A Perfect DJ's: "DJ Corner"
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This page is the concept of Dennis J. Barela as a result of working within the DJ, Music, Pro Audio, Recording and Entertainment Industries for decades. Our friends and associates have wanted a page to gain knowledge of the DJ industry that is more advanced than many of the resources currently available.
It is recognized there are many organizations such as the A.D.J.A. (American Disc Jockey Association) that are respectable resources within the industry. However, since their teachings and knowledge is mainly focused on the beginner to intermediate DJ, many have requested a forum to address advanced concepts.
This page will be updated regularly and you can feel free to email questions or comments to: dennis@aperfectdj.com.
Questions Links:
What is "line level" and how is it different from "phono level?"
What is the best way to set up "nearfield monitors" in my studio?
I have recording software that allows me to change the "sample rate." What's the difference?
At what level does sound damage hearing?
Why are you using plastic cases?
Is there such a thing as using too much studio foam?
Is there a "best way" to power on and off my different components?
I have agreed to provide my
sound system to be used as a P.A. for a live band. How much
wattage should I use?
A comment on "signal chain."
Question: (from Brian Rodriguez) What is "line level" and how is it different from "phono level?"
A:
Line level
Line level is a term used to denote the strength of an audio signal used to transmit analog sound between audio components such as CD and DVD players, TVs, audio amplifiers, and mixing consoles, and sometimes MP3 players.
In contrast to line level, there are weaker audio signals, such as those from microphones and instrument pickups, and stronger signals, such as those used to drive headphones and loudspeakers. The strength of the various signals does not necessarily correlate with the output voltage of a device; it also depends on the source's output impedance, or the amount of current available to drive different loads.
Overview
Consumer electronic devices concerned with audio (for example Sound cards) often have a connector labeled "line in" and/or "line out". Line out provides an audio signal output and line in receives a signal input. The line in/out connections on a computer sound card are generally unbalanced, with a TRS connector of 6.35 mm (1/4"), 3.5 mm (1/8" miniature) or 2.5 mm (3/32" subminiature). The connections on most other consumer equipment use RCA jacks. In most cases changing the volume setting on the source equipment does not vary the strength of the line out signal.
Nominal levels
A line level describes a line's nominal signal level as a ratio, expressed in decibels, against a standard reference voltage. The nominal level and the reference voltage against which it is expressed depend on the line level being used. While the nominal levels themselves vary, only two reference voltages are common: decibel volts [dBV] for consumer applications, and decibels unloaded [dBu] for professional applications.
The reference voltage for the decibel volt (0 dBV) is 1 VRMS, which is the voltage required to produce 1 milliwatt [mW]; of power across a 1 kilo ohm [kΩ] load.[1] The reference voltage for the decibel unloaded (0 dBu) is the voltage required to produce 1 mW of power across a 600 Ω load (approximately 0.7746 VRMS).[2]
The most common nominal level for consumer audio equipment is −10 dBV, and the most common nominal level for professional equipment is 4 dBu. By convention, nominal levels are always written with an explicit sign symbol. Thus 4 dBu is written as +4 dBu.
Expressed in absolute terms, a signal at −10 dBV is equivalent to a sine wave signal with a peak amplitude of approximately 0.447 volts, or any general signal at 0.316 volts root mean square (VRMS). A signal at +4 dBu is equivalent to a sine wave signal with a peak amplitude of approximately 1.737 volts, or any general signal at approximately 1.228 VRMS.
Peak to peak values are twice the peak values.
When digitized, the number of bits must be assigned to the entire peak to peak range with both negative and positive voltage values. That requires the use of one bit for a sign (+/−) leaving N−1 bits for the data values. Hence a 16 bit (CD standard) only has 15 bits for data which gives 215 (32,768 different values) for both positive and negative voltage values. 24 bits means there are 223 levels (8,388,608 levels) and 32 bits yield 231 (22,147,483,650 levels).
