NEWBIE ALERT: Hooking up a Signal Processor

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This topic contains 15 replies, has 3 voices, and was last updated by Profile photo of quebecguitare quebecguitare 5 years, 6 months ago.

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  • #50474
    Profile photo of quebecguitare
    quebecguitare
    Participant

    Hello, thank you for reading this post, I am new at mixing.

    I have done some reading and still trying to wrap my head around the basic concepts of AUX, FX and INSERTS. Still confused with the Send/Return thing…

    I am looking to purchase a A&H ZED-10 unit (I notice there is no section for the ZED line on this forum), but I am not sure if it will fulfill my LIVE scenario accordingly.

    Main Output:
    2 QSC K10 on for the room and 1 QCS K10 as a stage monitor

    Input:
    I have an Avid Eleven Rack with Stereo Effects and Signal Processing for Guitar as Input.
    -The unit has a Main Ouput with Stereo Left/Right XLR.

    Easy Peasy Question 1
    -I plan to use it directly in the mixer, my understanding is that I will use 2 mono tracks on the ZED-10 (XLR from the Eleven Rack above), which means that my EQ will have to be set at the same value, right?

    Question 2
    The ZED-10 has only TWO main output (Left/Right), but I will use 3 speaker.
    -I was thinking of using 2 XLR output to 2 Speaker and 1 TRS to XLR cable to the Stage monitor Speaker. My understanding is that these cable are not balanced, is that a problem (please add details if possible)?

    Question 3
    -I plan to use a signal processor such as a Lexicon MX400 Rackmount for Compression and Effect for a Classical Guitar using a mic as input for a live setup.
    Ref.: http://www.sweetwater.com/store/detail/MX400TRS

    -The unit has Analog left/right Output TRS and left/right Input TRS, I HAVE NO IDEA WHERE TO HOOK THAT ON THE ZED-10 Mixer!
    -The mixer has Aux out, FX out and Main Inserts.
    -Aux out and FX out have single jacks so I do not understand the relationship with my Lexicon input and output, knowing that I will want my effects to go out STEREO (is that important for Reverb)?
    -Could you PLEASE detail your explanation on how to hook up the Lexicon to the ZED-10?

    Question 4:
    Still confused with the Send/Return thing…
    -How does the Send/Return concept comes to play with the ZED-10 layout since there isnt any indication on the Lexicon or the ZED for these input/output?

    Thank you for your time.
    Best Regards!

    #50481
    Profile photo of cornelius78
    cornelius78
    Participant

    1:
    You could use 2x mono channels and pan them appropriately, or you could use one of the stereo channels (TRS connectors) if you have XLR-TRS adapters. The mono channels have a hpf + 3-band eq with a sweepable mid, whereas the stereo channels only have HF and LF shelves, so you can shape the sound better with the mono channels. You can eq them with different settings if you want, do whatever sounds best.

    2:
    The main + aux output sockets on the Zed10 (like most mixers) are balanced. The mains LR uses XLR sockets, the aux out is a TRS socket. The non-RCA inputs on the k10s are balanced too. XLR-XLR (standard mic) cables for the mains, either TRS-TRS or TRS-XLR for the aux. These cables should be balanced, allowing for long runs.

    3:
    You have two options in connecting the MX400 to the Zed10, it depends on whether you want the FX to apply to a single channel\multiple channels or the whole mix.

    One is to use the insert sockets on the LR bus. This would build the FX into the sound of the main mix (but you wouldn’t hear it in your monitor.) To do this you’ll need 2x specially wired insert cables, which are TRS on one end which break out to 2x TS on the other end. This picture explains it.

    The TRS end of one cable goes into the “Main Mix Insert L” socket on the Zed10. The two TS ends go into the “Input A Left” (tip: send) and “Output A Left” (ring: return) of the MX400. Sound goes from the L side of the main mix of the Zed10 up the cable to the MX400, gets processed by the MX400, then goes back to the Zed10, before hitting the XLR L output and your L FOH speaker. Same sort of process for the 2nd cable and the FOH R channel: TRS end into Zed10’s “Main Mix Insert R” socket, other ends into MX400’s ‘Input A Right” and “Output A Right” sockets.

    The second option is to use it in a send-return configuration, which would allow you to apply the FX to individual channel(s) which I’ve detailed in my answer to part 4.

    Either way, as long as you’re using both L&R outputs from the MX400 and you have them panned properly you’ll get your stereo return.

