SACD vs. CD - Unfair competition?

[WmAx]

That's YOUR definition of PROOF?! How laughable. You ought to have your naysayer's membership revoked for all the logical holes that you left open in that statement.

So, somebody put together an invalid test. Big deal, you disregard the findings and start over. How does that PROVE your case under more valid and equitable conditions? I thought so, just conjuring up more inneuendo to avoid having to do the work yourself. BTW, jumping the gun in a chess game does not make you a winner.

What? No one has yet demonstrated hi-res to be a cause in the first place. Cart before the horse syndrome? Soiunds like you've already accepted hi-res as being a real, plausible reason for difference when it's very low on the suspect list considering existing research/studies on bandwidth audibility. All mtrycraft did was tell you of a famous incident showing the desperation they have resorted to in order to falsify positive results since they have not been achieved elsewhere in a controlled/scrutinized test scenario.

-Chris

[hifitommy]

you are blathering. again.

[Sir Terrence the Terrible]

You have not been following the high-res vs CD debate much I can see. There are numerous of claims of the audibility of the high-res vs CD based on comparisons between redbook CD and SACD (e.g. threads at RAHE). These speculations have also been figuring in this thread, if you are not totally blind.

Since I don't visit or participate in RAHE, how in the heck am supposed to know what the contents of their threads are? If we are just comparing format vs format(and not taking into consideration anything else) I can see where a comparison can be made. Otherwise you cannot compare bit level, sample rate, they don't even use the same data carrier(LPCM vs DSD)

You claimed that it was solely mixed as a 5.1 format above. It's not. There is a stereo track from which I can use the analog output from a SACD player (or DVD-A) and record that signal in either 24/96 or 16/44.1 PCM and compare audibility between the tracks.

The stereo track from DSD is a downmixed version of the 5.1 track. So let me get this straight, you say you can convert a DSD stream into LPCM at 24/96 or 16/44.1khz and compare the audiblilty between tracks. Now I KNOW you are a phony. Converting a DSD stream to LPCM will completely negate any comparison that can be made. Common sense would dictate that when you convert one type of signal to another, the losses in the conversion would tilt the results in favor of the native signal. Worse, you would be passing that signal through the A/D and D/A of the SACD, redigitizing it into LPCM and then back through the D/A stages. Then you are comparing this to a signal that was recorded and played back in it's native form. Oooooo, very fair comparison(sarcasm off)

Now you say you write for the Swedish audio society? This rag must suffer from malnutrition in the accuracy area!!!!

I can record from live sources provided I can get a microphone that stretch up to at least 30 kHz or more or use a high-quality master. I can thus repeat those tests that have been made with previous negative results. You are saying I will be suprised. I will bring the report here.

The issue is the audibility of the medium. High-res vs. redbook CD. Nothing else.

T

The way you are proposing to do these tests will not reveal much of anything. You are converting signals from one format to another, and trying to compare them to unconverted signals. That is not the most pristine way of doing a test, and will tilt the results in favor of the unconverted format. The only way to conduct this test and get a fair result is to take your master, encode it into a DSD stream, and at 16/44.1khz. Using your mixing board, or a ABX comparorator that is level matched with full latency, blindly compare each track with the master. The one that comes closest to sounding like the master is the most transparent.

A sided by side comparison with no master in between will only tell someone that either they sound different from each other, or the same. But it won't say which is transparent, and closest to sounding like the master.

If this test is any indication to how previous have been run, I would highly suspect the results attained. Maybe you should allow someone else to design the testing methods, because it is apparent that you lack of experience in this area negatively betrays you.

[Sir Terrence the Terrible]

What? No one has yet demonstrated hi-res to be a cause in the first place. Cart before the horse syndrome? Soiunds like you've already accepted hi-res as being a real, plausible reason for difference when it's very low on the suspect list considering existing research/studies on bandwidth audibility. All mtrycraft did was tell you of a famous incident showing the desperation they have resorted to in order to falsify positive results since they have not been achieved elsewhere in a controlled/scrutinized test scenario.

