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Tech crew blowing speakers too frequently...

Discussion in 'Sound, Music, and Intercom' started by tenor_singer, Apr 23, 2004.

  1. tenor_singer

    tenor_singer Active Member

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    Hello!

    I want to start off by saying that I have a very dependible tech director. I think she is smart, dedicated and wants to do the best job possible. She is also seeking outside help with various aspects of sound and lighting.

    My question...

    We tend to pop the tweeters on our speakers an average of two times a year and almost always on Wednesday of tech week. My friend, who is an engineer and TD for an area theater, said that it was because we were clipping the sound.

    1. What is clipping the sound? I was under the impression that it meant turning down one wave frequency to almost zero while cranking the others.

    2. I think that they are popping because some fool always trips on the plug and unplugs the system from the wall sending a loud "thwack" through the speakers. Could this be causing it? If so that was this fall's reason.

    3. This spring, my TD turned on a body mic before checking it's volume on the board, which was pegged (some jack-off was messing around with the slides when she wasn't looking). The feedback that went through the system was amazing (unfortunatelly she didn't have the destroyer programmed yet). Would this cause the speakers to pop?

    4. My system has a 1000 Watt Amp and two EV 300sx speakers. Are they too weak for the amp?

    5. I will be showing my tech crew any responses to this post, so any additional advise you have for them would be great.


    If we were to rate my knowledge of sound engineering and lighting on a scale of 1 - 10, it would be a 6. My crew's have been trained to the best of my ability, so their level is similar. My ability to translate answers will be limited, so layman's terms please :D.


    Yours in theater,

    Tenor
     
  2. TimoteusR

    TimoteusR Member

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    I'll answer questions 1 & 4:

    here is a basic explanation. Your audio is being cut of at the tops (and bottoms ) of the sound wave this will cause your speaker to try and stay at the extreme positions (at forward or back) and switch quickly.

    More info at http://www.prosoundweb.com/studyhall/studyjump.php?
    pdf=clipping



    I have been considering these speakers also. If you are talking about the EV sx300e their RMS power is 300W, this means you'll want to provide between 480W-750W per speaker. 600W would be the optimum. The basic calculation you can use is

    RMS x 2 x 0.8 this is the minimum
    RMS x 2 x 1,25 this is the maximum

    http://www.prosoundweb.com/studyhall/studyjump.php?pdf=watts

    If you are running 1000W per speaker, that is a bit too much.


    If you are overpowering your speakers and have these thumps and pops coming through without filtering that will surely blow out stuff. Your friend is right. If you are running clipped signal to speakers at that level (overpowering) it is likely something will blow.


    Check the LAB archives at http:/www.live-audio.com/index.html for more info on sound stuff.

    I can give you additional tips if you list your sound equipment.

    --
    Timoteus Ruotsalainen
     
  3. TimoteusR

    TimoteusR Member

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    I forgot to ask about impendace. If you could post the amp's name and manufacturer it would help. If you are running the amp stereo with one sx300 per side you need to check how much that amp puts out at 8ohm.

    --
    TImoteus Ruostalainen
     
  4. tenor_singer

    tenor_singer Active Member

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    Thanks Timoteus!

    I tell my crew never to turn any one wave frequency (my mixing board has a low, two midlevel, and a high) completely off. I will adjust that to mean that they should not turn the high and low too far down while keeping the midlevel high.

    Will clipping occur if they turn the midlevel down while keeping the high and low too high?

    I have given your request for information to my TD and she will get that to me asap and I to you. Thank you very much for your help!


    Yours in theater!

    Tenor
     
  5. DMXtools

    DMXtools Active Member

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    Clipping occurs when you ask an amplifier to put out more power than it can. As Timoteus said, it basically "clips" the top and bottom bottom curves off the sound waves, leaving very sharp corners from where the wave is rising or falling to the flat top or bottom where the amp hits its limit. These sharp corners are loaded with high-frequency energy, which is why it blows the tweeters.

    A lot of more recent power amp designs have a built-in limiter (Peavey calls it DDT) to prevent this, but many of them let you switch it off (I don't know why). Look for a switch, probably on the back of the amplifier, labelled DDT or LIMITER or something similar and make sure it's on.

