The importance of Components?
Mar 10, 2009 at 4:19 PM Post #61 of 121
Quote:

Originally Posted by vcoheda /img/forum/go_quote.gif
recording quality shouldn't be included. you need to start with well-recorded music. otherwise, the rest is all for nothing.


I agree entirely that you have to start with "well-recorded" music. The problem is that every track and every album has been recorded and mixed differently. These differences might sound "well-recorded" on one system, not so "well-recorded" on another and on yet another might not be noticable. So how do you tell, with your particular system, what is "well-recorded" and what isn't?

G
 
Mar 10, 2009 at 4:52 PM Post #62 of 121
Quote:

Originally Posted by gregorio /img/forum/go_quote.gif
I agree entirely that you have to start with "well-recorded" music. The problem is that every track and every album has been recorded and mixed differently. These differences might sound "well-recorded" on one system, not so "well-recorded" on another and on yet another might not be noticable. So how do you tell, with your particular system, what is "well-recorded" and what isn't?

G



That's where double blind tests come into play...
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Because we wouldn't want to trust our ears
 
Mar 10, 2009 at 5:39 PM Post #63 of 121
Quote:

Originally Posted by gregorio /img/forum/go_quote.gif
I agree entirely that you have to start with "well-recorded" music. The problem is that every track and every album has been recorded and mixed differently. These differences might sound "well-recorded" on one system, not so "well-recorded" on another and on yet another might not be noticable. So how do you tell, with your particular system, what is "well-recorded" and what isn't?

G



This is an interesting problem you have presented.

If you adopt my stance (and many people have) that all decently made equipment apart from transducers sounds the same and you support blind testing, then it's not such a big deal to figure out what is well recorded.

If you take the stance that all equipment sounds different, good luck figuring out what is a good recording. You'll never be able to determine where the recording ends and where the equipment starts.
 
Mar 10, 2009 at 5:42 PM Post #64 of 121
Quote:

Originally Posted by olblueyez /img/forum/go_quote.gif
I think if you go back a few posts that you will see that I didnt bring it up. Right?


Sorry, I've lost you. What is the "it" in the statement above. I'm not trying to be a pain, I'm genuinely confused.

The joys of discussion on the internet
smily_headphones1.gif
 
Mar 10, 2009 at 5:51 PM Post #65 of 121
Quote:

Originally Posted by odigg /img/forum/go_quote.gif
This is an interesting problem you have presented.

If you adopt my stance (and many people have) that all decently made equipment apart from transducers sounds the same and you support blind testing, then it's not such a big deal to figure out what is well recorded.

If you take the stance that all equipment sounds different, good luck figuring out what is a good recording. You'll never be able to determine where the recording ends and where the equipment starts.



Not many people have had the luxury to blind test all the decently made equipment.

Some of use have to determine with our ears what is a good and bad recording
 
Mar 10, 2009 at 6:59 PM Post #66 of 121
Quote:

Originally Posted by gregorio /img/forum/go_quote.gif
I agree entirely that you have to start with "well-recorded" music. The problem is that every track and every album has been recorded and mixed differently. These differences might sound "well-recorded" on one system, not so "well-recorded" on another and on yet another might not be noticable. So how do you tell, with your particular system, what is "well-recorded" and what isn't?

G



well-recorded music is objective, at least in a comparative sense. go through your CDs (or tracks) one by one and separate them by good sound and less good. keep on repeating the process until you have a smallish pile - the best of the best. it is this music that you should use to evaluate gear. and it is likely that this small set of music will qualify as "well-recorded" regardless of the system.

but my larger point is that in general CDs (or individual tracks if recording quality greatly varies) fall into two broad categories - good sounding and less good or even bad. if you have listened to enough CDs/music, you should be able to tell which side something new falls on.
 
