Joe will wax on indefinately that that is the only way to capture the emotion of those old 78's. Quite a guy.
I found one thing consistent. There is a transient snap in real sounds that is simply lost as far as it has passed the microphone. I made many also classical recordings myself and sitting at the mixer with the direct feed from the microphones ( expensive, selected Sennheiser high voltage capacitors ), Stax electrostatic headphones on and off there is a starling loss of high frequency speed and energy. When my son playes the frensh horm 3 stories up, two closed doors inbetween i know it´s real. Instantly. I never heard that effect recorded.
Joachim, that´s exactly what I thought today when I attended
a string quartett concert (and each time I do so).
It´s evident from the first second they play. It´s
sad to hear that even with the best recording equipment it
seems to be impossible to overcome this deficiency for they
time beeing.
a string quartett concert (and each time I do so).
It´s evident from the first second they play. It´s
sad to hear that even with the best recording equipment it
seems to be impossible to overcome this deficiency for they
time beeing.
Difference is that you can wear the R at work, under a car, at the beach, or at the philharmonic(a).
Not sure if a T has as much stamina, never had one, the tick of an R sounds real nice.
(an O was pretty dissapointing, stopped 6 times in a couple of years. something to do with static skin, i was told.)
I was EXTREMELY disappointed with my Timex. When i was maybe 9 ot 10 they had a show in TV. There was a horse that wore a Timex at the leg. The horse was going through thick and thin and the Timex survived. I do not know how i got the money together but i got a Timex. I was a lot swimming at that time. I took it to the swimming pool and went home. On the next day rust was swimming in the glas...... the biggest ripp off i ever wittnessed.
Yes, they can. We have a PHD here in Germany that researched that for 30 years and wrote a recent book about it. I am a bit tired and have to leave early to Frankford to attend the Pro Light and Sound fair. When i am back i may have remebered his name. Back in the 90th Prof.Hawksford sat me on a chair. Two speakers where firing directly into my ears from the side. He could fly Helicopters over me and let it rain 10 meter UNDER me. Not to mention putting sound before and behind me. It sounded a little coloured but i was flabbergasted. Britsh Telecom has another system that is even more astounding. You do not even have to sit in an iron fist.
Joachim, many people have never heard really good sound, and they hear rumors that it is impossible, by people who would prefer to ignore hi end, and just make audio designs as cheap as possible. You have heard the difference, as I have, so we continue to improve audio as best that it is possible to do. Over time, I have always found that we make progress, all else being equal, in our designs. It may be differences in topology, parts selection, or design tradeoffs, but we do converge on what works better, and what does not work as well.
I definetely made progress over the last 20 years. I listen from time to time to my vintage speakers. For example the Virgo 2 from 1996 that a customer of mine has in Tokio. He collects classic speakers i made so he has also old Tempos and Allegrettos and around the corner are some Avanti 2 that have being used for monitoring at Lyra. As good as they are they can not compete with my recent designs. The newer ones sound even less colored and more dynamic. My system at home is also evolving. I recently listened to "Dog eat Dog" from Joni Mitchel. I have not heard that record for the last 20 years and i thought i knew it in and out. The sound it made was quite a shock. Huge stage and very alive, vibrant and dynamic. I remember it more like an average recording with fine music though. In total i whould say that my current system makes average recordings sound much better then they did in the old times. I made the oposite experience then many that say that a good system makes bad recordings really sound bad. Mine brings out the best in many recordings and i call that "musicality".
No processing allowed. I meant two mics/two speakers. I've heard several of the fakeries myself, but tricks they are.
I meant, if I wasn't clear, recording with a stereo pair and recreating EXACTLY the same 3D soundfield that a listen would hear. JJ from AT&T(?) is my source, this stuff does not interest me much. This is the A/B comparison available to most of us and to which the comments usually apply.
I agree that Joni Mitchel Dog eat Dog in vinyl is an outstanding recording.
