AFS problem & general setup.
What are we doing wrong? Originally had a problem with the AFS on the DRPA as detailed in this post ….
http://www.driverack.com/drug/view_thre ... forumid=16
We eventually sent the unit off for repair to Cimple Solutions, who ‘resoldered dry joints on main board’. On return the AFS worked ok for about 4 months, although still not as quickly and efficiently as when it was new. It has now stopped working again. Having begun to rely on the Driverack I didn’t want to run our system without one, so I’ve just bought a DBX260 so I can use this while we send the DRPA away for repair again.
BUT we can’t get the AFS to work on the 260 either. So it’s got to be us that’s doing something wrong! I’ll try to detail how we’ve set it up in the hope that someone can help us (Gadget?)
Unfortunately we don’t yet have a laptop that we can use for the GUI. Our system is A&H Mixwizard desk, 2 Mackie SRM450 tops, 1 Mackie 1501 sub. 2 further SRM450’s + other powered speakers (DB technolgy) for monitors. We’re trying to get the system running quickly, so have opted for similar setup to previous one using DRPA – ie stereo linked.. May change this later to also use it on monitors if we ever get good enough!
1. Connection
a. Left out from mixer to ch 1 input on DBX
b. Right out to ch 2 input
c. DBX out 1 to SRM450 left
d. DBX out 2 to SRM450 right
e. DBX out 3 to sub
2. DBX system setup
a. GEQ
b. insert 1 AFS
c. insert 2 Compressor (although don’t plan to use this)
d. main spkr Mackie C300
e. sub spkr Custom
f. Amp high custom
g. Amp low custom
3. Configure (section 3.5 in manual)
a. GEQ linked
b. AFS linkd (I think)
c. crossover 2x3
4. Cossover button
a. sub 45 – 118
b. tops 118 –
5. limiter, compression delay all OFF
As we’re using powered speakers it doesn’t seem that the gain structure setting advice in appendix A is applicable. I also read this on a post in the Mackie forum. So I presume I’m setting gain structure by simply ensuring each channel on the mixing desk stays within the green LED with the PFL in. Then individual channel faders on the desk are at or below 0db
So now to the AFS..
1. All mics on, faders at 0db. All speakers on (no monitors yet)
2. Set 12 filters, 4 fixed
3. Raise master faders and feedback occurs as the masters get towards 0db. The feedback carries on getting worse until it’s unbearable (even wih earplugs in). Filters do not catch it.
The first time I tried this the filters did actually catch a few of the feedback frequencies. We ran a gig last night but it certainly wasn’t working correctly. I couldn’t get enough volume out of the vocals without feedback – I had to manually ride the masters to control it. We ran through the whole setup again this morning, also trying the RTA process, but AFS won’t catch any now. AFS is switched on and lift after 5 sec.
Very upset – please help!
Wendy
http://www.driverack.com/drug/view_thre ... forumid=16
We eventually sent the unit off for repair to Cimple Solutions, who ‘resoldered dry joints on main board’. On return the AFS worked ok for about 4 months, although still not as quickly and efficiently as when it was new. It has now stopped working again. Having begun to rely on the Driverack I didn’t want to run our system without one, so I’ve just bought a DBX260 so I can use this while we send the DRPA away for repair again.
BUT we can’t get the AFS to work on the 260 either. So it’s got to be us that’s doing something wrong! I’ll try to detail how we’ve set it up in the hope that someone can help us (Gadget?)
Unfortunately we don’t yet have a laptop that we can use for the GUI. Our system is A&H Mixwizard desk, 2 Mackie SRM450 tops, 1 Mackie 1501 sub. 2 further SRM450’s + other powered speakers (DB technolgy) for monitors. We’re trying to get the system running quickly, so have opted for similar setup to previous one using DRPA – ie stereo linked.. May change this later to also use it on monitors if we ever get good enough!