Mar 10, 2009 at 8:03 PM Post #67 of 121
Quote:

Originally Posted by vcoheda /img/forum/go_quote.gif
well-recorded music is objective, at least in a comparative sense. go through your CDs (or tracks) one by one and separate them by good sound and less good. keep on repeating the process until you have a smallish pile - the best of the best. it is this music that you should use to evaluate gear. and it is likely that this small set of music will qualify as "well-recorded" regardless of the system.

but my larger point is that in general CDs (or individual tracks if recording quality greatly varies) fall into two broad categories - good sounding and less good or even bad. if you have listened to enough CDs/music, you should be able to tell which side something new falls on.



I agree but my question still stands. When you go through your CD collection and decide something sounds "bad" how do you know if it sounds bad because it is a bad recording or because it just sounds bad on your particular system? What happens if you get a new system?

OK, so this is a bit extreme, there are truly bad recordings out there, which will sound bad on pretty much anything. But most recordings are not obviously brilliant or rubbish but somewhere in between. Maybe it's a bit too harsh or bright, maybe there is too much bass, maybe it lacks clarity or stereo image. In these cases it's could be next to impossible to decide whether it's a slightly poor recording or just the way your systems plays it. It might sound good on another system and what sounded a little weak on another system might sound good on yours. How do you know?

The audiophile solution might be to go and spend $1,000 on a speaker cable so even the bad recordings sound great! (sorry, lol)
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G
 
Mar 10, 2009 at 8:28 PM Post #68 of 121
Quote:

Originally Posted by 883dave /img/forum/go_quote.gif
Not many people have had the luxury to blind test all the decently made equipment.

Some of use have to determine with our ears what is a good and bad recording



Quote:

Originally Posted by gregorio /img/forum/go_quote.gif
I agree but my question still stands. When you go through your CD collection and decide something sounds "bad" how do you know if it sounds bad because it is a bad recording or because it just sounds bad on your particular system? What happens if you get a new system?

G



The point of my original answer (in retrospect I should have stated it more clearly) is that if you think all equipment (except for transducers) sounds the same, it's mostly irrelevant what equipment you use. Just pick a source and amp from a decent (I think Denon is very decent) manufacturer and get some cables that are built well (Yes, Radio Shack is fine. If you want to go the distance, Blue Jeans would work).

My personal experience is that as long as the speakers are not changed, I can change anything behind it (as long as it's made decently) and the sound will be the same.

And let me clarify what I mean by decent. At minimum, I mean a flat frequency response. If a manufacturer is making something (again, the exception is transducers) and deliberately altering the frequency response, I don't think that is decently made.

Picking speakers is more difficult, but if you are trying to get an accurate representation of a recording, you'll want something that has a flat frequency response and a good off-axis performance.

A good (flat FR!) studio monitor in a nearfield setup would probably work best.
 
Mar 10, 2009 at 9:16 PM Post #69 of 121
Quote:

Originally Posted by odigg /img/forum/go_quote.gif
And let me clarify what I mean by decent. At minimum, I mean a flat frequency response. If a manufacturer is making something (again, the exception is transducers) and deliberately altering the frequency response, I don't think that is decently made.

Picking speakers is more difficult, but if you are trying to get an accurate representation of a recording, you'll want something that has a flat frequency response and a good off-axis performance.

A good (flat FR!) studio monitor in a nearfield setup would probably work best.



Certainly in my studio setup I have heard differences between equipment all over the place, even what would be considered fairly standard inconspicuous items such as when I switched master clock. So my personal opinion and that of the professional industry is that different equipment can make a substantial difference to the sound. In fact, certain bits of equipment in the recording chain is specifically chosen for it's sound colour, mic-pres are a good example and of course mics themselves.

My second point is getting a system which has a flat frequency response. Two things to think about here; 1. It's very difficult to buy anything which has a flat freq response, certainly you won't get it from nearfield monitors and you almost certainly won' get a good off axis response from nearfields.You want truly flat response across the spectrum you're going to need a great deal of money! 2. Mastering Engineers master the music based on an average of the likely systems used by consumers. As consumer equipment is by and large quite heavily coloured, listening to these mastered mixes on a fairly flat system will not give you the mix as intended by the mastering engineer.

