Professional audio page

    Using audio equipments

    Do you, the musician, desire that everyone in the audience enjoy your music at an appropriate volume with good tone quality and with all parts in musical balance? In the ideal situation, the performance room.s acoustics would be sufficient to achieve these goals. In reality, even a concert hall cannot satisfy the needs of every performance, much less a church sanctuary, a school theater or a multi-purpose room! When the acoustics are not sufficient, a sound reinforcement system is required.meaning microphones, mixers, amplifiers and speakers.Since musical sounds are complex, reinforcing them is a challenge. Quality equipment and a knowledgeable, experienced audio engineer are foundational to accomplishing this task.A good reinforcement system will provide a variety of microphones from which to choose. Once the microphones are selected, they must be positioned so that the sound captured has a pleasing tone color. The task of reinforcing a musical performance is complex, requiring quality equipment and a good engineer. Yet, that is not all that is required. You are needed, too! In most cases, the complexity of the setup and the intricacy of the adjustments make critical a "sound check".a dress rehearsal with the sound engineer to insure that everything has been done right. In general, judgements are quite different when it comes to the definition of sound, because everyone has his/hers own listening attitude and individual preferences. Also the music itself sets specific conditions. Measurable parameters, influencing the sound, are the room acoustic and the speaker system itself, i.e. the technology of the cabinet and speaker design. To get good sound out of your system, you need to have gooddevices and know how to properly use them. Here you canfind more than a few tips for this.Many common problems in PA systems are noise and distortion. Distortion can come from a LOT of places. It can come from improperconsole / effect / amp gain staging. It can come from clipping your poweramp. It can come from trying to make a weenie speaker do the job ofa stack of much larger and expensive speakers. You have got to knowwhere your problem really is before you can solve it.Noise to the system can generally get there through not well shielded cables or wrong signal sensitivity settings in the system (or just noisy sound source).Vital practice should be done in the performance setting shortly before the program. The engineer needs to hear the way you plan to sound when performing, so the musical pieces should be almost all ready. Many times, musicians work hard on their pieces while forgetting to rehearse with the engineer. On stage, the impact of the performance is diminished by feedback, abrupt volume changes, poor tone quality or a part being accidentally soloed out. All these effects lessen the quality of the presentation, disappointing the performer and reducing the audience.s enjoyment. A completed "sound check" almost eliminates the likelihood of these problems.

      Speakers and amplifiers

      Power amplifiers are deisgned to amplify the line level signal that enters to them to a signal strong enough to drive the speaker elements at the desired power. The relationship between a power amplifier and a loudspeaker is symbiotic, that is, each depends on the other. The wattage sent to the speaker by the amp determines the speakers output level while the impedance of the speaker determines the amplifiers load. As long as everything remaines within "normal" bounds, quality audio is produced. However, when one element falls outside the boundary, system damage may occur. When clipping occurs, several things can happen in a conventional amplifier. In extreme cases, protection circuits kick in. When amplifier is loaded with too low impedance load, the amplifier wil be overloaded, which will result (sooner or later) amplifier overheating (up to amplifier damage) or protection circuits to kick in. Amplifiers are typical rated according to their RMS power. Music and speech require very little RMS power, but have much higher instantaneous peaks. Most loudspeaker spec sheets show a Program rating that is double the RMS Wattage.Three options for matching amplifiers to loudspeakers:

      • 1. Match amplifier RMS output to speaker Program rating divided by two: Economical, and safe as long as the operator does not try to play the system louder than the amplifiers will go, which causes clipping and driver failure. The system will not be as loud and clean as it could be.
      • 2. Match amplifier RMS output to loudspeaker Program rating. This will give the loudest cleanest sound your loudspeakers can deliver. The most expensive and dangerous method, because instantaneous peaks can destroy the loudspeaker. A properly adjusted Peak Limiter is required to prevent this.
      • 3. Pick an amplifier with RMS power rating about 60% of speaker Program power. A good compromise between safety, economy and performance.
      Typically, providing 100% (3 dB) more power than the RMS rating of the speaker system produces the intended result. As a caveat, a hard limiter in conjunction with moderate compression will keep things from getting out of hand.

      Power amplifier have several controls. First, its imperative to understand that amplifier level controls are not "gain" controls. They do not control the amount of gain the amplifier produces. All power amplifiers are designed to produce a set amount of gain. The function of the level control knob is to adjust the signal level coming into the amplifiers input stage.Gain controls do not affect amplifier power. The amp has exactly the same power capability as with the controls turned up all the way; it just takes more signal level to hit full power. If the amplifier has a sensitivity setting, set it to match as closely as possible the output of our mixer and other gear (usually around 1.4v). Now turn on the amplifier and adjust the level controls to the desired sound level.Gain controls in aplifier allow you to optimize the gain structure of your audio system to maximize dynamic range and minimize noise and hum.

      Loudspeakers expect a source impedance somewhere near zero (a voltage source). Audio amplifier drive our speakers from essentially pure voltagesources, and the speakers are design to provide their responsefrom a constant voltage transfer, not a constant power transfer. Thus, as far as speaker systems are concerned, conjugate loadmatching is not only unnecessary, it's a bad idea. However, WITHIN speakers, conjugates are important becausepassive ladder-type crossovers ARE sensitive to the loadimpedance. Thus, in such cases, conjugate loadmatching IS used, and referred to as "zobel" networks.This is a matter internal to the speaker and one forthe designer of the speaker to deal with.Driving your typical speaker from typical voltage-cource amplifiersconjugate load matching simply is not an issue. A "mismatch"between the speaker and the rated load of the amplifier is notparticularly important unless you are trying for the ultimate inoutput power. If the speaker impedance is too high, the output levelwill be a bit low. If the impedance is low, you could get someoverheating in the amplifier.

      A considerable amount of distortion is caused when you try to make a modern a amplifier to give out more power than it can. This will sound bad. When an ideal amplifier clips, the input signal becomes flat-topped waves.Those flat-topped wave often but not always have more HF content than the input signal. The direction of the change depends on how much HF content the input signal hadto start with. The process of squaring sine waves tends to produce square waves which havea spectral content that falls off at 6 dB/octave. Modern music is often very bright, so in some cases clipping does nor cause the HF content so considerably cange.

      PA is designed for sound distribution. Our goal as professional sound engineers is to make quality sound delivery available to as much of the venue as possible or necessary. Speaker placement is integral to this end. Bass cabinets tend to be omnidirectional whereas upper cabinets tend to be uni-directional. Better cabinets are wedge designed to distribute sound in a wider pattern while maintaining a uniform appearance. Audio voids are to be avoided as much as possible.Today's pro grade speakers are designed by accoustical engineers and constructed to exacting specifications with the aid of computerized manufacturing techniques. Speaker arrays used to be sinonymous with flown, permanent speaker installation. PA manufacturers have adapted these advanced designs into an affordable, portable, road-worthy product.

      When connecting amplifiers to speakers look at the speaker impedance and the minimum impedance that the amplifier can handle. For example if your amplifier says that it's minimum impedance it can drive is 4 ohms, then you cna attach a spaker with impedance of 4 ohms or more to it safely. Attaching a spaker with higher imoedance than the imepdace to which amplifier is originally designed (the rated power is tols), just means that the available output power will tail offusually pretty linearly. If you attach a speaker with lower impedance than the minimum impedance of your amplifier, you risk in overloading and damaging your amplifier. Modern amplifiers generally work nicely without any load connected to them. Please note that some older amps don't like not having a load and canoscillate (those are rare, but such amplifiers have existed).

      Note on Hifi amplifier: Many home receivers/amplifiers have connections for two set of speakers. If those speakers are connected in parallel, many home amplifiers paralell the speakers! This meansthat two set of 8 ohm speakers show as one 4 ohm speaker load to the amplifier.With two sets of 4 ohm speakers you, this will be 2 ohm "nominal" loadto amplifier. Any home hifi amplifier/reciever will hate that load.

      There are many methods used to connect speaker cables to amplifiers. For amplifiers, the most popular termination device on professional products has been the dual banana. However, recent regulatory requirements in Europe have outlawed the use of the dual banana plug and forced users to terminate speaker cables with spade lugs or bare ends?an approach that is clearly not advantageous to the customer who wants to reconfigure his system or quickly change out a defective product. It is possible that similar regulatory controls will appear worldwide over the next few years. Neutrik? Speakon? connector is a special connector specifically designed for speaker connection applications (manufacturer says that it should not be used for other applications). The Speakon? connector meets all known safety regulations. Once wired correctly, the connector cannot be plugged in backwards, causing the type of inverted polarity situations that are common with banana hookups. It will provide a safe, secure and reliable method of interfacing your amplifier to the load. The Speakon? connector is nowadays widely used in professional audio field.