    A third option would be to go Avid Eleven – MX400 then MX400 to Zed10, you’d just need the appropriate adapters. Of course this would mean you’d only get the FX on the guitar sound, not any other mics you plugged in to the mixer.

    4:
    As opposed to inserting a FX processor on a channel/bus and building the FX into the channel/bus’s sound by returning the wet signal to the same channel/bus, a send-return configuration takes the output of a channel, sends it to an FX processor (the “send” part of “send-return,”) then the output of the FX unit is returned to the mixer on a different channel (the “return” part of “send-return.”) You then mix the original (dry) channel with the returned (wet) channel to achieve the sound you want. You can also mix different proportions of each dry and wet channel in different mixes, eg a lot of reverb in the FOH mix, minimal reverb in the monitor mix. You can’t do this with inserts, as the return happens on the same channel/bus, and once the FX is inserted into the channel/bus, it goes where ever the channel is routed: FOH and mons alike.

    To do a send-return configuration on the Zed10, you’d go from the FX out socket to the “Input A Left” of the MPX100 (your send.) You turn up the FX send pots (yellow) on the channels you want the FX on in order to get signal to the MP400.

    NB that the FX send is post-fader, so the red channel faders (or pots, as is the case on the Zed10) need to be turned up too. This also routes signal to the Zed10’s own FX engine. Ensure the returns from this engine (grey “FX to Aux” pot and yellow “FX Level” pot at the bottom are turned down, otherwise you’ll be hearing the Zed10’s own FX library in your mix.)

    Now that you’ve got signal going into the MX400, you need to return it to the Zed10 so it can be part of the mix. To do this, run cables from the “Output A Left” and “Output A Right” sockets of the MPX400 to one of the spare stereo channels on the Zed10 (your return.) You could also use a pair of mono channels too, and pan them appropriately. You could also use the “Playback” channel, assuming you’re not using it for anything else. You can then put this returned channel(s) at appropriate levels in your monitor using the grey “aux” pot(s), and in FOH using the red “level” pots.

    NB do not have the returned channel(s) feeding into the FX bus (yellow FX pot on return channel(s) should be turned all the way down,) otherwise you’ll have a feedback loop.

    The issue is that the FX out and aux out (which is already in use to feed your monitor) buses are mono. The FX bus will sum whatever channels you’re feeding it before sending it out to the MX400. If you’re sending the MX400 two sides a of a stereo channel, they are being summed before they hit the MX400, which would mean you weren’t running true-stereo anymore. To get around this you’d have to use 2 separate buses to feed the MX400, one for left and one for right. The problem is that the only other bus on the Zed10 is your aux out (which is already being used for monitors, and is pre-fader, which makes things awkward.)

    What you could do is use the Zed10’s recording bus as an FX send, as it’s post-fader and takes the channel’s pan into account. This would maintain the stereo image, however the send levels are locked post-fader at unity. If you use the recording bus and the RCA monitor (not aux) outputs, then you effectively get level control over the FX send too. If you do use the recording bus/monitor sockets with RCA outs, you’ll need adapters to get sound into the MX400, as the RCA IO on the MX400 are for S/PDIF, not analogue audio. As I said at the start of point 3, it really depends on which inputs you want the FX to apply, or if you want them to apply to the whole mix.

    HTH

    #50486
    Profile photo of quebecguitare
    quebecguitare
    Participant

    Thank you very much for your well articulated (and illustrated!) post.

    I gotta couple quick questions:

    If I need to use both effect and processing (compression, limiter etc.) from the LEXICON MX400, do I have to hook it up differently, knowing the processing usually go in front of the chain and effects at the end of the chain?

    In the case that I want to control the FX fader on each channel, if I understand well, I will need at least a dry and wet channel. That means, for an instrument it takes 2 channel each time (ex.: M1 and M2), am I correct?

    Given that I would use the Record Out and the Monitor Out both in RCA with the LEXICON MX400, is there anyway I can still record direct from the mixer?

    If I instead use the USB port with a laptop with Cubase installed, with a LEXICON VST REVERB PLUGIN, can I control the FX fader of each channel to blend in the effect?

    If not, how does this applies in a USB setup?

    Would you advise me to get a different mixer for my setup?

    More on my scenario…

    I plan to use this mixer for the following setup:
    1. Studio practice with only 1 QSC K10 as a monitor
    –2 Guitars into 2 Eleven Rack Units which have Main OUT in Stereo XLR
    –For these practices we use backing tracks. I was hoping to feed the backing tracks through the USB port of the ZED-10 using the A&H driver in Cubase.
    Question about stereo backing tracks: It it “enough” as an EQ to be able to control only HF and LF (the ZED-10 does not have mid on the stereo channel)?

