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What Do You Want to Know About Live Sound But Are Too Afraid To Ask? Dynamic Microphones
Old 2nd March 2014
  #1
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monocluster's Avatar
 

What Do You Want to Know About Live Sound But Are Too Afraid To Ask?

It's come to my attention that myself and a few select other regulars here have a much longer-than-average fuse when it comes to explaining basics (or not-so-basics) to "n00bs." I'd like to put an open offer on the table for seemingly-stupid questions without fear of internet-hate being reaped upon you.

Just a humble offer to bring my knowledge to the table, pay it forward, etc, etc. I'm always happy to explain things to people who actually care and I know some of the board vets will jump in on this too.

Ask away.
Cheers y'all.

-m
Old 3rd March 2014
  #2
Gear Head
 

PA for "one man band" with tracks.

Long time, full time professional player, with wide experience from Concerts, studio, clubs, now it's time for a solo act!
I am a keyboard player with an odd keyboard called Tyros.. it produces tracks as you play a chord!
I am very fussy about sound... I think good sound is more important than I once realized.. I am guessiing though the audiences know little about music, they DO have their opinions. So from my fussy pov and their pov, a "superior sound"** is important!
** not distorted, clear, punchy, not boomy, and able to handle reasonably middle loud volume of dance music for 50 - 150 people.
Due to my advanced age, I believe sound can be simultaneously MAins and to monitors at least to some extent.. So the speakers need to not be in front of me at keyboard.
I have qsc K10's Eon G2 15's a new EV 15 ZLX that is inexpensive and sounds smooth. I also own a Bose L1 Model 2! But I lost the power amp section!
Bose have 2 things going for them in spite of too pricey and not good bass- they sound great for voice, including speaking.. they sound similar as a monitor and 50-75 feet away. I have to make an agonozing decision to replace the amp section for a cool $1450 with cheap soft case!
That is an aside. I also have an inexpensive Newer Mackie 10 or so channel mixer which I used last night with K-10s on stands, I was not crazy about the sound.. not sure if it was my inexperience, or the K10's or the $250 Mackie.
I have found the drum machine sounds best, esp cymbals through the old Eon 15 G2. I may get another Eon, as one of them has crapped out.

Keyboard created- Drum tracks, bass, remainder of tracks, keyboard sounds, and voice through the TC harmonizer, is what I am dealing with. Any and all suggestions to this veteran player, newbie engineer, would be gratefully received.
Thank you kindly for this offer.
Old 3rd March 2014
  #3
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monocluster's Avatar
 

If I understand correctly, you are looking for some PA recommendations. The K10 is a great box, certainly in my top 3 picks. The EONs you mention tend to have a pretty nasty dip around 1.5k due to the crossover. [EDIT: The crossover is an internal filter circuit that splits the audio signal so the lows go to the woofer and the highs go to the tweeter. On that particular model, the crossover occurs at 1.5k but the eon 15's crossover is quite sloppy]

That is also a frequency range that tends to sound pretty funky with male vocals and keyboards in general (I know this because I toured for years as a professional piano-playing singer/songwriter), so the dip that "naturally" occurs there in the EON might actually be replicating the kind of EQ cut that I would make when mixing those instruments.

If your mixer will let you sweep the frequency of the mid band, try a 6db or so cut at 1.5k and that should give you a better sound through the K10.
Thanks for the question!
-m

EDIT #2: the K10 will have a much better response in both the highest and lowest frequencies than the eon will, so you were most likely hearing things that you were not used to hearing coming out of the eon. A good analogy is when you upgrade your standard CRT television to a nice HD plasma set and you all of a sudden notice how bad standard-def DVD's look. It seems like the quality got worse, but in truth you just hadn't noticed it before because the viewing method wasn't adequate.
Old 3rd March 2014
  #4
Gear Head
 

Quote:
Originally Posted by monocluster View Post
If I understand correctly, you are looking for some PA recommendations. The K10 is a great box, certainly in my top 3 picks. The EONs you mention tend to have a pretty nasty dip around 1.5k due to the crossover. [EDIT: The crossover is an internal filter circuit that splits the audio signal so the lows go to the woofer and the highs go to the tweeter. On that particular model, the crossover occurs at 1.5k but the eon 15's crossover is quite sloppy]

That is also a frequency range that tends to sound pretty funky with male vocals and keyboards in general (I know this because I toured for years as a professional piano-playing singer/songwriter), so the dip that "naturally" occurs there in the EON might actually be replicating the kind of EQ cut that I would make when mixing those instruments.

