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Do speaker wires/power cords make a difference? How about resistors and capacitors?
Follow Ups:
...there would still be a pack of homework-eating dogs to deal with :-).
:-)
Not to single out scientists exclusively, or anything like that, but I have to wonder if this is the main reason why scientists don't act like scientists so many times.
...which as many know is a big assumption to make with software, the resolving power of the method is inherently limited by the resolution of the soundcard used.To put it as simply as possible, unless the soundcard has enough resolving power and fidelity that you would be able to listen to music through it and use it for all playback chores, without being in the least dissatisfied with the playback quality of the soundcard, then this method would not be able to reliably detect any differences for a system that was at or above (less signal aberrations) the level of performance of the soundcard.
Or put another way, this method is inherently incapable of resolving to or below the limits of the soundcard used.
While this might seem trivial, I have yet to hear a soundcard, even the $500-600 pro grade ones, that are totally transparent and the equal of near-SOTA audio components or systems. Yet the author is claiming that a $200 soundcard is all you need to perform this kind of difference testing.
In order to make this clearer, I will provide an over-the-top-example:
lets say the soundcard used had an ACTUAL linearity such that it literally failed to respond to any data below the 15th bit, or put into a different framework, that pretty much everything below -90 dB full scale was obliterated. Now, given that we really cannot record in the real world all the way up to full scale, we have to leave some headroom, even with familiar signals and events, because the alternative, possible digital recording system clipping, is just not an acceptable alternative, we will probably have less than 14 bits of resolution to record the two events, and to try and create a difference file. If there were ANY difference between the two recorded events that were below approx. -84 to -85 dB of full scale (this presumes allowing approx. 5-6 dB of headroom, which would be pushing the digital recording pretty hot), it literally would not be discernible. If the two recordings were identical above the level of -85 dB, then there would be nothing in the difference file, just residual noise.The signal below -85 dB could be COMPLETELY DIFFERENT, and we would never even see/hear anything in the difference file, just that touch of residual noise.
I know that some folks are of the opinion, that if all the subtle sonic differences live below -85 dB, then they are automatically inaudible anyway, and thus, my example would be a moot point. But unfortunately for them, this is not the case.
One of my answer's to this is that there several things that occur at levels at or below -90 dB FS in a digital system that are quite certainly audible, and one non-trivial example is digital dither algorithms. As anyone who has actually made recordings for a living can tell you, the choice of the dither algorithm is not at all trivial, and that each one sounds quite different, each one has it's own sonic signature, and each one lives in it's entirety at levels below -90 dB FS. Sony's SBM, Apogee UV-22, POW-r, ExtraBit/MegaBitMax, etc. each one sounds different, each one has it's proponents in various studios, and each one only affects the signal at very low levels, typically below the -90 dB FS point.
If the overall tonality and sound of a recording can be altered by such algorithms that all work at such low levels, then this means that we can not ignore signal aberrations below -90 dB as insignificant, to the contrary, they are capable of having tremendous sonic impact.
Of course, the obvious answer is that you can find a sound card that has 'resolution' below the 15th bit, indeed, according to the specs on a spec sheet, there are cards that are 'linear' down to 17 or 18 bits.
But the bottom line is still: can you listen to the sound cards output and not detect any amusical occurrences, is the soundcard literally completely sonically transparent? If it is not, then it could be masking other components or systems aberrations and differences to whatever extent that the soundcard lacks ultimate sonic perfection.
Note that the listening test only invokes the DAC, the ADC (input) recording portion is totally ignored in this kind of 'test'.In order to really test the sound card, both the ADC AND the DAC need to be exercised, and for this, what is required, is to record the output of a very high quality 'source', say the output of a reference grade CD or SACD or DVD-A player, and then compare the net result of going in the ADC and out the DAC on the soundcard, and comparing that to the output of the 'original' source player.
I think that by now, you are beginning to get the picture, no sound card out there can pass this test, they all add something that is not 'musical' to the sound, when compared to the original 'source', there are various and sundry issues and problems, and again, the bottom line is that if the soundcard is less than perfect sonically, whatever faults it has, it will basically be incapable of resolving any differences below it's own limit's of true resolution and clarity, and thus, any difference files will be inherently limited to it's sonic accuracy.
One could refuse to acknowledge this, and insist that bits are bits, and that if a soundcard has more than 16 bits of resolution, and the FR is XX dB flat, etc, it is therefore perfect, and can provide all the capability needed in the first place. But now we are back to the same old impasse: if the specs say it is perfect, therefore it must be perfect, and can resolve everything that matters. On the other hand, if it does not record and listen completely transparently, it has some sort of inherent limitations, and thus cannot resolve below it's own limits. Now we are back to where we started, the test is said to be good enough by one group, and it is said to be inadequate by another group.
BTW, all of the above does not even bring into question the HUGE issue of whether or not an arbitrarily timed event can ever be compared to another arbitrarily timed event. Whatever am I talking about? If you record one set of components, and then make a change, and record another set of components, the actual timing of the sampling will not be synchronized.
Say we record a transient event, an infinitely fast and short impulse. In one recording event, it gets recorded exactly when the sampling interval is catching the peak, on another recording event, it captures the peak of the impulse dead in the middle of two of the sampling intervals, that is, you have two samples that would be at (approx.) half height of that single sample.
While this also seems trivial, and that the transient information has been preserved, there is no way to exactly line up the digital samples so that the two recordings could fully lineup if the two recorded events were indeed exactly the same, thus, there would be an irreconcilable difference residue that could not be "adjusted" away.
So even though you could have recorded two exact same events, there would exist a difference that would not be real. OK, in the example I gave, the grand total of the error would be limited to a few samples, and the rest would be random noise floor, etc. But what about a complex piece of music that you were using to make the recordings? ALL the samples would be off "a little bit", and there would be a constant tiny error due to the sampling limitations involved.
But that is not the only problem, soundcards in computers are very much subject to a form of digital recording and playback error known as JITTER. Now, we would have the sample to sample timing error CHANGING, and the jitter differences between the ADC and the DAC would only add to the difference error, not cancel.
Guess what most soundcards do not even bother to spec? ADC or DAC jitter. Who knows what a given soundcards level of jitter actually is?
This kind of error would also tend to blur and limit what the resolving power of the soundcard was and just how effective it could be in creating an accurate difference signal file.
The software could be perfect, and yet the soundcard is in the real world, where nothing is perfect, and thus this type of test IS definitely limited in it's resolving power.
Knowing that there are NO soundcards out there that are sonically perfect, it would be an open question as to just how many were "good enough" to provide meaningful results.
Note that the kinds of faults and problems I am mentioning almost all result in a false null, that is, little or no difference file content.
Of course, far too many objectivists will take such a result as "proof" that there are no sonic differences, but the actual truth would be, we just wouldn't know for sure. In other words, this kind of test actually and literally suffers from the same kind of limitations as all of the DBT type listening tests: failure to achieve a significant result will NOT mean there is no difference, just that we could not detect it with that particular set-up. Period, end of the truely scientific claims that can be made.However, I have absolutely NO doubt that the vast majority of objectivists will indeed claim that such had occured, and this time they would have the cachet of "scientific proof" that there was no difference, only it really wouldn't be true.........
It looks like this topic has been heating up since I was last here. Sorry I've been away and missed much of it, we had a critical office computer self-destruct and have had to scurry to get something up and running to replace it. Anyway..
----"To put it as simply as possible, unless the soundcard has enough resolving power and fidelity that you would be able to listen to music through it and use it for all playback chores, without being in the least dissatisfied with the playback quality of the soundcard, then this method would not be able to reliably detect any differences for a system that was at or above (less signal aberrations) the level of performance of the soundcard."
That is just plain nonsense. We do not need record and playback capability that reproduces things with perfect fidelity. We need only for it to respond *in some audible way* to the changes we are trying to detect. You neglect the fact that we are not trying to make something sound the same (as we wish in traditional listening tests in a stereo system) -- we are trying to see whether there is *anything audible* left in the processed difference.
It only matters that some audible trace of the difference is left in the record/play process. This is particularly true for this test --- but NOTICE THAT:
!! THIS IS ALSO TRUE FOR DETECTING CHANGES BY EAR. !!
Just look at where your reasoning leads ---Does an audiophile need perfect recordings in a perfect room in an otherwise perfect system in order to tell differences between cables, tweaks, etc? If so, I'd have to conclude that he must not really hear any of these differences, because no system or room is even remotely perfect. Do we agree with that conclusion?
Is every recording (that he uses to evaluate gear with) superior to every piece in his audio system? Even recordings that were made before the invention of the device/cable/tweak that is making the difference (wouldn't that tweak have to have been used in the recording studio, wired with better cable than is being tested?) One of my favorite recordings for ambience is Belafonte at Carnegie Hall, has nothing improved since that was recorded about 40 years ago?
For that matter, can an audiophile hear differences in a component when it is not the weakest link in his listening system because some less revealing component is in the chain somewhere? Can he detect a change of cables using loudspeakers less transparent than than either cable? (If so, I WANT those loudspeakers!!).
"In order to make this clearer, I will provide an over-the-top-example:
lets say the soundcard used had an ACTUAL linearity such that it literally failed to respond to any data below the 15th bit, or put into a different framework, that pretty much everything below -90 dB full scale was obliterated."Yes, that is over the top (one might even suggest, a "straw man"). But, ok, lets say we have that hypothetical soundcard. When recording, the signal being recorded is not below -90dB, it is dynamic varying with time over a wide range -- it contains large waveforms. Say there are differences between the cases being tested that, if isolated, are below -90 relative to fullscale, and are audible. So do these differences have *no effect* on the recordings made? Of course not. They will change values determined by the converter at many sample points. At some point, perhaps one sample value that had been on the edge of being +12,345 instead of +12,346 will get tipped over to +12,236 due to the tiny "difference" being made. An effect will show up in the subtraction result. And (back 'under the top), the actual dither or noise with real hardware actually allows signals to be resolvable below the least significant bit, just as sounds can be discerned below noise from analog sources.
'Under the top', still, in a real test with real gear, the difference track won't be 90dB reduced. In a very good null (made from a particularly quiet audio system), when turned up higher than usual, it will leave a soft, roughly white noise floor. But people get to listen to the result themselves, they don't need me (or you) to tell them what they can or can't hear or what they think may be significant.
And why the concerns about -90dB detections (reduced from the -120dB argument in an earlier post)? (That rules out use of vinyl, you know. An exceptional vinyl recording and setup can maybe get to 73dB S/N.) Are all possible differences between cables, etc., less than 90dB down? Because otherwise the example above could, at best, apply only to such circumstances.
****Should anyone who can solder want to really hear what 90dB below their listening level sounds like, you can make an attenuator to get that that with a series 330K resistor in the "hot" path from input to output, shunted to ground by a 10 ohm resistor on the output end. Just insert that attenuator into a line-level signal path and compare playing music with and without, at equivalent volume levels. No need to just read discussions about how low that is, hear it in actuality***
I have made a DiffMaker run using a decent (but not SOTA) soundcard. It was two recordings, made through two different cables, neither of premium status), and the resulting difference was of course not 90dB down, I'm sure. But when I listen to the difference at the same gain as used to listen to either one, nothing is heard in that case. Is the soundcard cutting the info out? To test, I took one of the recordings and dubbed my voice over it at a very low level, so that I couldn't hear it in the mix. When I subtracted the two recordings, I still heard nothing at appropriate gain. Until I turned the gain up, and then, there was my voice (but still no trace of the original recorded track). So, the differences must have been weaker than my overdubbed voice, at least to my ear in my room.