I totally agree that a good system should bring out the best in every recording... Mine does that and some more
And I say: Who cares IF just 2 loudspeakers can accurately reproduce EVERYTHING about an original recording? Most recordings are generated by a bunch of microphones, anyway. It is also, the sound quality, and the subtlety of the musical instruments, if passed through the reproduce chain with the best fidelity possible, that closest brings out the musical experience.
When I design, I try to cover as MANY bases as I can. I try for low harmonic and IM distortion, excellent stereo separation, virtual elimination of added higher order harmonic distortion, extended frequency response, and minimal TIM and PIM distortion.
There are probably even more factors, but I can't think of them at this time.
To me, it is a balance between low distortion and extra stage complexity, design convenience by using popular and easy to get parts, and smoother sound quality, brag rights like super slew rate, or ultra low distortion, and the reality of what is actually necessary from a listening point of view, etc.
Here is where the debates begin. Some think that most distortion is OK. Others try to lower distortion to almost unmeasurable by the best test equipment available, and think that it improves the sound quality.
Some roll off the audio bandwidth below 20 Hz and 20KHz in a fairly severe way, yet others demand that DC to Channel 5 is necessary.
There is a middle ground, folks! And with some care, it works OK, that is where I try to design.
When I design, I try to cover as MANY bases as I can. I try for low harmonic and IM distortion, excellent stereo separation, virtual elimination of added higher order harmonic distortion, extended frequency response, and minimal TIM and PIM distortion.
There are probably even more factors, but I can't think of them at this time.
To me, it is a balance between low distortion and extra stage complexity, design convenience by using popular and easy to get parts, and smoother sound quality, brag rights like super slew rate, or ultra low distortion, and the reality of what is actually necessary from a listening point of view, etc.
Here is where the debates begin. Some think that most distortion is OK. Others try to lower distortion to almost unmeasurable by the best test equipment available, and think that it improves the sound quality.
Some roll off the audio bandwidth below 20 Hz and 20KHz in a fairly severe way, yet others demand that DC to Channel 5 is necessary.
There is a middle ground, folks! And with some care, it works OK, that is where I try to design.
Expressing generalities is all that I have been able to do on this thread, so far. I will now attempt to get a little more specific.
Let us take audio bandwidth, for example.
I always attempt to keep the effective bandwidth, at least one octave on each side of the nominal frequency extremes as accurate as possible. This might be a 50Hz square wave, and a 10KHz square wave, as two extreme examples. To be able to reproduce these two extremes, accurately, is one of my design goals. To do even better is recommended, if practical, but probably not as important at the state of high fidelity as we practice it today. Why do I choose this set of limits?
Well, practical microphones have limitations, and so do loudspeakers. Also, recording systems, either analog or digital, have similar limitations, and sometimes even more so.
I select a 10uS risetime square wave or pulse to be accurately reproduced. This requires at least a 35KHz bandwidth, at the -3dB point, and hopefully even more, perhaps 100KHz for an Amp or preamp, as limiting it here will make it more difficult for the whole audio chain to perhaps achieve similar performance, as the different bandwidth limitations are in series and one will compromise the others, so where practical, why not get the best result?
Many will disagree with me. Some will state that 20KHz is hard enough, and most people do not hear above it. Others will state that phase and rise-time compromises have been found by double blind testing to not be detectable. Well, so much for double blind tests of this sort. I have other independent audio tests that show that extended bandwidth and phase shift are audible, and I would prefer to pass every test, not just one type, in designing for best audio performance.
Let us take audio bandwidth, for example.
I always attempt to keep the effective bandwidth, at least one octave on each side of the nominal frequency extremes as accurate as possible. This might be a 50Hz square wave, and a 10KHz square wave, as two extreme examples. To be able to reproduce these two extremes, accurately, is one of my design goals. To do even better is recommended, if practical, but probably not as important at the state of high fidelity as we practice it today. Why do I choose this set of limits?
Well, practical microphones have limitations, and so do loudspeakers. Also, recording systems, either analog or digital, have similar limitations, and sometimes even more so.