1. Connection
a. Left out from mixer to ch 1 input on DBX
b. Right out to ch 2 input
c. DBX out 1 to SRM450 left
d. DBX out 2 to SRM450 right
e. DBX out 3 to sub
2. DBX system setup
a. GEQ
b. insert 1 AFS
c. insert 2 Compressor (although don’t plan to use this)
d. main spkr Mackie C300
e. sub spkr Custom
f. Amp high custom
g. Amp low custom
3. Configure (section 3.5 in manual)
a. GEQ linked
b. AFS linkd (I think)
c. crossover 2x3
4. Cossover button
a. sub 45 – 118
b. tops 118 –
5. limiter, compression delay all OFF
As we’re using powered speakers it doesn’t seem that the gain structure setting advice in appendix A is applicable. I also read this on a post in the Mackie forum. So I presume I’m setting gain structure by simply ensuring each channel on the mixing desk stays within the green LED with the PFL in. Then individual channel faders on the desk are at or below 0db
So now to the AFS..
1. All mics on, faders at 0db. All speakers on (no monitors yet)
2. Set 12 filters, 4 fixed
3. Raise master faders and feedback occurs as the masters get towards 0db. The feedback carries on getting worse until it’s unbearable (even wih earplugs in). Filters do not catch it.
The first time I tried this the filters did actually catch a few of the feedback frequencies. We ran a gig last night but it certainly wasn’t working correctly. I couldn’t get enough volume out of the vocals without feedback – I had to manually ride the masters to control it. We ran through the whole setup again this morning, also trying the RTA process, but AFS won’t catch any now. AFS is switched on and lift after 5 sec.
Very upset – please help!
Wendy
Comments
Actually, would you call? there is just too much going on here for me to sort this out without talking to you... it could take a week one post at a time. I can assure you something is very wrong here... the 450's are very flat to begin with... I can have someone running around RIGHT in front of the speakers with a wireless SCREAMING (and often do) and NEVER have feedback... I have a 240 on the monitors... and the speakers will blow your hair back... and you can point the mic right @ the horns, 2 feet away, and have NO feedback...
I live in the centeral standard zone... calls before 10PM please, and after 7:30 am, 1-218-999-7100. I run my businesses out of my house so I'm here most of the time (except for weekend nights of course...
I think if you' do that we can get to the bottom of the problem...if not let me know and we can start the long and arduous back and forth process of question and answer...
Gadget
I tried to reply to your email directly but it was returned.
Wendy
Good to talk to you today, My email address is tzone@paulbunyan.net. Good luck with the new configs... enjoy the GUI its the bomb!
Gadget
I don't have a DR yet, but I'm putting together a new system for my church, and I'm making the DR260 a central part of it. In the meantime, I've been reading through all the archives and playing around with the GUI trying to learn all that I can. I'm fascinated by all this and just can't seem to get enough of it!
I just wanted to say that I appreciate all your posts. I've learned so much from you and others already. I can't wait to get the DR so I can put all of it to use.
Anyway, I saw here that you mentioned something about a feedback trainer. I've searched the archives and found mention of it in earlier posts, but I can't find what it is. Is this some sort of tutorial or white paper? Whatever it is, I'd be very curious to find out more.
Thanks,
Chris
Give me your email address and I'll send it to you..
G
Partial success tonight.
1. Gui Interface
Conneted this up at home and found it much easier to follow. Then had the problem of laptop with no serial port. Followed earlier posts about this and as suggested on wombatsound I bought an Aten USB to serial converter. The make is Roline not Iogear, but it does use the Aten driver. Sadly not got this to work yet so I had to set up at the venue again wihout the gui.
2. Setup
At the venue we set up a new user program so we were starting with a new slate. We seleced the same options as before, I previously forgot to mention the Output route setup for ch 1,2 & 3 at the end of the configure section. So 1 to High Left, 2 to High right and 3 to sub.
3. Gain structure
I kept the SRM450’s & sub off while I set up the gain structure. Unfortunately couldn’t use the test CD cos it had only recorded 51 tracks, so track 57 for setting gain wasn’t available. Instead I used a Steely Dan CD.
· Set input gain on CD channels on mixer to just below clipping. (EQ flat)
· Channel faders to 0. Lift master faders until start to clip (this got to +5dB
· On the DBX260 set the input mixer for ChA & ChB so the input level meters just below clipping. This ended up with Input 1 Level for ChA being set to +6dB and likewise for input 2 level for ChB.