    Heavy feedback is often at fairly high frequencies. Besides that, feedback can also push an amplifier into clipping very easily - and once again your tweeters get the brunt of it.

    A neat trick is to put a lightbulb in series with your tweeters. An 1156 (used as a back-up light on a lot of cars and available at most auto-parts stores) works well.
    Tungsten (the metal the filament is made of) has a very low resistance when it's cold, and won't interfere with the signal at normal levels. However, its resistance rises exponentially as it heats up. Power levels large enough to damage the tweeter are also large enough to light the bulb, raising its resistance. Then the bulb (under a dollar) takes the brunt of the power surge and protects the speaker. Note that this is just for the tweeter, not the whole speaker system. If you mount the bulb where it can be seen from the booth, the sound tech can watch and pull back the gain if the light comes on. In four years of running these tweeter lights in four cabinets I've burned out one bulb and NO tweeters.

    John
     
  6. Mayhem

    Mayhem Senior Team Emeritus Premium Member

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    Interesting trick John. You could also use "polyswitches" which do a similar job. Essentially, these components (which look like ceramic capacitors) go "open" when they get too hot and then as they cool down, they close the circuit again. You need to match them to the wattage and impedence of the speaker being protected and they work very well. The bonus here is that unlike the globe that needs to be replaced, backing off the power will cool down the polyswitch, allowing the gig to continue.

    Also, Mosfet power amps are prone to oscillation at higher frequencies whcih also will take out speakers. This is a lot harder to detect as it is past the audiable range. I do not know all that much about it but I am lead to belive that the polyswitches can offer protection from this as well.

    John - are you able to add to this?

    Another point regarding clipping - this actually means that you are sending DC or direct current to the speakers which is bad. The reason that I mention this is that you may hear someone suggest "checking if an amp has DC at the outputs". This is what they are checking for.
     
  7. TimoteusR

    TimoteusR Member

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    Thanks Dmxtools. Nice explanation. Clipping can occur in any amplifier. It can happen on your mic preamp, buss amp or power amp. Clipping basicly happens when you ask an amplifier to do more than it can. Turning up the "band" controls or equalizers as they are called by the pros (hi,mid,lo) can ask the amp to do more than it can. You should always (Remember, exception proves the rule) cut with eq as opposed to boosting. Be especially careful of boosting the extreme low (and high) end. You might not hear a change, as the speaker will not produce any sound at these frequencys (like asking your sx300 to produce 30hz) but it surely will heat up the speaker as it trys to.

    Do you know how to setup proper gain structure?


    If you could provide more info on your setup, we could help more. here are a couple of questions.

    What model speakers are you using?
    What is the make and model of your amplifier?
    Is there a mixer, if yes what make and model?
    Are there any other outboards such as effects or graphic eq's?
    Any mics or di's?
    What kind of cabling?
    How much power, or how big circuit brakers are you using?
    Any specific problems other than tweeters blowing?

    If you go with the light bulb trick, be sure to cover it up, and/or tell others about it. There's a story on LAB about a fellow, who was a little drunk, that got scared and ran away when he saw the lamp light. He thought the monitor wedge was possesed by demons...


    Ok, I re-read your post's and i would like to clear something up. Clipping happens when your amp (be it pre, buss or power) can't produce what it is asked to do. This happens to the whole signal (the amp doesn't sort the signal based on what frequencies it has. It does it's things to the whole signal). The "wave controls" (EQ) separet the signal into different bands and you can amplify them separetely. So turning some bands completely off will not cause clipping. It will just make the sound suck most times. BUT boosting any one of those controls might cause clipping. If any red lights are flashing, you are nearing the clipping point. Indicators marked PEAK are usually used.


    One way to demonstrate clippping is to use a piezo and a 440hz (A) signal. You can also use this to check for proper gain structure. Just put the signal through your mixer (with amps off or speakers disconnected, the sound get's annoing after a while). Then put everything on the channel your feeding the signal to to 0db. then connect your piezo to your mixer outs. Now turn the gain up on the channel trim until you hear a "sound" from the piezo.

