ravon said:
According to me the main differences in cables lie in its ability to preserve the accuracy of the low level signals (detail). If you lose that in any part of the system, the cables will have no or minimal effect.
I'm not sure what you try to achieve but I would say the system used for playback must be able to show differences in cable by itself. Again the music used should be well recorded acoustical instruments.
I'm also sceptical about changing from digital to analog to digital to analog. You lose detail with every conversion.
André
There's no doubt that the recording method should be superior to the signal source so that maximum transparency is obtained.Andre Visser said:
Does your system show differences in cable by itself?
Do you know a better way of recording than in 24/96? Do you prefer 32/192 or an analog tape recorder?
😀 Is that a trick question? I meant that you should be able to hear cable differences on the system that you want to use for playback.
No but what is the reason for the re-recording if you have to use a system that prove the point already?
It's not a trick question, I'm just checking if you think your system would be good enough to be used for that purpose. Can you hear differences between cables on your system?Andre Visser said:
If we agree that 24/96 is good enough to record signals in such a way that it is possible to use these recordings to identify differences on a good audio system you could make recordings with different cables which I could listen to and vice versa.
KSTR said:
Man, and I'm accused of being a geek..sheesh.
Personally, I'd use the following: three speakers...and.
Bass guitar or bass organ, one female vocal (Cheryl Lynn being my favorite), and perhaps a piano..
Run the parameters to put Cheryl 15 degrees to the right, both IID and ITD.
Run the parameters to put the piano 15 degrees to the left of center...this is a confounder.
Put your center speaker exactly 15 degrees to the right.
Use a hand held button to switch from Cheryl's voice coming only from the positioned speaker, and that of the derived image as presented by the two outer ones.
Now, using any combination whatsoever, compare the perceived location of Cheryl's voice as presented by the two outer speakers with the physical location of the "center" speaker. Either alter ITD and IID to present unity of location, or move the center speaker to the perceived location from the two outer speakers.
Or even, alter ITD and IID to MAKE the perceived and the real exactly in the same location.
Then play with the cables to determine if the cables make the real and perceived images drift w/r to one another. Add the Bass note any place you wish, to add confounding information...
Cheers, John
John,
ahem, not sure if I got this right... did you refer to individual mono tracks of vocals/bass/piano? Otherwise, what is the goal applying additional ITD/ILD to stereo signals, to compensate for an inperfect setup, wrt room acoustics symmetry? Normally a mono noise signal is used to adjust your sweetspot manually for correct ITD (distance) and ILD (balance pot at the amp) and maximum stability/area (toe-in/out).
The comparison between real source and its phantom image (regardless how this is synthesized) has a severe design flaw, because is doesn't match the different frequency responses of both direct and reverberant sound because this is impossible (without extreme DSP convolution-based tricks which can get you closer, for a small very confined listening space). It's a wonder that stereo still works in spite of this very obvious flaw (once you've tried trinaural, you know how severe this flaw indeed is). The problem is, this different sound by design cannot result in the same percieved localization. Stereo is basically still sort of circus trick, it's pure illusion. And the creation of a "realistic" soundstage ("realistic"; read: convincing illuision) is the point where the tricking skills of the recording engineer come into play...
- Klaus
ahem, not sure if I got this right... did you refer to individual mono tracks of vocals/bass/piano? Otherwise, what is the goal applying additional ITD/ILD to stereo signals, to compensate for an inperfect setup, wrt room acoustics symmetry? Normally a mono noise signal is used to adjust your sweetspot manually for correct ITD (distance) and ILD (balance pot at the amp) and maximum stability/area (toe-in/out).
The comparison between real source and its phantom image (regardless how this is synthesized) has a severe design flaw, because is doesn't match the different frequency responses of both direct and reverberant sound because this is impossible (without extreme DSP convolution-based tricks which can get you closer, for a small very confined listening space). It's a wonder that stereo still works in spite of this very obvious flaw (once you've tried trinaural, you know how severe this flaw indeed is). The problem is, this different sound by design cannot result in the same percieved localization. Stereo is basically still sort of circus trick, it's pure illusion. And the creation of a "realistic" soundstage ("realistic"; read: convincing illuision) is the point where the tricking skills of the recording engineer come into play...
- Klaus
ravon said:
OK so you will have to make the recording. 😀
Sound interesting. What CD Player and cables do you have? Music?
Andre Visser said:
I have a Rotel RCD-1072 CD player and I use DIY cables made from left over shielded twisted pair industrial signal cable which I buy per reel of 500 meters. Connectors are from Neutrik. Works great at very low cost. I also have a pair of vd Hul cables which I don't use because I don't like the orange color. If you're interested I could look around for some other cables and make a couple of recordings.
I guess music should be your choice. In what kind of sounds do you hear differences?
ravon said:
The Tripath TA-2020 chip which is in your Charlize has a switching frequency of several hundreds of kHz, that's a signal with a period in the 10*10^-6 second range. That switching frequency is not filtered out 100% so it's available on the output of your Charlize and it's right in your face when you listen to music.
If you're right there must be a lot of nasty audible modulation products available in the output of that Charlize. I don't understand how you will ever be satisified by the sound of such an amplifier if you are able to hear differences of 2*10^-6 seconds. 😀
Perhaps it's time for a doublecheck of that gut feeling before we discuss the resolution of a 24/96 recording compared to a 16/44.1 recording as we find it on CD's 😉
The 2*10^-6 s applies to our real life analog world, not hifi reproduction.