Digitized values run from 0 for zero voltage up to the maximum designed value for the circuit. There is no absolute maximum, and it depends on the circuit design.
|
Line levels and their nominal voltage levels. |
|||
|
Use |
Nominal level |
Nominal level, VRMS |
Peak Amplitude, VPK |
|
ARD, Germany |
+6 dBu |
1.550 (approximate) |
2.192 (approximate) |
|
USA professional audio |
+4 dBu |
1.228 (approximate) |
1.737 (approximate) |
|
Consumer audio |
−10 dBV |
0.316 |
0.447 |
The line level signal is an alternating current signal, meaning that its voltage varies for example from −2.192 V to +2.192 V. [3]
Impedances
Impedance bridging is employed to ensure that very little power is transferred and the line in circuit does not load down the output of the other device. When a line out signal, with its output impedance of around 100 Ω, is connected to a line in with an input impedance of 10 kΩ, most of the voltage appears across the input resistance and almost none of the voltage is dropped across the output. In effect, the output impedance of the source, and the input impedance of the line in form a voltage divider with a shunt element that is large relative to the size of the series element, which ensures that little of the signal is shunted to ground and that current requirements are minimized.
Line out
The signal out of line out remains at a constant level, regardless of the current setting of the volume control. You can connect recording equipment to line out and record the signal, without having to listen to it through the device's speaker, and without the loudness of the recording changing if you change the volume control setting of the device while you are recording.
The impedance is around 100 Ω, the voltage can reach 2 volts peak-to-peak with levels referenced to -10 dBV (300 mV) at 10 kΩ, and frequency response of most modern equipment is advertised as20 Hz - 20 000 Hz (although other factors influence frequency response). This impedance level is much higher than the usual 4 - 8 Ω of a speaker or 32 Ω of headphones, such that a speaker connected to line out essentially short circuits the op-amp. Even if the impedances would match, yielding the theoretical maximum power transfer of 50%, the power supplied through line out is not enough to drive a speaker.
Line in
Line in expects the kind of voltage level and impedance that line out provides. You can typically connect the line out connector of one device with the line in of another. However, doing this with a straight cable directly connected to both devices and having both devices on AC power, you may run into a ground loop; although some devices provide isolation by using an opto-isolator, which does not create a physical connection between the devices.
A line input has a high impedance of around 10 kΩ, as is often labeled as "Hi-Z" input (Z being the designator for impedance)..
Line level in traditional signal paths
Acoustic sounds (such as voices or musical instruments) are often recorded with transducers (microphones and pickups) that produce weak electrical signals. These signals must be amplified to line level, where they are more easily manipulated by other devices such as mixing consoles and tape recorders. Such amplification is performed by a device known as a preamplifier or "preamp". After manipulation at line level, signals are then typically sent to a device known as a power amplifier, where they are amplified to levels that can drive headphones or loudspeakers, which convert the signals back into sounds that can be heard through the air.
Most phonographs also have a low output level and require a preamp; typically, a home stereo amplifier will have a special phono input with a built-in preamp, which is much more sensitive than a line-level input. The phono preamp applies RIAA equalization to the reproduced sound.
Question: (from Joshua Hart) "What is best way to set up "nearfield" monitors in my studio?"
A: Many musicians and professionals working within the Pro Audio industry use speakers known as “near field monitors” that allow accurate monitoring of sound from any recorded or live media. Two speakers are preferred since the human brain is an excellent processor of information analyzing the “environment” that sound is generated from using the two ears that we have evolved with. This post focuses on some of the basics of “nearfield” speaker placement, how, where and why.
(It is my opinion that systems using more than two speakers are just “hype” in the industry since the human body has only two ears and does a great job of assessing environment size, depth, height, reflections, reverberations, location within the environment, etc. of sound. Further, the human brain can isolate sounds mixed with literally hundreds of sources to concentrate just upon one sound. This is a feat that is very difficult to accomplish using man-made devices. As an example: Hook up a good, quality microphone and listen to it through headphones. You will hear that lawnmower, passing cars, your dog, a helicopter, a bird singing, a trash truck, the refrigerator cycling on, the television, your shoes squeaking and kids playing outside. The human brain is able to assimilate all of these stimuli and more and isolate and analyze the sounds. Modern electronic equipment can then duplicate what the human ears and brain can do but at a cost.)
It used to be
that studios had huge wall-mounted speakers that required equally large rooms to
work properly. Commercial studios are carefully acoustically tweaked. I won’t
go into acoustic treatments here, but needless to say a properly tuned room will
make any monitor system work much better.
Nearfield monitors are smaller and
are designed to be placed closer to the listener. One of the benefits of this
is that you’ll hear more of the direct response from the speakers and less of
the room. This means that the room doesn’t have to be quite as perfect to get
decent results.
