Forums › Rave › Free Parties & Teknivals › A Technical Foundation To Building A Sound System: Part 3
The Sunsonic Criteria:
There are presently 8 Sunsonic Criteria. Some may include more than one element or requirement. Several of these elements and requirements are covered in more detail in later sections.
I. Speaker Cabinet Selection Should Emphasize Bass and Subbass Reproduction.
Dance music has a very important, yet subtle property that its appreciation and effectiveness are not improved by utilizing high midrange or treble volume levels. In fact, the exact opposite effect often occurs. Overly loud and harsh midrange and treble sound pressure levels cause a temporary reduction in perceived dynamic range in the listener, induce fatigue and stress, and over time cause hearing loss.
Of equal importance is the subtle property that subbass does not exhibit these properties to nearly the same degree. This is because the energy content of a sound is directly proportional to its frequency and amplitude. Subbass frequencies have large displacement amplitudes, but their frequencies are very low and their wavelengths are very long. As a result they couple more efficiently to the body itself than to the small inner dimensions of the ear, which is why you can feel the lowest subbass frequencies as much or more than you hear them.
The first criteria is thus as follows:
A. Employ subbass cabinets in a sufficient ratio to full-range cabinets such that the subbass power rating is at least as high as that of the full-range cabinets.
We highly recommend Cerwin-Vega T-36 and SL-36B bass bins, as these are inexpensive, clean, very reliable, and are compact and loud. Whatever you do though, use only horn-loaded or Helmholtz resonator-loaded subbass cabinet designs. If you have more money to spend, and require more portability or better space efficiency, look into NEXO or Funktion-One subbass cabinets. Avoid front-loaded bass cabinets, as they have much lower efficiency and as a result just don’t put out very much bass.
As an example, if you have 2 double 15″ full-range cabinets that are each rated at 1,500 Watts program power, you should have as many bass bins as are necessary to put out 1 to 1.5 times that amount of power. In this case 4 Cerwin-Vega T-36s would be an ideal match. A more general approach would be to figure you’ll need one single-18″ horn-loaded subbass cabinet per 15″ or 18″ driver in your full-range cabinets. That works out to 2 T-36 style cabinets per double 15″ full-range cabinet.
Note that the optimal ratio of subs to full-range cabinets does vary depending on the venue. Due to bass loading issues, outdoor venues require more bass bins to provide the same subbass SPL as in an indoor venue. The larger the air volume of a space, the more bass bins you need to produce the same subbass SPL. In addition, in an indoor venue, walls and corners can be used to provide 1/4 or 1/8th space bass loading, which results in far greater subbass efficiency and output level. Outdoor venues do not usually have walls, in which case the bass bins will only be half-space loaded. It takes 4 half-space loaded bass bins to produce the same output SPL as 1 eighth-space (corner) loaded bass bin. This is a subtle but very important property you must keep in mind when deciding where to put speakers. When doing sound in an indoor venue, if the speakers can be backed up into corners, you will get the most bass output. The second best option is to back them up against a wall. If that is not practical either, for instance if they have to be in front of a stage or something and more than 10 or 20 feet from any walls, then you are definitely going to need more bass bins than full-range cabinets. If you are fortunate enough to do sound in a small venue shaped like a simple square or rectangle, with nothing between the dancefloor and the wall and corners, you can back the speakers into the corners and you will get plenty of subbass, even with only 1 bass bin in each corner. Note that these loading issues apply only to bass and subbass. Higher frequencies have much shorter wavelengths and their levels are not significantly affected by speaker placement.
B. Select full-range cabinets that have extended low-frequency response, and non-harsh midrange and high-frequency response.
We recommend double 15-inch full-range models such as the Carvin TCS215, Cerwin-Vega V253, EV T-252+, EAW LA325, PAS RS-1.2, or the Yorkville E2152. These cabinets are well made and sound very good for what they cost. If you have more money to spend, look into NEXO, Funktion-One, or Meyer cabinets. We strongly discourage purchasing full-range cabinets that have only one front-loaded bass driver. If you don’t have large bass drivers with fairly high cutoff frequencies, your bass will not be well defined or have strong percussive impact. Also avoid cabinets that crossover the second 15″ driver such that it is used for midbass only.
Remember to keep your mid and high volume levels as low as possible, so they are clean, crisp, and audible, but not harsh or in-your-face. By having very loud subbass below 65Hz or so, you can then set the mids and highs fairly quiet, but the system will still sound and feel very loud. The resulting sound however will be very clear and not at all harsh to the ears, and at the end of the night your ears will have suffered no damage. In other words, below 65Hz is where you want most of your power going. You do need dynamic range and plenty of headroom for the mids and highs, i.e. the amps should still accommodate the peak power ratings of the cabinet, but keep them turned down low. This also applies to bass/mid-bass above 65Hz or so. Too much mid-bass gives a system a muddy, ill-defined sound.
II. Do Not Use Active Crossovers or Low Frequency Rolloff Filters with Passive Full Range Speaker Cabinets.
Active crossovers are used in almost all systems as a means of filtering out high frequencies passed to the amplifiers for the bass cabinets, and as a means of blocking low frequencies from going to the amplifiers for the full-range cabinets. Sealed, horn-loaded bass cabinets sound best when high frequencies are rolled off at 18 or 24 dB/octave, at a cutoff frequency of 60 to 100 Hz. Using a higher cutoff frequency can result in the bass sounding muddy, and in significantly less subbass power handling capability. Note that because bass is inherently non-directional in its propagation characteristics, there is no need to run bass cabinets in stereo. In fact, due to vinyl mastering requirements, the bass sounds on almost all records are in mono. This fact is helpful in simplifying the design and setup of systems. When running subs in mono, be sure that the mono source is a sum of the left and right channels. By summing the channels together you get a true mono sum signal for the subs (as opposed to running them in stereo or from the left or right channel only), which provides better bass accuracy, and partially cancels out low frequency vertical displacement and feedback that occurs with turntables. Also note that in no circumstances should full-range speakers be run in mono. Stereo is one of the most significant advances in the history of audio reproduction, but you would be surprised how many sound companies cut corners and run the entire system in mono, since it is easier to set up and requires fewer amplifiers.
Because of Criteria I, we do not require high volume levels at mid and high frequencies. Thus the average power level delivered to the full-range cabinets will be lower as a percentage of maximum rated power than the average power delivered to the bass cabinets. Because of this reduced SPL requirement, there is no need to run full-range cabinets through an active crossover. This results in much more bass being delivered from the full-range cabinets, which is extremely important for the system to have good transient response, and clean, accurate bass.
Most sound companies rely on the bass bins to reproduce all the bass, and send none of it to the full-range cabinets. Because bass cabinets generally have very low natural cutoff frequencies, and hence very poor transient response, it is impossible with this approach to obtain clearly defined, accurate bass. In general, only large drivers with high cutoff frequencies, such as 15″s or 18″s cut-off at 500Hz or higher, have the transient capabilities needed to make a system sound really good. The only way to get accurate transient response from a system which does not have large drivers covering the entire bass range is if the system is controlled by a sophisticated processor which has the ability to precisely phase align the various drivers. These types of systems are more expensive however and can be difficult to configure optimally.
The second criteria can be summarized as follows:
A. Do not filter the signal going to full-range cabinets.
This is one of our golden rules. It is OK to employ a rumble filter (i.e. sharp high pass filter with a cutoff of 15 Hz), though this is usually not necessary because many mixers, processors, and amplifiers already have subsonic filtering built-in.
Again, we cannot stress this enough, do not run the signal to your full-range cabinets through a crossover. Full-range cabinets are called “full-range” because they are designed to reproduce the full range of audible frequencies. Most sound providers / clubs / etc. fall victim to the mainstream industry practice of filtering the lows out from the full-range cabinets, and as a result their systems end up with incoherent, flabby sounding bass. This unfortunate practice came about for two reasons, (1) to allow full-range cabinets to put out more midbass, and (2) to reduce the likelihood of blown woofers in case of levels getting too high. As we previously discussed however, too much midbass (especially in conjunction with less subbass) gives systems a very unpleasing sound. And in regards to (2), this is a backwards mentality, which makes about as much sense as running no sound at all to the speakers to insure they can’t be overdriven. The better thing to do is be careful with your levels, and make sure you are using a high quality leveler in your system so that the levels cannot go too high even if the DJ cranks everything on the mixer.
B. Always use an active crossover or low-pass filter with bass bins.
Pretty much everyone knows this, but there have been times that we have played on other people’s systems and they were running the bass bins full-range. This results in too much mid-bass and substantially diminished subbass, and it sounds terrible. Also make sure the crossover or filter has a cutoff rate of 18dB/octave or higher, and that it sums the left and right inputs together, so that your subbass signal is a true mono sum of both inputs.
III. Avoid the Use of Compressors with Short Release Times.
Most sound system providers mistakenly use a compressor in place of what is called a leveler. A true leveler is simply a compressor which has the capability of utilizing long (5 Seconds or longer) release times, thus allowing the system gain to remain fairly stable while still ensuring the output signal level does not go too high. Many compressors simply do not have long enough release times to avoid the loss in dynamic range and pumping effects that result from fast compression. The problem with many compressors is that they have cheap level detector circuits which do not allow for release times of more than 1 or 2 Seconds. Digital compressors are less likely have this problem and can work well as levelers. Other products such as Auto Gain Controllers or Automatic Level Controllers are also available, but these usually don’t add any essential functionality over that of a basic leveler, can be higher in cost, and sometimes don’t work any better than a cheap compressor anyhow. Note also that many general purpose digital processors are now available which can take care of leveling, parametric EQ, timing/phase alignment, and crossover filtering functions, all in one inexpensive and compact unit. In any case, we cannot stress this enough, whatever you use must have a maximum Release Time of at least 5 full seconds. One product we do recommend is the Behringer AUTOCOM PRO-XL MDX1600, which offers a maximum release time of 5 Seconds, decent audio quality, and a price of under $100.
Maximum performance is obtained from a sound system when it is operated at a power level where the bass output is at or near its maximum, without significant distortion or risk of driver damage. Because few DJs have the sensitivity to consistently maintain the levels at exactly this optimum point, it is necessary to either have a competent sound person continually monitor and adjust the levels, or use a leveler. Most sound providers find it much easier to use a leveler (or unfortunately as is the usually the case, a cheap analog compressor) than to spend the whole night personally monitoring the levels. Still other sound companies will use neither a leveler or compressor, nor personally monitor the levels, but will instead set the levels fairly low so that even with the loudest record they cannot get too high. This results in the system generally being operated at 50% or less of its potential power, and in the levels on the mixer being run too high, which causes distortion. A similar but even worse alternative that is often employed by sound “professionals” is the use of lower power amplifiers in the system, such that the speakers cannot be blown no matter what the input levels to the amp are. A third alternative is installing mechanical or electrical limiting devices in or on the mixer, so that the levels cannot be set at higher than “7”, and the EQ controls are effective only in the negative direction. With the levels set appropriately and checked periodically, this approach works well, and results in improved sound quality since the mixer cannot be pushed into distortion. The ideal solution is the use of both a leveler and mechanical limiting devices in the mixer.
The reason compressors with short release times make things sound bad is because they cause a widely varying system gain that can fluctuate by a factor of 10 or more between bass peaks and valleys. This results in louder sounds being made quieter, in quieter sounds being made louder, and in the envelopes of the various sounds being distorted and flattened. The net result is the turning of clear, high dynamic range music into muddy, flat sounding music. It is analogous to turning the contrast all the way down and the brightness all the way up on a TV set – a clear image can be turned into a washed-out conglomeration of light.