    --
    TImoteus Ruotsalainen
     
  8. TimoteusR

    TimoteusR Member

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    A quick search on google gave me the following info. A clipped sine wave (square wave) isn't DC as it doesn't have the properties of DC. Whatever that means.

    The propeblem with clipping is that it make sthe speakers try to extend all the way to the front. Then after being there for a while jump all the way back. This causes stress to the speakers. It also heats the voice coils. When the speakers heats up, the impendace grows. This makes you push the fader even more thus heating the speaker more. This can also be caused by feeding signals (frequencies) above or below the speakers production range. Like 30hz to a mid speaker capable of 400hz-3khz. Amps should have DC protection. DC damages speakers as it has some properties of a clipped sine wave (square wave).

    I can't believe i wrote all that ;)

    --
    Timoteus Ruotsalainen
     
  9. DMXtools

    DMXtools Active Member

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    The problem with polyswitches is excessive hysteresis - once they trigger, you have to cut the power down to almost nothing to get them to reset. The lamp, as long as it doesn't blow, acts more as a compressor/limiter, with the resistance dropping rapidly as the power falls below dangerous levels.

    Not really, most of my experience has been with bipolar power stages.

    Actually, not true. DC at the outputs is generally the result of the failure of one or more transistors in the amplifiers. Also, being that it's a two-channel amplifier driving a pair of speakers with passive crossovers, DC is effectively blocked from the tweeters by capacitors in the crossover. Such a fault is much more likely to smoke the woofers.

    ROFLMAO!

    John
     
  10. Mayhem

    Mayhem Senior Team Emeritus Premium Member

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    Thanks for clearing those points up John.

    I am happy to say that I have never caused a pollyswitch to activate (touch wood), so I cannot comment on the required power reduction to bring them back online.

    You are right with the DC / crossover / tweeter scenario but my comment was more aimed at clipping in general. Although, it would still seem that I am somewhat off the mark still. Am I correct in my statement that clipping causes DC voltage to appear at the Amp outputs? I know that you are correct in your comment about DC often being casued by a faulty output device but I sure that I read/was told that clipping produces DC.

    Confused :?
     
  11. tenor_singer

    tenor_singer Active Member

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    I am not an engineer by trade (I actually teach mathematics and physics at a high school). How would you go about doing this? Would this be something that I would be better off asking an experienced person to do? I am sure my one friend would help me, but I really don't like bothering him that much (he lives a good hour away from my school).

    Thanks everybody for the information. Please keep it coming:D.

    Tenor
     
  12. TimoteusR

    TimoteusR Member

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    My non-scientific test say no DC will appear at the amp outputs when clipping. This was tested with a signal source, my mixer and a fluke 112 true RMS meter. I apllied the signal and tested for DC then I clipped the signal intenionaly by overdriving the preamp and checked for DC again. Clipping was verified with a piezo, and a rise in AC voltage. No DC was registered at any time

    --
    Timoteus Ruotsalainen
     
  13. tenor_singer

    tenor_singer Active Member

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    I will be editing this post as I find more information about my equipment from our TD.

    SPEAKERS:

    I have EV Sx300 speakers.

    I am not a sound engineer, so I have found EV's engineering specs on the net. Here is the web page:

    http://www.electrovoice.com/electrovoice/EVfiles.nsf/lookup/Sx300_EDS/$File/534694.pdf


    AMPLIFIER: Samson S1000

    Rugged stereo and mono-bridgeable power amplifier

    Produces 500 watts per channel into 4 ohms (stereo mode) and 1000 watts into 8 ohms (bridged mono mode)

    Linear frequency response for exceptional audio performance

    Dual temperature-sensitive, speed-controlled fans

    Stable bipolar design

    Front panel input level controls with 42 detents

    3-segment output LED metering

    Dual protection LEDs and relay-controlled outputs linked to protection LEDs

    Banana jack outputs with Speakon(tm) connectors

    XLR inputs and locking TRS inputs

    Parallel outputs to "daisy-chain" a number of amps together

    MIXER:

    I have a Behringer Eurodesk MX2442A

    Channel inputs 24
    Mono channels 16
    Stereo channels 4
    Subgroups 4
    INVISIBLE MIC PREAMPs 16
    EQ (mono channels)
    High, low, 2 x semi-parametric mids, low-cut
    3-band
    EQ (stereo channels) 4-band
    Aux sends 6
    Stereo aux returns 2
    Channel inserts 16
    Talkback mic internal
    Phantom power +48 V

    OUTBOARDS:

    I am not sure what this is. I do not have any inserts or sub-inserts other than a feedback destroyer.

