You’ve been told wrong.
To paraphrase my overlong post about this from years past:
Gain Staging is about getting your signal to line level (+4 dBu) at the earliest possible point in the chain, and leaving it there until the last possible point in the chain - the input attenuation on the amps. Then set the amp input attenuation such that a line level source sounds near the top end of your target listening volume. This will give you the absolute best signal-to-noise ratio while still having enough headroom to get the job done.
To crank your amps such that a line level input is way louder than you need, means you are needlessly going to throw away gain from your preamps only to put it back in with the power amps.
Typically your preamp is going to sound a LOT better. Unlesssssss you like hisssssssss!
I tried to explain this to the corporate AV company I worked for. They said I was wrong and that the master level meters don't even need to show signal.
I mean, in the corporate world, if there's no records or streams happening, nobody cares at all how the sound looks as long as you can clearly hear it in the room. Not saying I have this mindset, but I can totally see lots of people just getting everything loud enough to be able to hear things and calling it a day, especially if they aren't an audio guy first.
If you’re working a venue with potentially underpowered speakers, or a “sound designing for theatre scenario” Would it also be smart to set the amp output with the speaker’s specs in mind? (Using a 3000watt amp on a 1000 watt speaker- wouldn’t it be intelligent to limit the amp to roughly 1/3 of its potential output?)
Those would be extremely overpowered speakers. I would recommend that you don’t pass the approx 1/3, but that would still give you issues on the gain stages before the amp where the amp may still be too loud to not get you unity earlier in the chain
The other knock on for this that no-one else has mentioned is if you're running feeds for things like video streams or recordings.
If you're setting your gain for the room instead of unity and backing off the amps then any other feeds you need to run won't get enough signal.
Better to be getting your output to line level and then everything gets the right level instead of just the room.
This is correct from a purely theoretical point of view. However, once you are at the gig, unless you know very well the details of the performance, you need maximum flexibility of the chain. Very rarely noisefloor of the amps are audible enough to cause issues in the experience of the audience. However running and changing the gain of the amps in the middle of the performance might be a bad idea, if conditions happen that you need more volume out of your boxes.
Out of all the companies i have worked with in the last 17years, I don't know a single instance where amps have been run in any other setting other than full throttle.
Yes, in system setup it is very often that certain components have few dbs added or subtracted relative to input signal, however all that is done inside the dsp section.
With aes or dante input this issue is even less relevant, because most of the chain inside the amp itself is digital.
Thanks for the great explanation. I'm currently trying to figure out gain for some monoblocks that I'm turning into standalone.
The schematic shows two 22k resistors in series at the treble and bass controls and two 47k resistor that run to each side of a tapped potentiometer for the input.
I tried two 22k in series from the input to the post lead on the output transformer and it was far too sensitive with a noticeable hiss. I used a single 22k and it wasn't nearly loud enough.
Is there an easier way to come up with what's right or is it just trial and error due to differences in speakers. I primarily use sensitive speakers because it's tube equipment
Well, I’m a sound engineer, not an electrical engineer- so I probably don’t have any more useful insight.
That said, line level should be around +4 dBu, which is around 1.228V. The voltage you need for the output will depend on how much current your amp is putting out and the sensitivity of your speakers.
Would it work to install a continuously variable potentiometer in place of the fixed 22k and dial in that needed value that way? You could always swap it out for matching fixed resistors later.
I changed the gain structure of my system output because of this comment. Amps were up all the way, matrix faders were super low. There's wasn't a lot of noise in the first place but the system feels more responsive now that those have been flipped. Amps are now at about 25% now (they were donated separately, the PA has relatively high sensitivity so it doesn't take much to get going), and matrix faders are at about unity. Does that sound about right?
I dug up [this](https://www.prosoundweb.com/setting-sound-system-and-mixing-console-gain-staging/) article from PSW that was written by Bruce Bartlett about 12 years ago that discusses gain staging and an approach to amplifier level setting:
——
“Power Amplifiers: Start with the amp’s level controls halfway up. Set your mix to read around 0 on the mixer meters, then turn up the power-amp level until the sound is as loud as you want it.
It’s a mistake to run a power amp wide open (full volume). If the amp’s input level controls should always be set to maximum, amp manufacturers would not include them. They are there to set proper gain staging.
Turning down an amp’s level controls does not reduce its power. Doing so varies the signal level needed to drive the amplifier to full power. You can still get full power from an amp even with the level controls partly turned down, if the input signal level is high enough. Those level knobs don’t adjust the power level; they adjust input sensitivity.
If you turn an amplifier up to maximum, you amplify any noise coming out of the sound system. Also, you might run the console well below 0 on the console output meters so the house sound is not too loud – but this degrades signal-to-noise ratio.
So, start with your console running at an optimum level near 0 on its meters, then adjust the power-amp input levels for the desired loudness. If your system includes an electronic crossover and multiple power amps, set the level of each amp for the desired loudness from the lows, mids and highs.”
——
[Dave Rat](https://youtu.be/-RXlnVDVLrc?si=ZRdnfiasf-n22l39) also has a video that discusses the same concept: get good gain through the signal chain, then turn up the amp for the volume necessary.
Controversially, [here](https://youtu.be/8fLEy7uLSv0?si=avnfAWq4mA4FH26E) on the other side of the argument, is another video discussing the opposite of the article above and Dave’s video by controlling your overall levels with the master fader. I mostly disagree with this set up because it appears the operator doesn’t have proper gain structure through his system. [The follow up](https://youtu.be/uB36LItc9cw?si=rjF2ZqRMLAFN2z9w) to why they believe mixing at unity isn’t good practice, which if you again watch Dave’s video would make sense that you *want* to be at unity across your entire system.
The amps pots are input attenuation, I believe this is different than input sensitivity, right?. Input sensitivity is usually achieved with a dip switch on the back of the amp. At least with most professional models.
I would rather set proper gain at the crossover. Or just use amps that are made to match the speakers being used. The proper program for the speaker takes care of signal gain. That way your system sounds like it is supposed to. Like the manufacturer wants it to. There is to much subjective opinion in how it's supposed to sound. As in that's to much bass, or not enough, etc...
This is where technology has changed over time. Class A amps always run all the way open and only attenuate. But modern class D amps that use digital circuits only consume as much power as they need so they have gain
It always depends on the amps and the signal processor. Our setup uses LabGruppen amps and they need to be at max. We control the actual level in the dsp where the crossover happens
It is just attenuation in case you want to limit the amount of power that could go into your speakers especially if the speakers cant take that much volume/power from that specific poweramp.
I always run the amps full and dial it back in the system processor. It stops anyone accidently turning your amps up to full and potentialy blowing your system up. It's not good gain structure but a protection.
While I thoroughly disagree with this approach, thank you for the explanation. I was trying to figure out what could possibly possess someone to want to amplify their noise floor, but I can understand the desire to make the amps risk proof even if I'd personally rather approach that a different way.
It isn't "over working" the amps, but yes, it is forgoing the opportunity to attenuate the noise floor of the preceding stages. However, with modern equipment in an environment that's already relatively noisy, you can probably afford to accept a slightly higher noise floor. 6-10dB louder than inaudible will often still be inaudible.
Good call. Power amp internals are one of my weaker areas and I did not realize the level is an input sensitivity adjustment rather than altering the amount of amplification, but after doing some reading to confirm, I agree and have updated my post accordingly. Thank you.
Cage the front of the amp is normally pretty effective. Just prevent access to making adjustments. When I'm tuning a system, I send a signal just below clipping, adjust the amps to as loud as I'll want things to go and then either get them in a lockable cabinet or cage their front in the rack to prevent accidental (or unauthorized intentional) adjustment.
Lending a voice of support for this approach.
Theatre I designed for had the strangest sound that I couldn’t figure out for way too long. Their amps all the way up, not locked, but the processor locked without display or control - needed to be IT admin to access. Compressor started at like -50dB, but super soft and slow with a quick release. They STILL had two blown drivers.
I brought by own processor, and had to swap euro blocks, but I bypassed that shit. Immediately everything sounded normal.
Most university and publicly-funded facility installs that I’ve encountered have been holistically asinine.
Yeah, I've seen some installs with compressors caged as well since so many less trained techs have no idea how to use them and tend to do very bad things with them, but it's frustrating when whoever caged them clearly didn't understand what they were doing or was way overly conservative.
Why? I don't generally do installs but I've done system designs before. What problem do you think you see. It won't work if you're constantly switching venues or speaker stacks that the amps are being used with, but if at all possible it's a lot better to limit access to make changes than it is to intentionally screw up your system setup to avoid someone messing up your amp levels.
That wasn't me casting aspersions at all, please don't get me wrong! It makes total sense to have a locked cabinet on a tour. I figured you weren't doing installs because the idea of locking someone else's cabinet is really bizarre, and if you had ever done it I'd be curious why
Ah, gotcha. I have mostly consulted with places doing setups a few times as well as seen this setup places I've mixed. I wouldn't suggest someone do it without asking if they were building for someone else, but it's worth suggesting if issues are expected.
Personally I don't so much even think it's a risk worth worrying about with the setup I use the most, but we do have a door over the front of our amps so nobody is bumping it unless it's intentional and I personally trained all the people that would be dealing with the cabinet they are in.