· Outputs -Xover button. No need to change gain on ChA & B because level alread at top of green. Increased gain on Ch 3 by +10dB.
All this done with all Mackie speakers turned off As these are powered ther’s no need to set the gain for these – we keep the input attenuator at 12oclock. Read somewhere thatthis is corect.
So turned the system on and wow! It sounded so much better than ever before. Gain structure is obviously so important! We didn’t do any EQ or RTA wizard setup cos not enough time, also would rather do this properly once it’s warm enough to run through it outside. So on to the AFS …
4. AFS
Oh dear!Made sue AFS off before we started the AFS wizard. There was no one else in room so very quiet. Set up just 3 mics for test purposes. Set the gain for each mic to just below clipping and channel faders to 0.
· Enter wizard and toggle to AFS wizard.
· Followed instructions very carefully AFS is linked so both Ch A& B done together.
· Master faders down
· Select 12 filters, 4 Fixed.
· Musich High
· Slowly increased master fades, as they got towards 0dB the first feedback occurred The system was very loud but it took quite about 10 secs of increasingly loud feedback before it was caught. ( At east it did catch it!)
· Faders down and tried again, same thing happened - .2nd filter took even longer to catch. Ears hurting too much to try for a third.
· Page back to change the remaining 2 fixed filters to Live. Forward again to finish the process.
· Made sure AFS on and set to Live mode, Lift after 5sec.
· Got partner to try out mics – all very loud and clear, but as soon as mic brought towards front of stage and near speakers it fed back. The live filter did catch it, but again only once the feedback was really loud. I wouldn’t want it to get this loud during a real performance!
I’m so baffled, but stil hope I’m missing something somehere. I’ve read so much about this equipment now but can’t understand why noone else seems to have had this problem. As I said before I can’t believe there’s a fault with this brand new BX260 as well as with the older DRPA. So user error – but what??
The system is, however, already sounding better and I’m fairly sure we’ll get plenty of volume out of it without any feedback, but I just want this particular module to work as promised. We’ll certainly need it for the outdoor summer jobs when performers wonder around in front of speakers with radio mics!
Thank you so much for all your time Gadget and sorry to come back with more problems.
Email address is cdholi(at)yahoo.com.
Thanks,
Chris
well, you still need to get a PINK noise signal for the GAIN STRUCTURE...music is not the best way to do it...
One question I guess I should have asked is what Mic's you are using. Hopefully NOT condensor type...And DEFINATLY not omni directional. Also its important not to hold the mic right behind the capsule, blocking the rear openings of the mic...this turns the mic into a near omni directional mic.
Have you figured out what the primary feedback frequency's are?
My system, the feedback eliminator, you just barley hear the start of the feedback and its gone...
You said you did an \"auto EQ\" please describe what process you used for this...in great detail including what you did with the EQ curve after the process.
we'll get to the bottom of this yet!
Gadget
Thanks for having patience with me Gadget! Just got back from great sounding gig, we had no feedback, although I had to keep my hand on the master faders when it threatened a few times. I did manage to get the gui to work after much trial & error, but also got problems with the laptop so it was a bit inconsistent. But so much easier to set up and understand with the gui!
I realise I still need to use the pimk noise to get the gain structure done properly - I haven't done an auto EQ/RTA yet - want to wait till I've more time or can run it outside. Snowing here at mo so that's a no go-er yet. I did try the AFS again before the gig, after the sound check. We had 2 Sennheiser E865's on vocals, 2 SM57's on guitar amps, AKG c1000 for O/H. We had plenty of volume and great sound without any feedback during soundcheck. I simply want to use the AFS 'just in case'. So once band had gone and room empty (the gig was in an upstairs room of pub so this was quite easy) I set up AFS Wizard again. Raised masters to past performance level before it started to feedback, but the feedback was not caught. I didn't let it carry on for too long or too loud cos worried about damaging speakers. I also tried this again going straight to the live mode with floating filters - still no joy.
I then decided to notch out the frequency manually using the - thanks again for the feedback trainer, found 400 Hz first time. So took this out in the post PEQ. I didn't take any more out at that stage cos we had loads of volume anyway. But on previous occassions when we've tried this I think the frequencies were quite a bi higher (2-3KHz?)
