook which goes through a simplified version. 2. I have sat down with a scope, a signal generator, and converters. I have seen converters that did an excellent job of reproducing a waveform. I have seen some (like the PCM F-1) that did not, and then spent some time investigating why they did not. If the math says it should work, and actual bench testing shows it works, then to my mind, it works. --scott -- "C'est un Nagra. C'est suisse, et tres, tres precis." |
#392
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I found the holy grail that explains audiophile beliefs
bill "just use comcast", "you know the drill for no spam bots." wrote:
I hate to have to keep telling you that while the math was good with a steady amplitude sine wave it does not work with a decaying sine wave, like Cymbols (sp?) clanging, unless the decay rate is slow enough to build up a number of samples. Okay, here is an intuitive way of thinking about this: every aperiodic waveform can be thought of as a small section of a longer repeating waveform. Therefore any process that applies to a periodic waveform can also apply to an aperiodic one over a finite time period. The proof of this would be to look at distortion and sidebands generated by even a 15 KHz signal. While it may be reconstructed within reason, a side by side comparison of the original analog and the digitized output will show differences, distortion, phase delay, and some sideband spikes a KHz or so on either side of the desired waveform. But, if you actually measure it, these things do not happen. The only reason phase shift occurs is because of the bandlimiting required as part of the sampling system, and it is an implementation issue that has been overcome. The sideband spikes are the result of timing error on the sampling clock and we call it "jitter." It was a big issue back in the eighties but is pretty much dealt with today. Distortion can be caused by all kinds of things, including nonmonotonic converters. Again, an implementation issue that in no way invalidates the sampling theorem, and an issue that was dealt with a decade ago. This depends on the sampling rate, the filter, and what you consider tolerable. If the sideband spurs are 80 dB down with a noise floor of -120 dB then they will show up on a spectrum analyzer but I don't think anybody could hear them. This can even be an artifact of a clocked switched capacitor filter, which while nice, is not perfect. Sideband spurs are the result of clock timing errors. There is a canonical JAES paper from Stephen Harris on the subject, but the best intuitive description I ever heard was from alice!jj at an IEEE thing a few years ago. If you mail him he'll probably send you a copy. Have you ever seen anything man made that was totally flawless? Only mathematical models. That's what's so nice about them. --scott -- "C'est un Nagra. C'est suisse, et tres, tres precis." |
#393
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I found the holy grail that explains audiophile beliefs
Scott Dorsey wrote:
bill "just use comcast", "you know the drill for no spam bots." wrote: I strongly advise you to sit down with some graph paper, so you feel like you are 'enginering' and draw yourself a picture. You can live with your 'blind' faith. It's not blind, because: 1. I have done the math. You ought to, if only because it's really neat. If you don't like the original Shannon paper, get Stephen Harris' application note from the Crystal Semi databook which goes through a simplified version. 2. I have sat down with a scope, a signal generator, and converters. I have seen converters that did an excellent job of reproducing a waveform. I have seen some (like the PCM F-1) that did not, and then spent some time investigating why they did not. If the math says it should work, and actual bench testing shows it works, then to my mind, it works. --scott I would rather be reading something from Stephen Hawking or converting Einstein's theories of relativity into a C++ program, but I guess I can read some trivial paper and then return to this, ahhh, debate. -- Bill (Sleepless biker) Baka |
#394
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I found the holy grail that explains audiophile beliefs
I don't give a flying crap what Nyquist says because it is
aimed at steady state sine waves at that kind of a sample rate. No, it's not. The Nyquist criterion applies to band-limited signals. You don't seem to understand "superposition", either. |
#395
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I found the holy grail that explains audiophile beliefs
Scott Dorsey wrote:
bill "just use comcast", "you know the drill for no spam bots." wrote: I hate to have to keep telling you that while the math was good with a steady amplitude sine wave it does not work with a decaying sine wave, like Cymbols (sp?) clanging, unless the decay rate is slow enough to build up a number of samples. Okay, here is an intuitive way of thinking about this: every aperiodic waveform can be thought of as a small section of a longer repeating waveform. Therefore any process that applies to a periodic waveform can also apply to an aperiodic one over a finite time period. Uh-huh, If you watch a slow, say 4.42 Hz sample of a single cycle of a 2 Hz sine wave with random synchronization, and blip the sample points with a scope Z-modulation on a digital scope, then look at the single sampled reconstruction of that waveform, it will be different on every pass, assuming the digitizer is reset before each new wave. After you see the good in/garbage out, then try it with a 1 Hz wave, and while it will be better, it will still not be completely accurate. Don't forget your 2 Hz brick wall filter, which for this you can just write in C or C++. Then tell me about how the math doesn't lie to you. I have had the distinct pain in the ass of having to initiate PhD's into the real world, and had to fire one guy with a Masters from Cal Poly because he had a huge ego and nothing but a piece of paper to back it up. The proof of this would be to look at distortion and sidebands generated by even a 15 KHz signal. While it may be reconstructed within reason, a side by side comparison of the original analog and the