"If it is not, then it could be masking other components or systems aberrations and differences to whatever extent that the soundcard lacks ultimate sonic perfection."
Picked that sentence out because it illustrates a fundamental misconception. "Masking" is relevant to conventional listening tests. People may not be able to notice some things in the presence of other things. In DiffMaker tests, though, such masking "other things" are removed in the subtraction. Masking isn't a factor -- at all, unless possibly the noise is so high as to be noticeable. The test is between hearing something or hearing essentially nothing.
"...no sound card out there can pass this test, they all add something that is not 'musical' to the sound, when compared to the original 'source'..."
And there's another one. If I "add something" to A, and add the same something to B, then subtract A from B, that "something" vanishes, per the simplest algebra. Unless the added something can also make the difference we want to isolate VANISH (not change, mind you, but VANISH without a trace), the difference will be left.
"If you record one set of components, and then make a change, and record another set of components, the actual timing of the sampling will not be synchronized."
Everyone, please read the writeup about the test we are talking about before offering such objections. The synchronization is a main function that the software DOES, as is clearly described on the Audio DiffMaker page.
Besides, think it through: if the synchronization were incorrect, then the result wouldn't be silent, right? Synch (delay) errors, and gain errors leave a (false) difference. Silence only happens when all that stuff is right."Now, we would have the sample to sample timing error CHANGING, and the jitter differences between the ADC and the DAC would only add to the difference error, not cancel."
Ok, but all these things you object to will leave a difference error, not a silent null. And any lister listens to that error. If he hears nothing he considers significant, then that problem must not have been a factor, right?
If he hears something significant, then possibly the test has an error; or else a true difference has been found. It's easy to check for the error, by the way -- record the exact same situation twice, and subtract. If you hear something significant in that difference, then the setup or gear is inadequate for what you are testing."Note that the kinds of faults and problems I am mentioning almost all result in a false null, that is, little or no difference file content."
How so??? I see only ONE conceivable scenario in the post, which is that rather absurd case of the A/D converter that somehow knows what part of a signal is the -90dB part so it can ignore it in its conversions. EVERY OTHER problem you mention would result in a false DIFFERENCE. A NULL is very difficult to achieve, everything must go right for that to happen.
How could the other claimed faults result in a "false null"?
How would the faults mentioned above result in the two recordings being the same to within an inaudible degree, even if what they recorded WAS different to an audible degree?"Of course, far too many objectivists will take such a result as "proof" that there are no sonic differences, but the actual truth would be, we just wouldn't know for sure."
I really try to not offend, but: have you tried making a DiffMaker test? The statement above sounds EXTREMELY defensive to me. You are in effect implying that you fully expect the test to result in no differences you can hear, and are getting explanations all ready to defend against that. Why?
Have you tried it already and found no difference in things? Because I haven't tested many of the controversial things, myself. I don't have access to premimum cables or tweaks, don't have extra $$ laying around to buy any. (I'd also would prefer to just provide the tools and let others test themselves and make up their own minds). So, how do you KNOW that the tests will show "no difference" to most listeners?
There is only one set of DiffMaker recordings (that I know of) which has been posted anywhere, and it, in effect, reveals a difference (to my ears, at least) between different types of coupling capacitors (though the scenario I used is pretty much stacked in favor of that result).
May I also remind other inmates that the results of the DiffMaker test are not purely objective (though they could be modified to be, but then someone would have to decide what each number meant...). The result of DiffMaker is a WAV file that anyone interested listens to -- no graphs, no numbers, no one can say what someone else hears in that file.
Another correction, contrary to reports elsewhere, DiffMaker is not a commercial product, it is freeware. And I claim it to be only a tool, not a religion!
Bill wrote:
"We need only for it to respond *in some audible way* to the changes we are trying to detect.
AND
It only matters that some audible trace of the difference is left in the record/play process."I tried to explain this in my original post, but I didn't think it would have to be spelled out word for word.
OK, let's look at that example I made, of the soundcard with the last bit missing, everything below -90 dB is GONE.
Now, we record the output of DUT A, and of DUT B using that same soundcard. We play back the DIFF file, and low and behold, we hear nothing but a slight amount of noise. This means that there is no difference between the DUT A and DUT B, right?
Nope, in fact, they could sound quite different, and we would never even know it if all we used was the DIFFmaker test.
As an extreme example, let's say DUT A _ALSO_ lost the last bit of data, everything below -90 dB was GONE. Unit B does not do this, and is quite accurate all the way down to the 16th bit. Obviously, if we listened to DUT A vs. DUT B, we would hear this complete loss of low level information, the truncation of the audio signals. But the soundcard would be inherently unable to record and thus, show a difference, because IT TOO was limited to this level of resolution.
So how good the soundcard is, makes a difference in how well we can detect a difference between two other DUT’s being recorded.
I know that some have been proposing that we examine the difference between the input of DUT A and the output of DUT A, thus eliminating any loss of resolving power between the two units A and B, but the same problem rears it's ugly head, if we tested DUT A using that hypothetical soundcard, and adjusted the gain for the two different output levels, and listened to the resulting difference file, we would again hear virtually nothing but noise, thus we would erroneously conclude that DUT a was "perfect", that is, it had no difference between it's input and output. DUT could also be tested and also found to be "perfect", but in reality, the two would have quite different performance capabilities, yet the DIFFmaker test would not reveal ANY differences between the two DUT's.
This is why I am saying that whatever the limitations are of the soundcard used to make the recordings, this is the limit of the resolving power of the DIFFmaker test.
Now, I know that we are not dealing with soundcards limited to 15 bits of resolution, that was a gross example to make my initial point that the soundcard can and WILL adversely affect the results if it is less than perfect itself.
I once again point to widely accepted opinion and professional practice, that NO soundcard is considered sonically perfect or even sonically SOTA, much less a $150-200 soundcard, therefore, the DIFFmaker test is going to be limited in how much resolving power is availble to record differences.
If we postulate that no soundcard even approaches the SOTA sonic quality as exemplified by such components as Mark Levinson, Krell, Audio Research, etc, then we must also acknowledge that the DIFFmaker test will NOT be able to provide a valid difference file for one of these components, nor will it be able to provide all of the subtle differences present and of interest in audio cable, and many other audio components.
None of this addresses the obvious issue of the gain adjustment when trying to do a DIFF maker test for an input/output test of the same unit, obviously, if the soundcard used does not adjust ONLY gain, but introduces any other signal aberrations when a gain adjustment is made, then this distortion of the soundcard will be added to the resulting difference file.
In the hypothetical case we have been examining, instead of hearing just noise, we might hear some very small residual of distorted music, just enough to look as if the DUT was slightly less than perfect, BUT, with a difference that was very low. This might induce us to believe t6hat the DIFFmaker test had actually been wholly unsuccessful and had provided valid and worthwhile data, even though the entirety of the 16th bit and below was missing.Bill wrote:
"Just look at where your reasoning leads --- Z"Indeed, it leads to the inescapable conclusion that the DIFFmaker test is not magic or supernatural, but is limited in the same manner as all other audio systems.
Can we hear 'past' a weak link component? Yes, to a certain degree, but I think that it is wise to point out, that with a RECORDING process taking place, anything lost is gone forever, that once it has been 'left behind', it is never coming back, while with real world audio systems, we can can kinda sorta listen past a weak link component, and when that weak link component is upgraded, we can now hear what the other components are doing that we were not fully able to do belfore. However, if the soundcard is upgraded or replced, all the old recordings are still limited in what they can show or reveal, no new difference information will become known until we make that recording comparison again, using the new soundcard.
Bill wrote:
" Say there are differences between the cases being tested that, if isolated, are below -90 relative to fullscale, and are audible. So do these differences have *no effect* on the recordings made? Of course not. They will change values determined by the converter at many sample points. At some point, perhaps one sample value that had been on the edge of being +12,345 instead of +12,346 will get tipped over to +12,236 due to the tiny "difference" being made. An effect will show up in the subtraction result."Of course, some effect of a difference down at -93 dBFS will still alter bits higher up the ladder, BUT, these changes are greatly reduced, and would tend to fall, by definition, into that realm where they would register as if they were less than a true difference at -93 dB.
In other words, loss of the entire 16th bit will render the difference virtually deaf to signal aberrations below -90 dBFS.Bill wrote:
"And (back 'under the top), the actual dither or noise with real hardware actually allows signals to be resolvable below the least significant bit, just as sounds can be discerned below noise from analog sources."Sure, but now we are limited to 15 bits and a little more with dither, instead of 16 bits and a little bit ore with dither. BTW, dither designed to augment and extend a 16 bit system WILL NOT adequate exercise the 15th bit to "fix" it, the dither level would have to be increased correspondingly, and so, you argument falls apart in a very real sense.
Bill wrote:
"But people get to listen to the result themselves, they don't need me (or you) to tell them what they can or can't hear or what they think may be significant."Of course, but what they may not know, probably DON'T know, is that the difference file is still limited in a very real sense and manner, to the sonic quality of the soundcard.
It still boils down to whether or not you think that the specs are enough, or the sound as it listens is what matters, if the soundcard is not SOTA sonically perfect, then you can not test SOTA audio components with it and expect to see/hear all that might be there.
Bill wrote:
"And why the concerns about -90dB detections (reduced from the -120dB argument in an earlier post)? "I think that you must be referring to someone else's post here. My argument is that the known and obvious changes in the sound when different dither algorithms are used, is a form of proof that signal aberrations that are at or below -90 dBFS DO impact the sound. Each dither algorithm does sound different, and not just at low levels, but all across the range of the music signal. Music doesn't 'live' at 0 dBFS or even -10 dBFS, but rather, across a wide gamut of mini ranges as it swells and flows.
Bill wrote:
"(That rules out use of vinyl, you know. An exceptional vinyl recording and setup can maybe get to 73dB S/N.)"No, no, and no.
First, I have worked extensively with measuring and characterizing the vinyl playback system, I worked for Discwasher for years, and have actually done the research (in some cases, original and unique work). A well pressed virgin vinyl recording can reach a dynamic range in excess of 80 dB, from approx. a clean noise floor of -70 dB (where signals can be heard below for approx. 6-10 dB) in the midrange, up to about +10 to +12 dB above 0 dB (where 0 dB is defined as 3.54 cm/sec modulation velocity per channel, or 5 cm/sec lateral, the Shure tracking test record had levels of 25 and 30 cm/sec, which approximate to +18 dB).
If the record was from a direct to disc session, it could even exceed that range, up to about 85 to 88 dB. Some of the Sheffield direct to disc records regularly hit +18 dB, and had a noise floor of -70 dB or better, all in the midrange.
Second, it does not necessarily rule out the use of vinyl as a test medium, because as we are all aware, you can hear things on a vinyl record that allow you to discern differences between SOTA power amps, preamps, and audio cables. It is not JUST the dynamic range, or the low level resolution, but rather, the entire package of performance during the playback event. Where vinyl excels, is that when it is reproducing the music where it 'lives' at moderate to low levels, it is fairly linear and resolving, without some of the issues that plague digital audio.
Bill wrote:
"Should anyone who can solder want to really hear what 90dB below their listening level sounds like .... "Wrong. I did not say -90 dB below the music level. Music level implies the average level, not the peaks, and not 0 dBFS. Relative to the average music level, a -90 dBFS signal is really at somewhere around -50 to 55 dB from the average level of the music, which is quite a lot off from the -90 dB straw man you set up.
