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chasman
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#1
26th October 2007
Old 26th October 2007
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gain structure????

hi guys,

am looking for any and all help I can get on understanding gain structure.

have come to believe it is super important.

and yet for all the reading I've done, I think I am missing the boat big time on understanding what it's all about. and I have a feeling that if I could get at least some clarity, it would be a huge benefit.

sincerely,
Chas (as in multiple cha's......for instance the dance, the cha-cha)
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26th October 2007
Old 26th October 2007
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Most of the technical stuff is taken care of by the device makers - but it still pays to have a basic understanding of the different connecter types (TS, TRS, XLR etc), and the implications of impedance and voltage and balanced/unbalanced.

The first thing to get right is to make sure you are connecting the devices up correctly. That's a big subject in itself. For example, a low-z balanced line-level signal maybe terminated with XLR or 1/4" TRS - it doesn't really matter. But there is a huge difference between a 1/4" TRS being used for an insert point, or for a stereo headphone. Same connector, different applications.

But assuming you have bought off-the-shelf gear and connected it according to the handbook, a lot of these issues are taken care of by the device maker. Which leaves the gain structure issue.

The shortest explaination is: sweet spot. Analog devices have a range of extremes that go from too low & noisy, through to too hot and distorted. The sweet spot is somewhere in the middle. And sometimes distortion is desirable, so it's pretty much defined by what your ears hear.

The idea is to set everything so it's inputs and outputs are operating in their sweet spot - which can sometimes be a balancing act. You want to avoid unnecessary boosting of gain, followed by reductions in gain. Unless the abuse in intentional for a desirable distortion effect.

Get to know your boxes - play with the knobs, and learn what it sounds like when you abuse them, and what happens when you back off the gain. The only rule is that you like what you are hearing, and aren't generating excessive noise.

Meters and lights can be useful, but learn to trust your ears.
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chasman
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26th October 2007
Old 26th October 2007
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thank you kiwiburger.

what a fantastic explanation!!!
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26th October 2007
Old 26th October 2007
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I've been learning a lot about this stuff too.

Somebody please correct me if I'm wrong.

For the record, I'm purely OTB and I record to 1/2" tape. That's my setup.

Knowing the difference between mic level (-10db) and line level (+4db) is a
good start. I used to think that XLR and TRS meant one or the other. That's not necessarily true, only at the LINE-in input stage of the mixer versus the MIC-input. You want to use LINE-ins because you want whatever's going into you mixer and giving you a master level to read -Zero- at +4db. That's even keel.

It's kind of confusing but that's my understanding of A to B. That and you don't want to ride the gain heavily on any one device, just like you don't want all your jelly on one part of your toast. Even it out.

Also make sure your line-in 1/4" cables are TRS balanced, not mono instrument cables. Mono cables are good, but balanced as I've been told is key to getting a cleaner sound inside the control room.

Having enough preamp channels doesn't hurt either. I don't feel comfortable starting anything without at least 4.
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26th October 2007
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mic level is -50 -60 no?
#6
26th October 2007
Old 26th October 2007
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Quote:
Originally Posted by soupking View Post
I've been learning a lot about this stuff too.

Somebody please correct me if I'm wrong.

For the record, I'm purely OTB and I record to 1/2" tape. That's my setup.

Knowing the difference between mic level (-10db) and line level (+4db) is a
good start. I used to think that XLR and TRS meant one or the other. That's not necessarily true, only at the LINE-in input stage of the mixer versus the MIC-input. You want to use LINE-ins because you want whatever's going into you mixer and giving you a master level to read -Zero- at +4db. That's even keel.

It's kind of confusing but that's my understanding of A to B. That and you don't want to ride the gain heavily on any one device, just like you don't want all your jelly on one part of your toast. Even it out.

Also make sure your line-in 1/4" cables are TRS balanced, not mono instrument cables. Mono cables are good, but balanced as I've been told is key to getting a cleaner sound inside the control room.

