Old 12-20-2018, 07:35 AM   #1
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Default When does -18 DB matter

How important is it to keep your faders at -18 DB?

Much of this has to do with limits of headroom old analog gear and how the mixbus interacted but often people cite that many plugins are set up for -18 and that their response will be different if you don't provide it at that level.

How true is this and how are you supposed to know if a plug-in needs -18 because the bus certainly doesn't because Reaper doesn't care as it has more hHeadroom that anyone could possibly use?
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Old 12-20-2018, 07:50 AM   #2
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Originally Posted by Coachz View Post
How important is it to keep your faders at -18 DB?
It really only matters for the incoming signal (kind of but not really) or if you want to match the expectation of a preset in a non-linear plugin, and it doesn't really matter then sorta. The original reference to it is because of this...

In the analog world, we obviously can exceed "zero" but in digital, zero is the impassable ceiling. Knowing this, manufacturers need to account for it because if they made analog zero = digital zero, as soon as your analog signal exceeded analog zero, it would be clipping on the digital side. So, they "push down" where analog zero is on the digital scale so that from there to digital zero can cover all that >zero analog signal. With 24 bit converters many manufacturers happen to land "around" -18 dBFS. If you ran a 1k HZ sine wave into your interface line in at unity, it should show up in the DAW around -18 (or the level your sound card's manual says it uses, if it lists it).

This means that if you record and set your analog gear as you always would, there is a high chance it will show up in that -18 dBFS'ish RMS range in the digitally recorded file all by itself, with no need to adjust for that after the fact in the DAW. Once in the DAW, as I stated earlier, if you wanted to know the intention of some amp SIM preset that say's use -18 dB, it's only so that all this matches up and the preset they designed will react similarly to you as when they created it (because the sound of non-linear plugins is based how hard you hit them) - but at the end of the day, treat it like a real amp, turn the knobs until it sounds good.

One could make the point that by "gain staging" all your tracks to ~-18 dbFS RMS you have plenty of room to mix and do things that add gain as you go along with the summing of all the tracks but that's an organizational/workflow thing, not really an audio quality thing.
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Old 12-20-2018, 07:54 AM   #3
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You'll have to check the manual for each plugin, if you're worrying about how emulations are calibrated. Many have variable calibration options anyway, or at the very least an input trim knob. There is no standard for analogue modelling plugins in terms of calibration.

Personally, I don't worry about it at all. They are effects, not actual pieces of gear, so I just twist knobs till it sounds good.
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Old 12-20-2018, 08:00 AM   #4
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You can learn some here:
https://web.archive.org/web/20160102...es/digital.htm
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Old 12-20-2018, 08:28 AM   #5
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Originally Posted by Judders View Post
You'll have to check the manual for each plugin, if you're worrying about how emulations are calibrated. Many have variable calibration options anyway, or at the very least an input trim knob. There is no standard for analogue modelling plugins in terms of calibration.

Personally, I don't worry about it at all. They are effects, not actual pieces of gear, so I just twist knobs till it sounds good.
I've never seen a manual mention it. Do you know any plugs that are calibrated this way?
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Old 12-20-2018, 08:32 AM   #6
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Originally Posted by Coachz View Post
I've never seen a manual mention it. Do you know any plugs that are calibrated this way?
I've seen mention of several but that's it. If it is mentioned, it's often probably for the same reasons I'm discussing - so that there is some common ground between what they heard when designing a preset and what you hear when you try it out. When I buy a real amp, sometimes they will come with these little button position charts with settings and names by them, I'd usually fark with them for ten minutes, then throw it away and turn knobs till it does what I want to hear.

Otherwise, it's like an amp, the strength of the signal hitting the amp can vary wildly based on guitar, passive, active, single, double, pedals that boost, pedals that are a boost - anywhere from say ~50mv to 1,2,3,5 volts (I built and OD that will push out almost 7 clean volts if I ask it to). What do you do then? You plug in your guitar, you turn the knobs until it sounds right.

If we always followed what the recommendation was though, guitar distortion would have never been born.
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Old 12-20-2018, 08:54 AM   #7
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Faders aren't supposed to be at -18. The levels are.

It matters for analog-modeled plugins that use that as a reference level and analog gear that uses that as a reference level, typically things that emulate tubes, transformers, or discrete circuitry....they all behave in ways that changed based on your signal's relationship to a reference level. Being wildly away from it makes them perform wildly different from the designers' intentions.

Pretty much everything else modern uses floating point processing that doesn't care about reference levels.