Bill wrote:
"In DiffMaker tests, though, such masking "other things" are removed in the subtraction. Masking isn't a factor -- at all, unless possibly the noise is so high as to be noticeable. The test is between hearing something or hearing essentially nothing."Perhaps masking was a poor choice of words on my part, I was not referring to the classic 'masking' as applied to psychoacoustics and MP3's, but rather more in the nature of the poorly performing soundcard occluding or losing the low level signal information to one extent or another, and thus, 'masking' what was going on with regard to the difference file.
Again, in my hypothetical example, the soundcard will literally not record anything below -90 dB, it is MISSING, gone, poof! ALL differences below that level will be utterly and completely 'masked', or as a better choice of words, obscured. The difference file will contain virtually nothing but noise, or perhaps just enough residual of the original signal that it seems as though the test worked, but in reality, YOU COULD NEVER KNOW FOR SURE whether or not the soundcard was adequate to the task, or not, unless you had conducted a huge series of well done and executed controlled listening tests that incorporated numerous controls and verifications, as well as open testing, to try and verify or confirm that the soundcard was sonically transparent.
Without that kind of extremely costly and time consuming verification, you could not automatically assume that the soundcard was god enough, or did not have it's own flaws literally covering up those of the DUT's.Bill wrote:
"
And there's another one. If I "add something" to A, and add the same something to B, then subtract A from B, that "something" vanishes, per the simplest algebra. Unless the added something can also make the difference we want to isolate VANISH (not change, mind you, but VANISH without a trace), the difference will be left."Sorry, but the "added something" may not be a simple addition, but rather, an alteration of the signal, a signal aberration would again be a better term. In the case of the hypothetical soundcard which loses the 16th bit, nothing at all is added to the signal, just removed. However, in my extreme example, (in order to clearly make my point), the loss of signal content is a well defined and obvious thing, let's call it truncation to 15 bits. But what about a soundcard that is not so obviously distorting the signal? What if it is 15 1/2 bit s accurate (a not uncommon situation with a 16 bit based card), and that it has a frequency and level dependent loss of signal, these things WILL NOT necessarily cancel out with a set of recordings of DUT A and DUT B, or with an input/output comparison, but will still tend to obscure any low level differences that the DUT's do have that are below the level of soundcard corruption.
Bill wrote:
"Have you tried it already and found no difference in things? "Actually, I have tried to use a soundcard in a similar manner, by nulling the test signal with the output of a DUT, as well as to compare two DUT's with each other via a difference file.
I manually edited the timing to match the nearest sample, using both 'ways' {creating two diff files}when it was not coincedent with each other (almost all the time).
At first, I too thought that the soundcard should not matter, but found that using a SB16 just would not do the job; even after I tried a studio grade soundcard, the Echo/Event Darla, there was still too much contamination from the soundcard to do the job properly.
I sincerely hope that you do finally understand that the soundcard quality DOES matter, and that it can not be ignored, that it DOES NOT fully cancel out, only a portion of the soundcard distortions are cancelled out. Losses of the signal, non-linear or otherwise, simply do not cancel out. Non time coincident additions do not cancel out (computer EMI/RFI contaminants) either, and their resultant effect on the audio circuitry can be erratic or misleading with regard to how the DUT's AND the soundcard respond to them.
If you can not see this is the case, then I am sorry for you, it is not THAT hard to figure out, especially when it has been spelled out for you.
Jon Risch
It is pretty obvious from your reply, that you really didn't get most of the points I was trying to make, unless it is deliberate obsfucation on your part.I will reply point by point when time permits, as it is late here now.
Jon Risch
Bwaslo, you should have more experience before commenting.
First, sound cards DO effect the sound. IF they didn't, we would pronounce them perfect and not go any further. It would mean that if I added the AD-DA converter that I purchased for less than $200 on sale, from Leo's Music, a pro audio outlet, in series with my preamp, that it would be perfectly inaudible. We have found that this sort of thing IS audible.
Next, you act as if we think that all the subtle changes that we make in audio can be made obvious to everyone, everywhere, with cost effective audio equipment. No it can't, BUT we can detect differences in our own equipment that we have become used to, with audio sources that we know are recorded to a very high standard of quality.Technically, this difference test, while potentially being an important tool, will suffer from the limitations of the sound card. Perhaps, not as much as normally, because some of the more obvious harmonic and IM distortion will be cancelled out, that was generated by the op amps before and after the A-D processor, but the noise will still be added and will be at least 3 dB higher than the wire as it would normally be evaluated. This is because the subtraction of the coherent parts of the signal will not cancel the random effects of the noise from both samples taken. In effect, I would think that I would see mostly noise in the error signal. Another big problem is phase differences at both low and high frequencies. This might not be too much of a problem with a wire, but a component (such as a cap) would seriously suffer differences with another component.
It would seem that only the simplest things, like a wire, could be tested easily with this device without linear distortion adding and confusing the error output.
Another really important factor is the OMISSION of information caused the the sound card. If the sound card was so good in the first place, would it omit infomation? Listening tells me that it does do just that. If you omit information, how do you get it back, to see how it has been changed?
It should be noted that we have been trying to do this sort of thing for decades. We did it with 10 bits and a 50KHz clock rate, 1/3 of a century ago. We could not see any significant differences in good (Levinson JC-2 level) op amps, and I have always wanted even more resolution like 16 or even 24bits, but like Jon Risch, I doubt that it will be very effective. However, it would be a Godsend for us, IF it really worked.
John,I am quite well informed on this, I think, thanks. I bet I've tested and listened to many more sound cards than anyone reading this forum. I fight problems in (some of) them way too often, if fact. I read the disagreements, here, but they don't seem to hold water to me. Let me try to spell out why.
"First, sound cards DO effect the sound. IF they didn't, we would pronounce them perfect and not go any further"
Guys, just where did I say that sound cards don't affect the sound? I admit I'm not the best writer, but I know didn't write that, not ever. In fact I test soundcards regularly and have a pretty good idea what most can and can't do. OK, soundcards can and do affect the sound. There!
But, SO WHAT? So does every loudspeaker ever made. Yet, somehow, amazing as it is, people hear differences in equipment while listening through loudspeakers! Same with some amplifiers that can affect the sound. And the sound being changed was recorded with mics that affected the sound, too. Clearly these don't automatically invalidate testing, then, so why do they invalidate a differencing test?
Yet again: the soundcard CAN vary the sound, and still be quite usable, just as can the the incredibly imperfect loudspeakers we all use in domestic listening rooms. For the differencing test, the soundcard only has to respond to the difference, not reproduce it verbatim.
Yours and Jon's argument seems to be (correct me if my understanding is wrong) that:
(1) one can never detect a difference, even with mathematical processing, between two signals recorded with soundcards. Because the soundcards can affect the sound. Yet,
(2) a listener *can* regularly detect such differences, by ear. Even if a number of the components he listens through affect the sound, too. Even if they affect the sound more obviously than a soundcard does. Even if that listener is a constantly changing organism every minute, with changing moods, states of hunger, thirst, tiredness, age, etc. , never exactly the same for any two tests.
Can't you see why I disagree? Can the argument of "not enough resolution" be valid in the one case, yet not in the other?I can rather easily make two recordings with an intentional inaudible difference, (inaudible to me, of course, who knows about others!) which difference can be exposed and made audible by the software. But I'd like to find situations where there are recordings that have differences that ARE said to be audible and can ALSO be exposed -- revealing the reported difference for more listeners -- by differencing. Now, surely, having that difference be audible can't automatically make it even MORE difficult to detect via differencing, could it??? The program can make something that is inaudible audible, it is not hard to do that. It doesn't seem to me to be much of a stretch to think that it could also render something that is audible to some (but not to others) more audible, and reduce disagreement about the situation.
Note that I have also not said that every difference can be resolved by this using any equipment. I sense people think I'm trying to prove nothing is audible unless it can be measured. I'm not. I don't care much about proving something that can't be proven like that. I'm only trying to make audible differences, if they exist, more obviously audible by removing the other program material that can mask them. If someone as a result finds that a differencing convinces them there is no relevant difference between two things they test, based on what they hear (or don't hear) themselves with their own ears, that will be their conclusion (which they are free to make).
Obviously, if a differencing test were made through a telephone, the difference test would leave (at least) obvious noise, and no conclusion could be made. But it wouldn't leave silence. There might be signal behind that noise, but not apparent SILENCE altogether. So that just becomes a simple case of inadequate equipment or setup for the test. That can easily be checked with a dummy test. We can all agree that the test can't be useful in all cases. But I think it can be useful in very many cases. Particularly useful, I think, is the characteristic that lets people evaluate audibility more readily using their ears to hear the difference recording, without feeling challenged or being unduly influenced by expectations in either direction. But I see that some here do not agree. Que Sera.
"the noise will still be added and will be at least 3 dB higher"
Yes, noise will increase 3dB, but if the 3dB increased noise is not heard, I submit that it doesn't matter. The noise is always there, and it is always 3dB higher than some other level. (Anyway, even if it did matter, it wouldn't be hard to eliminate the 3dB increase: the tests could be recorded twice for each case and each recording set for a case averaged coherently. That would reduce noise by 3dB again, back to even. We could even average more than two for each situation and *reduce* the overall noise if we wanted to!).
"Another really important factor is the OMISSION of information caused the the sound card. If the sound card was so good in the first place, would it omit infomation? Listening tells me that it does do just that."
John, you are aware that such omission is in terms of your mind's rediuced ability to extract meaning and impressions from a signal that now less resembles the original signal you can relate to. We don't hear with spectrum analyzers. We can't hear degraded hash as well as we can notice degraded music. Information is a matter of context. But subtraction is not a matter of context or affected by masking (it has that very significant advantage over traditional listening tests for difference). The missing information is not from some buried signal completely vanishing from the larger signal with no effect, how could it *possibly* do that??
"Another big problem is phase differences at both low and high frequencies. This might not be too much of a problem with a wire, but a component (such as a cap) would seriously suffer differences with another component."
That is true, agreed (and experienced). Very small mismatches leave rather large difference results, the amount is easy to calculate, infact (see the Help file of the program). But that only means that care must be taken to handle these KNOWN effects. I've done such a test with capacitors, which of course make a first order filter (hp or lp, depending) in combination with the source and load resistances. It is a simple matter of adding resistances (or extra capacitance) to match the rolloff characteristics; at line levels, a 10 turn trim pot in shunt somewhere has done well. There is even a simple response analyzer system built into DiffMaker to help in adjusting these. What if the caps aren't matched well enough? Then, it will show up as an existing difference (which it is). Might not some characteristic of that trimpot cause an audible difference? Maybe, but if so, that will show as a difference, too. But, clearly (to me, anyway) neither has ANY way to show a false null. Mistakes make differences, not nulls. Presumably interesting difference results will result in further exploration to detect whether we are getting fooled by such things.
--
There are some comments in this forum that the DiffMaker test suffers from the same objections as do ABX tests: that the hardware used for the test can invalidate the result due to lack of transparency. I submit that is just not so. Hardware might degrade signals, so that they become less easy for a human to relate to, because they sound less like something humans care about (music). That might be a factor with human listeners involved with music during ABX. But after differencing, the listening is only for absense or presence of remaining signal, the main common signal is gone. The hardware simply doesn't know what part of the signal it sees is the original music and what is from the change made by the cable/tweak/etc. It's all just "signal" to it, one thing. I'm surprised that that, at least doesn't seem obvious to others here. Hardware must treat all parts of the signal about the same, it has no way to selectively strip one part off, based on how it got there, so that effects of the change completely go away.
....and sheesh, I'm tired of writing about this. Good night, all.