Having enough preamp channels doesn't hurt either. I don't feel comfortable starting anything without at least 4.

-10dbu (unbalanced) is not mic level and +4dbu is not ALL lines level.

actually a balanced cable doesnt give u a cleaner signal, it cancels noise that may have been acquired through contact with electrical sources (distance), balanced cables have differential amplifiers in the signal path, which changes signal amplitude (so it MODIFIES THE SIGNAL, THAT DOESNT MAKE A SIGNAL CLEANER). in the process of balancing a signal u can have more signal percentage loss vs unbalanced cabling depending in the quality of the components, signal loss is a fact in every type of cable,
balanced cables (in my opinion) are meant for long distances (that s why they use differential amps) balanced cables are better for short distances its a matter of the right choice for the needed aplication.

balanced cables: good for long distances (live sound) no noise (unless ground loops)

unbalanced cables: good for short distances (patching) (clean signal), but likely to acquire noise from electrical sources through distance.

i only use balanced cables in the control where i know i might have problems.

regarding gain structure, matching impedances are very important remember that if your input signal impedance is not 10 times over than your source signal impedance you ll increase your signal percentage loss.

signal/noise ratio is also very important.

sorry for el bad english
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26th October 2007
Old 26th October 2007
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Quote:
Originally Posted by analog modeling View Post
-actually a balanced cable doesnt give u a cleaner signal, it cancels noise that may have been acquired through contact with electrical sources (distance), balanced cables have differential amplifiers in the signal path, which changes signal amplitude (so it MODIFIES THE SIGNAL, THAT DOESNT MAKE A SIGNAL CLEANER).
actually a balanced signal uses 2 conductors instead of 1 . one conductor is charged positve and the other negative, the signal is carried on the positive. the postive and negative conductors both however carry the noise/hiss but being oppositely charged cancel each other out leaving only the positive signal thus resulting in a cleaner signal. in reality the noise is still there we just dont hear it, the signal itself is not modifiedcheersdave
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26th October 2007
Old 26th October 2007
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Quote:
Originally Posted by Weird Oddio View Post
actually a balanced signal uses 2 conductors instead of 1 . one conductor is charged positve and the other negative, the signal is carried on the positive. the postive and negative conductors both however carry the noise/hiss but being oppositely charged cancel each other out leaving only the positive signal thus resulting in a cleaner signal. in reality the noise is still there we just dont hear it, the signal itself is not modifiedcheersdave
neveer said a balanced cable used 1 conductor, if it were that way u couldnt cancel noise, pin2 carries the audio signal and noise, pin 3 carries the signal (this case noise) with an inversion of 180 degrees, my point is that the signal (pin2) is modified by a diff amp (that s why a balanced cable s signal has more amplitude than an unbalanced) electrical components are not totally accurate when processing the signal (in most cases its not exactly 180 degrees), when the components are not very good or defective (or bad soldered (welded?)) the timing between the 2 signals is not the same (among other things).

sorry for el english
#9
26th October 2007
Old 26th October 2007
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There's another piece of gain structure. Let's say you've got several devices chained together. Pre-comp-EQ-line in-fader-recorder. The gain of each unit affect the performance of everything downstream. The same applies to a stack of plugs in the virtual world.
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26th October 2007
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Quote:
Originally Posted by Weird Oddio View Post
one conductor is charged positve and the other negative, the signal is carried on the positive.
not to rip on you but you seem to have some confusion here... the cables aren't "charged" ... the literally carry the same signal except one is inverted... since the noise that the cable picks up is the same on both wires.... when the two signals are later combined with the inverted reinverted (? if you catch me) the noise on the reinverted line is now 180 degrees out from the original and subsequently cancells... the summing is done at the op-amp stage and makes use of whats called common mode rejection ratio... way cool trick... and it's the reason a mic run can be from stage to FOH...
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chasman
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26th October 2007
Old 26th October 2007
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thanks very much gents.