digitized output will show differences, distortion, phase delay, and some sideband spikes a KHz or so on either side of the desired waveform. But, if you actually measure it, these things do not happen. The only reason phase shift occurs is because of the bandlimiting required as part of the sampling system, and it is an implementation issue that has been overcome. The sideband spikes are the result of timing error on the sampling clock and we call it "jitter." It was a big issue back in the eighties but is pretty much dealt with today. I just looked at a graph heavy site on the subject a few days ago and with a 15KHz signal on a modern system there were spikes about 30 dB above the noise floor about every 1 KHz on every side. It wasn't a manufacturer's ad but a site that did independent analysis, and now i regret not bookmarking it, but it was just before I tripped into this quagmire. They seemed to be all about discrediting the technology, so I don't know how real the spectrum analyzer pictures were. The spikes were post digital filter so it could have been due to the switched capacitor clock getting through and causing strange things. I have a Tek scope but not an audio spectrum analyzer so I can't verify what I saw on a possibly bogus web site. Distortion can be caused by all kinds of things, including nonmonotonic converters. Again, an implementation issue that in no way invalidates the sampling theorem, and an issue that was dealt with a decade ago. This depends on the sampling rate, the filter, and what you consider tolerable. If the sideband spurs are 80 dB down with a noise floor of -120 dB then they will show up on a spectrum analyzer but I don't think anybody could hear them. This can even be an artifact of a clocked switched capacitor filter, which while nice, is not perfect. Sideband spurs are the result of clock timing errors. There is a canonical JAES paper from Stephen Harris on the subject, but the best intuitive description I ever heard was from alice!jj at an IEEE thing a few years ago. If you mail him he'll probably send you a copy. I can probably Google it. Everything else seems to be in their database. Have you ever seen anything man made that was totally flawless? Only mathematical models. That's what's so nice about them. To borrow a quote from Bob Pease again "Just because the schematic simulates in Spice does not mean the layout will work.". That one I can really attest to. --scott -- Bill (Sleepless biker) Baka |
#396
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I found the holy grail that explains audiophile beliefs
William Sommerwerck wrote:
I don't give a flying crap what Nyquist says because it is aimed at steady state sine waves at that kind of a sample rate. No, it's not. The Nyquist criterion applies to band-limited signals. You don't seem to understand "superposition", either. Too damned many self-proclaimed experts here. Read up on SDR, or Software Defined Radio, and see the tricks they pull with intentional undersampling. You might cringe at that trend. -- Bill (Sleepless biker) Baka |
#397
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I found the holy grail that explains audiophile beliefs
bill "just use comcast", "you know the drill for no spam bots." wrote:
After you see the good in/garbage out, then try it with a 1 Hz wave, and while it will be better, it will still not be completely accurate. Don't forget your 2 Hz brick wall filter, which for this you can just write in C or C++. NO. The filtering comes BEFORE the conversion. Not after. You cannot do the filtering in the digital domain. YOU MUST BE SAMPLING A BANDLIMITED SIGNAL, otherwise you get aliasing. Many of the artifacts you do see on digital scopes at high speeds, though have to do with the lack of anti-aliasing. That's an unfortunate side effect of pushing wideband stuff out to the limits of technology; dropping the anti aliasing filter is sometimes less of a problem than keeping it there. You will see the same thing on datalogger systems. Then tell me about how the math doesn't lie to you. The math does not lie. The implementation might very well lie, and lots do. I have had the distinct pain in the ass of having to initiate PhD's into the real world, and had to fire one guy with a Masters from Cal Poly because he had a huge ego and nothing but a piece of paper to back it up. and we call it "jitter." It was a big issue back in the eighties but is I just looked at a graph heavy site on the subject a few days ago and with a 15KHz signal on a modern system there were spikes about 30 dB above the noise floor about every 1 KHz on every side. It wasn't a manufacturer's ad but a site that did independent analysis, and now i regret not bookmarking it, but it was just before I tripped into this quagmire. That would seem about typical for something of the PCM 1610 level of technology. What is even scarier is that you will see lots of noise spikes well above the noise floor even with no modulation on some of the earlier converters. Some of this was due to digital stuff leaking into analogue lines, some of it due to idle tones on poorly-implemented sigma delta systems (which is how they got called "Bitscream" converters back then). Things are a whole hell of a lot better than they were back then. And back in the PCM 1610 days, you could hear all kinds of nastiness too. I'd be curious what the site was, though. I'd bet some folks out there like Prism and Lavry, though, might be willing to show real plots from a AP testset on real converters. Hell, even the crappy soundcards on Arny's site perform orders of magnitude better than anything available in the early eighties. They seemed to be all about discrediting the technology, so I don't know how real the spectrum analyzer pictures were. The spikes were post digital filter so it could have been due to the switched capacitor clock getting through and causing strange things. I have a Tek scope but not an audio spectrum analyzer so I can't verify what I saw on a possibly bogus web site. Switched capacitor filters are bad news for audio systems for the most part, because of those modulation effects. Earlier converters tended to use simple Bessel filters with dozens of stages chained