G
 
Mar 10, 2009 at 10:29 PM Post #70 of 121
Hello, Jazz. I've begun my experiments with sinewaves and filters.

Here's the first part, which deal with amplitude modulation. The second part will adress time smearing.

Amplitude modulation can be produced adding two close frequencies f1 and f2. The result is a sinewave of frequency (f1+f2)/2 modulated by a so-called "beating" of frequency f2-f1.

Let's generate a 1000 Hz sinewave :

01-sine1k.png


Its spectrum is one single peak at 1000 Hz.

The same for a 1200 Hz sinewave :

02-sine1,2k.png


Now, if we mix them together, the result is an amplitude modulated sinewave :

03-mix.png


However, if we perform the spectrum analysis of the modulated sine, we can still see the two original frequencies :

04-twinpeaks.png


This is very important to understand : a modulated sinewave is nothing more than two close frequencies. There is absolutely nothing below that accounts for the modulation.

In a spectral decomposition, the fundamental frequency would indeed be very low, since the wave repeats itself only after many oscillations. In our case, it would be 200 Hz, with harmonics at 400, 600, 800 etc.

It is interesting to list the complete decomposition with their relative amplitudes :

Frequency / level
200 Hz / 0
400 Hz / 0
600 Hz / 0
800 Hz / 0
1000 Hz / 1
1200 Hz / 1
1400 Hz / 0
1600 Hz / 0
All other harmonics : 0

The mystery of the missing low frequencies is now clear : the fundamental is there, but with a strictly null amplitude, and every harmonic has a null amplitude except the 4th and 5th ones, that are 1000 and 1200 Hz.

Now, what happens if we filter this modulated wave ? Let's use SoundForge parametric equalizer with the 1000-1200 Hz lowpass, with -60 dB at 1200 Hz (it can't do less).

With the RMAA software, I've plotted the frequency response of the filter :

05-SFfilter.png


Frequencies of 1000 Hz and below are not attenuated. Frequencies above 1200 Hz are reduced by a factor of -60 dB.

Here is the result :

06-twinfiltered.png


There is a strange side effect at the beginning, with smearing, then the wave goes on without modulation.

The spectrum shows that the 1200 Hz peak have indeed been reduced by 60 dB. Since the only significant frequency left is 1000 Hz, we get back the sine, unmodulated.

Last, let's just have a look at a native amplitude modulated sine. I start again from the 1000 Hz sine, but instead of adding another frequency, I directly ask for amplitude modulation in the software filters :

07-sf-am.png


This way, my sine will be multiplied by a 100 Hz sine. Let's see the result and its spectrum :

08-sf-am-spectrum.png


Amplitude modulation is there, and this time, we have not two, but three peaks on the spectrum analysis. I don't know exactly why. But we can see that there is still no low frequency at all.

Now let's see the first part of your arguments :

Quote:

Originally Posted by JaZZ /img/forum/go_quote.gif
15011d1236602570-importance-components-symphony-2x-os-filter-1.jpg


...which shows the preserved amplitude modulation even after proper (although unusually smooth) low-pass filtering with a cut-off frequency around 20/21 kHz.



"Preservation" is not the right word here, since the diagram reads "Response 20 kHz sinus". Amplitude modulation is therefore an artifact in this example. It shows that the original 20 kHz sine comes together with other near frequencies. That's not surprising since we are playing with anti-alias filters. The frequency in question is certainly the alias of the sine, which is 22050 + (22050 - 20000) = 24100 Hz.
Together, the original 20000 Hz sine and the parasite 24100 Hz one produce the modulation. This is common with very simple filters. The frequency response of the filter drops to zero at exactly 22050 Hz, but goes up again above 22050 Hz, which allows for the 24100 Hz alias to be there ! Like in this diagram,

sincfreq.gif


Which is the frequency response of a "zero-order hold" filter (i.e. the bare output of a DAC with no filter).