If your mixer will let you sweep the frequency of the mid band, try a 6db or so cut at 1.5k and that should give you a better sound through the K10.
Thanks for the question!
-m
So Eon has a 1.5 dip, and K 10 needs a 1.5 dip? Are you saying ( maybe coincidentally ) that putting a dip in the K10 will please my ear because I am accustomed to the 1.5 dip ( -6db dip ) in the Eon G2's?
OR does the K10 have the converse of the Eon, where it accentuates 1.5, thus I need to attenuate it?
Thank you
Old 3rd March 2014
  #5
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Quote:
Originally Posted by cowboycurtis View Post
So Eon has a 1.5 dip, and K 10 needs a 1.5 dip? Are you saying ( maybe coincidentally ) that putting a dip in the K10 will please my ear because I am accustomed to the 1.5 dip ( -6db dip ) in the Eon G2's?
OR does the K10 have the converse of the Eon, where it accentuates 1.5, thus I need to attenuate it?
Thank you
The K10's response is (more or less) flat. The eon's response is decidedly not. (I'm not just looking at published specs; I've actually bench tested these boxes with an analyzer in the shop and can tell you this first-hand.)

You're accustomed to hearing the "non-flat" (read: colored) response from your eon, and additionally, cutting some 1.5k content out of keyboards and vocals is pretty common because that range can be unpleasant. So when you made the move from a speaker which doesn't reproduce those frequencies well, to a speaker that does, you're all of a sudden hearing more of the unpleasantness that was in the signal content all along.

You should apply a judicious cut and see if that sounds any better to you.
Old 3rd March 2014
  #6
Gear Head
 

Quote:
Originally Posted by monocluster View Post
The K10's response is (more or less) flat. The eon's response is decidedly not. (I'm not just looking at published specs; I've actually bench tested these boxes with an analyzer in the shop and can tell you this first-hand.)

You're accustomed to hearing the "non-flat" (read: colored) response from your eon, and additionally, cutting some 1.5k content out of keyboards and vocals is pretty common because that range can be unpleasant. So when you made the move from a speaker which doesn't reproduce those frequencies well, to a speaker that does, you're all of a sudden hearing more of the unpleasantness that was in the signal content all along.

You should apply a judicious cut and see if that sounds any better to you.
Very very useful info, thank you. Have you checked the new inexpensive EV ZLX for flatness? It has a number of DSP modes to shape sound, I don't know if that ruins any chance to measure it for flatness or not! Let us know your thoughts on it. It is well liked on another forum.

How significant a factor is a $200- 250 10 channel mixer ( or a dumbie using it -) I kept it fairly flat but did dip the mids a tad ) versus a more expensive mixer for clarity of sound? I am guessing my inexperience with engineering my sound is the bigger factor. But this Mackie, I don't know! Thank you again.
Old 3rd March 2014
  #7
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Quote:
Originally Posted by cowboycurtis View Post
Very very useful info, thank you. Have you checked the new inexpensive EV ZLX for flatness? It has a number of DSP modes to shape sound, I don't know if that ruins any chance to measure it for flatness or not! Let us know your thoughts on it. It is well liked on another forum.

How significant a factor is a $200- 250 10 channel mixer ( or a dumbie using it -) I kept it fairly flat but did dip the mids a tad ) versus a more expensive mixer for clarity of sound? I am guessing my inexperience with engineering my sound is the bigger factor. But this Mackie, I don't know! Thank you again.
I have not had a chance to check out those EV boxes but I don't believe that they would have any significant variations in their frequency response. The eon is one of the only widely used boxes out there that just doesn't cut it for me (the Mackie Thump would be another one) generally speaking, any modern powered speaker will have an at least marginally flat response.

I can't think of any professional modern mixer that doesn't have a summarily flat response within the audible range. Some of them get a little funky once you get down towards DC (0Hz) and obviously response won't stay flat up into ultrasonics (>20kHz) but as a general rule a mixer is not something that is altering the frequency response of the system (EQ excluded, obviously).

Here is sound on sound's transfer function test of the X32 when it first came out, you can see it's basically flat from 20Hz to 20kHz. (whole article)

In short: in a perfect world, speakers and mixers are both flat, however in the real world, speakers have a much larger effect on spectrum than mixers should.
Old 3rd March 2014
  #8
Lives for gear
OK, Here's a question for you. Can you explain unitary gain or proper gain staging through a PA? What do you want in your gear to make it happen. I'll admit that I do know a little bit about it but I'm no expert. I thought that this would be a good question to pose for newbies(like me) to learn since its always being referred to.