I think that what Bwalso is saying that regardless of the measuring ABILITY of the soundcard, it is still a valid CONTROL, since we're using the same soundcard for both tests.If you have mediocre speakers but can hear an improvement when you change sources - agreed - it must be QUITE a significant improvement. The trick was that you used the same speakers to audition BOTH sources.
It's a no brainer that one would want to have the lowest noise floor possible to see even the most minute differences.
There is also a difference bewteen the dynamic range and SNR ratings of a codec ON a soundcard and what the actual SNR will be when a loop-back noise-floor test is done.
I also imagine one can't do bit-for-bit comparisons unless the stimulus is synced with the recording equipment for each iteration. Is this correct?
Once again, I applaud your effort in making this program. It is possible that it will supplant linear subtraction in many cases, such as the setup that Walt Jung and I used to measure caps more than 20 years ago. However, I think that comparing wires will be difficult with this test, but that does not show that wires have no sonic difference between them. That has not been my experience, and since I don't make wires, I would prefer that all 'well engineered' wires sounded the same. It would save me a good deal of time and effort, in making my products.
The bottom line is the resolvability of the AD-DA converters and their attendant circuitry, which seems to remove too much low level information, and this would not necessarily be made audible by cancelling out the main signal. It would seem to me that you are looking at differences residing at the 1 or 2 bit level in many cases.
Perhaps I am wrong. What differences have you found between common components with this test?
Hi JonWith an instrument like a microscope, TEF machine, telephoto photography or most processes, which extend your senses, there is a limit to its resolution, its ability to discern detail or detect anything.
You have written about this nicely as it applies to soundcards etc.
When recording, it is clear that the 24/96 option is audibly superior to 16/44, especially with quiet program.On the flip side, the usefulness of these devices is not entirely excluded by not having “resolution to infinity”. The strength of such tools in the range that spans between where your unaided senses leave off and where the resolution limit is.
In home audio, it is not that easy to do a null test which is what this software allows one in an easy way to audition.
In the case of say 80 dB of resolvable level, lets remember that while one might be able to detect some things –80 dB down, that is like hearing something which is .01% of comparative Voltage level, one hundred million to one in power..
Consider that if one hears anything with recorded music, it resides within the say 30-40dB dynamic span in range that most music occupies.
For the person who is sure they hear large differences between say two speaker cables, “if” those differences are electrical in nature, they would also be audible as a comparatively large difference between one end of the cable and another.
Same for amplifiers etc, what seems like an obvious audible difference with music is not produced by having two identical signals.
The down side if any I think is that hearing the null residual does not tell you anything about what to do, just what its signature sounds like.I find it interesting that many couch this entire thing as part of an Subjective vs Objective debate. It is weird that some have been taught to think that audio engineers don’t listen to what they make, I think there is more than just smugness too, it’s not understanding the process.
In reality, if you’re interested in getting to the bottom of something technical, any and all means becomes an valid option if it is actually illuminating. Adopting a specific posture is counter productive.
I think it is a “flaming issue” partly because of the significant level of BS mixed in by manufacturers looking to have a cool story to tell as opposed to same old same old.
The market / public is primarily educated by the manufacturers through the industry magazines (who in general participate in marketing it), I guess one shouldn’t expect a research oriented posture when it’s really about creating mystique and selling.
That leaves the small minority who bother to look further than hifi magazines.
Jon, are you going to NSCA ?
If so, stop by if you have time.
Best,Tom Danley
Tom wrote:
"In the case of say 80 dB of resolvable level, lets remember that while one might be able to detect some things > 80 dB down, that is like hearing something which is .01% of comparative Voltage level, one hundred million to one in power.. "I realize that -80 and -90 dBFS are way down there, but first let's put them in proper perspective.
On a 16 bit capable digital recording/playback system, you have approx. 96 dB of theoretical dynamic range available. You can record the average signal levels at 0 dBFS, you have to leave headroom for transients, or looked at another way, you have to allow for the crest factor of the signal. Live music is one of the most unpredictable signal sources, and the crest factor can be enormous compared to common test signals. Thus, it was common practice to place the VU meter "0 dB" for digital audio recording systems at approx. -26 dBFS, thus -96 dBFS becomes -70 db VU.
Now with a known signal source, such as a prerecorded bit of music, or an electrical recording of music while monitoring a capacitor or cable, you have a much better idea of the maximum signal level, and can push the digital system near it's recording limits, but good instrumentation practice will tend to keep the signal level some -6 dBFS to -10 dBFS down from the absolute maximum of 0 dBFS. Obviously, if you clip, the difference test will have a lot of false garbage and would be invalidated altogether.
Given that most sound cards that are 16 bit are NOT accurate down to 16 bits, this is not a trivial thing, we are now down to a situation not that far from what I was writing about, a loss of the laast bit's worth of accuracy, and not going all the way to the MSB by keeping the levels down, we are now down to a 14 bit recording system. To expect that to be able to capture the subtle details of an analog system, is just not being reasonable at all.
Remember, when I was talking about the dither algorithms? They all live at -90 dBFS and below, but they ALL sound diferent. For the typical soundcard recording the output from a digital audio workstation, and toggling baclk and forth betwen the various dither choices, it is entirely likely that the subtle differences would be entirely lost, that the difference file would only have very low level noises in it, along with some of the soundcards own colorations.
You mentioned a microscope, but trying to test high performance audio components with a $200 sound card is like trying to check the accuracy of a well built and executed Lab-grade microscope, with a magnifying glass from the dime store. Not going to be very revealing.
Tom wrote:
"For the person who is sure they hear large differences between say two speaker cables, “if” those differences are electrical in nature, they would also be audible as a comparatively large difference between one end of the cable and another.
Same for amplifiers etc, what seems like an obvious audible difference with music is not produced by having two identical signals."Actually, I was under the impression that the test method was to involve recording cable A's output, and cable B's output, and comparing them to each other. Same for the power amps, rather than trying to compare input to output for one device. As you noted, trying to interpret the I/O difference is going to be futile and frustrating, while comparing two different outputs is at least at the same signal level, and involves the same interface issues.
BTW, I will not be going to NSCA this year, sorry, I would have liked to meet you and chat.
Jon Risch
...isn't the situation very similar to DBTs? The shortcomings of the test platform swamp whatever differences might be detectable.For two decades (and I know you're with me on this) I've advocated the simple expedient of testing the test by submitting it to trained listeners who are familiar with real differences. Without such calibration the procedures are useless.
That is one way to do it, if it is don right. And, some do it this way.
You will test the "object" and only use there feedback with respect to the test that they passed.
x
nt
"The shortcomings of the test platform swamp whatever differences might be detectable."Obviously the goal of developing a useable test fixture would include minimizing that effect. It is not impossible to design this possibility out of the equation.
"I've advocated the simple expedient of testing the test by submitting it to trained listeners who are familiar with real differences."
This would not only be a good idea but imo necessary as a proof of concept. Of course choosing a test panel that can be trusted is another discussion.
"Without such calibration the procedures are useless."
I doubt any engineering team that could actually pull off the job would disagree with that statement.
x
--
The threshold for disproving something is higher than the threshold for saying it, which is a recipe for the accumulation of bullshit - Softky
And exactly how does one determine if someone is "trained" when no test methods can be applied to confirm?
Oh yes, I forgot. Self anointment.
Get real Clark Kent LOL.
At least we have a new one now.
Self determined "Golden ear" delusional audiophool "perceives" difference in transmogrified CD, wire X, cryo treated widget Y.
Sane but untrained peasants cannot "hear" this.
Reasons Why:
1) Their system lacks resolution. "Mid to Lo Fi" components and unblessed widgets.
2) They are deaf or lack proper Jedi training even though they listen to more live unamplified instruments than Wacko Joe "believe me I heard it" Audiophile.
3) Their components, wires and widgets are not sufficiently "Broken In" even 2yrs later.
4) Their sound card stinks.cheers,
AJ
The threshold for disproving something is higher than the threshold for saying it, which is a recipe for the accumulation of bullshit - Softky
Howdy John Risch,I"m wondering if much of the concern you have raised can be alleviated to at least be considered relatively insignificant by human hearing performance standards by meeting some appropriate test system (the PC used for the test and the signal source)design requirements.
1. Use a periodic reference signal which is designed to be guaranteed to be "good enough" as compared to the sound card specs such that the error between reference signal cycles is beyond the ability of the sound card to resolve, ie maybe a highly accurate signal generator putting out a square wave. This making it easier to get the effect of better time alignment of the before and after signals during comparison will give us a higher level of faith that waveforms sampled at different times through the same system can be used for comparison and assumed to be equal even though they are not the SAME waveform. I believe that an appropriate square wave contains all of the needed information but a dirac would be even better if you can somehow generate a good dirac (kidding about generating the dirac of course).
2. In order for this system to meet standards useful for human hearing comparisons it is necessary to guarantee sufficient sound card quality, ie accurate to the limits of human hearing for all the potential sources of signal digitizing/analog conversion errors. This will allow us to believe that the soundcard isn't introducing some audible distortion.
If you can meet both of these requirements then you will be able to test to the limits of human hearing which by my logic is good enough. Also you gurantee that audible differences will be visible in this tests output data. Am I wrong?
Point taken about unknown soundcard specs. Jitter, linearity, SNR, samplerate/conversion rate, etc can all be had in quantities that will be good enough to make each associated error boyond the limits of even the most high performance human hearing in existence. Granted the parts and associated board layouts don't come free but this isn't unubtainium to the well endowed design firm. I can't explain why the existing high end sound card manufacturers would not have already filled this niche if that is indeed the case as you have said. Maybe I am missing something here.
RE #1, nothing that you can do will take the sampling uncertainty out of the equation. It will still exist, and providing a square wave or impulse will not fix the problem or allow the software to align the signals any better (here, I am assuming that Mr. Waslo did a good job on the auto-alignment portion of his differencing software, which is not too terribly hard to do I am told). Nothing can be done either about the inherent and intrisnic amount of jitter the soundcard and computer system have, a reference sq wave or an impulse would not provide any better performance here.I tried to do this once for audio cables, create a method of differencing the signal from two different cables, usng a "studio grade" soundcard with excellent specs. Thus, I speak from direct personal experience on this exact subject.
Unfortunately, the sampling uncertainty was a major problem, and the really frustrating thing was, that even when I used the soundcard itself to generate the test signal, running out of one channel into the other, you would think that the timing would be exact, the same every time. Nope. Modern operating systems apparently can interfere with the exact timing of the signals out and into a soundcard, ANY system activity will interfere with the actual timing of the ouput and input, and the system is never really idle. Forget about making two channel recordings of two different analog events, there is no way to eliminate the sampling interval errors.
RE #2, if you have those tests, measurements and criteria availble, then I can guarantee a Noble prize for you, for you will have now succeeded where countless others have failed.
To this day, WE DO NOT HAVE A COMPREHENSIVE SET OF MEASUREMENTS THAT WILL ALLOW US TO FULLY DETERMINE WHAT THE LEVEL OF SONIC ACCURACY TO THE HUMAN EAR WILL BE PERCEIVED AS.
It is NOT sufficient to say "this soundcard has XX bits of resolution, it has FR from DC to light +/- YY dB, it has distortion under ZZ % THD", none of these measurements, nor any combination of the traditional and typical measurements, can even begin to come close to fully defining what will be heard or not heard.
Thus, this statement:
"Jitter, linearity, SNR, samplerate/conversion rate, etc can all be had in quantities that will be good enough to make each associated error boyond the limits of even the most high performance human hearing in existence."
is just not true.If you truly believe that the specs are fully sufficient to sonically characterize a audio component, then you won't even be bothering to perform such a test in the first place, and will likely also be willing to accept the results from any such "sonically perfect" soundcard differncing test, even though I have pointed out the fallacies involved with that type of testing.