-Chris

Here we go again, back to the issue of bandwidth when that was taken off the table several hundred replies ago. Chris you are sounding like a broken record because you really don't have a leg to stand on. Michael Bishop effectively cut them both off. Neither you, Mtry, or Thomas has participated in these test, so you have no way of TRUELY verifying anything that has to do with them. You don't know if they were desperate(inflammatory language shows a lack of effective communitcation skills) or just overlook a small variable. You are doing the exactly same thing as they are by picking and choosing what is legit and what is not based on your beliefs. The key to this is to shut your fat trap, and listen to the audio, and not sit around talking about it in a passive, indirect way. I know this is difficult for you to do, but give it a try, you may actually learn something in the process

The bottom line is that no test has effectively ruled that high frequency information is, or is not peceived in high resolution audio. That just means that some were effected, and other were not. I would say the maximum high frequency information that one can hear effects what can be perceived in the ultra sonic range. The test that AES ran tested the hearing of the subjects that participated. The ones that could hear the highest frequencies were the ones that tested with a positive result, and the ones that had some hearing damage(high frequency losses) produced the null result.

[Thomas_A]

Since I don't visit or participate in RAHE, how in the heck am supposed to know what the contents of their threads are? If we are just comparing format vs format(and not taking into consideration anything else) I can see where a comparison can be made. Otherwise you cannot compare bit level, sample rate, they don't even use the same data carrier(LPCM vs DSD)



The stereo track from DSD is a downmixed version of the 5.1 track. So let me get this straight, you say you can convert a DSD stream into LPCM at 24/96 or 16/44.1khz and compare the audiblilty between tracks. Now I KNOW you are a phony. Converting a DSD stream to LPCM will completely negate any comparison that can be made. Common sense would dictate that when you convert one type of signal to another, the losses in the conversion would tilt the results in favor of the native signal. Worse, you would be passing that signal through the A/D and D/A of the SACD, redigitizing it into LPCM and then back through the D/A stages. Then you are comparing this to a signal that was recorded and played back in it's native form. Oooooo, very fair comparison(sarcasm off)

Now you say you write for the Swedish audio society? This rag must suffer from malnutrition in the accuracy area!!!!




The way you are proposing to do these tests will not reveal much of anything. You are converting signals from one format to another, and trying to compare them to unconverted signals. That is not the most pristine way of doing a test, and will tilt the results in favor of the unconverted format. The only way to conduct this test and get a fair result is to take your master, encode it into a DSD stream, and at 16/44.1khz. Using your mixing board, or a ABX comparorator that is level matched with full latency, blindly compare each track with the master. The one that comes closest to sounding like the master is the most transparent.

A sided by side comparison with no master in between will only tell someone that either they sound different from each other, or the same. But it won't say which is transparent, and closest to sounding like the master.

If this test is any indication to how previous have been run, I would highly suspect the results attained. Maybe you should allow someone else to design the testing methods, because it is apparent that you lack of experience in this area negatively betrays you.

Oh. I have explained one of the previous tests already. You can read, or perhaps not? I am starting to wonder, since you have not understood anything what I've said at all. I want am analogue source signal which has frequency content >22 kHz, thus that can be the analogue signal from SACD, DVD-A, a master tape which has a frequency content >22 kHz (DAT high-speed), or directly fed by microphones that can reproduce at least up to 30 kHz. I record that in 24/96 and use that as the original signal. From this I can down-sample to e.g. 16/48 and test the audibility of that against the original. I can also record the same SACD or DVD-A signal in 16/44.1, not to loose the sync by downconverting 24/96-16/44.1.

Just like I said, I am comparing the audibility of high-res vs redbook CD only. Nothing else.

[WmAx]

Here we go again, back to the issue of bandwidth when that was taken off the table several hundred replies ago. Chris you are sounding like a broken record because you really don't have a leg to stand on. Michael Bishop effectively cut them both off. Neither you, Mtry, or Thomas has participated in these test, so you have no way of TRUELY verifying anything that has to do with them. You are doing the exactly same thing as they are by picking and choosing what is legit and what is not based on your beliefs. The key to this is to shut your fat trap, and listen to the audio, and not sit around talking about it in a passive, indirect way. I know this is difficult for you to do, but give it a try, you may actually learn something in the process

Refer to my last reply addressed to you. Nothing has changed.

-Chris

[WmAx]

Oh. I have explained one of the previous tests already. You can read, or perhaps not? I am starting to wonder, since you have not understood anything what I've said at all.

I would tend to agree with your suspicion(s) so far.

-Chris

[Sir Terrence the Terrible]

I would tend to agree with your suspicion(s) so far.

-Chris

You would. I think you talk to much, and don't listen enough. So, everyone has the negative personal opinions about each, but how does that advance your arguement(or lack of one)?

[WmAx]

You would. I think you talk to much, and don't listen enough. So, everyone has the negative personal opinions about each, but how does that advance your arguement(or lack of one)?

