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Author Topic: listening memory time when comparing a/b sources?  (Read 11487 times)

Peter Weihe

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Re: listening memory time when comparing a/b sources?
« Reply #105 on: December 28, 2004, 10:32:56 PM »

RKrizman wrote on Wed, 29 December 2004 00:14

s.

So this suggests the question should be asked as to whether the presumed benefits that people are getting from external summing boxes are from fixing a mythological "problem" with DAW summing, or are you simply benefitting from the euphonic analog component of the input circuitry.  In other words, take the Dangerous 2-Bus, which claims to provide extremely clean and transparent summing.  Could it be that its appeal lies in the degree to which it is NOT clean and transparent?  (the subtly of its alleged benefits leads me to think this may be the case)

Exactly. My tests with the Dangerous 2 Bus suggest that the coloration and compression is one part of the euphonic result. ( Page 4 and 7 on this thread) I could demonstrate to the software company that each mix bus in our test caused a specific degradation on the sound even on single signals. So did each converter, and these effects could be found again in the character of the mixes. However there were certain dynamic effects that all analogue mixes had in common, that the Digi mix didn't have although we had found significant differences among the analogue competitors. For instance the Helios did not at all have the harshness on fast attacks or sibilance that the Dangerous 2 Bus had to my ears. It smoothened the edges.  

Quote:

Would it possibly be worthwhile to design a test that isolates these factors?  You can't take direct outs of the D2B, so you can't test it, but other summing boxes and consoles certainly offer the the opportunity to figure out if it's the summing itself or the analog inputs that are providing the "out-of-the-box joy" that everyone is experiencing (or seeking).

We tried to isolate it with the custom made symmetrical summing bus that didn't have any  buffers per channel.
My technician tried to design it as pure as possible with a very stable power supply. The result was that this box was the most dynamic and it had the greatest bandwidth among the analogue mixers in the test. So  the single DA converters outputs went straight into the symmetrical buss. Nevertheless the mix was going more in the direction of the other analogue busses but was the cleanest and most transparent and open one. The classical engineers among the listeners and myself preferred it while two of the pop producers liked the 2 Bus's compression.

After all these tests I learned what kind of objective degradation had a subjectively positive effect on certain sounds and what was more destructive for others to my ears. So my conclusion is to use certain stages for these wanted effects.
I have made some experiments.The Helios glues the tracks together, softens harsh sounds and fattens the bass. I tried to insert 2 channels of the Helios in a group of my PT and  half of the effect was there. Then I played the individual tracks through the desk ,mixed them to two tracks and sent  it back into two PT inputs. Now I got the complete effect for these tracks that I had heard in the mixes. I left the more percussive sounds in the box and I was happy to get best of both worlds. I inserted SSL  bus compressor in another group and it gave the SSL sound .

Later I made your suggested test with only 8 channels of my Helios directs outs feeding the AD converters of my PT 192I/O. Again some part of the analogue mix sound was there, but not all of it.

My test and the discussions on the forum lead me to the following thought.
Each stage that our signals have to pass degrades the sound, some of it we like and sometimes we find it ugly depending on circuit design the sounds, fashion and musical style.
In addition to that analogue summing itself probably causes nonlinear distortion and dynamic effects and the harmonic results may exceed the digital bandwidth and effect the audible range.

But I better leave this rather complicated matter to the scientific department.

Best regards,

Peter.














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Peter Weihe

Peter Weihe

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Re: Analog summing again--a challenge to all proponents
« Reply #106 on: December 28, 2004, 10:45:05 PM »

bobkatz wrote on Sun, 26 December 2004 02:21


My contention is that the Dangerous 2 Buss is being sold on the wrong premise; its coloration or distortion is what people like, not its "superior summing mechanism"!
BK


I find that marketing concept misleading and confusing .


Best,
Peter

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Peter Weihe

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Re: listening memory time when comparing a/b sources?
« Reply #107 on: December 28, 2004, 11:13:57 PM »

Previous poster and late here..

Tone burst comparisons from 10hz to 50KHz is the key to seeing if a piece of equipment is dynamically transparent.


Tone bursts with a -10 solid state tone or multiplicity of odd order tones, as well.

Unfortunantly, this changes in testing, dependent on where it is tested and who is performing said test and the capture device.

Wild cans of fishing bait, properly referred to as "worms"

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lucey

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Re: listening memory time when comparing a/b sources?
« Reply #108 on: December 29, 2004, 12:45:07 AM »

RKrizman wrote on Tue, 28 December 2004 17:52


Brian,
Why don't you grab one from Fletcher and check it out.  I'm guessing you own those preamps already.

-R


a) Fletcher doesn't want my money
b) I dont really need a Folcrom, I use a console
c) I'm too lazy right now ... I was just fantasizing that someone else would do it!
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bobkatz

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Re: listening memory time when comparing a/b sources?
« Reply #109 on: December 29, 2004, 12:04:19 PM »

Thanks, Rick! An excellent summary (analog summing?) of the issues.


I have a digital 4 into 2 "summing box" from Graham Patten Systems. It contains TWO AES/EBU inputs and one AES/EBU output. Put any stereo source into input 1 of up to 24 bit length and it comes out as a perfect clone of the source. Same for input 2. That's why this is a SUMMING box. Addition... get it? The bottom line is that as long as the sum of the two sources does not exceed full scale, the output of this box is a PERFECT, underline PERFECT mix of the two sources.