    MICROPHONES:

    I use audio technica boundary microphones (3... l-stage, c-stage and r-stage). I keep the center up only unless the action moves to l or r stage.

    I also use audio technica wireless uhf microphones (10 - 16 depending on the show). I am not sure of the specs, but they are not programable (again... a very poor school district).

    CABLING:

    Um.... my sound engineering friend got it for me and said plug it there and there. That is all I can say. It is a heavy gauge wire, though with speaker easy ends (at least that is what a sales person told me they were)...

    POWER:

    I run my sound system off of a 20-A breaker that feeds 7 different outlets around our gymnasium.



    Thanks again for the help.

    Tenor
     
  14. TimoteusR

    TimoteusR Member

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    Is your amplifier a Samson S1000?
    http://www.samsontech.com/products/productpage.cfm?prodID=77&brandID=2


    If it is i suggest you run it with the following configuration:

    The amp will only about 340W per speaker (according to the 2xRMS rule 480W would be minimum). Underpowering speakers makes it easier to clip the amp when trying to get more volume. Just run one speakon cable to each speaker and connect on speaker per amp channel. If you run a stereo signal to the amp you can use this setup with pannings.


    When using the setup make sure that your gain structure is good. Also make sure that the CLIP light on the amp (it's the third light from left to right on the amp face) isn't flashing (and flashing on every kick drum hit is very bad). This light tells you the amp is clipping. In other words your trying to make the amp provide more than it can.

    I wonder if Samson uses one power supply for both channels. If so you could use a trick that I learned from Dave Rat of RatSound. It can be found on LAB. I think it's called "Fighting for power". Basicly you reverse polarity on one of the input signals and reverse it again on the output. This way the L channel will pull "+" power while the R pulls "-" power. It supposedly gives greater headroom. At least the description sounds good.

    Whatever you do, be sure to document everything carefuly. You could even make a drawing with every component and how they are conneted together. This helps troubleshooting.

    --
    Timoteus Ruotsalainen
     
  15. TimoteusR

    TimoteusR Member

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    I'll give you some basic sound info on your setup.

    First of all. The mixer has a 3 band EQ for mic channels (1-16). It has hig,mid and low bands. The high band is at 12khz and the low band at 80hz. The middle on has two knobs. The top one controls the amoun of boost or cut (from -15db to +15db). The bottom one controls the frequency between 100hz and 8khz. So if you want to cut 4khz turn the frequency knob to the middle and the cut/boost knob to 9'o clock to cut the frequencies. Stereo channels (17-24) have 4 band EQ. They have the same high and low frequencies at 12khz and 80hz. they also have a high-mid band at 3khz and a low-mid band at 500hz. These work just like the high and low eq's on the mic channels. Turn left to cut or to right to boost at the frequency printed on the mixer. Just remerber cutting is always better than boosting. If you need to boost highs you could try lowering the lows and boosting overall volume. In addition to the eq's your mixer has a low-cut filter at 75hz. What it basicly does is cut all frequencies under 75hz by a certain amount. This filter should be enabled on at least vocal mics. If you are using only the sx300 (no subs or other speakers) I would enable it on everything except for drums/bass and maybe piano.

    Your mixer is advertised to have 6 auxes but in reality only 4 can be used at the same time. auxes can be either pre- or post fader. That means the send is either taken before (pre) or after (post) the channel fader. Prefade send are used for monitors. This way adjusting the faders will not affect the monitors (if run from FOH). Postfade sends are used for effects. On your mixer 1-2 are switchable either pre/postfade and auxes 3-4 (and 5-6)
    are permanently postfade.


    Remember amps DON'T HAVE VOLUME CONTROLS. They are called sensitivity controls. They control how much input is needed to drive the amp to maximum output.

    I'll post a guide for setting proper gain structure later.