Anthony
Old 3rd March 2014
  #9
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....and what would you do different if it's a digital desk?
Old 3rd March 2014
  #10
Quote:
Originally Posted by AnthonyG View Post
OK, Here's a question for you. Can you explain unitary gain or proper gain staging through a PA? What do you want in your gear to make it happen. I'll admit that I do know a little bit about it but I'm no expert. I thought that this would be a good question to pose for newbies(like me) to learn since its always being referred to.

Anthony
There are any number of great videos on YouTube concerning this and most of them follow the same concepts. The idea being you keep your channel faders set to unity while you PFL (Pre Fader Listen) each channel's input, adjusting the gain (not the fader), and measure it on the meter to get each channel set so it has ample headroom and is at the appropriate relative volume to the other channels. What you want in your gear are two things, a PFL switch for each channel, and a gain knob.

The tricky part in all of this is that signals will have nominal levels as well as peaks when you set them up. Peaks will indicate how much headroom each channel has whereas nominal (fairly steady indicators) tell you the relative loudness of that channel compared to other channels. If you systematically go through each individual channel during the soundcheck you should have a relatively decent mix of all instruments when you're done with all faders set at unity. During performance you may need to tweak the faders slightly to get what sounds good to your ear, but they shouldn't have to vary too greatly if the gain staging was done correctly. The process would be the same on a digital desk, just selecting each channel and PFL'ing it appropriately.

You can choose any level on the meter you like as the baseline for the signals. For example, you want your vocals to stand out over the instruments, so you choose to have their nominal signal levels at unity on the meter when PFL'ing them. That would mean they would likely peak at 2 or 3 db above that on occasion still giving you plenty of headroom. Drums are different because they really only have peak levels given the sound decays so quickly. Therefore you might target the peaks for the snare and kick drum at around 2 or 3 db above unity on the meter to ensure they punch through the mix. Bass you might target 3 or 4 db below unity on the meter to keep it's peaks just below unity. Lead guitar slightly above that so that leads stand out in the mix.

This all assumes, of course, that once everything has been gain staged the musicians have the discipline to leave their volume settings alone. That's why I typically have all the musicians start by playing something together so they get their stage sound where they can hear themselves and each other appropriately before I start gain staging.
Old 3rd March 2014
  #11
Lives for gear
Thanks for a great answer. Here's a quick follow up question I hope. Is there a practical difference between a "PFL" button and a "Solo" button? I've seen both on some mixers and one or the other on other mixers. When choosing your mixer do these functions vary and make a difference.

Thanks, Anthony
Old 3rd March 2014
  #12
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Audiobond's Avatar
 

Sometimes PFL and Solo are the same, sometimes they aren't. It all depends on the terminology that particular manufacturer uses...

PFL stands for Pre Fader Listen. As the term implies, you are then listening to that channel BEFORE (PRE) the fader affects the level. Useful for setting input gains as you are hearing/seeing the level that is going thru the channel strip and then use the fader to adjust the output of that channel.

On a typical live console, this sends only the selected channels into the cue mix (ie headphones at FOH) for the engineer to hear only those channels but leaves the main mix alone. Many consoles also have a seperate AFL/PFL (or cue mix) meter to show the level of that channel(s) which can be helpful if the consoles channel meters show you the level AFTER the fader affects the level. AFL, btw stands for AFTER FADER LISTEN and is more common to see on group or aux outputs as opposed to individual channels, altho many consoles allow you to switch between the two. AFL also leaves the main mix alone. Both allow the engineer to hear what he needs to in the headphones to make adjustments or listen for problems without interrupting the main mix.

SOLO is typically a term for studio. Also known as SIP or SOLO IN PLACE, it behaves more like you would expect to hear in a studio in that it will mute all other channels that aren't selected even in the main mix. Obviously you don't want to do this during a show! Nothing like trying to hear just the vocal in the headphones and all of a sudden the whole mix drops out in the house and the singer is now naked! Again, many consoles offer the option to toggle this feature on and off as some engineers do like to use this feature during soundcheck etc...