one after the other to make the filter sharp. Modern converters use oversampling or sigma-delta stuff so the only filter required is a first order rolloff. Sideband spurs are the result of clock timing errors. There is a canonical JAES paper from Stephen Harris on the subject, but the best intuitive description I ever heard was from alice!jj at an IEEE thing a few years ago. If you mail him he'll probably send you a copy. I can probably Google it. Everything else seems to be in their database. JJ is a good guy that is willing to sit down with someone at a conference, and answer stupid questions with good intuitive answers. He has helped me a whole lot over the years. Have you ever seen anything man made that was totally flawless? Only mathematical models. That's what's so nice about them. To borrow a quote from Bob Pease again "Just because the schematic simulates in Spice does not mean the layout will work.". That one I can really attest to. I have a personal hatred of Spice, mostly due to trying to get undergrads to understand that the Spice model doesn't encompass all the actual effects in the real world, and that if it did, it would be harder to design than the circuit itself. But the CONCEPT of Spice... I love the concept. --scott -- "C'est un Nagra. C'est suisse, et tres, tres precis." |
#398
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I found the holy grail that explains audiophile beliefs
On Fri, 27 Oct 2006 00:28:58 GMT, bill
wrote: Too damned many self-proclaimed experts here. I think it may be more a case of everyone here having been passed through the eye of this same needle at some time past. Some more publicly (like me and you) and others more privately. It's non-intuitive, so there's an eye to be passed through. But public or private, the Revealed Truth is very cool and well worth the effort. It's almost, but not quite, as cool as learning about dither, possibly the coolest concept in technology. Anywho, all good fortune, and glad to hear your voice here, Chris Hornbeck |
#400
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I found the holy grail that explains audiophile beliefs
In article ,
says... Chevdo wrote: In article , says... If "the scientific method" were the definition of science, then the best scientists would be those who where the best at running scientific experiments. But despite the need to carefully desingn, run, and interpret experiments, the best scientists are those with the best insight, the best imaginations. The first hypothesis has to come from -- and it doesn't come from experiment. It comes from imagination. Doesn't matter whether those people are practicing science or not, you're talking about superior people, not superior 'scientists'. Science without the scientific method is 'woo-woo', aka 'flummery'. It's how we got crap like chiropracty and homeopathy that we for some reason now can't get rid of. The fact that a genius needs to be in charge of the lab janitors to generate exception results, and that even the lab janitors have to be cognitively superior to, say, the fader janitors who call themselves 'audio engineers', and that even those fader janitors have to be cognitively superior to the original mop & broom janitors, is superfulous. However, if geniuses simply relied on their geniusness, rather than the scientific method, we'd be absolutely NOWHERE, as we were for thousands of years until we finally realized that anything not proven by scientific method is WORTHLESS. It was the Catholic church who stifled invention and burned people at the stake for heresy in questioning 'God'. That put us back an easy 1,000 years. So? There were plenty of geniuses who were around before the scientific method, and they unfortunately didn't get very far. Church again. Read about what happened to some of the inventors back around Galileo's time. I already know all about that. Does the name Bruno 'ring a bell'? If Aristotle had been around in the 16th century, he'd probably rival or exceed Newton, and if either of them had been in the 20th century they'd probably rival or exceed Einstein. Aristotle and Pythagoras were true geniuses before their time but the Roman empire (conquer, rape, and burn) set things back a bit, not to mention all the warring for no good reason. Ever wonder what was in the library of Alexandria that has been lost forever? Now you've ventured into erroneous territory. The Romans modelled themselves after the ancient Greeks, they certainly didn't 'set things back'. Obviously you didn't put as much effort into learning about Roman geniuses as Greek ones. Anyway, it was the catholic church, again, who purged much of the ancient Greek manuscripts in the middle ages. Most of those we have access to today were hidden by byzantine monks, who often wrote over them with boring christian prayer text so that they would be saved for future use (the Archamedes manuscript, for example, was written over with christian prayer by a monk looking to protect it, so he wrote perpendicular to the original text, making it much easier for modern scholars to decypher what was originally there..) Einstein had the benefit of the access to the entirety of Newton, Aristotle, and many other geniuses at his disposal, without which even he wouldn't have got very far. The point is, research was totally unreliable before the scientific method. Now it is far more reliable (though not totally, as evidenced by the amount of flummery that still gets published in peer-reviewed science journals), and the various political machinations to shape and direct research by pharmacutical companies, and over-zealous and/or ideologically biased academics. Let's see. There's primitives in office, like Bush declaring stem cell research is immoral and against God's wishes. Ring a bell? Yes, which I think I covered in the sentence you're responding to, but I also don't think that the 'anti-science' crusade the Bush Administration is on is as successful as those who lament it most passionately seem to think. They can't even push creationism into public school in Kansas. Somehow I don't think they're going to achieve any massive dumbing-down of the population at large. |
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