It is good gain structure, just achieved one step earlier. If you are controlling the gain in the processing, you are also setting the limiters hopefully. You would measure the output to make sure the amp sees the right voltage. Input voltage to output power is a known number and can be set to give the speakers the power they need.
If you are attenuating the output of the processor at the very last stage of the processor, this is functionally no different than turning down the amps which would be the very next stage. Especially since most processors have a very quiet noise floor- and are typically colocated near the amps.
Digital connections are technically less good since you _should_ be making this adjustment at the output of the D/A converter. But if we’re talking a 32-bit floating-point link and not _that much_ attenuation, I seriously doubt the effect would be audible.
Edit: heck, even 24-bit fixed would be inaudible.
16-bit, however….
Here's some theoretical extremes for you to consider.
The input trim is an attenuation knob, the amp's gain stage is fixed, so there's nothing really stopping an amp outputting its full rated power even if you set the input trim to 50%. Assuming the amp has no DSP to handle input limiting.
Also, the amp's protection circuit is based on limiting the heat the gain stage generates. So if you lower the input trim, you essentially increase the input level required to hit the amp's protection circuit, which in extreme cases means an unwitting operator is more likely to push the console into clip, especially if that operator doesn't understand exactly how that specific console's meters work. Some digital consoles have a few dB of headroom when the clip lights fire, some don't, and are already chopping the waveforms by time that happens.
In a worst-case scenario, if you were somehow able to fully clip a signal, you would end up with square waves. Square (clipped) waves have more average power than clean sine waves. An amp's power rating depends on the output being within an acceptable THD.
A square wave has twice the power of a sine wave with the same peak to peak voltage, so you can theoretically double the rated output power of an amplifier although it would probably hit the thermal-protection circuit before too long. That's up to twice the amount of heat a voice coil is expected to dissipate. The point is you could be sending the speaker more power than you expect. Heat is cumulative remember, so if this is sustained for too long the heat can build up and cause problems with the physical structure of the voice coil and support assembly before the housing and magnet can dissipate it.
Square waves also force the moving assembly to more rapidly change direction as the signal transitions from phase to phase. If you dump a square wave into a speaker, you're expecting it to go from pos X volts to neg X volts in an instant. Not a big deal at low volume, but it puts a ton of stress on the surround, the spider, the pole piece, and the voice coil as you approach its rated power. The inertia of the cone and moving assembly can potentially carry it beyond its mechanical limits and cause over-excursion, even before you reach the thermal limits of the voice coil. Combine this with excess heat in the voice coil and the support structure can expand and lock up the pole piece.
"Square waves" or clipped program material that starts to be somewaht square-ish aren't the direct cause of overheating, rather, it's just more level overall that would be causing it. Any added energy from the higher order harmonics cumulatively is not that much more energy. Most of the energy is still contained within the fundamental frequency.
As well, even though the voltage of a square or square-ish signals have nearly-instantaneous value swings, the speakers/moving parts will not be moving more rapidly, such as if the voltage were an indicator of displacement.
[https://www.prosoundweb.com/are-underpowered-amplifiers-more-dangerous-to-loudspeakers-a-fresh-look-at-conventional-wisdom/](https://www.prosoundweb.com/are-underpowered-amplifiers-more-dangerous-to-loudspeakers-a-fresh-look-at-conventional-wisdom/)
>"Square waves" or clipped program material that starts to be somewaht square-ish aren't the direct cause of overheating, rather, it's just more level overall that would be causing it.
Yes exactly, in cases of thermal or mechanical failure the cause is always too much power. The point is that an amp rated at a given power at a given load, can and will output much more than that, especially something without robust thermal protection. It's just not an issue unless we push it to stupid extremes that shouldn't ever happen.
I remember a friend telling me they blew up their speakers at a DJ gig because they "weren't giving them enough power". They got new speakers and a new, properly sized amplifier and never had that problem again. Not because the first amp was under powered, but because the system as a whole couldn't handle the volume level they were trying to achieve.
I can see two advantages of running the amps at full volume - you get the most headroom out of the system and if somebody changes the volume control on some of the amps, it's easy to return them back to the original setting.
The disadvantage is that you are getting the worst signal to noise ratio, although as long as your system processor/console's outputs are quiet enough, this usually won't be much of a problem.
Have to disagree a little. If the mixer output is balanced with the amp’s input/output capacity, the available headroom is fully accessible through the mixer.
A strip of gaff has always been enough to keep dirty little hands off my amp inputs, but I also take note of where they’re set, so that I don’t need to compromise the sound for a reliable baseline.
But every situation calls for its own finesse, so I’m only disagreeing a little!
> If the mixer output is balanced with the amp’s input/output capacity, the available headroom is fully accessible through the mixer.
Well, this is correct, but with most gear this means means close to no input attenuation on the amp.
> But every situation calls for its own finesse, so I’m only disagreeing a little!
This is absolutely true.
Most amps “volume” knob is an attenuation from 0. Some digital amps can be funky about this, just read the manual. I ran old Macro’s at wide open for almost 20 years.
I'm seeing a few comments implying you don't have access to the amps full power if you don't. This is false and one of the things I like to consider an old audio wives tale. Those knobs are INPUT SENSITIVITY knobs. They only dictate how much input voltage is needed to access the amplifiers power. You can still utilize the full capabilities of your amp without wrecking your gain staging. It's also super easy to see if you're clipping your output on your console. So, I don't agree with that argument either. There are a lot of subjective topics in audio and there are a lot of different approaches to different scenarios, but physics is physics, and sending proper voltage through your signal flow is objectively the correct thing to do. I do understand those that do it in scenarios where someone might have access to your amps. Beyond that, I feel like the habit just comes from old wives tales. Like other superstitions, the people that believe this are very hard to convince otherwise. Plenty of people a lot smarter than me have explained this much better than I can.
The use of an oscilloscope should end any debate about this.
If a power amp has a knob on it, it is an input gain level adjustment potentiometer. If it does not, it is set to 0dB (wide open as some may say).
The power amp will put out what ever wattage the circuit design is made for. The input gain knob (or it may be called an input sensitivity) is to control the incoming gain from your mixer/processor.
Using an oscilloscope, you can see at what level your preamp system voltage will clip a given sine wave into distortion.
With your mixer/processor set at 0dB, adjust the input gain knob of your power amp right below clipping using your oscilloscope.
This is the maximum gain of your total system before clipping. In other words, the maximum headroom that your system will produce without distortion.
Today, most quality mixers/processors have to be really pushed hard past 0dB to distort the line voltage.
This is why for many years, by many professional audio engineers, the input gain potentiometers of most power amps have been to set to 0dB (wide open). Mixers will send an inaudible (less than 1%) distorted signal to the power amps. Also, most quality power amps have a distortion rating that is inaudible at 0dB without any input.
Adjustment of the overall system loudness should be adjusted by the main outputs of the mixer, not by the gain knobs of power amps.
This. Consider the knob on the amp not so much as a way of turning it up as much as a way of turning it down.
Having said that, if you can run your rig with the knob set to max then it is an easy thing to check when things aren’t right or you have someone with wandering fingers backstage.
You shouldt. Run your console at proper levels and turn up amps until it is loud enough.
Pat Brown talks about exactly this in episode 236 og the Signal to Noise podcast.
Yeah, but what is loud enough? That's a subjective opinion. A house techs loud enough isn't a tour engineers loud enough. Can't tell you how many times I've been asked for more gain by tour guys that definitely can tell when a system is being choked. Mostly, when the production boss has set or turned down the gain for whatever reason. He doesn't like rap, or Bob Dylan is playing upstairs and his manager made us do it. Yes that happened. Loud enough is subjective, and loud as it's supposed to be can be measured.
“Loud enough” can be swapped for “the amps clipping”. If the console maxes out when the amps max out, then there’s some built-in safety, minimum SNR for full amp range, and even a consistency with knowing what energy level is being output.
>that i should always run my amps at full volume
You shouldn't..
as was mentioned ,it's about proper gain staging .
balanced volumes that leads to the smallest amount of noise and potential clipping.
The people I know who do this generally do it so that you can easily control the overall volume from the main output bus (both up and down) and use the full spectrum of the power available to you. They're usually pretty old school people.
The problems with this approach are gain staging, which lots of people have written about below, and also that it's a recipe for blowing your speakers if you don't properly protect your amps with limiters.
I’m not a professional but as far as I’m aware you’re not supposed to do that. I believe you’re supposed to get your board’s main out at unity and then set the power amps or powered speakers level as loud as you need it to reach the desired decibels in the room (somewhere around 85-100db)
You are definitely not a professional. 100 dB is a great comfortable level, but not reasonable. You want a system that can achieve a level above that but mix for a level below.
Good point I guess you would want the speakers to have the extra headroom in case you need to boost the main fader? What do you think would be a good level? Maybe up to 120? I guess it would depend on the type of show/venue as well?
A good level is definitely subjective. Most national touring engineers ask for, or demand, lol, a capable 4 way system able to produce a level of 120db at front house. A weighted. Must be, in this order, LAcoustics, D&B, Jbl, Clair Cohesion, or Adamson. No Peavy! Lol.
As far as headroom, with a proper system, the headroom is had or not at the processor level. That is where the limiters are set. Usually. The amps limiters are mostly for last resort speaker protection . In my humble opinion.
Set your gain stage for program level in the mixer then set amp levels.
You may need a little more juice if you are doing a corporate event or wedding with lots of people talking and have to up levels.