See my reply to Dick Hertz down below, and I pose the same question to you that I posed to him at the end: why not create the ultimate systam based on one of these sonically perfect $200 soundcards?
"nothing that you can do will take the sampling uncertainty out of the equation."No, but you can design with the intent of reducing audible distortions in the final product by addressing specificly noted shortcomings of past effeorts with state of the art technology if necessary until the specific complained about effect gets minimized to the limits of technology or to the point that the problems are effectively eliminated.
"providing a square wave or impulse will not fix the problem "
my bad I only recomended squares and impulses since I believe that communications theory says we will be able to get more accurate transfer function approximations from the device under test using these types of stimuli than we would using some random audio sample. I didn't mean to imply that I KNOW that these particular waveforms are any more useful in guaranteeing perfect time alignment of the sample sets being compared, other than what may be implied by the fact that a very strong reference quality signal generator market exists with a wide range of highly accurate commercial products being widely available which could potentially be leveraged if it were deemed to be part of a solution which worked better to achieve the desired results. On the contrary....
"I am assuming that Mr. Waslo did a good job on the auto-alignment portion of his differencing software,"
With all due respect, I gottta say that's brave. I'm proposing here and now that doing this correctly may even be what lies at the root of why previous attempts at this particular type of analysis may not have yeilded the desired results. I'm not an expert on how to or the motivation of which spot on a signal to choose which to trigger but I'm starting to think triggering on the signal properly and the assumptions made deciding just where to trigge may be critical to getting good results. I believe this is quite a bit more precise and critical than the simpler time aligniong software (cakewalk etc.) out there used to line up music tracks for musician song editing would allow for. For example: I'm guessing the nuances of time and frequency information differences between cables under test might be quite a bit more subtle than the time alignment and signal stretching effects necessary to satissfy a drummer that his drum beat track is sufficiently lined up with his mate guitar riff in order to sufficiently "rock on". In other words we may require much more accurate time alignment software/hardware capabilities such as on the order of what digital sampling scopes are capable of in their triggerring algoriyhms and circuitry rather than what a musician software/hardware package allows for. It would be interesting to know how this software stacks up and what assumptions ar being made. Not that I am enough of a signel processing whiz to be useful at analyzing the algorithms myself without more research.
"Nothing can be done either about the inherent and intrisnic amount of jitter the soundcard and computer system have, a reference sq wave or an impulse would not provide any better performance here."
If you are referring to the synching of the A to D and D to A operations in time with the signal being sampled as well as to each other, well I have a couple things to say about that. First, If the errors of each domain conversion are reduced to below the level of what is audible then it is my belief that this is not necessary in the first place as long as proper time alignment algorithms are used for doing the difference comparison. Perhaps this downfall of modern equipment you are seeing is related to this application pushing the limits of what the hardware is intended to do, ie the combination of error due to digitizing and due to re generation of the analog signal. Perhaps with this in mind, if it indeed turns out to be part of the problem the next generation of or special niche of soundcards will be designed to address this issue.
If in fact you are referring to the variation of clock to clock edge timing accuracy in an already synched system, well, that just speaks to some unfilled sound card niche since these soundcard manufacturers of these "pro" caliber soundcards you mention later in your post need to quit cheaping out. That of course is assuming you have heard some digital technology which is jitter free enough that these dream spec sound cards we are going to dream up might aspire to have if the soundcard manufacturers creating a new card were to try to cater to the desires of a consumer in this potential market.
"To this day, WE DO NOT HAVE A COMPREHENSIVE SET OF MEASUREMENTS THAT WILL ALLOW US TO FULLY DETERMINE WHAT THE LEVEL OF SONIC ACCURACY TO THE HUMAN EAR WILL BE PERCEIVED AS." and "
admittedly my statement regarding technology being able to address all humans hearing capabilities is more or less unproveable. Sorry I take it back. I still believe the point I was trying to make is valid though that if you weren't having success identifying clearly audible differences in a black box under test using equipment which is good enough perhaps it is the assumptions made during the data analysis that is flawed not the equipment. This brings me back to the point I raised above about whether best practice regarding signal triggering and the assumptions used to time align the before and after comparisons are being used to gurantee accuracy of the desired result.
d
I admitted the overly assertive tone is not appropriate and have asked for forgiveness. Please see my post above. Besides I am allowed to make a subjective evaluation aren't I? For all you know I am deafer than mister Magoo.
.
...wouldn't the limitations of the sound card and everything else in the chain cancel each other out? Ultimately the idea is to show where the differences, if any, are between the two items in the test. Therefore, it doesn't matter as long as the comparisons are made through the same computer, software, sound card, etc. It's like saying red and green look different through blue sunglasses. While that is true, RELATIVELY speaking one can discern there is a difference between the two colors. As I understand it, that's the purpose of the program, to point out whatever differences there might be between the items being tested.
I suggest that you re-read my post again.The bottom line is that if the soundcard can not adequately resolve certain levels or types of audio signals (errors of subtraction), then there will be nothing there to "difference", the sonic differences between the two recordings will be lost as they were incapable of being recorded in the first place.
If the sound card commits errors of addition, and adds a certain form of distortion to the signal, yes, it will be added to both recordngs, but once again, it will be blurring and hiding what either recording may be doing or not doing AT OR BELOW that level of signal abberation.
Even though both recordings will be exposed to the same level of contamination, and this level of contamination will 'cancel out', the fact remains that nothing at or below the level of contamination will be able to be examined via the difference method.
In other words, unless the soundcard is literally perfect, it has the capability to hide and to mask other devices sonic signatures, and I personally know of no soundcard that is sonically perfect.
Of course, if you tend to believe that such a thing exists (the sonically perfect soundcard), then you probably don't believe that very many sonic diference exist between audio components in the first place, and will be perfectly content to accept the specs as all that is needed to define a sonically perfect soundcard (or other audio component).
At this point, we have another dead end where someone has decided 'a priori' that something is sonically "good enough", and this is at the heart of the whole matter and debate.If you truly believe that a $200 soundcard is sonically perfect, then why not build yourself the ultimate system around one? People who have gone the computer based audio route have found that even the multi-thousand dollar outboard USB/firewire DACs are not providing sonic perfection, the same issues that hold true for traditional component based audio are factors with the computer based audio. Only now, they have to deal with noisy computer power supply fans, the huge amount of digital based EMI/RFI pouring out of most computers and contaminating the audio signal, and the not insignificant task of interfacing such a computer based ssytem to the rest of a sound system (AC power line grounding isues, etc.).
Remember, the recordings have to go through the ADC input as well as the DAC output, try a "loop thru" on a sound card sometime, they are not sonically transparent.
Jon Risch
...or any other audio device was sonically perfect. I was suggesting that there probably was one out there that had enough accuracy to make this test at least something one might want to investigate further. Now I also realize there are golden ears in this forum who could hear things no dog or bat could hear, and that in itself would render any test other than a subjective listening test useless. On the other hand, I would also suggest that there is an overwhelming number of people who do not have the hearing acuity of the august few. For those unfortunate folks, this test might be instructive.
> Now I also realize there are golden ears in this forum who could hear
> things no dog or bat could hear, and that in itself would render any
> test other than a subjective listening test useless.A controlled subjective listening test is a perfectly valid form of measurement. Unfortunately, I think you will find they will also refuse to use such tests as well because of past results.
They are simply asking you to accept their assertions about perceived sound differences (quite reasonable in many cases) and their assertions that it is due to not fully understood defects in various hardware devices audibly altering the sound field impinging on their ears (utter nonsense in almost every case).
We're asking two very different things:1) That you check it out yourself.
2) That you quit demanding "proof" for your eyes instead of checking it out yourself with your ears.
And in some cases, we're simply stating that we hear these differences and couldn't care less about measurement geeks cries to the contrary.
x
Andy, you just don't know what you are talking about, BUT you certainly have strong opinions about what we can or cannot hear.
We have seriously addressed ABX and double blind tests in general for about 30 years. I have measured, participated, and debated with the founders and major sponsers of ABX testing since 1979, when I put my first LTE in print to rebut Dr. Lipshitz, et al.
The problem is that ABX testing is a very poor way to detect small differences, except for level and frequency response.
Some here, like Jon Risch and Clark Johnson, are virtual experts in ABX testing, yet they don't necessarily use it to prove audio differences.
It is just too prone to Type 2 error! Even proponents of the test have admitted this, but they get shot down too, by the 'hear-no-difference' crowd.
In the end, it is easiest just to trust your ears. Of course, you can make a mistake on occasion, but so what? It is better than never hearing any differences in an ABX test, when they are apparent in open listening, and reappear when you listen a second time, or compare products.
Why do you think that people rarely do ABX testing anymore, and test boxes are not as available as in the past?
...take note of this:
> Andy, you just don't know what you are talking about, BUT you
> certainly have strong opinions about what we can or cannot hear.I suggest you reread the post you have replied to when it comes to claims about what you hear.
> The problem is that ABX testing is a very poor way to detect small
> differences, except for level and frequency response.What is the basis for this statement? Does it involve things that can be measured and independently checked?
> Some here, like Jon Risch and Clark Johnson, are virtual experts in
> ABX testing, yet they don't necessarily use it to prove audio
> differences.Experts? Clark openly admits science does not apply to audio so how can he be an expert. I do not know much about Jon other than a few wacky statements about cables which would seem to be rather a hindrance to a claim to be an expert in any area related to scientific thought.
> In the end, it is easiest just to trust your ears.
Only if you are completely ignorant of how sound perception works and do not wish your results to be accepted by mainstream audio or science.
> Of course, you can make a mistake on occasion, but so what?
Without knowing what depends on the consequences the question is not answerable.
> It is better than never hearing any differences in an ABX test, when
> they are apparent in open listening, and reappear when you listen a
> second time, or compare products.What is wrong with this? It would appear perfectly normal to me.
> Why do you think that people rarely do ABX testing anymore,
In the audiophile area? I would assume most are aware that the outcome is not going to be what they want and so quite sensibly avoid it. In other audio areas like computer audio I believe it is widely used.
> and test boxes are not as available as in the past?
Because there is no market for them. I was surprised there was ever a market in the first place.
More wackiness: "I do not know much about Jon other than a few wacky statements about cables." Any C & V there?
Andy wrote:
"Experts? Clark openly admits science does not apply to audio so how can he be an expert. I do not know much about Jon other than a few wacky statements about cables which would seem to be rather a hindrance to a claim to be an expert in any area related to scientific thought."It is easy to be uninformed, but that is still no excuse for not even bothering to look. I seem to recall one of the more famous objectivists that used to post here that insisted that other people look things up for themselves. I am not claiming to be an expert on DBT's, but I have studied things a bit, done some controlled listening tests (which are the basis for my DIY audio cable designs), and provided DIY information over a wide range of topics that is pretty much well accepted and taken to be valuble and worthwhile as DIY projects.
Well, I'll make it easy by providing the URL's and info here.