That was my opinion. I felt like chiming 'in' per say.

Re: your other assertions: It does not advance my arguments. I already submitted my arguments. This assertion I made in reply to Thomas was not meant as as addendum to past arguments I have had with you. You irrationally responded to the past arguments --- I let it be known that I have no reason to discuss these issues with you further -- specifically because you seem to lack comprehension. SO far this still stands.

-Chris

[Woochifer]

Proof? No, that is how you imagine this evidence as. It is evidence you cannot dismiss. No, you can if you want, no one is stopping you.

Evidence, proof, whatever you offered up constitutes neither. Try coming up with another word if you're up to it. Nice of you to evolve the discussion into yet another word twisting exercise. In case you forgot or intentionally decided to overlook, you're the one who's making the assertion that high res should be eliminated as a causal variable. All I've stated all along is that I lack the information and access to source material to draw my own conclusion about the causal effects. I'll leave you to obsess all you want over the causal effects, since I sure don't.

Since you're making the assertion, show me the evidence, proof, DBTs, peer reviewed findings, or whatever else you typically demand of others who make assertions on this board. If this is all the "evidence" that you can offer when actually making an assertion, maybe it's better that you just revert back to inneuendo, condescention, and mudslinging.

Invalid test for you that is. That is your assertion.

Invalid test for me? No, you were the one who cited the Sony test as faulty, and therefore supportive of your conclusion. Tell me again, how that "evidence" confirms that the resolution is not a potential causal effect? I thought so, just more spin for its own sake.

I suppose you can point to some evidence of any kind? Oh, your listening comparison and speculations? Yes, that is a given. Any other? How telling.

I've stated all along that I don't have the answer, so you are asking me for evidence to that effect? You crack me up, thanx for the good laugh.

And you're only half right on the last point. I do the listening, you do the speculating. When you come up with the evidence and proof that you claim is so easy to verify, let us know.

[Woochifer]

What? No one has yet demonstrated hi-res to be a cause in the first place. Cart before the horse syndrome? Soiunds like you've already accepted hi-res as being a real, plausible reason for difference when it's very low on the suspect list considering existing research/studies on bandwidth audibility. All mtrycraft did was tell you of a famous incident showing the desperation they have resorted to in order to falsify positive results since they have not been achieved elsewhere in a controlled/scrutinized test scenario.

-Chris

Go ahead and perpetuate the spin all you want. All that I've said is that I don't have the information and access to source material to come up with my own assessment over the relative importance of the various causal effects. If the resolution is one of the known variables when I do a disc comparison (which I'm stuck with since I don't have access to master sources), then who am I to eliminate it as a causal factor? Mtry tried to cite a faulty test that Sony did as "evidence" that resolution should be eliminated as a causal effect. To me, a faulty test proves nothing one way or another. You can draw all the inneuendo and conspiracy conclusions that you want, but that hardly qualifies as evidence or proof of anything.

[Sir Terrence the Terrible]

Oh. I have explained one of the previous tests already. You can read, or perhaps not? I am starting to wonder, since you have not understood anything what I've said at all.

If you knew what you were talking about, it might making it easier to understand what you write. I am begining to wonder if you passed a basic english composition course.

I want am analogue source signal which has frequency content >22 kHz, thus that can be the analogue signal from SACD, DVD-A, a master tape which has a frequency content >22 kHz (DAT high-speed), or directly fed by microphones that can reproduce at least up to 30 kHz.

Why would you use a analog signal from a SACD or DVD-A? That signal is not pristine as it has passed through the A/D and D/A circuits. I do not know of any DAT player that can record signals up to 30khz(typical sampling rates are 32. 44.1 and 48khz which limits their frequency response to 24khz). Will this signal be from a pre-recorded SACD or DVD-A? That would be stupid because it probably has been through post production processing. Why would you choose a microphone with a high frequency limit of 30khz? That would make the source the limiting factor and not the formats themselves. SACD is -3 at 50khz, and 24/96khz is down about 5 db at 48khz. The microphone should be flat to 50khz so the source is not the limiting factor. Will this source be close miked? It sure better or you will lose any ultra high frequency information that would be present to the air.

I record that in 24/96 and use that as the original signal. From this I can down-sample to e.g. 16/48 and test the audibility of that against the original. I can also record the same SACD or DVD-A signal in 16/44.1, not to loose the sync by downconverting 24/96-16/44.1.