Quote:

Originally Posted by JaZZ /img/forum/go_quote.gif
Since there are no frequencies above 22 kHz left, it's not aliasing in the classic sense. You're free to call it like this, but it's not caused by «forbidden» frequencies.


The simple fact that the sine is modulated shows that the response of the filter is not finished at 22050 Hz, and that there are further lobes beyond the diagram.
Unless you mean that there is another frequency below 22050 Hz that causes the modulation. But I fail to see how it could have been introduced, since the original is a pure 20 kHz sine.
The "sampling" at 44100 Hz only introduces new frequencies above 22050 Hz, not below (except noise). (Actually, it's the D/A conversion that introduces them).

Quote:

Originally Posted by JaZZ /img/forum/go_quote.gif
A «proper» filter in your sense is one with a massive filter resonance -- that would indeed do the trick.


Not necessarily. If your filter had no further lobes, it would just destroy the 24100 Hz alias, and attenuate slightly some frequencies below 22000 Hz. Think about the effect on the spectrums above. We would just get a slighty attenuated 20 kHz sine with very little smearing, and no modulation at all.

Quote:

Originally Posted by JaZZ /img/forum/go_quote.gif
Since you call it «smooth», it doesn't seem to be similar to a classic antialiasing filter which calls for ultimate sharpness. It's absolutely possible to create an unobtrusive low-pass filter, as MP3 encoders and the like show.


That's exactly the kind of filters used for anti-aliasing. They are called "brickwall" in comparison with Butterworth filters, that are usualy very soft, like 6 dB per octave, for example, while we need here minus infinite within a fraction of octave.
But they are smooth in comparison with real sinc filters.

For example, here is the frequency response of my Yamaha CD player (model CDX-860, 450 euros in 1991) :

Standard-chute.png


That's sharper than a speaker filter, but far from a "sinc" brickwall, that would be a vertical line.

Quote:

Originally Posted by JaZZ /img/forum/go_quote.gif
So your orange world view dictates you that my listening impressions aren't trustworthy beforehand -- and the filter variants are there to fool the customers...


You have perfectly stated your position on that matter : you are not interested in blind testing. I have no problem with it, but am myself only interested in blind testing. So I'll just deal with the technical side of the discussion
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Mar 10, 2009 at 11:51 PM Post #71 of 121
Quote:

Originally Posted by milkweg /img/forum/go_quote.gif
I read in the amp forum that the DAC on your Meier amp is having some issues. Have you read about it?


Just right after your hint. I have no problems with it in my main setup, and on my computer where I use the optical connexion there's also no issue. With coaxial connexion I get short dropouts when switching the desk lamp on or off. Not a real problem, though -- I just noticed it by provoking it after reading the thread.
.
 
Mar 10, 2009 at 11:52 PM Post #72 of 121
Quote:

Originally Posted by Pio2001 /img/forum/go_quote.gif
Hello, Jazz. I've begun my experiments with sinewaves and filters.
...
Amplitude modulation can be produced adding two close frequencies f1 and f2. The result is a sinewave of frequency (f1+f2)/2 modulated by a so-called "beating" of frequency f2-f1.



It took me a while to understand the intention behind your AM experiments..., but...

Quote:

This is very important to understand: a modulated sinewave is nothing more than two close frequencies. There is absolutely nothing below that accounts for the modulation.


Indeed. So the amplitude modulation in my examples -- more precisely the Symphony's filter 1 with 2x oversampling -- is indeed an aliasing product.