    2. Live with 3 QSC K10, 2 for the room, 1 for stage monitor
    –1 Guitar into 1 Eleven Rack unit
    –Using stereo backing tracks

    3. Live with 3 QSC K10, 2 for the room, 1 for stage monitor
    –1 solo Classical Guitar Hooked up to a mic
    –Using either reverb from the LEXICON MX400 or from a VST plugin on a Laptop

    P.S.: Should a solo guitar live setup use 2 mics (same or different) or just one?

    Thank you very much for your time HTH.
    Best Regards

    #50496
    Profile photo of cornelius78
    cornelius78
    Participant

    Major edit: Just realised I’d been referencing the Zed10 FX instead of the vanilla Zed10 in my previous post. Ignore what I said about yellow pots (you weren’t going to use them anyway, the ones you were going to use to send channels to the FX bus are now grey.)

    If I need to use both effect and processing (compression, limiter etc.) from the LEXICON MX400, do I have to hook it up differently, knowing the processing usually go in front of the chain and effects at the end of the chain?

    If you wanted processing (eg a compressor) on the start of a channel’s signal chain you’d use the channel’s insert point. If you then wanted FX (eg a reverb) at the end you’d need either another insert point, or to use a send-return. Unfortunately the Zed10 doesn’t have insert points per channel, it only has inserts on the main LR. Bear in mind that in a live situation you probably don’t want too much compression in monitors (particularly wedges) as it’s a recipe for feedback. What you could do is channel>fx bus>input A of MX400 (compressor) > output A of MX400 into input B of MX400, then Output B of MX400 return to Zed10. That would effectively get you compression followed by reverb on a single channel. It all depends on which channels you actually want to have which FX applied.

    In the case that I want to control the FX fader on each channel, if I understand well, I will need at least a dry and wet channel. That means, for an instrument it takes 2 channel each time (ex.: M1 and M2), am I correct?

    Yep, you need a dry channel, the send (the fx bus) and a spare channel (or 2) to use as the FX return (the “or 2” is if the return is to be stereo, eg for a stereo delay.) You can use a pair of spare mono channels and pan them or you can use a stereo channel (which is automatically panned.)

    Given that I would use the Record Out and the Monitor Out both in RCA with the LEXICON MX400, is there anyway I can still record direct from the mixer?

    You don’t necessarily need to use the record out and the monitor out to feed the MX400, that was just an idea so that you’d effectively get a master send level control on the Zed10 for the FX bus feeding the MX400. You could just use the record out by itself (as long as the MX400 has input trims.) Some recording options include recording the main mix via the USB socket to the computer, getting creative with the phones/monitor sockets, or you could just y-split the main outs into an analogue recorder (or the line-in socket on your PC.)

    If I instead use the USB port with a laptop with Cubase installed, with a LEXICON VST REVERB PLUGIN, can I control the FX fader of each channel to blend in the effect?

    Yes. You send individual channels to the FX bus using the grey “FX” pots on the channel strip. You then send the FX bus (acutally it ends up being the aux + fx bus, aux on L, FX on R) to the USB socket, using the Record Out+USB source out selector switches. This sends the audio to your computer. Use you DAW to listen to the R side of the stream: that’s the fx bus. Apply whatever plugins you want to the track in your DAW. Have the DAW output the returns from that track to USB L+R. The stereo return will appear on the “Playback” channel of the Zed10. You can then turn up this channel in the main mix and/or the aux using the relevant pots on the Zed10. If you decide you want more reverb on one channel and less on another you can use the fx pots on the individual channel strips. p27 of this manual explains it.

    Would you advise me to get a different mixer for my setup?

    The Zed10 seems like a capable mixer for your setup, just, provided you don’t want to get too fancy with FX routing. The next step up would be something like a PA series/mixwiz, with inserts and direct outs, more post-fade buses to feed your outboard FX units and a multitrack USB option.

    You’ve listed some scenarios, which is good, and the Zed10 can handle all of them to a degree.

    Your scenarios:

    1.
    Backing tracks can come through on the USB port and end up on the playback/stereo 2 channels. Chances are they are already mastered and pretty well eqed anyway, so you shouldn’t really need to eq them anymore, so a lack of eq shouldn’t be a problem. If you’re playing them through your DAW chances are you can use an eq plugin on them in the DAW if you decide you do need to eq them.