But, as another consequence of floating point processing, it will never harm your audio to use it. So, when in doubt, you might as well have RMS/VU meters hovering around that level on average and let the peaks go wherever they go (as long as you're not unintentionally clipping a plugin or something).
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Old 12-20-2018, 09:34 AM   #8
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Quote:
Originally Posted by Coachz View Post
I've never seen a manual mention it. Do you know any plugs that are calibrated this way?
Here's one off the top of my head (page 8): https://www.lsraudio.com/demos/VLB525_User_Manual.pdf

There is often a little screw for calibration, or a hidden menu. If the plugin has VU meters then you can simply shoot for an average of 0 on that.

For pure digital plugins it doesn't matter, I don't know of any that aren't floating point now so headroom is of no concern.
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Old 12-20-2018, 09:40 AM   #9
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Oh, and if I'm mixing, I don't care what the levels are on individual channels. All that matters is the level hitting the master. I have Klanghelm VUMT on my monitor fx, when I bring in all the audio, I select all items and bring them down until VUMT doesn't overshoot 0 too much on the loudest parts, when calibrated to 0 dBVU = -18 dBFS, dealing with individual item gain only if relative levels are WAY off.

Obviously in a mix only the total level of all tracks matters, not what individual tracks are at.
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Old 12-20-2018, 03:09 PM   #10
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Quote:
Originally Posted by Coachz View Post
I've never seen a manual mention it. Do you know any plugs that are calibrated this way?
Klanghelm stuff is made to run at -18db. A lot of Waves stuff too.
When looking for that info in manuals it is often reference to as the: Nominal level. Or normal level of operation.
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Old 12-20-2018, 05:14 PM   #11
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Quote:
Klanghelm stuff is made to run at -18db. A lot of Waves stuff too.

oh they do? ok,so perfect: try smashing the inputs way above them levels while maintaining a 32bit recording+playback chain: v 16bit v 24bit chain...results?? sound the same??

⃝hopefully most will soon realize there is very little to fear,or worry about regarding levels using 32bit in+outs..
the only concern after that is 20/24bit interfacing --which will still clip like a beaaatch..
solution-32bit interfacing=no worries=ever.

with 32bit recordings-- the mix of sounds becomes the thing that all need focus on-- wether the mixes sound great @ both very low levels--and--very high levels via speakers.

all plugins,speakers and amps are actually having to work less- by using stronger input+output signals..

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Old 12-20-2018, 06:11 PM   #12
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I don't ignore Bri1 because I appreciate absurdity.

That's also why I want to talk about my JSFX Infernal Diseased Dire Rat. It's a "model" of a modified clone of a Rat style distortion pedal that I have actually built in meatspace. It's not really a part-for-part circuit simulation, but it does use some real part values to determine various things, and more to the point here, all of the internal processing works on "real voltage" numbers.

In order to understand that, you need first to understand how audio signals are represented inside the floating-point audio engine. Basically, 0dbFS, being the limit of a fixed point file, is treated as our unit of measurement in floating point. That is, a centered, symmetrical wave that just hits 0dbFS swings between -1 and 1.

Now, I mentioned above that my converters are calibrated so that 0dbFS = 20db above the nominal level. That nominal level is +4dbu, so the maximum input for those holes is +24dbu. Put that into your favorite calculator and you'll find that it means that it takes an input voltage of +/- 21.78V to hit 0dbFS.

I assume that I'm going to record my guitar through a buffer into a line input at unity gain, so that the recorded value in floating point will be whatever voltage it put out divided by that 21.78. So the first step in the plugin is to multiply it back by 21.78. Now if the guitar puts 1V, the variable that holds that signal level is at 1. If I spit that back out to Reaper as is, it would hit 0dbFS. We're not done with it yet, though.

The next step is a couple of filters, but then we get to the gain element. Now, this one has a somewhat larger maximum gain than stock, a theoretical maximum of almost 3500 times (about 70db) in the pass band. So, with the gain slider all the way up, we're multiplying the signal by about 3500. Now if that input had been 1V, the variable that holds the signal is at 3500. The biggest number that variable can actually hold is something like 6 billion in 32 bit floating point, and exponentially bigger in 64 bit, so even when it's this big, we're nowhere near running out of headroom.