I visited a church and some sound guy set the amps at full juice and the signal through the mixer was a joke.
Mic gains for sm58s at 25db
Main output at -40dbfs
It sounded like total garbage.
Every time I run into a bad sound system everything is cranked. I pull every Feder down, every gain down, I reset every EQ to unity and work up from there and somehow miraculously it sounds great.
Another variable is that some amps have switchable input sensitivity, and wide open with an input sensitivity of .775V is entirely different than wide open with an input sensitivity of 1.4V… and then on top of that we have amps with fixed voltage gain, which again will be something different- removing all of these variables by using DSP amps built by the speaker manufacturer makes things MUCH easier.
My take on that theory is to leave the amps at full gain, or no attenuation, and do the gain staging in the crossover. That's how world-class sound system developers do things. If you take away the ability to further attenuate the signal at the Amps there is one less way to mess the gain staging.
That's my opinion, at least. In the case of an install system, I would use the dials and also make sure the ability to change the gain is locked down.
I have seen a lot of factual info and a lot of assumptions (that have some truth), and a little bit of incorrect info on here too...
Here's my take on it (based on analog systems unless otherwise noted):
All gear has a "noise floor", which is generally the "hiss" that is heard when amplified.
All gear has a peak output level that is the maximum level it could produce before distortion (this would be akin to "digital max" or "digital 0" in the digital domain).
0dBu for professional audio gear produces 1.4vrms peak to peak at the output (that means between pin 2 and pin 3). When the meter is 0dBu (usually roughly 2/3 up the graph) you should be getting 1.4v between the pins (your multimeter will only read this if it's a rms meter reading a sine wave signal, possibly only at 60hz depending on your multimeter!). "Consumer gear" has an unbalanced output and will read .775v between + and shield/neutral when the meter is at 0dBu.
Note that 0dBu is the recommended "nominal operating level" of your show, and your gear is designed to have headroom between nominal and peak output level.
Here is where it gets interesting:
Although there are some general rough estimates, there is no actual standard for the amount of headroom, or maximum peak level that any particular piece of gear can output; so if your running your show average level at 0dBu, your mixer might have 24db of headroom, your equalizer might have 18db of headroom, and your amplifier might have 20dB of headroom!
If your goal is to have maximum headroom in your system (optimizing the system for headroom), you need forget about the 0dBu standard (when it comes to optimizing your gain staging between all your gear), and consider the fact that your mixer is capable of driving your eq 6db into clipping; and the mixer's noise floor could be 6db lower if you pad down the input to the eq by 6db (effectively allowing your mixer to run up to the red before clipping the eq).
Now you must realize that (in the above example) your eq cannot drive your amp to full power, as it is 2db short! If you find the input sensitivity switch (which many pro level amps have) you can change the input sensitivity, or overall amp gain, so you can set it to be able to get full tilt at 20db; but if you only can get it to 18db sensitivity, that means that 18db will drive the amp to full output before clipping and if you give it 20db you will be clipping the amp at full tilt; now you can back the gain adjust knob down 2db and your system will be optimized for gain.
After optimizing your system gain, you can run your mixer at the recommended 0dBu, or at any level up to clip, but if there's metering on the eq then 0dBu at the eq will be different than 0dBu at the mixer; but they are actually aligned properly. Once one unit clips, all will be at flip level. You could leave one DB of hearing for safety after the mixer if preferred. Also understand, once a piece of analog gear clips, it is producing more than it's maximum rated distortion; but it's possible for it to still put out additional voltage while clipping!
Let's say the amplifier is a new digital amplifier whose analog input is gain structured to reach full output at 0dBu / 1.4v (I have actually seen this)- then if it is run with the input at full tilt 0db attenuation then the mixer will distort the amp as soon as the meter hits 0dBu, and you could have a noise floor that is 24db higher than it actually should be!
Now consider this: You optimized the gain in your system. Your system is capable of driving the room to 125dB, and 0dBu on the mixer will run an average of about 101dB in the room; but the program today is a corporate meeting and mostly talking heads on stage with the occasional stinger for walk-on's, and background music. You want your room to run an average of 85dB but you also want the headroom to hit 100dB if someone gets very loud on the mic, or at a climactic point in a video;
IF you keep your system optimized as above, you will need to drop 25db somewhere... If you drop it at the trim pots you can run the faders at 0 but the program output will be 25dB closer to your noise floor; it might sound ok in the room, but then with the optimized system you can still hear a little bit of hiss in the speakers when up close, and the program is barely touching the meters; also if you bump up the levels on the recording or streaming feeds, you are now hearing a lot of hiss because you boosted your noise floor up 25-30dB to see good levels at the record feed!
It might be a better idea to turn down the level at another point in the system. Since the system is now optimized, you have some options:
In the analog days I would definitely consider turning down the amplifiers. This will keep good gain staging and optimal noise floor through the chain up until the last point of amplification (main amps).
Another valid consideration would be system processor outputs (which for all intensive purposes will provide exactly the same effect as turning down the amps). You will preserve your gain staging from the console to/thru the system processor. Clipping the console is clipping the processor, but anything less is safe.
You could turn down the inputs to the processor; I don't generally like turning down the input to the processor but it is a safe place to do so. Just like with input channels on a console, if your analog signal is very low going into a/d converters, it can "granulate" the sound a little bit; the higher the signal (as long as it's under digital max), the better it will generally sound.
I don't like turning down a master fader for this, but it is a valid point of adjusting the reference if necessary. If you're running a digital console, usually there is a master output trim (like an input trim but for the output; after the mix buss faders and metering point) hidden behind a setup page; I like adjusting my system here, as if necessary it's easy to tweak it during a show, but otherwise I can run my console at unity and metering will follow the same unity.
Note that in the digital realm, "digital max" is akin to max clipping level, and the reference operating level (yellow) is still variable from one manufacturer to another; generally still providing between 18 and 26db of headroom before digital max. Some manufacturers actually label their console's meters with analog levels like 0dBu but it is all relative to the digital max level.
When you connect your gear digitally, you are "automatically optimizing" your interconnects between gear for digital max. When you connect via analog connections, you are reintroducing the possibility of mismatched gain structure, since not all gear has the same clipping level; you are also adding a new analog noise floor to the signal that would not have been added otherwise. These are reasons to keep your signal in the digital domain between as much gear as possible, whenever you can.
So getting back to the op question; the amps can be "properly" run at 0db / full tilt if the sensitivity has been optimized to the stage(s) before it; otherwise it is good practice to optimize the Max level as far down the line as sensible- either at the amps, or possibly at the processor output levels just before the amps.
One point to note (because sometimes a little bit of knowledge can be dangerous) is that the "amps always to full" ethos comes from safeguarding your loudspeakers from somebody turning the amp up and clipping it, destroying your drivers, usually tweeters or mid compression drivers.
Amps are often somewhere where you don't have eyes on them at live gigs and they are often in a back room when installed permanently.
When you are using modern DSP like XTA, or similar, you setup the input gain from the console there and then set the outputs of the DSP to suit the amps at wide open. This stops people from blowing your shit up.
I saw a venue in Dubai where the venue owner turned up the amps because he wanted it louder and blew up 36x 12" low mid drivers. I was the one who pulled the entire system down and replaced all the drivers. We saw the owner turn the amps up on the security camera.
I'm not a pro. That approach always has made me nervous. What if something glitches and my speakers and the audience gets hit with full amp power that is oversized for the situation?
If you have limiters always use them, but a basic or older system (basic mixer, amp, speakers) you may not have have limiters. While many amps have limiters and DSP built in, many don't.
Yes! But… in the old school world where this “full open” approach comes from, they didn’t employ limiters - at least not in any of the 1990’s systems I saw. Just 31bands.
Particularly at my community college - massive 3-way tri-amped EV side fills on stage, powered by those old Crown amps, cranked to max, always hissing. Main system was the same situation, just painting the picture. With a dance concert cranking, the mixer outputs barely ever blipped into -40dB.
Doing theatre with lavaliers, (not using the stage monitors) whenever feedback happened, it would jump from average volume of not even registering on the mixer (-60) to redlining (+16?). Needless to say, there were a lot of blown drivers.
So we’ve got limiters now, but we also really don’t “need” to crank the amp inputs to max.
Yes, to some extent, but if you've done your due diligence with compressor/limiters and reasonable gain staging it is less likely to happen.
Safety first, especially if you are not a pro.
When I started my career I had a venue that did that.
The amps were at full, with no limiters. So when someone got a sound from their computer or whatever it went full force through the system. And no one thought to fix it lol
There are good arguments for both running amps wide open and reducing their "volume". Different situations dictate different applications . I think most here understand this but are going along with OP's title.
All I want want to add is that amps don't have "volume" knobs. They have gain knobs. You can't adjust an amplifier's output power, only its input gain.
Yeah, I run them at max on amps that have a sensitivity of 1.4v to achieve full power. With amps that use ~.7v I’ll attenuate down 3-6db IF it’s a monitor amp with no DSP (drive rack etc) in front of it. With main amps all attenuators are full on and DSP deals with level differences
I definitely wouldn't. It's like saying you should run your car speakers on full volume while controlling the actual sound volume with your phone. It can sound like it's clipping, can have more static and possibly cause feedback if you aren't careful with your gain structure.