Traditionally, most live boards offer PFL. Some (especially lower end ones) may call it "SOLO" and mean PFL as opposed to SIP. Some call it SOLO/PFL just meaning PFL but I guess throw the "SOLO" term in there for less experienced users that may not know what PFL stands for. It can be a little confusing, especially to a newer operator. Generally, on a live board, if it doesn't say SOLO IN PLACE specifically, you are probably safe assuming it is a PFL, but... Check the manual, or just hit the button during soundcheck or setup to see what it does and make sure you aren't going to ruin a performance.

Good luck!
Old 3rd March 2014
  #13
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Audiobond's explanation of unity gain mixing is spot on, and I'll just add to it since you asked about the PA system.
Mixing unity-gain optimizes your gain structure from the mixer "upstream" to the mics. It is also in your best interest to do the same thing downstream.
Any device that comes downstream of the mixer (system processor, inserted effects, amps, etc) should basically follow the same rules.
If your output from your mixer is directly driving the amp, for example, you max out the mixer, using pink noise or other signal, until it is "just barely not clipping", meaning lighting up everything on the output bus except for the clip LED. No more headroom.
Then you feed this into the amp (cover your ears!) and turn down the amp's level set knob until the amp is also just-barely-not-clipping.
Note: the knobs on the amp are NOT volume knobs. The amp always runs at full tilt. they just set how much signal is required to bring the mixer to full output. By optimizing in this way, you will make sure that everything clips / maxes out at the same point, which means no headroom is wasted anywhere and your system is running at max efficiency.

System processors and crossovers can be a little different because often times gain changes are required between system elements, but as long as you have a basic handle on the concept of proper gain structure, you can make intelligent decisions when these situations arise.

Likewise for any inserted effects: they should generally operate at unity gain. When you toggle the effect between engaged and bypassed, there should not be a marked difference in level of signal. We use effects to generate effects, not additional gain.

Again, some considerations here are due when compressors / limiters are used since these devices by their very nature are used to affect the level of the signal, but the same concepts still apply.

The result is that it is very easy to look at your system and clearly see where any cuts/boosts are happening, and that your system is fully optimized, with no bottlenecks and no wasted headroom.


Digital/analog desks make no difference in terms of gain structure, they're both just different tools to do the same job.

Understand this concept well and you will always have a job.

Thanks for the great question, and to audiobond for jumping in with a great explanation as well.
Old 3rd March 2014
  #14
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Quote:
Originally Posted by AnthonyG View Post
OK, Here's a question for you. Can you explain unitary gain or proper gain staging through a PA? What do you want in your gear to make it happen. I'll admit that I do know a little bit about it but I'm no expert. I thought that this would be a good question to pose for newbies(like me) to learn since its always being referred to.
Beyond the mixer, unity gain for a system is basically that what goes in is what comes out or that the signal level out of a device is the same as the signal level into that device, thus every device functions as a 1:1 or unity gain device.

What is more common now is to set system gain downstream of the mixer based on all devices clipping simultaneously. Since not all devices accept or support the same nominal or maximum input and output levels, this approach allows that to be accommodated in a more effective manner. Here are some relevant resources:

AV: How Do You Set Sound System Gain Structure? - Pro Sound Web
Ins & Outs Of Gain Structure
http://www.naterecording.com/gainstructure.pdf


One aspect of system gain structure that seems to often be misunderstood is amplifiers. One common misconception is that the level controls on an amplifier control the amp output. The reality is that an amplifier is always capable of its full output and in almost all amplifiers the level controls simply attenuate the input signal to varying degrees. Thus the level controls on an amplifier are varying the input signal level and not the amp output. Turning down the input level controls on an amplifier means that you then need a greater input signal level to get the same output, however it does not alter the potential output of the amplifier.

That leads into understanding amplifier sensitivity, a spec to which many never pay attention. Sensitivity for an amplifier is the input signal level that will result in the full output from the amplifier. It is essentially the inverse of amplifier gain, a low gain amplifier will require a greater input signal level (a higher sensitivity) to obtain a certain output than will a high gain amplifier (a lower sensitivity).

Why this matters to system gain structure is that an input signal level above the sensitivity value for an amplifier does not result in any additional output yet most source devices provide and most amplifiers can accept an input signal level significantly greater than the amplifier's sensitivity. It is quite common to have a mixer or system processor that can output up to a +24dBu signal level and an amplifier that will accept that signal level but may require only a +4dBu or +5dBu level to generate its full output.