Except, of course, the pedal runs off 9V, and the opamp that does that gain can only get to within about a volt of each rail, so only swings about 7V before it clips off. Inside the pedal, there's a DC bias voltage that makes the signal swing around 4.5V, but in the plugin I set the "opamp" saturation stage's limit at +/- 3.5. Then it goes to the actual diode clipper which is really the same saturation algorithm, but with limit set to about +/- 0.7. That actually is the maximum peak output from a standard Rat pedal, BTW. Doesn't seem like much, a total peak to peak less than 1.5V, but in the world of guitar it actually is pretty big, especially when you figure that it's basically a square wave so that the RMS is about equal to the peak level.

Now if I spit that 0.7 out to Reaper, it hits -3dbFS which is just absurdly loud. I kind of intend this thing to interface with other things that operate on the same basic principle, so I need to divide back down by that same initial factor of 21.78, which comes to 0.032, which is like -30dbFS which seems small, but it's about the same level I'd expect to see if I plugged in the actual pedal to a line input.

BUT if I wanted to, I could make this modular and just leave it in volts between modules. I could pass that +70dbFS signal coming out of the gain section to a properly calibrated saturation plugin and Reaper itself wouldn't bat an eye or clip it in between or anything. I don't currently have anything that does that, and I'm not sure why I ever would, but I definitely could.
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Old 12-21-2018, 05:36 AM   #13
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lolz here-- feel the abrasiveness--like sandpaper
op asked question "when does -18db matter?"
= gave an honest answer--it matters 0 while using 32bit recording+rendering.
why worry about such things???
ooooh i know....because digital plugins and other things were calibrated to ancient analog standards...ooooo i get it now= how silly to not see.

why emulate-when you can innovate.?
>to be original-1 must come with original ideas/concepts/ways of doing---or, just follow like sheep,asleep.
making music is no different-- to be original these days is not an easy tasking!!
it's all been done before__right.?
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Old 12-20-2018, 10:47 AM   #14
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Quote:
Originally Posted by karbomusic View Post
In the analog world, we obviously can exceed "zero" but in digital, zero is the impassable ceiling.
Somebody on another forum wrote something similar. I know that you know what you mean here, and so did he, but I just feel the wording in both cases might be confusing to some people, so I'm copying my reply from that thread. Much of it has been covered above already, but...
Quote:
Originally Posted by ashcat_lt
Please don't confuse 0dBVU with 0dbFS.

In analog, the mark that says 0 is usually the nominal or normal expected operating level. It tells us nothing about the actual limits of the circuit. Most times there is 15-20db above that mark before you hit the rails and start actually clipping. Things often get curvy before that point. Sometimes we like how that sounds.

In digital, the 0 mark is the limit. That's exactly as loud as a fixed point digital file can get, and exactly as far as your converter (ADC or DAC) can get. The digital part is perfectly linear right up to that point and then just suddenly clips off. We usually don't like how that sounds. Sometimes the analog electronics get curvy before the digital part clips off, so sometimes it's not so bad.

This is exactly the reason we have that -18dbFS rule of thumb. Our converters are designed to top out at more or less the same level as the analog gear we attach to it. We said earlier thats usually +15 to +20dbVU, and very often +18dbVU. So the limit of our digital system (0dbFS) is set to the limit of our analog system (+18dbVU) and therefore the nominal analog level of 0dBVU hits digital at -18dbFS. But actually that's just a rule of thumb, and if you want to know how your converter works, you have to look at its specs and usually do some math. They never make it easy. I think my line inputs are calibrated to -20dbFS, but I've seen some with as little as 12db headroom above analog nominal.

Worth mentioning too that the analog VU meters are showing you an average very much like (close enough to) a short-term RMS or LUFS level. It's normal for actual peak levels (the ones that actually matter when we're worried about the limits) to be much higher, and on some sources if you have then right on 0dBVU will still end up distorting or clipping. This confuses a lot of newbs. The digital meters they're looking at are almost always peak levels, but they heard -18dbFS, so they shoot for that and end up quieter than they really need to be. That means nothing in digital really, but it means you're running the analog end of things much lower than it's nominal level and much closer to its noise floor. That's only a problem if it's a problem, but in a lot of cases it can be noticeable. It's complicated quite a bit by the fact that for most of us at home with all in one interfaces, we don't get VU meters. The only meters we have are those in our DAW, which are usually peak based.
But also yes, just turn the knobs til it sounds good. I don't have any preamps worth overdriving, so I just shoot for out of the noise floor but not clipping. In the mix, you just do what you need to do. The way to know if you're hitting a plugin too hard is to listen.