My AES paper is preprint #3178, "A User Friendly Methodology for Subjective Listening Tests", presented at the 91st AES convention, October, 1991. I can send a free copy to anyone who requests it via e-mail, or if you are stubbornly independant, you can purchase a copy at the AES website it is available for $5, and can be ordered from:
http://www.aes.org/publications/preprints/search.htmlPosts about ABX/DBT:
A discussion of selected ABX/DBT issues:
http://www.audioasylum.com/forums/prophead/messages/2190.html
http://www.audioasylum.com/forums/prophead/messages/2579.html
http://www.audioasylum.com/forums/prophead/messages/2580.html
and related:
http://www.AudioAsylum.com/forums/prophead/messages/3100.html
and
http://www.audioasylum.com/audio/general/messages/88709.htmlI have also given a paper on a new test signal I developed, a copy of this can be had for the asking, the text is at:
http://www.geocities.com/jonrisch/PhiSpectral1.htm
"A NEW CLASS OF IN-BAND MULTITONE TEST SIGNALS"
by Jon M. Risch, presented at the 105th Convention of the Audio Engineering Society, 1998 September 26-29th, San Francisco, California as preprint #4803Some of the other DIY subjects that are not audio cable related are:
DIY Acoustics projects and advice/info
See: http://www.geocities.com/jonrisch/a.htm
(and in point of fact, I will say that I _am_ an expert on acoustics, I even have a patent for a new and basic acoustic principle.
Loudspeaker with Differential Flow Vent Means, #6,549,637
http://patft.uspto.gov/netacgi/nph-Parser?Sect1=PTO1&Sect2=HITOFF&d=PALL&p=1&u=/netahtml/PTO/srchnum.htm&r=1&f=G&l=50&s1=6,549637.PN.&OS=PN/6,549637&RS=PN/6,549637DIY Isolation Transformers and Balanced Power
http://www.geocities.com/jonrisch/catch2.htmDIY AC Line Filters
http://www.geocities.com/jonrisch/surge.htm
AND
Digital Audio Component Isolation Filter
http://www.AudioAsylum.com/audio/tweaks/messages/43988.html
and
http://www.AudioAsylum.com/audio/tweaks/messages/44145.html
Steve Eddy schematic (top picture)
and
my additional "tweak" for the Digital Iso Filter:
http://www.AudioAsylum.com/audio/tweaks/messages/47160.htmlA New and Unique Speaker Crossover Topology:
http://www.geocities.com/jonrisch/LBIseries.htm
http://www.geocities.com/jonrisch/LBIseries2.htm
and so on (see bottom of pages for next page)Other patents I hold:
Record Handling Device, #4,452,480
http://patft.uspto.gov/netacgi/nph-Parser?Sect1=PTO1&Sect2=HITOFF&d=PALL&p=1&u=/netahtml/PTO/srchnum.htm&r=1&f=G&l=50&s1=4,452480.PN.&OS=PN/4,452480&RS=PN/4,452480Augmentation Amplifier #5,586,194
http://patft.uspto.gov/netacgi/nph-Parser?Sect1=PTO1&Sect2=HITOFF&d=PALL&p=1&u=/netahtml/PTO/srchnum.htm&r=1&f=G&l=50&s1=5,586194.PN.&OS=PN/5,586194&RS=PN/5,586194Patent Pending:
Method and apparatus for creating a virtual third channel in a two-channel amplifier APP #20060013406Other work I have been involved in:
Fundamental research into vinyl record playback:
http://www.audioasylum.com/audio/general/messages/444040.html
http://www.audioasylum.com/audio/general/messages/441848.htmlDesign and UL liason for an AC power line filter and surge suppressor:
http://www.audioasylum.com/audio/general/messages/133254.html
Jon Risch
Proofs of an Absolute PolarityAbsolute polarity is an interesting phenomenon (wherein) those who don't hear the effect mostly doubt the opinion of those who do. (John Roberts, AES) A newly-devised test (herein called triple-blind) once-and-for-all assesses polarity audibility variously among audio engineers, hobbyists and musicians. Results decisively affirm the sensation; many trial subjects moreover testify that Absolute Polarity's palpable reality constitutes an essential addition to the audio engineering armament.
Preprint Number: 3169
Convention: 91 (September 1991)
> It is easy to be uninformed, but that is still no excuse for not even
> bothering to look.Thanks for the CV but I think you may have lost the plot somewhat. Why should I look you up? In addressing John's statement I have not misrepresented the basis of my statement and your strong promotion of cables makes clear your relationship with the scientific method.
Not that I have followed any of your links to find out one way or the other but there is nothing in your CV that would prevent you from being an unscientific loony.
The AES is a club for the audio industry and getting papers accepted at their conferences means that the organisers considered the paper to be of interest to their members (or they were struggling to fill up the programme) it does not confer any scientific merit on the contents. To do that you need to get it accepted, reviewed and published by a reputable scientific journal. The reviewers will require adherence to the scientific method which can be a big stumbling block for enthusiastic audiophiles. Even then, subsequent work may show it to be wrong but it will generally be accepted as "scientific work".
Patents also confer no scientific merit whatsoever on what is patented. In fact it is easier to patent nonsense than it is to patent something sensible because there is more competition in the latter category. You will notice that it is quite common for the wacky audiophile products like Klaus's resonating cups to claim patents or patent applications to help support the marketing.
(nt)
z
That pretty much pegs your mind set and attitude. Now I know not to waste anytime on your replies.
Jon Risch
Andy, you are: "Cruisin' for a bursin' " ('Grease') with such slander. After being a member of the AES for more than 40 years, I might have some criticisms, but you have just slandered (or is is libeled?) an organization that was founded to help audio engineers make better audio designs. In fact, it has deviated from that path in recent decades, because it was taken over by the Ph.D. academics, and peer review has been used with a vengeance as a tool, to reduce new ideas and to remove progressive ideas from the pages of the 'JAES'.
Is it any better in any other scientific society? I really doubt it. Politics and prejudice always seems to be part of the picture, and I don't know of any scientific society with really clean hands, or a perfect record in weeding out 'faked research', 'attempts to steal vital information' or 'active suppression of new ideas'. This appears to go with the territory of trying to make a new and better understanding of physical reality; be it hard science, or a better audio product.
When Lord Kelvin ( you know, the guy who we named the degree scale after) said "X-rays are a hoax!" in 1900, he was former President of the British Royal Society. Allowing such sloppy thinking, without proof, must imply that the British Royal Society is not very scientific either.
d
There is no market for ABX boxes, because people found them opaque to listening differences. This means either that the differences didn't exist (your take on this) or that the test hides the differences (my take from experience). Your opinion on this should get you out of the discussion of audio in general, why do you bother? Or is it because you like to heckle people who do hear differences in audio products?
Where do you get off telling us what 'science' is? Are you some sort of 'expert' on the subject?
> This means either that the differences didn't exist (your take on
> this) or that the test hides the differences (my take from
> experience).Implicit in your response is an assumption that what is heard is only a property of the sound that impinges on you ears. A scientist or even someone with a bit of common sense does not make this assumption.
You are welcome to put forward the view that the ABX box modifies the signal in some unmeasureable way. This is only a view rather than a scientific hypothesis because it is untestable if one cannot measure it. I am afraid to those with some understanding of the basis of science your view will not get taken seriously because the subject clearly lies in the scientific domain.
> Your opinion on this should get you out of the discussion of audio in
> general, why do you bother?I am not following you. I have a technical interest in sound and audio, particularly the former, but this is an audiophile site and so it has only marginal relevance. My interest here is in audiophiles.
> Or is it because you like to heckle people who do hear differences
> in audio products?Heckle? I ask questions about posts and, unlike most here, answer the ones I am asked in turn.
Elsewhere in this thread I have been asking you whether you can measure the differences you hear and have explained why this is a relevant question. You have not answered.
> Where do you get off telling us what 'science' is?
What is and what is not science is relevant when thinking about how much weight to give various statements. Sound and audio lie in the scientific domain unlike, say, religion and much of music and so statements about the performance of audio based on "unscientific" information is relevant.
> Are you some sort of 'expert' on the subject?
The scientific method is taught at school and so there really is not much to be expert about.
> You are welcome to put forward the view that the ABX box modifies the signal in some unmeasureable way.>More likely this type of forced-choice test affects the receptive/cognitive abilities of the ear/brain interface to identify small audible differences other than gross loudness or frequency response.
This type of test has never been scientifically validated for this purpose.
What is the test's sensitivity for various types of audible differences?
x
...subtle audible differences - some that you may not even be aware you are hearing, as the copy says - with a clock radio?> ...wouldn't the limitations of the sound card and everything else in the chain cancel each other out?>
...this method is inherently incapable of resolving to or below the limits of the soundcard used.This is like the shunt test SM was big on to "prove" that his Citation 11 preamp is audibly "perfect". Let's pipe it through the monitor loop of an equally mediocre Marantz 3300 and see if we can tell the difference (using freebee cabling throughout). Nope. Ok, it's perfect!
...can the differences be heard? I would venture to say there are sound cards available capable of better resolution than most listeners' ears.
Given that virtually every megabuck preamp I've heard has its own signature (as determined by completely bypassing it), I'd be curious as to which sound card(s) you refer to.
I also imagine that they would be able to resolve a signal as well as some of the best preamps. Yes I know it will have a sonic signature that some will say they hear. But for the purposes of Mr. Waslo's testing procedure, the Lynx should do nicely.
disagrees with your assertion about the audibility question?
I don't see it. With regards to the sound card and whatever limitations it may have, that issue seems to be critical to those who don't see Mr. Waslo's program as workable.
OK, soundcards can and do affect the sound. There!rw
And the only reason the sound card issue was even part of the discussion was because Jon Risch raised it."Or put another way, this method is inherently incapable of resolving to or below the limits of the soundcard used."
x
What can be learned by knowing these differences? Ie how does one distinguish differences caused by distortions or noise as compared to differences caused by more of the intended information being contained in the signal? Wouldn't a perfect reference signal be necessary for meaningful comparisons?
...until major work is done in the experimental arena.
.
Hi.As I already said, if one can't hear the difference, change hobby.
An iPot or a pocket radio can do the job.If one's audio gears can't deliver the sonic difference, upgrade it.
If one refuses to admit hearing the difference, God bless.
The question is: do you like or dislike such sonic difference given sonic difference is always there.
I don't doubt the Audio DiffMaker will do its job to screen out the sonic difference. But if whatever screened out by this "tweak" be noises instead of listenable music, it won't help us music lovers.
It may do a good job for those who prefer looking at the screen rather than listening to real music. But not for a rational subjective like yours truly.
c-J
"It may do a good job for those who prefer looking at the screen rather than listening to real music"One does listen with Diffmaker. It makes an audio file of the difference (between audio recordings of gear playing real music), not something on the screen. (Hey, but aren't you reading THIS on a screen?? What are you preferring at the moment?)
Understood that you may not care about such research; many of us do, though. And after all this forum is for "Technical and scientific discussion of amps, cables and other topics" (see upper right of the Prop Head page). And we can be both technical and appreciators of real music, just as you can both post on a computer and like real music.
Hi.If such sound analysers can split hair musically instead of grahically, I will be listening.
Tell me how the DiffMaker analyses the musical difference. They can compose songs to show different levels of difference?
Pardon my being ignorant of these new beasts.
No, DiffMaker doesn't analyze anything. The brain of the listener does that part.DiffMaker only provides a difference (ok, a result of any difference, soundcards not being nearly as good, apparently, as loudspeakers) for him to listen to. It does it by simple subtraction (after some very tedious and exacting level and time matching).
No, the result isn't necessarily music, just sound to hear or not hear. But then, again, this is a Technical forum!
I haven't read any responses yet. It will be interesting to see who likes the idea and who doesn't.
Where?
____________________________________________________________
"Nature loves to hide."
---Heraclitus of Ephesus (trans. Wheelwright)
All you have to do is read my post with your head somewhere outside of your ass to know this.
