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Old 27th September 2009, 08:42 PM   #6221
mige0 is offline mige0  Austria
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Originally Posted by markus76 View Post
No, but why should it be "loudspeaker-free"? I'm talking about multichannel (Holman is working on 10.2), virtualization (Smyth Realiser, Beyerdynamic Headzone) or wave field synthesis.
I thought so
- but - we not even get consens about a simple loudspeaker - not to mention the gaps in cosidering electronics as to be seriousely contributing to depth of listening experiance (and I certainly do not refer to enhancers of any kind)
I'm afraid your suffering will last..


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Old 27th September 2009, 09:12 PM   #6222
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Originally Posted by mige0 View Post
I thought so
- but - we not even get consens about a simple loudspeaker - not to mention the gaps in cosidering electronics as to be seriousely contributing to depth of listening experiance (and I certainly do not refer to enhancers of any kind)
I'm afraid your suffering will last..


Michael
We would easily reach consensus on loudspeaker design if we had a better understanding what the limitations of a specific recording/reproduction technique are. We're still lacking meaningful measurement categories. E.g., we know that first reflections are a factor but we don't know how to describe that in meaningful measurement categories. See http://www.aes.org/e-lib/download.cf...86&name=harman

Best, Markus
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Old 27th September 2009, 09:16 PM   #6223
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....Our production preamp was 3 V/uSec, and the prototype was 13 V/uSec. Keeping the rest of the circuit the same - including the same parts for RIAA...

How do you keep slew rate the same without changing something in the circuit?
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Old 27th September 2009, 09:31 PM   #6224
gedlee is offline gedlee  United States
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Originally Posted by markus76 View Post
E.g., we know that first reflections are a factor but we don't know how to describe that in meaningful measurement categories. See http://www.aes.org/e-lib/download.cf...86&name=harman

Best, Markus
We know that there is little consensus on the importance and effects of these. Toole and I differ considerably on this point as an example.
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Old 27th September 2009, 09:34 PM   #6225
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I should add that the two problems mentioned in the previous post are nearly universal in consumer audio, whether the price of $500 or $50,000. Phono preamps commonly overload, slew, or both, and DACs, CD players, and HT receivers have substandard, slew-prone analog stages. Preamps and power amps commonly have multiple regulator stages with inadequate phase margins, so they overshoot or even "ring" for a few cycles when the device is hit with a step function in the current domain. The ringing can fall in the audio band, lending an annoying upper-midrange coloration that does not show up in a steady-state measurement.

Design oversights and outright faults make marginal recordings sound much worse, since the brief transients of mistracking, overload, or slewing are exaggerated and stretched-out by the electronics. They are as common in $50,000 heavily advertised equipment as it is in mid-price Japanese consumer electronics. You have to go to high-end studio gear - EMM Labs, Weiss, et al to get pro-quality engineering. It basically isn't there in the consumer sector.

It's not the job of the speaker to cover up defects in the electronics - but conversely, there are speaker faults that make problems in the electronics sound a lot worse. Resonances and stored energy in the 1~8 kHz range will certainly draw attention to problems in the source or electronics - so the worst case would be a beat-up record, a CD with a hot mix or outright clipping, electronics that slew or clip (clipping does occur in CD analog electronics), and a speaker with resonances, delayed energy, or significant diffraction in the 1~8 kHz range. Unfortunately, this describes many, if not most, consumer and high-end systems.

My concern is a subtle bias developing towards a handful of "audiophile" recordings to avoid the scenario above - in a sense, a type of signal conditioning. I've heard many systems where I knew the technical faults existed - as seen on the tech bench - so the hapless audiophile ends up listening to only a handful of magazine-approved recordings, just so they don't have their ears burned off when they listen to their expensive systems. The volume goes down, the type of "acceptable" music gradually changes, and finally, no music goes through it at all - instead, it gets used to play movies or TV shows. Does this describe any audiophiles you know?

Many audiophiles stop enjoying music altogether, since their systems are so tempermental, and sound so bad on so many recordings. They've spent a lot of money, they're enjoying it less, and they don't know why. What for? Why would anyone do that? To impress a handful of audiophile buddies how "accurate" their system is? That's a pretty thin and unrewarding pleasure, and hardly worth thousands of dollars, much less tens of thousands of dollars.

This completely defeats the purpose of a music reproducer, in my view. I feel it is much more fruitful - and healthy for the industry as a whole - to isolate and remove the technical faults that make mediocre recordings sound much worse. These faults are known, measurable, and identifiable. It's not black magic, just having the right instrumentation and knowing how to use it.

Last edited by Lynn Olson; 27th September 2009 at 10:00 PM.
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Old 27th September 2009, 10:05 PM   #6226
gedlee is offline gedlee  United States
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Lynn

I guess what I don't understand is how the electronics knows if its a good or bad recording. I mean you seem to be implying that it makes bad recordings sound worse, but somehow doesn't affect good recordings. That to me means that there must be something in the signal that differentiates a good recording from a bad one.

Also, I somehow find it hard to believe that all electronic circuits engineers are incompetent. Looking at the effects of out of band signals was fundamental in automotive - it was ALWAYS done. And that could be some pretty bad sounding stuff. At any rate, any engineer who worked in automotive would certainly know to do this. Did none of them ever make it into audio design? And these problems are not so tough to deal with. A simple cap in the right place and of the right type and value and "poof" problem gone.
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Old 27th September 2009, 10:08 PM   #6227
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Originally Posted by john k... View Post
How do you keep slew rate the same without changing something in the circuit?
Change a 709 externally compensated op-amp for an internally compensated 5532 op-amp, both with identical external current-helper circuits (so they stay in Class A mode, despite current demands from the RIAA-compensation feedback circuit).