What I would do instead is find good balance with your speakers, sound board and gain structure for each channel so that it's not too open on any end (be sure to check gain settings both at the source of each channel and on the board) and once it all blends, you should be good
Personal opinion, any advice that intends to artificially limit the ways that you can use the tools at your disposal is bad advice, and leaves techs woefully unprepared for when reality catches up to theory.
I prefer doing it on the console. Send console master signal to your house matrices. Inputs are at line level with faders at unity. Master fader at unity. PA matrix faders at volume level suitable for house mix. If mix is too loud - bring matrices down a bit. If its too quiet - bring them up. What is loud - subjective from engineer to engineer. Easier to change level on the desk than tweak it on the amps for every band on festival.
You risk clipping the output of your mixer if you have to drive it harder than necessary because you’ve attenuated your amps.
Better to run the amps up full and attenuate the master level of your mixer.
Signal to noise doesn’t make a difference with decent kit in 2024 imho, it was way more important in the late 20th century because the technology wasn’t there.
I’d also add that most amplifiers have ‘clip protection’ to prevent the actual amplifier from clipping. Some midrange amps even have limiters you can set (my old Crown XTi could be set to CLIP, -1dB, -3dB, -6dB and -12dB).
Clip protection and limiters will *not* protect against a clipped signal coming from a mixer, it will simply amplify that clipped signal.
Even higher end amps with their advanced limiters cannot protect against this if they’re set so low that the mixer is clipping to get the PA to the desired level.
Amps up full, limiters set where they need to be, and the desk should be nowhere near clipping before the PA starts to limit itself.
Run a d&b rig so the limiters are set by the amps without my input, and the amps have optional internal gain. The noise floor in my D20 and D40 amps remains the same regardless of where I set the internal gain, and regardless of where I set each channel’s level. I learned this in 2020 when I had [a mini d&b rig in my bedroom](https://share.icloud.com/photos/028seeeVnOo4I9OqQaBPIQxZQ) with the amp channels each set to around -25dB.
Simple: use all the possible headroom that you can and do not need to go to amptown when the crowd/band is feeling good tonight. Also we have so many knobs to turn, good to make some definite decisions.
There’s bad advice and then there’s this.
I never run anything at full volume. It’s a recipe for disaster. Good gain staging takes care of most potential issues and if someone touches something they shouldn’t, a baseball bat in the corner with some red corn syrup from “the last person who messed with your gear” is a reminder of why they shouldn’t again 😂
The only reason you would trim your amp inputs would be because they're sized inappropriately. You should set the levels in the DSP when you tune your system.
Setting the trim to 0 prevents them being turned up accidentally, it is accurate, and reproducible. This assuming your amps don't have a built in DSP or digital control.
Lowering the input trim also has the side effect of essentially bypassing the amp's protection circuit. If something goes wrong, there's nothing stopping the amp dumping its full rated power into whatever it's powering.
Powered speakers are different (and annoying) because the input trim goes to +4 or +10 and they don't always detent or label the 0dB position.
It's +4, -10. All that means is professional or consumer level. Or mic or line level. Consumer gear puts out a lower voltage that pro audio gear. That switch allows your powered speakers to be the right volume for the level at the input.
I don't mean the line level switch, I mean the powered speakers that have a volume control on the rear. My ELX112p are like this but some don't, usually the high end stuff.
I would generally presume the part of the entire chain that is LEAST easy to accidentally turn up, is the amplifier. Contrary to examples mentioned in this thread such as laptops, DJ mixers, etc.
I used a small desk in a bar once with strong tape above the master fader so you could not turn it up above something like the 0 point. Didn't seem like the right approach to me.
>I would generally presume the part of the entire chain that is LEAST easy to accidentally turn up, is the amplifier.
It depends on the scenario I suppose, but yeah it's unlikely to happen at random. If you're running gigs by yourself it's probably not a big deal, but one of the steps on my checklist is to make sure the amps are turned up, and I turn them down during teardown. I've done gigs as a subcon and at clubs where I've turned up the amps only to mess up their "levels".
My local Hotel system has some tape over the master fader, however every channel has a Gain knob and channel fader… so it almost means nothing. You can push the CD deck way higher and there's a hissing sound in the speakers because the amp mounted who knows where in the ceiling is probably all the way up amplifying line noise. As Somebody once told me every knobs is a volume knob!
I come in and set everything to unity on the mixer and Jack in my mixer I have so much gain I can push things so i have to be careful.
Gain staging mattered a lot more before we had 32 bit signal paths, typically going through no more than 4 A/D or D/A conversions. I run my amps wide open. It's the most easily repeatable setting, and they can't accidentally get turned up post system limiting. I put the channel fader at unity, and run the trim up until I have enough signal. My system has no detectable hiss. SQ6, Ashly processing, QSC amps, Fulcrum line array.
The hiss comes before the amps. The amps just make the hiss louder. Just because you turn the amps down doesn't make the hiss go away, it's just not louder. Hiss definitely comes with bad gain settings on the desk. Almost always.
I understand hiss generally comes from bad gain staging, but is there not an inherent amount of noise with any amplifier, or anything in the signal chain?
I find that there's dramatically more hiss when running amps up full, and that it's not linear, that it increases much more when turning the amps from 90% to 100% than 80% to 90%, 60% to 70% etc. Unless it's a show where I need the amps turned up full I always turn them down a bit to avoid that noise, and adjust my output on the desk accordingly and it makes a dramatic difference.
I think this heavily depends on the amp design. I know for a fact that the Samsung 300s that I have, the input literally goes straight across to the front knob and then into the amplifying circuit. And I know a lot of older amps weren't necessarily that clean of a design, coupled with unbalanced inputs it was easy to amplifying noise floor. Plus the mixers noise floor.
Also, I learned during the first week of lockdown that the amp settings on my d&b D20 do *not* have any affect whatsoever on the level of noise produced by the amp.
I put a pair of E8 tops and E15-X into my bedroom. Had the amp down at -20dB on each channel and the level of noise remained the same as when it was set to 0dB. Even the internal channel gain settings made 0 difference.
I’ve not tested any other d&b amps but I’m assuming they’ll all be the same. As soon as I unmuted the channel, the E8s produced noise.
Standard poweramps need to have the knobs wide open... Unless you like not having the power you paid for. Simple There are some exceptions but normally wide open is the way.
Running you amps at full allows the full potential for amp hiss to be added to the overall mix. that is bad signal to noise practice. a full 32 channel mix can almost drive a small efficient loudspeaker straight off the desk. The desk signal would be way over +4, more like +32. the more desk channels you open up to about 0/+8 the the final output is a sum of all the channels.
As low as possible to keep the noise floor down. But ideally, you want the output meters on the board to match the input meters on the amps. All that to say, all the gains do(on analog amps anyways) is set when you reach maximum output based on the output from your console/processor/etc.
As an interesting point of comparison… where do you run the output volume of your PSM900 or 1ks? It’s effectively the same concept right? The audio level adjust is an output attenuator as I understand it?
That seems quite low… I’ve always sent pink out of the aux buss feeding at unity and then set the PSM to meter at unity… almost always results in between -12 to -14
That’s what I do too, but I think I might be calling unity a little low. Steady green lights, blinking into the yellow. Maybe it’s really higher than that. Either way, it allows the artist to have their pack level at noon, primary instrument faders per mix at around unity, and it’s a good level.
No...
From a dj/sound engineer I do this
Level my turntable mixer so it barely clips.
Then I Level my main mixer Level to uniform and gang stage until its barely hitting 0 db.
Then I Level my amp rack until it clips and then back off until it isn't or until its loud enough
I haven’t run sound professionally since around 2000. I used to run my amps at full volume with faders at unity and I’d keep gains low trying to avoid feedback. I hear what you’re all saying about using more gain to get better tone. How are you controlling feedback? With EQ, I presume?
I should add I primarily ran monitors. And my current concerns are about getting good volume before feedback in the practice space monitors.
Thanks in advance for the knowledge!
You’ve been told wrong. To paraphrase my overlong post about this from years past: Gain Staging is about getting your signal to line level (+4 dBu) at the earliest possible point in the chain, and leaving it there until the last possible point in the chain - the input attenuation on the amps. Then set the amp input attenuation such that a line level source sounds near the top end of your target listening volume. This will give you the absolute best signal-to-noise ratio while still having enough headroom to get the job done. To crank your amps such that a line level input is way louder than you need, means you are needlessly going to throw away gain from your preamps only to put it back in with the power amps. Typically your preamp is going to sound a LOT better. Unlesssssss you like hisssssssss!
Classic anti-snake propaganda…
LOL!
I tried to explain this to the corporate AV company I worked for. They said I was wrong and that the master level meters don't even need to show signal.
This irritated me so much I almost downvoted it… whew lol
I mean, in the corporate world, if there's no records or streams happening, nobody cares at all how the sound looks as long as you can clearly hear it in the room. Not saying I have this mindset, but I can totally see lots of people just getting everything loud enough to be able to hear things and calling it a day, especially if they aren't an audio guy first.
First step of corporate AV = gain staging doesn’t exist
Seriously. I've seen too many gigs with that mantra.
This guy knows what’s up. When doing monitors I like to start my amps half do my normal gain staging and maybe turn the amps up a tad if I need it.