LOL--------Hoist on your own petard.
____________________________________________________________
"Nature loves to hide."
---Heraclitus of Ephesus (trans. Wheelwright)
Are you really that stupid?
Hoist on your own petard.
____________________________________________________________
"Nature loves to hide."
---Heraclitus of Ephesus (trans. Wheelwright)
Where are they was a new question. Do try to keep track.
Auteur Lise Aubut
____________________________________________________________
"Nature loves to hide."
---Heraclitus of Ephesus (trans. Wheelwright)
!
Sorry the relevance was too much for you to understand.
____________________________________________________________
"Nature loves to hide."
---Heraclitus of Ephesus (trans. Wheelwright)
Let's see. We've got Arrogance, then Pretension, and now Condescension.I guess it's nice that someone out there has such a high opinion of you.
Apparently not. The point is that the search area for finding AS's alleged similar suggestions would be very large, possibly too large to be practical.
____________________________________________________________
"Nature loves to hide."
---Heraclitus of Ephesus (trans. Wheelwright)
It's your pompous use of French added to your already arrogant and tiresome game of junior philosopher to which I was responding.
Well, it doesn't do much good to talk to him in English, does it?
____________________________________________________________
"Nature loves to hide."
---Heraclitus of Ephesus (trans. Wheelwright)
!
...with all that French stuff, you've probably turned on a few inmates. :)
"...with all that French stuff, you've probably turned on a few inmates. :)"Pity. French is an official language in Canada.
I quoted two bits of an Edith Butler song, words by Lise Aubut:
"I live in the Milky Way . . ."
"Address useless over the billions of miles . .
A nice song, contrasting the human situation against the vast universe.
____________________________________________________________
"Nature loves to hide."
---Heraclitus of Ephesus (trans. Wheelwright)
it would leave them with too much time on their hands. Important things might then get accomplished......gardening, home repairs, etc.
"I always play jazz records backwards, they sound better that way"
-Thomas Edison
I would tend to agree, based on how few have actually tried to use the free software. Not that it will necessarily provide the answer.....UNLESS it found an unexplainable difference that all could agree they heard when the difference is played by itself.
I see mostly arguments here about how it couldn't work in every situation for everything possibly audible. Well, duh!
No one will EVER be able to prove anything is inaudible -- can't test everyone at all times with all systems with all rooms, takes an infinite number of tests, even if you did have a method to rule each case out.
But you only need to show something IS audible in one case to prove the opposite!
Not exactly. It would mean folks like Clark Johnsen would have to find a real job.
d.b.
(nt)
Dan is a key contributor to alcoholics. We're still anxiously awaiting his NFB article with SM.
Have you learned the math for postage due yet?
Learned to read your own schematics correctly yet, Dan? Why do you insist on making it personal? What a jerk.
We killed his business!I don't know about you, but I have never heard any of the RE Designs stuff. Consequently, I've never commented on how they sound. I've always assumed it was good. If anything, I backed him on one occasion when SM attacked the high cost of high end gear (including his own).
Since you are a dysfunctional illiterate, and since you prefer tubes without negative feedback, which will give you lots of noise and distortion, along with hefty FR deviations, it would be a total waste for you to listen to relatively well designed equipment in general.
And BTW: who cares what you have listened to or not? Your opinion doesn't matter in the grand scheme of things, it only matters to the self delusional like yourself.
You make me puke;
d.b.
Since you are a dysfunctional illiterate...Yes, that explains my six figure salary.
...since you prefer tubes without negative feedback
You really need to do your homework better. My VTL amp uses about 15 db of NFB.
rw
Fortunately for you, audio perversion is accepted in this society. Some of us think it should be treated just like any other disease. Surgical removal sounds attractive from this end.
You make me puke;
d.b.
Dan, we feel sorry for your going out of business, but this happens all the time. It is not our fault.
Or did the intellichip do it for you?
Folks like you, Mike Cruller, E_-Static, have taught me well. Since you don't know enough to respond on a technical level, the only thing available to you are insults, lies, half truths, personal attacks, etc. etc.
I am not fooled by so called professionals who write for magazines that give positive reviews to the CLC, because in reality you folks are just bent on making "high end" the home for the dysfunctional, neurotic, and the ignorant.
First you trash specifications,then you trash standards, then if you are successful in doing both of those, you can sell any garbage you want.
Congratulations; it worked. You helped take a thriving business driven to improve things and turned it upside down.
d.b.
You are such an idiot, Dan. If you would look down below you would see my arguments with Geoff Kait regarding his products. If you had read anything else I have written you would KNOW I don't support those kinds of products and then you wouldn't write this kind of rubbish because it is patently false.Dan, who is the personal attacker here? It is YOU!!
"First you trash specifications,then you trash standards, then if you are successful in doing both of those, you can sell any garbage you want."Maybe if pseudo engineers like yourself would stop blindly using virtually meaningless metrics for sound performance (like THD) and get to work to define some REAL metrics and standards then I would be whole hearted endorser of good measuring gear (because then it might actually sound good too). Get off your lazy butt and find out what really is important and how to specify it. I have no patience for meaningless metrics. I don't use them in the lab and as a result our data is well correlated with the real phenomenon.
The idea of correlating measurements to experience never even occurs to you, does it?
I just love it when people accuse me of not listening. That’s when I think all the audio meetings and presentations I went to, and all the times I went to someone else’s place to listen, or out to a retail shop to listen to whatever. I also think about all the time I spent listening to various recordings during prototype work on all the gear I designed and built. I also can reflect on all the time I spend in my Lab/office listening to music either for the sheer pleasure of it or while I am working on a project.
I can even think to my formal training in music, practicing and playing up to 12 hours a day and those ear training courses I got to take every semester. Are you aware of how much you really have to listen if you are playing music with other people? Trust me when I tell you that it gets pretty intense either when either you are reading music or doing improvisation and sometimes a bit of both. My minor was Tympani; which calls for the timpanist to change tuning on the fly. My teacher made me practice those intervals in a “quote” melody, which was without a key center and definitely atonal. I could only take an hour of that a day as it was just too morbid.I am saving this post on my computer: and I will use it the next time someone decides that since I happen to design audio equipment and take electrical engineering seriously, that my only ability is to read a meter.
Morricrab: you and your ilk, are nothing more than arrogant fakers. I am not fooled, and neither are the thousands of others who left “high end” because of dysfunctional illiterates like yourself.
So answer the question: Do you have your Clever little clock yet? The magazine you write for endorses it, in fact it even gave it an award.
d.b.
> The idea of correlating measurements to experience never even occurs
> to you, does it?But there is a catch. In order to correlate measurements and listening impressions in a valid manner (for doing science) one must perform controlled subjective listening tests or else nobody except audiophiles will take any notice of the results. As you know, in controlled listening tests almost all the differences perceived by audiophiles disappears.
How would you suggest Dan goes about correlating measurements and subjective impressions?
...in controlled listening tests almost all the differences perceived by audiophiles disappearsSuch controlled listening tests rarely, if ever, uses the highest resolution audio gear. The "best" cabling used in Roger Russell's comic book on wires is 12 gauge zip. Versus 24 gauge zip. Yawn. Tests prove what they prove on that which is tested. Nothing more. Remember that scientific thing?
Tell us the gear used in any one of the "best examples" of testing to which you refer. Oh, that's right. As a scientist, you cannot divulge the conditions on which a test is conducted in order for others to replicate the results. Such details are irrelevant to the scientific method, right?
You don't even know the difference between marketing and fact.
You don't have the qualifications, you don't have the background, and you don't have the intellect, all you have is the persistence to keep showing your ignorance.
You make me puke;
d.b.
Zero content?
my wife's poor health mainly. Now that it's done it such a relief not to have to be polite to idiots like your self. Have you begun to seek treatment yet? It's long overdue.
BTW: Do you have your Clever Little Clock yet?
d.b.
Follow moniker to answer your question.
Such a pity, it's the natural progression for all of you subjectivist dysfunctional illiterates. So tell me, what are you going to do when HP gives the CLC a glowing review? My bet is you'll dance to whatever tune he plays.
You make me puke;
d.b.
I'll be paying him a visit next week, so I'll ask him just for you. :)
The spirit of Bud Fried is alive and well.
d.b.
It is likely HP met Bud sometime in the past. I always liked his full name: I M Fried. Evidently, he was one of those "perverts" (audiophiles) as you put it. This is from the company website:"About Fried Products Corporation: Fried Products Corporation was founded by a group of audiophiles , including the late Bud Fried, in 2004 to preserve, protect and advance the art and science of loudspeaker design and production.
If anyone here needs a psych evaluation its you to get over the hatred and resentment you feel over the hifi business. Go on, Dan, lie on the couch and tell me your problems. Get it all out. Tell me about your mother. Were you abused as a child? Do you have suppressed memories that manifest themselves in various unpleasant ways?Most of the rest of us are quite well adjusted and having fun with a hobby. Sure we take it somewhat serious because most of have a fair amount of money invested and we want high performance for that money. However; very few people go running around insulting just about everyone who doesn't think like we do (only you and SM seem so crass and arrogant).
I hope you realize just how far you are now from a sophisticated, well educated engineer you claim to be. Your utter lack of civil or indeed even semi-decent behavior is shocking given your claimed level of expertise.
FWIW I have lots of clocks, one of them is pretty clever (it updates the time automatically from satillite so I guess that makes it pretty clever) but none have anything to do with audio and none ever will.
I find it funny that because we disagree on what kind of amplification works the best you think that it puts me into bed with Geoff Kait and his ilk. Wake up and pull your head out of your you know what!
I have heard literally DOZENS of amps designed like you advocate and not one sounds realistic. Not one. So something must be wrong then with those designs. I am not anti-transistor as both amps I own use them partially (hybrids). For now I am anti-feedback because experience shows me (because I listen to all kinds of amps without initial prejudice) that amps with feedback in general sound worse than those without. Also, many articles I have read suggest that feedback causes more problems than it cures...particularly from an audible distortion POV.
Just because this flies in the face of your orthodoxy is not my problem. Maybe you should really investigate the phenomenon (it is hardly an isolated incident) and see if YOU can make a no feedback design with decent measured specs (some exist you know). Then compare the sound of that design to your previous ones. Or do you lack the talent to give that a real shot? Safer to use the feedback crutch, right? Afterall the meter says its better nevermind your ears (you claim to listen but...??).
BTW, how many amps did you ship that were flawed like the schematic? Maybe feedback covered up the problem...sounded the same with it right or wrong...right? But hey it measured the same. LOL!
Thank you once again for proving that you are totally and utterly dysfunctional and illiterate I guess it pays to advertise eh?
Do you own the Clever Little Clock yet, inquiring minds want to know!
d.b.
"Do you own the Clever Little Clock yet, inquiring minds want to know!"Can't your read? I answered that question. Guess that illiteracy is spreading, oops you caught it.
Well then; according to the magazine you write for you're just not getting the most out of your audio system. Get in gear there Morricrab, time's a wasting.
BTW: Does your editor know of the sterling recommendation you gave to something he gave an award to? Are you trying to embarass your "esteemed editor"?
d.b.
You have some very strange idea about the way things work with a web magazine...seek psychological help. Do you think its a dictatorship where everyone has to agree with what the editor gives an award too?
See link: and remember this is watered down form what really happens.
d.b.
