What are your "secrets" tricks/tips for live sound.

[Darren_Darren]

This one might need as STICKY.

 

I was perusing the Recording forum and saw a sticky at the top that offered recording "secrets" for those of us novices.

 

I think that might be a good one here. Although we have a sticky for definitions...we don't have one for information applying to applications. FOR EXAMPLE: you can define what a Hi-Pass filter does but what are some cool secrets or tips to make you a better sound dude using it?

 

What are your "Secret Tips" for LIVE SOUND & PRODUCTION?

[Craig Vecchione]

 

Originally posted by Darren_Darren FOR EXAMPLE: you can define what a Hi-Pass filter does but what are some cool secrets or tips to make you a better sound dude using it?

 

 

I don't know if it's cool, and it certainly isn't secret, but regarding the high-pass on a mixer channel, you engage it when that channel is used for vocals, to keep un-needed low frequency caused by handling and nearby sources out of that channel.

[RickJ]

 

Originally posted by Craigv regarding the high-pass on a mixer channel, you engage it when that channel is used for vocals, to keep un-needed low frequency caused by handling and nearby sources out of that channel.

 

 

I engage them on everything except the kick drum and bass. Sometimes, even the bass strip gets its button pushed. (Keyboards are iffy, and I ask the muso beforehand if (s)he will be looking for that Genesis-style low drone.)

 

This works for me whether the subs are part of the mains or run off an aux.

 

To the original question: There are no secrets. There are only opinions. Ask specific Q's and you'll receive any number of replies as to how things are done.

 

What you'll find out is, there are many techniques out there, and you'll be able to experiment to find out what works best for you.

[groove.77]

Well, I dunno if it's a trick, but I often find that running the mids knob on the kick drum channel around 3:00 gives it more thump without having to crank the channel.

[Scodiddly]

Many skills, but few tricks I can think of.

 

Here's one: Plagued by low-frequency feedback? Try switching the polarity of the mic(s). Doesn't always work, but enough times it'll make a big diffence.

 

When tuning monitors, get them as loud as possible, then turn them down before the band shows up. You'll have a lot more headroom that way.

[rangerkarlos]

Keep stage volumes low. Point amps at the players head. Tell the crowd tip their waitress/bartender.

[groove.77]

Smiley face EQs are bad.

[OneEng]

Wow. I don't know if anything I know is a secret, but I do know a thing or two that I have found is little known information:

 

 

I am certian there are more, but these are some of the most obveous.

[MrKnobs]

The stuff OneEng posted is great!

 

I like to run 15 band (2/3 octave) graphics on each sub buss, so I can reach over and change the EQ of the vocals, guitars, drums, or keys without touching the channel strips or the main EQ. For example, if the vocals are muddy, I can cut a little around 200 Hz and boost a little presence around 4k. If they're harsh, I can pull down the sliders around 2k. Anything that gets done to one vocal gets done to them all, and individual tweaks are done on the channel strips. This is a big part of the sound I get.

 

If the mixer I'm using doesn't have fully parametric EQ on the channel strips, I will insert my outboard parametrics on select channels, especially on the drums. The drums are perfect for parametric EQ, because they make essentially the same tone every time you hit them, not a moving target. You can focus tightly in on whatever frequency (example snare crack) you like or don't like and boost it or lower it. Essential to a great drum sound.

 

If I'm mixing a low budget band that I know will have a not so good drum set, I'll often stick a trigger to the kick and maybe the toms. I'll quickly select and tune a sample to the drum being struck in sound check, using one that sounds a lot like the real drum except better, of course. After tuning the sample to the same pitch as the real drum, I'll blend the two sounds together for a sound that's better than either by itself.

 

Often, in a very reverberant room, there's no point in adding MORE reverb to the mess. So, for the lead vocal I'll likely use a short delay instead of verb, to make it sound fuller but still clear.