Rooms have modes, which are frequencies that resonate particularly strongly in the room as a result of the geometry. Because rooms usually have width, length, and height, they tend to have three modes. As a result of this, it’s a good idea to have each of these modes line up in different spots of the frequency spectrum so that you don’t end up with a huge bulge in one frequency range. This means that you either need to have a room with dimensions that aren’t cube-like or are irregular. It also helps to have the speakers placed such that the speakers aren’t the same distance from any two walls to avoid setting up the same resonance in two dimensions. Symmetry to the left and right if the walls are parallel, however, can help balance the reflections from those walls reaching your ears differently.
Generally you’ll want to have your speakers spaced apart from each other the same distance they’ll be from your ears. Basically it should look like an equilateral triangle. The speakers should be angled roughly along those triangle angles. The “tweeters” should be directly facing your ears as the frequencies that they produce tend to be more “directional” than other frequencies.
It’s a good idea to have some sort of acoustic treatment on the ceiling, behind you, to the left, and to the right in the middle between your listening position and the speakers because this is where sound will bounce from the speakers to your ears.
A good set of speaker stands (or some rubber feet if the speakers are on a desk) can go a really long way in reducing their mechanical coupling with the floor or with your desk. Auralex® makes a great product called MoPADS that work excellent in isolating vibrations to a desk or speaker stand. If the speakers are coupled, they’ll cause the bass to transfer into the room’s materials, causing more annoying resonances. Be aware that even if one of your speakers are placed in a room where the back wall is different from the other (i.e. a corner) you will even notice an increase or decrease in tonal and/or bass response for that speaker thus producing an inaccurate representation of the sound and it's environment.
Question: (from James Muratone) "I have recording software that allows me to change the sample rate. What is the difference?"
A: This question is asked a lot by DJs, music lovers and recording enthusiasts alike. The "sample rate" (or "sampling rate") is the number of samples (or pieces of information) per unit of time. The time is usually referenced in "seconds" of time. It must be noted that digital music is only a computer or other digital device's digital representation of an analog signal or "sound wave."
All sound creates a sound wave that varies in size depending upon it's frequency. The more samples you have or assign, the better the representation of the original wave form. I won't go too far into frequencies at this point but it should also be noted that the more samples you have the greater the demand on your computer to reproduce.
Many times when music files are compressed or converted to another file type they lose a lot of their original fidelity due to the fact that the computer and/or program is "simplifying" the signals involved in the music. This is why many recording enthusiasts and music aficionados will demand only the highest of sampling rates available regardless of the size of memory that the file will occupy on your hard drive or how taxing they can be on your computer RAM ("random-access memory") to reproduce.
If you look at the diagram to the right, you will see that a higher sampling rate (top picture) is an 880 Hz sine wave sampled at 88.2 kHz which results in a much better representation of the original sound wave. Compare this to the bottom picture which is the same sine wave sampled at 44.1 kHz. Remember that at it's core, the information that a computer recognizes are "on" and "off" signals. This is why you notice abrupt angles involved in the representation of the original sound wave.
When converting files, many times your program will offer the choice of "variable bitrate" (VBR) or "constant bitrate" (CBR). Some recording software will even allow you to choose the actual bitrate (CBR) that you record at (again at the cost of space). VBR allows the program you are using to choose the bitrate that is most advantageous to achieve high fidelity while keeping the file size down. The advantage of using a VBR is that it produces a higher quality-to-space ratio as compared to CBR.
The most popular file format for the past decade or so has been "mp3" files. These files do a pretty good job even when offering a VBR to produce high fidelity files while maximizing your file space. Microsoft tried to get in on the action by developing the file technology known as "wma" but the file type has never really become as popular as Microsoft would have liked.
Question: (from various) "At what level does sound damage hearing?"
A: Every day we
experience sound in our environment such as the sounds from television and
radio, household appliances and traffic. Normally, we hear these sounds at safe
levels that do not affect our hearing. However, when we are exposed to harmful
noise (sounds that are too loud or loud sounds that last a long time) sensitive
structures in our inner ear can be damaged, causing noise-induced hearing loss (NIHL).
These sensitive structures called "hair cells" are small sensory cells that
convert sound energy into electrical signals that travel to the brain. Once
damaged, our hair cells cannot grow back.
NIHL can be caused by a one-time exposure to an intense “impulse” sound, such as
an explosion, or by continuous exposure to loud sounds over an extended period
of time such as noise generated in a woodworking shop.
It should be noted that long or repeated exposure
to sounds at or above 85 decibels can cause hearing loss.
