Impious

Sen. Bunning blocks unemployment benefits extension - Yahoo! News

Tell him it's a stupid idea and a stupid way to go about doing it. Why does he think it's going to be "better" to have the subs in a poorly designed and inadequate ported enclosure?

Not an ideal ported driver. If you want to go ported, build a new enclosure of appropriate size. Don't port the one you have.....it's too small. Here's the owner's manual: http://www.rockfordf...ated=1083259380

I believe the bill that did not pass was an "emergency" bill. It would have only extended benefits another month until a "long term" bill is passed. Here is an exert from another article; So while some people may end up screwed in the short term, it does not appear this is a permanent change to unemployment benefits.

Any idea the actual model number for your subwoofers? The two owners manuals I could find on RF's website for the older model Punch Z's like yours showed the specs varied some between the two models, but neither were particularly well suited for ported enclosures. Also, typically not a good idea to port a sealed enclosure as ported enclosures are typically required to be slightly larger. Just off a glance, after you subtract out the volume of the port your current enclosure would be too small to function ideally as a ported enclosure for the subwoofers. So overall.....probably not a good plan to do what you wish to do.

Three 10's have slightly less cone area than two 12's. Three 10's sealed seem pointless if you can fit two 12's sealed. Two 12's would be less expensive, take up less room and have equal output capabilities. As for the 10's port verse 12's sealed.....that's a personal decision that I can't much help you with.

The JL Audio link in my prior post explains the proper way to calculate port length when using multiple ports. I don't have time right now to mess around with that calculator to be able to tell you how to interpret the information it's giving you.

The port end correction factor increases the effective length of the port. Therefore, for a given tuning frequency you would need to decrease the physical length of the port by the end correction factor so that the effective length was correct for the tuning frequency.

Or try a different enclosure

Keep in mind a clamp test isn't a very useful or accurate method for testing amplifier power. To the average user (such as yourself), the numbers stated in that thread are pretty useless. That was my point Increasing power is most useful to competitors where tenths of a decibel can be important. To the average user (such as yourself) even doubling power output won't result in a significant audible benefit, especially when considering the added expenses. But people lose sight of this fact because of all the attention raw power figures are given on the internet. Also was mentioning it to point out that if you have a 1500w amp now and you want to significantly increase audible output don't run out and buy a 2kw or 2500w amp as it won't make a much if any audible difference and all you'll have accomplished is spending money for pretty close to nothing.

As bromo said, the issue is with the difference between physical port length and effective port length. On the carstereo.com calculator when you mark slot port "yes" the calculator reduces the length of the port by 1/2 of the port width to account for port end correction. If you notice the difference between the two calculators is 12", which is half of 24", which you apparently entered as the port width on the carstereo.com site. The 12volt site does not account for port end correction. For more on port end correction factor, see here: JL Audio - Car Audio Systems Without knowing details on the layout of the port in the enclosure, I can't tell you which is accurate. If you are using an enclosure wall as the sidewall of the port, you will need to account for end correction. It would seem odd to have a port that's only 4" tall & 24" wide unless it's horizontal across the bottom of the enclosure. If the port is vertical in the enclosure I would assume the 4" is the width and 24" is the height? Either way, you would only need to subtract 2" from the port length for end correction factor as opposed to the 12" initially calculated (i.e. on the carstereo.com site, put 4" for the width and 24" for the height and mark slot port "yes" if you are using the enclosure wall as a a sidewall for the port).

Keep in mind that even under ideal conditions you would have to double your power to gain 3db in output, which is an audible but not dramatic increase in sound. After you factor in power compression & other factors, even doubling power will yield less than a 3db increase.

This definition isn't very accurate either. For "phase cancellation" (destructive interference) to occur between two or more speakers the speakers do not have to be out of phase. For destructive interference to occur, the soundwaves simply need to interact with each other while the soundwaves are not in phase. The speakers may in fact be in phase while this is occurring. This phenomenon is very prevalent in car audio where we sit at different distances from the two speakers or in home theater where the listening positions can vary. There are other ways for destructive interference to occur. In home audio speaker design the center-to-center distance of drivers in speaker may cause destructive interference. The crossover can also cause two drivers in a given speaker to be out of phase since the crossover will shift the phase of the signal (90* phase shift for every 6db/oct increase in slope), in which case you must wire the drivers out of phase in order for the soundwaves to be in phase at the crossover point. You can also have destructive interference with a single speaker due to reflections. The reflected sound may reach the listener out of phase with the original sound. Not nitpicking your definitions.....but also do not want people misunderstanding the information being provided.

The sentences highlighted in red are rubbish and should be excluded from any viable definition of DF. For more information on damping factor see here: Damping Factor

Ehh...not really. I would maybe define beaming as a term used to describe the narrowing off-axis disperson of a loudspeaker as frequency increases due to the relationship between the wavelength of the sound wave and cone diameter. When you say "maybe" I question your definition. How does Cone Diameter directly affect beaming? I used "maybe" because I hadn't given more than a moments thought to the wording of my definition, thus it was subject to revision if I came up with something that was worded either more cleverly or more comprehensively.