I've stopped panicikng because using the PEQ is probably a better way to fix feedback than relying on the AFS, although I'd still find it tricky to find the feedback during the gig. We're having another session at the pub Sunday afternoon, so we'll try and get the gain structure sorted correctly, may also try the auto EQ.
Any more thoughts before we go?
BTW most condensors are too sensitive fo live use... one exception is the Shure 87/beta 87... but the system needs to be extrremely well tuned to use condensors... Thats why I asked what mics you are using...
Let me know... BTW, also, when you go to do the Auto EQ , use the close proximity on axis (6-8 feet max aimed @ the spot between the horn and woofer one side only, Return everything below 150 hz and above 700 hz to zero and set those frequencies by ear.(see the auto EQ posts/tutorials)
Gadget
I'm still a bit confused, because surely if a certain mic is prone to feedback that's the idea of the AFS - to catch it and notch it out? Perhaps I'm expecting too much of the system? Thanks to you I certainly understand better how to use the system now and I can easily get more than enough volume without feedback, but it's frustrating that I can't get a feature which I know has worked in the past to operate properly now. The speed at which the AFS kicked in at outdoor gigs last summer was phenomenal. I can't think what I'm doing differently to stop it working now.
Regarding the mics, I don't always use the C1000, because I quite often work with acoustic bands with no drummer. The E865's are 'stage condensors' but I think these are great mics - really clear and have good pick up range. They are super-cardiod, but they are meant for stage use - the spec states \"ecellent feedback response\"
We'll be off again this afternoon for another go, so I'll report back later. Thanks for the tip about Auto EQ. I'll do some morr eading about that now.
Wendy.
Let me ask you this, Did you set the laptop up, and were you monitoring the functions with it? I'm going to send you a screen shot of the way I set up my screen, perhaps you ave done something simillar. Note that the EQ is directly over the RTA (I have the RTA mic hooked up and running so I can see the systems response at all times
Check your Email
Gadget :P
Thanks for all the info - you've certainly given me some homework!
I did manage to keep the laptop connected today when we had another test session (during gig last night the laptop died, probably dodgy battery).
So this afternoon's progress was as follows
1. Played pink noise from test CD, only strange thing was that I couldn't get the level of output 3 to mono sub to rise above -12dB, this was with xover gain at +20. Thought this was too drastic so put it back to +6. The odd thing was that when we pushed the low pass filter frequency cut off point on the sub up into the range covered by the tops, ie big overlap, the output level on the sub did rise to match the tops - suspect lack of low frequency in the pink noise?
2. Ran through the Auto EQ fairly successfully - I'll try and send you some screenshots later. That was fun and again brilliant to see it work on the gui!
3. The dreaded AFS. Same as usual happened. AFS on, ran AFS wizard and kept the windows open on the laptop - I chose just 2 fixed. Made sure fixed mode selected on laptop. I just had 4 mics on - 2 E865's and 2 SM57's. Raised master faders and did manage to catch one feedback frequency - great to see that notched out on the laptop. I tried to catch a 2nd but the feedback got too loud and was still not caught. I set to just one fixed and then went to live mode. My partner went on stage and sang while we brought a 2nd mic out in front of the speakers. The level was way loud - probably beyond normal performance level. So when the 2nd mic started to feedback nothing happened - ie the feedback just got louder without any filters catching it. The AFS was definitely on and in Live mode. So my problem is that the filters are not setting at all in live mode and they take too long to set in fixed mode. I supose if I let the feedback continue long enough in live mode it would eventually set, but this is no good for live performane - it might damage the speakers as well as make the audience deaf! I don't want to bore you to death with this - the system does work fine; I hope to never have to use the AFS, but on the odd occasion I may need it.
You say that you left the RTA mic hooked up and running - I didn't realise you could do this. Where do you keep the mic, by the mixing desk at the back of the room?
Got to go out for a curry now but I'll connect the 260 to the PC tomorrow and get some screenshots. Also off to do some reading ....
Wendy.
My emails to you keep getting returned (spam checker?) so can't send you the screen shot.