As a result, good system gain structure leads to it being quite common to want to attenuate the input signal to the amplifier so that the maximum input signal level to the amplifier equals the amplifier sensitivity and thus the amplifier output clips at the same time as the output of the device feeding the amplifier (some prefer that the amplifier clip slightly before the other devices and thus apply a bit less attenuation). That attenuation may be achieved via a pad before the amplifier or via reducing the output signal level of the device directly before the amplifier in the signal chain, however probably the most common method is via the amplifier input level controls. Keep in mind the point above, adjusting the amplifier input levels is simply reducing the input levels to the amplifier and not reducing or limiting the potential amplifier output.
Old 3rd March 2014
  #15
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Thanks again guys for your great answers.

Anthony
Old 4th March 2014
  #16
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Electrolytic's Avatar
 

Quote:
Originally Posted by Audiobond View Post
.

SOLO is typically a term for studio. Also known as SIP or SOLO IN PLACE, it behaves more like you would expect to hear in a studio in that it will mute all other channels that aren't selected even in the main mix. Obviously you don't want to do this during a show! Nothing like trying to hear just the vocal in the headphones and all of a sudden the whole mix drops out in the house and the singer is now naked! Again, many consoles offer the option to toggle this feature on and off as some engineers do like to use this feature during soundcheck etc...

Traditionally, most live boards offer PFL. Some (especially lower end ones) may call it "SOLO" and mean PFL as opposed to SIP. Some call it SOLO/PFL just meaning PFL but I guess throw the "SOLO" term in there for less experienced users that may not know what PFL stands for. It can be a little confusing, especially to a newer operator. Generally, on a live board, if it doesn't say SOLO IN PLACE specifically, you are probably safe assuming it is a PFL, but... Check the manual, or just hit the button during soundcheck or setup to see what it does and make sure you aren't going to ruin a performance.

Good luck!
your description of Solo in place is incorrect.
SIP works like Solo, expect that you aren't just hearing what has been cued, instead you hear the cued channel with all the other channels 'dimmed' by -20db or whatever the amount is set to. it is like hearing that track 'highlighted' in place with how it sits with the rest of the mix.

also this would not affect the main L&R mix to tape and like wise not effect a Live mix to FOH either as this is a control room section feature just like any mixer with an operator monitoring section.
Old 4th March 2014
  #17
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What does gain staging really mean?

Right I need help.

I'm not new to this

However my formal training happened in a studio.
When setting the gain on a pre amp the pfl button is used.
the school of thought is to to get every channel to sit at 0dB nominal on the meters because that gives the best possible signal quality. not to low that signal to noise ratio (SNR) is an issue nor not to hot that Harmonic distortion becomes a problem. 0dB has been defined by the engineers that designed the pre amp as the best comprise and the thus the best signal.

ok

so everything gets gained for optimal signal, the mix happens on the faders. that could mean the hats are as strong or stronger than everything so the fader is pulled down to like -10dB or -15dB. In that region the resolution of the fader is smaller and an incrementally adjustment could have a large effect, unlike around nominal where the most precise control can be found.

ok

I use the above method to mix bands live. usually vocals at nominal with everything else pulled down accordingly with the master set appropriate to the room at the time.
I work with a guy who mixes the first band on the gain pots with all faders at unity, the second band on the faders, then the third band on the groups. That is how he builds a solid mix and can knows he can fall back on all faders at unity as a base line mix if things go to far one way.

Now historically the carbon track faders on analogue desks where not very good quality when used far below the unity region where the most control is. so engineers would gain things to sit in the region.
Now with digital desk there is no need to do this as the fader is for data not physical resistance value.

ok so getting back to setting gains on preamps, in the digital realm getting the optimal balance between SNR and harmonic and intermodulation distortion is important but also is getting the most efficient bit depth for the signal at the A/D converters as well. This is usually -18dBFS on the meters - this is your optimal nominal point defined by the engineers, this is equivalent to 0dBu on an analogue voltage meters.

so

getting the best signal on a digital desk where the faders do not degrade quality is surely the better way to go?
who's right here?


in general on any desk:

why do you guys run all your faders at nominal?
why do you adjust your pre amp gains to suit this fader arrangement yet it is not the best signal quality? is this because a rough mix is in place and thus when using other functions like aux sends to monitors the levels are not widely different and behave the same so the mix to them is more consistent?
are you doing it to utilise the range of the faders better?

please I don't understand the reasoning behind your school of thought.
Old 4th March 2014
  #18
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Audiobond's Avatar
 

Quote:
Originally Posted by Electrolytic View Post
your description of Solo in place is incorrect.
SIP works like Solo, expect that you aren't just hearing what has been cued, instead you hear the cued channel with all the other channels 'dimmed' by -20db or whatever the amount is set to. it is like hearing that track 'highlighted' in place with how it sits with the rest of the mix.

also this would not affect the main L&R mix to tape and like wise not effect a Live mix to FOH either as this is a control room section feature just like any mixer with an operator monitoring section.
I guess it depends on the console you are using. I rarely if ever use the SIP function, but the times I have, the way I described it is how it functioned. And for that matter on any console I've ever heard of. In fact, a quick google search on the subject supports my statements.

