flakko

if you dont have access to an o-scope or any fancy equipment like scott said, try turning the gains down a little bit until the symptoms go away. it may not be as loud but you are saving both your sub and your amp from damage. if you have a digital multimeter (its only liek 20 bucks if you dont) lookie here: http://www.soundsolutionsaudio.com/forum/i...?showtopic=3704 follow that

oh hoh! nice. kowinkidink i was looking for some RCA's

about level matching: "Okay, now that you’ve had some time to absorb all the previous information, I’m going to step it up a little bit. Let’s first look at the challenges we face in building any kind of speaker system. First is level matching. It’s very difficult to find a tweeter and a woofer that have the exact same sensitivities, and will operate at the same volume given the same power. Second is impedance variations. As you move through a speakers frequency response the impedance varies. As you get towards the upper end of response the impedance starts to rise considerably. This is demonstrated in the following graph. This a factor of inductance of the voice coil. As you can see, the impedance is rising the higher you go. This causes a drop in output. A Zobel will level that out, giving you a flatter response. Third is spikes and dips that cause your response to be something other than flat. Let’s start in order here. Level matching is done using an L-Pad. Generally, the tweeter is much more sensitive than the woofer. So the tweeter needs to be attenuated. The L-Pad circuit can be introduced into your passive crossover. It can be done 2 ways. First is fixed. If you know the amount of attenuation, you can get the appropriate resistors and build the network right into the crossover. The problem with this is if you don’t get it right, you need to completely redo it. The second way is to use a variable L-Pad. Parts Express has these and in a car audio application using passive crossovers they are a MUST in my opinion. http://www.partsexpress.com/webpage....Group_ID=1 96 The drawback is you need to build around the L-Pad and mount it somewhere that is accessible. However, it does allow you to adjust your tweeter to compensate for different mounting locations. These are what upstage kits, like CDT use. Next is the Zobel network. The Zobel network compensates for the rise in impedance due to voice coil inductance. Pretty much enough said. Third, and this is where crossovers start to get REALLY tricky, and even I’m a little fuzzy in this area. Depending on your baffle, mounting locations, varying distances of voice coils, your response will be something other than flat. If your baffle is too narrow, (as in a tower speaker) your mid drivers can experience a spike in response of around 6db starting at around 100 hz and flattening off at around 1k hz. This is called baffle step. You use baffle step compensation to, well, “compensate” for it. I’m not going to go into it a whole lot, and baffle step is just one example of how a speaker reacts to it’s environment. I will say that there is a compensation circuit that will cover just about every peak, and dip. What does this all mean? It means that you can drive yourself CRAZY trying to build the perfect crossover for any application. There’s a point of diminishing returns when a crossover just becomes too over-engineered. It’s safe to say that a decently built crossover will contain 3 parts, the actual crossover network, the L-Pad, and the Zobel. The other networks don’t really come into play until you NEED them. Baffle step can be fairly accurately predicted, but other “notch” type filters are usually used to compensate for something unpredicted and unwanted. It’s possible to “notch” out a peak caused by a Butterworth type crossover at the crossover point, but it’s just better, if it’s a problem to pick a different alignment. LEAP is great, as are a lot of other programs out there. LEAP allows you to build your crossover ‘to the enclosure’ for the best possible results. LEAP can only be effective when the surrounding variables of a driver are controlled. For example, the difference between a well built speaker enclosure, and a door frame. LEAP can predict and design a crossover to compensate for the way the speaker will react within a given enclosure by simply inputting the dimensions. LEAP is going to “leap” out of your computer and kick the snot out of you if you try to enter door frame parameters. So, if you are looking to go the passive crossover route, you can pretty easily design a crossover as long as you know which speakers you’re going to be using. You add your three essential items and voila you have a working crossover. However, based on the above information that’s all you’ll get. You won’t get any kind of compensation for the way the speakers will react in the car. And believe me, they will react BADLY. So, what do we do to compensate for these unknown and highly erratic, unpredictable peaks and dips in our frequency response? We EQ them out. An equalizer will do everything that a notch filter will do, and is effectively an “Active Notch Filter”. We couple the EQ with an Active Crossover, and you get a system that is highly adjustable and can correct for the many, less than desirable, effects your car will have on your system. Add Time Alignment and you can achieve results comparable to a cheap home audio setup (given that you spend $1000.00 on high quality speakers). By going active, we’re essentially, trying to take the “car” out of the “car audio”. Active processing is also used in home audio and is highly desired by many audiophiles out there. However, in my opinion, the difference between a well built tower with a passive crossover and an active setup in home audio is not nearly as dramatic as an active to passive setup in car audio." whew. iz done. other info can be seen over in the link given

