Sound stage = dynamics range. Dynamic range = difference between the most intensive signal that can be reproduced without extreme distortion and the least intensive signal that won't be masked by the self noise of preamp. It is impossible to have a good sound stage without great dynamic range, because reverberation is info that's most easily lost (it's the same sound, but much less intensive, heard after a tiny period of time). Music recorded in some specific space (church), with low number of mikes is ideal for such tests. If you can hear exact position of instruments + specific reverbation of the room where it's recorded, you shall probably have other important aspects of good sound reproduction as well. If sound stage is lost, such huge problem should be measurable, somehow.
German magazines STEREOPLAY and HiFiVision produced a series of CDs (30 years ago) with exact drawing of instruments positions in the church, dimensions of the church hall, which part of the church is marble and which part is wood, where main Schoeps mikes are, how high they are, the method of mike positioning (ORTF), where is the big Neumann stereo condenser overhead (if used), which PCM recorder is used, etc. That's reference, so when you listen to your system you should have all these clues heard in proper way. But we all cannot have the same real reference, so subjective impressions of individuals has at least some merit.
German magazines STEREOPLAY and HiFiVision produced a series of CDs (30 years ago) with exact drawing of instruments positions in the church, dimensions of the church hall, which part of the church is marble and which part is wood, where main Schoeps mikes are, how high they are, the method of mike positioning (ORTF), where is the big Neumann stereo condenser overhead (if used), which PCM recorder is used, etc. That's reference, so when you listen to your system you should have all these clues heard in proper way. But we all cannot have the same real reference, so subjective impressions of individuals has at least some merit.
If Bonsai's impression of how a Marantz amp sounds is so identical to mine (and to some other people too), than is ceases to be subjective and becomes (a sort of) objective. It becomes a reference. That's the best and the most precious reference one can have, a impression of another man (that's why we read books, we want to share experiences with other human beings). More objective than that is above human capabilities.
That' s why I intend to build Ovation nx.
That' s why I intend to build Ovation nx.
When I speak of high fidelity I always mean acoustic instruments in real spaces. Compressor/limiters,etc. are used with close miking, and heavily processed recordings. Popular music has only artificial reverberation, that's not high fidelity, you can not use that as reference. That's not real space but the sound of digital delay processors. In some way that's not sound stage at all, but some random construction of it. But, if properly applied, recordings with such artificial "space" can be used for comparisons.
I still have Ortofon LP test record, and there is a track on it which is used to evaluate dynamic range. It's just an ordinary recording in real space (church) which has it's reverberation. Test record says that if you can hear this reverberation in proper way, you have good dynamic, because if you don't, either would loud passages be distorted or proper reverberation would be lost. Ok, dynamic is not equal to sound stage but is conditio sine qua non for proper sound stage.
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I know of at least 4 Sydney DIY Audio members who have heard that design.
In fact, I had one myself .(Was that the Ultra Fidelity?) There were a couple of designs by David Tilbrook including an NE5534 based preamp where the output could be tapped off at the output of the 1st NE5534.The Sydney members who heard that one agreed that it sounded better from the 1st output.(a couple of us had that one too) So much for opamps being transparent, even though 100s were used in mixing desks.
Alex
Opamps are more transparent on 'sparse spectrum' signals and that's generally what's presented to channel strips in mixing consoles - a single instrument. After all a single sinewave gives very low THD from a typical opamp and that's as sparse as signals get.The critical opamps in a m a console are those handling the complete mix - i.e post the mix bus and fortunately there's not normally a long chain of them there as there is in a channel strip.
THD is not the only possible distortion in a loudspeaker... IMD is also making its job, and it can be easily measured by using a couple of simultáneous test tones. It depends heavily on the speaker design, but I have seen measurements of a Vifa XT25 tweeter, an old but "good" tweeter with a better behaviour than a lot of tweeters available in mid-end and hi-end loudspeaker designs, with 3kHz+6kHz test tones (and then with 2.5kHz+2.6kHz), and it generated several IM products inside de audio band... very visibles by using ARTA. I don't remember exactly the IMD value, but I guess it was about 0.1%... still greater than the pre-amp distortion.