Why in the hell would you record in 24/96khz if the comparison is 16/44.1khz? Why would you downsample to 16/48khz if this test is about SACD vs CD. Secondly, going from 24/96khz to SACD is a stupid, unnecessary conversion. In order to be fair, you need a DSD stream encoded directly from the microphone, no conversion at all. Otherwise you compromise the test. In order for the test to be valid, all formats must be in their native format.
Thirdly, SACD cannot be recorded into 16/44.1khz without conversion. It runs at 1 bit 2.822mhz and is not LPCM. You also cannot record a DVD-A signal at 16/44.1khz because the format runs off of a 48khz platform per specfication set by the DVD audio group
You cannot record a SACD signal to 16/44.1khz without conversion from DSD to LPCM. Not pristine. You also cannot cleanly or pristinely downsample from DVD-A to 16/44.1khz because the decimation process from 48khz to 44.1khz introduces some distortion and degredation.

Just like I said, I am comparing the audibility of high-res vs redbook CD only. Nothing else.

If the comparison is high res vs redbook you are not going to cleanly get there with what you propose. If you are simply camparing high rez to redbook then the test would be as simple as a live feed to a recorder at 16/176.4khz, and the same feed to a recorder at 16/44.1khz, level matched. . That is the only way you do not introduce distortions into the mix. I however thought we were talking about SACD vs CD since this is the topic of the thread(damn, you are like a roach with the lights going on, scrambling all over the place)

Your main problem is that you don't have a very good understanding of recording practices, sample rates, conversion, or studio technology at all. You are proposing to record with sample rates that will introduce distortion during downsampling. You are taking DSD signals and converting them to LPCM signals. You want to use equipment that actually introduces weakness to the process, and you never make mention to the reproduction devices at all(speakers or headphones????)

When a person introduces a testing methods the way you have, one can only come form a few conclusions. 1. you don't know what you are talking about. 2. You do not understand the technology, OR 3. a combination of both 1 and 2 keeps your from designing a non compromising testing procedure.

Please go back to the drawing board and try this again. There are enough holes in this testing procedure to double as a water sprinkler.

[Sir Terrence the Terrible]

That was my opinion. I felt like chiming 'in' per say.

Can you chime on topic??

Re: your other assertions: It does not advance my arguments. I already submitted my arguments.

Those were arguements. Wow, as much profoundness as a puddle of spit

This assertion I made in reply to Thomas was not meant as as addendum to past arguments I have had with you. You irrationally responded to the past arguments --- I let it be known that I have no reason to discuss these issues with you further -- specifically because you seem to lack comprehension. SO far this still stands.

-Chris

Chris, the way Michael cut you down to size shows that you lack comprehension. You looked like a feeble old woman trying to cross 5th Ave during rush hour in your responses to him. Since you don't know me well enough to assume what contexted my resposes were to, your assumptions a far off base. You also don't know me well enough to evaluate the rational, or irrational nature of them. Stick to what you know, pretending to know much about nothing. You are really good at that.

Its rather amusing to me that you clam up when you do not have an effective argument to present. That might be a good thing because it keeps you from look so stupid as poor Thomas does.

[Thomas_A]

If you knew what you were talking about, it might making it easier to understand what you write. I am begining to wonder if you passed a basic english composition course.

Keep wondering.

Why would you use a analog signal from a SACD or DVD-A? That signal is not pristine as it has passed through the A/D and D/A circuits.

Does not matter. A signal is a signal, whether it is from a signal generator, LP playback, CD, DVD-A, SACD or live source. If I capture this signal and if it changes audibly from the original after any processing event, then there is a positive result in a DBT. If not, it's a negative. The purpose of the test is to investigate whether a wider bandwith, higher sampling rate, or more bits are audibly different from standard 16/44.1.

I do not know of any DAT player that can record signals up to 30khz(typical sampling rates are 32. 44.1 and 48khz which limits their frequency response to 24khz).

http://perso.club-internet.fr/farzeno/piopor.htm

Will this signal be from a pre-recorded SACD or DVD-A?

Most probably not.

That would be stupid because it probably has been through post production processing. Why would you choose a microphone with a high frequency limit of 30khz?

There are mics which are linear up to 50 kHz. The at least 30 kHz limit was said because you previously claimed we cannot hear above 18 kHz. Thus, it should not matter for audibilty if there is a roll-off at 30 kHz. Or? Changed your mind?

That would make the source the limiting factor and not the formats themselves. SACD is -3 at 50khz, and 24/96khz is down about 5 db at 48khz.