There are applications where shorter release times are appropriate, such as in microphone processing, or in the mastering of a record or CD, as these will sound louder overall with fast compression. However this additional loudness is not necessary on a sound system which is already quite loud. Added loudness at that point will usually make the system sound overly loud and lacking in dynamic range and clarity.
With the proliferation of inexpensive digital mixers that has occurred, it is now possible to obtain a full featured digital mixer with built-in dynamics and leveling capabilities, parametric EQ, MIDI control options, scene memories, and multiple bus sends for around $400. (Not to mention numerous inputs, effects, and very high signal quality.) This is a cost effective and highly flexible option that can do all of the processing required for even complex sound systems. We recommend the Tascam TM-D1000, Yamaha ProMix 01, or Fostex VM200 for these tasks, though any digital mixer with fully parametric EQ, dynamics (with release times of at least 5 Seconds), and multiple bus sends will do. The above mixers can be found on ebay or other websites at any time in the range of $300 to $500. The price and features offered by a number of digital processors are now quite comparable however, and these are more compact, and easier to set up due to their dedicated nature. We have heard good things about the dbx DriveRack Series.
An advantage of more expensive sound systems from NEXO, Funktion-One, Turbosound, and other high-end manufacturers is that they have a thorough system design which includes full-featured processors that provide optimized limiting, crossover filtering, and timing and phase alignment for each speaker cabinet in use. These systems may not offer true leveling functions however, in which case a leveler is still needed for preventing over-compression.
Set your compressor/leveler so there is no more than 3dB of gain reduction when the mixer is at it’s 0dB meter reading, with an attack time of around 30mS and a release time of at least 3 Seconds. This will ensure the overall output level is maximized, and that the inter-channel levels will stay balanced. Because cartridges typically have +/- 1.5 dB of inter-channel gain tolerance, not to mention the variation in the record itself and in the mixer, it is important to have something to compensate for that. A compressor set for 3dB of gain reduction, infinite compression ratio (i.e. a leveler), and dual mono rather than linked operation, is ideal for this purpose. It is important when doing this to ensure that no more than several dB of gain reduction can result however, or the DJ will not have tight control over the EQ, i.e. if the DJ changes the bass setting on the mixer, the compressor/leveler will compensate in the opposite direction, resulting in the system being unresponsive to the mixer’s EQ, and in the mids and highs “pumping up” when the bass gets quieter. It is also a good idea to put a note on the mixer which states something to the effect of “Attention DJ: Please keep output level near the 0dB mark. Going above this will not actually make things any louder, but will result in decreased mixer responsiveness and in increased distortion.” If the club is likely to have different mixers in use, you may want to post a note in the DJ booth instructing the DJs to set the levels on the mixer so that there is no more than several dB of gain reduction that shows up on the compressor/leveler. Different mixers have different gain structures and thus “0dB” is not the same on all mixers.
Perhaps the most important point of all, is to securely lock the compressor/leveler and amps behind a clear plexiglass panel. The DJ should always be able to see the compressor/leveler so it is clear how much gain reduction there is, but no one but you should be able to touch the knobs! If you leave any knobs accessible to anyone it will not be long before you end up with blown speakers, terrible sound, or both. Also do make sure the amps and rack equipment have plenty of ventilation so they will not overheat.
It is a nasty situation that occurs when a compressor is used with too short of a release time or with too much gain reduction. Surprisingly however, the mainstream audio industry and the majority of clubs and sound professionals seem to be entirely unaware of this, and run their systems in such a way that the resulting sound is over-compressed, with very little dynamic range or transient detail.
In 1996 the Sunsonic MX500 DJ Mixer introduced an Auto Gain Limiter function which ensured the mixer output would always be at an optimum level, guaranteeing optimum sound quality and eliminating the need for an external compressor/limiter/ALC. This model is no longer available however, and for a number of years no other DJ Mixer manufacturers offered this feature. Rane stepped up and introduced this feature into their MP44 DJ Mixer, though its release time is very short, making it necessary to modify the internal limiter circuitry for it to be of any use.
A simple yet very effective and high quality internal leveler can be added to DJ Mixers by anyone skilled in electronics. Install an internal motorized volume control (such as those found in most consumer Stereo Receivers), which is set to reduce the output level if the topmost LED’s light in the mixer’s level meters. Then use a buffered RC circuit to slowly return the volume control to its normal setting. (We can supply a parts list and schematic at no charge, or can make this modification to a mixer for you should you not have access to a local electronics expert.) This is a highly cost effective approach which has the advantage of levelling the signal in the mixer, as opposed to after the mixer, where the signal may already have gone into distortion.
IV. Place Speaker Cabinets as Closely Together as Possible.
Sound is simply the variation of air pressure. Because air is relatively thin, pressure differentials propagate through it relatively slowly. (Sound travels at about 1000 feet per second in air.) A difference of only 30 feet between two speaker cabinets thus results in a delay of 30 milliseconds between when the sound from one will arrive at the other. If the effect of such a delay is examined on a sound such as a snare drum or hi-tom, it is obvious that the combining of two such sounds with delays of more than a few milliseconds between them has destructive results. Many clubs fall victim to the “let’s put speakers everywhere” mentality, and have speakers separated by distances of 100 feet or more. If viewed from a music theory perspective, it is clear that a 100 foot separation results in approximately 100mS of delay. For even a fairly slow tempo of music, such as 120bpm with a 4/4 signature, 100mS equates to almost a full 1/16th note! Few people would disagree that any band or DJ who was always 1/16th note off, would be kicked off the stage. Clearly then it should be an emphasis of good club design to minimize the distances between speakers.
As a side note, to clear up a couple of misconceptions we often run across, adding delay to one speaker would not fix this scenario. It might work if you always stood a certain distance from each speaker, but anywhere else and it would likely sound worse. Also note that delay occurs in the air, not in the speaker wire. (Electricity travels almost one million times faster in wire than sound travels through air.)
A follow-on point to this is that it is always better to use a small number of high powered speakers than a larger number of lower powered speakers. The greater the surface area from which sound is radiated, the larger will be the delay spread, and the more drastic will be the resulting timing misalignments and interference patterns. This is analogous to dropping stones into a still pool of water. If only one stone is dropped in, a single group of pristine circular waves will radiate out. If several stones are dropped in however, a complex pattern of frequency dependent nulls and peaks will be created, and the waves from each source will arrive at different places at different times. We have played on many “patchwork” systems which have 20 or more cabinets seemingly from numerous manufacturers and various historical eras, and it is often surprising how little sound can come out of so many cabinets, and how poor the overall sound quality is. A simple system with 4 T-36s and 2 double 15″ full range cabinets will generally put such systems to shame.
Sunsonic systems are always set up in one of two configurations:
A. A single row of bass bins with full range speakers at each end (on top):
This configuration delivers the most accurate and most intense bass, because all the bass bins are right next to each other. With 6 or 8 bass bins, there is still plenty of separation between the full range cabinets to deliver excellent stereo imaging.
B. Two stacks of speakers, separated by no more than 25 feet.
This is a less desirable setup for smaller systems, but works well for larger systems.
For maximum bass response, always place all bass cabinets on the floor, with their backs directly against a wall, and preferably in a corner. The lower the air volume within one wavelength of a bass-bin (i.e. within about 30 feet), the better the impedance match will be between the air and the bass drivers. As a result, two things will happen. First, the acoustic power conversion efficiency of the drivers will go up exponentially due to the improved impedance match of the low impedance cone driver to the normally high impedance air. Second, the relationship between bass SPL and distance from the cabinet will become more linear as opposed to inverse exponential. This means the bass will travel further, due to the horn loading effect created by a corner made by two walls and the floor. In this type of corner, a bass speaker only has to push air in 90 degrees in each axis, i.e. it is 1/8th space loaded. An 1/8th space loaded bass cabinet will have several times the acoustic power conversion efficiency as a 1/2 space loaded (i.e. on the floor but not near any walls) cabinet. Considering these points, it’s clear that you can get much more bass from the same bass-bins simply by placing them correctly.
A side note here is that the DJ does not necessarily need to be behind the speakers. Raves and clubs are often set up this way, likely due to carryover from the typical live sound emphasis on performer-audience dichotomy, but it is often counterproductive to use this approach for dance music. If the speakers are backed against a wall for example, and are facing the DJ, it is not a problem at all for the DJ if he or she has a good loud monitor (with a separate booth monitor level control on the mixer of course). In fact, this set up is highly preferable to many DJs, as they can then between mixes for example turn down the booth monitor and hear the main speakers exactly as the audience is hearing them.
A final note relating to speaker placement is to avoid at all costs setting up two systems in the same room. Because of the logarithmic sound perception mechanisms of the ear, it is ruinous to the music and sound quality in both areas if significant acoustic coupling exists between them. The only practical solution for this is to insure there is a completely separate room for each system (i.e. a full floor-to-ceiling wall between them), or, if the systems are outdoors, that cabinets from different systems are always at least 50 feet apart and facing opposite directions. We have seen promoters do this too many times: They want to get as many DJs on the flyer as they can (so it looks like a bigger party and hence more people will want to go, and so all those DJs will go and promote for the party and bring all their friends), but they don’t seem to realize that the systems are going to clash terribly, making the entire venue sound like a giant echo chamber.
V. Take all steps to insure feedback will not occur with turntables.
Sound systems which put out more than 1,000 Watts of bass must have solid, vibration-proof footing for the turntables, or feedback can occur. There are several approaches to solving this problem. Without them, the power handling capacity, dynamic range, and accuracy of the system can be drastically reduced.
A simple and effective approach to preventing feedback is to use an electronic feedback destroyer (a.k.a. feedback suppressor or feedback eliminator). There are now many models available at low prices. A well prepared sound company should always have one on hand. Note however that feedback suppressors are a secondary fix, which address the symptoms of feedback, but not the cause. If the turntables are on solid footing, the system will be cleaner sounding, more stable and accurate, and will not be dependent on a feedback destroyer to fix everything.
The ideal footing for turntables is a vibration-proof table placed directly on a concrete floor or on the ground. Vibration-proof materials include cinder blocks, which are cheap but heavy (making them better for permanent installs than for mobile sound systems), or wire shelving (such as Metro or Nexel), which also has many accessories available such as wheels or rack mount rails. With wire shelving it helps to weigh the bottom shelf down with some cinder blocks, sandbags, or your amplifiers, so the needle will be less likely to skip if the table is bumped. A more cost-effective table can be made from an ordinary banquet table by securely attaching 4x4s, or 4 or 5 layers of 3/4″ 2’x6′ pieces of particle board to the top. Another option is to pour 1″ or so of concrete into the bottom of a DJ coffin. Note that the turntables should not be left inside of a DJ Coffin which has not been reinforced like this. DJ coffins are usually made of thin wood that easily vibrates and conducts feedback. The foam lining inside some cases can also be deceiving – it looks like it should reduce feedback, but in reality it can allow feedback to bypass the vibration dampening systems that turntables have built into the feet. Set the turntables directly on the cinder blocks/concrete/4x4s/wire table, and make sure these are on solid ground. Stages, scaffolding, etc. should be avoided when possible.