      On the audio amplifier market there are different kind of amplifiers, most important of them being PA amplifiers and HiFi amplifiers. PA amps tend to be optimized for heavy duty higher-power use. Hi-Fi amps tend to be designed for home use. PA amplifiers are designed usually so that they will deliver lots of power reliably to the load. PA amp will be designed for far greater output than a Hi-Fi amp. It may have a cooling fan which would be audible in a home situation. PA amps are frequently more noisy physically: mainly the cooling fans, but sometimes buzzing transformers, etc. This noise is not a problem in noisy environment where PA systems are generally use, nut could be annoying at home. The mechanics of PA amplifier is typically heavily built rack-mountable case that can take hard use on the road. PA amplifiers generally have professional audio connectors, typically balanced XLR connectors or 6.3 mm jacks. PA amps may have lower sensitivity (+4dB professional line level vs. -10dB consumer line level). This makes them more difficult to interface to things like consumer preamps, etc. A PA amp will normally be fed from a mixing board. A home system probably needs a front end with switching for various inputs. PA amps are frequently more noisy electrically: Optimizing them for high power sometimes involves trade-offs with low- level signal to noise ratios. Note that most PA amps are never heard at the distances and quiet ambience where Hi-Fi amps are usually found. Being a "PA" amplifier does not impart any inherent superiority or inferiority to any particular "Hi-Fi" amp in sound quality. There's absolutely no reason why a powerful PA amp can't sound perfectly smooth and detailed - and many of them do. The only downside is that they usually have quite noisy cooling fans. There are good and bad products on both categories. A small, quality PA amp can be a useful substitute for a Hi-Fi power amp.

      PA speakers are generally quite unsuitable for home listening. They're often designed to be loud, not to be smooth and detailed. PA speakers are also often designed in such way that they sound good on some distance, and still sound good on longer distance. A large PA speaker could sound very bad if you sit just few meter away from it. A lot of PA speakers sound rather rough in a living room. PA speakers may not be what you want for domestic music replay. Many PA speakers are designed with certain directivity pattern in mind more than very accurate frequency response, because controlled directivity is needed when building spaker system that consists of many speakers stacked or hanged together. If the directivity is not right in those situations, the overall sound quality will be bad. Slight frequency response errors can easily be fixed in PA system with a proper equalizer if needed. Mechanical construction of PA speaker is usually very rugged for life on the road.

      If you are lookign for good speakers for home use, you might find it interesting to look at speakers sold as studio monitors rather than ones sold as hi-end hi-fi. Even a medium-priced pair of nearfield monitors placed the right distance from your ears (a few feet) may give you a VERY pleasant surprise.

      And remeber always that you usually get what you pay for. Be aware that $1,000 is peanuts in the world of high-quality speakers. And when buying speakers it is always a good idea to listen to the speakers well with the intended material you plan to play for them before buying them. All speakers have their good and bad sides, some spakers are better for some uses than soem other, and no speaker will do all situations well.

        Amplifier specs and operation

        When people refer to "amplifiers," they're usually talking about stereo components or musical equipment. But this is only a small representation of the spectrum of audio amplifiers. Amplifier is in general just an electronic device that simply produces a more powerful version of the audio signal that is coming in to the amplifier. In other words the amplifier generates a new audio signal based on the input signal and the amplification factor defined to amplifier circuit (can be adjustable or fixed).

        The amplifiers are generally divided to preamplifier amplifiers. Pre-amplifier is an amplifier that takes a quite weak signal (typically from milliolts to few volts) and outputs an amplified signal (typically 1-4V signal level). The pre-amplifiers have often adjustable amplification factor (volume control) and possibly other controls (for example audio source selector, tone control etc.). Power amplifier is an amplifier that is designed to drive the speaker. It can supply the neeeded power to the speaker (signal level typically few volts to tens of volts and currents typically up to many amperes). In a small amplifier -- the amplifier in a speaker phone, for example -- the final stage might produce only half a watt of power. In a home stereo amplifier, the final stage might produce hundreds of watts. Output amplifiers are generally designed to have fixed amplification (some models have adjustable attenuators in front of final state). Most home hifi amplifier are devices where pre-amplifier and and power amplifier are built into same equipment. In professional amplifier world the pre-amplifiers are typically inside house mixer, and the amplifiers that drive the speaker include just the power amplifier part.

        The component at the heart of most amplifiers is the transistor. The goal of a good amplifier is to cause as little distortion as possible. The final signal driving the speakers should mimic the original input signal as closely as possible.

        There are many different kind of amplifiers and techniques for amplifiers. Sound enthusiasts are fascinated with variations in design that affect power rating, impedance and fidelity, among other specifications. The amplifier operation is generally divided to different amplifier classes:

        • CLASS A: The positive and negative output transistors each handle 100% of the audio signal- they are biased so their zero-signal output current idles halfway between zero and maximum. When the audio current in one transistor increases, the current in one transistor increases, the current in the other decreases; as a result, their voltage move together. In some designs (in preampkifiers for example) on the of the transformers is replaced with a resistor. The primary advantage of class-A operation is inherent lack of distortion. However, a serious flaw is the extreme heat loss at idle. Class A amplifiers are generally only used on pre-amplifiers and some "high-end hifi" amplifiers.
        • CLASS B: Class B amplifier has two output transformer, one for positive and other for negative half of the audio signal. So each transistor control only its half of the waveform. When the waveforms are combined properly, we still get the complete output waveform, but we have eliminated the large idle current. If the waveforms don?t joined together perfectly, we get annopying zero-crossing distortion (frequently called crossover distortion and heard as slight gargling or rattling sound during quiet parts of the program).
        • CLASS AB: One popular method is to compromise between class A and B and operate the amplifier in class AB. Bu permitting a small idle current to flow, we get a small amount of idle heat, but we eliminate any chance of "dead space" between the positive and the negative waveforms. Most professional and hifi power amplifiers nowadays operate in AB mode. This amoplifir class provides both acceptable power consumption and well acceptable sound quality.
        • CLASS C: When each transistor controls less than 50% of the waveform, we call this mode class C. This mode is not usable for audio.
        • CLASS G: This mode uses two or more sets of output transistors connected to different supply voltages. The goal is to reduce the heat loss in class A or B amplifiers. The main problem is to ensure seamless transfer from the low-voltage to the high-voltage transistors to avoid any small glitches similar to zero-crossing distortion, but this techniques has been successfully used on some amplifier (for example QSC Series Three and original QSC MX series)
        • CLASS H: This class uses a single bank or output transistors connected to a low-voltage supply, along with some means of switching them to a higher-voltage supply when required. This method has the same thermal benefits as class G, but it avoids the second bank of output transistors, thus reducing the size and cost of the amplifier. The QSC EX series uses this technique.

        The newest player in the aqmplifider game are so called "digital amplifiers". Sometimes those are referred as amplier classes D, E, F and T. The so-called "digital" or class "D" amplifiers use pulse-width modulation of a square wave that is then filtered to analog. A class-D amplifier is one in which the output transistors are operated as switches. When a transistor is off, the current through it is zero. When it is on, the voltage across it is small, ideally zero. In each case, the power dissipation is very low. This increases the efficiency, thus requiring less power from the power supply and smaller heat sinks for the amplifier. Pulse width modulation is a process that generates different length pulses. A square pulse can have any width: It can smoothly go from "always off" to "always on". The output pulse width is determines by the input signal voltage. The output filter "integrates the area under the curve" The speaker gets an analog signal, just like any other amplifier output. The advantages of class "D" are very high efficiency (lower power consumption and less heat) compared to traditional class "A", "AB" or "B" amplifiers. The have been around in experimental form since the '70s, but they seem to be gaining in popularity due to the large number of power amplifiers required for multichannel surround sound and because of power saving possible on the portable equipment. There has been also some trials in using the class D technology with professional audio amplifiers, and there has been some amplifiers that are built to very small case, weight almost nothing, do not need massive colling fans, and still generate considerable amount of power. The primary disadvantages of class D technolofy is the complexity and sound quality. The speed requirements for the switching transistors are 50 to 100 times greater than for linear audio amplifiers. The high-frequency switching causes radio interference, and many practical problems must be solved to attain the same audio fidelity that we expect with linear amplifiers. Today the class "D" switching amplifiers don't attain the performance of the highest quality traditional designs, but they might eventually. The complexity of the designs also nowadays causes the class D designs to be somewhat more expensive than traditional designs, but this is changing as this technology comes more and more in mass production. The term digital amps" is a misnomer. There are two categories: Analog-controlled class D (switching amplifiers with an analog input signal and an analog control system) and Digitally controlled class D (amplifiers with a digitally generated control that switches a power stage).