Not to say you are wrong in your description for a given console (whatever that may be), but I am not wrong either. No fighting intended!

As for the studio comparison regarding SIP... Now that you mention it and I think about it, I honestly cant remember how it affects what is going out the master to 2 track as I haven't done studio work in forever and didn't solo things while printing, so no real frame of reference there. My apologies if that part was incorrect.
Old 4th March 2014
  #19
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Audiobond's Avatar
 

As to your other point/question, Electrolytic...

I get both schools of thought. Wanting to run the channel at an optimal level (ie. PFL and set to 0dB-ish) makes total sense and is generally the way I set a console up. But I also understand wanting to keep the faders near unity where the resolution is finer. Occasionally I will trim the input down on particularly hot channels a bit to keep the fader from being completely buried in the armrest, but for the most part I agree with your school of thought.

I would tend to say that running channels at/near 0dB (assuming analog... as you mentioned -18 dBFS for digital) is more TECHNICALLY correct for gain staging and optimal use of the EQ/Dynamics. But there is something to be said for being able to make small moves by having the fader closer to unity. To each their own as long as it sounds good in the house, I suppose.
Old 4th March 2014
  #20
Quote:
Originally Posted by Audiobond View Post
I would tend to say that running channels at/near 0dB (assuming analog... as you mentioned -18 dBFS for digital) is more TECHNICALLY correct for gain staging and optimal use of the EQ/Dynamics. But there is something to be said for being able to make small moves by having the fader closer to unity. To each their own as long as it sounds good in the house, I suppose.
Agreed. Especially on your last statement.

In my mind LIVE gain staging is less about absolute optimization of the signal and more about getting the mix right, or as close to right as possible. In a recording studio situation I might choose to think differently.

Because you only get one shot at a correct mix in a live situation there's a security in knowing you have a "gravity point" at unity you can rely on if all else fails. The dynamics of the mix may change slightly from song to song or even within a song, but I can react faster on fader moves knowing that unity is where everything started and I can see how far I'm veering from that at any point in time. But ultimately I have to respond with my ears and not with my eyes, so unity is my reference point and my ears and faders are my refinement adjusters.
Old 4th March 2014
  #21
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Quote:
Originally Posted by Electrolytic View Post
your description of Solo in place is incorrect.
SIP works like Solo, expect that you aren't just hearing what has been cued, instead you hear the cued channel with all the other channels 'dimmed' by -20db or whatever the amount is set to. it is like hearing that track 'highlighted' in place with how it sits with the rest of the mix.

also this would not affect the main L&R mix to tape and like wise not effect a Live mix to FOH either as this is a control room section feature just like any mixer with an operator monitoring section.

This may be true for recording consoles, but not at all true for live boards.

Depending on the console, Solo In Place will either effectively mute (Avid/Yamaha), or actually mute (Midas) every channel except the one being soloed. It is like using the PA as your "headphones" to listen to a single channel at a time.

This can be very handy during soundcheck to hear a channel within the context of the PA, but NEVER DURING A SHOW OR RECORDING. In live consoles, it is NOT a control room function, but is an actual destructive solo that replaces the FOH/Main bus mix with the single channel that you pull up. On Midas (digital) boards, it goes a step further and will affect your aux mixes as well, as any non-soloed channels in SIP mode will be totally muted.
Old 4th March 2014
  #22
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Electrolytic's Avatar
 

good god I hope that needs the goldeneye key to be activated, using that accidentally would be soul destroying.
Old 4th March 2014
  #23
Gear Addict
 
monocluster's Avatar
 

Quote:
Originally Posted by Electrolytic View Post
Right I need help.

