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Question re AFS setup

Ian AnthonyIan Anthony Posts: 14
edited August 2013 in PA Configuration Wizard

I'm a solo vocalist and fairly long term user ( three years I think ? ) of the Driverack PX unit, which I use with the following basic system for live events.

Laptop using 'Showmagic' software as backing track sound source and linked DMX lighting controller.

Mackie ProFX mixer
Lexicon vocal FX unit
dbx 231 EQ twin channel 31 band eq

Mackie SM450 speakers ( up to 6 of these available )
Mackie 1501 subs ( two available )

Shure SM58 beta wireless UHF radio microphone.

Shure SM58, old style VHF radio microphone ( as backup only ).

Gain setup throughout the system is set at 'unity' and signal gain strength through the various parts of the system is set at optimum levels for best performance without clipping or distortion.

I'm getting really good results from using the auto setup system, particularly when running the eq section for different venues. ( now using the recommended indoor system of microphone on floor at approx 25 feet, one speaker at a time to get a flat speaker response, followed by careful 'tweaking to taste' on the separate twin channel dbx 231 eq unit ).

My problem at this time concerns the setup of the AFS section.

I normally set a 'medium' response and set 6 filters 'fixed' and 6 filters 'live' with a five second release on the live filters.

If I use my old SM58 microphone as my live microphone when setting up the AFS, I can slowly bring up the mains until feedback just starts to creep in and then slightly increase it until a frequency is grabbed by the AFS and reduced.

I can continue as suggested to carry on increasing the mains until AFS is complete. No problems here and very little stress on my speakers when done slowly and carefully.

However, if I use my newer SM58 'beta' with the super-cardioid cartridge as my live microphone and begin to slowly increase the main volume control on the mixer, the system does not feed back in a steady and controllable way. Even with painfully slow and careful increase of the mains, with no build up or warning the system will suddenly shriek with feedback which is pretty much uncontrollable and I have to react like lightning to turn down the mains and prevent speaker stress/damage.

The other thing I noticed when this happens is that all six of the 'fixed' feedback filters are assimilated in the split second of howling feedback?

I have considered using the older VHF microphone for the AFS setup and using my newer UHF microphone during the gig, but this doesn't seem to make sense as their reaction to out of balance frequencies appears to be quite different?

Any ideas as to how I can deal with this please?



  • GadgetGadget Posts: 4,915
    Your NOT going top like this but the Beta 58 is just about my LEAST favorite microphone on the planet! :twisted: It has a very hyped upper mid/low HF particularly in the 2K area.. and to me sounds terrible, even with way too much equalization...

    The fact is, this mic needs a different monitor position than a cardioid model like the SM58... the 58 wants the monitor basically at the base of the mic stand, the mic cable pointing at the monitor.

    Also, the hyper-cardioid/super-cardioid mics have maximum rejection at the 10 and 2 positions..
  • Hi Gadget, thanks for getting back to me on this, even though you have rubbished my new microphone :roll:

    I don't think I've explained the problem properly that I'm getting during setting up the AFS?

    I routinely set up any stage monitors in the positions as you suggest in your reply, so I'm definitely getting the best feedback resistance out of the super-cardioid feature on the SM58 beta. I don't think speaker/monitor positioning is the issue here?

    The problem is, during the AFS setup, when I get the mains way up towards the point where feedback begins to occur, it's impossible to get the feedback to come in slowly and under control, as I can do if I use my older SM58 cardioid microphone during the AFS setup.

    When I use the SM58 beta, the feedback just runs away with an almighty squeal and fills up all six of my fixed frequencies in a split second!

    I've tried the most careful and slow increase of the main faders to try and isolate the feedback frequencies one by one, but I can't get this to happen when I use the SM58 beta as my live microphone, no matter where I position it in relation to speakers/monitors etc.

    I'm wondering whether to set the system up with a lead microphone ( say a traditional SM58 ), directly after the auto EQ section. This would at least identify the feedback prone frequencies for the particular room I'm setting up in?

    I know this isn't ideal, as I will be using a different microphone during the performance, but it seems to be the best I can get at the moment?

    I'd appreciate any advice you can offer and also, what wireless microphone would you personally recommend as an alternative to the Shure SM58 beta?