Does your headunit have any adjustments? EQ, crossovers (both lowpass and subsonic/infrasonic/highpass), level settings, etc?

Really nothing unusual there. There are a few possible explanations. First, your ear is more sensitive to sound at ~65hz than it is at ~35hz. A sound or tone at ~65hz may be perceived as louder to your ear than a sound or tone at ~35hz even though the actual SPL (as would be measured by a microphone) of the ~65hz tone is lower than the ~35hz tone. SPL competitors are well accustomed to this, and is the reason the experienced competitors will tell you that what is louder to the ear won't necessarily be louder on the mic. This effect has been studied and is known as the Equal Loudness Contour. As you can see from the graph, a 35hz tone requires approximately 10db higher SPL to sound equally loud as a 65hz tone to the ear. Second, just because the enclosure is tuned to 35hz doesn't necessarily indicate there will be a peak in the output at that frequency. Depending on the parameters of the sub and how it responds in the enclosure you have it in, it's possible the output is actually lower at 35hz than it is at 65hz. It's also possible that the output from the subwoofer is pretty equivalent at 35hz and 65hz and you are simply experiencing effects of #1 above and #3 below. Third, I'm not sure where you have the enclosure located or the specifics of the room, but you may be experiencing the effects of some room nodes which will affect the frequency response from various listening positions. While rooms in a house don't have the "cabin gain" that we experience in car audio, there will be locations in the room where the sound waves have constructive and destructive interference which will affect the frequency response. Well, yes, if you lowpassed the subwoofer and that lowpass filter attenuated 64hz you could of course make it sound less loud than 35hz. However, that's not the mechanism(s) causing what you are experiencing.

Keep in mind the gain is not a "power knob".....having it turned up less than half way does not mean you are not receiving full power output. It is possible to obtain full power output from an amplifier with the gain turned to it's lowest setting, depending on the input voltage from the source. That said, there are different theories on how to properly set the gain with a DMM. One theory is to use a 0db test tone so that the amplifier doesn't clip the signal. However, this decreases the amount of power output during typical music playback since music is at a level below 0db way more often than it is at 0db. So another theory is to use attenuated test tones; -3db, -6db, -10db. This gives you more power output while playing music at the risk of clipping the peaks of the signal. But before we really get into resetting the gain, it would be more useful to know what enclosure you were using, what your other settings were set at (lowpass filter, highpass or infrasonic filter, bass boost, Q, etc), and what exactly you think is lacking as well as your expectations. We may end up chasing our tail around proper gain setting when the problem is actually somewhere else. When you use a bass boost or EQ where you increase or decrease the level of the signal at a particular frequency, you are not only adjusting the frequency marked on the dial. You will also be increasing or decreasing the level of the frequencies surrounding (both above and below) the center frequency. For example, if you set the EQ center frequency to 40hz, and you turn the db dial to +3db, you are not only increasing 40hz by 3db....you are also increasing the output level of frequencies above and below 40hz (by an amount less than 3db, however). The Q setting changes the "bandwidth", or how wide of a frequency range above and below 40hz you are affecting. The lower the Q, the wider the range of frequencies. The higher the Q, the narrower the range of frequencies. Unfortunately without quite a bit of information it's hard to determine a "proper" setting of the Q....so generally it's best to set it to where ever obtains the best results by ear.

What enclosure are his current subwoofers in? Something as simple as a new enclosure for his current subs may give him what he's looking for at the cost of about $30 in supplies and an afternoon building it. Type R's can wang chung in the right enclosure, if that's what he's looking for.

I actually chuckled when I saw the pictures. I've seen a lot of things on the forums over the past 8 years....can't say I've ever seen anyone use chimney and sewer pipes as a makeshift port before. No harm, no foul when it comes to testing though.....hopefully you'll learn a little something in the process.

The Epicenter works by looking for harmonics and filling in what might be a missing or attenuated fundamental that was filtered out during mastering process or somewhere in the reproduction chain (such as the source unit). If, for example, it finds a signal at 100hz (1st harmonic), 200hz (2nd harmonic) and 400hz (3rd harmonic) but a weak signal at 50hz (the fundamental), it will "restore" or "recreate" the 50hz signal. So it's only really useful if the fundamental is truly "missing". If the music you're listening to simply has limited bass content, then you are using the Epicenter outside of it's intended purpose. It's not supposed to be used to add bass to songs which were composed with limited bass content, only songs that had the bass unnaturally filtered out at some later point. In general with most of today's music and sources, an Epicenter is pretty useless and does more damage than good. Most people still using them are intentionally (although maybe unknowingly) using them incorrectly. Shameless plug: Damping Factor What exactly is it you want to know?