I'm not new to this

However my formal training happened in a studio.
When setting the gain on a pre amp the pfl button is used.
the school of thought is to to get every channel to sit at 0dB nominal on the meters because that gives the best possible signal quality. not to low that signal to noise ratio (SNR) is an issue nor not to hot that Harmonic distortion becomes a problem. 0dB has been defined by the engineers that designed the pre amp as the best comprise and the thus the best signal.

ok

so everything gets gained for optimal signal, the mix happens on the faders. that could mean the hats are as strong or stronger than everything so the fader is pulled down to like -10dB or -15dB. In that region the resolution of the fader is smaller and an incrementally adjustment could have a large effect, unlike around nominal where the most precise control can be found.

ok

I use the above method to mix bands live. usually vocals at nominal with everything else pulled down accordingly with the master set appropriate to the room at the time.
I work with a guy who mixes the first band on the gain pots with all faders at unity, the second band on the faders, then the third band on the groups. That is how he builds a solid mix and can knows he can fall back on all faders at unity as a base line mix if things go to far one way.

Now historically the carbon track faders on analogue desks where not very good quality when used far below the unity region where the most control is. so engineers would gain things to sit in the region.
Now with digital desk there is no need to do this as the fader is for data not physical resistance value.

ok so getting back to setting gains on preamps, in the digital realm getting the optimal balance between SNR and harmonic and intermodulation distortion is important but also is getting the most efficient bit depth for the signal at the A/D converters as well. This is usually -18dBFS on the meters - this is your optimal nominal point defined by the engineers, this is equivalent to 0dBu on an analogue voltage meters.

so

getting the best signal on a digital desk where the faders do not degrade quality is surely the better way to go?
who's right here?


in general on any desk:

why do you guys run all your faders at nominal?
why do you adjust your pre amp gains to suit this fader arrangement yet it is not the best signal quality? is this because a rough mix is in place and thus when using other functions like aux sends to monitors the levels are not widely different and behave the same so the mix to them is more consistent?
are you doing it to utilise the range of the faders better?

please I don't understand the reasoning behind your school of thought.
Solo-In-Place on most live desks is destructive and mutes all other channels to the main outputs.

-18dbFS is only equivalent to 0dBu on midas, yamaha, some other console manufacturers do it differently.

The studio practice of "saturating" the preamp comes from trying to combat analog tape hiss, getting each track's signal as hot as possible. With modern recording equipment this isn't really necessary anymore, nor is it necessary or desirable live. Preamps amplify noise, too, so we generally don't want it working harder than it needs to be. (Exception here for dual-stage desks like the midas pro series which have analog pre for color then digital trim for gain setting) so I'm not sure why you think this creates an inferior signal quality, it's actually better.

Most motorized faders are 10-bit (1024 set resolution) which is linear, not logarithmic so the highest "definition" is in the 0db range.

It is also the only way to "normalize" gain structure through the entire aux matrix.
Old 4th March 2014
  #24
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monocluster's Avatar
 

Midas SIP is covered by a plastic flip top to prevent accidental engagements.

Think about musical theater with 24 body mics. If you don't have your gain structure set up properly, you will need faders all over the place to produce balanced dialogue. If you set everything up at 0db, you can throw any faders to unity and the dialogue will be at the right level.
Old 4th March 2014
  #25
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Jhana's description is correct...
Old 4th March 2014
  #26
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Quote:
Originally Posted by Electrolytic View Post
why do you guys run all your faders at nominal?
why do you adjust your pre amp gains to suit this fader arrangement yet it is not the best signal quality? is this because a rough mix is in place and thus when using other functions like aux sends to monitors the levels are not widely different and behave the same so the mix to them is more consistent?
are you doing it to utilise the range of the faders better?

please I don't understand the reasoning behind your school of thought.
The way you describe setting up the gain structure of a mixer, analog or digital, is the generally-recommended way; set the head amps to get a signal level hovering around 0dB on the main/PFL meter, then use the channel faders to set the amount of that channel in the mix. However, there is a competing method used by a lot of people, especially live, and that is to set all faders at unity (or -6dB or similar reference point), then create the basic mix using the head amps. The resulting configuration allows you to use the unity gain on the channel faders as a "close enough" mark. In the olden days, it had the additional advantage of mitigating the capacitive effect of the fader hardware available at the time; modern mixers are transparent enough that any EQ change based on fader position is minimal to nonexistent.