What the turntables are placed on can be just as important as anything else in the sound system. We have seen sound companies bring 10 or 20 cabinets to an event, and then put the turntables on a flimsy table. Feedback then occurs, requiring the levels to be kept turned down below the point of noticeable feedback. In this scenario the 10 or 20 cabinets that the promoter paid to rent, and the sound company spent hours moving around and setting up, are not putting out half the power they could, and sound bad – there might as well have been half as many speakers.
For venues that do not have solid flooring available, the floor will conduct feedback, and you will need to use some sort of cushioning material to isolate the turntables from the floor. Another alternative in these cases is to suspend a rectangular structure of 4x4s from the ceiling using chain or rope. The turntable feet should rest directly on the 4x4s. This is also good for preventing needle skips from the floor movement that occurs from people jumping up and down when they are dancing. Keep in mind that even though a stage or wooden structure may appear solid, unless it is concrete or dirt, or very thick wood (4″ thick or more), bass will still vibrate it and feedback can result.
We have done testing of various materials which are available for preventing feedback, and summarized the results in the table below. For this test, a Cerwin-Vega T36 bass bin was placed 2 feet from a standard 2′ x 4′ particle board banquet table, on which were placed 2 Technics SL1200MK2 turntables. The needle was put in the last inner groove of a record where it loops continuously and there is no music. The floor used for the test was carpet above a thin layer of concrete and wood. The system gain was then increased until just before the incidence of significant sustained feedback, and the gain was noted. The winner of the test is a new product released Aug. 2003 in the US called “FREEFLOAT” (see picture below). These are inexpensive and work great (available from United Record’s Accessories Page.) One issue with these however is that they can go flat, so it is good to have a backup in place such as some foam cushions underneath. They also make the turntables somewhat wobbly, which might be annoying to some DJs (especially scratch DJs), although it did not cause the needle to skip for us when cueing records.
We believe the best overall solution is to put the DJ Coffin on top of 2 small patio cushions. This works well (a 15dB improvement in this test) while allowing the turntables to remain very stable. The FREEFLOAT product is more wobbly, but might be better for very large sound systems or exceptionally flimsy flooring. This test likely is not highly representative of other sound system configurations, and does not consider other factors such as skip resistance from shock. It is also likely that other inflatable products could work very well.
Keeping the bass bins at least 15 or 20 feet from the turntables can be helpful with larger systems, as heavy subbass can vibrate the turntables themselves, regardless of what they are on. As is mentioned in other sections of this document however, in order to minimize intercabinet delays the bass bins should not be more than 30 feet from the DJ or from each other.
VI. Use as Few Processing Components as Possible.
The ideal sound system consists of turntables, a mixer, leveler, low-pass filter or crossover (for the bass cabinets only), amplifiers, and speakers. Nothing else. Many sound providers think that the more EQs, compressors, dynamics processors, and various other fancy looking devices they can put in a rack, the better. While many types of processing components do have their proper place, generally they do more harm than good if not used in exactly the correct manner.
Each piece of equipment that is added to a signal chain causes phase and dynamic range degradation of the original signal. Considering the high level of production quality with which most underground dance records are produced, there is rarely a need for processing of any kind. An exception to this is for equalizing a small room that has bad resonance peaks in the frequency response. Generally however, an EQ will do more harm than good. Even if a room is optimally EQed with a Real Time Analyzer or other tools, that does not always mean it is going to sound good. As the room becomes filled with people, as different tracks are played, or depending on where you stand in the room, an “optimized” EQ setup can end up sounding worse than if no EQ was used at all. If used minimally, with the settings closely monitored as the event progresses, EQs can be beneficial, but otherwise they are more often misused than used correctly. Perhaps one of the most inappropriate uses of EQs is for boosting the low and high frequencies (i.e. the infamous “V” curve). Apparently, many sound providers have lost much of their hearing capability in the high frequency ranges, because they seem to think it is necessary to boost those ranges by as much as 10dB. Another problem with EQs is that they are often designed such that the signal is passed through each filter one by one, each of which introduce some phase degradation. If you must use an EQ, use a Parametric EQ with 3 to 5 variable Q, sweepable bands. These are much better for almost all EQ applications and have better phase performance.
On a related note, we highly recommend the use of active / self-powered speaker systems (speakers which have built-in amplifiers). These can provide large increases in accuracy and power output, and at the same time greatly simplify system setup. (We recommend Meyer’s MTS-4As, though they are a bit overpriced.)
Worthwhile of mention again is that an inexpensive digital mixer or system controller can in fact handle all the signal processing requirements of even a large sound system. Because all processing occurs digitally inside a single mixer/processor at high bit depths, there is less degradation in signal quality and reliability than would occur with several processing components chained together. We also highly recommend the Behringer DEQ2496, which incorporates all necessary dynamics and EQ functions into one high-quality compact unit, and is priced at only ~$300.
VII. Minimal Phase Delay Crossover Networks Should be Used in Full-Range Speakers.
The higher the order of filtering in a speaker’s crossover network, the higher the level of phase distortion that will result. The derivative of phase delay is group delay. If a sound system does not have a constant group delay, different delays are induced to different frequency components of an audio signal. This is much more detrimental to dance music than most other types of music, because percussion is its most important element. Percussion sounds in general are defined by an initial large amplitude impulse (caused by the striking of two surfaces together) followed by a decaying resonance (caused by one of the surfaces oscillating at its natural resonant frequency). An impulse is the only kind of sound that does not have a characteristic frequency – in fact, if an impulse is evaluated in the frequency domain, it is seen to consist of all frequencies simultaneously. When such a sound is fed to a bass cabinet and a full range cabinet, the sound will pass through the various crossover filters to each driver in each cabinet, where the respective frequency components will be radiated. The more filtering done in each crossover, the greater the delays and time differences generally will be between when each component is radiated from each driver. Without minimal phase delay crossovers, the initial precise millisecond “kick” of a clean kick drum sound will come out of each driver at different times, resulting in a skewed, unrealistic sounding drum.
The reason sharp cutoff filters are used in most full-range speakers is because they allow higher broadband power levels to be sustained by each driver. For the purposes of dance music however there is not as much need for exceptional power handling capability in the mid and high frequency drivers. (We use overload protection circuits on these drivers anyhow.) Often it is not practical to replace existing crossover networks within higher-end speaker cabinets, because they are usually highly customized to the electrical and acoustic properties of the cabinet and drivers. Replacing them with generic minimal phase delay crossovers can result in degraded frequency response and may circumvent overload protection circuitry in the original crossover. Fortunately higher-end cabinets are often designed with some degree of phase correction built into the crossover system, and they may already sound just fine even if not a minimal phase design. With many cabinets however it is often a good idea to seek out the assistance of an experienced Electrical Engineer, who can take a closer look at the system phase response, and if necessary can modify the existing crossovers to a 6dB/octave design with characteristics closely optimized to the cabinets and drivers.
Often a good time to upgrade to a minimal phase crossover is while other repairs are being made. A type of speakers used by Sunsonic in the mid-90’s for example were a pair of Sonic double 15″ full range boxes using a Peavey 22XT compression driver. The 22XTs have a very harsh sound and a pole in the frequency response at 8KHz. After many uses the seams in the cabinets started coming apart due to the vibration, and the cheap wood and construction practices with which they were made. The crossovers were mounted on plastic mounting cups, which had become heavily cracked. The existing crossovers were recycled and replaced with a Sunsonic designed and built 6dB/octave crossover with a 750 Hz cutoff frequency, a compensation circuit for the pole in the 22XTs frequency response, and a light bulb overload compression circuit. (Light bulbs are natural power compression circuits – their resistance increases in proportion to filament temperature). This was all assembled onto a heavy duty metal mounting plate, using RTV sealant and 12 gauge stranded wire throughout, and the cabinets then reinforced with screws and wood pieces. The cabinets then delivered hard, precise bass and percussion that would be expected only from a $1,000+ cabinet (but at a cost of only $350 plus $50 in parts). An A/B comparison was done between the first cabinet after it was done and the second before taking it apart, and the difference was amazing.
An exception to this criteria can be made in cases where the crossover filtering is done in a high-end digital processor which can apply timing and phase corrections based upon manufacturer-supplied parameter settings specific to each speaker cabinet. Note however that these types of systems are substantially more expensive and can sound bad if not configured correctly.
VIII. Amplifiers should be able to supply the program power rating of the speakers.
Accurate reproduction of percussion is the most important requirement of sound systems for underground dance music. As was mentioned before, the major defining element of percussion sounds is their initial large amplitude impulse. Accurate reproduction of impulses requires a system with very high dynamic range and high available power. For this reason, amplifiers must be matched to speakers such that each speaker has available to it at least its full rated program power. (Program power is generally defined as twice the continuous rms power rating of a speaker.)
In the case of bass bins, Impulses are of such short duration that they are mostly filtered out of the signal going to the bass bins, so they do not technically need as much headroom as full range cabinets. However, bass bins are more likely to present large reactances and greater average power consumption than full-range cabinets, and so this criteria applies to them as well, if for slightly different reasons. If you had an unlimited budget it would be even better if your amps were able to supply the full peak power rating of the speakers (peak power is generally defined as twice the program power rating of a speaker, or four times the continuous rms power rating), but the difference in quality will generally be insignificant enough that the added expense would not be justified.
Amplifier power ratings are different for different load impedances. Due to increasing resistance in driver coils as their temperature increases, and irregularities in driver impedance curves, impedances seen by an amplifier are often substantially higher than expected. In addition, voice coil reactances can cause momentary impedances to be presented to an amplifier which are equivalent to 4 to 10 or more times greater of a load than the speaker’s rated impedance. As such it is suggested to not run amplifiers at their lowest rated output impedance. For an amplifier to maintain tight control over a voice coil in these conditions, it must have plenty of available headroom, a very high damping factor, and the speaker cables used must have near-zero resistance. (We use only 8 AWG (6.5 mm2) cabling to subbass cabinets.) Also note that only heavy gauge AC Power extension cords should be used. We advise the use of no smaller than 10 AWG (4.0 mm2) extension cords. Using thinner AC Power cords than that can cause significant voltage drops and can result in your amps clipping much sooner and in their power output capability being reduced by 25% or more.
As an example, to drive 4 full-range speakers which each have a 4 Ohm nominal impedance, and 1000 Watts program power handling, an amplifier capable of delivering at least 2000 Watts per channel at 2 Ohms is required for optimal sound quality. Note that you must know what your amplifiers can put out at all load impedances. For example, an AB Intl. 9620 Amplifier is rated as follows: 2000 W/ch @ 2 Ohms, 1500 W/ch @ 4 Ohms, or 900 W/ch @ 8 Ohms. A Cerwin-Vega T-36/750 bass bin is rated at 1000 Watts Program power at 8 Ohms. If you wanted to push 4 T-36s from an AB 9620, that would be a 4 Ohm load per side, which would give you 750 Watts into each T-36. If you did not take the load impedance into account, it would be easy to make the mistake of thinking that it is a 4,000 Watt amp, and so it should be perfect for driving four 1,000 Watt speakers. This would be true if the speakers were each 4 Ohms, but T-36s are 8 Ohms. As a result, an AB9620 is not enough to make 4 T-36s sound as good as they can. We have done side by side tests with bigger and smaller amps and you really can hear the difference. Bass bins sound tighter and cleaner when they have plenty of available power behind them. A Crest Pro 9001, QSC Powerlight 6.0, Crown Macro-Tech 5002VZ, Lab-Gruppen fp6400, or Camco Vortex 6 would be much better for pushing 4 T-36s. These amps are all either very heavy or fairly expensive however. The problem is that T-36s have an unusually high load impedance for their power rating. This means you either need a 5K or 6K Watt amp (in stereo mode) to push them comfortably, or you could instead use 2 smaller amps in bridge mode. For example a Crest CA9 or Crown K2 in bridged mode would be an excellent match for 2 T-36s.