  • DraDra Posts: 3,777
    I had a similar experience with an "E6 countryman" type ear worn mic. At the end of the day, I believe that it is simply the mic / room is either pretty stable or is high unstable in broad bands. Make sure that you don't have the channel looping back into itself in some crazy way.
    How loud is the mic before the feed back goes all crazy?
    Also try reducing the input gain on the channel strip by about 1/3rd and try again.

  • GadgetGadget Posts: 4,915
    even though you have rubbished my new microphone :roll:

    Hi Ian... Sorry, but one thing you can expect from me is I tell it, like it is... and I don't pull any punches...I HAVE an SM58beta and I have had many many bands come in with those and I can't tell you how I HATE to see them on stage! They create problems in the mains, and even more so on the monitors and I tune my stuff to the NUTS...I have an article in the FAQ section where I get 121.7dB from a monitor Using an SM58 or Audio Technica Pro10 HE or an Audix Om5...and I could aim the mics right down the center of the horn less than a foot away (just like a singer that grabs the mic, and swings it right down to the monitor WILL do! :roll: )

    Now, if I take the 58 beta and do that all HELL will break loose! Look, every mic/monitor combination has a point of no return..(same with the mains, but here the ROOM usually becomes the victim...)no matter HOW well you tune the system there will be a point where, if you turn it up enough it WILL feed back, usually when you get to that point there is NO bringing it back even if you butcher the EQ to the point where the sound is clearly HORRIBLE!!! :shock: My guess is that you have reached that point with that mic...BTW I'm not the only one that isn't in love with that mic...and I really AM sorry.. but you didn't ASK before you got it :mrgreen:

    It's always a good idea to do your equalization, then defeat it, and see how badly you butchered the sound...I have found that if I use the Auto EQ, then take the MAJOR trends and do those with the PEQ's.. the MINOR GEQ changes are fine, anything over 3dB is done with the PEQ's...but then I use either the 260, or 4800 for monitors...With the 260 you have 9 PEQ's per input and 4 PEQ's per output and the 4800-... well UFDA!

    If you want to test this for yourself, go to the FAQ section, find the 121.7 monitor thread, set it up like the post, then simply change the mic from the SM58 to the 58 Beta... I bet the gain drops by (FAR) more than 6dB (3dB is a relative doubling of sound pressure...)

    Look, the Beta works for some singers because of their voice...but if the singer is wrong for the mic.. well.. that doubles the damage! I'm not saying that those that the mic works for couldn't have a BETTER experience with something else, just that it works for them, and in my book, ANYTHING that takes a lot of EQ to make it work is WRONG for my system...I have the mic simply to assuage my clients that demand it... otherwise it's a paper weight!

    So, to summarize, Yes, you explained it fine, that mic is just... I don't know.. possessed?
  • Hi guys, thanks for your excellent and detailed replies to this post, much appreciated!

    Dealing with the replies in chronological order, my first thoughts are on DRA's reply...

    I think both yourself and Gadget have hit the nail on the head when you say that the basic design of the microphone ( Shure SM58 beta ) is probably responsible for the problem?

    Maybe Shure have gone a bit too far in trying to 'bring out the vocal in the mix' and the artificially enhanced frequencies on the cartridge are just a bit too strong for their own good?

    The cabling and routing setup on my system is definitely all ok and I don't think there is any problem here as other microphones tested on the exact same system and room don't behave in this way.

    To be fair, the feedback doesn't crash in until I am way up on the mains and microphone fader, with very little headroom left in my PA, so I can push volume levels very high in the live performance without too much risk of feedback. I'd just like to finish the AFS setup on the driverack without blowing my speakers due to massive and very sudden feedback !!!!

    Re your suggestion to reduce the microphone channel gain during the AFS setup, I'll give that a go this weekend at my next gig and see how the system reacts. I think it's a very good suggestion from you and hope it may assist :wink:

    Reply to Gadget's thoughts.......

    Hi Gadget, thanks for coming back to me again on this problem.

    I'm still giggling to myself at your intro line about telling it like it is and not pulling punches...Hey you're a sound engineer, what do I expect, Christmas spirit! LOL :wink:

    Seriously though, the more I hear about the challenges of using the SM58 beta, as opposed to the basic SM58 and other similar and less radical microphones, the more I'm convinced about your opinion that it is the beta microphone itself that is the issue?

    To be honest the main reason I purchased the SM58 beta was on the recommend of a number of experienced artists and that my old SM58 runs in the VHF band, rather than UHF on the beta. As you'll know, the VHF system was/is prone to interference from other radio systems, unlike the UHF setup.