Edit - Actually in some situations where either I don't have a lot of time to dial things in or I don't really trust the source to maintain its level or I feel like I might need to match the recording some other time, I'll just turn my pres all the way down. It's the only place on the knob that I know I can find exactly every time. Well...the other end, too, but that's obviously not appropriate in most cases. Noise is better than clipping most of the time.

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Old 12-20-2018, 10:52 AM   #15
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Somebody on another forum wrote something similar. I know that you know what you mean here, and so did he, but I just feel the wording in both cases might be confusing to some people, so I'm copying my reply from that thread. Much of it has been covered above already, but...
Yep thanks for posting, I'm usually searching for the worded answer that includes some obviousness/aha moment without getting too wordy, then it still seems too wordy for something that is as simple as it sort of is.
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Old 12-20-2018, 11:01 AM   #16
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in reaper-it only matters on which recording formats are used really-- try this.
1.make 3 input tracks-all same input.
2.right click+ set each track to a different recording format: eg: 1x 24bit wav - 1x vbr mp3 - 1x wav adpcm.
3.make your source input almost reach 0db on input meters.
4.add same input fx on each track-eg: reaeq-set to +6db of output gain there,so your input signal 'seems to clip on input levels'. < (try much higher clip levels plz,something ridiculous like +48db)

5.record a small passage--making sure 'auto mute' does not engage while recording....
6. try normalizing each file,then null each file against each other.


^ conclusions,or theories from any other users =??
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Old 12-20-2018, 10:58 AM   #17
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The way to know if you're hitting a plugin too hard is to listen.
Amen.
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Old 12-20-2018, 11:00 AM   #18
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Amen.
Agree with the reverend.
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Old 12-20-2018, 11:02 AM   #19
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If everybody would just read the Yamaha Sound Reinforcement Manual in high school, I'd probably type a lot less.
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Old 12-21-2018, 01:10 PM   #20
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http://geir-music.blogspot.com/2016/...n-staging.html
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Old 12-21-2018, 01:28 PM   #21
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Another example: some Airwindows plugins require you keep the level significantly below 0 dBfs otherwise you'll get obvious (and subjectively ugly) unintended distortion. The most recent one I can think of is Channel6. Someone commented how it's a "distortion box" lol. Meanwhile if used properly it has a subtle effect.

So... -18 dBfs? It might be relevant even in the floating-point domain of your DAW. Or some other level, for that matter.

As for the signal hitting your A/D converters? Yeah analog is still analog. Bri1, you're out to lunch on this. Arguing that impedance doesn't matter with analog gear too? Dude come on. I'm not even getting into this. You're just flat-out wrong and need to do your research.
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Old 12-21-2018, 02:02 PM   #22
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Quote:
Arguing that impedance doesn't matter with analog gear too? Dude come on. I'm not even getting into this. You're just flat-out wrong

lol--who? where? what? when did that get mentioned previous?
waffle.
fwiw- people either _ get it,or,not. =)
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Old 12-21-2018, 06:25 PM   #23
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Quote:
Originally Posted by Bri1 View Post
lol--who? where? what? when did that get mentioned previous?
waffle.
Here:

Quote:
Originally Posted by Bri1 View Post
lolz--why? type more.
that news is ancient history-we are in the now! =)
"
YAMAHA SoUNd REINfORCEMENT HANdbook Part I - Paqe 8-18 It is important for you to know whether the output of a particular piece of equipment is supposed to be matched or bridged, or whether that doesn’t matter. When there is an impedance mismatch (which means the source and load are not right for one another, whether matched or bridged), the results can range from improper frequency response to excess distortion to incorrect operating levels to circuit failure. In terms of specifications, it is important to know what impedances were used when measuring the specs in order for the specs to be reproduc- ible. equipment, many manufacturers have contributed to significant confusion. 8.6. 1.1 Output Impedance EXAMPLE: What is the source impedance of the output in this specification? Output Impedance: 600 ohms If you guessed a 600 ohms,” you may be right. On the other hand, you may be dead wrong! Sometimes, instead of specifying the acutal source impedance of an output, a manufacturer will specify the impedance of the load into which the output is designed to oper- ate. In the above example, it is entirely possible that 600 ohms was not the source impedance, but rather the intended minimum load impedance. © 1987 "

ok m8- you carry on living in 1987--people work with modern tools+equipments..
really,it's >where< we are going though,that matters...that^ matters not!! heh heh heh!
You're mocking the usefulness of this information. It's still useful. If you don't realize that, you're ignorant.