Both measured very similarly on the Radford distortion analyzer with 0.0008% distortion residue, with a slight margin, not surprisingly, favoring the 5532 at the highest frequencies. But not a huge difference, really.

Easy to compare with A/B switching. Small subjective differences in "clarity" and all those vague audiophile words, but very obvious differences in surface noise. It took a fast storage scope to see what the glitches on the record surface really looked like - really narrow impulses, definitely fast enough to slew a preamp.
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Old 27th September 2009, 10:23 PM   #6228
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We know that there is little consensus on the importance and effects of these. Toole and I differ considerably on this point as an example.
That doesn't surprise me. There are virtually no scientific studies, so everybody with strong opinions is "right".

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Old 27th September 2009, 10:30 PM   #6229
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Originally Posted by gedlee View Post
Lynn

I guess what I don't understand is how the electronics knows if its a good or bad recording. I mean you seem to be implying that it makes bad recordings sound worse, but somehow doesn't affect good recordings. That to me means that there must be something in the signal that differentiates a good recording from a bad one.

Also, I somehow find it hard to believe that all electronic circuits engineers are incompetent. Looking at the effects of out of band signals was fundamental in automotive - it was ALWAYS done. And that could be some pretty bad sounding stuff. At any rate, any engineer who worked in automotive would certainly know to do this. Did none of them ever make it into audio design? And these problems are not so tough to deal with. A simple cap in the right place and of the right type and value and "poof" problem gone.
I don't know about the automotive industry, so who knows, maybe it's better than consumer audiophilia. It's entirely possible. The design teams in consumer electronics are smaller than you'd expect - one to three engineers in nationally famous companies - with marketing staffs several times larger. Sad but true. Worse, the turnover is high, so product continuity is not good.

RF suppression and appropriate test gear is almost non-existent from what I've seen in the industry. It is even more rare in the boutique sector, despite the crazy prices. Those prices are spent on marketing, not research.

The slewing problem for DACs, CD players, and HT receivers only exists in one stage, the I/V converter, or is buried deep within the DAC chip itself, for voltage-mode devices (where it cannot be removed or bypassed). That's the real merit of passive I/V conversion - a resistor and lowpass cap to ground aren't going to slew. But this approach is not that common.

The specs are better with active I/V conversion, and that's what all the application handbooks show. Embarrassingly enough, even $5,000 to $30,000 audiophile products use analog circuits right out of the manufacturers application notes. Real engineering is more rare than we think, which is why so many boutique systems sound so dreadful. I've gone to lots of CES shows and found out the company selling $100,000 systems only has one engineer - or even more upsetting, had one engineer, who left in disgust when he didn't get paid. I am not making this up.

To answer the first point, there are classes of source defects that expose faults in the playback electronics. A CD that uses up the top bit can actually overshoot in the analog electronics in the active-filter section, and clip the electronics. That changes a momentary flat-topping into a much more malign clipping in the op-amp, which in turn saturates the Class AB output section in the op-amp itself.

I've seen the 20 MHz comb spectra coming out of a Burr-Brown PCM 63K converter for myself. It is real. The more HF content the CD has, the more energy there is in that comb spectra (more transitions in a given unit of time, all of which have extremely fast rise times). If the analog electronics are fast enough, or are passive up to the lowpass filter, a "bad" CD merely sounds too bright. With electronics that slew, though, it sounds very distorted, especially with music that has a lot dense upper-mid content, like massed choral works. A few parts changes, poof, problem gone. Many more CDs are then listenable.

How many CD players use passive I/V conversion or ultrafast electronics? Very very few. The problem is simply ignored, since reviewers don't care or even know about it, and it doesn't show up on conventional measurements, so the chip vendors don't care either. Out of sight, out of mind.
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Old 27th September 2009, 10:41 PM   #6230
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Originally Posted by Lynn Olson View Post
Change a 709 externally compensated op-amp for an internally compensated 5532 op-amp, both with identical external current-helper circuits (so they stay in Class A mode, despite current demands from the RIAA-compensation feedback circuit).

Both measured very similarly on the Radford distortion analyzer with 0.0008% distortion residue, with a slight margin, not surprisingly, favoring the 5532 at the highest frequencies. But not a huge difference, really.

Easy to compare with A/B switching. Small subjective differences in "clarity" and all those vague audiophile words, but very obvious differences in surface noise. It took a fast storage scope to see what the glitches on the record surface really looked like - really narrow impulses, definitely fast enough to slew a preamp.
Well that is hardly not changing anything but slew rate. No?

A slew rate of 13v/usec would allow a max output voltage of 41 V. 3V/usec 9.5 V at 50 K Hz. Both of these would be in excess of what a power amp would require for full output, and typically a lot less that would be expected of any phono stage. 3V/usec would allow close to 2V peak output at 250k Hz. I don't doubt that the different opamps sounded different, especially back then, but I would have a hard time believing the slew rate was the issue. As I recall, back in the late 70's, early 80's lots of things were blamed on slew rate limitations and we saw the market flooded to ultra wide band preamps and power amps. Some were very good, some not so good.
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