you are all very generous.
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26th October 2007
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Let your ears decide ...other stuff doesn't much matter.
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27th October 2007
Old 27th October 2007
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Quote:
Originally Posted by analog modeling View Post
-10dbu (unbalanced) is not mic level and +4dbu is not ALL lines level.

actually a balanced cable doesnt give u a cleaner signal, it cancels noise that may have been acquired through contact with electrical sources (distance), balanced cables have differential amplifiers in the signal path, which changes signal amplitude (so it MODIFIES THE SIGNAL, THAT DOESNT MAKE A SIGNAL CLEANER). in the process of balancing a signal u can have more signal percentage loss vs unbalanced cabling depending in the quality of the components, signal loss is a fact in every type of cable,
balanced cables (in my opinion) are meant for long distances (that s why they use differential amps) balanced cables are better for short distances its a matter of the right choice for the needed aplication.

balanced cables: good for long distances (live sound) no noise (unless ground loops)

unbalanced cables: good for short distances (patching) (clean signal), but likely to acquire noise from electrical sources through distance.

i only use balanced cables in the control where i know i might have problems.

regarding gain structure, matching impedances are very important remember that if your input signal impedance is not 10 times over than your source signal impedance you ll increase your signal percentage loss.

signal/noise ratio is also very important.

sorry for el bad english
Okay, now I want to know more about gain staging.

So how does one measure impedence if he/she's supposed to match it?

So mic levels not coming in at -10db? It's my understanding that at least unbalanced connectors are supposed to provide that. Mic is -50, 60?

And if so, what does that have in connection with getting the signal to line level and matching impedance? Like okay, we've got A, what's B, in order to get to +4?

Know what I'm asking? Anyone? Anyone?
#14
27th October 2007
Old 27th October 2007
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I think the concept that helped me the most was understanding that you never want to let an audio signal drop beneath the level that it entered the system at.
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#15
27th October 2007
Old 27th October 2007
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Quote:
Originally Posted by soupking View Post
So mic levels not coming in at -10db? It's my understanding that at least unbalanced connectors are supposed to provide that. Mic is -50, 60?
No, no and yes as a general rule

Kind regards

Peter
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27th October 2007
Old 27th October 2007
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Quote:
Originally Posted by Bob Olhsson View Post
I think the concept that helped me the most was understanding that you never want to let an audio signal drop beneath the level that it entered the system at.

That's not a bad rule of thumb. Because every active electrical circuit you pass your signal through will add noise. So unless you are getting some gain from the circuit, you are increasing the noise without increasing the signal - i.e. the signal to noise ratio is getting worse.

The obvious exceptions are DI boxes and Reamp boxes. A typical DI box takes an Instrument signal, and reduces it down to Mic level. A typical reamp box takes Line level and reduces it down to Instrument level. I tend to prefer passive boxes for this, because a passive (transformer) DI or reamp doesn't add noise.

But sometimes the slight noise that active circuits add is compensated for by other benefits - e.g. buffering, impedance matching, or just the tone or harmonic distortion an active device can bring to the party.
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27th October 2007
Old 27th October 2007
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thanks alot.

can you advise me about the knobs on my Yorkville 250 watt powered mixer???

lets say I use only one channel. I plug a Pearlman Tm2 mic into a Pacifica mic pre and then plug that into channel one on the mixer.

the Pacifica has a gain knob (and a pad switch -20 dB). the Yorkville mixer has a Channel level knob that goes from 1 to 10. the Yorkville also has a Main Master knob that goes from 1 to 10.

how would you set the gain on the Pacifica and the knobs on the Yorkville??

one HUGE thing that I don't get here is how the Main Master knob differs from the Channel level knob on the Yorkville powered mixer????

Kiwiburger, can you help me out here?
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27th October 2007
Old 27th October 2007
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I don't have any of that gear, but similar stuff (Neve, API, Mackies which I only use for monitoring).