Correct. If we can hear above 22,050 Hz as you stated we cannot.

The microphone should be flat to 50khz so the source is not the limiting factor. Will this source be close miked? It sure better or you will lose any ultra high frequency information that would be present to the air.

Yes, so I've stated previously. Below you see a link with 24/96 example of rattling keys with ultra-high frequency contents.

Why in the hell would you record in 24/96khz if the comparison is 16/44.1khz?

Why not, if I want to test the audibility between 24/96 and 16/44.1?

Why would you downsample to 16/48khz if this test is about SACD vs CD.

It's CD vs high-res. The test is whether a signal with higher sampling rate, a wider bandwidth or more bits is audibly different/better from redbook CD.

Secondly, going from 24/96khz to SACD is a stupid, unnecessary conversion.

Agreed. SACD is stupid, therefore I will not convert 24/96 to SACD.

In order to be fair, you need a DSD stream encoded directly from the microphone, no conversion at all. Otherwise you compromise the test.

No. You have not understood the test.

In order for the test to be valid, all formats must be in their native format.

I am not testing all formats.

Thirdly, SACD cannot be recorded into 16/44.1khz without conversion. It runs at 1 bit 2.822mhz and is not LPCM. You also cannot record a DVD-A signal at 16/44.1khz because the format runs off of a 48khz platform per specfication set by the DVD audio group.

I will not do any conversion.

You cannot record a SACD signal to 16/44.1khz without conversion from DSD to LPCM. Not pristine.

Se above.

You also cannot cleanly or pristinely downsample from DVD-A to 16/44.1khz because the decimation process from 48khz to 44.1khz introduces some distortion and degredation.

Agreed. That's why I record from the source signal directly to 24/96 and to 16/44.1.

If the comparison is high res vs redbook you are not going to cleanly get there with what you propose. If you are simply camparing high rez to redbook then the test would be as simple as a live feed to a recorder at 16/176.4khz, and the same feed to a recorder at 16/44.1khz, level matched. . That is the only way you do not introduce distortions into the mix. I however thought we were talking about SACD vs CD since this is the topic of the thread(damn, you are like a roach with the lights going on, scrambling all over the place)

See my first post in this thread.

Your main problem is that you don't have a very good understanding of recording practices, sample rates, conversion, or studio technology at all. You are proposing to record with sample rates that will introduce distortion during downsampling.

Se above.

You are taking DSD signals and converting them to LPCM signals. You want to use equipment that actually introduces weakness to the process, and you never make mention to the reproduction devices at all(speakers or headphones????)

See spec of the speakers earlier in the thread. That is what I can use. Headphones can probably be any of the STAX models.

When a person introduces a testing methods the way you have, one can only come form a few conclusions. 1. you don't know what you are talking about. 2. You do not understand the technology, OR 3. a combination of both 1 and 2 keeps your from designing a non compromising testing procedure.

Well, you do not qualify as a scientific reviewer so I don't care what you think.

Please go back to the drawing board and try this again. There are enough holes in this testing procedure to double as a water sprinkler.

Here is some excercise for you:

http://64.41.69.21/technical/sample_rates/index.htm

Good luck.


T

[Sir Terrence the Terrible]

Keep wondering.

Don't have to, it's readily apparent

Does not matter. A signal is a signal, whether it is from a signal generator, LP playback, CD, DVD-A, SACD or live source. If I capture this signal and if it changes audibly from the original after any processing event, then there is a positive result in a DBT. If not, it's a negative. The purpose of the test is to investigate whether a wider bandwith, higher sampling rate, or more bits are audibly different from standard 16/44.1.

It does matter. That is why I think you are a fake, you are sloppy and overly complicated which contaminates results.




http://perso.club-internet.fr/farzeno/piopor.htm

You still have A/D and D/A conversion to worry about so using a DAT is stupid, unclean, and unwise.

Most probably not.[/qoute]

Incomplete answer which shows you had every intent on doing it, but changed your mind when called on it.

There are mics which are linear up to 50 kHz. The at least 30 kHz limit was said because you previously claimed we cannot hear above 18 kHz. Thus, it should not matter for audibilty if there is a roll-off at 30 kHz. Or? Changed your mind?

It doesn't matter if we cannot hear above 18khz or not, if the object is to compare high rez to redbook CD, then you don't won't your microphone to roll off the output before the format does. More sloppiness and lack of forethought. Signs of profound inexperience!

Correct. If we can hear above 22,050 Hz as you stated we cannot.