Quote:

Now let's see the first part of your arguments :

"Preservation" is not the right word here, since the diagram reads "Response 20 kHz sinus". Amplitude modulation is therefore an artifact in this example. It shows that the original 20 kHz sine comes together with other near frequencies. That's not surprising since we are playing with anti-alias filters. The frequency in question is certainly the alias of the sine, which is 22050 + (22050 - 20000) = 24100 Hz.
Together, the original 20000 Hz sine and the parasite 24100 Hz one produce the modulation.
This is common with very simple filters. The frequency response of the filter drops to zero at exactly 22050 Hz, but goes up again above 22050 Hz, which allows for the 24100 Hz alias to be there ! Like in this diagram...
...which is the frequency response of a "zero-order hold" filter (i.e. the bare output of a DAC with no filter).


Now you're searching too far.

Looking at this graph again...

sinuskurven.jpg


...it's easy to find the reason for the amplitude modulation: The aliasing is already there on the CD!

It's not caused by a violation of the Nyquist theorem (obviously, because there are no frequencies above 22.05 kHz stored on a CD). It's just a consequence of «improper» filtering: While on the CD the filtering is entirely missing, the Symphony's said filter 1 represents an «improper» implementation instead of a «correct» «reconstruction filter». Obviously a sharp filter slope is mandatory for eliminating the amplitude modulation. Which brings us back to the resonance aspect of the «reconstruction».

I'm curious about your time-smearing measurements. Thanks for your effort so far!
.
 
Mar 11, 2009 at 1:34 AM Post #73 of 121
Quote:

Originally Posted by olblueyez /img/forum/go_quote.gif
Yes, dig them up so we can shoot them down.


You can't shoot them down because your argument is purely subjective opinion and not objective scientific testing.

And all these graphs being posted mean nothing if the areas being measured are outside of the human hearing range.

I think it is time for me to move on because I am done discussing anything more with tweako idiophiles.
 
Mar 11, 2009 at 2:19 AM Post #74 of 121
Quote:

Originally Posted by milkweg /img/forum/go_quote.gif
ABX testing has already proven that long ago. Do I really need to dig up articles on this subject for you too?


Quote:

Originally Posted by milkweg /img/forum/go_quote.gif
I read in the amp forum that the DAC on your Meier amp is having some issues. Have you read about it?


Quote:

Originally Posted by olblueyez /img/forum/go_quote.gif
Yes, dig them up so we can shoot them down.


Quote:

Originally Posted by odigg /img/forum/go_quote.gif
I'm just being pedantic, but ABX testing has provided a large amount of evidence supporting the claim that cables make them no audible difference. It has not proved there is no audible difference because by stating "proved there is no difference" you have stated a claim that violates the important scientific idea of fallibility.



Is there any point in doing so? You've already established you don't believe in blind testing. Would a couple of papers on the subject make you change your mind?

Where is thread going again?



Quote:

Originally Posted by olblueyez /img/forum/go_quote.gif
I think if you go back a few posts that you will see that I didnt bring it up. Right?


Quote:

Originally Posted by odigg /img/forum/go_quote.gif
Sorry, I've lost you. What is the "it" in the statement above. I'm not trying to be a pain, I'm genuinely confused.

The joys of discussion on the internet
smily_headphones1.gif



So, what you are saying is that YOU have no idea what YOU are talking about right?
 
Mar 11, 2009 at 2:33 AM Post #75 of 121
Quote:

Originally Posted by milkweg /img/forum/go_quote.gif
You can't shoot them down because your argument is purely subjective opinion and not objective scientific testing.

And all these graphs being posted mean nothing if the areas being measured are outside of the human hearing range.

I think it is time for me to move on because I am done discussing anything more with tweako idiophiles.



I use my ears to evaluate my equipment, not articles claiming amps and cables all sound alike as you do, so I say that the term Quote:

tweako idiophile


describes you , not me. My hearing is as good as my vision and I am quite capable of driving a car. Your posting stems from a pathetic cry for attention and has nothing to do with the sound quality of stereo equipment. Furthermore, anyone with their head screwed on straight knows that what counts when choosing equipment are testimonies from actual users of the equipment and what that equipment sounds like. ABX that.
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