Let me expand upon what a “decibel” is and how music compares to other
sounds.
Many times an event facility owner will come out with a “decibel meter” or start
talking about decibel levels and local ordinances. It may be surprising to some
that many common sounds that we as humans are subjected to on a daily basis are
actually LOUDER than most DJ rigs.
Editor’s note: It is again my humble opinion to mention that much of today’s popular music is specifically designed to irritate anyone over the age of 25 with exaggerated bass frequencies that do not occur naturally in life. (Sorry, irritating adults is an age old tradition that will never cease) A good DJ or pro audio “sound man” will take this into consideration and compress and/or attenuate these frequencies depending upon the age of the audience and/or local ordinance.
The decibel (abbreviated dB) is the unit used to measure how loud
a sound is. The decibel scale is a little odd because the human ear is
incredibly sensitive. Your ears can hear everything from your fingertip brushing
lightly over your skin to a loud jet engine. In terms of power, the sound of the
jet engine is about 1,000,000,000,000 times more powerful than the smallest
sound that your ears can just barely hear. That's a big difference!
On the decibel scale, the smallest audible sound (near total silence) is
0 dB. A sound 10 times more powerful is 10 dB. A sound 100 times more powerful
than near total silence is 20 dB. A sound 1,000 times more powerful than near
total silence is 30 dB.
Here are some common sounds (including my personal pet peeve: "the dreaded
leaf blower") and their decibel levels:
50 dB
Refrigerator
50 - 75 dB Washing machine 50 - 75 dB Air
conditioner
50 - 80 dB Eelectric shaver
55 - 70 dB Dishwasher
60 - 85 dB Vacuum cleaner
60 - 95 dB Hair dryer
65 - 80 dB Alarm clock
70 - 115 dB Leaf blowers
75 - 85 dB Flush toilet
80 dB Ringing telephone
110 dB Baby crying
90 - 115 dB Subway
120 dB Ambulance siren
130 dB Jackhammer, power drill
130 dB Percussion section at symphony
140 dB Airplane taking off
95 - 110 dB Motorcycle
110 dB Symphony concert
110 dB Car horn
110 -120 dB Rock concert
112 dB CD player on high
117 dB Football game (stadium)
150 dB Firecracker
157 dB Balloon pop
162 dB Fireworks (at 3 feet)
163 dB Rifle
166 dB Handgun
170 dB Shotgun
Question: (from Gil Leslie) "Is it worth the money to send my out of warranty "modular" amplifier to the manufacturer for service or should I just buy a new amplifier?"
A: You bring up a great point. Many of today's amplifiers are "modular" in design. What this means is that servicing and repair is easier and less expensive for the manufacturer and the consumer because different components are "plug and play" and easily replaceable. This concept is more along the line of Henry Ford's idea of assembling a product (in his case, cars) that can easily be manufactured and repaired by having "like parts" that are easy to swap in and out.
The only problem is that amplifiers (like automobiles) these days are much more complicated. They have lots of computer chips, sensors and data gathering gizmos in them that transfer huge amounts of data to computers and other componentry. With "modular" designs this data can get lost with detachable plugs over time. Why? Because oxidation can wreak havoc with these connections that plug in and out. Even systems using gold plating of the metals can become the victim of oxidation over time.
My recommendation: If you are "mechanically inclined" I recommend that you (always unplug any electronic devices prior to attempting disassembly) disassemble the casing of the unit and locate all plug-in type connectors. Unplug the connectors and apply a product made by "Caig" called "Deoxit" for most common contact metals and "ProGold" for gold connectors. It is a product that not only cleans the connection but actually improves the "conductivity" of the metal allowing for more voltage and/or data to transfer through the connections. After applying this product wait about 10 minutes for all of the solvents to dry. Then reassemble the unit making sure that all connectors are re-connected and try it out.
I have known many people to find that this remedy can be a life saver in that you may revive that old amplifier and very well may save yourself a few hundred dollars in the process.
Please let me know how this works out.
Question: (O.K. I've
been asked this question often:)
Am I a "Martin man" or a "Taylor
man?"
A: Some things need to remain confidential.
Question: (From an "unknown
DJ" who performed at Falkner Winery) "Why are you using
plastic cases?