Note that most amplifier manufacturers make several different grades of amplifiers. Consumer grade amplifiers should be avoided at all costs, as they can easily overheat and shut down (yes, they will turn off right in the middle of your show believe it or not) and their audio quality is not as good. If you have a little more money to spend, look into CAMCO, Lab-Gruppen, or Crest’s Pro 200 Series amplifiers. These are also very high quality, but are much lighter and easier to transport.
This document © David X (United Records, Sunsonic Sound System) – http://www.unrec.com/ & http://www.unrec.com/sunsonic/sound2.htm
great thread :bounce_m:
raaa U ROCK dOC :weee:
Thanks Doc,raaaraaaraaaraaa
Generally a good lowdown, but there are some points that need to be made wrt the assumptions (and plain engineering in a few cases)…Myths are prevalent in Audio engineering in general, but PA’s are particularly bad for it in my experience, and this contains a few…:crazy::groucho:
Dance music has a very important, yet subtle property that its appreciation and effectiveness are not improved by utilizing high midrange or treble volume levels. In fact, the exact opposite effect often occurs. Overly loud and harsh midrange and treble sound pressure levels cause a temporary reduction in perceived dynamic range in the listener, induce fatigue and stress, and over time cause hearing loss.
Of equal importance is the subtle property that subbass does not exhibit these properties to nearly the same degree. This is because the energy content of a sound is directly proportional to its frequency and amplitude. Subbass frequencies have large displacement amplitudes, but their frequencies are very low and their wavelengths are very long. As a result they couple more efficiently to the body itself than to the small inner dimensions of the ear, which is why you can feel the lowest subbass frequencies as much or more than you hear them.
Total nonsense IMO (sorry). The aim of building a PA should always begin with linearity in mind. Reproduction should preferably contain no colouration from the equipment (in practice impossible, but we try), or pre-eq based on any type of content that may be played. If you can’t reproduce Carmina at 10K and sweetly, as well as Fatboy Slim, or an acoustic folk band, you aren’t trying hard enough.
And the perceived dynamic shortening alluded to is usually the result of distortion, not high frequencies. It is true that you need a higher SPL of sub than higher frequencies. No comment on the body coupling theory tho’. Need more research to have an opinion on that…
As an example, if you have 2 double 15″ full-range cabinets that are each rated at 1,500 Watts program power, you should have as many bass bins as are necessary to put out 1 to 1.5 times that amount of power. In this case 4 Cerwin-Vega T-36s would be an ideal match. A more general approach would be to figure you’ll need one single-18″ horn-loaded subbass cabinet per 15″ or 18″ driver in your full-range cabinets. That works out to 2 T-36 style cabinets per double 15″ full-range cabinet.
This depends on the setup (active/passive. 2,3,4, or 5 way active, projected environment – ie outside, in a building etc). I’m currently driving seriously bottom heavy – 2KW electrical to sub and lo-mid. about 800W total high-mid and high. SPL is a different matter though. The 100W RMS that drives each of the highs makes them capable of 114dB SPL with the attached flare (they are ceramic 2″ CD drivers, and have a 1/2m horn flare). My sub bass bins are driven by an 800W RMS peavey each in bridge mono mode, and put out about 115-120dB unless they are being driven really hard (they can peak at up to 140ish). So the SPL’s are roughly the same top to toe.
Electrical input power is vastly bottom heavy (a speaker cone that is 18″ in dia takes more power to move electrically than a 2″ – that much is logical. A speaker is a linear motor in essence, and a bigger motor is required if you want to do more work)
As for horn loads, and Helmholtz resonator tubes…I’ll not get started… Suffice to say – SORT THE NEAR FIELD BEFORE YOU START BUILDING LONG THROW.
Taking absolutely no account of phase differential in that last sentence (even a 2 way rig has phase issues, and speaker placement is fundamental to dealing with this. All speaker placement, not just subs). This seems to be a kind of flipside view of the Bass has no directivity myth (so you can sum it in a stereo field and it will make no difference). I’ve posted at length about this somewhere too…:rant::rant::rant:
or better yet, go for 3 or 4 way active. Then the response can be altered to suit the venue… If you don’t want to do this, select full range boxes that don’t distort in the mid-high range (and get the phase alignment with the subs right, so you don’t have dropouts or reinforcements across the frequency spectrum colouring and knackering the sound)
Harsh/In your face again means distortion (not relative SPL’s). It is distortion that does most damage to hearing (due to unnatural movement (ie sharp clips to it’s travel) of the timpanic membrane being transmitted to the inner ear and hammer/anvil then being modulated again by more sharp clipping pressure waves.) At high volumes this translates to a sort of synchronisation damage (the movement being translated is sharply different to the incoming wave causing excess stress as it tries to re-equalise – in speakers eventually blows the coil, or damages the driver cone – in people can easily rupture the timpanic membrane, or damage the delicate bone structures of the ear). If you have ringing in your ears when you leave a venue, that is damage being done to your hearing.
The muddy ill-definition (notwithstanding phase probs which are likely too) is more likely to be a bad speaker design for, or bad placement of the subs (flabby sound is usually due to a too loose box design for the driver being used), or a system in which the speakers are interacting out of phase, and causing drops in places, and peaks in others…
Active crossovers are used in almost all systems as a means of filtering out high frequencies passed to the amplifiers for the bass cabinets, and as a means of blocking low frequencies from going to the amplifiers for the full-range cabinets. Sealed, horn-loaded bass cabinets sound best when high frequencies are rolled off at 18 or 24 dB/octave, at a cutoff frequency of 60 to 100 Hz. Using a higher cutoff frequency can result in the bass sounding muddy, and in significantly less subbass power handling capability. Note that because bass is inherently non-directional in its propagation characteristics, there is no need to run bass cabinets in stereo. In fact, due to vinyl mastering requirements, the bass sounds on almost all records are in mono. This fact is helpful in simplifying the design and setup of systems. When running subs in mono, be sure that the mono source is a sum of the left and right channels. By summing the channels together you get a true mono sum signal for the subs (as opposed to running them in stereo or from the left or right channel only), which provides better bass accuracy, and partially cancels out low frequency vertical displacement and feedback that occurs with turntables. Also note that in no circumstances should full-range speakers be run in mono. Stereo is one of the most significant advances in the history of audio reproduction, but you would be surprised how many sound companies cut corners and run the entire system in mono, since it is easier to set up and requires fewer amplifiers.
First comment true. Rest….. Active crossovers do indeed filter frequency bands, and feed them to the relevant amps (in order to maximise the characteristics of both venue and speaker). 18 or 24db/octave is a fairly good rolloff (most actives use this rolloff, usually with a 4th order Linkwitz, or Linkwitz-Riley filter setup). However – the active crossovers then amplify the output level, to send a full input level to each amp (passives truly do filter which seems to be his confusion point, and at the speaker end instead of pre-amplification, so the filtered level is lost to a certain extent.)
As for vinyl – I have an awful lot of vinyl that doesn’t mono-sum the bass – so I have no idea where that comes from. And bass is not non-directional. This myth comes from the fact that the dispersion pattern of bass frequencies is roughly spherical – circles are non-directional from inside, therefore lets leap to a bizarre conclusion and present it as fact…
Running in mono does take one of the phase planes out of the setup calculations though – making it easier. Most hire rigs also have comp/limiters on them to protect the equipment, and they are easier to run in mono and coupled.
As for turntables – if you are having trouble…move them. Or get better suspension. Low frequency vertical displacement sounds like management speak for I haven’t a clue what phase is to me… A bit like porn stars being “adult feature models”.
Bass is stereo though, and should be placed in your stereo field like all other frequencies… Anything else is just lazy.
Another attack of technobabble. Lower power electrically does not=lower SPL. The rest is based on this flawed assumption, and should be ignored as nonsense.
He has heard of phase alignment then. Someone tell him you can do it by ear, and a few basic calculations (that even the mathematically challenged like me can perform)…No special equipment necessary.
Most big rigs are more than 2 way active in my experience (only the semi-pro’s or hobbyists tend to use 2 way, because it doesn’t give the required control and feed possibilities that a big rig needs). Hire rigs for dry hire – sure. Less knobs = less chance a monkey will fiddle and blow it up. Engineered rigs – no way. Engineers are fond of control…
And transient responses are more dictated by the speaker cabs than anything (the electricity/speaker interface is by far the most in-efficient, and the dynamic range of the driver coupled with it’s counter tension from the box size and air flows are far more likely to be the culprit of inaccurate transient response). A well built box includes good transient response (which is why I always design and build my own).
The natural cutoff of the box design (usually the point starting from the response curve exceeding 3db/octave dropoff below the box tuning frequency) can be very tight and managed. If you insist on long throw, or 1/4 tube resonators and no near field, you will have naff transient response though (unless you are away in the next field in which case it could sound quite accurate.)
A. Do not filter the signal going to full-range cabinets.
This is one of our golden rules. It is OK to employ a rumble filter (i.e. sharp high pass filter with a cutoff of 15 Hz), though this is usually not necessary because many mixers, processors, and amplifiers already have subsonic filtering built-in.
Again, we cannot stress this enough, do not run the signal to your full-range cabinets through a crossover. Full-range cabinets are called “full-range” because they are designed to reproduce the full range of audible frequencies. Most sound providers / clubs / etc. fall victim to the mainstream industry practice of filtering the lows out from the full-range cabinets, and as a result their systems end up with incoherent, flabby sounding bass. This unfortunate practice came about for two reasons, (1) to allow full-range cabinets to put out more midbass, and (2) to reduce the likelihood of blown woofers in case of levels getting too high. As we previously discussed however, too much midbass (especially in conjunction with less subbass) gives systems a very unpleasing sound. And in regards to (2), this is a backwards mentality, which makes about as much sense as running no sound at all to the speakers to insure they can’t be overdriven. The better thing to do is be careful with your levels, and make sure you are using a high quality leveler in your system so that the levels cannot go too high even if the DJ cranks everything on the mixer.
A 20Hz cut is also useful due to the fact the power requirement to make that frequency in any way meaningful is ridiculous, so you save power and direct it towards the meaningful spectrum. Infrabass is also dangerous if you do manage to get it to levels it can make a difference too…
If you do insist on 2 way only, the next advice is good for sure. If you go fully active however, there are no passives in the rig (filtering twice is dumb – esp as the decent filter – the active network – is up signal path from the naff one – the speaker passive (usually just a few capacitors and a coil).
The problem with clubs/venues however is that the don’t usually hire engineers to install, don’t hire them to run, and don’t hire to maintain the sound equipment. And every man and his dog (and for sure every DJ – otherwise termed gorilla’s with vinyl) knows how to engineer audio – resulting in mayhem, and a round of working out which speaker/amp is buggered this time, and who’s been at the processing this time, with what half arsed notion…:rant::rant::rant:
Simple fact – pay peanuts, get sent monkeys.
Pretty much everyone knows this, but there have been times that we have played on other people’s systems and they were running the bass bins full-range. This results in too much mid-bass and substantially diminished subbass, and it sounds terrible. Also make sure the crossover or filter has a cutoff rate of 18dB/octave or higher, and that it sums the left and right inputs together, so that your subbass signal is a true mono sum of both inputs.