        All amplifiers have a maximum power limit. The voltage at the amplifier output can only go as high as the voltage in the dc power supply. If the signal tries to exceed this limit, it "hits the ceiling", and the waveform becomes flattened. This problem, called clipping because it looks like the top of the waveform has been clipped off, results in the familiar ?blatting? sound of an overdriven amplifier. Increasing the supply voltage adds cost and weight to the amplifier, so amplifier power has a big effect on price. Amplifiers have a minimum rated output impedance, which should be equal or less than the impedance of the loudspeaker load. As the impedance of the loudspeaker gets lower, more current will be drawn from the amplifier. This is why, up to a point, the amplifier power rating increases into lower impedances. However, the increased current puts a greater strain on the amplifier components and the power supply. At some minimum impedance, the strain will get so high that the power-supply voltage sags or the transistors overheat. Any further decrease in impedance will cause the amplifier circuitry to collapse, resulting in less power, or it could even cause amplifier failure.

        The ac power comes into the amplifier through the ac cord, is controlled by the on/off switch, and usually goes through a fuse or circuit breaker, which cuts off ac power in case of massive overload. It then reaches the power transformer, which is in the heart of the power supply. The simplest and least expensive transformer is the E-I type, which is generally cubic-shaped (roughly equal height, length, and width). This type is widely used. The U-I type is more expensive, but it is easier to make in a flatter shape that can fit into low-profile amplifiers. The toroidal type is built on a donut-shaped core, which has the best magnetic properties. It can be made quite flat, it weighs somewhat less and is has low hum emissions, but it is the most expensive. Once we have scaled and isolated the ac power through a transformer, it is rectified with a rectifier. Typical large capacitors are connected to the output of the rectifier. The capacitor fills, or charges up, to the peak voltage of the rectified wave-form. If the capacitor is large enough, it stays pretty full between the peaks, and we get an almost perfectly smooth dc voltage. The size and weight of power-supply components has been somewhat reduced over the last 20 years, but progress has been slow because we are only refining the same basic technology.

        The only great change in power supply technology has been switch mode power supply used on some amplifiers. A switch more power supply first rectifies the incoming ac and smooth it with capacitors. Then high-speed switching transistors to convert the dc power to a high-frequency ac waveform that is passed through the switchign transformers. The switch mode power supplies typically operate at 50kHz to 100kHz frequency. Higher freuquency needs a specially constructed amplifier but allows usigg a smaller size transformer. In addition to the primary benefit of greatly reduced weight, switch mode power supplies can control the operation of the high-frequency transistors to compensate for variations in ac voltage and load currents, thus improving both kinds of power-supply regulation. The ultimate result will be more consistent amplifier performance, but the audio industry must solve problems of cost, reliability and radio/TV interference caused by the high-frequency switching.

        Many amplifier use protection circuitry. The lower the impedance of the load, the greater the current drawn from the amplifier, and the greater the heat generated in the output transistors. If too many loudspeakers are connected to the amplifier, or if the ends of the loudspeaker wire touch together by accident, the load impedance goes very low, and the current flow becomes dangerously high. If the flow is not limited, the output transistors will burn out. Therefore, amplifiers need some kind of short-circuit protection. There are also other thing where protection is needed. Common protective circuits include turn-on and turn-off muting, shut-down or muting in case of excessive temperature, protection against radio pickup (RFI), and dc fault protection.

        Speaker specs

        The ohms in a loudspeaker's specification tells you in broad terms whetherthe 'speaker will suit your amplifier. Ohm is a measure of resistance (ormore accurately for alternating currents, impedance) The higher the number,the higher the resistance, and therefore the less current the 'speaker willdraw.Today, most speakers are rated at 8 ohms, some at 4 ohms, so today'samplifiers tend to be designed to work with 'speakers of nominal impedance4-8 ohms.Speaker impedance ratings are very "nominal" and most'speaker's impedance will go down by almost half its rating, and up byseveral times it's rating depending on the design. The nominal power for speakers is defined as the continous power that can be applied to the speaker for 24 hours. This nominal power is measured by pink noise signal. The nominal power is applicable to both a single chassis/driver and complete box. Sometimes nominal power is also referred as thermal power, (according AES/ANSI specs). The maximum power is defined for woofers and boxes only. It is measured by applying sinusoidal signals of 250 Hz and lower such that the speaker is neither damaged nor produces unwanted output.

        Distributed speaker systems

        100V- or 70V-Systems are referred to as 'constant-voltage distributed audio systems'. The constant voltage system is the most economical way to install a multi-speaker sound reinforcement system. This was typically used (years ago) to power large numbers of horn type speakers in outdoor events and as a cheap and cheerful way of running speakers for musack purposes around large buildings or even show relays. This system is still used nowadays for some applications because it allows many speakers to be attached to one amplifier without running into impedance problems. In an installation where you need to run a large number of lower volume loudspeakers, such as a paging system, a restaurant background music system, or a church install, the easiest solution is often a 70-volt speaker distribution system. 70V/100V line systems are easy to wire, easy to expand and are still used in a major way in factories, shopping centres, schools and other environments to this day to play background music, do paging and for evacuation systems.

        The term "100V system" or "70V system" relates to the maximum output voltage of the amplifier. 100V is the usual voltage in Europe, 70V in the United States. The actual voltage used is pretty much the highest local regulations don't consider mains so in the EU we mainly use 100v, presumably in the US the cut offs 70V. A higher voltage up to 200V can be used too for very long cable runs and higher power requirements. To generate this high voltage, the amplifier is equipped with a step-up transformer, which transforms the regular output voltage, in the 15 to 30 Volts range, up to the necessary 100V or 70V respectively. There are direct 70 volt amps out there, and there are normal amps powering the 70 volt systems. A bigger amp can deliver more current and hence drive more speakers, but it won't be any louder with a same set of speakers.

        The main difference to a regular low-impedance system (4 or 8 Ohms) is the way, individual loudspeakers are connected to the loudspeaker line. A large number of single loudspeakers, each equipped with a step-down transformer, can be connected to one single output line. Individual speakers have transformers of suitable ratios to draw their rated power from the line. Each speaker's step-down transformer has a relative high impedance at the primary side to connect to the 100V line. The secondary side of the transformer matches to the speaker itself (mostly 8 Ohms). There are speakers with multi-tap tranformers and volume controls in them, so with suitable speakers it is possible to adjust the volume levels of different speakers locally without affecting the rest of the system operation.

        Also a much smaller wire diameter (AWG) can be used in 70/100V than in a low-impedance system, because increasing voltage and decreasing current minimizes the amount of current flowing in the wire. This solution was borrowed from the electrical power line distribution system years ago. Requirements for long audio distribution came about and the 25 and 70 volt line levels were developed for this purpose. The higher the distribution voltage the lower the losses because of the resistance of the wire to the speakers.A distribution transformer is required to step up the output voltage of the amplifier so that the current flow is kept as low as possible. 25V, 70V, 100V and sometimes even more than 200V are used.

        Many loudspeakers can be placed across the output by using distribution transformers. The input taps of the distribution transformer let one choose the power drawn from the line and the output taps let choose the connected loudspeaker (4 Ohms, 8 Ohms, 16 Ohms). The downside of the use of those transformers is, that they always degrade the sound quality in a certain way (especially the low end). Most audio transformers pass a low frequency of 100 Hz without major loss. If the amplifier is producing power at 30 Hz and feeding it to the transformer it will saturate the core and reflect a short to the amplifier resulting in a loud, possibly damaging, surge or crack to the speaker or a blown speaker fuse on the amplifier. So you won't get big thumping bass or very high output powers with a 70/100V system, but there are many applications that do not need those properties.

        The 70 volt system offers the following benefits compared to "low impedance" system: Lots of speakers on one amp, no need to home run each speaker, higher voltage allows use of smaller wire, speakers can easily be added and removed, economical, no need to calculate impedance (just total power) and EASY to design. Disadvantages of 70 Volts system are: Limited frequency response and the system is considered high voltage by codes.

        There are many commercial products that operate at a constant voltages using transformers. Usually found in 25V, 70V and 100V sizes, these transformers are connected to the amplifier on the primary side and then send one pair of relatively thin stranded wire from device to device. In the case of the 70V audio transformer, a mono audio signal is fed and kept at a constant 70V signal. The 70V voltage is either generated directly with a special amplifier with 70V output, or using conventional 4/8 ohm power amplifier wired to suitable transformer to boost up the voltage to 70V.

        The 70V keeps the signal from degrading but does not have the same fidelity associated to a standard 4 or 8 Ohm system connection. Less wire, longer distances without degradation, better coverage capabilities and easy installation.