I like to work from the beginning to the end - so start with the mic end. Depending on the source, you may or may not need to use the pad switch (either on the mic, if it has one, or on the preamp).

Generally, you don't want to use more gain than you need to - because excessive gain increases noise.

I only see gain knobs on the Pacifica. (I can't tell from the photo if these are switches for different transformer taps, like on a Neve - or continuous pots, like on an API - either way, that's all you appear to have, so it's really easy).

I don't see any metering on the Pacifica - and frankly, metering can be misleading. Experiment to see what happens if you overload the input, or if you crank up too much gain. It should become obvious how far you can push it.

But be aware that the output of the Pacifica could be overloading the next stage, which I presume is your Yorkman powered mixer.

At this stage, I would question whether the Yorkman even needs to be in the circuit at all. What is it adding? Apart from noise (which all active circuits certainly add).

Is this for recording into a DAW? What is your A/D converter? I would normally connect straight from the preamp into my A/D, and directly to the DAW. IF I needed monitoring, I would take a spare output from the preamp and send that to my monitoring mixer, but avoid tracking the mixer sound. (Because in my cause my Mackies do too much damage to a quality signal).

But if the mixer must be inserted, make sure the mixer input is set to Line level, not Mic.

Power mixers are generally a compromise for simple live sound applications. Not normally used for recording.
#19
27th October 2007
Old 27th October 2007
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some pretty good explanations here! i just want to add that as long as you are in the analog world, keep an eye (use your ears too) on your meters; keep them all aligned to 0dBVU wherever you have an analog VU or Led meter on your gear. Since 0dBVU is normalled to +4dBu, you are modulating 100% (remember you have some numbers going like 20-40-60-80-100 on your VUmeters wich are not dBs??!) accoring to the specifications of this normaling. wich means you have the healthies signal.
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27th October 2007
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Quote:
one HUGE thing that I don't get here is how the Main Master knob differs from the Channel level knob on the Yorkville powered mixer????
To help your understanding - every amp, whether it's a preamp or a power amp, or a guitar amp or whatever - has an input and an output.

An amp may, or may not, provide seperate gain or attenuation controls for the input, and/or the output. They are very seperate things.

For example - you can overload an input. That can cause clipping or distortion.

Or you can feed it with a signal that is too weak. That requires further gain upstream, which will also boost any noise that is generated by the input stage.

A mixer, by definition, has a bunch of input stages that are connected to a least one output stage. Each channel usually has a trim pot, which often ranges from Mic level (lots of gain) to Line level (no gain). Set this first - listen for noise at one extreme, or distortion at the other. Mic and Line levels can vary between devices - don't just assume.

Ideally the channel fader should sit around 0dB (unity gain) when in normal use - obviously musical considerations might require a lower setting.

Nothing is rigidly set in stone - that's why you are given knobs in the first place. There are applications for just about any knob position - choosing the best settings is an acquired skill.
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27th October 2007
Old 27th October 2007
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thanks Kiwiburger.

when I record into my Yamaha aw1600 professional audio workstation, I go right in from the pacifica. (no yorkville powered mixer. I monitor with headphones.)

I was wondering about singing thru the yorkville powered mixer for kicks (no yamaha recorder). any idea how the Channel level knob differs from the Main Master knob????


ps: there is a little more to this question. I have an old tubeworks mosvalve 100 watt guitar amplifier that has 2 channels....... clean and fuzz. that said, this thing actually has 3 master knobs.
there's a top dog master. and the clean channel has a volume and another master knobs. the fuzz channel has a gain and a drive knobs (for the 2 tubes) and its own master.

my point is that when i see all these gain and master knobs (and channel level, etc.) these days it just kind of makes me go hmmmmmmmm.

specifically, when singing thru the Yorkville for kicks, would you turn the Main Master up more, and the Channel level less.......the other way around........both even?????? just looking for tips from the very knowledgeable.....maybe I should just experiment more......

sincerely thanks very much,
Charlie
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27th October 2007
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When you have a tube amp with a fuzz channel, the choice is purely artistic. If you wanted ultra clean, you would set the input stages fairly low, and the output stages fairly high. If you wanted maximum fuzz, you would set the output fairly low and the inputs fairly high.