Won’t go here (2-way…why?) Summed bass (lazy!!!). Good crossover network (a given really)
Most sound system providers mistakenly use a compressor in place of what is called a leveler. A true leveler is simply a compressor which has the capability of utilizing long (5 Seconds or longer) release times, thus allowing the system gain to remain fairly stable while still ensuring the output signal level does not go too high. Many compressors simply do not have long enough release times to avoid the loss in dynamic range and pumping effects that result from fast compression. The problem with many compressors is that they have cheap level detector circuits which do not allow for release times of more than 1 or 2 Seconds. Digital compressors are less likely have this problem and can work well as levelers. Other products such as Auto Gain Controllers or Automatic Level Controllers are also available, but these usually don’t add any essential functionality over that of a basic leveler, can be higher in cost, and sometimes don’t work any better than a cheap compressor anyhow. Note also that many general purpose digital processors are now available which can take care of leveling, parametric EQ, timing/phase alignment, and crossover filtering functions, all in one inexpensive and compact unit. In any case, we cannot stress this enough, whatever you use must have a maximum Release Time of at least 5 full seconds. One product we do recommend is the Behringer AUTOCOM PRO-XL MDX1600, which offers a maximum release time of 5 Seconds, decent audio quality, and a price of under $100.
Compression is a black art in many respects (despite the fact it is essentially very simple – an auto gain compensator – ie volume control, with fast response). Compressors are also (to quote a friend who is in the business) “not as fast as my bloody ears and hands”. Limiters (the protection systems referred to as levellers here) is essentially a compressor
set to a hard knee, fixed threshold, infinite gain reduction ration (resulting in a hard power level). A decent active crossover network usually has this built in to it’s output stage.
Compressors set wrong are responsible for more damage than not being there at all (dry hires use them for protection though – if set right they do protect the gear. Knacker the sound usually, but protect the gear)
Pumping usually comes from as noted short release time, but can also come from an uncoupled compressor compressing the stereo image independently. Couple it to set gain reduction over the whole souind and not just 1 side.
This is the most sense spoken in 1 paragraph so far. Introduce DJ’s to the concept of UNITY GAIN – with sharp objects if necessary. A real pro will not need this usually, as they are professional, and their livelihood is based on sounding good.
An engineer is the most valuable part of the setup (and will pay you back far more regularly than a peak limiter ever will)
I have employed the monkey limiter (rewired the main output of the DJ mixer to bypass the onboard eq, while leaving the monitor output running through it). Rude, but oh so satisfying when you get a gorilla who willnae take the telling, and likes being poked with sharp things….
And good headroom and amp/speaker matching is essential – you have learning you need to go and do if you have pro aspirations and don’t believe this…
There are applications where shorter release times are appropriate, such as in microphone processing, or in the mastering of a record or CD, as these will sound louder overall with fast compression. However this additional loudness is not necessary on a sound system which is already quite loud. Added loudness at that point will usually make the system sound overly loud and lacking in dynamic range and clarity.
More good sense – we’re on a roll….:weee::weee::weee:
Or you could pick up one of the older analogue mixers, the sound of which digital systems fail to reproduce. I like Soundcraft desks personally (the only item in the signal chain for PA I allow to purposefully colour the sound). Old analogue desks (I have both a spirit auto, and a very old 400B) can add a really nice, mellow warmth (due to slight distortion introduced at the harmonics of the program material by analogue components). Fairly similar to the warmth of a really good tube amp as it’s vacuum tubes distort the signal. Digital does not do this, and any digital distortion is most unwanted in your signal path. Leave that for the artists to play with.
I picked up my 400B (in flight case) for £195, and my spirit desk with a load of other stuff (multidyne, lexicon reverb, MOTU patch panel, Mac G4, M-audio sound card etc etc) for £700. I never run the rig without it.
good design a given natch… Engineers are still needed for setup. complicated processing is useless if the rig is assembled half arsed. And all the processing in the world will never change this fact. Turbo have some of the best engineers in the business – and that is what shows, and is what made their name synonymous with good PA (they’ve been at it since nearly the beginning of sound reinforcement, and know that their engineers are the most important processor in their rigs…:wink:)
But can introduce the aforementioned pumping and general warble mentioned earlier
More sense here. I personally use a 6-12db limit threshold on the crossover hard limit, but I also never dry hire, meaning there is always a competent individual who can hear any prob before it starts…and can deal with clueless monkeys… Unity gain is something you can never drill into DJ’s enough, and all the stuff about more fader != more volume is one they seem unable to grasp for the most part.
It is a nasty situation that occurs when a compressor is used with too short of a release time or with too much gain reduction. Surprisingly however, the mainstream audio industry and the majority of clubs and sound professionals seem to be entirely unaware of this, and run their systems in such a way that the resulting sound is over-compressed, with very little dynamic range or transient detail.
In 1996 the Sunsonic MX500 DJ Mixer introduced an Auto Gain Limiter function which ensured the mixer output would always be at an optimum level, guaranteeing optimum sound quality and eliminating the need for an external compressor/limiter/ALC. This model is no longer available however, and for a number of years no other DJ Mixer manufacturers offered this feature. Rane stepped up and introduced this feature into their MP44 DJ Mixer, though its release time is very short, making it necessary to modify the internal limiter circuitry for it to be of any use.
A simple yet very effective and high quality internal leveler can be added to DJ Mixers by anyone skilled in electronics. Install an internal motorized volume control (such as those found in most consumer Stereo Receivers), which is set to reduce the output level if the topmost LED’s light in the mixer’s level meters. Then use a buffered RC circuit to slowly return the volume control to its normal setting. (We can supply a parts list and schematic at no charge, or can make this modification to a mixer for you should you not have access to a local electronics expert.) This is a highly cost effective approach which has the advantage of levelling the signal in the mixer, as opposed to after the mixer, where the signal may already have gone into distortion.
All uncommon good sense (and confusing in light of the belief in the earlier mythology…)
The only addition I would make is that over compression is responsible for many of the ills that beset venue rigs (and in some cases good hired rigs when the performer insists on mastering compression pre mixer feed – no names mentioned, but a very well known, (and excellent sounding on their CD’s,) outfit are especially guilty of this)…
I’m gonna stop here for a bit… I will come back and edit the rest later. I have (as usual) more to say….:wink:
Sound is simply the variation of air pressure. Because air is relatively thin, pressure differentials propagate through it relatively slowly. (Sound travels at about 1000 feet per second in air.) A difference of only 30 feet between two speaker cabinets thus results in a delay of 30 milliseconds between when the sound from one will arrive at the other. If the effect of such a delay is examined on a sound such as a snare drum or hi-tom, it is obvious that the combining of two such sounds with delays of more than a few milliseconds between them has destructive results. Many clubs fall victim to the “let’s put speakers everywhere” mentality, and have speakers separated by distances of 100 feet or more. If viewed from a music theory perspective, it is clear that a 100 foot separation results in approximately 100mS of delay. For even a fairly slow tempo of music, such as 120bpm with a 4/4 signature, 100mS equates to almost a full 1/16th note! Few people would disagree that any band or DJ who was always 1/16th note off, would be kicked off the stage. Clearly then it should be an emphasis of good club design to minimize the distances between speakers.
As a side note, to clear up a couple of misconceptions we often run across, adding delay to one speaker would not fix this scenario. It might work if you always stood a certain distance from each speaker, but anywhere else and it would likely sound worse. Also note that delay occurs in the air, not in the speaker wire. (Electricity travels almost one million times faster in wire than sound travels through air.)
A follow-on point to this is that it is always better to use a small number of high powered speakers than a larger number of lower powered speakers. The greater the surface area from which sound is radiated, the larger will be the delay spread, and the more drastic will be the resulting timing misalignments and interference patterns. This is analogous to dropping stones into a still pool of water. If only one stone is dropped in, a single group of pristine circular waves will radiate out. If several stones are dropped in however, a complex pattern of frequency dependent nulls and peaks will be created, and the waves from each source will arrive at different places at different times. We have played on many “patchwork” systems which have 20 or more cabinets seemingly from numerous manufacturers and various historical eras, and it is often surprising how little sound can come out of so many cabinets, and how poor the overall sound quality is. A simple system with 4 T-36s and 2 double 15″ full range cabinets will generally put such systems to shame.
Sunsonic systems are always set up in one of two configurations:
A. A single row of bass bins with full range speakers at each end (on top):
This configuration delivers the most accurate and most intense bass, because all the bass bins are right next to each other. With 6 or 8 bass bins, there is still plenty of separation between the full range cabinets to deliver excellent stereo imaging.
B. Two stacks of speakers, separated by no more than 25 feet.
This is a less desirable setup for smaller systems, but works well for larger systems.
For maximum bass response, always place all bass cabinets on the floor, with their backs directly against a wall, and preferably in a corner. The lower the air volume within one wavelength of a bass-bin (i.e. within about 30 feet), the better the impedance match will be between the air and the bass drivers. As a result, two things will happen. First, the acoustic power conversion efficiency of the drivers will go up exponentially due to the improved impedance match of the low impedance cone driver to the normally high impedance air. Second, the relationship between bass SPL and distance from the cabinet will become more linear as opposed to inverse exponential. This means the bass will travel further, due to the horn loading effect created by a corner made by two walls and the floor. In this type of corner, a bass speaker only has to push air in 90 degrees in each axis, i.e. it is 1/8th space loaded. An 1/8th space loaded bass cabinet will have several times the acoustic power conversion efficiency as a 1/2 space loaded (i.e. on the floor but not near any walls) cabinet. Considering these points, it’s clear that you can get much more bass from the same bass-bins simply by placing them correctly.
A side note here is that the DJ does not necessarily need to be behind the speakers. Raves and clubs are often set up this way, likely due to carryover from the typical live sound emphasis on performer-audience dichotomy, but it is often counterproductive to use this approach for dance music. If the speakers are backed against a wall for example, and are facing the DJ, it is not a problem at all for the DJ if he or she has a good loud monitor (with a separate booth monitor level control on the mixer of course). In fact, this set up is highly preferable to many DJs, as they can then between mixes for example turn down the booth monitor and hear the main speakers exactly as the audience is hearing them.
A final note relating to speaker placement is to avoid at all costs setting up two systems in the same room. Because of the logarithmic sound perception mechanisms of the ear, it is ruinous to the music and sound quality in both areas if significant acoustic coupling exists between them. The only practical solution for this is to insure there is a completely separate room for each system (i.e. a full floor-to-ceiling wall between them), or, if the systems are outdoors, that cabinets from different systems are always at least 50 feet apart and facing opposite directions. We have seen promoters do this too many times: They want to get as many DJs on the flyer as they can (so it looks like a bigger party and hence more people will want to go, and so all those DJs will go and promote for the party and bring all their friends), but they don’t seem to realize that the systems are going to clash terribly, making the entire venue sound like a giant echo chamber.
V. Take all steps to insure feedback will not occur with turntables.
Sound systems which put out more than 1,000 Watts of bass must have solid, vibration-proof footing for the turntables, or feedback can occur. There are several approaches to solving this problem. Without them, the power handling capacity, dynamic range, and accuracy of the system can be drastically reduced.
A simple and effective approach to preventing feedback is to use an electronic feedback destroyer (a.k.a. feedback suppressor or feedback eliminator). There are now many models available at low prices. A well prepared sound company should always have one on hand. Note however that feedback suppressors are a secondary fix, which address the symptoms of feedback, but not the cause. If the turntables are on solid footing, the system will be cleaner sounding, more stable and accurate, and will not be dependent on a feedback destroyer to fix everything.