I select a 10uS risetime square wave or pulse to be accurately reproduced. This requires at least a 35KHz bandwidth, at the -3dB point, and hopefully even more, perhaps 100KHz for an Amp or preamp, as limiting it here will make it more difficult for the whole audio chain to perhaps achieve similar performance, as the different bandwidth limitations are in series and one will compromise the others, so where practical, why not get the best result?
Many will disagree with me. Some will state that 20KHz is hard enough, and most people do not hear above it. Others will state that phase and rise-time compromises have been found by double blind testing to not be detectable. Well, so much for double blind tests of this sort. I have other independent audio tests that show that extended bandwidth and phase shift are audible, and I would prefer to pass every test, not just one type, in designing for best audio performance.
Now, what about distortion? As we know, there are many sorts of distortion measurements. In the early days, harmonic distortion was popular, usually with an oscillator and a wave analyzer, that is sort of like a mechanically driven spectrum analyzer, giving one frequency at a time. Sometimes we motor drove them to be able to sweep the whole bandwidth, most times not.
Later, about 1960, SMPTE IM became VERY popular. It was cheaper, faster, and could more easily have a lower residual distortion than harmonic distortion measurements at the time. We used it for a lot of amp and preamp testing, until the middle '70's, when cost effective designs like the Sound Technology became available, that could give you both harmonic (THD) and IM (SMPTE). The distortion products could be looked at by an oscilloscope and/or a wave analyzer, to evaluate the different harmonic structure. Both IM and THD would give a distortion residual that could be evaluated. (more later)
Later, about 1960, SMPTE IM became VERY popular. It was cheaper, faster, and could more easily have a lower residual distortion than harmonic distortion measurements at the time. We used it for a lot of amp and preamp testing, until the middle '70's, when cost effective designs like the Sound Technology became available, that could give you both harmonic (THD) and IM (SMPTE). The distortion products could be looked at by an oscilloscope and/or a wave analyzer, to evaluate the different harmonic structure. Both IM and THD would give a distortion residual that could be evaluated. (more later)
Now, we have arrived at the 'dawn' of hi end design, where most solid state designs that are well known today, and they were designed initially with IM and THD harmonic distortion, usually with a Sound Technology 1700 or an HP HD analyzer. Advanced measurements were sometimes made with an HP 3581 wave analyzer or HP 3580 Spectrum analyzer, but these were very expensive instruments in the middle 1970's, so they were not in every designer's lab. However, for research, the HP3581 was used in Matti Otala's Government lab for TIM measurements, at that time. It took me an extra 15 years, before I could afford one for myself, for example.
Well, what about today? So far as I have described, I have noted products that do NOT use FFT processing. This is the breakthrough in today's equipment. FFT makes it possible that YOU can use your own computer and with a little care, you can make a pretty good spectrum analyzer. Perfect? Not really, but cost effective and useful. What are the trade-offs between lab equipment and home computers with a few added programs? For most people, very little difference, except for the cost and convenience of the home computer option. However, the window of measurement is necessarily limited by the clock rate of the computer A-D converter and this can constrain one into a measurement area that is below commercial lab equipment that can also be used to make more extended and precision measurements. This can lead to a false sense of 'security' that one has done everything OK. After all the computer measurement can't find any problems, how could the human ear? (more later)
Well, what about today? So far as I have described, I have noted products that do NOT use FFT processing. This is the breakthrough in today's equipment. FFT makes it possible that YOU can use your own computer and with a little care, you can make a pretty good spectrum analyzer. Perfect? Not really, but cost effective and useful. What are the trade-offs between lab equipment and home computers with a few added programs? For most people, very little difference, except for the cost and convenience of the home computer option. However, the window of measurement is necessarily limited by the clock rate of the computer A-D converter and this can constrain one into a measurement area that is below commercial lab equipment that can also be used to make more extended and precision measurements. This can lead to a false sense of 'security' that one has done everything OK. After all the computer measurement can't find any problems, how could the human ear? (more later)
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