Wendy[/img]
I do have a firewall, perhaps if you zip it and send it in a compressed format it will go through. Again My email address is:
tzone@paulbunyan.net
Please try setting the # of fixed filters to 12, make sure all the other switches in the FBX are properly set and try that, with a single mic. No monitors..lets try and see if we can set a baseline...also, lets try different settings in the FBX and see if perhaps one of the settings isn't functioning properly...If that were the case you should try a \"hard reset\" see the manual...for the proceedure.
Yes, you can monitor the RTA in realtime by selecting 'MIC' in the input select button in the RTA module, you can also monitor the inputs in the same fashion, and compare what is coming into the DR and what is coming out of the speakers. This can yeild some interesting information. and you can get a feel for what the speakers are doing to the sound...Anyther bonus is, with the GEQ over the RTA, if you set the displays up right you can see a peaking frequency/feedback, and grab the fader right above it in the GEQ and pull that fader down. Or use it to see that perhaps a gentle falloff of high frequencies is occuring, and help deal with it using the GEQ ior PEQ's.
I'd also be interested to know if the feedback shows up on the RTA display. There are a couple of different thoughts about mic placement, for this purpose I would like to have you place the mike on a stand about speaker center high and about 6 feet away, when doing the show, however if you can't get it on the ceiling in the nearfield put it back near you @ FOH. There are some things to know about this but lets talk about that once we have the FBX working...
Keep me informed, Oh, and send me a simple \"testing\" email, and lets see if it goes through...OK?
Gadget
<\"mailto:tzone\"@paulbunyan.net>: host ......... said:
Mail from ...............
rejected; Anti-spam/mail-abuse (in reply to RCPT TO command)
Got to dash to work now, but here's sth I've just remembered, when I was setting the gain structure up yesterday I opened the input mixer for ChA, but by mistake I started lifting the fader for input 2 (this should stay on -INF). I already had the AFS window open and I suddenly noticed that a whole load of filters were being taken out. Not sure which mode the AFS was in. This was being done when the FOH speakers were off. I don't kow if this is signifcant or not - probably of no relevance whatsoever!
Info about monitoing in your latest post is really useful - can't try this out again yet cos taken rig down. I'll try single mic going in to DBX to a single speaker and see if the FBX works.
COntact your internet provider and see if you can get some help, I had a heck of a time with Anhony, a fellow I help in Ruissia!
Is the place you bought the Driverack unable to help you?
What eacctly do you mean by this? More detail please
\" already had the AFS window open and I suddenly noticed that a whole load of filters were being taken out. Not sure which mode the AFS was in. This was being done when the FOH speakers were off. \"
Gadget
The shop just said AFS's are never very reliable. I'll try the UK distributers once I've followed your latest suggestions.
Forget the last bit in my previous post - I think it may have been the RTA window I was looking at.
I'd like to really thank you again for all the effort you dedicate to this forum Gary, I don't know how you find the time. I've learnt loads this past week, but know I've a long way to go yet! It's a relief to have someone knowledgeable you can ask for help.
Wendy
Got to go out now, but will try to send the config file and results from last night later.
Wendy
Phew!!
WG
Gary
I didn't think the Auto EQ Or AFS Wizards were available thru the GUI.
Steve
DRA
I have read here on the forum that a \"feedback trainer\" exists. Is it possible to get a copy?
I have just purchased a Driverack 260 for our church. Haven't had a chance to set it up yet. I am reading everything on this forum to get a many pointers as possible before doing the setup and installation.
The sound in the main auditorium space is just terrible. The space was not tuned or EQ'ed so after doing a lot of research I settled on the Driverack 260.
Our Mixer setup: We have an Mackie Onyx 1620 Mixer with firewire option which is our Main Mixer (small church 150-200 people). We also have a Phonic 1860II mixer/poweramp combo (looking at purchasing a new power amp but pretty well blew the budget for this year). Currently the Main L/R output of the Mackie is inserted into the phonic 1860II power amp section just after the main output faders and EQ.
Speaker Setup: Driving a cluster of three Wharfedale Pro LIX-12 Speakers (3-horizontal-tightly spaced, one against the the other with the two outboard speakers angled about 25-30 degrees off the center axis) and suspended from the ceiling. Now here is where things gets interesting, the system was installed by a Music Store and they hooked the right channel of the 1860II Right Output Channel to Two of the WPro LIX-12's in parallel and the Left channel to the other remaining LIX-12. To say the least the system is unbalanced both electrically and acoustically. I believe it would be best to disconnect the center speaker in the cluster, and then at least we would have a more balanced system.