Frequencies in music (good for eqing): "Understanding Frequency - What does What A lot of you know that I am a bit of an sq fiend, I am one of the few people on here that often turns of the sub just to check it's still running...I personally like sub bass as an anchor for the low end and to add warmth to the music. To me, mid and midbass is the key. Mid and Midbass IS critical. let's look at the frequency response of some instruments and see where all the action is. I am not going to go into the the differences of Fundemental and Harmonic frequencies, and how they interact (I can if need be), this is more of an overview of what frequencies make up what. For those of us who listen to acoustic music, apposed to synthesised dance music, for want of a better description, I shall take some drums, bass guitar, electric guitar, and vocals. I shall work through the frequencies rather than the instruements, as this will allow us to see where there are complimentary frequencies (different instruments produce the same sound). 50hz (usually sub bass) this freq is where all the boom is, if you want more boom on foot drums and bass guitar, boost, to reduce, cut. 100hz( usually mid bass) this is the hard bass sound, it gives drums that solid feel, boosting here will harden the drums/bass guitar, as well as adding warmth to guitars. A cut will reduce boom on guitar and add clarity. 200hz (either midbass/mid) Boost to add warmth to vocals and guitar, reduce to clean up vocals 400hz (usually mid / large Horn) Boost to bass in general, reduce to decrease cardboard sound low drums. 800hz(usually mid/horns) Boost to add clarity and Punch to bass, this is the one that digs you in the ribs , cut to reduce tinnyness to guitars 1.5khz (mid/tweet/horns) Boost to add clarity to bass guitar, reduce to impreve dullness of guitar 3khz (mid/tweet/horns) Boost to increase pluck on bass guitar, attack on guitar and high drums, increases clarity of vocals. Cut to reduce breathy sound on vocals. 5khz(mid/tweet/horns) boost for vocal presence, low drum attack, piano attack, and guitars, reduce to distance background. 7khz(usually tweet/horn) boost, more attack on low drums, percussion and bring life to dull vocals, also sharpen elctric guitar Cut to reduce siblance 10khz (tweet/horn) increase to brighten vocals/guitar and piano cut to reduce siblance 15khz (tweet/horn) increase to brighten vocals/guitar and piano highs Right, looking down this list we can see that if we want a good solid bass line ( add @100hz), that's not boomy (cut @ 50hz), with good punch (add @ 800hz), with good attack ( boost at 5-7khz), most of the action is in the midbass and midrange area, with only boominess being in the sub area. This also highlights one of the main benefits of horns( the huge range they cover). Understanding these frequencies also allows for fine tuning things like stage height (more attack of drums gives perception of a higher stage), and adding depth, ( make background sounds more distant)"