 

In a REALLY reverberant room, if time allows, I'll set up a pair of full range speakers maybe 20 ft forward from the stage and time align them to the mains. This pushes the direct field much farther out into the audience and the reverberant field buch farther back, partially negating the effects of the bad sounding room.

 

Live drums can really be tightened up with gates. Having a whole bunch of mikes open at once but not in use (like when you mike every tom) muddies up the drum sound horribly. So I'll gate the toms, kick, but usually not the snare. The better the set of gates you have the better this will work. Frequency keyed gates can be set very precisely, since each drum has a characteristic repeating tone. If your gates allow you to set a gentle gain reduction (instead of off/on) any missed "grace" notes will be much less noticeable. Blending the gated close mikes with the two (or one) overheads gives the most natural sound.

 

A crazy guy who used to do sound for Joe King Carasco showed me some unusual (to me) uses for a harmonizer. His rider required four! I told him dude, nobody needs four harmonizers! He used them to deepen drums in the FOH or just add a subharmonic, and also to feed the monitors with just a one cent detune which can greatly reduce feedback since each subsequent loop back no longer converges to the same pitch. This is not something I'd do everyday, but he proved to me it works.

 

Sound check these days has unfortunately become a luxury at some venues. That sucks, but you have to deal with it. It's extra essential then, that you get a good LINE check before the show. Nothing worse than the recording artist stepping up to his mike and singing in silence because the cable or snake channel is dead or disconnected! Check it once, then check it again.

 

In a situation like that, with no sound check, keep everything moderate on the board faders except the vocals. Then quickly add everything smoothly starting with drums. The audience will forgive a lot if the vocals and drums are right and the guitars and keys are a little low at first. You can usually count on the bass player to be heard even with his slider down.

 

Another trick to dealing with a reverberant room is to customize the group EQs by putting a dip in them right where the room is most reverberant. Hey, if the room is gonna carry 300Hz, let it. You can't fix a time domain problem in the frequency domain, but sometimes you have to try.

 

Board tapes: Lots of bands want a board tape, because, like Ms. Flier, they're very nervous about what they sound like out front. But, if you give them a board tape made in a small club, you'll never convince them that the vocal heavy, guitar and bass starved mix they hear is not what the audience heard. Instead, tell them to be patient and then do a board tape at the next large outdoor gig. If you're really lucky, and your band closes the night, you can play the board tape right back through the mains and let the band stand out front and hear EXACTLY what they sounded like.

 

I've actually had grown men kiss me when I've done this!

 

And they never again worried about what it sounded like out front.

 

Terry D.

[Bro' FF]

Wow, GREAT info you guys! I have cut and saved the above and will store this for future use!

[ezstep]

The only secret (and it's no secret) that I have learned is not to try to bluff my way through something that I am not familiar with. I will be the first to ask questions to learn more. I have received much good advice on this forum, and I have tried to leave a little advice of my own. If I don't understand a particuar post, I often PM someone to further explain it to an old dog (me). Thanks, guys.

[RickJ]

I've gotten away from blindly patching gates into the drum kit mics. I like to treat the drum kit as one instrument rather than seven or eight individual sources.

 

Try placing the mic as close to the head as possible on a 45-degree angle. The drum head becomes a planar surface to make use of the mic's boundary effect. Place the rest of the mics similarly on each drum, and also place them at 45-degree angles to each other. This pattern all points to a "sweet spot" where the drummer sits -- allowing the audience to hear what the drummer is hearing. Each mic will naturally cancel its closest rivals, and the mics that are farther away will enhance the "kit" sound.

 

(This technique is used on Dave Davies' latest solo album, and the drum sounds are phenomenal.)

 

Two exceptions: (1) If an inept drummer can't tune a drum well enough to get rid of an obnoxious ring I patch in a gate, and (2) the kick drum. Most times, a gate with 10 to 20 or so dB of attenuation is necessary to prevent the kick from taking off on a wild feedback ride through Sub City.

