Here's one paper on ultrasonic content content in audio, it is amusing due to the year he had a hard time finding computer sound hardware or source material for some of his experiments. My take away was that eliminating the IM distortion of a typical tweeter is essential in doing ANY of these experiments. Also it shows how hard these things are to do and catch all the confounders. I have posted this at least twice before.
http://www.davidgriesinger.com/intermod.ppt
http://www.davidgriesinger.com/intermod.ppt
Here's one paper on ultrasonic content content in audio...
Interesting. The author weaves together a good, coherent explanatory story based on what he knows. And confidence that is story is true has been shown to be primarily a function of its coherence, much more so than its ultimate physical truth. I recently came across some mention of some poorly understood extra-basilar mechanism(s) involved in some subtleties of hearing. As we continue to discover more and more about how the brain and senses work, models may have to undergo some updating from time to time. Good article though, thanks for the link. It's was helpful and illuminating. My guess would be that there may turn out to be more to the story that has yet to be told.
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Last time I looked at a filter textbook it was clear that some filter types introduce no ripple at all into the passband: Butterworth and Gaussian for a start. Interestingly, people often seem to prefer the sound of filters which do introduce ripple.bear said:Last time I looked the different filter types all introduce varying amounts of ripple into the passband.
Also, remember that the appearance of ringing can be caused simply by removing higher harmonics. This means that the ringing was always there, but cancelled by anti-ringing which the filter has now removed.
I thought so, until replaced a 12ax7 , by 6n1pi.Opamp rolling is like inserting a different tube type in any amplifier position... yes it should sound the same unless your audio system is compromised.
More detail up there, likeable sound, the 12ax7 was already damn good.
But now i´m thinking about this situation, if it´s just because of tube change. ? Well, curves looks better..
Another reason, why it sounds different, is gain- (100% 12ax7; 6n1p ~38%) = DAC has to play much louder to have same loudness.
Bigger amplitude - more, if not all of the dac´s 32bits are put into work.. Also input amorph tx gets bigger workout too..
You change one damn thing, but you indirectly affect the rest.
Here's one paper on ultrasonic content content in audio, it is amusing due to the year he had a hard time finding computer sound hardware or source material for some of his experiments. My take away was that eliminating the IM distortion of a typical tweeter is essential in doing ANY of these experiments. Also it shows how hard these things are to do and catch all the confounders. I have posted this at least twice before.
http://www.davidgriesinger.com/intermod.ppt
Scott, I think you invited me to accompany you to Dick Burwin's place (if he still had visitors)? And then you say you don't want any email conversation?
Look up the State Accompany HF ribbon - now an old design, there are newer more better ribbons of this sort on the pro market. It was spec'd at <1% THD @ 129dB SPL. What to you estimate the THD to look like run 30dB lower? Curious.
So I agree, (putting it bluntly) many tweeters are not very good. (I was going to use a single word that starts with "s" and ends with "k", but I thought better of it). 🙄
And regardless of the apparent specs, or what Earl Geddes says or found, the bottom line is that with all these apparent issues and confounders, one can still discriminate between opamps (or preamps, or power amps) that this is the element under consideration.
I agree fully that "confounders" are a major issue, and one of the reasons that I take exception with the generalizations drawn (often by the authors of) many published papers.
_-_-
Also, remember that the appearance of ringing can be caused simply by removing higher harmonics. This means that the ringing was always there, but cancelled by anti-ringing which the filter has now removed.
Some filters can produce ringing in the passband, not just the stopband. It may take some energy, or forcing function, to excite the ringing, but how long it rings after that depends on filter pole locations, no?
Scott, I think you invited me to accompany you to Dick Burwin's place (if he still had visitors)? And then you say you don't want any email conversation?
I simply don't want to carry on an email discussion about the sound of op-amps behind the scenes. Now that DBT is off the table, at least for now, I see it will be a long wait. For clarity, again what is the hypothesis? That no two op-amps sound the same, which would be very hard on you?
If the tests are sighted then they are biased...
All humans are biased in many ways all the time. Does't mean it's always helpful to them.
Demian. iirc Dafos was recorded on 1/2" two track 30 ips tape. Plenty of bass.
Most CDs I've tested have a pretty much brickwall HP filter - aka the low bass is non-existent. Why, I dunno.
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Dafos was recorded with Keith Johnson's 1/4" 3 track recorder at 15 IPS.
Mickey Hart first heard the track at my house on stacked Entec subwoofers. He almost choked on the banana he was eating when he herd the big crash. What they did was lift the big drum array and drop it on the stage.
Unfortunately due to stupid contract BS Reference lost the rights to it and its now a Rykodisk item. I'm not sure what Rykodisk is using as a master since the 3 track is not compatible with much of anything. I don't know if Keith ever transferred it to digital. I'll check. If so we could get a sense of its spectrum.
There is no reason to limit bass on a CD. The system can go to DC. Most ADC's have an internal high pass filter. The ones I checked are at 1 Hz.
Most vinyl has high pass filters and bass summing. Bass summing can backfire if the actual bass content has lots of out of phase components from spaced microphones. You just lose bass. however low bass out of phase causes big vertical movement of the stylus, which can cause mistracking and even cause the cutter to part company with the disk, really bad.