    Ah well, looks like I'll be shopping for a more suitable microphone for christmas! If you can recommend any suitable microphone systems for a male vocalist - baritone-tenor, mainly singing middle of the road 'rock' and some ballads, I'd be very grateful.

    Once again, thanks for the detail in your replies to my post and for the benefit of your experience in live sound production, it's invaluable.

    Anybody want to buy a Shure SM58 wireless microphone and base unit ?????? :?

  • GadgetGadget Posts: 4,915
    You know, you can get different capsules and put them in the handset, although the problem is only (in my opinion) exacerbated by the wireless part of the system... most of my comments were directed to the wired model, the wireless only add instability..

    If you have read much of my stuff you'll know that speakers respond non linearly to the volume of the system, that is to say that as the volume goes up, what was stable and flat @ 105dB won't be @ 115dB, and the louder you go the more non linear it gets...you really need to tune the system for the volume you intent to playat.

    Also, room dynamics will at some point go strait into the lue. After that point (where the room starts to break up) the sound will be generally more and more UN-intelligible, and generally blows up into these fits of feedback and resonances. Perhaps you are pushing the system too far when your doing the FBX?? After all, any more than double what you want to hit volume wise is just driving the speakers into their non-linear mode anyway :wink:
  • Hi Gadget,

    After some thoughts I'm going for the following action........

    I'm hiring a Shure PGX2 SM58 handheld radio microphone to use directly alongside my SM58 BETA.

    I'll use both microphones to run through a test of the auto EQ and AFS systems on the driverack, just to compare typical results and then make a decision whether to purchase a replacement SM58 cartridge to swop for the BETA version.

    I think I probably will end up buying the SM58 cartridge as it seems to be less radical than the BETA and more balanced/easier to use when going through an EQ and AFS for each new venue/room I play at.

    I take your point on the differing response from loudspeakers at various volumes, so I'm going to set up a test day when I can run an 'outdoor' test and get to save a flat profile for my speaker(s) and save it for use as a reference point when I set up in a new venue. I can use the flat EQ profile to start off with, then tweak the EQ to get the best profile for the room?

    Can I ask your opinion on this following setup routine for EQ and AFS at a new venue, bearing in mind that I am a solo vocalist working to backing tracks, often with limited setup time prior to live performance?

    1. Set up all PA ready for use, i.e. mixer, active speakers, microphones 'on', onboard and individual channel EQ set 'flat', correct gain structure and mikes/speakers in correct positions for live show.

    2. Run Setup Wizard, programming in speaker types etc.

    3. Set main speaker levels for balance/pan with dbx m2 RTA microphone in central position, in general area of audience position.

    4. Move RTA mike to a position on floor ( on a towel ) about 25 feet away from and directly in line with one main speaker. Turn down opposite channel main speaker ( after noting previously set balance position ).

    5. Run Auto-EQ Wizard, ( In Response A ( flat ) / Precision Medium setting ), with pink noise volume at similar level to live performance volume setting.

    6. Turn up the speaker that had been turned down prior to the Auto-EQ so that we now have both channels producing a balanced volume.

    7. Place the 'live' vocal microphone that I will be singing with on a stand in the approximate position that it will be used during the performance. Run the AFS filter setup routine, using six fixed and six live filters ( set at 5 second release ) Music medium - very narrow notch.

    8. Save and name preset for this room.

    9. Do a final check of the sound mix for the room by playing a good quality recorded track at a good volume and using my 'ear' to assess quality and balance. If necessary adjust 'flat' EQ to reduce or boost frequencies to achieve a good pleasant sound for the room.

    10. Play backing track and sing live through system to set vocal to backing track volume levels and vocal FX levels. Make any final small tweaks to EQ if necessary.


  • GadgetGadget Posts: 4,915
    I think you make this more difficult than you have to...Once you set the HPF and crossover, set the L/R balance and sub tops balance you STORE that, this becomes you basic setup for EVERY room... then if you have the above setup AND a flat outdoor EQ this becomes the default setup and you recall it for each new venue...

    The above are not not usually room dependent, other than perhaps bass performance in the room. Standing waves and Modes will set up in the rooms ... different rooms have different dimensions and different waves will set up.