I find it funny you keep referring to Lynn Olsen, Mr. DIY SET guy. If you look at his own webpages you will see he is deep into tube gear. Yet because he gave your amp a good rating once you use him as a paragon of audio journalism or something (not that he is not but it is not the reason).Just to show how much more he is in my philosophical camp than yours (or am I in his?) here is a quote about his Amity and Karna amplifiers:
"One thing I really like about the Amity and the Karna is the extraordinary transparency and vividness of tone color; to me, they are better than anything else I’ve heard. What’s interesting is that intrinsic low distortion sounds quite different than low distortion attained through feedback techniques; low-distortion-via-feedback has a characteristic “clean” sound, akin to a well-designed transistor amplifier, but the tone colors tend to be flattened out and diminished compared to a no-feedback design.
When the devices themselves are linear it sounds quite different. As before, the sound is “clean” in the sense of absence of coloration, but there’s so much more. Tone colors are vivid; textures are right there in front of you - you hear the sound of skin on drums, the woodiness of a cello, the weight and scale of a piano, and the emotional inflections of a singer. You stop listening to hi-fi and start experiencing music. This is the realm of the direct-heated triode, the most linear device of all, but also the most tempermental.
"
I didn't realize I was part of some larger "ilk." LOLIf you don't mind my saying so, you have much more in common with brother Banquer than you might realize. Now, why don't you two knuckleheads kiss and make up, maybe form an anti Machina Dynamica Alliance. Then everyone will be happy. LOL
"maybe form an anti Machina Dynamica Alliance"I can't possibly think of a bigger waste of time. Why would I waste energy beyond posting to debate anything your products "do"? Get real.
Machina Dynamica is just the logical conclusion to subjectivist dysfunctional illiterate audio.
First you throw away specifications, then you throw away standards, then you can sell anything you want, such as snake oil.
You and your ilk asked for it Morricrab, deal with it.
d.b.
"First you throw away specifications, then you throw away standards"I never advocated throwing anything away. I advocated improving them so that they are more meaningful. Big difference.
You have not debated anything yet. Where did you ever get that idea? All I've gotten from you is the old, "I'm a PhD and I know that science can't explain these things so they can't possibly work," routine. Repeatedly.It is you you suffers from lack of reality. Maybe get out of that chemical lab and get some fresh Swiss air more often.
~ Cheers
Geoff go look at past discussions where I brought up ideas about light/matter interaction. Dang, Dan is right this illiteracy is just spreading all over. Go see a doctor I think you may not be able to read for much longer.BTW, I was in Gstaad last weekend, beautiful Swiss alps covered in snow and a lunar eclipse to boot (it made everything sound better...really if you haven't tried listening during an eclipsed moon then you can't say it doesn't help. HA!)
You didn't bring up anything of consequence; you only clained that you knew "something" about it. Big deal!And what if you were to win a "technical argument" (as unlikely as that might be) - where would that leave us? Ans: Still back at square one in terms of whehter or not the Chip works. I can see you are going to be a difficult case.
BTW I climbed Dents du Midi many many years ago, and saw solar eclipse from the summit. I win again!!
"You didn't bring up anything of consequence; you only clained that you knew "something" about it. Big deal! "Not true. I think its too late for you now the illiteracy could be irreversible.
"Still back at square one in terms of whehter or not the Chip works."
Not really, already tried and found it doesn't work so I stopped trying to find out why this "something" doesn't work. No sense chasing phantoms when there are plenty of real problems to solve.
"BTW I climbed Dents du Midi many many years ago, and saw solar eclipse from the summit. I win again!!"
Oh there was a competition about something? I guess it says everything about your personality...jerkus maximus. Yep in that category you certainly "win".
Your "arguments" are like a Swiss clock: same old movements over and over again. Name-calling and lack of substance is the essence of your game, and this has become quite boring. I always enjoy "debating" with people like you - well, up to a point. It's like shooting fish in a barrel.
Except you are the fish in that barrel.
I can never tell when you are being serious and trying to be cutting or trying to be humorous. Maybe you just can't think of original material.I know what you're thinking: "Why is this person hounding me? I am a reviewer for a major audio publication. I must know something!! LOL
You are just one long running clichè Geoff. Used cars come to mind. "aus lediderhosen" get the picture.
The feeling is mutual.There is just no getting through to some people.
You weren't civil or nice even when you were running a company and trying to sell to us "idiots". My guess is that your nastiness on this forum cost you a lot of business so that even if your product was the best thing since sliced bread there are some here who wouldn't give it a second thought. Think about it.
I got a lot of compliments from my customers on the way I stood my ground here and elswhere. You just can't buy that integrity these days.
Do you own the Clever Little Clock yet? Inquiring minds want to know?
d.b.
Like the ones who think your a schmuck would write to you and say "Gee Dan your such a schmuck I am not going to buy anything from you". Most likely they simply don't waste the time and don't bother to contact you.
Well I can say this: To the best of my knowledge I don't have any customers like you; they are far more intelligent in matters related to audio.
Have a nice day
d.b.
LOL!
"It would mean folks like Clark Johnsen would have to find a real job."You wrote that, which sounds like "the only thing available to you are insults, lies, half truths, personal attacks, etc. etc.," which you wrote also. Your statement started it, Dan, so what do you really have to complain about?
(nt)
nt
You teased us way back when and have not delivered the goods. :)
What's the point? there are so few of you audio perverts left that it really doesn't matter. You have isolated yourselves rather well as far as I can tell. Both Pro and HT will have nothing to do with perverts like you, and the perverted philosophy that goes with it. So when are you going to get treated for your condition? Will a frontal lobotomy do the trick? Heck; we can even arrange to have wired up with some of your favorite cable.
You make me puke;
d.b.
Should I even bother? What's the point?Wouldn't that serve to further your crusade to marginalize the high end? I would have thought that you and SM would delight in such. Such a pairing sounds like a match made in heaven. Demonstrate your superiority over the "perverts".
If you recall, it was your idea. :)
The crusade you need is a mental health one. I'll bet you could spend less on a qualified mental health professional for counseling, than you have spent on cables.
d.b.
I don't have to make this up!
...if you find one, then next you can look for the answer to the conservative/liberal debate.And then to the Jewish/Palestinian one.
...it will always tell you there's a difference!BFD!
It won't let you assess the difference between your vinyl playing string quartet music or an MP3 of Britney Spears, either! Must not be worth anything....Is there anyone, anywhere, who thinks there's no difference between vinyl and digital? Has that ever been doubted or contested? Why would anyone even conceive such a test?
The test isn't going to tell you better or worse, or what you'll like better. Just whether there might be anything changed to BE better or worse, which is the question for things like cables and CD treatments. And if so, to maybe show what that difference sounds like all by itself.
Odd that people are shooting the test down before even seeing any results from it, kind of early to get defensive, no? What if the test proves a difference is happening with these things?
dc
How many times has the subjectivist crowd accused the objectivists of the same thing?
It's a nasty habit any way you look at it.
There is an example recording, of capacitors making a difference, that can be downloaded from the site where DiffMaker is.
Will a second generation recording of a system in use be good enough to resolve the subtle differences involved? So you bring your $200k worth of recording equipment to your listening room and make a baseline recording. Then change something and make a new recording. Then perform the analysis between the two signals.Sounds great on paper, but impractical and fraught with unvalidated assumptions.
You are saying that an audible difference in signals might yield NO difference between recordings of that signal and the unchanged signal?In other words, that whatever has changed will be *totally invisible* to the less than $200K recording process and will cause no difference at ALL between the tracks made with it (kind of like the way vampires don't cast reflections in mirrors)?? Are there signal characteristics that get removed (not just get distorted or corrupted or mixed with added noise -- but vanish *completely*) in passing through gear that doesn't cost enough $$?
I can think of two cases where that could possibly be true:
...Signals outside the bandwidth of the lower$$ gear-- but soundcards that record at 192kHz have bandwidth as high as any used in studios, and can be had for under $150. So that can be easily avoided.
...Or signals so low in level that they are below the noise floor of the sub$$ gear -- but do all the audio benefits of tweaks and cables only occur at -105dB below the peak playback levels?
..Maybe there are other classes of stealthy sounds in one recording that can leave no trail in another recording, who knows for certain? Pretty darn unlikely, though, I'd say.The "not good enough equipment" argument maybe holds when talking about degrading a signal, garbage combining with other garbage sounding like the same garbage. But the test mentioned detects whether any change has happened at all.
Are you going to multi-mike the second generation recording in a user's listening room? Use a minimal mike approach? Will that be the same approach as was taken with the first generation of the recording? Is that going to retain all the original recording's perspective? And S/N ratio? If I understand the approach correctly, we are re-recording a recording in another room. Twice.
EStat,"Is that going to retain all the original recording's perspective?"
Sorry, I didn't follow that the first time I read it, and need to respond.
I need to get it clear that IT DOESN'T MATTER if the perspective is the same, or if the recorder's response isn't dead flat or if the S/N is degraded some, etc.. Diffmaker is looking for differences, only. As long as the recording situation is the same for both recordings. Things like bad S/N will make a difference in the two recordings (noise isn't the same every time and doesn't subtract out). All non-silly errors I can think of that are makable by the program will be in the direction of showing a (possibly nonreal) difference, rather than showing "no difference" (that takes an *awful* lot of things going right to achieve).
From the other post:I guess I was misled by the block diagram whereby you first "make a Reference recording". What what you've said, you capture line or speaker signals which are not really "recordings" in the usual sense.
I now notice that this is your product, so naturally you have devoted far more time to understand its principles! I genuinely wish you good luck in getting real audio designers (as opposed to the hobbyists here) to use the tool.
I suspect you haven't read the documentation that comes with the program. No offense, but there's a lot written there about that. Here's a summary:Recording with mics in a normal listening room isn't likely to be usable, the thing is insanely sensitive to even teeny differences. Things like the mic or speaker moved several mils, very exact positions of reflective sources including the person doing the recording in the room, even in other rooms that open into the measurement room, maybe air currents and temperature changes). I've not obtained a good silent difference track with acoustical signals when testing with no changes at all between recordings (a good verification test for setups).
For similar reasons, I doubt it would work with turntable sources either (though I haven't tried) because of changes in the vinyl after the first playing. Maybe if it were done on different days after the plastic has relaxed again it might work, if the turntable speed was consistent enough.
But electrical signals can be recorded from speaker terminals or at line levels between any components in the chain. As long as what is being changed comes before the place where you record from, you should be able to pick up differences in the audio electrical signal.
So devices to test might include cables, CD treatments, contact enhancers, AC Power enhancers, vibration control devices, etc. (unless they are having their effect on the listener by some way other than through the electrical audio signal).
So things like speaker drivers (which are generally agreed to make a difference anyway), room resonators, other stuff that doesn't act on the electrical signal probably can't be tested with Diffmaker, at least not without special isolated conditions and extreme care. It doesn't do everything, but neither does any other test. Use where appropriate.
I'm really not out to prove that things don't make a difference (that couldn't be proven anyway unless every possible situation was evaluated). But the reverse is provable, and by only ONE universally repeatable example! I keep seeing finger-wagging audiophile posts here and elsewhere about how engineers aren't measuring the right thing. So here's an attempt to do that. I can't do scientific investigation and find those measurements without actual evidence (as opposed to testimonials), so the idea was to get evidence where it may be gettable, and also to maybe give me and others a feel for the degree of difference being made (or not made).
I'm not trying to take away anyone's toys, OK?
It will take only one repeatable example of an audible effect (that isn't explained by current theory) to allow actual research on it to start (and fame for the lucky engineer who gets first crack at it, too). Help me out here, if you want to (not just EStat, anyone else, too). If you really want someone to "measure the right things", that is.
...you then have my blessing. Mess with my toys however... ;-)Seriously though, sounds like you've created a valuable tool.
x
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