All humans are biased in many ways all the time. Does't mean it's always helpful to them.
I guess bear was just trying to be honest/cooperative when he said if he heard no difference he would say so. At the time he didn't understand what I was saying, I wanted to make it clear that sighted listening and promising to be honest is a nice gesture but it won't cut it.
I thought so, until replaced a 12ax7 , by 6n1pi.
More detail up there, likeable sound, the 12ax7 was already damn good.
But now i´m thinking about this situation, if it´s just because of tube change. ? Well, curves looks better..
Another reason, why it sounds different, is gain- (100% 12ax7; 6n1p ~38%) = DAC has to play much louder to have same loudness.
Bigger amplitude - more, if not all of the dac´s 32bits are put into work.. Also input amorph tx gets bigger workout too..
You change one damn thing, but you indirectly affect the rest.
I think there is some confusion here. The digital level should be constant, unless you are using digital volume control ahead of the DAC. The signal in the digital input transformer is unchanged regardless. If the output of the I/V amp is lower then you need to turn up the volume after that stage. Usually thats a good thing, gets the system noise lower at the volume control and all the stages are operating at a better S/N point. Most audiophiles are afraid to turn the volume control past 1/4 and the system should be operated with the volume at 3/4 or more for normal serious listening.
If you take a device such as E.S. has proposed and it is in a seal container, not way to see the internal parts or if you placed ten different speakers behind a curtain which you could not possibly see through how is that not considered blind testing? Does the protocol for doing blind testing require that you actually need to be blindfolded or just that there is a second person who is controlling the test conditions? I'm sure that people here on many forums cheat all the time to prove their particular viewpoints and ingrained beliefs but discounting the cheats how do you actually do a wide scale test such as using Ed's boxes of unknowns without having absolute control of the test conditions? Can a public DBT even work or is this in fact just more fantasy and delusional thinking?
I tried to propose a possible DBT protocol that would not require several people working together to implement. Similar to placebo based testing. If sighted gave no info on the content it should be pretty blind.
I don't know if Keith ever transferred it to digital. I'll check. If so we could get a sense of its spectrum.
There is no reason to limit bass on a CD. The system can go to DC. Most ADC's have an internal high pass filter. The ones I checked are at 1 Hz.
I have the original LP and the RR CD purchased a long time ago on release. The drum drop is what smoked a pair of Quicksilver PA's being reviewed the same ones that modulated the lights and had so much distortion that the gamelan track sounded out of tune.
KJ's work is certainly wonderful but some CD's like the Kodo drumming I find have quiet passages so quiet they are not practical for normal listening. Another one BTW to find some serious bass.
Demian,
A quandary I have often thought about is this thought that we need massive amplifier power for peak SPL and bass notes and at the same time that most of the time we aren't listening at anywhere near a level to need that amp power. If you are truly only needing a few watts power to listen in ordinary conditions all that extra power just seems to increase the noise level at lower volumes, so what is the balance we really need to both listen at lower level and when we crank it up? Do we really need two different amplifiers for the two conditions or is there an optimum balance that a well trained EE would settle on?
A quandary I have often thought about is this thought that we need massive amplifier power for peak SPL and bass notes and at the same time that most of the time we aren't listening at anywhere near a level to need that amp power. If you are truly only needing a few watts power to listen in ordinary conditions all that extra power just seems to increase the noise level at lower volumes, so what is the balance we really need to both listen at lower level and when we crank it up? Do we really need two different amplifiers for the two conditions or is there an optimum balance that a well trained EE would settle on?
This is an interesting read.
http://www.ti.com/lit/an/sloa013a/sloa013a.pdf
Ran across while looking for in more depth data on compensation.
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http://www.ti.com/lit/an/sloa013a/sloa013a.pdf
Ran across while looking for in more depth data on compensation.
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If you take a device such as E.S. has proposed and it is in a seal container, not way to see the internal parts or if you placed ten different speakers behind a curtain which you could not possibly see through how is that not considered blind testing?
The protocol as stated by jcx allows unlimited sighted listening in preparation for the blind test. This has two problems, one building this device without any artifacts/clues is difficult and two, how do you proctor the blind part.
I guess bear was just trying to be honest/cooperative when he said if he heard no difference he would say so. At the time he didn't understand what I was saying, I wanted to make it clear that sighted listening and promising to be honest is a nice gesture but it won't cut it.
Seemed quite clear to me what you need on your end to proceed with expending some effort on your part. Bear seems to have other needs with respect to his availability. So, my understanding is no mutual agreement was ever reached.
Demian,
A quandary I have often thought about is this thought that we need massive amplifier power for peak SPL and bass notes and at the same time that most of the time we aren't listening at anywhere near a level to need that amp power. If you are truly only needing a few watts power to listen in ordinary conditions all that extra power just seems to increase the noise level at lower volumes, so what is the balance we really need to both listen at lower level and when we crank it up? Do we really need two different amplifiers for the two conditions or is there an optimum balance that a well trained EE would settle on?
Most power amps have the highest SNR at full power of any electronic stage. With the higher signal level is much easier to get the 100+ dB SNR numbers needed for power amps.
Back to opamps and their sounds. . .
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