 

OneEng, I have to disagree about not using HPF's on the vox strips. (If it's a sweepable HPF and it's set to 400Hz, then I agree.) But the standard fixed 75Hz or 80Hz filter is well below the range of virtually all singers. It's been my experience that warmth in the voice happens at 160 or so (males) and 320 or so (females). IMO, any advantage that leaving the HPF off on vox strips is far outweighed by the stage rumble that enters the system through the mic stands.

 

PS: Mr.K, I believe Lee is a "Mr." not a "Ms."

[OneEng]

RickJ,

 

IMO, any advantage that leaving the HPF off on vox strips is far outweighed by the stage rumble that enters the system through the mic stands.

 

I have a Allen&Heath MixWiz which has 100Hz HPF's on each channel. Just using my ears and A/B testing it, there was a signifigantly more pleasing response without the HPF being on the vocal channels. When everything in a song is at high SPL, I doubt that this is noticable; however, when the vocals are out front in a quiet part of the song, it is quite noticable.

 

Your point is quite valid though. We get away with this only because we keep the stage volume down. A loud backline is a problem no matter what you do. In a case where the band backline (especially the bass) is loud, I am sure that your point is very valid indeed.

 

If you don't have high bass stage volume, then you don't have to worry about PA rumble caused by the hot vocal mics picking up the stage backline.

 

There is also the individual system aspects to be mindfull of. My system gets autoequalized in each venue ensuring that I don't have any inhearant rumble problems from the room and speakers. Keeping a well room eq'd system from rumbling is much easier than one that has a resonance mode or two in the low frequency range.

 

To conclude, if you are experiencing low rumble problems, I would highly suggest trying to lower the stage volume on the bass, guitar and keyboard first. Second, I would adjust the main equalizer to eliminate the resonance mode. If none of that worked, I would then start moving to the channel strips and clicking on the HPF's as you suggest.

 

I can not stress enough how important it is to have low stage volume. High stage volume brings in a pleathra of problems. The solutions to all of the various problems it brings all result in a loss of overall tone. The best cure for high stage volume is in-ear monitors. The worst problems with in-ear monitors are that they are not as cost effective as conventional monitors (someone is going to argue here for sure), and that many old-school musicians won't use them.

 

In my experience, the basic theories that get great PA tone are:

 

[abzurd]

On the mixwiz I like the 100HZ high pass. It's right about where you'd cross over the subs. Using it keeps the selected voices or instruments out of the subs and reduces the number of point sources for those instruments. I think it cleans up the mix considerably, especially in your standard bar, a structure seldom engineered to sound good.

 

Also makes it far easier to balance the subs with the mains because you're only really worrying about the kick, bass guitar, low part of the keys, and possibly the floor tom. Vocals, guitars, other drums aren't affected if you have the subs a little too hot.

[agedhorse]

 

Originally posted by MrKnobs Another trick to dealing with a reverberant room is to customize the group EQs by putting a dip in them right where the room is most reverberant. Hey, if the room is gonna carry 300Hz, let it. You can't fix a time domain problem in the frequency domain, but sometimes you have to try. Terry D.

 

The single best piece of advice presented here ever.

 

Read this again and again, write it on the chalk board 100 times whatever you must do

 

YOU CAN NOT FIX A TIME DOMAIN PROBLEM IN THE FREQUENCY DOMAIN.

 

You can make an attempt at masking it but you can not solve it, and can do a great deal of damage to what's left by unceremoniously hacking at it with an eq!

[Dookietwo]

 

Time delay your mains from the backline sound. 1.75ms per foot is a common figure to use. This makes the sound more focused.

 

Hi

About .89ms per foot is closer to the normal for delaying .

The 1.75 ms would be good for 2 feet .

Dookietwo

[agedhorse]

 

Originally posted by Dookietwo Time delay your mains from the backline sound. 1.75ms per foot is a common figure to use. This makes the sound more focused. Hi About .89ms per foot is closer to the normal for delaying . The 1.75 ms would be good for 2 feet . Dookietwo

 

No, this is fixed time offset, which has a different and dubious solution of it's own.