    When you get to the new room, you simply recall the default setup, run your auto EQ pass, compare that to the default setup and see what can be done WITHOUT the EQ changes...look at the EQ and determine if there is a cause for the changes and if there are any causes (like a window, or concrete block wall) and if you can use aiming to get the sound on the people.. and off the hard surfaces. Unfortunately the SRM 450's spray sound all over the place...making it VERY difficult to keep the sound on the people, and off the reflective surfaces...

    After you get the system the way you want it you store THAT setup to a different location, with a corresponding name.
  • DraDra Posts: 3,777
    Since you are looking catastrophic feedback protection, don't worry about the fixed filters, set them all to live and speech. The speech setting grabs faster.
    Here is something else you might try. The AFS grabs feedback based on gain increase INSIDE the DRPA. Increase the gains in the x-over +10db and decrease the amps' gains -10db, then try again. When done, set back to normal gains. This is also assuming that you have the systems gain structure set up already.

  • Hi guys, my thanks to Gadget and Dra for your excellent comments and suggestions. I'm learning much more about live sound-management than I thought I would when I first started on this post!

    Also, apologies for the delay in getting back to you on these comments. Usual Christmas madness with gigs, shopping, eating and drinking etc. all stopping me from getting on my laptop! I also managed to pick up a full dose of the 'flu' just after Christmas Day, so the last three days have been spent semi-comatose in bed. Got a gig on New Years Eve, so hope I'm recovered for that one :?

    OK, I've booked my local community hall for a day next week when I can run a series of different setups on my rig, such as different microphones and powered speaker combinations. My aim is to create a series of saved DRPX profiles for my different powered speaker systems ( these will be full outdoor tests on the playing fields immediately outside the club ) looking to create 'flat' EQ's which I can use as the basis for setting up at different new venues as they occur. Thanks to Gadget for his coments on how to shorten the process down effectively but still get the same quality results.

    Also want to try out the effects of using 'speech' settings on the AFS? ( thanks to Dra for this ). I like the sound of using only a few or no fixed filters when running the AFS ( this will take less frequencies out of my mix ) and leave the rest as 'live' with a five second release setting?

    My concern with using the 'speech' setting is that it may be too wide a 'slice' taken out of my mix and therefore have too severe effect on the overall sound??? I suppose the only thing to do is try it and see what I get!

    Also made a nice discovery re the microphone challenge.... I can unscrew the top section of my old SM58 radio microphone ( it's about seven years old but working perfectly ) and screw it onto the base of my new SM58 beta radio microphone! This allows me to sing with the SM58 cartridge, which I've worked out that I prefer, but I can also take advantage of the UHF radio system provided by the SM58 beta system :D

    I never would have believed that these two transmitters would be interchangeable, because of the age difference, but they interface perfectly. Ah well, nice when something works out this way. Thought I was going to have to buy an updated SM58 cartridge at about £175!

    Re the Mackie SRM450 speakers, specifically Gadgets comments on their wide sound dispersal and potential room reflections and clashes, I'm looking at various speakers with a more 'directable' sound output, so any suggestions on the way forward here would be appreciated.

    I'll let you know how I get on with my day out at the community hall? Hopefully I won't shatter too much glass :wink:

  • GadgetGadget Posts: 4,915
    Well, I think that the wider filters will definitely have more of an affect on the overall sound,,, however I never really had much use for the feedback filters if the system is basically flat, and not mad loud and obnoxious...I must admit I am not of the "It must be loud ot be good" department, but I don't think 90dB average 'C' weighted is reasonable for live rock music either...

    I'm providing for a lovely little gal (88 lbs soaking wet) with a HUGE voice, TINY stage, reverberant space (tongue and groove pine all the way around the 8X12 stage and a 4 piece rock band...3 powered Anchor 8" Extreme powered monitors, 2 washing the stage nestled in to the FOH speakers, and one for the drummer... Melody has my wireless, and 6 nights a week walks down the stairs singing right past the monitor and stands in front of the FOH speakers and belts out journey like only current Journey front man does.. and never a whisper of feedback...

    These are no lightweight tops by any means, they are OAP S122 with a 3" voice coil 2" exit B&C D750 horn driver and a 400 watt Neo woofer that I covered 2000+ people with using only 4, flown and some LABsubs..it was insane...

    THIS is what it should be like, because to have to depend on the feedback suppressor is to fix issues is a train wreck waiting to happen...