    2.
    If you’ve got stereo backing tracks coming through via USB then you won’t be able to use that USB input as an FX return as well (unless you mix backing tracks + reverb return in the DAW, and use the DAW for level control of each. It’s complicated to keep on top of, but possible.) It would be simpler to use an analogue out from your PC into an analogue input on your Zed10 for the backing tracks, and keep the USB IO exclusively for FX via the DAW. That way you keep separate level control of both on the ZED10.

    3.
    This is probably the easiest application. Mic into channel 1, FX either via USB and DAW or via MX400. You could even use both DAW and one side of the MX400 for FX simultaneously, and use the other side of the MX400 as an insert on the mains if you wanted.

    If you can give a little bit more detail on your scenarios, particularly wrt which fx you’d like on which channels, we can get a better idea of whether the Zed10 suits your intended application.

    PS.
    You can mic a single guitar with 2 mics if you want, but you need to control the mic positioning and process the signals really well for it to be worth it. It’s a lot easier to do in a studio with a DAW than it is to do live. Some dual-micing techniques require inverting the polarity of one of the signals in order to negate the phase cancellation, but the Zed10 doesn’t have polarity invert switches, so you’d need to use a deliberately mis-wired cable/adapter to create the effect. IME it’s it’s easier to just use a single mic (either a dynamic pointed at the sound hole, or a condenser a bit further back acting as an ambient mic,) then process to taste (usually a bit of stereo verb/chorus works well to space it out a bit.)

    HTH is interweb speak for “hope that helps,” they’re not my initials.

    HTH

    #50499
    Profile photo of quebecguitare
    quebecguitare
    Participant

    Hi cornelius78, once again thank you for your attention to details. It’s a delight to read you.

    Just a couple quick questions…

    Bear in mind that in a live situation you probably don’t want too much compression in monitors (particularly wedges) as it’s a recipe for feedback. What you could do is channel>fx bus>input A of MX400 (compressor) > output A of MX400 into input B of MX400, then Output B of MX400 return to Zed10. That would effectively get you compression followed by reverb on a single channel. It all depends on which channels you actually want to have which FX applied.

    For a live solo classical guitar scenario without backing tracks…
    I do not understand how an effect could be sent differently from processing (Compression), from a single signal processing unit like the MX400. How is it possible that the Compression will be at the front of the chain and the Reverb at the end?

    Further more, “feedback being a known issue” in stage monitors, is there anyway that I can, with my 3 QSC K10 setup, avoid feeding the stage monitor with Compression, but leaving other effect such as Reverb and Chorus (for solo classical guitar input mic to a channel).

    On a slightly different topic…
    I am eye balling this ZED-10 on an Ebay auction in Japan.
    http://www.ebay.com/itm/ALLEN-HEATH-USB-equipped-compact-mixer-ZED-10-/311431415622?hash=item4882c24746

    Here is the description of the Item:
    Condition: New

    Brand: ALLEN & HEATH
    Manufacturers Part No. ZED-10
    Shipping Weight 4.8 Kg
    Handling start date at Amazon.co.jp 2010/6/11

    Note*
    This product is available in Japan. Almost products have Japanese in its instructions or contents. Please understand that for your shopping. Also, when you buy Japanese electric products from us, please make sure about Japanese electronic equipment. The voltage and the plug vary across countries. You may need the electric transformer and the conversion plug to use that kind of the products.

    Check Plug/Outlet Type in Japan here.
    http://treehouse.ofb.net/go/en/voltage/Japan
    Voltage, Frequency and Plug/Outlet Type in Japan – Treehouse Cityguide
    treehouse.ofb.net

    I’ve asked the seller details about the Main power specs of the unit but he couldn’t answer me. They are selling a lot of stuff and there is a language barrier…

    I wanted to ask you:
    Does A&H ZED-10 ships to all country with the same POWER AC MAINS IN ~
    100 – 240V~ 47-63Hz 15W ?

    Do you think this could be a problem?

    I live in Indonesia, 100-240v 50Hz do fine as the power voltage here is 230v.

    Thank you
    Best Regards.

    EDIT: Do you think the Bezel would be in Japanese (I doubt)?
    Also, what do you make of the 2010 date as a reference (from amazon) in the product description, was there any revision done to the ZED-10 since 2010?
    Is there any issue that you see with that date?

    Thank you
    Regards

    #50534
    Profile photo of cornelius78
    cornelius78
    Participant

    For a live solo classical guitar scenario without backing tracks…
    I do not understand how an effect could be sent differently from processing (Compression), from a single signal processing unit like the MX400. How is it possible that the Compression will be at the front of the chain and the Reverb at the end?