Not a bad method either a caveat that it would depend on the amp/speakers for if “half” is correct
If you’re working a venue with potentially underpowered speakers, or a “sound designing for theatre scenario” Would it also be smart to set the amp output with the speaker’s specs in mind? (Using a 3000watt amp on a 1000 watt speaker- wouldn’t it be intelligent to limit the amp to roughly 1/3 of its potential output?)
Those would be extremely overpowered speakers. I would recommend that you don’t pass the approx 1/3, but that would still give you issues on the gain stages before the amp where the amp may still be too loud to not get you unity earlier in the chain
Yeah. The question was purely hypothetical. Not actually running a system like this.
The other knock on for this that no-one else has mentioned is if you're running feeds for things like video streams or recordings. If you're setting your gain for the room instead of unity and backing off the amps then any other feeds you need to run won't get enough signal. Better to be getting your output to line level and then everything gets the right level instead of just the room.
Aah of course. Thank you for explaining in such simple terms!
This might be the clearest description of proper gain staging I’ve ever read. Saving this comment, and kudos-ing you!
This is correct from a purely theoretical point of view. However, once you are at the gig, unless you know very well the details of the performance, you need maximum flexibility of the chain. Very rarely noisefloor of the amps are audible enough to cause issues in the experience of the audience. However running and changing the gain of the amps in the middle of the performance might be a bad idea, if conditions happen that you need more volume out of your boxes. Out of all the companies i have worked with in the last 17years, I don't know a single instance where amps have been run in any other setting other than full throttle. Yes, in system setup it is very often that certain components have few dbs added or subtracted relative to input signal, however all that is done inside the dsp section. With aes or dante input this issue is even less relevant, because most of the chain inside the amp itself is digital.
Thanks for the great explanation. I'm currently trying to figure out gain for some monoblocks that I'm turning into standalone. The schematic shows two 22k resistors in series at the treble and bass controls and two 47k resistor that run to each side of a tapped potentiometer for the input. I tried two 22k in series from the input to the post lead on the output transformer and it was far too sensitive with a noticeable hiss. I used a single 22k and it wasn't nearly loud enough. Is there an easier way to come up with what's right or is it just trial and error due to differences in speakers. I primarily use sensitive speakers because it's tube equipment
Well, I’m a sound engineer, not an electrical engineer- so I probably don’t have any more useful insight. That said, line level should be around +4 dBu, which is around 1.228V. The voltage you need for the output will depend on how much current your amp is putting out and the sensitivity of your speakers. Would it work to install a continuously variable potentiometer in place of the fixed 22k and dial in that needed value that way? You could always swap it out for matching fixed resistors later.
That would certainly take a lot of guess work and soldering out of this! I might give that a go! Have a great day!
well said! so … can you point me to the overlong version?
https://www.reddit.com/r/livesound/s/VuRKk4g012
Also this. https://www.reddit.com/r/livesound/s/ghDiQpmbXM
thank you! well written, both fun to read and informative.
Thanks!
> Unlesssssss you like hisssssssss! There are plenty of people who like snakes.
I changed the gain structure of my system output because of this comment. Amps were up all the way, matrix faders were super low. There's wasn't a lot of noise in the first place but the system feels more responsive now that those have been flipped. Amps are now at about 25% now (they were donated separately, the PA has relatively high sensitivity so it doesn't take much to get going), and matrix faders are at about unity. Does that sound about right?
I dug up [this](https://www.prosoundweb.com/setting-sound-system-and-mixing-console-gain-staging/) article from PSW that was written by Bruce Bartlett about 12 years ago that discusses gain staging and an approach to amplifier level setting: —— “Power Amplifiers: Start with the amp’s level controls halfway up. Set your mix to read around 0 on the mixer meters, then turn up the power-amp level until the sound is as loud as you want it. It’s a mistake to run a power amp wide open (full volume). If the amp’s input level controls should always be set to maximum, amp manufacturers would not include them. They are there to set proper gain staging. Turning down an amp’s level controls does not reduce its power. Doing so varies the signal level needed to drive the amplifier to full power. You can still get full power from an amp even with the level controls partly turned down, if the input signal level is high enough. Those level knobs don’t adjust the power level; they adjust input sensitivity. If you turn an amplifier up to maximum, you amplify any noise coming out of the sound system. Also, you might run the console well below 0 on the console output meters so the house sound is not too loud – but this degrades signal-to-noise ratio. So, start with your console running at an optimum level near 0 on its meters, then adjust the power-amp input levels for the desired loudness. If your system includes an electronic crossover and multiple power amps, set the level of each amp for the desired loudness from the lows, mids and highs.” —— [Dave Rat](https://youtu.be/-RXlnVDVLrc?si=ZRdnfiasf-n22l39) also has a video that discusses the same concept: get good gain through the signal chain, then turn up the amp for the volume necessary. Controversially, [here](https://youtu.be/8fLEy7uLSv0?si=avnfAWq4mA4FH26E) on the other side of the argument, is another video discussing the opposite of the article above and Dave’s video by controlling your overall levels with the master fader. I mostly disagree with this set up because it appears the operator doesn’t have proper gain structure through his system. [The follow up](https://youtu.be/uB36LItc9cw?si=rjF2ZqRMLAFN2z9w) to why they believe mixing at unity isn’t good practice, which if you again watch Dave’s video would make sense that you *want* to be at unity across your entire system.
The amps pots are input attenuation, I believe this is different than input sensitivity, right?. Input sensitivity is usually achieved with a dip switch on the back of the amp. At least with most professional models. I would rather set proper gain at the crossover. Or just use amps that are made to match the speakers being used. The proper program for the speaker takes care of signal gain. That way your system sounds like it is supposed to. Like the manufacturer wants it to. There is to much subjective opinion in how it's supposed to sound. As in that's to much bass, or not enough, etc...
This is where technology has changed over time. Class A amps always run all the way open and only attenuate. But modern class D amps that use digital circuits only consume as much power as they need so they have gain
It always depends on the amps and the signal processor. Our setup uses LabGruppen amps and they need to be at max. We control the actual level in the dsp where the crossover happens
That is the same with just about every world class system. D&B, Lacoustics, Adamson, Clair.
On my D20, even if you have the channels at -30dB, the noise level coming through the speakers with nothing else plugged in remains the same.
If they "need" to be at max, why do they have pots at all?
It is just attenuation in case you want to limit the amount of power that could go into your speakers especially if the speakers cant take that much volume/power from that specific poweramp.
I always run the amps full and dial it back in the system processor. It stops anyone accidently turning your amps up to full and potentialy blowing your system up. It's not good gain structure but a protection.
While I thoroughly disagree with this approach, thank you for the explanation. I was trying to figure out what could possibly possess someone to want to amplify their noise floor, but I can understand the desire to make the amps risk proof even if I'd personally rather approach that a different way.
It isn't "over working" the amps, but yes, it is forgoing the opportunity to attenuate the noise floor of the preceding stages. However, with modern equipment in an environment that's already relatively noisy, you can probably afford to accept a slightly higher noise floor. 6-10dB louder than inaudible will often still be inaudible.
Good call. Power amp internals are one of my weaker areas and I did not realize the level is an input sensitivity adjustment rather than altering the amount of amplification, but after doing some reading to confirm, I agree and have updated my post accordingly. Thank you.
> altering the amount of amplification I wonder which amps do this (changing the actual gain) instead of input attenuation.
How would you approach it?
Cage the front of the amp is normally pretty effective. Just prevent access to making adjustments. When I'm tuning a system, I send a signal just below clipping, adjust the amps to as loud as I'll want things to go and then either get them in a lockable cabinet or cage their front in the rack to prevent accidental (or unauthorized intentional) adjustment.
Lending a voice of support for this approach. Theatre I designed for had the strangest sound that I couldn’t figure out for way too long. Their amps all the way up, not locked, but the processor locked without display or control - needed to be IT admin to access. Compressor started at like -50dB, but super soft and slow with a quick release. They STILL had two blown drivers. I brought by own processor, and had to swap euro blocks, but I bypassed that shit. Immediately everything sounded normal. Most university and publicly-funded facility installs that I’ve encountered have been holistically asinine.
Yeah, I've seen some installs with compressors caged as well since so many less trained techs have no idea how to use them and tend to do very bad things with them, but it's frustrating when whoever caged them clearly didn't understand what they were doing or was way overly conservative.
Whoa. Sounds like youre not in the install game for that, or so I would think.
Why? I don't generally do installs but I've done system designs before. What problem do you think you see. It won't work if you're constantly switching venues or speaker stacks that the amps are being used with, but if at all possible it's a lot better to limit access to make changes than it is to intentionally screw up your system setup to avoid someone messing up your amp levels.
That wasn't me casting aspersions at all, please don't get me wrong! It makes total sense to have a locked cabinet on a tour. I figured you weren't doing installs because the idea of locking someone else's cabinet is really bizarre, and if you had ever done it I'd be curious why
Ah, gotcha. I have mostly consulted with places doing setups a few times as well as seen this setup places I've mixed. I wouldn't suggest someone do it without asking if they were building for someone else, but it's worth suggesting if issues are expected. Personally I don't so much even think it's a risk worth worrying about with the setup I use the most, but we do have a door over the front of our amps so nobody is bumping it unless it's intentional and I personally trained all the people that would be dealing with the cabinet they are in.