Quote:
Originally Posted by ashcat_lt View Post
To be fair, that particular quoted section regarding matched/bridged is kind of antiquated information. It might be important if you're working with actually ancient gear or maybe "faithful" reproductions thereof. Nothing in the modern world is ever run matched. Everything we ever use (at least this side of the power amp and really usually even then) would rather be bridged from a low source impedance to a higher load impedance. It really is just not a question any more.
I can name a few examples of gear in the room in which I'm currently sitting that sound different and operate to different degrees of efficiency with different impedances. Not to mention tube amps...impedance is a pretty make-or-break thing with that (in more ways than one).

It's true that a lot of gear is more forgiving now. But there's something to be said for understanding this. It's helped me.
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Old 12-21-2018, 08:00 PM   #24
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Brit1,
recordings dynamic range resolution and freq. samplerate are highly dependant (determined) by your hardware - specifically ADC (analogue to digital converter). The most expensive devices get 21~22 bit (useful bits). Most consumer devices will get you around 18~20 bit.

32 bit data stream is ideal for the CPU without conversion to 24. Also better filtering of noise or/and distortion where you can store that data and just trim it off (filter out).
Big disadvantage is the huge file size. Hence longer rendering times.

No matter the bit depth or freq. samplerate, In The Box you will always have a ceiling voltage (which is way lower than any even consumer grade analogue equipment) - that is how computers work (~3V, 5V for USB and and only 12V for powerful consumption of GPUs).
3V or 5V are good enough for ADAC needs - computers do not come with 100W power audio amps built-in. That is for efficiency and low temperature, low Watt consumption.

Mine NI Komplete Audio 6 max. voltage output is +12 dBu, 3.0 V(rms)
Maximum Wattage output is 12mW (milliWatts)

0VU (rms) = +4dBu (is the reference value for professional grade equipment = -18dbFS), which is ~1.228V (rms , not Peaks!)
Peaks can reach even +18dBu () (14dBu above 0VU or in digital is -4dbFS, 14 above -18dBFS). Still have at least -3dbFS headroom for ISP and related dynamic Peak changes.

GPUs are different and require more Watts that probably won't change soon (mine GTX 1060 max. power consumption is 120W and can reach 94°C during heavy work).

Nevertheless even with 32bit won't change the voltage ceiling. In acoustic terms the "ceiling" is around 120dB on average. They say a bit is usually allocated for 6dB (average value for perceived change in loudness in half 1/2 or 50%).

120dB : 6dB/bit = 20 bit
120dB : 3.75dB/bit = 32 bit

Most people will need 0dB to 90dB anyway for Music. 90 / 6 = 15 bit (wow, even lower than the CD standard - mindblown!!!)
90 : 3.75 = 24 bit (mindbanged!!!)

and you need -18dBFS (RMS at 300~350ms time windows) which will assure that even the True Peaks (inter-sample peaks → ISP) are always below 0dBFS with enough headroom for post-processing.

Nothing stops you from pushing the data in digital processing way above that (with 64 bit processing you can translate your sound in the range above "+200dBFS" but that will have to be lowered back at the final stage wen the mix has to go in the DAC).

Reminder: it is nonsense to have positive dbFS values!
Because... well, voltage!
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Old 12-21-2018, 09:37 PM   #25
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I can name a few examples of gear in the room in which I'm currently sitting that sound different and operate to different degrees of efficiency with different impedances. Not to mention tube amps...impedance is a pretty make-or-break thing with that (in more ways than one).

It's true that a lot of gear is more forgiving now. But there's something to be said for understanding this. It's helped me.
Which of your gear actually wants to matched? You're talking about proper bridging, which IS important, but isn't actually touched in that particular quote. A passive guitar wants to be bridged properly to the front end of a tube amp just like any other source/load combination, but the only part of the amp where the user might have to worry about impedance to keep from blowing something up is the speaker connection, and that's the other side of the power amp.


The part of the quote about knowing what the spec is actually saying IS pretty relevant even today, though. We still can't always be sure what they really mean half the time.


Edit so as not to double post (again in this thread): A little bit about practical gain staging and when/why impedance kind of matters.

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Old 12-21-2018, 02:04 PM   #26
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Originally Posted by JamesPeters View Post
Arguing that impedance doesn't matter with analog gear too?
To be fair, that particular quoted section regarding matched/bridged is kind of antiquated information. It might be important if you're working with actually ancient gear or maybe "faithful" reproductions thereof. Nothing in the modern world is ever run matched. Everything we ever use (at least this side of the power amp and really usually even then) would rather be bridged from a low source impedance to a higher load impedance. It really is just not a question any more.
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