A midwoofer Scan Speak 8545k tested with 100 & 350Hz, and then with 100 & 1kHz, didn't show IMD in both tests, at least above the setup's noise floor.
ezavalla,
I would think that those tests for IMD would have very different results at different output levels. The distortion product of those tests is usually done at a rather low 1 watt @ 1 meter test function and this could be quit different at say 10 or 25 watts of input power. Modulation distortion in a loudspeaker is very much tied to spl output. Doppler shift is going to be much greater at higher output and this will have a direct effect on IMD.
I would think that those tests for IMD would have very different results at different output levels. The distortion product of those tests is usually done at a rather low 1 watt @ 1 meter test function and this could be quit different at say 10 or 25 watts of input power. Modulation distortion in a loudspeaker is very much tied to spl output. Doppler shift is going to be much greater at higher output and this will have a direct effect on IMD.
That´s partially true, but the results are not that different.
Yes... but I am talking about "home listening levels", so from 1 to 5 watts of input power is all that is needed to "fill" the room until you almost cannot hear to somebody beside you.
It is also very much tied to the frequency range of operation, and this is the cause because tweeters are "so vulnerable" to IMD.
May be.. or may be not... I don't know
Even when exists a couple of methods to try to quantify Doppler shift, I'm not quite sure about its importance at normal input power levels.
Ezavalla,
I guess it would also have a lot to do with the efficiency rating of the speakers under test. Yes a compression driver or horn loaded system may work wonders at 1 or 2 watts with over 100db output but not with an 86db per watt speaker. Then we are going to get into much higher amplifier outputs to produce anywhere near the same spl output. And something like the 6 1/2" speaker that I have built that will get into the 110db range with the lower efficiency range is going to have a very long cone travel while also producing a 35hz wavelength with a top end range of about 2.5khz before rolling off. Very different than a 15" cone driver only going as high as 600hz flat before cone breakup as a comparison.
I guess it would also have a lot to do with the efficiency rating of the speakers under test. Yes a compression driver or horn loaded system may work wonders at 1 or 2 watts with over 100db output but not with an 86db per watt speaker. Then we are going to get into much higher amplifier outputs to produce anywhere near the same spl output. And something like the 6 1/2" speaker that I have built that will get into the 110db range with the lower efficiency range is going to have a very long cone travel while also producing a 35hz wavelength with a top end range of about 2.5khz before rolling off. Very different than a 15" cone driver only going as high as 600hz flat before cone breakup as a comparison.
That one dates from around 1981 and I still have mine, replete with Signetics 5534ANs. Sadly, friends I was trying to impress with my build, dis'd it outright, whatever mods I made and I did try every sensible recommendation I could find. This was nothing to do with op-amp quality but the overall design, power supplies and layout, as I later found in an abbreviated version, using the same active circuit designs.
With its forest of shielded wiring, flashy 20-LED meters, calibration tone and many controls, it was aimed at wanna-be pro audio guys rather than audiophiles, IMO.
But that's a deflection away from the question, which I'll state again. When you mentioned 1% as your figure of distortion, were you talking about THD or some other measure?
Most certainly - yet David Greisinger found IMD in practice was more significant in electronics than in tweeters. A two-tone IMD test of a tweeter wouldn't be at all representative of music as the crest factor's too low.
Would you care to justify that statement? Or is it just one of the "Golden Rules" of audiophile dogma?
Of course soldering the devices in is going to be a more reliable method ultimately, though I have used turned-pin gold plated sockets. But other than that I can't see any reason why it should make a significant difference.
Even assuming you have some sensible reasons for making that assertion, the PCB is DSPT and this means changing opamps is impossible without ruining them and probably the board too. So I would then be stuck with the original shitty sounding Fairchild 5532s. Or perhaps they just needed to "break in" (Golden Rule number xx)