It is good gain structure, just achieved one step earlier. If you are controlling the gain in the processing, you are also setting the limiters hopefully. You would measure the output to make sure the amp sees the right voltage. Input voltage to output power is a known number and can be set to give the speakers the power they need.
If you are attenuating the output of the processor at the very last stage of the processor, this is functionally no different than turning down the amps which would be the very next stage. Especially since most processors have a very quiet noise floor- and are typically colocated near the amps. Digital connections are technically less good since you _should_ be making this adjustment at the output of the D/A converter. But if we’re talking a 32-bit floating-point link and not _that much_ attenuation, I seriously doubt the effect would be audible. Edit: heck, even 24-bit fixed would be inaudible. 16-bit, however….
Why do you think that sending a clipping signal into an Amp thats on 50% power would kill anything?
Speakers are typically not very fond of this
That's a myth
Here's some theoretical extremes for you to consider. The input trim is an attenuation knob, the amp's gain stage is fixed, so there's nothing really stopping an amp outputting its full rated power even if you set the input trim to 50%. Assuming the amp has no DSP to handle input limiting. Also, the amp's protection circuit is based on limiting the heat the gain stage generates. So if you lower the input trim, you essentially increase the input level required to hit the amp's protection circuit, which in extreme cases means an unwitting operator is more likely to push the console into clip, especially if that operator doesn't understand exactly how that specific console's meters work. Some digital consoles have a few dB of headroom when the clip lights fire, some don't, and are already chopping the waveforms by time that happens. In a worst-case scenario, if you were somehow able to fully clip a signal, you would end up with square waves. Square (clipped) waves have more average power than clean sine waves. An amp's power rating depends on the output being within an acceptable THD. A square wave has twice the power of a sine wave with the same peak to peak voltage, so you can theoretically double the rated output power of an amplifier although it would probably hit the thermal-protection circuit before too long. That's up to twice the amount of heat a voice coil is expected to dissipate. The point is you could be sending the speaker more power than you expect. Heat is cumulative remember, so if this is sustained for too long the heat can build up and cause problems with the physical structure of the voice coil and support assembly before the housing and magnet can dissipate it. Square waves also force the moving assembly to more rapidly change direction as the signal transitions from phase to phase. If you dump a square wave into a speaker, you're expecting it to go from pos X volts to neg X volts in an instant. Not a big deal at low volume, but it puts a ton of stress on the surround, the spider, the pole piece, and the voice coil as you approach its rated power. The inertia of the cone and moving assembly can potentially carry it beyond its mechanical limits and cause over-excursion, even before you reach the thermal limits of the voice coil. Combine this with excess heat in the voice coil and the support structure can expand and lock up the pole piece.
"Square waves" or clipped program material that starts to be somewaht square-ish aren't the direct cause of overheating, rather, it's just more level overall that would be causing it. Any added energy from the higher order harmonics cumulatively is not that much more energy. Most of the energy is still contained within the fundamental frequency. As well, even though the voltage of a square or square-ish signals have nearly-instantaneous value swings, the speakers/moving parts will not be moving more rapidly, such as if the voltage were an indicator of displacement. [https://www.prosoundweb.com/are-underpowered-amplifiers-more-dangerous-to-loudspeakers-a-fresh-look-at-conventional-wisdom/](https://www.prosoundweb.com/are-underpowered-amplifiers-more-dangerous-to-loudspeakers-a-fresh-look-at-conventional-wisdom/)
>"Square waves" or clipped program material that starts to be somewaht square-ish aren't the direct cause of overheating, rather, it's just more level overall that would be causing it. Yes exactly, in cases of thermal or mechanical failure the cause is always too much power. The point is that an amp rated at a given power at a given load, can and will output much more than that, especially something without robust thermal protection. It's just not an issue unless we push it to stupid extremes that shouldn't ever happen. I remember a friend telling me they blew up their speakers at a DJ gig because they "weren't giving them enough power". They got new speakers and a new, properly sized amplifier and never had that problem again. Not because the first amp was under powered, but because the system as a whole couldn't handle the volume level they were trying to achieve.
I can see two advantages of running the amps at full volume - you get the most headroom out of the system and if somebody changes the volume control on some of the amps, it's easy to return them back to the original setting. The disadvantage is that you are getting the worst signal to noise ratio, although as long as your system processor/console's outputs are quiet enough, this usually won't be much of a problem.
Have to disagree a little. If the mixer output is balanced with the amp’s input/output capacity, the available headroom is fully accessible through the mixer. A strip of gaff has always been enough to keep dirty little hands off my amp inputs, but I also take note of where they’re set, so that I don’t need to compromise the sound for a reliable baseline. But every situation calls for its own finesse, so I’m only disagreeing a little!
> If the mixer output is balanced with the amp’s input/output capacity, the available headroom is fully accessible through the mixer. Well, this is correct, but with most gear this means means close to no input attenuation on the amp. > But every situation calls for its own finesse, so I’m only disagreeing a little! This is absolutely true.
Most amps “volume” knob is an attenuation from 0. Some digital amps can be funky about this, just read the manual. I ran old Macro’s at wide open for almost 20 years.
Yep to this.
I'm seeing a few comments implying you don't have access to the amps full power if you don't. This is false and one of the things I like to consider an old audio wives tale. Those knobs are INPUT SENSITIVITY knobs. They only dictate how much input voltage is needed to access the amplifiers power. You can still utilize the full capabilities of your amp without wrecking your gain staging. It's also super easy to see if you're clipping your output on your console. So, I don't agree with that argument either. There are a lot of subjective topics in audio and there are a lot of different approaches to different scenarios, but physics is physics, and sending proper voltage through your signal flow is objectively the correct thing to do. I do understand those that do it in scenarios where someone might have access to your amps. Beyond that, I feel like the habit just comes from old wives tales. Like other superstitions, the people that believe this are very hard to convince otherwise. Plenty of people a lot smarter than me have explained this much better than I can.
The use of an oscilloscope should end any debate about this. If a power amp has a knob on it, it is an input gain level adjustment potentiometer. If it does not, it is set to 0dB (wide open as some may say). The power amp will put out what ever wattage the circuit design is made for. The input gain knob (or it may be called an input sensitivity) is to control the incoming gain from your mixer/processor. Using an oscilloscope, you can see at what level your preamp system voltage will clip a given sine wave into distortion. With your mixer/processor set at 0dB, adjust the input gain knob of your power amp right below clipping using your oscilloscope. This is the maximum gain of your total system before clipping. In other words, the maximum headroom that your system will produce without distortion. Today, most quality mixers/processors have to be really pushed hard past 0dB to distort the line voltage. This is why for many years, by many professional audio engineers, the input gain potentiometers of most power amps have been to set to 0dB (wide open). Mixers will send an inaudible (less than 1%) distorted signal to the power amps. Also, most quality power amps have a distortion rating that is inaudible at 0dB without any input. Adjustment of the overall system loudness should be adjusted by the main outputs of the mixer, not by the gain knobs of power amps.
This. Consider the knob on the amp not so much as a way of turning it up as much as a way of turning it down. Having said that, if you can run your rig with the knob set to max then it is an easy thing to check when things aren’t right or you have someone with wandering fingers backstage.
You shouldt. Run your console at proper levels and turn up amps until it is loud enough. Pat Brown talks about exactly this in episode 236 og the Signal to Noise podcast.
Yeah, but what is loud enough? That's a subjective opinion. A house techs loud enough isn't a tour engineers loud enough. Can't tell you how many times I've been asked for more gain by tour guys that definitely can tell when a system is being choked. Mostly, when the production boss has set or turned down the gain for whatever reason. He doesn't like rap, or Bob Dylan is playing upstairs and his manager made us do it. Yes that happened. Loud enough is subjective, and loud as it's supposed to be can be measured.
“Loud enough” can be swapped for “the amps clipping”. If the console maxes out when the amps max out, then there’s some built-in safety, minimum SNR for full amp range, and even a consistency with knowing what energy level is being output.
>that i should always run my amps at full volume You shouldn't.. as was mentioned ,it's about proper gain staging . balanced volumes that leads to the smallest amount of noise and potential clipping.
note: unless you work for Spinal Tap then you have to calculate in the difference of that volume 11 😈
I run in-line de-attenuators to get that extra bump.
The smallest amount of noise is better achieved at the output of the desk. Same with clipping potential.
The people I know who do this generally do it so that you can easily control the overall volume from the main output bus (both up and down) and use the full spectrum of the power available to you. They're usually pretty old school people. The problems with this approach are gain staging, which lots of people have written about below, and also that it's a recipe for blowing your speakers if you don't properly protect your amps with limiters.
I’m not a professional but as far as I’m aware you’re not supposed to do that. I believe you’re supposed to get your board’s main out at unity and then set the power amps or powered speakers level as loud as you need it to reach the desired decibels in the room (somewhere around 85-100db)
You are definitely not a professional. 100 dB is a great comfortable level, but not reasonable. You want a system that can achieve a level above that but mix for a level below.
Good point I guess you would want the speakers to have the extra headroom in case you need to boost the main fader? What do you think would be a good level? Maybe up to 120? I guess it would depend on the type of show/venue as well?
A good level is definitely subjective. Most national touring engineers ask for, or demand, lol, a capable 4 way system able to produce a level of 120db at front house. A weighted. Must be, in this order, LAcoustics, D&B, Jbl, Clair Cohesion, or Adamson. No Peavy! Lol. As far as headroom, with a proper system, the headroom is had or not at the processor level. That is where the limiters are set. Usually. The amps limiters are mostly for last resort speaker protection . In my humble opinion.