The ideal footing for turntables is a vibration-proof table placed directly on a concrete floor or on the ground. Vibration-proof materials include cinder blocks, which are cheap but heavy (making them better for permanent installs than for mobile sound systems), or wire shelving (such as Metro or Nexel), which also has many accessories available such as wheels or rack mount rails. With wire shelving it helps to weigh the bottom shelf down with some cinder blocks, sandbags, or your amplifiers, so the needle will be less likely to skip if the table is bumped. A more cost-effective table can be made from an ordinary banquet table by securely attaching 4x4s, or 4 or 5 layers of 3/4″ 2’x6′ pieces of particle board to the top. Another option is to pour 1″ or so of concrete into the bottom of a DJ coffin. Note that the turntables should not be left inside of a DJ Coffin which has not been reinforced like this. DJ coffins are usually made of thin wood that easily vibrates and conducts feedback. The foam lining inside some cases can also be deceiving – it looks like it should reduce feedback, but in reality it can allow feedback to bypass the vibration dampening systems that turntables have built into the feet. Set the turntables directly on the cinder blocks/concrete/4x4s/wire table, and make sure these are on solid ground. Stages, scaffolding, etc. should be avoided when possible.
What the turntables are placed on can be just as important as anything else in the sound system. We have seen sound companies bring 10 or 20 cabinets to an event, and then put the turntables on a flimsy table. Feedback then occurs, requiring the levels to be kept turned down below the point of noticeable feedback. In this scenario the 10 or 20 cabinets that the promoter paid to rent, and the sound company spent hours moving around and setting up, are not putting out half the power they could, and sound bad – there might as well have been half as many speakers.
For venues that do not have solid flooring available, the floor will conduct feedback, and you will need to use some sort of cushioning material to isolate the turntables from the floor. Another alternative in these cases is to suspend a rectangular structure of 4x4s from the ceiling using chain or rope. The turntable feet should rest directly on the 4x4s. This is also good for preventing needle skips from the floor movement that occurs from people jumping up and down when they are dancing. Keep in mind that even though a stage or wooden structure may appear solid, unless it is concrete or dirt, or very thick wood (4″ thick or more), bass will still vibrate it and feedback can result.
We have done testing of various materials which are available for preventing feedback, and summarized the results in the table below. For this test, a Cerwin-Vega T36 bass bin was placed 2 feet from a standard 2′ x 4′ particle board banquet table, on which were placed 2 Technics SL1200MK2 turntables. The needle was put in the last inner groove of a record where it loops continuously and there is no music. The floor used for the test was carpet above a thin layer of concrete and wood. The system gain was then increased until just before the incidence of significant sustained feedback, and the gain was noted. The winner of the test is a new product released Aug. 2003 in the US called “FREEFLOAT” (see picture below). These are inexpensive and work great (available from United Record’s Accessories Page.) One issue with these however is that they can go flat, so it is good to have a backup in place such as some foam cushions underneath. They also make the turntables somewhat wobbly, which might be annoying to some DJs (especially scratch DJs), although it did not cause the needle to skip for us when cueing records.
We believe the best overall solution is to put the DJ Coffin on top of 2 small patio cushions. This works well (a 15dB improvement in this test) while allowing the turntables to remain very stable. The FREEFLOAT product is more wobbly, but might be better for very large sound systems or exceptionally flimsy flooring. This test likely is not highly representative of other sound system configurations, and does not consider other factors such as skip resistance from shock. It is also likely that other inflatable products could work very well.
Keeping the bass bins at least 15 or 20 feet from the turntables can be helpful with larger systems, as heavy subbass can vibrate the turntables themselves, regardless of what they are on. As is mentioned in other sections of this document however, in order to minimize intercabinet delays the bass bins should not be more than 30 feet from the DJ or from each other.
VI. Use as Few Processing Components as Possible.
The ideal sound system consists of turntables, a mixer, leveler, low-pass filter or crossover (for the bass cabinets only), amplifiers, and speakers. Nothing else. Many sound providers think that the more EQs, compressors, dynamics processors, and various other fancy looking devices they can put in a rack, the better. While many types of processing components do have their proper place, generally they do more harm than good if not used in exactly the correct manner.
Each piece of equipment that is added to a signal chain causes phase and dynamic range degradation of the original signal. Considering the high level of production quality with which most underground dance records are produced, there is rarely a need for processing of any kind. An exception to this is for equalizing a small room that has bad resonance peaks in the frequency response. Generally however, an EQ will do more harm than good. Even if a room is optimally EQed with a Real Time Analyzer or other tools, that does not always mean it is going to sound good. As the room becomes filled with people, as different tracks are played, or depending on where you stand in the room, an “optimized” EQ setup can end up sounding worse than if no EQ was used at all. If used minimally, with the settings closely monitored as the event progresses, EQs can be beneficial, but otherwise they are more often misused than used correctly. Perhaps one of the most inappropriate uses of EQs is for boosting the low and high frequencies (i.e. the infamous “V” curve). Apparently, many sound providers have lost much of their hearing capability in the high frequency ranges, because they seem to think it is necessary to boost those ranges by as much as 10dB. Another problem with EQs is that they are often designed such that the signal is passed through each filter one by one, each of which introduce some phase degradation. If you must use an EQ, use a Parametric EQ with 3 to 5 variable Q, sweepable bands. These are much better for almost all EQ applications and have better phase performance.
On a related note, we highly recommend the use of active / self-powered speaker systems (speakers which have built-in amplifiers). These can provide large increases in accuracy and power output, and at the same time greatly simplify system setup. (We recommend Meyer’s MTS-4As, though they are a bit overpriced.)
Worthwhile of mention again is that an inexpensive digital mixer or system controller can in fact handle all the signal processing requirements of even a large sound system. Because all processing occurs digitally inside a single mixer/processor at high bit depths, there is less degradation in signal quality and reliability than would occur with several processing components chained together. We also highly recommend the Behringer DEQ2496, which incorporates all necessary dynamics and EQ functions into one high-quality compact unit, and is priced at only ~$300.
VII. Minimal Phase Delay Crossover Networks Should be Used in Full-Range Speakers.
The higher the order of filtering in a speaker’s crossover network, the higher the level of phase distortion that will result. The derivative of phase delay is group delay. If a sound system does not have a constant group delay, different delays are induced to different frequency components of an audio signal. This is much more detrimental to dance music than most other types of music, because percussion is its most important element. Percussion sounds in general are defined by an initial large amplitude impulse (caused by the striking of two surfaces together) followed by a decaying resonance (caused by one of the surfaces oscillating at its natural resonant frequency). An impulse is the only kind of sound that does not have a characteristic frequency – in fact, if an impulse is evaluated in the frequency domain, it is seen to consist of all frequencies simultaneously. When such a sound is fed to a bass cabinet and a full range cabinet, the sound will pass through the various crossover filters to each driver in each cabinet, where the respective frequency components will be radiated. The more filtering done in each crossover, the greater the delays and time differences generally will be between when each component is radiated from each driver. Without minimal phase delay crossovers, the initial precise millisecond “kick” of a clean kick drum sound will come out of each driver at different times, resulting in a skewed, unrealistic sounding drum.
The reason sharp cutoff filters are used in most full-range speakers is because they allow higher broadband power levels to be sustained by each driver. For the purposes of dance music however there is not as much need for exceptional power handling capability in the mid and high frequency drivers. (We use overload protection circuits on these drivers anyhow.) Often it is not practical to replace existing crossover networks within higher-end speaker cabinets, because they are usually highly customized to the electrical and acoustic properties of the cabinet and drivers. Replacing them with generic minimal phase delay crossovers can result in degraded frequency response and may circumvent overload protection circuitry in the original crossover. Fortunately higher-end cabinets are often designed with some degree of phase correction built into the crossover system, and they may already sound just fine even if not a minimal phase design. With many cabinets however it is often a good idea to seek out the assistance of an experienced Electrical Engineer, who can take a closer look at the system phase response, and if necessary can modify the existing crossovers to a 6dB/octave design with characteristics closely optimized to the cabinets and drivers.
Often a good time to upgrade to a minimal phase crossover is while other repairs are being made. A type of speakers used by Sunsonic in the mid-90’s for example were a pair of Sonic double 15″ full range boxes using a Peavey 22XT compression driver. The 22XTs have a very harsh sound and a pole in the frequency response at 8KHz. After many uses the seams in the cabinets started coming apart due to the vibration, and the cheap wood and construction practices with which they were made. The crossovers were mounted on plastic mounting cups, which had become heavily cracked. The existing crossovers were recycled and replaced with a Sunsonic designed and built 6dB/octave crossover with a 750 Hz cutoff frequency, a compensation circuit for the pole in the 22XTs frequency response, and a light bulb overload compression circuit. (Light bulbs are natural power compression circuits – their resistance increases in proportion to filament temperature). This was all assembled onto a heavy duty metal mounting plate, using RTV sealant and 12 gauge stranded wire throughout, and the cabinets then reinforced with screws and wood pieces. The cabinets then delivered hard, precise bass and percussion that would be expected only from a $1,000+ cabinet (but at a cost of only $350 plus $50 in parts). An A/B comparison was done between the first cabinet after it was done and the second before taking it apart, and the difference was amazing.
An exception to this criteria can be made in cases where the crossover filtering is done in a high-end digital processor which can apply timing and phase corrections based upon manufacturer-supplied parameter settings specific to each speaker cabinet. Note however that these types of systems are substantially more expensive and can sound bad if not configured correctly.
VIII. Amplifiers should be able to supply the program power rating of the speakers.
Accurate reproduction of percussion is the most important requirement of sound systems for underground dance music. As was mentioned before, the major defining element of percussion sounds is their initial large amplitude impulse. Accurate reproduction of impulses requires a system with very high dynamic range and high available power. For this reason, amplifiers must be matched to speakers such that each speaker has available to it at least its full rated program power. (Program power is generally defined as twice the continuous rms power rating of a speaker.)
In the case of bass bins, Impulses are of such short duration that they are mostly filtered out of the signal going to the bass bins, so they do not technically need as much headroom as full range cabinets. However, bass bins are more likely to present large reactances and greater average power consumption than full-range cabinets, and so this criteria applies to them as well, if for slightly different reasons. If you had an unlimited budget it would be even better if your amps were able to supply the full peak power rating of the speakers (peak power is generally defined as twice the program power rating of a speaker, or four times the continuous rms power rating), but the difference in quality will generally be insignificant enough that the added expense would not be justified.
Amplifier power ratings are different for different load impedances. Due to increasing resistance in driver coils as their temperature increases, and irregularities in driver impedance curves, impedances seen by an amplifier are often substantially higher than expected. In addition, voice coil reactances can cause momentary impedances to be presented to an amplifier which are equivalent to 4 to 10 or more times greater of a load than the speaker’s rated impedance. As such it is suggested to not run amplifiers at their lowest rated output impedance. For an amplifier to maintain tight control over a voice coil in these conditions, it must have plenty of available headroom, a very high damping factor, and the speaker cables used must have near-zero resistance. (We use only 8 AWG (6.5 mm2) cabling to subbass cabinets.) Also note that only heavy gauge AC Power extension cords should be used. We advise the use of no smaller than 10 AWG (4.0 mm2) extension cords. Using thinner AC Power cords than that can cause significant voltage drops and can result in your amps clipping much sooner and in their power output capability being reduced by 25% or more.