        The speakers used in these constant voltage systems will have a transformer with connections called taps. Taps (usually you can find find multiple choices) on those transformers are based on a Wattage specification. The number of speaker transformers and the size of amplifier connected at the head end determine what tap is used. You will be surprised at the amount of volume available from a one Watt tap. In most cases, the sound systems you see in malls, amusement parks, office building and paging systems use this form of wiring at low wattage taps. If you need to adjust the volumes on different speakers different, you can tune the soudn putput levels by selecting different wattage taps.

        Distributed audio systems are often mono systems. The fact that the system is mono does not mean that sound is "bad" or AM quality. The audio quality of mono system can be as clear as with a stereo system, is it just lacking the "stereo image". Often time?s a mono signal will provide you with more information and more fidelity than a stereo signal in applications where multiple speakers are used and listeners are not in the "sweet spot" for sound (stereo sound will sound good only on limited "sweet spot" are between left and right channel speakers).

        • A 70 Volt Meter Attenuator for Sound Reinforcement - Many times a sound reinforcement system that uses the industry accepted 70 volt interface system needs to be metered. Benchmark SPM-220 and SPM-320 meter systems as well as the RPM-1, VU-1 combination can easily do the job and allow the operator to see the audio levels both in VU (average) levels as well as in PPM (peak) mode. In other words by using the peak metering capability of the Benchmark meter systems, an operator can, at any place on the system, not just the amplifier room, see whether or not his amplifier hit clip.    Rate this link
        • A Flat Response - the distributed-mode loudspeaker is set to revolutionize sound system design, audio media and home cinema systems    Rate this link
        • Loudspeaker time delay DATA SHEET - for helping standard method for improving system synchronisation and intelligibility is with the use of signal time delay units    Rate this link
        • Transformer Tek-Notes - Here you'll find a list of technical notes and information for the transformers used in distributed speaker systems.    Rate this link
        • 70 Volt Systems Explained - In an installation where you need to run a large number of lower volume loudspeakers, such as a paging system, a restaurant background music system, or a church install, the easiest solution is often a 70-volt speaker distribution system.    Rate this link
        • 70-Volt Speaker System - Life gets complicated when you have several speakers that you want to hook up together to put the sound in more areas, or handle more power. Rather than wrestle with the nightmare of speaker impedance matching, many large haunts use a 70-Volt speaker system.    Rate this link

      Sound mixing

      An audio mixer is a device that takes mutiple audio inputs and allows the user to blend them together for a single output. Your console mixes signals at "line level" and achieves maximum dynamic range when this is done at or near "unity gain." Each microphone input has a pre-amplifier which adds gain ) to bring the mic level up to line level. The gain required varies with the Sound Pressure Level of the source, distance from source to mic, the microphone?s sensitivity. . It is the operator?s responsibility to adjust the channel gain with the "trim" control and set subsequent levels for unity gain. Properly adjusting the gain structure of your mixing console is the key to consistently realizing its optimum sonic performance. Unfortunately, many system operators pay little or no attention to establishing a proper gain structure. Instead, they may experiment with trim or gain controls, channel faders, submix faders and master faders until the sound quality is not too objectionable. This trial-and-error approach invites higher than necessary noise levels or excessive distortion. Modern consoles brandish a bewildering array of features, with varying degrees of benefit. Just as a computer should be purchased to match the software requirements, a console should be bought for the needs of a particular environment. Current digital technology has given the audio community a very powerful tool with the advent of small digital mixers. Digital consoles offer an enormous amount of features and benefits compared to analog consoles. A digital mixer will let you store, and then recall, all of your fader, EQ, FX and processor settings with a push of a button.

        Record mixing

        TV and radio mixing

        • Mix Minus - "Mix-minus" is one of the terms most often used during the installation of a broadcast-to-telephone interface. Unfortunately, it's also one of the most confusing. This is a brief primer on mix-minus to help you avoid frustrations during your next installation. A production console with auxiliary buses that can be assigned to create a mix-minus will help very much on making radio talk shows.    Rate this link

        DJ mixers

        An audio mixer is a device that takes mutiple audio inputs and allows the user to blend them together for a single output. Typically a DJ would have a combination of turntables and CD players feeding into a mixer while the output is sent to an amplifier. Most standard DJ mixers come with (at least) two channels, a set of three equalizers for each channel and a crossfader to fade smoothly between each channel. Each audio source (eg. a turntable or CD player) is connected to one channel on the mixer. Each channel has a volume slider which controls the amount of volume that the source will output from the mixing board. There is usually also a LED meter to indicate how many decibels a channel is emitting. Knobs or sliders could be used used for sound volume control. The equalizers for a channel usually consist of three knobs that allow the user to adjust the low, middle and high frequencies (sometimes there is only one common equalizer for master output only in cheap mixers). Typical DJ mixers have a crossfader. The crossfader is one of the most important aspects of a mixer for DJs who want to perform tricks while mixing. A crossfader is designed to predictably control the outputs of two separate mixer channels based on the relative position of the fader's knob between its endpoints. Sometimes faders have extra option called "hamster switch" which reverse the polarity of the fader. Some advanced cross faders have curve adjustment option also. Typical connections for DJ mixer are that all normal audio input (phono and line) use RCA connectors. The outptu connectors typically also use RCA connectors and give line level output (some high-end DJ mixers have also balanced line level outputs which use XLR connectors). Generally you just plug the output of the mixer to your power amplifier input. The microphone connector in DJ mixer is most often 6.3 mma jack or XLR connector. Practically all DJ mixers have also built-in headphone amplifier for listening to the signal you are mixing or individual channels (or mix of them in some mixers with advanced cue options). Headphone output is typically 6.3 mm stereo jack (TRS).

        • Evolution of the DJ Mixer Crossfader - The DJ mider crossfader was originally developed as a control for implementing smooth fades from one program source to another by fading between two independent sources. The needs over years have somewhat changed and so have the implementations.    Rate this link
        • Hamster Switch Fer Dat Ass - Hamster switches reverse the polarity of the faders. In other words, when you flip the hamster switch on the crossfader, the right turntable is then on the left, and the left turntable is then on the right.    Rate this link
        • Hamster Switch Fer Dat Ass - Hamster switches reverse the polarity of the faders. In other words, when you flip the hamster switch on the crossfader, the right turntable is then on the left, and the left turntable is then on the right.    Rate this link
        • Introduction to DJ'ing and Mixing    Rate this link
        • Technical Secrets of the Crossfader - A crossfader is designed to predictably control the outputs of two separate mixer channels based on the relative position of the fader's knob between its endpoints. It's a simple sounding task but there are many different ways the job can be done, electrically and mechanically. This document describes some of the most commonly used ones. Most crossfader circuits are implemented in one of two basic schemes.    Rate this link
        • Turntablism: Frequently Asked Questions (FAQ) - The term "Turntablism" was first coined in 1995 by DJ Babu (Beat Junkies) to describe a form of advanced turntable music stemming from Hip-Hop DJ'ing. Turntablist is a person who uses the turntables not to play music, but to manipulate sound and create music.    Rate this link

        Taking care of mixer

      Compressors, limiters anf gates

      The audio compressor, is a pretty useful item, and one which you need to add to your system at some point if you are recording any type of audio, but especially vocals. The Compressor automatically adjusts and maintains the signal levels as they go to H/Disk or Tape to be recorded. If you use a normal compressor, nothing occurs until the threshold is breached. But when that happens, the compression cuts in. On a Hard Knee compressor, this full amount of compression (as set by the Ratio) is applied in full, as soon as the input level rises above the threshold. Lets say you have set a RATIO of 4:1, this means that compressor allows only 1db of signal level increase at the output, for every 4 db in input singnal level rise above the threshold setting.Soft Knee compressors apply compression gradually as the signal approaches the threshold level. As the input signal gets within about 10db of the threshold level, the Soft Knee compressor starts to gently apply compression, but with a very low Ratio, which increases proportionately as the Input level gets nearer to the Threshold setting, so that by the time the Input level actually reaches the Threshold level, the compressor is applying its gain reduction at the full level as set by the Ratio Control.Hard Knee compressor is the most commonly used compressor type.Some units allow you to switch between a Hard & Soft Knee function.