Not interested. I am interested in what YOU do, not what someone else did. You cannot continue to use someone elses work, I am interested in what YOU would do. So far I am left unconvinced that you know what you are talking about

Yes, so I've stated previously. Below you see a link with 24/96 example of rattling keys with ultra-high frequency contents.

Once again, stop pointing to someone elses work. If you are so knowledgeable, I am sure you can think things up for yourself.

Why not, if I want to test the audibility between 24/96 and 16/44.1?

In case you have forgotten, the title of this thread is SACD vs CD unfair comparison. That means we are not talking DVD-A vs CD. You are making me dizzy with all of this spin action you are trying to put on this topic. You are busted, but your over blown ego won't let you admit that you don't know what you are talking about( a problem you and Chris share). Stevie Wonder can see that

It's CD vs high-res. The test is whether a signal with higher sampling rate, a wider bandwidth or more bits is audibly different/better from redbook CD.

Go back to the title of the thread. You cannot quote the noise levels of one format, then change to a completely different one. That is called spin, and this is a no spin zone. Try again buddy

Agreed. SACD is stupid, therefore I will not convert 24/96 to SACD.

You are not qualified to make this statement. What format did you bring to the consumer? None, and I am sure the engineers that created the SACD know alot more than you about digital audio.

No. You have not understood the test.

That is because your testing method is overly complicated, full of audio degrading and unnecessary conversions, and downsampling. That is not a test that would pass mustard at AES

I am not testing all formats.

You are not testing any formats with what you propose, you are testing the quality of the conversion and downsampling process. Hardly what I would call a SACD vs CD comparison which is the crust of these threads

not do any conversion.

That is not what is outlined in your previous post. More spin I see.

agreed that's why I record from the source signal directly to 24/96 and to 16/44.1.

Once again you are off topic. This is SACD vs CD not DVD-A vs CD. Damn I am getting dizzy. Spin, spin spin!!!

See spec of the speakers earlier in the thread. That is what I can use. Headphones can probably be any of the STAX models.

Doesn't matter what you use based on your testing standards, it would not get the result that is desired.

Well, you do not qualify as a scientific reviewer so I don't care what you think.

And you are no audio journalist either, so this is irrelevant. I have probably done more DBT that you have anyway, so that would make me just a little more qualified than yourself.

Here is some excercise for you:

http://64.41.69.21/technical/sample_rates/index.htm

Good luck.


T

Your are not qualified in any way to pass out homework assignments. This has nothing to do with the subject at hand. You should change your name to spinmiester.

There is nothing here that is relevant to the topic at hand. Stay on topic, and choose a testing method based on sound recording practices and comparison. Do not point to work that you haven't done yourself. Anyone can do that.

[mtrycraft]

Here is some excercise for you:

http://64.41.69.21/technical/sample_rates/index.htm

Good luck.


T

You don't play fair Facts always get in the way.

Tilt. Overload

[Sir Terrence the Terrible]

Here is some excercise for you:

http://64.41.69.21/technical/sample_rates/index.htm

Good luck.


T

You don't play fair Facts always get in the way.

Tilt. Overload

Were are the facts? How do I know from what source these signal come from, the quality of the signal, processed or unprocessed. I see no facts here. Try again old inexperienced one.

[WmAx]

Its rather amusing to me that you clam up when you do not have an effective argument to present. That might be a good thing because it keeps you from look so stupid as poor Thomas does

It's pointless to argue with someone(you) who does not demonstrate the ability to comprehend the difference between speculation and substantiation and the relative value(s).

-Chris

[hifitommy]

"It's pointless to argue with someone(you) who does not demonstrate the ability to comprehend the difference between speculation and substantiation and the relative
value(s)."

so why do you insist on demonstrating that disability?

[Thomas_A]

Were are the facts? How do I know from what source these signal come from, the quality of the signal, processed or unprocessed. I see no facts here. Try again old inexperienced one.

In case you can't read:

"24 bit 96 KHz "reference" samples were made by using 2 B&K 4007 1/2" condenser microphones powered by an Audio Technica phantom power unit, preamplified using a Benchmark Media mic preamp, and recorded using a CardD Deluxe in a 800 Mhz Pentium 3 computer located in another room. They were closely miced on-axis in a fairly small dead space. Therefore the transients are very well-defined and harmonic-rich, technically speaking. They also have relatively low amounts of background noise (mostly acoustic). They may sound quite "dry" to your ears.