The thing about gain structure is that often there are multiple amp stages - and each one will have it's own noise and distortion strengths and weaknesses.

Sometimes the best distortion sounds are a combination of several types of distortion. Guitar players know this well, and exploit various combinations of stomp boxes and preamp and power amp settings.

The trick is to know exactly what each stage does. Sometimes, you are forced to pass signal through a stage that has a weakness. If it has a bad distortion sound, then you would tend to keep the levels low. If it is noisy, you would tend to keep the levels high.

Experimentation is the key. Just beware about excessive gain and protect your ears. Start with every knob minimised is a safe starting point.
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27th October 2007
Old 27th October 2007
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kiwiburger,

am speechless with regards to your generosity.

thank you,
charlie
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27th October 2007
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Quote:
any idea how the Channel level knob differs from the Main Master knob????
With a mixer, you are setting up a "mix" - a balance of several sound sources. The Master knob allows you to set the final output level without disturbing the mix balance you have set up.

For example - in sound check you might establish a good balance, but when the hall fills up with people that make a lot of noise and absorb a lot of sound, you want to increase the overal level, without destroying the balance.

Rough rule of thumb - the lead instruments should have the channel settings around unity gain - 0dB. If you were driving an external power amp, I would suggest setting the master at unity - 0dB too, and adjusting the power amp gain to provide the maximum level required. And until you actually need that maximum level, set the Master level lower.

In the case of a powered mixer, the Master is probably your amp gain - so just set it to the level you need.

Generally, mixer circuits are well behaved at around unity gain. Or around the mid position of a pot or fader - but that's a gross generalisation with zillions of exceptions.

It's generally better to cut than to boost - because cutting also lowers noise along with signal. Boosting boosts noise, along with signal. So if everything is sweet at full volume, with the channel faders sitting around 0dB, then mixing to lower levels is no problem.

It's when you have the channel faders set at max, and then realise you need a bit more volume on that channel - you are screwed. Or, if your channel fader is ridiculously low, you could be getting lots of hiss along with your signal.
#25
27th October 2007
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Quote:
Originally Posted by moracspace View Post
I can see that not just the original poster is confused on this subject.
Anything i've said that you disagree with? I'm keeping it pretty simple on purpose ... basically i'm pretty simple.

Too much maths and unitless "dB" figures don't really help imo.
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27th October 2007
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Quote:
Originally Posted by Kiwiburger View Post
That's not a bad rule of thumb. Because every active electrical circuit you pass your signal through will add noise. So unless you are getting some gain from the circuit, you are increasing the noise without increasing the signal - i.e. the signal to noise ratio is getting worse.

The obvious exceptions are DI boxes and Reamp boxes. A typical DI box takes an Instrument signal, and reduces it down to Mic level. A typical reamp box takes Line level and reduces it down to Instrument level. I tend to prefer passive boxes for this, because a passive (transformer) DI or reamp doesn't add noise.

But sometimes the slight noise that active circuits add is compensated for by other benefits - e.g. buffering, impedance matching, or just the tone or harmonic distortion an active device can bring to the party.
A DI doesn't change the level of anything. It is just a transformer with a high impedance TS jack input on one end and a low impedance male XLR ouput connector on the other. The purpose of this is impedance bridging. It basically converts a high impedance instrument signal to a low impedance mic signal, which can then form an impedance bridge with your line level mixer input (after you have supplied the 40 to 60 or so decibels of gain via a preamp to get a mic level signal up to line level).

You probably already know what impedance matching is if you have ever done any research into hooking up passive speakers to a power amplifier. You need to make sure that the source and load impedance is the same and this ensures maximum power transfer. If the load has lower impedance than the source then your speakers draw more power than the amp is capable of under nominal conditions and it overheats.