 

The problem that Terry and I are talking about is where sound waves colide in time and create a diffraction pattern, a repeatable series of dips and peaks.

 

Now add all the sources of time offset into the equation, including the band, the subs, the left mains, the right mains and the reflections from the building itself, and you alter the energy in that pattern, and the pattern itself.

 

So what you are measuring is not necessicarily what your ear is hearing. The ear and brain do a lot of processing to get to where it wants to be, and corrects for a lot of problems. The electronics do not see it the same way, and even though you may see a big hole at 1kHz on the RTA, it may not be there, it's just that the measureable in phase energy is not there.

 

SPL is the vector sum of all the incremental bits of energy, and when the vector directioon (phase) is not pointed at your reference zero degrees, the numbers are no longer real, but then contain an imaginary part that conveys the phase information.

 

It's like you have a buchet 1/2 full of cheep beer, and a bucket 1/2 full of expensive beer. When you add them together, you have a full bucket of beer, but what kind of beer is it... could be cheap, could be expensive, could be in between OR it could be undrinkable. But, it's doubtful that either brewer would claim it as theirs.

[Scodiddly]

 

Originally posted by MrKnobs Yep. Standing waves, modes.

 

 

One of those situations where inverting the mic polarity can occasionally save you ass, incidentally.

[OneEng]

On the mixwiz I like the 100HZ high pass. It's right about where you'd cross over the subs. Using it keeps the selected voices or instruments out of the subs and reduces the number of point sources for those instruments. I think it cleans up the mix considerably, especially in your standard bar, a structure seldom engineered to sound good.

In fact, many bars appear to be structured to sound bad on purpose! Actually, I think they just give little or no reguard to live sound. For example: Mirrors make a place look bigger. They do nothing good for live sound though.

 

I appear to be outnumbered on the HPF I still like the warmth I get with the bottom enabled on the lead vocals. I will look into it again to be sure, but the last time I checked, the vocals definately sounded better with it OFF.

[RickJ]

OneEng,

Absolutely! Mics always sound better when they are allowed to pick up the full range of their sources. Problem is, they are mixed with other sources, and that's where the challenge lies. If there's an 8x10 SVT cab behind the lead vox mic, you had better HPF the vox, or you'll hear more bleed from the bass into the lead vox than you want. It makes for a sloppy mix.

 

Re: Standing waves: I had a kid come up to the sound booth in the club tonight and tell me that the bass (kick & bass guitar) was killing him -- he couldn't enjoy the show. I left the booth and asked him to show me where he was standing. He showed me. I moved two or so feet to the left and said, "Stand here." He bought me a beer.

 

My new motto: If you don't like the way it sounds, move two feet in any direction. It seems to be true whether it's a cover band in a dance club or a major at an arena.

 

Pretty sad if you think about it, but you can't fight room acoustics without a million dollars worth of gear and acoustic treatments.

[Dookietwo]

 

Originally posted by agedhorse No, this is fixed time offset, which has a different and dubious solution of it's own. The problem that Terry and I are talking about is where sound waves colide in time and create a diffraction pattern, a repeatable series of dips and peaks. Now add all the sources of time offset into the equation, including the band, the subs, the left mains, the right mains and the reflections from the building itself, and you alter the energy in that pattern, and the pattern itself. So what you are measuring is not necessicarily what your ear is hearing. The ear and brain do a lot of processing to get to where it wants to be, and corrects for a lot of problems. The electronics do not see it the same way, and even though you may see a big hole at 1kHz on the RTA, it may not be there, it's just that the measureable in phase energy is not there. SPL is the vector sum of all the incremental bits of energy, and when the vector directioon (phase) is not pointed at your reference zero degrees, the numbers are no longer real, but then contain an imaginary part that conveys the phase information. It's like you have a buchet 1/2 full of cheep beer, and a bucket 1/2 full of expensive beer. When you add them together, you have a full bucket of beer, but what kind of beer is it... could be cheap, could be expensive, could be in between OR it could be undrinkable. But, it's doubtful that either brewer would claim it as theirs.