      Equalization means selectively boosting or cutting bands of frequencies to improve the performance of a sound reinforcement system. Equalization can do when used properly:

      • Improve the naturalness or intelligibility of a sound reinforcement system by emphasizing the frequency ranges most critical for speech (improvement is usually noticable but not dramatic)
      • Increase the overall output level of a sound reinforcement system by reducing the system's output in the frequency bands at which feedback occurs (helps somewhat but not very dramtically)
      En equalizer cannot make a poorly designed sound reinforcement system work satisfactorily. Every well-designed sound reinforcement system is subject to the laws of physics described by the Potential Acoustic Gain equation. Equalizartion cannot improve intelligibility problems caused by reverberation, reflections, mechanical vibration, high background noise levels, or other problems caused by the location or physical design of the room. Approach equalization gently and slowly! After every adjustment, listen carefully to the resulting sound. The goal in sound sound reinforcement system is to improve sound quality as well as increase the gain before feedback. When the system is loud enough and/or clear enough, stop equalizing! There is also other use for equalizers: improving sound quality on the sound playback systems. Well, take budget loudspeakers, for instance. An unfortunate truth regarding budget loudspeakers is they usually don.t sound verygood. This is often due to an uneven frequencyresponse, or more correctly stated, a non-flat power response.A conceptually ideal loudspeaker system has a flat power response.response.This is where equalizers can help overcomesuch response deficiencies. By cuttingfiltering a little here, and (less frequently)boosting a little there you can establish a very acceptable power responseand an improved sounding system.Equalizers when uses correctly can help to get rid of reasonable amount of frequency response problems in speakers. To solve the problems you need to know what is wring and how to correct it, otherwise the results can be worse than without equalizer. In stereo system you need to make identical equalizer adjustements to each loudspeaker. What you can do with an equalizer has it's limits: No equalizer equalizer will improve a speaker system performance to good if high multiples of +12 dB frequency boots are applies. There are some people who say that no graphic equaliser has a place in a very good system, because they usually introduce far more problems than they cure. In practice if the system is so good that it sounds good without equalizer, you don't need any. If there are problems, then it might be worth to try if using an equalizer to see if it can be used to make the sound better.The problem with multi-band equalizers is the filter topology that causes distortion of various kind. Any sort of filtering implies phasedelays, and the phase delay is determined by circuit topology. Depending the application in some cases those delay and phase problems can cause more problems than frequency response correction gives you.

      Use of delay lines in audio systems

      Simply put, a delay takes an audio signal, and plays it back after the delay time. The delay time can range from several milliseconds to several seconds. The delay is one of the simplest effects out there, but it is very valuable when used properly. A little delay can bring life to dull mixes, widen your instrument's sound, and even allow you to solo over yourself. The delay is the also a building block for a number of other effects, such as reverb, chorus, and flanging. Audio delays have also uses in large multi-speaker audio systems. Multi speaker sound can provide a greatly enhanced audio playback quality at large spaces when properly implemented. When designing any sound system, there are some standard issues that need to be addressed in order to provide a good listening experience to all seating areas. Many systems, whether they be single or multi-channel, can utilize the benefits of delay speakers. In a nutshell, delay speakers are generally designed to cover the rear portions of a room that are too far away from the main speakers to be covered at the same sound level as the front seats. The key to a good delay system is that the signal that feeds these speakers is delayed so that it coincides with the arrival of the direct sound from the main speakers. In fact, a good designer will add an additional bit of time to the delay processing so that the ear will actually localize up front to the primary speakers, even though the delay speakers may be louder. We tend to localize to the first arrival, even if it's slightly lower in level. So we add just a little additional delay to fool the brain, but not enough to make it sound like an echo or a separate source. The nice thing about delay speaker systems (if they are properly set up) is that they seem to be invisible. It is difficult to discern any sound coming from them until they are turned off. Once a sound system has to project to audiences greater than a few hundred feet from the stage, delay speakers become an important tool. These are speaker stacks that are (usually) positioned in the middle of the audience and pointed towards the crowd in the back. Delay stacks are used for several reasons:

      • They allow you to keep the sound levels directly in front of the stage at a reasonable volume, and not deafen those seated down in front, while getting the needed volume to the audience seated far from the stage.
      • The distance of the air attenuates high frequencies. Therefore, there will be a noticeable loss of brightness several hundred feet from the stage speaker stacks unless a properly positioned and equalized delay stack can correct for this effect.
      • There may be obstacles between the speakers and the audience, putting some of the listeners in an acoustic shadow of main speakers
      However, it.s not quite as simple as just putting in some additional speaker stacks and amplifiers a few hundred feet from the stage. In order to get the sound from both speakers to arrive at the listeners. ears at the same time, we need to add enough delay to the remote speaker so that it "waits" for the sound from the stage to catch up. Sound travels at approximately 1130 feet / 300 meters per second. At that speed, and given the way the human hearing system works, most people will hear a distinct echo when sound arrives from two sources that have a difference in their path length of more than 40 feet. This also equates to a difference of approximately 35 milliseconds or .035 seconds. This is critical for certain types of program material that is percussive or speech oriented. There are a few considerations when selecting delay units for delay stacks. First, pick a digital delay that's designed to do the job. Please note that enironmental conditions can effect the needed delay. Temperature affects the delay because as temperature rises, the air gets less dense. As you may remember from high school physics, the speed of sound of various media is dependent on density. So as air temperature rises, the speed of sound slows. This doesn't seem like a big deal, but at a few hundred feet from the stage, the needed delay can easily vary by 10 milliseconds or more from the morning to the afternoon. 10 milliseconds is very noticeable to many listeners. Dealing with delay stacks can be tricky, but they're often the only way to cover a large crowd while maintaining any kind of control. Delays do not answer all sound problems. At best, they enable you to adjust theaudience's perception of where the sound sources are, if you keep the leveldifferences within the limits of the Haas effect. Most engineers, set delays by ear (automatic solutions are rare and expensive). The process is usually the following: The basic maths for setting delays is really easy, back of theenvelope stuff: for a rough start, 1 ms per foot is close enough. Once youhave settings based on this dialed it, it's an easy matter to tweak on thefly. Start testing with a simple click or beep sound - maybe 500-Hertz pulses at 1-second intervals (or some clicking sound from microphone). First, position yourself very close and a little to the side of the delay speaker stack. This allows your one ear to hear the main speaker, and the other to listen to the delay stack. Now gradually bring up the volume in the delay stack until the volume level approximately matches the sound from the stage. At this point, you'll hear two distinct clicks, with the sound from the remote speakers happening first and the sound from the stage next. Now start adjusting the delay until the two separate clicks merge into a single sound. Now put on some real music with percussion and "walk through" the delays from the stage on out, listening for any echo zones as you walk. Finally, if possible, put on some program of the actual music style. Various music genres will need different adjustments to sound best. The technology to implement audio delays has advanced over years.Back in the prehistoric days of audio (before digital processing), audio people used various devices such as tape-delay loops and longs lengths of coiled plastic hose fitted with horn drivers and microphones. With the invention of advanced electronics making an audio delay at good quality at reasonable price has became possible. Electronics devices have used bucket brigade delay lines for a long time, but they have their limits (limited dynamic range and limited maximum delay). Modern digital delay instruments use A/D-converter, fifo memory and D/A-converter to implement the delay. Nowadays when the memory is cheap, it is easy to get long soudn delays with a very good sound quality.


      Complex effects are produced by the combination of simpler audio effects which can be easily implemented. These effects include delay, echo, reverberation, chorus, ring modulation, frequency shifting, vocal morphing, auto wah, flanging, distortion, and pitch shifting. A complex audio effect can be constructed from these fundamental audio effects by simply varying the amount of each effect applied. The result is the ability to model an environment or shape a voice with powerful effects to create a real-world experience or something that a normal physical world is not normally capable of making in any easy way. Some commonly used audio processing effects explained:

      • "Compression" is usually implies some sort of frequency independent amplitude control -- sort of an invisible hand that is constantly tinkering with the volume control according to some defined strategy.
      • "Loudness compensation" is usually static with respect to time and corrects for the level dependent characteristics of the human auditory system.
      • "Limiting" is a type of compression that happens suddenly. Everything below a threshold is passed unchanged. Above that threshold everythinging is "chopped". Limiting is often crude and very abrupt. While one shouldn't try to push this analogy too far, one could think of "Limiting" as a safety chain and "Compression" as an airbag. Limiting is very inexpensive to include in a design, often pennies or less.
      • "Filters" are digital filters for shaping of the audio spectrum.
      • "Delay" is an effect that just delays the sound a predefined amount of time. Delays can be experienced in acoustical spaces. A sound wave reflected by a wall will be superimposed on the sound wave coming from the source. If the wall is far away, such as a cliff, we will hear an echo. We can generate the same effect electronically by using a delay effect and mixing the original signal with the delayed one. There are many delay-based audio effects such as vibrato, flanger, chorus, slapback and echo.
      • "Echo" is a device used to simulate the sound of sound wave reflected by a wall will be superimposed on the sound wave coming from the source. An echo unit generally includes a delay unit and some kind of mixer that mixes the delayed and derect signal. Echo units generally have some for of feedback from the device output to delay input to simulate the situation of sound bounching back and forth between walls in the space, all the time attenuating.
      • "Modulators and Demodulators": Modulation is the process by which parameters of a sinusoidal signal (amplitude, frequency and phase) are modified or varied by an audio signal. In the field of audio processing modulation techniques are mainly used with very low frequency sinusoids. Wah-wah, phaser and tremolo are typical examples of amplitude modulation and vibrato, flanger and chorus are examples for phase modulations of the audio signal.
      • "Nonlinear Processing": Audio effect algorithms for dynamics processing, valve simulation, overdrive and distortion for guitar and recording applications, psychoacoustic enhancers and exciters fall into the category of nonlinear processing. They create intentional or unintentional harmonic or inharmonic frequency components which are not present in the input signal. Harmonic distortion is caused by nonlinearities within the effect device.
      • "Expander" is an effect opposite to the compressor. An expander widens the dynamic range of the incoming signal (for example to compensate the compression done on the other parts of signal chain).
      • "Noise Gate" is a device that only passes through audio signals that have higher volume than the set threshold. So if the incoming audio signal is stronger than the threshold, everyhting gets through as it is. If the input signal has lower volume than threshold, the noise gate does not give out any sound (=output is just complete silence).
      • "Warping": Time warping aims at deforming the waveform or the envelope of the signal. Frequency warping modifies its spectral content, e.g., by transforming an harmonic signal into an inharmonic one or vice-versa. Applications for those techniques are shifting inharmonic sounds, inharmonizer, extraction of excitation signals, morphing and classical effects.
      • "Automatic volume level control": Changing the volume level automatically is just slightly different than compression. Its fast attack and very long decay.
      • "Vocal Removal" are designed to remove vocal tracks from a stereo recording. Sometimes a vocal can be removed almost completely, but just as often the results are disappointing. In most cases you'll be able to reduce the vocal level, but some audible remnant of the original performance will probably remain. The idea is simple: You can reduce the level of a vocal (or other lead instrument) in a stereo recording by taking advantage of how vocals are generally recorded: in mono and placed centered in the mix. Since the vocal track is present in both the left and right channels equally, you can, in theory, remove it or at least reduce its level by subtracting one channel from the other. Instruments panned away from center will not be removed, although the tone of those instruments will probably be affected.
      • "Reverb": Reverberation is the result of the many reflections of a sound that occur in a room. It's very tempting to say that reverb a series of echoes, but this isn't quite correct. 'Echo' generally implies a distinct, delayed version of a sound, as you would hear with a delay more than one or two-tenths of a second. With reverb, each delayed sound wave arrives in such a short period of time that we do not perceive each reflection as a copy of the original sound. Even though we can't discern every reflection, we still hear the effect that the entire series of reflections has.
      • "Phasing": The phase shifter (or phaser) achieves its distinctive sound by creating one or more notches in the frequency domain that eliminate sounds at the notch frequencies. The notches are created by simply filtering the signal, and mixing the filter output with the input signal. The filters can be designed so that we can independently control the location of each notch, the number of notches, and even control the width of the notches. This can lead to many interesting sonic possibilities.
      • "Ring modulator": A ring modulator is a simple device that can be used to create unusual sounds from an instruments output. It effectively takes two signals (each with some frequency), and produces a signal containing the sum and differences of those frequencies. These frequencies will typically be non-harmonic, so the ring modulator can create some very dissonant sounds. For this reason, ring modulation is not a widely used effect.
      • "Chorus": Just as a chorus is a group of singers, the chorus effect can make a single instrument sound like there are actually several instruments being played. It adds some thickness to the sound, and is often described as 'lush' or 'rich'. Chorus is implemented with a delay line that is used as a variable length delay line (delay time changes over time).
      • "Flanger": Flanging has a very characteristic sound that many people refer to as a "whooshing" sound, or a sound similar to the sound of a jet plane flying overhead. Flanging is generally considered a particular type of phasing. Flanging is created by mixing a signal with a slightly delayed copy of itself, where the length of the delay is constantly changing.
      Traditionally the effects have been simple analogue electronics, but DSPs have changes this largely over last decade. With advancements in the capabilities of DSP we can tackle increasingly complex tasks that cannot be performed by analog components. The entertainment industry has fully taken advantage of this fact through its development and use of digital audio effects.

    Connectors, connections and wiring

    There are many different connectors and wiring practices used in audioword. The most common connectors used are:

    • XLR: An XLR is a quite larger (about 5cm long, and 2.5 cm diameter) with (generally) 3 conductor pins (or recepticles) in a triangular pattern shrouded by the cover. As used on almost all pro audio equipment to carry balanced audio signals. XLRs are most typically used in microphone circuits, and PA system cabling. XLR connectors are commonly used in professional audio systems microphones and equipment interconnections. The audio signals are transported as electrical signal between pins 2 and 3. Pin 1 is used for shield ground. The origin of the XLR connector was the Cannon X Series connector. It was fitting the demands of the audio community except the missing latch. Cannon rearranged the pins and added a latch.
    • RCA: The RCA (sometimes called also Cinch) type is the regular consumer type used for unballanced audio. RCA is a lightweight small coaxial connector, with a centre pole that sticks out a little further that the shield flangy ring thing, and is quite small. The signal goes between a center conductor, and the shield or return side, which is usualy referenced to the case or outer sleeve section. The signal carried in this connector is usually consumer line level or sometimes low level signals from LP player. Practically all RCA connectors that are prone to noise problems, this is probably the number one source of bad connections. Typical consumer AV equipment (like VCRs), may have an audio source impedance of up to 5-20 Kohms, which cna cause problems with long cables (high frequency rolloff and easy pickup of noise). NOTE: RCA connector was originally intended for use at RF *inside* equipment and racks. Never was intended for audio with that long signal pin which mates before the shield!
    • 6.3 mm PHONO: This is a connector type original used for manual telephone patch panels. In audio world this connector is used for patch panels, equipment interconnections, some microphone connections and headphone connections. The stereo version of 6.3 mm (1/4 inch) PHONO plug is used to carry, depending on aplication, stereo headphone signals or balanced line level signals in equipment interconnections. Mono 6.3 mm PHONO plug is generally used to carry unbalanced line level audio signals in audio equipment interconnections. In some applications the same connector is also used for microphone level signals generated by microphones or instrument pickups (for example in electric guitars). In some applications 6.3 mm phono connector is used to carry speaker signals (not very recommended practice).The common ? inch stereo phone plug was originally designed by the Bell Telephone Company around 1880, for use on telephone switch boards. That is why it is called a phone plug.
    • 3.5 mm PHONO: This is a miniature version of PHONO connector. 3.5 mm (1/8 inch) stereo PHONO plug/jack is commonly used in portable CD players, small radios and PC soundcards to carry stereo headphone signals or line level audio signals. In PC soundcards this connector is also used for mono electret microphone connections where that connector carries micrphone signal and microphone bias voltage.
    • 2.5 mm PHONO: This is a very tiny version of PHONO plug. It is used in some applications to connect microphones to wireless transmitters or video cameras. The most commonly used version is mono version, but also a stereo version of this connector. The wiring of the connector can vary from equipment to equipment but is on same general line as other PHONO connectors.
    • BATAM: This connector looks somewhat like a stereo PHONO jack which has a size between 6.3 mm and 3.5 mm PHONO jacks. This connector is used in some professional audio patch panel applications to carry balanced audio signals.
    • TT: This connector looks somewhat like a stereo PHONO jack which has a size between 6.3 mm and 3.5 mm PHONO jacks. This connector is used in some professional audio patch panel applications to carry balanced audio signals.
    • Banana plugs: 4mm banana plugs are very traditional speaker connectors. Banana plug is good connector for speaker signals, both mechanically and electrically. You can see those connectors on many audio amplifiers and speakers. Banana plug has it't problems. First you need two separate connectors to connect one speaker cable. Secondly the electrical safety is a problem, because exposed banana connector connected to powerful can have dangerous voltages on it (speaker signal can be easily tens of volts in amplitude) and on some countried the banana plug is too similar to pins on mains power plug (especially in European countries). Those safety problems have caused compliance problems in European countries (CE marking). Nowadays there are also CE compliant innovative banana plugs that are approved worldwide as a loudspeaker connector, but the audio industry seems to be moving away from banana connectors (mostly to use Speakon connector for speaker connections).
    • Spade: High current spade lugs are sometimes used to terminate speaker cable. A spade lug can be easily connected to screw type terminals on speakers (those same terminals can also take bare wire).
    • Speakon: A type of connector used for speaker connections developed by the company Neutrik. Has four or eight contacts (depending on the model). The plug locks into the jack so it can't be pulled out. It has become as widely accepted as the standard professional speaker connector due to its electrical handling and mechanical features. Neutrik NL4FC 4-Pole Speakon Connector is nowadays propably the most commonly used speaker connector type used in professional audio world (there is also 2-pin version of this connector with same mechanical size as 4 pin). All Speakon Connector contacts on both connectors are touch proof, so the the connectors meet strict safety requirements. Speakons are designed for high-power use, they can handle 250V voltage and 20A continuous current, os they are more than adequate for even highest power audio systems. Speakon connectors are "non-shorting".
    • Other connectors: There must be 50 different random audio connectors on the market. They are designed for special market. Those special connectors are most often used with multi-pair cables (there is no single standard for those) and for some application where "standard" connectors are suitable (for example miniature connector versions in very small equipment).
    In audio world there are two different interface types: unbalanced and balanced. The unbalanced interface is simple wiring/interface with one signal wire and shield around it which acts like a common ground reference for both systems connected. The normal hifi system intercnnection cables (usually terminated with RCA connectors) are unbalanced connections. Unbalanced connection is simple and inexpensive, but not problem free. Unbalanced connections can quite easily pick up interference, especially ground loop interference. Unbalanced interface is generally adequate for small home hifi systems, but large professional sound systems need something better, because unbalanced cables and are very prone to picking up noise, especially at low signal levels from devices such as microphones.