Figuring out which frequency to crossover drivers: "I'm going to use a couple of examples, and some response graphs to help you understand this. I'll start with the Peerless Exclusive 7". Scroll to the bottom of the PDF where the response chart is. http://www.madisound.com/pdf/peerless/830883.pdf You'll notice the "blue", "red" and "green" response graphs. Labeled at the bottom you'll see "On Axis", "30 Degrees", "60 Degrees" respectively. Looking at the graph you can see that the upper end response of the driver lowers dramatically the further off axis you play them at. Now if your driver side door sits 60 degrees off-axis of your listening position (which most doors fall in that area) then you can get a good idea of what the upper end response will be. In this case the graph shows about 1750 hz before it starts to collapse, and is probably useable up to about 2200 hz. Your passenger side driver will yield a considerably higher response due to it be much closer to on-axis than the driver side, so you might start to hear some bias from the passenger side should you try to run the set up to 3500 hz. Now let's look at the Vifa MG 4" midrange http://www.madisound.com/pdf/vifa/mg10md09-04e.pdf Here you'll notice that on-axis response is great. Near 15k flat, with extension up to 20k. Npdang tested this driver and mentioned that it can almost be used without a tweeter. By the response graph we can see that. However, now let's say we're building some kick pods, and due to some reason, we can't get them completely on-axis, but rather 30 degrees off-axis is the best we can do. You'll see that the 30 degree off-axis response graph basically tells us we can use these midranges up to about 5k before any real degradation of response. Pretty nice. Now, for low end response. For tweeters, the general rule of thumb is twice the Fs (Resonant Frequency) at 12 db. A higher slope (i.e. 18 or 24 db) can get you closer to the Fs, but we'll use 12 db for now. Let's first look at the most common tweeter on this board... The LPG http://www.madisound.com/pdf/lpg.pdf The Fs of this tweeter is 1850 hz. Doubled that is 3700 hz. Now, try coupling that with a 7" driver mounted in a door 60 degrees off-axis, and you have quite a gap between 2000-3700 hz. Almost a full octave. Now to be fair, let's look at the upper end extension. The on-axis, 30 and 60 deg graphs look almost flat up to 20k. Very nice. These can be mounted in some sail panels firing horizontally across your front stage and you can get great results from them. Next is the Seas Neo tweeter. http://www.madisound.com/pdf/seas/h1396.pdf Here is a tweeter with a much lower Fs. 1170 hz, using our rule, can be crossed at 2340 hz. The specs say 2500, so we're pretty close. With a 24 db slope you could get 2200 hz out of them. These would be much better to mate with a set of Exclusives mounted 60 degrees off-axis. However, their top end is nowhere near that of the LPGs. You can definitely see that these would lack the top end "sparkle" that so many people refer to when talking about the LPG's. These will be much more neutral and laid back on the top end. Not a bad thing, as a lot of music doesn't go any higher than 15k. You will also notice a huge difference between on and off-axis. A major consideration when considering how to mount them. These are just some suggestions on how to "guess" at appropriate crossover points. The graphs give you a good idea of how to tell how a driver will perform in a given installation. These graphs in no way indicate how a driver will sound, nor how they will perform at the upper and lower limits of their capabilities."

