What is responsible for soundstage in a preamp design?

[Chrisfromthepast]

When mixing in ear monitors, I feel bad for the singers because the headphone output from the desk always sounds better than the wireless packs.
Even if you optimize your gain structure to stay out of the companding regions of the PSM1000’s there is a restricted bandwidth and reduced dynamic range that I just can’t get past.
Drummers usually insist on a hardwired headphone amp because the maximum dynamic range is essential.
Im sure the difference is measurable in a number of distinct parameters that Shure is well aware of.
I never really thought of it as depth, but playback sources sound different, and this definitively warrants a listen to the lead singer’s mix as it comes out of her pack rather than the desk.
Anyway, I would posit that the OP’s MOTU 4 has a preamp or converters that do not accurately represent the signal, and they should use a signal path that brought them joy in the past. It could be neutral of have a bunch of even harmonics, that don’t matter. The subjective reference for how something “should” sound for a given song/ performance/ performer is an essential reference point for any engineer. Unfortunately it only comes with critical listening experience.
“This kick mic sounds one legged” lacks depth for sure

 

[Lampie519]

Just pick something you like for the job you are doing...xformer hysteresis and power supply lag...rail voltages...an ic at 5vdc or tube at 250+vdc...time for audio signal to get thru the circuit could all apply...if it sounds great...use it...if it sounds like crap...don't
PLENTY OF USEFUL INFO ON THIS SITE !!!

That is all fine but the question remains for someone who has not yet tried all options. It was observed in the initial entry that the different amps had different “sound stages” depending on topology, not on mic placement or effects used. If audio frequency is DC then no high frequency capacitors are needed in the feedback loops are they? These are there to prevent amps from oscillating and that is because of phase shift errors. Opamps btw are getting better and faster so these are no more required but in case of a feedback loop that includes a transformer these are needed still.
Funny that these simple things are overlooked here…,

 

[svyet]

Sure, you can do that with compression, by changing the balance between direct sound and reverb tails, and you can do that with eq, by increasing or decreasing the presence region around a couple to a few kHz. I don't know that I would describe a source being statically moved slightly forward or slightly back as "3D," which speaks to the point of some of the earlier posts that the terminology is evocative, but not well defined. Whether the thought that the term evokes in my mind is the same as the thought in your mind, or the OP mind is difficult if not impossible to tell without some actual recordings to refer to.
I suppose it is possible that some preamp designs have some kind of amplitude compression or frequency response change that varies with time or signal level, but I don't have any that do, so I can only speculate that it might be possible in theory.

A quick stream of consciousness on the topic:

I'm also inclined to taking a cue from compressors in general (whether hardware or hardware emulations), and even moreso with compressors that feature stereo and dual mono modes. Maybe the sense of "3D-ness" (or more simply "dimension") the OP is talking about could be related more simply to exaggeration and non-linearities (i.e. inability of analog circuitry to reproduce sound exactly the same at any given moment due to tons of physical factors). One example of exaggeration would be when compressing drum overheads for example..all of a sudden some of the finer and quieter sounds (cymbals and room sound in particular) are brought up and exaggerated in a way that adds "space" to the overall sound. Non-linearities in hardware circuitry compared to the linearity of a typical digital signal can also add to a sense of "space" much like the non-linearities of a room (reflection time, etc.). I think of when switching between stereo and dual mono mode on a compressor, it can exaggerate (or collapse) that sense of space due to difference in processing the left and right channels. It's not being done in a clean, clear, and clinical way, and often it sounds great for that very reason. Much like saturation is ultimately distortion (correct?), perhaps the sense of dimension is nothing more than the combined non-linearities that our ears are familiar with in the real world (or they themselves produce), as opposed to a clinical and linear reproduction of a sound lacking in the same exaggerations and non-linearities of the real world in real time. That could also explain a higher quality microphone's ability to produce (and perhaps even exaggerate) a sound much closer to what the human ear can perceive and is familiar with. Applied to preamp design, I think it's the same principle. Different designs have different exaggerations and non-linearities that contribute to the perceived "dimension," "space" or "3D-ness." I would assume that since stereo is ultimately a mono Left channel and mono Right channel, it applies to mono in principle. It's of course more noticeable in stereo since we have two channels of different exaggerations and non-linearities at play instead of just one channel with non-linearities. But if listening to a mono source processed through analog gear or emulation and playing through both channels, the exaggerations and non-linearities could still be perceived when compared to a more clinical signal. I think that all makes sense..

 

[emrr]

The arguments against feedback seem to come from a lack of understanding around the related changes made to every other aspect of design, layout, and parts specification. In the transition from none to some it allowed many things to be done more cheaply and more sloppily, since it’s a simple band-aid that can mask many a head wound. So thank capitalism and profit seeking? It’s pretty much been figured out since that transition. Don’t forget that most professional preamps post-WWII use in excess of 20dB NFB loop, somewhere. Much of that equipment is not faulted in any way by those faulting NFB for the destruction of soundstage. Basic lack of holistic comprehension. Now, I do love a preamp lacking NFB, for creative reasons, and also one with lots….for creative reasons. Either approach can also be totally F’ed up.