"
A: While cleaning up this past weekend I was approached by a DJ who performed right next door. Like other encounters that I have had in the past he asked (in a laughing voice), "Why are you using plastic cases?" (as he was proud of his own anvil-style flight cases)
It never ceases to amaze me how little the majority of DJs know about the pro audio industry. I do not use "plastic" cases. Most cases that I use are manufactured by SKB or Pelican. These cases are made of composite materials and were originally developed by the U.S. military and currently used today in the armed forces, police and rescue services because of their superior design.
Even though they may be more expensive they are lighter, more durable, secure, water-tight (can actually be used as flotation devices) and protect equipment better. Since most of my equipment can be heavy almost all of these cases are on wheels as well. Most importantly, they are lifetime guaranteed. This is important for equipment that has to endure the punishment of the road. I have never had a problem with warranty issues from these manufacturers as they have always stood behind their products.
My advice for the serious DJs out there: Choose quality, professional equipment if you are serious about your profession and want to protect your investment.
Question: (From Brian Nunez) "I am building a new studio. Is there such a thing as using too much studio foam?"
A: Thanks for asking. Yes, you can actually install too much studio foam. "Studio foam" or other types of sound insulation are used to stop sound waves from bouncing around a room. They are also known a "diffusers" because they stop the repeated bouncing around of sound waves within a "space."
Think about it: If you blindfold yourself and
someone walks you outside your brain knows that from the lack of sound waves
bouncing around (and from the sounds of nature) that you are outdoors. If
someone then leads you into a bathroom you can tell that as well when you start
talking because there is a lack of sound-absorbing material in bathrooms.
In fact, many of the tiles, floors, counters, mirrors, glass, fiberglass, etc.
actually encourages the sound waves to bounce back and forth more than usual.
The sound waves are "reverberating."
The brain can also tell if you are in a large room verses a small room as well. Over years your brain has taken notes as to what different rooms and areas sound like. Once again, the brain is an amazing tool for interpreting space.
If you use too much sound absorbing materials in a studio you can create an "anechoic chamber." These chambers are commonly used in manufacturing of products to assess how products "naturally" sound or are affected by sound. They are used for speakers, automobiles, tools, appliances, etc.
When you walk into an anechoic chamber most people who are not accustomed to them instantly get an uneasy feeling. This is because their brain is not receiving any type of information or feedback about the "space" that you are in and it becomes human nature that your body releases adrenaline and gets into a defensive mode because it is not familiar with this environment and is having problems assessing what you are encountering.
Remember that sound waves are very similar to ripples in a
pool in that they bounce off walls in various directions and some sound
waves get absorbed by other materials and furniture within the room. You
want a studio to "sound" like a room but you also do not want this effect to be
exaggerated. This happens with parallel walls, corners and materials that
do not absorb sound waves (again, like in a bathroom). Corners especially
are known for accumulating and exaggerating low end frequencies. If you
want to hear "true" sound from your studio monitors, care must be taken in
corners.
There are various sound absorption devices specifically made for corners. These devices are commonly known as "LENRD's" or "low end node reduction devices." Lots of low end may be what consumers want these days but if you are needing to judge your mixes you need accuracy and not exaggeration. To the right is an example of some of these devices. If a sound engineer or producer adds too much bass trying to "amaze" the listener without listening to the mix in a good studio, the result is a mix that sounds like mud.
It should also be noted that some "studio monitor" manufacturers have even designed monitors that can "listen to" and asses the room and make adjustments as necessary to achieve a "natural sound." The more we learn about the amazing devices our ears and brain are the better we can create great studio products thus making better audio recordings.
Question: (From Alex Grabowski) "I recently had a problem with a power outage that caused a loud noise in my system when everything was powered back on. Is there a "best way" to power on and off my different components?"
A: Any time you have a power outage with any pro audio equipment it is crucial that you immediately make a dive for your power strip and/or power conditioner to switch it to the off position. Why? Because more times than not, someone who "tripped" the circuit or accidentally unplugged your power supply will most likely reset the circuit or plug back in the power very quickly in an effort to avoid looking like the dork they are. The problem with this is that your equipment can sustain very high input signals from equipment still being turned up thus causing damage to amplifiers and/or speakers.
This even brings up another point: It is very important to always have a stage flashlight on hand for such problems. Many manufacturers like Surefire, Maglight and others produce good flashlights that you can have on your belt at any given time for problem solving. Dark stages are part of the business so a professional should always be prepared. These manufacturers even have small yet powerful models that can be worn with stage clothing or formal wear.