LOL....always nice to see manufacturers combating ignorance, even if it's just a brief footnote! Glad you guys find this thread useful

What is sensitivity? The sensitivity of a driver is typically defined as the sound pressure level of the loudspeaker given 1 watt of power input, measured at a distance of 1 meter, measured on-axis with the loudspeaker and measured in infinite space with the loudspeaker mounted to an infinite baffle. While the later part is typically assumed but not stated, the former part is the reason you generally see “1w/1m” or “2.83V/1m” stated along with the sensitivity specification. But wait, is 1w/1m equal to 2.83V/1m? The answer is only for an 8ohm impedance driver. We know this because of ohms law, where Power = Voltage^2/Resistance. If the load is anything other than 8ohm then the resulting sensitivity rating will not be a 1w measurement! If the driver is 4ohm and the measurement is a 2.83V measurement, we can find the amount of power input with the formula 2.83^2/4 = 2w. This means that sensitivity will be overstated by 3db compared to a true 1w measurement. I have seen many unscrupulous companies take a dual 2ohm/coil subwoofer, wire the coils in parallel and then rate sensitivity at 2.83V, overstating sensitivity by 9db! Due to these and other variations in rating methods and some manufacturers not even stating a method, it can often be difficult to directly compare one subwoofer’s sensitivity to another’s. However, if you are able to obtain the Thiele-Small parameters for the subwoofer there is one surefire way to calculate an accurate sensitivity measurement for a driver. And that is with the following formulas; Efficiency (N0) = 9.64 * 10^(-10) * Fs^3 * Vas / Qes *To express as a percent, multiply by 100 *Vas in liters Sensitivity (SPL) = 112 + 10*Log(N0) The Efficiency (N0) calculation expresses, typically as a percentage, the amount of power input that is converted to acoustic power. With the sensitivity calculation we can convert the efficiency of the driver to a 1w/1m sensitivity rating. As long as the T/S parameters are accurate, utilizing this formula will put any two drivers on a level playing field and allow you to directly and accurately compare sensitivity (or efficiency) between drivers. Okay, but higher is still better because higher means it’s louder, right? Not necessarily the case. There are a few issues here that need to be discussed to better understand why. From a driver design perspective, we can see from the above formulas that there are two ways in which to increase sensitivity for a given driver diameter; increasing Fs or decreasing Qes (or both). Adjusting either of these two parameters, however, is ultimately going to affect how the driver responds in a given enclosure. And this relationship, as it relates to enclosure size, sensitivity and low frequency extension, has been defined through a rule known as Hoffman’s Iron Law. As you can see from the link (which I highly suggest you read), it’s already been covered in relative detail elsewhere on the site. But it’s important to us here, so it’s worth touching on again. In short, Hoffmans Iron Law states that we can only have two of the following three; Low frequency extensionSmall enclosureHigh sensitivity What’s important to take away from this is that if you want high sensitivity and a small enclosure, you will necessarily sacrifice low frequency extension. Or, conversely, if you desire extended low frequency output from a small enclosure, you must necessarily sacrifice sensitivity. While a given driver may have a higher rated sensitivity, that driver may actually have less output in the subbass region once the effects of the enclosure are taken into consideration since it may begin it’s rolloff at a higher frequency. And given in car audio we typically require reasonable enclosure volumes, having a high sensitivity can actually be disadvantageous. The second factor we need to consider is how output is achieved. Sensitivity isn’t what determines maximum output. Output is a function of air displacement. The maximum amount of linear displacement (Vd) you can achieve is limited by your cone area (Sd) and linear excursion (Xmax). As you increase the amount of displacement at your disposal, you increase the potential linear output you can achieve. Even though a driver may have a higher sensitivity, if it’s maximum linear displacement (Vd) is less than that of a driver with a lower sensitivity, the lower sensitivity driver will have the potential for higher levels of output. We also have to consider the effects of Power Compression (and I again urge you to read the link). Consider the effects of power compression applied to our scenario. A higher sensitivity driver may, for example, begin to experience more power compression at a lower output level than a lower sensitivity driver. If this is the case, then as output is increased beyond that level the higher sensitivity driver would gain less output for each increase in power and experience more parameter shift. The basic summary is that after many factors are taken into consideration, it is impossible to state unilaterally that higher sensitivity drivers will better than lower sensitivity drivers. Generally choosing one driver over another based on sensitivity alone is a bad path to follow. Sensitivity is just one of a multitude of parameters that all conspire together to define the performance of a loudspeaker. And in my humble opinion, it is one of the less important factors to consider. As with anything, there are compromises to be made and it's necessary to find the best set of compromises for your particular situation.

http://en.wikipedia.org/wiki/Thiele_small

Ehh...not really. I would maybe define beaming as a term used to describe the narrowing off-axis disperson of a loudspeaker as frequency increases due to the relationship between the wavelength of the sound wave and cone diameter.