Each test file was prepared by downsampling the reference file to the indicated sample rate, and then upsampled to the indicated sample rate of either 16/44 or 24/96."

[Thomas_A]

Your are not qualified in any way to pass out homework assignments. This has nothing to do with the subject at hand. You should change your name to spinmiester.

There is nothing here that is relevant to the topic at hand. Stay on topic, and choose a testing method based on sound recording practices and comparison. Do not point to work that you haven't done yourself. Anyone can do that.

I cite from the first post in this thread:

"Let's consider the following points:

(1) I can not find a scientific research project demonstrating audibly benefits to humans of a wider bandwidth then CD offers.

(2) I can not find definitive research of SACD vs CD releases, to find alternative explanations.

(3) I can not find reason for larger then 16 bit wordlength for audio playback, especially when properly dithered, which can effectively remove the quantitazation noise and allow the theoretical limit of CD of 96dB to be approached and/or met."


If VmAx think I drifted off topic, I am sure he would say so. Besides, the "work" that is done by Arny Krueger on the webpage I referred to, I am now sure you have not looked at the page. These are free listening tests which anyone can perform.

Your lack of scientific thinking and skills is obvious. I choose to record a signal in 24/96 and in 16/44.1. If nobody hear a difference between 24/96 and 16/44.1 in a DBT, then all possible differences between the two samples, including the most obvious - higher sampling rate, are inaudible. Simple. One of the questions posed in the initial post has already been solved. It's an unfair comparison since the CD and SACD are mixed differently. The remaining issue is whether high-res provides anything more that is audible compared to redbook CD.

BTW,

the K622 recording may be interesting. The interview also have some points about DSD:

http://www.stereophile.com/musicrecordings/804k622/

T

[Thomas_A]

It does matter. That is why I think you are a fake, you are sloppy and overly complicated which contaminates results.

You are wrong. The signal itself should contain the matter to be tested, but that's implicated in the test question itself. If it is the importance of wide bandwidth that should be examined, it should contain high bandwidth. The quality of the signal is not important. Most often, test signals such as dirac's, pink noise, and short transients are most revealing.

You still have A/D and D/A conversion to worry about so using a DAT is stupid, unclean, and unwise.

You cannot understand a "wire bypass test" or "before/after" test, that's very clear I can see.

Incomplete answer which shows you had every intent on doing it, but changed your mind when called on it.

Incomplete answer? Where are your answers to my questions? DBT? ABX? Results? Statistics? Dream on.

It doesn't matter if we cannot hear above 18khz or not, if the object is to compare high rez to redbook CD, then you don't won't your microphone to roll off the output before the format does. More sloppiness and lack of forethought. Signs of profound inexperience!

My previous quote:

"The test included a high-quality analog musik signal with a bandwith of 46 kHz recorded with high-speed DAC (fs=96 kHz) with two Earthworks microphones. This was the original signal. Signal 2 was the same but filtered analog with 3rd order butterworth at 25 kHz. Signal three, the original was converted to normal DAT standard (48 kHz).

Two different tweeters were used, both which were ±0.5 dB up to 20 kHz ±30°. The first falls soft above 26-28 kHz, the second continue 20-80 kHz ±9dB. If the peak at 28 kHz is exlcuded it was within ±4 dB.

There were no signifcant audible differences between the three program material used with any of the tweeters."

Not interested. I am interested in what YOU do, not what someone else did. You cannot continue to use someone elses work, I am interested in what YOU would do. So far I am left unconvinced that you know what you are talking about

I am overwhelmed over the interest of what I do. Care to share the data from your DBTs again?

Once again, stop pointing to someone elses work. If you are so knowledgeable, I am sure you can think things up for yoursel

Yes I can. I have a reference list of my scientific work if you want to know.

In case you have forgotten, the title of this thread is SACD vs CD unfair comparison. That means we are not talking DVD-A vs CD. You are making me dizzy with all of this spin action you are trying to put on this topic. You are busted, but your over blown ego won't let you admit that you don't know what you are talking about( a problem you and Chris share). Stevie Wonder can see that

Well it takes a while to understand science. I can't help if it makes you dizzy.

Go back to the title of the thread. You cannot quote the noise levels of one format, then change to a completely different one. That is called spin, and this is a no spin zone. Try again buddy

I've read it. This question is solved. There are different mixes of SACD and CD. The remaining questions in VmAx first post is the topic now.

You are not qualified to make this statement. What format did you bring to the consumer? None, and I am sure the engineers that created the SACD know alot more than you about digital audio.