You can use it to combine a dry mix with the wet output of a reverb, and if the sum does not exceed full scale, you won't have to add dither, either. The only time you have to add dither is if you are truncating wordlength OR if you performed a level change, which this box does not do.

BK
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maxdimario

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Re: Analog summing again--a challenge to all proponents
« Reply #110 on: December 31, 2004, 11:48:10 AM »

Peter Weihe wrote on Sun, 26 December 2004 00:49

Further investigations.

The clear, bright modern IC designs like the 2 Bus and the Manni bus sounded a bit more synthetic on strings to our ears  than the Helios. The Helios on the other hand was not as crisp as the Manni buss.
I tested whether a simple run of the Manni mix and the 2 Bus mix through the Helios could bring back the charming sound of the strings .However it was the same as in my stereo test, that Bob Katz had suggested, it just softened the edges but the synthetic aspect stayed.
Obviously these modern IC design busses added that light harshness that we interpreted as being synthetical and the Helios couldn't  make up for it afterwards.
In other words a Mackie mix bus will not sound like a vintage Neve just by inserting two 1081 in the signal path.
You can't get rid of the artifacts.
My dream is either a perfect digital emulation of different classical summing busses plus some world class converters for inserting classical analogue gear, or a transparent, open neutral sounding analogue mixer with inserts for classical analogue gear.
After all my tests with single sources I guess that  the GML 9100 mixer probably is the perfect tool for this task.



I believe that when people (including myself years ago) listen to different audio circuits and hear a different sound, they naturally believe that it is the added distortion that 'creates' the sound.

Guitar players know that a tube amp, especially a simple 50's or 60's tube guitar amp feels different than playing through a transistor or op-amp based amp.

In a good guitar amp the sound will 'jump off' the pick and the end result is that the player feels 'connected' to the sound.

Plugging a guitar into a transistor or op-amp based circuit feels sluggish and tight.  The feeling of the music (which is mostly based on minute timing differences) gets lost.

Peter will know what it feels like to record or play through a digital guitar effect/amp emulator. Right Peter??

although distortion can 'add' to the sound, or modify it in an artistically pleasant way I believe that a very big issue is what DOESN'T get lost in the audio path.

The big problem with digital from a listener's point of view is that, although digital does not add substantial amounts of distortion, it does DETRACT a fundamental part of the sound which is what gives us listeners the 'feel' and definition of the recorded sounds.

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Bill Mueller

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Re: listening memory time when comparing a/b sources?
« Reply #111 on: January 01, 2005, 12:21:01 PM »

ted nightshade wrote on Fri, 03 December 2004 10:21

Digitally emulating analog summing is a laudable goal- but I would be much more interested in digitally emulating acoustic summing in the air.

Can it be that digital summing is "more perfect" than summing in real acoustic space? Are there really anomalies in summing in the air, which sounds so very different than digital summing? There's no crosstalk or harmonic distortion in the air, I don't think...


Real acoustic summing in air implies quite a lot of variables.

I have to imagine that your are speaking about the human experience of listening to multiple musicians playing live, not standing ten feet in front of a single point, full bandwidth source, listening to a group of musicians processed and summed through microphones, amplifiers and speakers.

In the case of the live experience, you have multiple moving sources, multiple pathways (even from each source) summing with different phase interaction at multiple points between the players and then between the players and the listener (crosstalk). You have all of the environmental conditions (under Haas and beyond Haas) as well as structure born transfer and radiation that occur at different speeds than air.

The rate of transfer (time domain distortion if you will) changes, depending on temperture, humidity and air pressure (1130 fps at 72degrees and sea level). Increasing the air pressure, (closer to sea level), increases the effeciency, speed and bandwidth (to a limit of course). Increase the temperture and these things increase as well, but with a different mechanism, (the air molecules contain more energy).

These conditions also change the frequency response dramatically (harmonic distortion). Air transmits low frequencies much more effeciently than high frequencies.

Directionality varies as well by temperture related refractive effects.

These effects are easy to discern at an open air concert when the distance from the source is greater than 100', however they occure in the nearfield as well.

I would say that digital summing is extremely simple when compared with acoustic summing. Only in the last few years have computers become powerful enough to predict accurate room responses for anything other than a cube. That is why creating digital reverb algorithms is such an art.

When you think about it, we and those before us, have been attempting to simulate acoustic summing for over one hundred years.

Best Regards,

Bill

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bobkatz

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Re: listening memory time when comparing a/b sources?
« Reply #112 on: January 01, 2005, 06:05:45 PM »

Bill Mueller wrote on Sat, 01 January 2005 12:21




When you think about it, we and those before us, have been attempting to simulate acoustic summing for over one hundred years.





I think the whole issue of "acoustic summing" is a red herring. What we have to become aware of is how the ears react to the reflections of the environment around the musician and how to accurately capture them. Even SIMPLE approaches in terms of recording musicians in stereo with stereo pairs to capture a good-sounding environment in two channels at a time produce far superior results compared to multi-mono power panning. Once you learn how to do this, then the issues of "electrical" versus "acoustic" summing become irrelevant to the task at hand of creating the illusion of a natural-sounding depth environment.

BK
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