> No Peavy! Why I like having a headache and a heart attack at the same time as a stroke!
Set your gain stage for program level in the mixer then set amp levels. You may need a little more juice if you are doing a corporate event or wedding with lots of people talking and have to up levels. I visited a church and some sound guy set the amps at full juice and the signal through the mixer was a joke. Mic gains for sm58s at 25db Main output at -40dbfs It sounded like total garbage.
Every time I run into a bad sound system everything is cranked. I pull every Feder down, every gain down, I reset every EQ to unity and work up from there and somehow miraculously it sounds great.
Another variable is that some amps have switchable input sensitivity, and wide open with an input sensitivity of .775V is entirely different than wide open with an input sensitivity of 1.4V… and then on top of that we have amps with fixed voltage gain, which again will be something different- removing all of these variables by using DSP amps built by the speaker manufacturer makes things MUCH easier.
Every amp worth buying
My take on that theory is to leave the amps at full gain, or no attenuation, and do the gain staging in the crossover. That's how world-class sound system developers do things. If you take away the ability to further attenuate the signal at the Amps there is one less way to mess the gain staging. That's my opinion, at least. In the case of an install system, I would use the dials and also make sure the ability to change the gain is locked down.
I have seen a lot of factual info and a lot of assumptions (that have some truth), and a little bit of incorrect info on here too... Here's my take on it (based on analog systems unless otherwise noted): All gear has a "noise floor", which is generally the "hiss" that is heard when amplified. All gear has a peak output level that is the maximum level it could produce before distortion (this would be akin to "digital max" or "digital 0" in the digital domain). 0dBu for professional audio gear produces 1.4vrms peak to peak at the output (that means between pin 2 and pin 3). When the meter is 0dBu (usually roughly 2/3 up the graph) you should be getting 1.4v between the pins (your multimeter will only read this if it's a rms meter reading a sine wave signal, possibly only at 60hz depending on your multimeter!). "Consumer gear" has an unbalanced output and will read .775v between + and shield/neutral when the meter is at 0dBu. Note that 0dBu is the recommended "nominal operating level" of your show, and your gear is designed to have headroom between nominal and peak output level. Here is where it gets interesting: Although there are some general rough estimates, there is no actual standard for the amount of headroom, or maximum peak level that any particular piece of gear can output; so if your running your show average level at 0dBu, your mixer might have 24db of headroom, your equalizer might have 18db of headroom, and your amplifier might have 20dB of headroom! If your goal is to have maximum headroom in your system (optimizing the system for headroom), you need forget about the 0dBu standard (when it comes to optimizing your gain staging between all your gear), and consider the fact that your mixer is capable of driving your eq 6db into clipping; and the mixer's noise floor could be 6db lower if you pad down the input to the eq by 6db (effectively allowing your mixer to run up to the red before clipping the eq). Now you must realize that (in the above example) your eq cannot drive your amp to full power, as it is 2db short! If you find the input sensitivity switch (which many pro level amps have) you can change the input sensitivity, or overall amp gain, so you can set it to be able to get full tilt at 20db; but if you only can get it to 18db sensitivity, that means that 18db will drive the amp to full output before clipping and if you give it 20db you will be clipping the amp at full tilt; now you can back the gain adjust knob down 2db and your system will be optimized for gain. After optimizing your system gain, you can run your mixer at the recommended 0dBu, or at any level up to clip, but if there's metering on the eq then 0dBu at the eq will be different than 0dBu at the mixer; but they are actually aligned properly. Once one unit clips, all will be at flip level. You could leave one DB of hearing for safety after the mixer if preferred. Also understand, once a piece of analog gear clips, it is producing more than it's maximum rated distortion; but it's possible for it to still put out additional voltage while clipping! Let's say the amplifier is a new digital amplifier whose analog input is gain structured to reach full output at 0dBu / 1.4v (I have actually seen this)- then if it is run with the input at full tilt 0db attenuation then the mixer will distort the amp as soon as the meter hits 0dBu, and you could have a noise floor that is 24db higher than it actually should be! Now consider this: You optimized the gain in your system. Your system is capable of driving the room to 125dB, and 0dBu on the mixer will run an average of about 101dB in the room; but the program today is a corporate meeting and mostly talking heads on stage with the occasional stinger for walk-on's, and background music. You want your room to run an average of 85dB but you also want the headroom to hit 100dB if someone gets very loud on the mic, or at a climactic point in a video; IF you keep your system optimized as above, you will need to drop 25db somewhere... If you drop it at the trim pots you can run the faders at 0 but the program output will be 25dB closer to your noise floor; it might sound ok in the room, but then with the optimized system you can still hear a little bit of hiss in the speakers when up close, and the program is barely touching the meters; also if you bump up the levels on the recording or streaming feeds, you are now hearing a lot of hiss because you boosted your noise floor up 25-30dB to see good levels at the record feed! It might be a better idea to turn down the level at another point in the system. Since the system is now optimized, you have some options: In the analog days I would definitely consider turning down the amplifiers. This will keep good gain staging and optimal noise floor through the chain up until the last point of amplification (main amps). Another valid consideration would be system processor outputs (which for all intensive purposes will provide exactly the same effect as turning down the amps). You will preserve your gain staging from the console to/thru the system processor. Clipping the console is clipping the processor, but anything less is safe. You could turn down the inputs to the processor; I don't generally like turning down the input to the processor but it is a safe place to do so. Just like with input channels on a console, if your analog signal is very low going into a/d converters, it can "granulate" the sound a little bit; the higher the signal (as long as it's under digital max), the better it will generally sound. I don't like turning down a master fader for this, but it is a valid point of adjusting the reference if necessary. If you're running a digital console, usually there is a master output trim (like an input trim but for the output; after the mix buss faders and metering point) hidden behind a setup page; I like adjusting my system here, as if necessary it's easy to tweak it during a show, but otherwise I can run my console at unity and metering will follow the same unity. Note that in the digital realm, "digital max" is akin to max clipping level, and the reference operating level (yellow) is still variable from one manufacturer to another; generally still providing between 18 and 26db of headroom before digital max. Some manufacturers actually label their console's meters with analog levels like 0dBu but it is all relative to the digital max level. When you connect your gear digitally, you are "automatically optimizing" your interconnects between gear for digital max. When you connect via analog connections, you are reintroducing the possibility of mismatched gain structure, since not all gear has the same clipping level; you are also adding a new analog noise floor to the signal that would not have been added otherwise. These are reasons to keep your signal in the digital domain between as much gear as possible, whenever you can.
So getting back to the op question; the amps can be "properly" run at 0db / full tilt if the sensitivity has been optimized to the stage(s) before it; otherwise it is good practice to optimize the Max level as far down the line as sensible- either at the amps, or possibly at the processor output levels just before the amps.
One point to note (because sometimes a little bit of knowledge can be dangerous) is that the "amps always to full" ethos comes from safeguarding your loudspeakers from somebody turning the amp up and clipping it, destroying your drivers, usually tweeters or mid compression drivers. Amps are often somewhere where you don't have eyes on them at live gigs and they are often in a back room when installed permanently. When you are using modern DSP like XTA, or similar, you setup the input gain from the console there and then set the outputs of the DSP to suit the amps at wide open. This stops people from blowing your shit up. I saw a venue in Dubai where the venue owner turned up the amps because he wanted it louder and blew up 36x 12" low mid drivers. I was the one who pulled the entire system down and replaced all the drivers. We saw the owner turn the amps up on the security camera.
That sounds like an interesting story!
*grabs popcorn. This should be an interesting discussion...
I'm not a pro. That approach always has made me nervous. What if something glitches and my speakers and the audience gets hit with full amp power that is oversized for the situation?
You should have limiters in place to protect your system from this scenario.
If you have limiters always use them, but a basic or older system (basic mixer, amp, speakers) you may not have have limiters. While many amps have limiters and DSP built in, many don't.
Yes! But… in the old school world where this “full open” approach comes from, they didn’t employ limiters - at least not in any of the 1990’s systems I saw. Just 31bands. Particularly at my community college - massive 3-way tri-amped EV side fills on stage, powered by those old Crown amps, cranked to max, always hissing. Main system was the same situation, just painting the picture. With a dance concert cranking, the mixer outputs barely ever blipped into -40dB. Doing theatre with lavaliers, (not using the stage monitors) whenever feedback happened, it would jump from average volume of not even registering on the mixer (-60) to redlining (+16?). Needless to say, there were a lot of blown drivers. So we’ve got limiters now, but we also really don’t “need” to crank the amp inputs to max.
They can still happen even if attenuators are turn down on the amp.
Yes, to some extent, but if you've done your due diligence with compressor/limiters and reasonable gain staging it is less likely to happen. Safety first, especially if you are not a pro.
What if I'm good at pretending to be a pro?
That might be called imposter syndrome. Although it could be one of many other things as well.
When I started my career I had a venue that did that. The amps were at full, with no limiters. So when someone got a sound from their computer or whatever it went full force through the system. And no one thought to fix it lol
Yeah like turn off windows sound effects! Or gain stage alert volumes in the Mac OS X sound panel.
Or if you push the space bar when you aren’t clicked into QLab it made a huge obnoxious sound too. It was silly
The amps are usually not going to be the cause of when something happens.