    --
    Timoteus Ruotsalainen
     
  16. tenor_singer

    tenor_singer Active Member

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    Timoteus,

    Thank you for all of your information. I really wish that my theater background was in technical theater. I truly feel as though I am guessing every time I set up my sound system and light system and many of the explainations here today are right at that threshold of my understanding.

    To be honest, when I get equipment I take a couple thousand dollars (which is the result of two to three years worth of fundraising by my students) and tell my TD friend not to spend more than that and to use his head with purchases. He was the one who bought me the system I described earlier.

    I think next year I will have people from area colleges come in and give inservices to my tech kids. They're great boys and girls, but their knowledge has been hampered by my lack of knowledge.... which pretty much amounts to plug that there and there and don't ask why because I don't know... lol. As a teacher this truly upsets me because I am failing at my job of showing them the true approach to technical theater. However I can block the hell out of a show.

    Again thanks!


    Tenor.
     
  17. TimoteusR

    TimoteusR Member

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    Tell your TD friend that he did a great job ;) If budget is 2K$ then the system you have is pretty good, actually really good considering you have 3 boundry mics + wireless. If I were to upgrade something it would probably be the power amplifier. Used rocks in power amps. you can get more out of those speakers (you can almost double the power going to them) if you want. And when buying remember. Buy what fits your budget and gets the job done period. If you don't need something at that time (like more power) don't buy it.

    I'm writing a guide right now. It's about 2 pages long already. If you like it you can use it as a guide if you mention my name somewhere in the material ;) It's even custom made for you. lol

    Make sure the college kids don't abuse your system. Many people in this industry feel that Behringer sucks. It does if used in wrong places (pro audio) but in your case it's perfectly good.

    If you don't understand something please ask. there are no such things as stupid questions. As a teacher you must have same kind of situations with your students as you're having right now ;) maybe you can learn something.

    And i'm pretty confident the only way to learn this stuff is to do it. Take you students and put the system on, lower the main faders a little so the volume won't be overwhelming if you mess up. Then take the manual and start trying different things. Have one of your students talk to a microphone (they can take turns doing whatever they like. Maybe imitating a sports game or singing or telling jokes. play with the eq's. one of the students might have a minidisc player which you can use to record the practices ;) they might enjoy it.

    And remember if things go wild and you don't know what to do, just shut down the power amp from the front switch.

    --
    Timoteus Ruotsalainen
     
  18. TimoteusR

    TimoteusR Member

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    Ok I have waaaayy too much time ;)

    Here's a small guide for setting up your system. If ControlBooth wants to use this somewhere (maybe a general guide *hint, hint*) you are free to do so as long as i get credit. This goes to all other previous post in this thread.

    If your setup doesn't change you shouldn't have to do this more than once. Make sure the amps are turned off and the feedback distroier is unplugged. Remember how it was plugged in case you want to change back to that setup. If you just want to test the gain structure you can skip the microphone part.


    Let's start with your microphone's. I can't comment on the boundary mics other than test different setups. Try moving them around a little. Or try adding a plexiglass sheet under them and hanging them. If it's possible try and locate the wireless receivers near the stage. If you can position them on either SR or SL (stage right or stage left). Them use line level cabling to go back to your mixer. You can still plug them into the same channels. Just use the "Line in" jack located near the "mic input" (it's the middle jack). Then activate the Low-cut filter on every channel (as your not using any sources with low frequency content this will also clear up some of the stage wash). You could also play with completely turning down the low eq's (at 80hz) if your only using vocals. Remember if it sounds goo it IS good.

    If you have headphones you can connect them to the mixer to monitor the signal there (these are optional). Connect headphones to ?headphones 1? jack in the master section (where connectors are, it should be labeled). Make sure ?mon/ctrl? is on and all others (turn the phones volume off) are off in the phones section. Make sure that in the mon & ctrl room section all buttons are off and the pot is at 12?o clock.