Of course I am qualified. It suffers from noise and when filtered only marginally better performance than PCM 16/44.1. This view is shared by many and among recording engineers.

That is because your testing method is overly complicated, full of audio degrading and unnecessary conversions, and downsampling. That is not a test that would pass mustard at AES

That's your opinion. You have not published anything so you don't know.

You are not testing any formats with what you propose, you are testing the quality of the conversion and downsampling process. Hardly what I would call a SACD vs CD comparison which is the crust of these threads

Nope the unfair comparison SACD vs CD has been solved. It is unfair because they are differently mixed. The question is whether higher sampling rate or more bits are audible compared to 16/44.1.

Once again you are off topic. This is SACD vs CD not DVD-A vs CD. Damn I am getting dizzy. Spin, spin spin!!!

Read the initial post of this thread. The main question between SACD and CD has been solved. It is not possible to do the comparison if not the mixes are identical. They are very seldom the same. An exception might be the K668 Stereophile project (cited from Stereophile):

"Sidebar 1: K622: The Music

K622: Mozart Clarinet Concerto MFSACD017/MFLP017

Antony Michaelson, clarinet, with the Michaelangelo Chamber Orchestra conducted by Robert Bailey
Flute: Andy Findon, Helen Keen
Bassoon: Brian Sewell, Francesca Carpos
Horn: Richard Watkins, David Wythe
Violin: Adrian Levine (concertmaster), Kathy Andrew, Alex Balanescu, Sue Briscoe, Gordon Buchan, Beverly Davison, Ruth Erlich, Jonathan Evans-Jones, Alison Kelly, Pauline Lowbury, Rona Murray, David Ogden, Julian Tear, Paul Willey
Viola: Marina Ascherson, Rachel Bolt, Tim Grant, Rusen Gunes
Cello: Naomi Butterworth, Mike Hurwitz, Judith Serkin, Jonathan Williams
Double bass: Paddy Lannigan, Steve Williams

LP Side 1: Allegro (12:38)
LP Side 2: Adagio (7:58), Rondo (Allegro) (8:51)

SACD Hi-Rez Layer
1: Allegro (pure DSD) 12:38
2: Adagio (pure DSD) 7:58
3: Rondo (Allegro) (pure DSD) 8:51
4: Allegro (DSD transfer from analog tape) 12:38
5: Adagio (DSD transfer from analog tape) 7:58
6: Rondo (Allegro) (DSD transfer from analog tape) 8:51

SACD "Red Book" Layer
1: Allegro (PCM downsampled from DSD) 12:38
2: Adagio (PCM downsampled from DSD) 7:58
3: Rondo (Allegro) (PCM downsampled from DSD) 8:51
4: Allegro (PCM transfer from analog tape) 12:38
5: Adagio (PCM transfer from analog tape) 7:58
6: Rondo (Allegro) (PCM transfer from analog tape) 8:51

Recorded by Tony Faulkner in Henry Wood Hall, London, on November 19, 2003. LP mastered by Stan Ricker and pressed by RTI, Camarillo, California"

Doesn't matter what you use based on your testing standards, it would not get the result that is desired

That's only your opinion. If there are two conversions from one master tape, you claim that the one closest to the master is the best one. That's very true. If there is one that is more close, that implies that there must be a difference between the two converted ones. Thus, I need not compare the samples to the master; I can compare them to each other. The question is if whether a higher bandwidth is audible. Your desire is not the issue.

And you are no audio journalist either, so this is irrelevant. I have probably done more DBT that you have anyway, so that would make me just a little more qualified than yourself.

Please entertain me. Where were we... yes..the DBT results, can you share?

[mtrycraft]

you are blathering. again.

Coming from you? LOL. You cannot comprehend the issues. Priceless indeed.

[mtrycraft]

The test that AES ran tested the hearing of the subjects that participated. The ones that could hear the highest frequencies were the ones that tested with a positive result, and the ones that had some hearing damage(high frequency losses) produced the null result.

You keep referring to the AES running this test.
Please cite the work. It better be good, not that sloppy conference paper.

[mtrycraft]

Mtry tried to cite a faulty test that Sony did as "evidence" that resolution should be eliminated as a causal effect. To me, a faulty test proves nothing one way or another. You can draw all the inneuendo and conspiracy conclusions that you want, but that hardly qualifies as evidence or proof of anything.

Faulty? You read it? You don't like the answer?
You have a better one? Didn't think so. Come back when you have something.