    Most professional audio devices are connected via balanced interfaces and cables to minimize pickup of stray electrical noise. Balanced circuits have an inherent ability to only pass audio signals and reject unwanted noise. Balanced refers to the fact that there are two symmetrical signal lines and one ground, while unbalanced uses just one signal line in reference to ground. Normally, XLR connectors are used in most balanced devices while unbalanced consumer gear normally use mini-plug connectors. The purpose of a balanced line is to transfer a "signal" from one place to another while rejecting "ground noise", which is not whitenoise or hiss, but power line related hum and buzz. To accomplish this noise rejection, two signal lines are used and the impedance of the two lines to ground must be equal or "balanced" same impedance. Typically those two lines have the same audio signal 180 degrees out of phase with each other. It is a popular belief that the signals must have opposing polarity and equal amplitudes, or symmetry. Signal symmetry has nothing to do with noise rejection (as long as the signal is sent as difference between two signallines the system). The noise induced on the two lines can be cancelled out at the other end by taking the signal difference of the signal from two signal wires. This difference is clear audio signal, because the noise coupled to the cable is common mode noise between those two signal wires (twisted pair wiring and balanced impedances make sure that both electrostatic and magnetic noise is couled simularly to both signal wires, this makes is common mode noise).

    A balanced system must rejects noise even when there is no signal and, infact, this is usually how system noise testing is done. The ground in balanced line simply provides a current return path and a shield ground. Theoretically balanced lines work well well without ground connection, but in practice the real life system work better with the ground in place. Balanced systems can be used without signal ground with somewhat reduced noise cancelling properties (a typical telephone line uses balanced audio on unshielded twisted pair). Input stages to GOOD mixers use high quality (i.e expensive) transformers to match and isolate the balanced circuit. Many less expensive mixers use very precise differential input amplifier configuarationin the sound input to do the same as transformer have traditionally done (some opamp based this kind of circuits are better than some other). The technology of balanced-line audio wiring is quite trendy today in professional audio world, and there is quite a bit of information (and misinformation) in the popular press about it. Balanced-lines do not enjoy any magical properties. They do have some potential advantages for some systems which could justify the moderate extra cost and complexity involved in their implementation. Using balanced interconnection can reduce any system noise caused by ground loops, RF, power lines etc. Balanced connection will not increase the slew rate of an audio system, affect cable properties, improve distortion or improve the individual audio component noise.

    It is possible to use balanced mics on unballanced amps and vice-versa. You can buy transformers to convert between the two. They also normally have animpedance change switch, to allow matching to different impedances. In some applications just direct wiring will work (but not always).

    Impedance is a necessary measure on many audio interconnections. For most audio applications impedance is assumed to be measured at 1kHz frequency. Most current audio equipment (both pro and consumer) has low output impedances, frequently lower than 600 ohms. Pro equipment frequently hasoutput impedance of 100 ohms.Most modern audio equipment (pro and consumer) has input impedances much higher than 600 ohms, typically greater than 5-10 kOhms. "600 ohms" is an old (approaching 100 years!) standard related to telephoneline impedance in our great-grandparents' era. Some pro audio equipment isdesigned with switchable artifical loads that make their input impedance 600ohms for interfacing with older audio components. 600 ohms matching is not used in modern professional audio equipment much, although you might still see 600 ohms on some technical specifications. A common rule of thumb is to assume consumer output impedance is around 1 kohms, and input impedance is around 10 kohms. Consumer, single-ended, mic-level inputs tend to be around the range of 1-5K, frequently with bias voltage to operate electret condenser capsules(i.e. computer sound-cards, consumer camcorders, etc.)

    The most common cable types used in audio connections are:

    • Shielded twisted pair: This cable type is most generally used to carry balanced microphone signals and balanced line level signals (works also for unbalanced). Multi-pair cables (like "snake cables") used to carry many audio signals generally consists of many shielded twisted pair wires inside one large cable. Balanced connections and good shielded twisted pair cables are needed to transfer audio signals for long distances. Shielded twisted pair cable good cable to carry both balanced and unbalanced audio signals. Typical this type of microphone cable has around 70-150 ohm/km resistance on conductors, typically around 50-70 pF capacitance between the conductors and around 90-130 pF capacitance from conductor to shield.
    • Shielded single conductor: This cable type is used generally to carry unbalanced audio signals. This cable type has one centdal signal 1 conductor and a ground shield around it (coaxial construction). This cable type is sometimes called "high impedance cable", because typically used with high output impedance equipments like home hifi equipments (tape, CD player, phono player), musical instruments (electric guitar) and some microphones (for example hobby recording and computer microphones). Shielded single conductor cable is optimal for unsielded audio signals. It is not suitable for balanced audio signals.
    • Star Quad: Some mic cables use "Star Quad" wiring where there are actually four, rather than two, signal conductors; they are intricately braided together and then paired up at the ends so that they behave like two conductors that are very close together physically. Star Quad increases the capacitance, but it reduces noise.
    • Unshielded twisted pair cable: This is the cable type used in telephone and modern network wiring (structured cabling systems). This cable type is not very suitable for professional audio use. With suitable adapters this can be used to carry digital audio signals and also analogue audio signals with limited performance. You can interface balanced audio signals directly to unsielded twisted pair with quite usable performance. Unshielded twisted pair cable is not suitable for unbalanced audio because such cable will pick up easily lots of interference (=poor shielding against interference).
    • Unshielded wire pair: This cable type is used for speaker connections from the power amplifier to the speaker cabinet. This kind of cables has generally thick copper wires in them to keep their resistance low (avoid power losses and effects to sound quality). Speaker cables generally do not need shielding in a form of twisting or external shield layer, because the speaker signal levels are very high (easily from many volts up to tens of volts) and low impedance (amplifier output impedance is fraction of ohms, speakers typcally 4-8 ohms), so so they are not suspectible in picking audible noise.
    • 75 ohm coaxial cable: This cable type is used to carry digital audio signals that are in S/PDIF (IEC60958) and AES/EBU 75 ohm/BNC version (AES-3id-1995 standard) formats.
    • 110 ohm shielded twisted pair: This cable type is used to carry digital audio signals that are in AES/EBU format.
    Sometimes you might see another kind of cable type division:
    • Speaker cable: Speaker signals are high power, low impedance, unbalanced. Because they're high power, interference and hum are very small by comparison, so the wires can be unshielded, and triboelectricity is not a problem. The ideal speaker cable has big, low-resistance conductors, with low capacitance between them. Shields are unnecessary.
    • Microphone cable: A mic signal is very low power, low impedance, balanced. It is quite susceptible to external interference, which is why it's balanced. Because the source and load are low impedance (and mics aren't as susceptible to instability as amps are), capacitance is not a big problem; but anything you can do to help balance the signal will reduce noise. A typical mic cable is a shielded twisted pair cable. Some mic cables use "Star Quad" wiring where there are actually four, rather than two, signal conductors; they are intricately braided together and then paired up at the ends so that they behave like two conductors that are very close together physically. Star Quad increases the capacitance, but it reduces noise. The ideal mic cable has at least two and maybe four signal conductors, plus a shield. Capacitance and triboelectricity are not problems.
    • Instrument cable: An instrument signal is low power (but more than a mic), high-impedance, unbalanced. Like a mic signal it is susceptible to external interference, but different sorts: it is more sensitive to things like fluorescent lights and neon signs, less sensitive to motors and power lines. Because the source and load are high impedance, any capacitance in the cable creates a low-pass filter. That is, it reduces the high frequencies in the signal. For that reason, you want low capacitance. But also, the high impedance means that triboelectric signals, which would get drained away by low impedance, can be a problem: when you wiggle the instrument cable you can hear noise from your amp. To deal with this, manufacturers add layers of intermediate insulators that are actually somewhat conductive. The ideal instrument cable has one signal conductor, surrounded by a shield, and is fairly low capacitance. Triboelectric shielding is useful.
    Snake cables are used to connect multipleaudio channels in low-level (microphone)and high-level (line) componentry such asconsole board equipment for recordingstudios, radio television stations, postproductionfacilities, and sound system installations. A snake cable is s flexible cable with many individually shielded twisted pair cables inside. Snake cable is commonly terminated to many XLR connectors, a stage box with many connectors in it or to a multi-pin connector that connects to equipment rack. The thickness of wires used in snake cables can vary somewhat, but typically the wires itself are 22 AWG or 24 AWG sizes.