    Now turn all other microphones off except for one. I'll use SL boundary mic. Turn all controls on that channel to 12'o clock position except the aux sends. Turn the Aux sends completely off (fully CCW, counterclockwise). Make sure that "Mute", "1-2" and "3-4" buttons are in their off position (up, not pressed). Make sure "Main mix" is on. Now ask another person to go to SL and talk to the mic. You could also borrow a boom box and use that if you can't find a person willing to talk while you tweak. Press the "Pfl/solo" button on your mixer. Then make sure the "Channel mode" button in the master section (Solo section on the left) is in the up position (pfl) and the level button is pointing at 12'o clock. Turn both the channel fader for your mic and the and the master faders down. Now with your person talking on stage (or boom box) start turning the "Gain" button on your channel strip clockwise. The left meter in the channel section should show signal. Keep turning the "Gain" until the peaks are hitting the lower red area or your channel peak led starts to flicker. This is the most level you can get from the mic pre without clipping. If you have headphones, you can turn the volume up (in the phones section) careful and hear what the mic picks up. If the person were to go closer to the mic and start screaming you would see your peak leds light. If the peak leds start lighting you need to back off on the ?Gain? to compensate for the higher level the mic is sending. Press the channel ?Pfl/solo? button to turn it off. Now turn the master fader to 0db. Put the channel fader somewhere around 0db now your mixer should be outputting the best signal possible. You should be seeing movement in both of the main meters.

    Next you can connect your mixer to the power amp. Make sure the amp is off. Run two XLR (round plug with three prongs) cords to your amps (one left and one right) inputs. They are the upper jacks have the text ?balanced 0dbm? next to them. If the current cables are guitar cable ( 1/4? plug, look at the plug if it has 2 sections it is a TS cable, Tip Ring, and if it has 3 sections it is a TRS, Tip Ring Sleeve cable) you can use them. If you have TS cables you might want to upgrade them at some time to TRS or XLR cable (XLR cable = mic cable). Make sure the switch on the back is set to the middle position (stereo). The bridge position will make amp sum both Left and right power amps into one amp. If you bridge the amp you have to use both of the ?+? sides to feed your speakers (don?t ask why, If you wan?t to read LAB has a pretty good explanation). The parallel setting will feed both amp sections from one signal. This way you only have to run one cable to the amp if you want mono. Set the front panel controls all the way CCW (counterclockwise, the amp will be least sensitive in this setting, It will need more input signal to achive the full power at output). Turn the mixer?s main faders off. Turn the power amp on. Slowly turn up the master faders to 0db. Now you should hear the person on stage speaking into the microphone from your heaphones and from the speakers. Now turn the amp's sensitivity control slowly clockwise until you reach the desired volume. Make sure the CLIP light on the amp never lights if i does light that means it's time to back off on the main faders at your mixer

    Now you should have one mic working as I should be. Just repeat the mic procedure for all your mics. Here it is once again in a shorter version:

    1. Choose a channel
    2. Turn all controls to 12?o clock (auxes off)
    3. PFL the channel
    4. Adjust gain until the yellow leds are lighting on peaks (maybe one of the red leds sometime)
    5. release PFL
    6. raise channel fader.

    --
    Timoteus Ruotsalainen
    [email protected]
    Finland
     
  19. TimoteusR

    TimoteusR Member

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    Here are some links to basic stuff on audio:

    http://www.churchsoundcheck.com/mag820.html
    http://members.cox.net/pasystem1/noframe.htm
    http://www.prosoundweb.com/studyhall/ab_index.php
    http://www.prosoundweb.com/studyhall/sr_index.php
    http://www.soundcraft.com/palz.asp
    http://www.prosoundweb.com/studyhall/ci_index.php



    http://www.prosoundweb.com/live/articles/daverat/amptricks.shtml

    you can try the above amp trick if you want. Dave Rat's test amp showed about a 14% rise in power with this. I have confirmed with Samson that their amp uses one power supply for both channels so this should work fine. If you want to try it, you need a "speakon polarity reverse" cable and a "XLR or TRS polarity reverse" cable. If you show the article to your TD friend he'll know what you mean.

    --
    Timoteus Ruotsalainen
     
  20. Nephilim

    Nephilim Active Member

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    Modern amps are supposed to do that by default, at least switchmode amps do. The reason you get "zero point crossing" distortion with those (except class I from Crown) is because the switching between the + and - power reservoirs can cause tiny glitches in the signal. I don't recall is AB class amps split like that also.
     

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