I do not run my faders at unity when I run a board; I prefer to use the head amps for their designed purpose, lifting the signal out of the noise floor, and I discourage the unity fader method in general. I do realize its value when working with novice sound technicians; if you need to leave the board in the hands of someone who can count the number of times they've sat at a board on the fingers of one hand, it's easier to say "all faders at unity is an okay mix, stay around there" versus having a different "normal" for each fader. But, it's a crutch.
Old 4th March 2014
  #27
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Quote:
Originally Posted by monocluster View Post
Think about musical theater with 24 body mics. If you don't have your gain structure set up properly, you will need faders all over the place to produce balanced dialogue. If you set everything up at 0db, you can throw any faders to unity and the dialogue will be at the right level.
I disagree. If you did your sound check properly and set up your lavs for a 0dB average level, balancing dialog will result in the faders around the same level anyway, especially if your lavs are in similar places (i.e. lapels). Also, unity on the faders isn't always going to produce balanced dialog even if you set them up that way; relative proximity of one player to another, the actor's head position, relative volume during the performance as opposed to the soundcheck/rehearsal, etc. And it's no bad thing to have the faders all over the board; they indicate what you're actually doing with the mix instead of hiding information.
Old 4th March 2014
  #28
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edva's Avatar
Quote:
Originally Posted by Liko View Post
The way you describe setting up the gain structure of a mixer, analog or digital, is the generally-recommended way; set the head amps to get a signal level hovering around 0dB on the main/PFL meter, then use the channel faders to set the amount of that channel in the mix. However, there is a competing method used by a lot of people, especially live, and that is to set all faders at unity (or -6dB or similar reference point), then create the basic mix using the head amps. The resulting configuration allows you to use the unity gain on the channel faders as a "close enough" mark. In the olden days, it had the additional advantage of mitigating the capacitive effect of the fader hardware available at the time; modern mixers are transparent enough that any EQ change based on fader position is minimal to nonexistent.

I do not run my faders at unity when I run a board; I prefer to use the head amps for their designed purpose, lifting the signal out of the noise floor, and I discourage the unity fader method in general. I do realize its value when working with novice sound technicians; if you need to leave the board in the hands of someone who can count the number of times they've sat at a board on the fingers of one hand, it's easier to say "all faders at unity is an okay mix, stay around there" versus having a different "normal" for each fader. But, it's a crutch.
This is true, however, it is also true that many, if not most, very experienced live sound engineers, including myself, do generally start a mix with the "all faders at unity" (or some equal setting) method. Note, I am not saying this is right or wrong. It is just the reality of what is usually done in professional scenarios. IME and IMHO.
Old 4th March 2014
  #29
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Quote:
Originally Posted by Liko View Post
I disagree. If you did your sound check properly and set up your lavs for a 0dB average level, balancing dialog will result in the faders around the same level anyway, especially if your lavs are in similar places (i.e. lapels). Also, unity on the faders isn't always going to produce balanced dialog even if you set them up that way; relative proximity of one player to another, the actor's head position, relative volume during the performance as opposed to the soundcheck/rehearsal, etc. And it's no bad thing to have the faders all over the board; they indicate what you're actually doing with the mix instead of hiding information.
For dialog/theater/corporate, I think it's important to get a starting point at (fader) unity though. Yes, you will be mixing and moving faders all night, but I like to know for sure that a fader at unity will be a safe and nominal level.

It's not how I set levels for concerts though. I usually start at PFL then bring the faders up. If you are running nominal levels at the meters and all your faders are stuck in the -30 range, then turn down your VCAs, or the master, or... your amps... or go with a smaller rig next time.
Old 4th March 2014
  #30
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Quote:
Originally Posted by Electrolytic View Post
in general on any desk:

why do you guys run all your faders at nominal?
why do you adjust your pre amp gains to suit this fader arrangement yet it is not the best signal quality? is this because a rough mix is in place and thus when using other functions like aux sends to monitors the levels are not widely different and behave the same so the mix to them is more consistent?
are you doing it to utilise the range of the faders better?

please I don't understand the reasoning behind your school of thought.
At least in my mind a couple of reasons. One is keeping the channel faders in the most useful or effective positions. If setting the preamp gain to get a certain level and then adjusting the fader for the desired level ends up with a fader near the top of its travel or down where a small movement makes a big change then that can limit the effective use of that fader. So I may not set all faders to nominal 0, but I do want to try to have the faders where I can make finer adjustments and also where I have sufficient range to make the changes that I think I'll need for the mix (e.g. sufficient range to raise a source level as required for a solo).

I also agree with monocluster, the self noise or noise floor of most live sound microphones and other sources is typically greater than the noise floor of the mixer inputs so why boost the source noise at the preamp only to then attenuate the signal at the faders and sends? It seems like the greatest dynamic range with real world live sound sources would be achieved by not amplifying the signal any more than necessary at the preamp.
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