    As for recommendations, Danley sound makes probably the best sounding and speakers with the best pattern control... it comes at a price however...I'd say if not Danley, look for something in the 40X60 coverage in a 12" .. I like the D.A.S. RF1264's here and are probably the most cost effective speakers in their class...
  • DraDra Posts: 3,777
    Even using the Speech setting, the filter is much less intrusive than "Q" of a 31 band EQ. Plus,the benefit that it centers on the feedback, unlike the EQ ISO center that either requires a much deeper cut to remove or more than one center pulled.

  • It's been a long time since I posted re problems with the AFS section of my DRPX but I've finally managed to get enough time to have a serious try at sorting this out so here's my update :)

    After swopping my own DRPX for one borrowed from another rig, it seems that there is a fault on my own units AFS section, so bought a new unit last week and fitted it into my rack.

    Took the new DRPX et al down to my local rugby club and put it through a full outdoor eq session, saving six different settings of volume ( medium or loud ) with eq set at 0 ( flat ) or preset 'C' ( mid bass boost and high end tail off to reduce listening fatigue ).

    I also tried several different 'notch' settings on the AFS auto setup, to compare its effect on overall sound. Glad to report that using six fixed filter/six live on medium sized notch had no noticeable effect on the mix ( not to my ears anyway! ) so I'm confident to use this in my sound-check, after starting from my previously saved flat eq pre set.

    Also tried AFS set up with my Lexicon vocal effects unit activated for hall reverb and studio delay. This has previously made my rig more prone to feedback at high volumes, but after setting flat eq and running AFS setup, the sound system is much more stable and no longer prone to feedback caused by the addition of vocal FX.

    One slight puzzler was that during the process of setting overall system gain, I got my mixer set just below clipping, DRPX inputs/outputs nicely set just below clipping, but I couldn't get the clipping indicator led to activate on my Mackie speakers, even with my main mixer volume set to full ( + 10 db I think ? ). The Mackie speaker itself was howling loud and I was reluctant to keep trying too many variations as I understand that pink noise can damage the high range speaker at high volume?

    All limiters/compression etc. were turned completely off on the DRPX so not sure what to try now, if anything? I would have liked to have found the speaker clipping point so I could finish off by setting the right level of soft knee limiting for any 'oops' moments, but I ran out of time and brain energy at this point. More fun experimenting next week on my day off! :wink:

    EQ preset 'c' gives the most pleasing mix and seems to get the best out of my Mackie SRM 450 v2's. Not the best speakers in the world, but this eq setting improves them very noticeably and transforms them from being slightly harsh in the top end lacking in mid bass to providing a much warmer, present sound which is really good to listen to and a huge improvement on the basic response from the speaker.

    I now take the modified sound signal from the DRPX straight into a dbx 31 band twin eq for any slight tweaks I might want to make before/during the show.

    All in all a great result for my system sound and I'm very happy and confident to take the improved mix out on the road for my live show.

    Thanks again to Gadget, Dra and all other contributors to this post, and special thanksfor the time and effort that has gone into the forum briefing posts on auto set up, auto eq etc. The info is a huge help in getting to grips with the DRPX.
  • DraDra Posts: 3,777
    Where do you have the Mackie's volume knob set? If it is already crazy loud, no compression would be a good thing. Set the compression like you want it (as far as attack, release, etc) at the volume you are at , then raised the threashhold 2 or 3db.

  • GadgetGadget Posts: 4,915
    Hi again Ian!
    Best regards
  • Hi Dra,

    I'm trying to set a limiter rather than compression, to protect my speakers?

    I normally run the Mackie 450s with the gain knob set at 12 o'clock.

    Hoping to set the limiter so that it prevents clipping, now that the system gain structure is properly set up as per the forum advice post.

  • DraDra Posts: 3,777
    The Mackie's would probably have to be really stressed to go into actual clipping. Their internal limiters will prevent that. My suggestion stands for setting the limiters as well. Powered speakers are extremely (if even possible) difficult to do a true and accurate gain structure. 1- Because the limiter in the speaker cannot be turned off, and 2 - You must do bad things to the speaker since it has to be on during the torture.

  • Makes sense to me Dra.

    I think I've got the gain system set to its best levels via the mixer and DRPX settings so I'm happy that I've done all I can to protect the speakers from being over driven and to get the best out of my sound rig.

    I'm going to do some experimenting with compression on my live vocal next ( via my Lexicon 200 effects unit ).

    If its ok I'll do the basic set up of the compressor and come back on a new thread if I get any interesting results.

    Thanks again for your help and advice.

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