As an example, to drive 4 full-range speakers which each have a 4 Ohm nominal impedance, and 1000 Watts program power handling, an amplifier capable of delivering at least 2000 Watts per channel at 2 Ohms is required for optimal sound quality. Note that you must know what your amplifiers can put out at all load impedances. For example, an AB Intl. 9620 Amplifier is rated as follows: 2000 W/ch @ 2 Ohms, 1500 W/ch @ 4 Ohms, or 900 W/ch @ 8 Ohms. A Cerwin-Vega T-36/750 bass bin is rated at 1000 Watts Program power at 8 Ohms. If you wanted to push 4 T-36s from an AB 9620, that would be a 4 Ohm load per side, which would give you 750 Watts into each T-36. If you did not take the load impedance into account, it would be easy to make the mistake of thinking that it is a 4,000 Watt amp, and so it should be perfect for driving four 1,000 Watt speakers. This would be true if the speakers were each 4 Ohms, but T-36s are 8 Ohms. As a result, an AB9620 is not enough to make 4 T-36s sound as good as they can. We have done side by side tests with bigger and smaller amps and you really can hear the difference. Bass bins sound tighter and cleaner when they have plenty of available power behind them. A Crest Pro 9001, QSC Powerlight 6.0, Crown Macro-Tech 5002VZ, Lab-Gruppen fp6400, or Camco Vortex 6 would be much better for pushing 4 T-36s. These amps are all either very heavy or fairly expensive however. The problem is that T-36s have an unusually high load impedance for their power rating. This means you either need a 5K or 6K Watt amp (in stereo mode) to push them comfortably, or you could instead use 2 smaller amps in bridge mode. For example a Crest CA9 or Crown K2 in bridged mode would be an excellent match for 2 T-36s.
Note that most amplifier manufacturers make several different grades of amplifiers. Consumer grade amplifiers should be avoided at all costs, as they can easily overheat and shut down (yes, they will turn off right in the middle of your show believe it or not) and their audio quality is not as good. If you have a little more money to spend, look into CAMCO, Lab-Gruppen, or Crest’s Pro 200 Series amplifiers. These are also very high quality, but are much lighter and easier to transport.
This document © David X (United Records, Sunsonic Sound System) – http://www.unrec.com/ & http://www.unrec.com/sunsonic/sound2.htm
safe alot of very useful info in there, many thanks !!!
Edit done here, cos the edit button vanished from the thread post?
Sound is simply the variation of air pressure. Because air is relatively thin, pressure differentials propagate through it relatively slowly. (Sound travels at about 1000 feet per second in air.) A difference of only 30 feet between two speaker cabinets thus results in a delay of 30 milliseconds between when the sound from one will arrive at the other. If the effect of such a delay is examined on a sound such as a snare drum or hi-tom, it is obvious that the combining of two such sounds with delays of more than a few milliseconds between them has destructive results. Many clubs fall victim to the “let’s put speakers everywhere” mentality, and have speakers separated by distances of 100 feet or more. If viewed from a music theory perspective, it is clear that a 100 foot separation results in approximately 100mS of delay. For even a fairly slow tempo of music, such as 120bpm with a 4/4 signature, 100mS equates to almost a full 1/16th note! Few people would disagree that any band or DJ who was always 1/16th note off, would be kicked off the stage. Clearly then it should be an emphasis of good club design to minimize the distances between speakers.
As a side note, to clear up a couple of misconceptions we often run across, adding delay to one speaker would not fix this scenario. It might work if you always stood a certain distance from each speaker, but anywhere else and it would likely sound worse. Also note that delay occurs in the air, not in the speaker wire. (Electricity travels almost one million times faster in wire than sound travels through air.)
All true (and proof of at least a basic grasp of phase alignment) Bit flaky in the detail though. Placing speakers close together in any stereo field will always cause phase distortion (dropouts and reinforcements of different frequencies, or constructive and destructive interference as it is usually known). All due to the basic physics of sound and it’s propagation and interaction when 2 sources collide. The Hyperphysics site has some good info on the physics of why and how. Unless a listener is in the sweet spot of the audio field, in which phase is aligned etc (only 1 spot usually), this problem occurs. The usual solution is to try to minimise interference across the stereo field by limiting this interaction (use of constant directivity speakers which have a narrow dispersion, not pointing speakers at each other etc). And hundreds of milliseconds are unnecessary here from a problem causing perspective – a wave coming from nearly aligned speakers can be 180degrees out of phase and be only a few milliseconds split. If the program material is an exact duplicate, you will hear virtually nothing (180 degrees means inverted which would cancel out the other speaker)
Phase delay (and inversion) is used by crossover networks to compensate for the fact that a stack is difficult to build if you have to ensure all the drivers are aligned (which would be perfect phase alignment). The calculations I mentioned earlier, and the fine ear tuning are all done to set these delays (and hence phase align) a stack.
All spot on. A side point though – by speakers he means individual boxes, not multiple driver boxes which are far more powerful (it’s called sympathetic coupling – if you drive 2 speakers in a box with exactly the same program – ie linked to the same amp source, they will put out an exponentially louder sound)
A. A single row of bass bins with full range speakers at each end (on top):
This configuration delivers the most accurate and most intense bass, because all the bass bins are right next to each other. With 6 or 8 bass bins, there is still plenty of separation between the full range cabinets to deliver excellent stereo imaging.
B. Two stacks of speakers, separated by no more than 25 feet.
This is a less desirable setup for smaller systems, but works well for larger systems.
No comment other than I always wait to see a space before deciding stack configs etc.
Dose of dogwaffle here. air impedance isn’t a factor – the air is the medium for propagation, and the matching is done with a good box design. Also true of the stuff about limiting the free air volume to improve resonance. If you have well designed boxes (which are resonators – that is their job), why change their characteristics with placement unless unavoidable? Efficiency and power conversion may if you’re lucky go up due to the horn effect (standing wave reflections will make it sound mince tho’, unless you baffle the aforementioned corners). But that doesn’t change the inverse square law of propagation. That’s why it’s a law of physics, and not a guideline of physics….
Resonance and use of horn effects will cheat a bit by focussing, and using waves of set frequency (the fundamental) to cause it and it’s related harmonics to throw a greater distance (in speakers, what is meant by long or short throw – long are horns that propagate their resonant frequencies a greater distance, but with less spread – the standing waves are only at the points in which all frequencies are constructive – known as nodes in the sound field)
The ripple on a pond referred to earlier btw is a transverse waveform (and as such can’t propagate through a gas or liquid – the ripples only travel over the surface)
Sound waves are longitudinal in nature (think slinky ripple – constant forward/backwards motion almost like a spiral), and so behave differently. It strikes me that the attack of nonsense comes from taking the metaphor (which was a sound one) too far, and believing that is how sound waves propagate.
Hyperphysics is again a good place for a technospeak free description of all of this. The Wisconsin prof’s experiment to show a visual representation of frequency and wavelength using a pipe, methane, a sine generator and a match is splendidly eccentric too:weee::weee::weee:raaa
Agree here – lock them away and drop nuts through a small hole in their box…:wink:
Good sense too. Cacophony is exactly that – the noise of a Cacodemon…:groucho:
Sound systems which put out more than 1,000 Watts of bass must have solid, vibration-proof footing for the turntables, or feedback can occur. There are several approaches to solving this problem. Without them, the power handling capacity, dynamic range, and accuracy of the system can be drastically reduced.
A simple and effective approach to preventing feedback is to use an electronic feedback destroyer (a.k.a. feedback suppressor or feedback eliminator). There are now many models available at low prices. A well prepared sound company should always have one on hand. Note however that feedback suppressors are a secondary fix, which address the symptoms of feedback, but not the cause. If the turntables are on solid footing, the system will be cleaner sounding, more stable and accurate, and will not be dependent on a feedback destroyer to fix everything.
The ideal footing for turntables is a vibration-proof table placed directly on a concrete floor or on the ground. Vibration-proof materials include cinder blocks, which are cheap but heavy (making them better for permanent installs than for mobile sound systems), or wire shelving (such as Metro or Nexel), which also has many accessories available such as wheels or rack mount rails. With wire shelving it helps to weigh the bottom shelf down with some cinder blocks, sandbags, or your amplifiers, so the needle will be less likely to skip if the table is bumped. A more cost-effective table can be made from an ordinary banquet table by securely attaching 4x4s, or 4 or 5 layers of 3/4″ 2’x6′ pieces of particle board to the top. Another option is to pour 1″ or so of concrete into the bottom of a DJ coffin. Note that the turntables should not be left inside of a DJ Coffin which has not been reinforced like this. DJ coffins are usually made of thin wood that easily vibrates and conducts feedback. The foam lining inside some cases can also be deceiving – it looks like it should reduce feedback, but in reality it can allow feedback to bypass the vibration dampening systems that turntables have built into the feet. Set the turntables directly on the cinder blocks/concrete/4x4s/wire table, and make sure these are on solid ground. Stages, scaffolding, etc. should be avoided when possible.
Another run of sense…:bounce_fl:bounce_fl:bounce_fl. My solution is to have a Coffin that weighs a lot, but is extremly robust (it’s been run over by a double wheelbase van carrrying 1 ton or so of rig – it was 1/2 on the road, and 1/2 on the kerb too. Not a dent – few scratches to the paint, but hey – it was all good). I usually put it across 2 metal trellis stands which presents a small footprint in the event of wooden floors etc – less contact for sound to travel across (bearing in mind that longitudinal waves propagate far faster in solids than in air or liquid, minimising the propagation route means less sound transmitted). Hanging from the ceiling is a good solution too.
Feedback destroyers are the last resort when the mechanical options are exhausted – processing unnecessarily is just adding more noise to the signal path.
For venues that do not have solid flooring available, the floor will conduct feedback, and you will need to use some sort of cushioning material to isolate the turntables from the floor. Another alternative in these cases is to suspend a rectangular structure of 4x4s from the ceiling using chain or rope. The turntable feet should rest directly on the 4x4s. This is also good for preventing needle skips from the floor movement that occurs from people jumping up and down when they are dancing. Keep in mind that even though a stage or wooden structure may appear solid, unless it is concrete or dirt, or very thick wood (4″ thick or more), bass will still vibrate it and feedback can result.
We have done testing of various materials which are available for preventing feedback, and summarized the results in the table below. For this test, a Cerwin-Vega T36 bass bin was placed 2 feet from a standard 2′ x 4′ particle board banquet table, on which were placed 2 Technics SL1200MK2 turntables. The needle was put in the last inner groove of a record where it loops continuously and there is no music. The floor used for the test was carpet above a thin layer of concrete and wood. The system gain was then increased until just before the incidence of significant sustained feedback, and the gain was noted. The winner of the test is a new product released Aug. 2003 in the US called “FREEFLOAT” (see picture below). These are inexpensive and work great (available from United Record’s Accessories Page.) One issue with these however is that they can go flat, so it is good to have a backup in place such as some foam cushions underneath. They also make the turntables somewhat wobbly, which might be annoying to some DJs (especially scratch DJs), although it did not cause the needle to skip for us when cueing records.
Yup – freshair was the winner if you can get away with it (that inverse square law again):wink:
sounds good
Yup – unplugged my bass bins in a hard wired case before, cos I used a couple of cheap XLR’s without the clip. Bass just rattled them loose. (another benefit of fully active though – didn’t need to get in then and there to rewire – rig was at full tilt, and would have been impossible. Active meant I could retune the remaining speakers/amps to take up the slack. Lost a bit of sub (below about 35Hz), but sounded good still, and only a small volume drop which most people didn’t even notice until I pointed it out.)