"The way two drivers combine at the crossover frequency depends on the magnitude and phase at the crossver frequency, where both drivers are playing the same frequency. Strictly speaking, of course, it's the acoustic phase that matters. But let's say that the two drivers are equidistant to your ears, so we'll just focus on electrical phase for now. If two drivers have the same amplitude, and are 0 degrees out-of-phase, the combination will be 6dB hotter in amplitude. If two drivers have the same amplitude, and are 90 degrees out-of-phase, the combination will be 3dB hotter in amplitude. If two drivers have the same amplitude, and are 180 degrees out of phase, the combination will perfectly cancel. And now for a couple popular ones: First order, 6dB per octave. Each driver is down 3dB at the xover frequency, and each driver experiences a 45 degree phase shift at the crossover (but in opposite directions). So the relative phase between the drivers is 90 degrees at the xover frequency. This means the drivers will have a combined response that's 3dB hotter ... and each was down 3dB at xover, meaning a combined result that's perfectly flat. This is a good thing ... but the shallow roll-off/up renders this solution almost useless (or at least not real popular anymore). The 12dB Butterworth Alignment provides 12dB roll off/up for each driver, and each driver amplitude is down 3dB at the crossover frequency. Each filter "order" also provides 45 degrees of phase shift at the crossover frequency ... so the woofer will experience a negative 90 degree phase shift (second order), while the tweeter will experience a positive 90 degree phase shift (second order). This means that the drivers are naturally 180 degrees out-of-phase at the crossover frequency. This is kind of a bad thing, because it means a deep cancellation null at the crossover (again, we're assuming same distance from each driver to your ear). To solve it, many people wire the tweeter out-of-phase at the crossover. This puts the drivers back to a 0 degree phase relationship (inversion provides exactly 180 degree phase shift), which will result in a +3dB amplitude bump (since each one was down 3dB). Can be addressed by underlapping xover freqs a bit. The 24dB Linkwitz-Riley was discovered by the challenge to find a crossver where each driver would be down 6dB at the crossover ... instead of the typical 3dB ... and each driver would experience a 180 degree phase shift (in opposite directions, of course) ... meaning a "perfect", 6dB summation at the xover. It was discovered that the cascade of 2 second-order butterworths satisfy all requirements, while also providing a relatively steep roll-off/up." "Any linear filter is completely characterized by two things : it's MAGNITUDE response versus frequency, and it's PHASE response versus frequency. Together, these two functions define the COMPLETE FREQUENCY RESPONSE fo the filter. Oftentimes, we are more interested in the first derivative of the phase response, giving rise to a new function of frequency called the GROUP DELAY ... which will also, of course, be a function of frequency. A "Butterworth" response is also known as Maximally Flat Magnitude. Mathematically, it is determined by setting as many derivatives as possible ... of the filter's MAGNITUDE function ... equal to zero at DC (for a low-pass filter). The resulting low-pass shape will have a MAGNITUDE response that is as FLAT as possible ... until it kinda starts to rolloff, of course Applicable to low-pass, high-pass and band-pass of course. This filter has a phase response that, while not as good as our next candidate, still isn't "bad". A "Bessel" response is also known as Maximally FLat Group Delay. Mathematically, it is determined by setting as many derivatives as possible ... of the filter's GROUP DELAY function ... equal to zero at DC (for a low-pass filter). The resulting low-pass shape will have a GROUP DELAY response that is as FLAT as possible. Applicable to low-pass, high-pass and band-pass of course. The magnitude response will still look low-pass (or whatever you want), but will not roll-off as quickly or sharply as the Butterworth. The transient time domain behavior of any filter ... including both of these, and any other linear system ... is completely determined by the COMPLETE frequency response (magnitude and phase, or group delay). The Bessel is generally considered to be a bit "better" ... less overshoot, less ringing for higher orders ... because it's phase is better behaved. But the Butterworth isn't bad, and given it's sharper roll-off, it's generally preferred for most crossover designs. Excepting, of course, the Linkwitz-Riley. From a simple filter perspective, it's simply a cascade of two Butterworth responses. It won't be as flat as a true Butterworth of equal order, but it's strength is the way the two drivers combine at crossover, as discussed above."