Have you heard a Charlize, Ravon? They're very good. The filters seem pretty effective though a harsh distortion becomes noticeable at high volumes.
The recording idea is interesting. If done using agreed conditions - we could use 2 recordings of the same music as a standardised DBT.
KSTR said:
Individual mono tracks..
I concur it has problems when reverberant is considered.
Ignore that. Stick with first arrival.
The game is stereo. Work within that game...😉
Course, the two channel market is going away...
Cheers, John
ps.... I am not interested in the "tricking skills" of the recording engineer.
If it requires "tricking skills", then the game has been lost..
Andre Visser said:
I see a photo of a blonde woman, I have feelings that she is mysterious and beautiful. "Any ideas of how to measure those are welcome."
All I know is measurements feel good to me but not as good as a woman...... and I could supply measurements that prove this. 😀
ravon said:
If the differences are as large as reported it will not only be possible but very easy.
John,
Even with no reverberant field there is a big difference between any combination of two sources forming a phantom sources and a real source. If you were to look at the eardrum signals (with ear-channel miniature mics) there woulnd't be much correlation, both in time and spectral domains.
It's our ability of gestalt recognition that does the trick. If you have a mono phantom center (zero ITD/ILD) this is one hell of comb filtering, giving drifting and shifting comb filter different signals at each eardrum. Still we are able to assign that a phantom position, but (and that's my point) this phantom source has different perception than a real source. One common observation is that centered phantom sources are localized a little bit high (slightly from above the speakers acoustic center) than that same signal as real center source (and as single sided "full pan" sources as well). It's related to the HRFT difference resulting from those different incident angles. A sound apparently coming from the center but having the HRTF of +-30° sources (including the crossttalk) seems to trick our brains to the perception of an elevated center source probably because this combined HRTF has some components that an elevated cnter source would have. Since we have learned to assign a sound source dirrection with real sources, we also tend to assign a phantom source to a location that is closest to a thing we know from real-world experience.
And what I meant with "engineering tricks" is then stuff like this: If you want a center phantom that is localized "below the horizon" you need to apply some processing (not only simple EQ) to produce a situation that tricks our brains into percieving the wanted effect. Same with depth positioning, etc... The problem and the art is to use these tricks (for lack of a better name, I'm german after all) wisely and in an artful way -- creating that perfect illusion were are after (and which is strongly individual, that is my firm belief). Cheap and too obvoius tricks don't work for long term satisfaction, there I fully agree.
- Klaus
Even with no reverberant field there is a big difference between any combination of two sources forming a phantom sources and a real source. If you were to look at the eardrum signals (with ear-channel miniature mics) there woulnd't be much correlation, both in time and spectral domains.
It's our ability of gestalt recognition that does the trick. If you have a mono phantom center (zero ITD/ILD) this is one hell of comb filtering, giving drifting and shifting comb filter different signals at each eardrum. Still we are able to assign that a phantom position, but (and that's my point) this phantom source has different perception than a real source. One common observation is that centered phantom sources are localized a little bit high (slightly from above the speakers acoustic center) than that same signal as real center source (and as single sided "full pan" sources as well). It's related to the HRFT difference resulting from those different incident angles. A sound apparently coming from the center but having the HRTF of +-30° sources (including the crossttalk) seems to trick our brains to the perception of an elevated center source probably because this combined HRTF has some components that an elevated cnter source would have. Since we have learned to assign a sound source dirrection with real sources, we also tend to assign a phantom source to a location that is closest to a thing we know from real-world experience.
And what I meant with "engineering tricks" is then stuff like this: If you want a center phantom that is localized "below the horizon" you need to apply some processing (not only simple EQ) to produce a situation that tricks our brains into percieving the wanted effect. Same with depth positioning, etc... The problem and the art is to use these tricks (for lack of a better name, I'm german after all) wisely and in an artful way -- creating that perfect illusion were are after (and which is strongly individual, that is my firm belief). Cheap and too obvoius tricks don't work for long term satisfaction, there I fully agree.
- Klaus
janneman said:
Yes Jan. These children grow up (?) to be called "Audiophiles". If they "hear" something, it must be real. Never can it be a figment of the imagination. Never.
jneutron said:
Hi John. You've lost me here. How is the stimulus in a blind test at the home of superlistener A, listening to cd/LP X, any different from the stimulus from cd/LP X being played by superlistener A, during normal psychogenic listening sessions?
jneutron said:
I believe it was Soundmind who referred to it as tinkering. I once asked John the same thing I always ask Subjectivists, including here in this thread (even though I know they can't answer😉 ), what was the design/engineering process used by the manufacturer to create the "good' sounding capacitor. More tinkering? Top secret unknown to science measurements? Alchemy?
Ok, back to the thread topic.
Andy
Andre
Rabbitz
Analog SA
etc, etc.
You claim to be able to "hear" differences in cables that measure the same.
Please post the measured L, C and R of any 2 cables you have compared against each other, in your systems, in your home.
Or post the excuse(s) why you can't/won't. That will also provide an answer.
We'll worry about output/input impedance, etc. later.
Gentlemen?
cheers,
AJ
Cal Weldon said:
But we haven't discussed how carpet spikes affect the comparison of cables....
Cal Weldon said:
Yeah, I agree.. there are too many people making snide remarks.
I always assume when people do that, or start yelling, they are not thinking rationally, or objectively 😉
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and..........
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it reminds me too much of my Ex
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Lock 'er up Cal !! I mean the thread .......
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