    You daisy-chain the FX processors (side A and side B) within the MX400, eg the output of side A (compressor) feeds the input of side B (reverb.)

    Further more, “feedback being a known issue” in stage monitors, is there anyway that I can, with my 3 QSC K10 setup, avoid feeding the stage monitor with Compression, but leaving other effect such as Reverb and Chorus (for solo classical guitar input mic to a channel).

    An unconventional method (because you can’t un-assign channels from LR on the Zed10,) would be to do the following:

    Mic into ch1, ch1 turned down in aux and LR, PFL ch1, feed MX400 side A from PFL bus. Side A set up as reverb and y-split its returns; one to spare stereo channel 1 on the Zed10, the other to side B of Mx400. Spare Stereo channel 1’s aux 1 send turned up for foldback, but not turned up in FOH. Side B (with compressor settings) of MX400 returns to spare stereo channel 2 on Zed10. This 2nd spare stereo channel is the only one that gets turned up in FOH. That way you get reverb only in your monitor, and reverb + compression in mains.

    Does A&H ZED-10 ships to all country with the same POWER AC MAINS IN ~100 – 240V~ 47-63Hz 15W ?

    TBH I’m not sure, I’ve never unboxed one. I do know that in some locals, the local distributors will unbox and re-pack with a local power cable. IME there are some companies (like Dell,) who ship their “international versions” of their products to Australia, and we end up with manuals in 5 different languages and at least 3 different power cables. You’re buying off ebay, who knows what you’ll get. I don’t know what things are like in Indonesia, but here in Australia you can buy a brand new power cable from a department store for about $5 AUD (because here we’re not allowed to simply chop the end off and fit a new plug, unless we get a licensed electrician to sign off on it (for which they know they can charge $$$,) despite the fact that some of us learned to do it when we were 10 and have managed not to set fire to anything… stupid protected industry… we also can’t legally crimp\install our own CAT5 cables without having got the relevant registration +endorsements (600 hours of approved supervised experience, exams + registration fees… but I digress…)

    Do you think the Bezel would be in Japanese (I doubt)?
    Also, what do you make of the 2010 date as a reference (from amazon) in the product description, was there any revision done to the ZED-10 since 2010?
    Is there any issue that you see with that date?

    I very much doubt the silk would be in Japanese. All the pictures I’ve seen on the internet all show a roman alphabet.

    No idea re the 2010 date. I’m not overly familiar with the mixer, I’ve only used one a couple of times, I haven’t kept up with revisions like I do with digital mixers in terms of features added with new firmware. There might be some hardware difference, eg the new ARs for GLD/Qu are a differnt colour and now have locking XLRs, but with analogue equipment they tend to just release a new model with more buses, or an inbuilt FX processor etc, eg Zed10 > Zed10 FX > Zed 12 FX etc.

    #50558
    Profile photo of quebecguitare
    quebecguitare
    Participant

    Hi cornelius, I can’t thank you enough for your time and lengthy replies. I think we’ve been to the bottom of that.

    I just bought the ZED-10 from Japan.

    I wanted to ask you… what do you think of the dbx DriveRack PA2?
    http://www.sweetwater.com/store/detail/DriveRackPA2

    Any cheaper substitute for our scenario?
    I am looking to get the best possible sound out of our QSC at the best budget and user friendly. I am an amateur…

    Do you think it could be a good investment for our setup and different scenarios or it’s a loss of money?

    Thank you
    Rgards

    #50576
    Profile photo of cornelius78
    cornelius78
    Participant

    I think speaker management is always a good idea, providing you set it up properly. As with lots of things in audio, it’s entirely possible to go too far and butcher the sound if you set it up wrong.

    The PA2 is an amalgamation of devices: a 3-way crossover, a peq, a sub-harmonic synthesizer, compressor/limiter and line delays among other things. IMO it’s good for what it is. An (appropriate) cheaper substitute would depend on what you wanted to use it for: eg if you only wanted to use it as a comp/limiter as a bit of protection for your speakers/peoples’ ears, then a standalone 2-channel compressor would probably be cheaper. If you only wanted a FOH eq, then a standalone 2-channel peq would probably be cheaper. You’d be hard pressed to find something cheaper will all of the above functions in 1RU though. It also has built in presets for your QSCs, which is handy.