 

[JohnRoberts]

Back in the 80s when I was still writing my "Audio Mythology" column for RE/P magazine, in one issue's column I published a joke glossary of audiophile terms with some amusing definitions.

JR

 

[Newmarket]

Back in the 80s when I was still writing my "Audio Mythology" column for RE/P magazine, in one issue's column I published a joke glossary of audiophile terms with some amusing definitions.

JR

Okay. We definitely don't want to see that

 

[emrr]

That's the thing with weeds, you can kill them only so long as you stay on the job.

 

[JohnRoberts]

Okay. We definitely don't want to see that

I'm sure its around the WWW somewhere... As I recall I didn't make it up but Richard Elen (Editor of Studio Sound) sent it to me, and I published it with his permission.

A lot of those audiophool terms write their own jokes...

JR

 

[Flatpicker]

Guess I'll throw my hat in here...

Back when I had more time on my hands I spent a considerable while recording short identical audio clips of acoustic stringed instruments and vocals on separate tracks using different preamps and then compare the recordings back to back.

When comparing an API-ish preamp to a Jensen-ish preamp (the main difference was steel lam xfmer vs. high nickel lam xformer) I noticed that I could hear much more of the room (i.e. - what I call 3-D) with the Jensen, whereas the API circuit seemed to strip away the room and sound more 'up front' like in an isolated booth. Now keep in mind this was in my bonus room with roof-pitch walls and no acoustic treatment other than a couch and carpeted floor.

Later after building a fully acoustic treated studio in my basement, I repeated the test and this time the Jensen circuit sounded almost as 'up-front' as the API circuit. I guess the moral of my story could be that high nickel picks up more room than steel (more sensitive?), but it starts to become a moot point when the room is properly treated.

And speaking of which, for those of you who don't know, room treatment changes EVERYTHING.

 

[abbey road d enfer]

Funny that these simple things are overlooked here…,

We're probably too thick...

 

[Lampie519]

Don’t forget that most professional preamps post-WWII use in excess of 20dB NFB loop

I own some of the old German V7x modules and if you take a closer look you will see that the fredback loops do not include the transformers and never beyond 2 stages. Distortion can be minimized by connecting 2 stages with similar distortion therefor cancelling some of these. This is a very old trick done even before feedback was invented and works similar to a balanced or push pull stage where some harmonics get canceled.
You should be able to measure the “propagation” delay by using a differential input of your oscilliscope setting the gain correctly and see the “spikes” of the delayed waveform.

 

[svyet]

Guess I'll throw my hat in here...

Back when I had more time on my hands I spent a considerable while recording short identical audio clips of acoustic stringed instruments and vocals on separate tracks using different preamps and then compare the recordings back to back.

When comparing an API-ish preamp to a Jensen-ish preamp (the main difference was steel lam xfmer vs. high nickel lam xformer) I noticed that I could hear much more of the room (i.e. - what I call 3-D) with the Jensen, whereas the API circuit seemed to strip away the room and sound more 'up front' like in an isolated booth. Now keep in mind this was in my bonus room with roof-pitch walls and no acoustic treatment other than a couch and carpeted floor.

Later after building a fully acoustic treated studio in my basement, I repeated the test and this time the Jensen circuit sounded almost as 'up-front' as the API circuit. I guess the moral of my story could be that high nickel picks up more room than steel (more sensitive?), but it starts to become a moot point when the room is properly treated.

And speaking of which, for those of you who don't know, room treatment changes EVERYTHING.

I think this goes hand in hand with what I was saying about how compression also brings up the room and therefore exaggerates the dimensional qualities of the source track - if in fact those dimensional qualities already exist. Paired with the non-linearities of hardware (or hardware emulations) - and particularly in stereo - those dimensional qualities become even more exaggerated by little inconsistencies and offsets here and there in how the source is perceived, and probably more closely represents and exaggerates what we hear in real life particularly in the form of reflections, decay, etc.

It's possible however that the OP might be referring to that sense of 3-D ness even after the room as been eliminated. I would still chalk that up to the non-linear behavior of hardware (or hardware emulations). Similar to the idea that two identical hardware preamps may not sound identical, and maybe contribute to added "dimension" even in mono compared to a more linear mono source. Of course then its exaggerated when two or more tracks are then added. I think this is the idea behind something like Plugin Alliance's TMT, where each channel of TMT is different from the next and can give very quick results by emulating the non-linearity of most hardware. I think it's pretty neat to have that extra layer of non-linearity when mixing, and to see how it can definitely add just a little extra something to individual tracks and a mix.