Circuits are said to be bridged when the load impedance is at least 10 times greater than the source. This ensures maximum voltage transfer which is mostly what you want when transferring signals between audio devices. Every cable connection that you make should be bridging circuits.

You can sort of hear the effects of not forming an impedance bridge if you plug an electric guitar straight into the input of a non DIed/non preamped input on your soundcard and try recording it. It will be very very quiet because you don't have the gain boost normally provided by the preamp. And your guitar will sound horrible with really weird sounding bass, loads of treble loss and sometimes distortion because you are getting less than ideal voltage transfer.

Whenever you are chaining devices in this way you want each device to form an impedance bridge with the next device in the chain, all the way through to your mixer input. A mic usually has 10 times less resistance than a typical mixer input anyway so it already forms a bridge when you hook them up. You just need a preamp to get the signal up to line level. You do not need a DI for a mic when plugging them into a soundcard or a mixer. You do however need a DI for high output impedance devices like guitars. Guitar pickups in general can have different impedances as a result of the number of windings but its roughly in the same kind of range that a DI box can be a cure all. 'Hot' passive pickups will have more windings and thus more impedance. Active pickups have less windings and are lower impedance lower level output devices making these easier to stick straight into a mixer input without loads of treble loss. It mostly still sounds like crap without a DI/preamp though in my humble experience.

And no you can't really fix it with an EQ after the mixer, or adding loads of gain after conversion. It just increases the noisefloor in proportion with the peak level of the signal (on the order of 40 to 60 decibels) and you get a horrible amount of hiss which pretty much always sounds terrible.
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#27
27th October 2007
Old 27th October 2007
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hey kiwiburger,

thanks very very much again. and thank you robobaby.
#28
28th October 2007
Old 28th October 2007
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Quote:
A DI doesn't change the level of anything.
Can't agree with you there.

I agree with the need for impedance matching - but once again, it's something that can be exploited for artistic effect.

I have several different types of DI box, and apart from the ones that have a Line Level output, they generally have a Mic level output. Which is much lower than Instrument level, which tends to be somewhere between Mic and Line level.

My simplistic view of what a passive DI transformer does: A guitar pickup needs to be presented with a hi-z load, otherwise it tries to deliver too much current, at the expense of voltage. So plugging a guitar directly into a low-z Mic preamp is usually a disaster (but maybe it's a cool sound, so try it).

So a DI transformer has an input coil with lots of windings to present a hi-z load to the guitar pickup. On the secondary side, we want to present a low-z source to drive the Mic preamp.

The turns ratio is typically chosen to match a nominal 50 kΩ signal source such as the magnetic pickup of an electric guitar to a 100 Ω input.

A spin-off effect of having a transformer with a big input:output turns ratio is that the output voltage will be decreased. That happens to be what we want anyway - because the guitar output is a lot hotter than the output of most mics.

Talking in huge generalisations here - because some condensor mics can practically be a line level. But generally, dynamic mics and ribbons have very low output, compared to a guitar pickup.

And active DI boxes can pretty much be designed to do whatever gain increase or decrease you need.

Sorry to be pendantic, but I just can't agree with this statement.
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#29
28th October 2007
Old 28th October 2007
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thanks guys.
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28th October 2007
Old 28th October 2007
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Quote:
Originally Posted by chasman View Post
hi guys,

am looking for any and all help I can get on understanding gain structure.

have come to believe it is super important.

)
Its one of those phrases people use when they want to pontificate

The reality is only an idiot wouldnt listen to what he was doing and not adjust gain as needed. Its not secret knowledge but its brought up as such by some to explain why things might not sound right--when in fact it almost always Not gain related.
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If you just used the word MUSICAL in your post... You just repeated a term, you heard from some pansy, that has absolutely no meaning.
Congratulations.....Your a follower.
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