    I think it’s a good investment in that it’s protection for your speakers/the audience’s ears, if you later decide to add a sub to the FOH setup it can cope with that too, and that adding the opportunity to have FOH eqs and line delays that you wouldn’t otherwise have, all of which can improve the overall sound, providing you set it up properly. It’s also good in that it can save presets, so if you have a gig in one place you can dial in the eq for that room and save it. Next time you’re in that room you can simply recall the preset you previously saved.

    One thing I will say is that IME auto feedback destroyers are really only useful for dry speech. IMO they are a combination of too slow and too drastic for music: they tend to make it sound worse. Also, re the unit’s AutoEq, while it will get you some of the way there, IME you almost always get better results if you tune by ear (or start with the auto eq then go back and manually adjust the resultant curve.) If you are going to use the AutoEq then you should probably invest in a measurement mic too. If you’ve got a wireless router and a smartphone/tablet you can adjust settings remotely too, which can be handy.

    #50583
    Profile photo of quebecguitare
    quebecguitare
    Participant

    Thanks Again cornelius.

    There is a bunch of dbx DriveRack PA2 on Ebay. Do you think people are getting rid of them because they are no good? Or because it’s a tool for amateurs, hence a good start for me?

    http://www.ebay.com/sch/i.html?_from=R40&_trksid=p2047675.m570.l1313.TR0.TRC0.H0.Xdbx+DriveRack+PA2.TRS0&_nkw=dbx+DriveRack+PA2&_sacat=0

    Would you have any advice on gear to start our setup for the given scenarios?

    Any suggestions for doing the sound engineering for multiple backing tracks from different sources (variable mastering).
    This must sound like a stupid question but I’ll try: does an EQ have the ultimate last word of making multiple backing tracks sound the same (in an evening program) and mix well with a guitar signal that would hypothetically remain on the same patch for all backing tracks (eleven rack guitar processor)?

    Does this sounds like a good start (I have very limited knowledge in sound engineering)?
    3 QSC K10
    A&H ZED-10
    dbx DriveRack PA2
    882i Sonic Maximizer

    AKG C1000S Mic for LIVE Flamenco Guitar Gig (any better suggestions for live Flamenco Guitar microphone?)

    What would you suggest for a measurement mic for the AutoEQ of the dbx DriveRack PA2?

    Any comments about the setup?
    I would imagine this PA system would be for outdoor gig as well.

    Thank you
    Best Regards

    #50584
    Profile photo of cornelius78
    cornelius78
    Participant

    Re PA2s on Ebay:
    I think the influx of speaker management processors is largely due to the uptake of digital mixers. Most digital mixers (including A&H’s iLive, GLD and Qu series) offer geq and/or peq on outputs, compressor/limiting and line delay, making some of the functionality of the PA2 redundant. Some people also use the output processing on digital desks as a crossover too (though personally I’m against that practice,) further reducing the need for something like a PA2 in their system.

    Re eq having the last word:
    TBH I don’t really understand the question. I do know that if you use the eq to notch a few of the frequencies of the backing track that the guitar would normally stomp on, you can get the guitar sounding clearer in the mix without needing to boost its level as much. Also, if you were to compress both guitar and backing tracks with separate compressors, but have the threshold lower on the backing track channel, this would allow the guitar to naturally sit on top of the backing track, without needing to ride the guitar fader as much.

    Re your gear:
    I wouldn’t use the sonic maximizer. Used very sparingly when mastering a 2 track it can be a good thing, but live on guitar tracks is just asking for trouble, IMO. Makes the sound gets bassy and shrill at the same time (essentially a disco (smiley-face) eq,) and you lose your mids: not ideal for a track that supposed to be guitar-centric.

    I’m not a guitarist, (though I’ve started learning bass) and I have almost no experience in micing acoustic guitars. With the gear\bands I’ve had access to, I’ve just been putting ’57s \ e906s on cabs for electrics, and acoustic guitars have been put through Radial\Whirlwind\etc DIs.

    Re measurement mics:
    Lots of people use a Behringer ECM8000, DBX also do their own measurement mic (mostly because they’re cheap.) I standby what I said before though: while if the AutoEq worked 100% properly and could use these mics to make the speakers sound 100% “flat” in the room (at least in the spot you took measurements,) your ears would still do a better job of making the speakers sound “good” in that room. “Good” doesn’t always mean “flat.”

    No comment on using the gear outdoors. I’ve seen K series setup outdoors, but the weather was good and I wasn’t really paying attention to how big a space they needed to fill. Generally speaking, being outdoors (eg if your did a gig-in-a-field) you deprive yourself of some of the reflective surfaces (eg walls and ceilings) that would normally provide you with some natural reverb. Your indoor mix might sound a lot thinner than you were expecting when done outdoors, due to the lack of natural reverb normally provided by the room. Often you end up wanting to use an FX processor to add some more verb to the mix to make up for it. You won’t know how much until you get there though.