The ideal sound system consists of turntables, a mixer, leveler, low-pass filter or crossover (for the bass cabinets only), amplifiers, and speakers. Nothing else. Many sound providers think that the more EQs, compressors, dynamics processors, and various other fancy looking devices they can put in a rack, the better. While many types of processing components do have their proper place, generally they do more harm than good if not used in exactly the correct manner.
Each piece of equipment that is added to a signal chain causes phase and dynamic range degradation of the original signal. Considering the high level of production quality with which most underground dance records are produced, there is rarely a need for processing of any kind. An exception to this is for equalizing a small room that has bad resonance peaks in the frequency response. Generally however, an EQ will do more harm than good. Even if a room is optimally EQed with a Real Time Analyzer or other tools, that does not always mean it is going to sound good. As the room becomes filled with people, as different tracks are played, or depending on where you stand in the room, an “optimized” EQ setup can end up sounding worse than if no EQ was used at all. If used minimally, with the settings closely monitored as the event progresses, EQs can be beneficial, but otherwise they are more often misused than used correctly. Perhaps one of the most inappropriate uses of EQs is for boosting the low and high frequencies (i.e. the infamous “V” curve). Apparently, many sound providers have lost much of their hearing capability in the high frequency ranges, because they seem to think it is necessary to boost those ranges by as much as 10dB. Another problem with EQs is that they are often designed such that the signal is passed through each filter one by one, each of which introduce some phase degradation. If you must use an EQ, use a Parametric EQ with 3 to 5 variable Q, sweepable bands. These are much better for almost all EQ applications and have better phase performance.
Yup – except for the bit about 2 way crossovers. All the rest is true. I add another mixer between DJ and rig (cos I’m a control freak, and the mixer colours the sound pleasantly). EQ should be unnecessary if you set up right, but as noted – use a parametric (or semi parametric) if you do. The aforementioned mixer has a semi parametric EQ (2 sweeps, with hi/low shelf). The EQ is only to achieve a particular ambience though (I play more than dance through any rig I build, and different program types benefit from EQ’ing properly.)
Worthwhile of mention again is that an inexpensive digital mixer or system controller can in fact handle all the signal processing requirements of even a large sound system. Because all processing occurs digitally inside a single mixer/processor at high bit depths, there is less degradation in signal quality and reliability than would occur with several processing components chained together. We also highly recommend the Behringer DEQ2496, which incorporates all necessary dynamics and EQ functions into one high-quality compact unit, and is priced at only ~$300.
Matter of preference IMO (cannae stand powered speakers myself – the sound is coloured in a way I don’t like – same for digital processing mixers). Low tension cabling to speakers is a bonus though.
The higher the order of filtering in a speaker’s crossover network, the higher the level of phase distortion that will result. The derivative of phase delay is group delay. If a sound system does not have a constant group delay, different delays are induced to different frequency components of an audio signal. This is much more detrimental to dance music than most other types of music, because percussion is its most important element. Percussion sounds in general are defined by an initial large amplitude impulse (caused by the striking of two surfaces together) followed by a decaying resonance (caused by one of the surfaces oscillating at its natural resonant frequency). An impulse is the only kind of sound that does not have a characteristic frequency – in fact, if an impulse is evaluated in the frequency domain, it is seen to consist of all frequencies simultaneously. When such a sound is fed to a bass cabinet and a full range cabinet, the sound will pass through the various crossover filters to each driver in each cabinet, where the respective frequency components will be radiated. The more filtering done in each crossover, the greater the delays and time differences generally will be between when each component is radiated from each driver. Without minimal phase delay crossovers, the initial precise millisecond “kick” of a clean kick drum sound will come out of each driver at different times, resulting in a skewed, unrealistic sounding drum.
The reason sharp cutoff filters are used in most full-range speakers is because they allow higher broadband power levels to be sustained by each driver. For the purposes of dance music however there is not as much need for exceptional power handling capability in the mid and high frequency drivers. (We use overload protection circuits on these drivers anyhow.) Often it is not practical to replace existing crossover networks within higher-end speaker cabinets, because they are usually highly customized to the electrical and acoustic properties of the cabinet and drivers. Replacing them with generic minimal phase delay crossovers can result in degraded frequency response and may circumvent overload protection circuitry in the original crossover. Fortunately higher-end cabinets are often designed with some degree of phase correction built into the crossover system, and they may already sound just fine even if not a minimal phase design. With many cabinets however it is often a good idea to seek out the assistance of an experienced Electrical Engineer, who can take a closer look at the system phase response, and if necessary can modify the existing crossovers to a 6dB/octave design with characteristics closely optimized to the cabinets and drivers.
Often a good time to upgrade to a minimal phase crossover is while other repairs are being made. A type of speakers used by Sunsonic in the mid-90’s for example were a pair of Sonic double 15″ full range boxes using a Peavey 22XT compression driver. The 22XTs have a very harsh sound and a pole in the frequency response at 8KHz. After many uses the seams in the cabinets started coming apart due to the vibration, and the cheap wood and construction practices with which they were made. The crossovers were mounted on plastic mounting cups, which had become heavily cracked. The existing crossovers were recycled and replaced with a Sunsonic designed and built 6dB/octave crossover with a 750 Hz cutoff frequency, a compensation circuit for the pole in the 22XTs frequency response, and a light bulb overload compression circuit. (Light bulbs are natural power compression circuits – their resistance increases in proportion to filament temperature). This was all assembled onto a heavy duty metal mounting plate, using RTV sealant and 12 gauge stranded wire throughout, and the cabinets then reinforced with screws and wood pieces. The cabinets then delivered hard, precise bass and percussion that would be expected only from a $1,000+ cabinet (but at a cost of only $350 plus $50 in parts). An A/B comparison was done between the first cabinet after it was done and the second before taking it apart, and the difference was amazing.
An exception to this criteria can be made in cases where the crossover filtering is done in a high-end digital processor which can apply timing and phase corrections based upon manufacturer-supplied parameter settings specific to each speaker cabinet. Note however that these types of systems are substantially more expensive and can sound bad if not configured correctly.
All true as far as it goes. Don’t use passives except for as hi-fi boxes/monitors, so don’t really have an opinion.
Active setup involves applying these phase/timing corrections each setup (a pain sometimes, but the efficiency gain from active rigs makes it more than worthwhile)
Accurate reproduction of percussion is the most important requirement of sound systems for underground dance music. As was mentioned before, the major defining element of percussion sounds is their initial large amplitude impulse. Accurate reproduction of impulses requires a system with very high dynamic range and high available power. For this reason, amplifiers must be matched to speakers such that each speaker has available to it at least its full rated program power. (Program power is generally defined as twice the continuous rms power rating of a speaker.)
In the case of bass bins, Impulses are of such short duration that they are mostly filtered out of the signal going to the bass bins, so they do not technically need as much headroom as full range cabinets. However, bass bins are more likely to present large reactances and greater average power consumption than full-range cabinets, and so this criteria applies to them as well, if for slightly different reasons. If you had an unlimited budget it would be even better if your amps were able to supply the full peak power rating of the speakers (peak power is generally defined as twice the program power rating of a speaker, or four times the continuous rms power rating), but the difference in quality will generally be insignificant enough that the added expense would not be justified.
Amplifier power ratings are different for different load impedances. Due to increasing resistance in driver coils as their temperature increases, and irregularities in driver impedance curves, impedances seen by an amplifier are often substantially higher than expected. In addition, voice coil reactances can cause momentary impedances to be presented to an amplifier which are equivalent to 4 to 10 or more times greater of a load than the speaker’s rated impedance. As such it is suggested to not run amplifiers at their lowest rated output impedance. For an amplifier to maintain tight control over a voice coil in these conditions, it must have plenty of available headroom, a very high damping factor, and the speaker cables used must have near-zero resistance. (We use only 8 AWG (6.5 mm2) cabling to subbass cabinets.) Also note that only heavy gauge AC Power extension cords should be used. We advise the use of no smaller than 10 AWG (4.0 mm2) extension cords. Using thinner AC Power cords than that can cause significant voltage drops and can result in your amps clipping much sooner and in their power output capability being reduced by 25% or more.
Amplifier comment is right. Impedance waffle is nonsense though. Impedance is by nature irregular, and increases with frequency (range could be 4ohm – 50ohm across the audio spectrum). The ratings given are usually using resistive 4 and 8 ohm loads, and only show the amp demonstrates acceptable characteristics being loaded in that way, and outputs this power at each load respectively. Some power amp manufacturers give a 2 ohm rating too, but driving at this impedance is not a good idea generally (lots of reasons, starting with the beta droop effect (large signal nonlinearity) in BJT output stages increasing to unacceptable levels, and ending with the fact that due to the reactive nature of a 2ohm load, certain frequencies will cause dips that could go as low as 1ohm – with the associated effect of thermal and clipping distortion)
Higher impedance is not damaging BTW. Use of AC power cable (15amp rated at least) is refreshing to see though. No talk of gold or whatnot plated contacts, and directional low capacitance wire either…:bounce_g::bounce_g::bounce_g:
Note that most amplifier manufacturers make several different grades of amplifiers. Consumer grade amplifiers should be avoided at all costs, as they can easily overheat and shut down (yes, they will turn off right in the middle of your show believe it or not) and their audio quality is not as good. If you have a little more money to spend, look into CAMCO, Lab-Gruppen, or Crest’s Pro 200 Series amplifiers. These are also very high quality, but are much lighter and easier to transport.
Load matching is a given really if you want to run at optimum efficiency (although driving with a higher impedance load will make the sound cleaner – due to the aforementioned effects at 2ohm being present at a lesser degree with higher impedances)
I try to drive at 8ohm if possible (only my subs are 4), but using an amp capable of delivering the necessary power (amp headroom is essential to any rig). I tend to use Peavey amps these days (generally the older MOS-FET designs are favoured), because they are almost indestructible, and will run happily, with nary a grumble for years. I use an old M2600 to drive my highs – last made pre 1975 AFAIK. It’s older than me, but has never once broken down in the 14 years I’ve owned it. All the other amps I use are the same (CS800’s – 2 of, a PV1200, and the only non Peavey – a Hartfield 800)
All an amp should do (in a perfect world) is turn the signal coming in into a bigger one going out with no other changes. Perfect linearity being impossible in the real world however, choose amps that’s colouration characteristics please you, and ones that will do what is asked without throwing any fits. I used to have a Yam CP2K, but will never get another. Nice sound, but highly tempremental. And see getting repair parts for them…Yam trying to maximise the profit by having you buy, then come back for repair regularly – bit like the theory with cars and having them break after a certain point so you get a new one.
If you get old workhorses, built when pride of workmanship meant more than profit, you can’t go wrong IMO. Having said that – he’s right about Crown. They make some beauties if you have pockets deep enough.
I can stop now…Finally…:weee::weee::weee:. I had seen an earlier incarnation of this FAQ a few years ago. It evolved a bit since then, but the sacred bovines really needed taking out and shooting…:groucho::groucho:.
All in the interests of more partying though, so all good nevertheless..:groucho::love::love::love:
0
Voices
5
Replies
Tags
This topic has no tags
Forums › Rave › Free Parties & Teknivals › A Technical Foundation To Building A Sound System: Part 3