"A crossover consists of two or more filters designed to split the desired frequency range into parts. Each part then being assigned a dedicated drive unit (speaker). A 2-way crossover has two filters: a high pass and a low pass, and is usually used to integrate a mid-bass and a tweeter. Each filter then has several characteristics, one of which is it's slope (or roll-off), measured in decibels per octave (dB/oct). An octave being a factor of 2 of the original frequency. Thus 1kHz has octaves at 500Hz and 2kHz. The rate of roll-off is always a factor of 6 (for ease of circuit design and calculation). The most popular types being 6dB/oct, 12dB/oct, and 24dB/oct. Further to the numerical rate of roll-off, filters are often asigned an 'order' number. Order numbers are again a factor of 6, thus a 1st order filter equates to 6dB/oct (1*6), 2nd order 12dB/oct (2*6), and 4th order 24dB/oct (4*6). The other important filter characteristic is the filter point (crossover point where necessary). The filter point is that chosen at the design stage and equates to the 'half-power point', or the frequency where output is -3dB (-3dB equating to a 50% reduction in power). *** This assumes the filter design uses -3dB, some designs are -6dB at the filter point ***. For example: a 1kHz high pass filter. In it's most basic sense everything above 1kHz passes, everything below is discarded. That's not particularly accurate. The frequencies below 1kHz are not discraded, rather they are attenuated according to the rate of roll-off of the filter. Assuming 6dB/oct: The output from the filter is -3dB at 1kHz, an octave later (500Hz) the output is -9dB (ie the output is already -3dB, and the filter reduces that by a further -6dB). Another octave lower (250Hz) and the output will be -15dB. If the filter were 2nd order, 12dB/oct, the output would be -15dB within the space of an octave, ie at 500Hz. -15dB equates to over 95% reduction in power. This should indicate the importance of high order filters, especially within high power systems, and especially with tweeters. With a 6dB/oct filter, low mid-bass frequencies will still be entering the tweeter. And even though the output is considerably reduced, if enough power is available, the percentage reduction may not be enough to stop tweeter damage. Just for reference: -3dB = 50% reduction, or a factor of 2 -6dB = 75% reduction, or a factor of 4 -12dB = 93.75% reduction, or a factor 16 -18dB = 98.4375% reduction, or a factor 64 -24dB = 99.609375% reduction, or a factor 256"

are they made by fi?? or is it just the same cone/logo design?

if its anything liek the viper (Directed), i would honestly ditch the orion...

research by forums, and online "papers". also talking to some people, and a little trial and error to top it off. if you like DIY stuff, go to www.diymobileaudio.com

heard a couple of MX's and i heard no mechanical noise. maybe cuz of the invertendess.

Airspace is airspace. dimensions dont matter. as long as it fits!

Well on my ephatch.com forum someone said this.. and he knows his stuff Yellow-top for deep cycling and car-off use Red-top- for big instantaneous current and fast, turning over large high compression engines, etc. Don't like to be deep cycled. It's a toss-up for car audio, if you like to sit and jam with the car off then the answer is un-disputably the yellow-top. For daily use I recommend the red-top due to it's uber low internal resistance, HUGE peak current capacity, etc. so i was thinking the red top would do fine for what i need. dont know much about batteries, but from the reading ive done, the Red top is a good starter battery, and the yellow is a good backup battery.

yea. twas started by me **embarrassed** i was regurgitating what i was told. no hard feelings right? right?

great feeling to actually hear the difference between amps no? welcome

looks like you did

the xxx thermal handling (on the site) is 2kw.

that power looks tempting...

You should look into a deep cycle battery instead of a starting battery. A lot of users here swear by Kinetik batteries. http://www.kinetikaudio.com/hc1400.asp You also need to complete the Big 3 upgrade http://www.soundsolutionsaudio.com/forum/i...p?showtopic=200 I would run at least 1/0awg to a distro in the rear of your car and then split it to 4awg to each amp. I cant shell out that much for a battery. As for the big 3, I honestly dont think I could do it correctly lol. w/o the right electrical system, you wont use your amps to thier full potential. I upgraded to an odyssey batt, and my voltage became much more stable. im still downsizing my system tho. 65a alt sux

looked it up. i believe its 2.5"

planning a sealed box for a 18" rl-p, and i want this thing to be solid. first things first, im gonna resin it. should i do the entire inside, or just the edges? i was thinking mainly the edges to give it a nice tite seal. can i use resin instead of caulk? or should i caulk first, then resin? 2nd, instead of using a 2x baffle, i was thinking about bracing it. i saw this bracing thing from bcae: http://www.bcae1.com/spboxnew2.htm and was going to stick 2 of those in the box. (dims are 42x12.5x18.75) also maybe add some polyfill. the box + the bracing and displacement is going to be around 4.18 cubic ft w/o the fill thanks in advance

booh that sux.

**Rubs ears** WOOOOOOOOSAAAAAAAH calm down buddy, its all fun and games

yea...why did you let your system reach 8v? its not ma audio that sucks, its the install that had the problem lol BACK DOWN ON THE GAINS!

i like your balls. not your pecker