What do you think about the speaker re-config before installing the 260? I could possibly drive the center speaker off a separate amp from the 260 as another possibility.
RTA & Space EQ: This was never done with the old setup. Awaiting delivery of the dbx RTA-M mic and then will eventually EQ the space. We do have some room modes which will eventually require acoustic treatment and some base trapping but will leave that for a project next year.
Advice on the speaker hookup would be helpful and is it possible to get a copy of the feedback trainer?
I don't think you need to disconnect the ctr channel why not balance it by mixer volume. You are feeding one side of the amp with one channel (say left main out)and the other with other main right?
Give me your email address and I'll see if I can send it to you .. the trainer that is..
Remember you cannot equalize a room only hope to equalize the system with respect to the room...
Be advised that the majority of the info you need is on the \"former forum\" under \"user submitted white papers\", and \"tutorials\"
Gadget
Your amp (head) is probably not bridgable, but hear it goes.
3 cabs fanned from center.
Center cab (#2) used as primary room source.
Left (#1) and right (#3) cabs used as near field fills left and right.
Amp is bridged and receives a mono signal to both inputs.
#1 & #3 are connected to the left and right outputs of amp.
#2 is connected accross the +'s of the 2 outputs.
#2 will have a +3db (I think, maybe its 6db, I can't remember which) output bump for the main area and +0db for the front fills.
What you have is 3 speakers, 8 ohms each. The amp (I'll make up numbers here) is 200w @ 8 ohms and 400w @ 4 ohms. 1 speaker by itself pulls 200w with it's 8 ohms and uses it all itself. The 2 speakers pull 400w with their combined 4 ohms and split the power up and use 200w each. All three cabs are using the same power. No problem with what they've done.
DRA
Gadget, I sent you my private e-mail for the feedback trainer. Much thx in advance.
I am reading the Former Forums now and hopefully will avoid repeating questions that have already been answered.
Dra: Thx for the advice on the speaker hookups. I looked at the paper you mention at the Peavey site and it looks to be very close if not identical to what I want.
BTW my Phonic 1860 II Powerpod Mixer has the following spec's:
2X300W/4 ohms
Bridge 600W/8 Ohms
Here are the details they mention on the outputs:
\"These are 4-pin speakon connectors that produce
speaker-level signal from the built-in power amplifier
output.
The speakon marked “L/R/BRIDGE“ can be used
for left channel output, right channel output as well
as bridge mono output with the 4-pin structure. Pin
1+ and pin 1- are for left channel, as pin 2+ and pin
2- for the right channel. Using pin 1+ and pin 2+,
gives you a bridge mono output.
The speakon marked “R� is only for right channel,
using pin 1+ and pin 1-.
The minimum required load for left and right channel
is 4 Ohms, and 8 ohms is the required load for bridge
mono. This ensures the proper function of the powered
mixer.\"
Info: The Powerpod has a switch on the back that they describe as follows:
\"POWER AMP
There are two channel input level controls and an
operation mode switch in this section. These two
VRs help set the input sensitivity of the power amplifier.
The range is from +18dBu to +4dBu.
The operation mode switch allows user to select
between stereo and bridge mono. When switched to
bridge mono, only the left VR and the L/R/BRIDGE
speaker output connectors are functional.\"
At this time the switch is in STEREO position.
Here is how the music store wired the system:
RIGHT Speakon Connector: 1+, 1- to Right Speaker
L/R/BRIDGE Speakon Connector: 1+ to Left and Center Speaker (+),
1- to Left and Center Speaker (-), 2+ and 2- NOT Connected
Amp Output in Stereo Mode.
Based upon what you have indicated the AMP Output should be in L/R/Bridge Mode and wired as you suggest.
Gadget, do you know?
DRA
Wayne
I don''t think so or you would have seen that I provided a very detailed \"getting started\" set up set of FAQ's.
There is a feedback trainer here...
http://sft.sourceforge.net/
Gadget