 

 

Hi

I run Smaart live to time align my foh to the loudest source on stage . Sometimes drum monitor or git . Depends on the band .

 

Smaart is not time blind .

 

Now add all the sources of time offset into the equation, including the band, the subs, the left mains, the right mains and the reflections from the building itself, and you alter the energy in that pattern, and the pattern itself.

 

 

The above is normal in all multi source live sound app .

What you were talking about is time aligning mains to the stage .

To use the loudest source is normal .

It takes sound roughly .89 ms to travel one foot .

If you use 1.75 ms and your foh speakers are 25 feet away then you would be adding 48ft aprox . to the foh delay time using your calculations . 25 times 1.75 = 43.75 ms . or 48 feet .

 

Because sound travels at .89ms per foot ( aprox ) a setting of 22.25 ms would be closer . 25 times .89 = 22.25 ms .

 

If the loudest source on stage is the snare in this example and it is 25 feet back from the mains and if I used 1.75 ms as a starting point my mains would be -23 feet back in time compared to the arrival time of the accoustic snare at the mains . Again 1.75 ms times 25 feet equals 43.75 ms or 48 feet back in time . Subtract 25 from 48

( the distance the snare is back from the mains ) and the mains are still in time 23 feet back in time from the snare .

 

Even if you included the Hase effect ( spelling ? ) of roughly 10 ms . You would hear the the natural sound of the snare before the snare sound comes out of the pa . Roughly 12 feet in time later or 10.68 ms . This is enough to be heard .

 

Mulit sound sources on stage are normal .

When you time align the pa to the mains you have to make a choise .

What do you want to time align the mains to ?

Where do you want the system to be aligned at ?

Impluse alignment or phase alignment ?

Do you have the proper test equipment to carry this out ?

 

Again with no test equipment it is better to use .89ms as a basic starting point . Sound travels roughly one foot in .89 ms .

 

Some people add 7 to 10 ms for hase effect . 25 times .89 plus 8ms . If the mains are back in time a few ms than you would think the main sound is coming from stage . If you are off more than that most people will hear the different arrival times and locate them .

 

I have had my mains 37 feet in front of a band before . The snare being 44 feet back . Using 1.75ms per foot I would have to add 77ms or 86 feet to my mains ? Way to much .

Dookietwo

 

I wish this group had spell check as my french to english could be better..........

[agedhorse]

 

Originally posted by Dookietwo Hi I run Smaart live to time align my foh to the loudest source on stage . Sometimes drum monitor or git . Depends on the band . Smaart is not time blind . Now add all the sources of time offset into the equation, including the band, the subs, the left mains, the right mains and the reflections from the building itself, and you alter the energy in that pattern, and the pattern itself. The above is normal in all multi source live sound app . What you were talking about is time aligning mains to the stage . To use the loudest source is normal . It takes sound roughly .89 ms to travel one foot . If you use 1.75 ms and your foh speakers are 25 feet away then you would be adding 48ft aprox . to the foh delay time using your calculations . 25 times 1.75 = 43.75 ms . or 48 feet . Because sound travels at .89ms per foot ( aprox ) a setting of 22.25 ms would be closer . 25 times .89 = 22.25 ms . If the loudest source on stage is the snare in this example and it is 25 feet back from the mains and if I used 1.75 ms as a starting point my mains would be -23 feet back in time compared to the arrival time of the accoustic snare at the mains . Again 1.75 ms times 25 feet equals 43.75 ms or 48 feet back in time . Subtract 25 from 48 ( the distance the snare is back from the mains ) and the mains are still in time 23 feet back in time from the snare . Even if you included the Hase effect ( spelling ? ) of roughly 10 ms . You would hear the the natural sound of the snare before the snare sound comes out of the pa . Roughly 12 feet in time later or 10.68 ms . This is enough to be heard . Mulit sound sources on stage are normal . When you time align the pa to the mains you have to make a choise . What do you want to time align the mains to ? Where do you want the system to be aligned at ? Impluse alignment or phase alignment ? Do you have the proper test equipment to carry this out ? Again with no test equipment it is better to use .89ms as a basic starting point . Sound travels roughly one foot in .89 ms . Some people add 7 to 10 ms for hase effect . 25 times .89 plus 8ms . If the mains are back in time a few ms than you would think the main sound is coming from stage . If you are off more than that most people will hear the different arrival times and locate them . I have had my mains 37 feet in front of a band before . The snare being 44 feet back . Using 1.75ms per foot I would have to add 77ms or 86 feet to my mains ? Way to much . Dookietwo I wish this group had spell check as my french to english could be better..........