 

[gyraf]

I noticed that I could hear much more of the room (i.e. - what I call 3-D) with the Jensen, whereas the API circuit seemed to strip away the room and sound more 'up front'

What you hear here is the Barkhausen quantisazion or -distortion. It's simply a threshold function determining the minimum-energy-level transferable because of ferromagnetic "cell" or crystal size, and the related energy content required to flip these. The better magnetic conductivity, the smaller crystals and the longer the "tails" of natural-room reverb..

If you want to hear something extreme in this direction, try a nano-crystalline or amorf-core mic stepup transformer - to me that was an almost scary experience

/Jakob E.

 

[svyet]

What you hear here is the Barkhausen quantisazion or -distortion. It's simply a threshold function determining the minimum-energy-level transferable because of ferromagnetic "cell" or crystal size, and the related energy content required to flip these. The better magnetic conductivity, the smaller crystals and the longer the "tails" of natural-room reverb..

If you want to hear something extreme in this direction, try a nano-crystalline or amorf-core mic stepup transformer - to me that was an almost scary experience

/Jakob E.

Interesting. Could this be one or maybe "the" source of what the OP (and some of the rest of us) might be referring to?

 

[Newmarket]

Guess I'll throw my hat in here...

Back when I had more time on my hands I spent a considerable while recording short identical audio clips of acoustic stringed instruments and vocals on separate tracks using different preamps and then compare the recordings back to back.

When comparing an API-ish preamp to a Jensen-ish preamp (the main difference was steel lam xfmer vs. high nickel lam xformer) I noticed that I could hear much more of the room (i.e. - what I call 3-D) with the Jensen, whereas the API circuit seemed to strip away the room and sound more 'up front' like in an isolated booth. Now keep in mind this was in my bonus room with roof-pitch walls and no acoustic treatment other than a couch and carpeted floor.

Later after building a fully acoustic treated studio in my basement, I repeated the test and this time the Jensen circuit sounded almost as 'up-front' as the API circuit. I guess the moral of my story could be that high nickel picks up more room than steel (more sensitive?), but it starts to become a moot point when the room is properly treated.

And speaking of which, for those of you who don't know, room treatment changes EVERYTHING.

Indeed it does. As for the differences heard I'll suggest a likely cause might be different impedances seen by the mic. Altering mic characteristics. Depends on the mic obvs. What mic was it ? Disclaimer: I'm not that much of a mic expert beyond the basics.

 

[freshtapescent]

@OP may have been better served with a link to a double-blind ABX tester. Any of us can speculate what OP’s perception might be picking up on and the answer might be in the thread somewhere…?

Let OP upload two files, describe them in any way they want (one’s gonna be 3D, one’s gonna be 2D) and then we can tell him what’s different about them empirically instead of creating an exhaustive list of ‘who the hell knows’ and chasing our tails as OP sighs and slinks away.

 

[emrr]

What you hear here is the Barkhausen quantisazion or -distortion. It's simply a threshold function determining the minimum-energy-level transferable because of ferromagnetic "cell" or crystal size, and the related energy content required to flip these. The better magnetic conductivity, the smaller crystals and the longer the "tails" of natural-room reverb..

/Jakob E.

I have some 1970's RCA preamps that are essentially API's (I think what people call Huntington 2520's) but with UTC 'A' series 80% nickel input transformers. I set them up with some 50% nickel Jensen output transformers. They give a very different impression from the steel API/CAPI transformers.

Another twist in the recipe is that many of the transformers used in preamps 50+ years ago were multi-purpose mic/line types that could take higher levels (UTC A series +15dBm for instance), that interacts differently with a low level mic signal than something like a Jensen spec'd at +2dB max.

 

[user 37518]

I own some of the old German V7x modules and if you take a closer look you will see that the fredback loops do not include the transformers and never beyond 2 stages. Distortion can be minimized by connecting 2 stages with similar distortion therefor cancelling some of these. This is a very old trick done even before feedback was invented and works similar to a balanced or push pull stage where some harmonics get canceled.
You should be able to measure the “propagation” delay by using a differential input of your oscilliscope setting the gain correctly and see the “spikes” of the delayed waveform.

Can you let us know the delay figures you've encountered and the op-amps being used? The question is not whether there is delay, the question is whether the delay is significant. What you argue for has been debunked many, many times since maybe the late 80s? The only ones still publicly making an argument for TIM are some audiophools and quacks, they are also the type of persons who prefer directional cables and such.

 

[abbey road d enfer]

Indeed it does. As for the differences heard I'll suggest a likely cause might be different impedances seen by the mic. Altering mic characteristics.

The effects of mic loading have been known since long. When RCA introduced their ribbon mics, they specified using an input stage with an unloaded-secondary transformer, which results in tilting the frequency respons upwards. Of course it enhances the HF contents of ambient sound, which makes it more or less pregnant.

 

[Lampie519]

the question is whether the delay is significant

Yes, i think it is but i can’t back that up with actual numbers unfortunately.
Direction of cables i don’t think matter even if it can be heard initially, but over time this can change. Not exactly sure why but that it does is very audible.
Same with electrolytic capacitors, they need time to settle so A/B comparison is difficult.