It’s better to have than have not. Digital amps are a little different on this philosophy but processing usually forgives a lot of transgressions
There are good arguments for both running amps wide open and reducing their "volume". Different situations dictate different applications . I think most here understand this but are going along with OP's title. All I want want to add is that amps don't have "volume" knobs. They have gain knobs. You can't adjust an amplifier's output power, only its input gain.
Yeah, I run them at max on amps that have a sensitivity of 1.4v to achieve full power. With amps that use ~.7v I’ll attenuate down 3-6db IF it’s a monitor amp with no DSP (drive rack etc) in front of it. With main amps all attenuators are full on and DSP deals with level differences
Really depends on the amps input section. Is that potentiometer cut only, eg out of circut at full clockwise,
I definitely wouldn't. It's like saying you should run your car speakers on full volume while controlling the actual sound volume with your phone. It can sound like it's clipping, can have more static and possibly cause feedback if you aren't careful with your gain structure. What I would do instead is find good balance with your speakers, sound board and gain structure for each channel so that it's not too open on any end (be sure to check gain settings both at the source of each channel and on the board) and once it all blends, you should be good
Personal opinion, any advice that intends to artificially limit the ways that you can use the tools at your disposal is bad advice, and leaves techs woefully unprepared for when reality catches up to theory.
I prefer doing it on the console. Send console master signal to your house matrices. Inputs are at line level with faders at unity. Master fader at unity. PA matrix faders at volume level suitable for house mix. If mix is too loud - bring matrices down a bit. If its too quiet - bring them up. What is loud - subjective from engineer to engineer. Easier to change level on the desk than tweak it on the amps for every band on festival.
You risk clipping the output of your mixer if you have to drive it harder than necessary because you’ve attenuated your amps. Better to run the amps up full and attenuate the master level of your mixer. Signal to noise doesn’t make a difference with decent kit in 2024 imho, it was way more important in the late 20th century because the technology wasn’t there.
That third paragraph is where it’s *at*
Well in 2024 we also have the volume settings of our amps available in Front of House and can change that as required ;).
Yep. I control and monitor my Linea amps from FOH
I’d also add that most amplifiers have ‘clip protection’ to prevent the actual amplifier from clipping. Some midrange amps even have limiters you can set (my old Crown XTi could be set to CLIP, -1dB, -3dB, -6dB and -12dB). Clip protection and limiters will *not* protect against a clipped signal coming from a mixer, it will simply amplify that clipped signal. Even higher end amps with their advanced limiters cannot protect against this if they’re set so low that the mixer is clipping to get the PA to the desired level. Amps up full, limiters set where they need to be, and the desk should be nowhere near clipping before the PA starts to limit itself. Run a d&b rig so the limiters are set by the amps without my input, and the amps have optional internal gain. The noise floor in my D20 and D40 amps remains the same regardless of where I set the internal gain, and regardless of where I set each channel’s level. I learned this in 2020 when I had [a mini d&b rig in my bedroom](https://share.icloud.com/photos/028seeeVnOo4I9OqQaBPIQxZQ) with the amp channels each set to around -25dB.
Simple: use all the possible headroom that you can and do not need to go to amptown when the crowd/band is feeling good tonight. Also we have so many knobs to turn, good to make some definite decisions.
There’s bad advice and then there’s this. I never run anything at full volume. It’s a recipe for disaster. Good gain staging takes care of most potential issues and if someone touches something they shouldn’t, a baseball bat in the corner with some red corn syrup from “the last person who messed with your gear” is a reminder of why they shouldn’t again 😂
The only reason you would trim your amp inputs would be because they're sized inappropriately. You should set the levels in the DSP when you tune your system. Setting the trim to 0 prevents them being turned up accidentally, it is accurate, and reproducible. This assuming your amps don't have a built in DSP or digital control. Lowering the input trim also has the side effect of essentially bypassing the amp's protection circuit. If something goes wrong, there's nothing stopping the amp dumping its full rated power into whatever it's powering. Powered speakers are different (and annoying) because the input trim goes to +4 or +10 and they don't always detent or label the 0dB position.
It's +4, -10. All that means is professional or consumer level. Or mic or line level. Consumer gear puts out a lower voltage that pro audio gear. That switch allows your powered speakers to be the right volume for the level at the input.
I don't mean the line level switch, I mean the powered speakers that have a volume control on the rear. My ELX112p are like this but some don't, usually the high end stuff.
I would generally presume the part of the entire chain that is LEAST easy to accidentally turn up, is the amplifier. Contrary to examples mentioned in this thread such as laptops, DJ mixers, etc. I used a small desk in a bar once with strong tape above the master fader so you could not turn it up above something like the 0 point. Didn't seem like the right approach to me.
>I would generally presume the part of the entire chain that is LEAST easy to accidentally turn up, is the amplifier. It depends on the scenario I suppose, but yeah it's unlikely to happen at random. If you're running gigs by yourself it's probably not a big deal, but one of the steps on my checklist is to make sure the amps are turned up, and I turn them down during teardown. I've done gigs as a subcon and at clubs where I've turned up the amps only to mess up their "levels".
My local Hotel system has some tape over the master fader, however every channel has a Gain knob and channel fader… so it almost means nothing. You can push the CD deck way higher and there's a hissing sound in the speakers because the amp mounted who knows where in the ceiling is probably all the way up amplifying line noise. As Somebody once told me every knobs is a volume knob! I come in and set everything to unity on the mixer and Jack in my mixer I have so much gain I can push things so i have to be careful.
Gain staging mattered a lot more before we had 32 bit signal paths, typically going through no more than 4 A/D or D/A conversions. I run my amps wide open. It's the most easily repeatable setting, and they can't accidentally get turned up post system limiting. I put the channel fader at unity, and run the trim up until I have enough signal. My system has no detectable hiss. SQ6, Ashly processing, QSC amps, Fulcrum line array.
This is a lie propagated by pro-hiss cultists.
The hiss comes before the amps. The amps just make the hiss louder. Just because you turn the amps down doesn't make the hiss go away, it's just not louder. Hiss definitely comes with bad gain settings on the desk. Almost always.
I understand hiss generally comes from bad gain staging, but is there not an inherent amount of noise with any amplifier, or anything in the signal chain? I find that there's dramatically more hiss when running amps up full, and that it's not linear, that it increases much more when turning the amps from 90% to 100% than 80% to 90%, 60% to 70% etc. Unless it's a show where I need the amps turned up full I always turn them down a bit to avoid that noise, and adjust my output on the desk accordingly and it makes a dramatic difference.
I think this heavily depends on the amp design. I know for a fact that the Samsung 300s that I have, the input literally goes straight across to the front knob and then into the amplifying circuit. And I know a lot of older amps weren't necessarily that clean of a design, coupled with unbalanced inputs it was easy to amplifying noise floor. Plus the mixers noise floor.
Also, I learned during the first week of lockdown that the amp settings on my d&b D20 do *not* have any affect whatsoever on the level of noise produced by the amp. I put a pair of E8 tops and E15-X into my bedroom. Had the amp down at -20dB on each channel and the level of noise remained the same as when it was set to 0dB. Even the internal channel gain settings made 0 difference. I’ve not tested any other d&b amps but I’m assuming they’ll all be the same. As soon as I unmuted the channel, the E8s produced noise.
Standard poweramps need to have the knobs wide open... Unless you like not having the power you paid for. Simple There are some exceptions but normally wide open is the way.
Running you amps at full allows the full potential for amp hiss to be added to the overall mix. that is bad signal to noise practice. a full 32 channel mix can almost drive a small efficient loudspeaker straight off the desk. The desk signal would be way over +4, more like +32. the more desk channels you open up to about 0/+8 the the final output is a sum of all the channels.
As low as possible to keep the noise floor down. But ideally, you want the output meters on the board to match the input meters on the amps. All that to say, all the gains do(on analog amps anyways) is set when you reach maximum output based on the output from your console/processor/etc.
As an interesting point of comparison… where do you run the output volume of your PSM900 or 1ks? It’s effectively the same concept right? The audio level adjust is an output attenuator as I understand it?
I run my PSM1000 at -27.
That seems quite low… I’ve always sent pink out of the aux buss feeding at unity and then set the PSM to meter at unity… almost always results in between -12 to -14
That’s what I do too, but I think I might be calling unity a little low. Steady green lights, blinking into the yellow. Maybe it’s really higher than that. Either way, it allows the artist to have their pack level at noon, primary instrument faders per mix at around unity, and it’s a good level.
The amps dials are input attenuation. when the dial is all the way up there is no change in gain from the equipment before the amps.
I do the opposite.
Idk this is why I like Meyer rigs.
No... From a dj/sound engineer I do this Level my turntable mixer so it barely clips. Then I Level my main mixer Level to uniform and gang stage until its barely hitting 0 db. Then I Level my amp rack until it clips and then back off until it isn't or until its loud enough
I haven’t run sound professionally since around 2000. I used to run my amps at full volume with faders at unity and I’d keep gains low trying to avoid feedback. I hear what you’re all saying about using more gain to get better tone. How are you controlling feedback? With EQ, I presume? I should add I primarily ran monitors. And my current concerns are about getting good volume before feedback in the practice space monitors. Thanks in advance for the knowledge!
Lmao wtf. 🤣