 

 

To what point in the house do you time align? If you move that point, ALL of the equations change. What happens in reality, is that it is IMPOSSIBLE to time align to more than 1 point in the room at any instance. If you work out the math, then move the point sideways, the delta between the speakers and the center of the stage along will change enough to allow you to drop all the decimal points.

 

What you will find is that modern test gear will allow you to think you are getting more precise than is physically possible, since the equation to be solved is changing with measurement point. If I remember correctly (Terry, help me out here... it's been a few too many years) the solution is staticly indeterminate because the number of equations required to solve the problems increases by at least N+1 times the amount of data present to solve the equations with.

 

I have to deal with this all the time, and the only way to get a workable solution is to ASSUME something to eliminate one of the variables and make A solution possible. If you make a good assumption, the solution will be a good one.

[RickJ]

That's easy, Agedhorse: Tune the system so the the perfect listening position is two or three feet above where your hands rest when they're not moving faders.

[Dookietwo]

 

Time delay your mains from the backline sound. 1.75ms per foot is a common figure to use. This makes the sound more focused.

 

AggedHorse

 

I was replying to you tip . ( Above )

 

I'm saying 1.75 ms is the wrong amount . That .89 ms is closer to an average time to start at . When I enter 100 feet in my dsp for delay lines it shows 89 ms . 10 feet is 8.9 ms and 1 foot is .89 ms .

 

When you setup and your speakers are 18 feet from the backline the time alignment is way off to begin with . Time skew is not correct no matter where you are .

If you notice in my last post I said you have to determine where you want your alignment to be .

With a standard setup . Two midhighs and subs . You could pick a spot on the dance floor around 1/3 of the way out and 1/3 of the way between the stacks . This would give you two spots that would be roughly aligned . Much better than NO alignment at all .

With different sub placement sometimes you have to make a choise as to what bandpass you want to align to . If your dealing with a loud bass player and your subs are center stage you may want to time align the subs to him then phase align the mids and highs to that . Or if the loud source on stage is the snare you would go with the mid highs and align the subs to them . Again using smaart and a chosen sourch on stage you can get better alignment than none at all .

 

http://support.siasoft.com/Downloads/binary/TechNotes/CaseStudy6/case6.pdf

 

Above is a link to give you a better understanding of what I do .

Again delaying mains to backline is a normal thing to do and is a good tip . I just know that 1.75 ms is to much to add per foot .

Reread your first post . Is there any chance the 1 in front of the .75 is a typo ? .75 ms per foot would be closer .

 

As a side note . My digital mixer has delays on each channel .

I can if I want delay every channel seperate and knowing the distance from the mains to the source align them to a chosen spot in the room . Even in a stereo setup . I am doing this New Years eve .

 

Dookietwo

[agedhorse]

I have found in practice that much of this delay and time align thing affects the sound too far to the right ofthe decimal point to be of much use, other than some basic issues relating to speaker placement.

 

Since I do mostly larger halls, as you gat farther out from the stage, the delay issue rapidly disappears into the insignificant digits.