    #50660
    Profile photo of quebecguitare
    quebecguitare
    Participant

    Re eq having the last word:
    TBH I don’t really understand the question. I do know that if you use the eq to notch a few of the frequencies of the backing track that the guitar would normally stomp on, you can get the guitar sounding clearer in the mix without needing to boost its level as much. Also, if you were to compress both guitar and backing tracks with separate compressors, but have the threshold lower on the backing track channel, this would allow the guitar to naturally sit on top of the backing track, without needing to ride the guitar fader as much.

    “needing to ride the guitar fader as much”

    Yeah I forgot to mention that I will perform and do the mixing at the same time for this system. I do not have a sound engineer.

    Thus you will understand my problem, I need to know if I can start an evening program without “needing to ride the guitar fader as much” if the PA2 and Mixing EQ on the ZED-10 is setup to a position, giving that I will use many backing tracks that have different levels, noise and type of mastering.

    Could you give me more details on how to compress both guitar and backing tracks with seperate compressors, to “allow the guitar to naturally sit on top of the backing track” ?

    Can I use VST plugins through USB for this or I have to get hardware (DBX 160, dbx 266XS ??)?

    What would you suggest?
    Do you flat out the EQ on the ZED-10 once you have the DBX PA2 between the PA and the FOH speakers?

    Do you also manage the stage monitor in the DBX PA2 system?

    Following your advice I will not buy the BBE Sonic Maximizer for Live Setup.
    We’ll definetely get a diagnostic mic to start with the AutoEQ of the DBX PA2, then tweak from there, saving presets as well for each venues.

    Thank you,
    Best Regards

    #50691
    Profile photo of cornelius78
    cornelius78
    Participant

    Thus you will understand my problem, I need to know if I can start an evening program without “needing to ride the guitar fader as much” if the PA2 and Mixing EQ on the ZED-10 is setup to a position, giving that I will use many backing tracks that have different levels, noise and type of mastering.

    This is what your sound check is for. However before you start it might be a good idea to normalize your backing tracks against each other so that despite their different dynamic ranges, at least they start at a similar base line throughout the gig. This will allow you to set up your guitar gain and fader to be at a sensible level, and if you leave yourself enough headroom with your preamp, you’ll be able to control the guitar dynamics through your playing style, rather than having to reach over and mess with faders all the time, which I think is a better way to go for a soloist who is also mixing.

    Could you give me more details on how to compress both guitar and backing tracks with seperate compressors, to “allow the guitar to naturally sit on top of the backing track” ?

    When you set up the compressor on the backing tracks, you set it with a threshold low enough such that it kicks in before the compressor on your guitar\mains kicks in (you need to experiment with levels with backing tracks and guitar being played at the same time to get it right.) That way, when things start to get loud, the backing track gets compressed first before the guitar/mains do, which allows the guitar track to sit on top and have some “breathing room.” You should be able to get compressors in your DAW to do this. This should mean that if the backing track gets loud, and for whatever reason the guitar doesn’t (perhaps you’re finger-picking) your guitar would still be audible without you having to turn the guitar fader up/the tracks fader down. It really depends on the style of music.

    Do you flat out the EQ on the ZED-10 once you have the DBX PA2 between the PA and the FOH speakers?

    IMO the FOH eq (PA2) should be used to make the speakers sound good in the room, eg flattening them if that helps, and dealing with room resonances, perhaps knocking out feedback if that can’t be done on the channel eq etc. The individual channel eqs should be used to “fix” individual channels, eg use the channel eq on the backing tracks channel to take out a some of the freqs that your guitar is likely to stomp on, and use the guitar channel’s eq to take out a few of the problem freqs of the guitar (eg sound hole resonance) or enhance brightness/presence. Remember it’s usually better to cut rather than boost.

    Do you also manage the stage monitor in the DBX PA2 system?

    It would be ideal if your monitor also had some processing on it too, however the PA2 only has 2 inputs. If you were running FOH with a mono source you could use one side of the PA2 for FOH, and the other side for your stage monitor. If you want stereo FOH, then you’ve run out of inputs on the PA2. For speaker management on stereo FOH + monitors, you’d need either a speaker processor with more inputs, or another PA2 unit.

    #50696
    Profile photo of quebecguitare
    quebecguitare
    Participant

    it might be a good idea to normalize your backing tracks against each other so that despite their different dynamic ranges

    Thanks! I’ll do that in Cubase.