This brings up another issue: It is very important to "maximize" your signal going into your amplifier to reduce stress on the amplifier, lower noise and prolong the life of your amp. To do this you must be familiar with your "signal chain." A signal chain is the music signal that passes from component to component to amplifier. It is important that the volume be maximized just to the threshold of distortion. Usually, this is at approximately 70% to 80% of volume. Many pro audio devices will show this as a "0" reading or a nominal level. Although distortion is not much of an issue with digital devices (like computers) it is important to note that many digital and analog devices are able to increase the signal to make up for signal loss in other areas of the signal chain. (Many digital components can be set at 100%)
Another point to make is that consumer audio has an output of -10db while pro audio equipment will have an output of +4db. This is why you should always try to avoid mixing consumer and pro audio equipment since the "makeup gain" required may force you to induce some distortion and/or noise in the process. If you have properly optimized your signal, you should only have to minimally increase the output at the end of your signal chain.
Amplifiers, however, most commonly use an "attenuator" for volume adjustments. What this means is that when on "0" (completely counter-clockwise) the amplifier will infinitely reject any signals coming in. As you turn up the attenuator it allows more and more signal to be introduced into the amplifier's input circuitry thus creating volume. It should be noted that attenuators allow the amplifier to achieve maximum output almost regardless of where they are turned to depending upon the strength of the input signal. A big mistake that I see with DJs is that they turn the attenuators all the way up an then adjust the volume from other components (like their mixer). This is the wrong thing to do since the amplifier will then produce much more noise, heat, distortion (not to mention shortening the amplifier's life) and be susceptible to damage from uncontrolled signals or someone turning the AC power back on (or plugging back in your AC power) before you can shut down your equipment.
When it comes to a signal chain there definitely is a correct way to turn components on and off. Generally you "power on" your signal chain from the top down and you "power off" the components from the bottom up.
What this means is "power up" your computer, mixer then your components (equalizers, compressors, sonic maximizers, effects processors, crossovers, etc.) followed last by your power amplifier. This lessens the chance of damage to your equipment caused by a stray signal.
To "power off" always turn your amplifier off first then power off your components, mixer and computer in the reverse order. Again, this lessens the chance of causing damage to your amplifiers and/or speakers.
Question: (From Derek Overton) "I am usually a DJ but I have also agreed to provide my sound system to be used as a P.A. for a live band and I want to have enough power for the vocalists but I want to make sure that I have enough power. How much wattage should I use?"
A: When determining the amount of power needed to be heard above a live band the general rule of thumb is to have ten times the power of your lead guitarist. This is usually about 1000 watts RMS minimum since most stage guitar amplifiers are about 100 watts. This amount of power assures that you will have the necessary amount of headroom to get the job done without worrying about feedback. It's also recommended to use a compressor on the vocals so that you don't have spikes in the signal which can also contribute to feedback.
Why use this method? Since the lead guitar and the vocals are fighting to be brought to the front of the mix and share many of the same mid-range frequencies, it is important to use this math as a rule of thumb. (Note: So what all competes with these midrange frequencies? Vocals, guitar and the "crack" or "snap" of the snare drum)
Many times bass players will use considerably more wattage than guitarists but you should not worry about the bass player because:
a) It takes more power to reproduce electric bass frequencies.
b) Bass guitar frequencies usually do not compete with the same frequencies as vocals so the vocals should still be prevalent and centered in your mix.
Follow-up email & Q: "What kind of reverb should I use on the kick drum?"
A: You almost never use any kind of reverb on the kick drum since it tends to "mud-up" the mix and contributes to a sloppy "tempo" or what musicians call "the pocket." The surrounding room or outdoor areas should contribute enough "life" to the sound of the kick drum naturally while keeping the mix "tight." Still, unless you are performing sound reinforcement services for a concert setting and all instruments are mic'd, there should be no reason to mic or amplify the kick drum as long as it's a professional drum kit.
Comment:
Please guys, I cannot say it enough about how important your signal chain is. I could really get into discussing exactly what a "signal chain" is but what it boils down to is that your sound is only as good as your worst piece of equipment. Whether that be your extension cords, power strip, power conditioner, audio card (not a "sound" card!), components, amplifiers, speaker and signal cables and speakers, etc.
Please be a professional and invest in quality "pro audio" equipment that will keep your signal chain pristine!
Thanks for your questions. Dennis J. Barela
This page will be updated regularly and you can feel free to email questions or comments to: dennis@aperfectdj.com
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