    When you set up the compressor on the backing tracks, you set it with a threshold low enough such that it kicks in before the compressor on your guitar\mains kicks in

    Brilliat Stuff! I had no idea you could that in real time… thanks!

    By the way the guitar style are this:

    Evening Program:

    -Compression needed and soundcheck-
    Set1 – Jazz standard with backing tracks (clean Gibson Style Input)
    Set2 – Heavy Rock Fusion with Backing Tracks (High Gain)

    -Solo-
    Set3 – Solo Nylon Guitar without (Backing Track)

    Remember it’s usually better to cut rather than boost.

    Brilliant Tip, thanks!

    It would be ideal if your monitor also had some processing on it too, however the PA2 only has 2 inputs

    There is something I don’t understand about the DBX PA2 I/O. I thought the 2 inputs we’re for the mixer and the 3 sets of Output for the FOH speakers.

    Ref: http://www.sweetwater.com/store/detail/DriveRackPA2

    I/O list PA 2 x 6 PA Management
    2 Input (XLR female)

    2 Output (XLR male) for High
    2 Output (XLR male) for Mid
    2 Output (XLR male) for Low

    I think the setup is this:
    the ZED-10 output goes in the 2 Input (XLR female) of the DBX PA2 and the FOH speaker in monitor goes either in the High, Mid or Low ?

    On my QSC K10’s I have 2 Channel at the back of the speaker. If I use only one channel, does that mean I use only half of the 1000W specs of the speaker?

    Could you advise on how to hook up the K10 with the DBX PA2 knowing that there are high, mid and low output, assuming that these are for the speakers?

    Maybe I can setup these input accordingly in the DBX PA2, configuring them as QSC K10 bypassing the high, mid and low original setup and plug in both channel of each K10 in the Left/Right of the output of the DBX PA2 ?

    If it’s not possible, would there be anyway to add input through a small box that would allow me to manage 3 K10 without having to get a second DBX PA2 ?

    Thank you
    Regards

    #50699
    Profile photo of cornelius78
    cornelius78
    Participant

    I think the setup is this:
    the ZED-10 output goes in the 2 Input (XLR female) of the DBX PA2 and the FOH speaker in monitor goes either in the High, Mid or Low ?

    No. There are 2 inputs to the PA2. Normally you’d use them for FOHL and FOHR. There are 3 outputs per side for use when you use the 3-way crossover functionality. Left-Lo, Left-Mid and Left-Hi. Same deal for the right. If you’re not using the crossover functionality (eg because you don’t have a sub/aren’t running bi-amped/tri-amped mains, your FOH speakers have their own XO network, as is the case with the K10s,) then you can just feed your FOHL speaker from the Left-Hi output, and your FOHR speaker from the Right-Hi output. Ignore the other 4 outputs. Perhaps it would help to think of them as inputs 1&2 and outputs 1L, 1M, 1H, 2L, 2M and 2H. If you’re running FOH from a mono source, you could use input1 and its associated outputs for FOH, and input2 and its associated outputs for your monitors.

    On my QSC K10’s I have 2 Channel at the back of the speaker. If I use only one channel, does that mean I use only half of the 1000W specs of the speaker?

    No. Look at a block diagram for the k10s. The two inputs are summed together before they get processed and XO’d, and only then do they hit the amplification stage.

    Could you advise on how to hook up the K10 with the DBX PA2 knowing that there are high, mid and low output, assuming that these are for the speakers?

    See above (or the PA2 manual/quickstart guide.) If not using the XO functionality, use the Left-Hi and Right-Hi outputs only, ignore the other ones.

    If it’s not possible, would there be anyway to add input through a small box that would allow me to manage 3 K10 without having to get a second DBX PA2 ?

    There isn’t if you want to maintain your stereo image. FOHL and FOHR are two different signals. They each need their own processor if they are to remain separate and be sent to two different speakers. If you wanted to crunch them together before the PA2 you could, and that would free up a PA2 channel to use for the stage monitor. If you know you don’t need the full functionality of a PA2 for the stage monitor (eg you don’t need an XO, and geq+peq + an AutoEq + a limiter + a line delay + and RTA etc,) then perhaps a stand alone peq/geq to knock out feedback would be a sensible compromise for your stage monitor.

    #50905
    Profile photo of SteffenR
    SteffenR
    Participant

    quebecguitare you should pay Cornelius a lot of money for that crash course in live sound reinforcement… 🙂

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