    For installed jobs, where you must have an approved cable type (NEC approved in USA, other approvals in other countries), choose your preferred cable type from the approved cable types. For non-installed jobs, where usually cable flexibility is more important than rating, you can usually chooseyour preferred cable type more freely. z

    It is a common misconception that cable is either "digital" or "analogue". In fact any decent double-shielded coaxial cable of the correct impedance will cope with both types of signal up tovery high frequencies. Because digital signals are fast changing signals, the cable must have right impedance for the used system. 75 ohms is the right impedance for coaxial digital audio connections (S/PDIF). For AES/EBU audio connections a 110 ohms (+-20%) shielded twisted pair cable is recommended (preferably impedance in 100 ohms to 120 ohms). Standard analog audio cable impedance is typically in range of 45 ohms to 70 ohms, so it is not generally the right cable for carrying digital signals designed for other impedances. A cable designed for digital signals in mind generally carry also analogue audio well if needed.

    The three most important electrical components of wire are resistance, capacitance and inductance. High resistance will decrease the audio signal (especially affects speaker connections). High capacitance will roll off high frequency response (espically if used with equipment that has high output impedance). High inductance can alter the tones in various ways, depending on the circuit thy are inserted into.

    Transmission line effects are essentially irrelevant in audio wires at analogue system. At audio frequencies (20Hz..20kHz) Transmission line effects are essentially indistinguisable from lumped-parameter behavior of the cable. For transmission line effects to dominate over these more normal behaviors, the cable would have to be very long (for transmission line effects to be significant even at 20,000 Hz,the cable would have to have length of many kilometers). Transmission line effect (like characteristic impedance)have effect only on digital audio transmission. The notion of cable impedance has no validity at audio frequencies until the cable becomes truely long, as on the order of miles/kilometers. In practice the cable impedance is only relevant in transmission of digital audio signals (which is high frequency digital signal).

    A cable in audio applications for carrying microphone and line level signals can be modeled as a low-pas filter. A first-order high-cut (or low-pass) filter is formed by an output's source impedance and the capacitance of the cable. The frequency at which a filter attenuates 3 dB is called its "corner frequency". With short cables (low capacitance) and low output impedances, the corner frequency typically occurs well above the audio band. With longer cables and higher output impedances,the corner frequency drops and may drop into the audio band. The formula for the corner frequency is:

    F = 1 / (2 * PI * R * C) 
    where F is the corner frequency in Hz, PI is 3.14, R is the source's output impedance in Ohms, and C is the cable's total capacitance in Farads. A first-order filter has a slope of 6 dB per octave. This means that beyond the corner frequency, the response will drop 6 dB for each doubling of frequency. Generally it doesn't seem likely that you would get detectable loss even at 20kHz unless you have one or more of these conditions: unusually high source impedance (many kilo-ohms), unusually high capacitance cable or unusually long cable length (tens of meters).O ne meter of typical shielded audio interconnection cable (RCA cable) has typically capacitance of around 100 picofarads.

    Impedance is an electrical term that refers to how much a device impedes the flow of current and is measured in ohms. There are different impedances used in audio interconnections. Usually the most often heard terms are "high impedance" and "low impedance". While there is no set standard, low impedance usually refers to a range of between 150 and 800 ohms. Most professional audio microphones are low impedance. High impedance generally refers to impedance from few kilo-ohms to tens kilo-ohms. Impedance, simplified, is the resistance in the circuit to audio signals. Each audio input has an impedance, as does each output. In older products, transformers were typically used between equipment inputs and outputs, and matched impedances were critical to transferring power from an output to the next input. This system used typically 600 ohms impedance. Development of present audio circuit techniques permits much lower output impedances, and input circuits which require almost no audio input power. These newer input stages are typically high impedance, often 10 kOhms or higher. These inputs amplify the input voltage without demanding power from the prior equipment output. For accurate voltage amplification (so power can be ignored), it is generally accepted that an audio input impedance should be at least 10 times the output impedance of the prior device.

    The impedence matching in the analogue audio electronics world in a very simple context is the following: You can drive a high impedance load with a low impedance source. You cannot drive a low impedance load with a high impedance source. If you try to drive a low impedance load with ahigh impedance source it acts like a shorting circuit, drawing too much current. In the case of audio this will make the sound lower and most likely a bit tinny sounding, sometimes distorted. There are a lot of complicated reasons for this to happen.

    Properly grounding all equipment is important to ensure noise free operation of audio systems. In some specific cases also ungrounded system can work quite satisfactorily (many small home hifi systems are ungrounded). Following proper grounding procedures ensures the best possible audio in every situation. A ground loop is a an annoying thing which can cause humming noise headaches for audio system installer. A ground-loop is created whenever two or more pieces of mains-powered equipment are connected together, so that mains-derived AC flows through shields and ground conductors, degrading the noise floor of the system. The effect is worst when two or more units are connected through mains ground as well as audio cabling, and this situation is what is normally meant by the term "ground-loop". Typical ground loop occurs when you connect two grounded audio equipments powered from different power outlets in different rooms together. Other common situation is a system where there are groudne equipment and connections to house central antenna networks.Please note that ground currents can also flow in systems that are not galvanically grounded; they are of lower magnitude but can still degrade the noise floor.

    Sometimes questions should I ground XLR connector shell (= connect to pin 1) or not. Generally the advice is to not connect the XLR shell. The shell of an XLR is going to get connected to the chassis (and thereforethe chassis ground) of whatever it plugs into.. hence the general view that the shell does not get connected to any of the wires/shields/etc. in a normal balanced XLR cable. This keeps signal and chassis ground separated andavoids propagating problems with chassis ground of one component into the next component. If the XLR shell were grounded,every where the XLR shells in a multiple cable run touch ground/truss/scaff etc. introduces a potential earth loop. There are also anternative view: every where the shell isn't connected supplies a HF EMC gap (makes wiring more suspectible to RF noise; the slight riskof RFI problems). Thats why there are two schools on this subject. Personally I leave the case floating on the grounds that the gap in the shielding is too small to form an effective aerial at any frequency below 500Mhz. My recommendation is not to ground the shells. If in special cases RF noise becomes a problem, then try connecting the XLR connector shield to pin 1 (ground) through a small ceramic capacitor (100pF-10nF) to reduce RF-pickup.

    How about using UTP cable to carry audio? Unshielded twisted pair is suitable cable to carrybalanced signals (balanced audio, 10/100Base-T Ethernet,telephone, etc.), but is far from optimal for unbalancedsignals (like home hifi audio interfaces with RCA connductors).To properly transfer unbalanced signal over UTP thesignals need to be balanced (there are baluns for this). If you are carrying unbalanced audio signals through someshort distances, I recommend you to use a cable with coaxial construction (typical shielded audio cable)or use shielded twisted pair cabling (best cable for balancedaudio, works well also with unbalanced signals).

      Useful wiring accessories

      • A Direct Box can be inDIspensible - DI-box converts high impedance signal to a low impedance and converts unbalanced signal to balanced    Rate this link
      • Using patch panels properly - This document is an explanation of how to use the Neutrik Patchlink SPL (NYS-SPP-L) 1/4-inch balanced patch panel in basic studio applications. It also explains offers four possible configurations of its jack modules. A lot of the information presented here applies to many of the popular patch bays on the market.    Rate this link

      Routing audio through computer networks

      • RAVE (Routing Audio Via Ethernet) Application Notes - RAVE is a signal transport system that allows you to route multiple channels of audio over standard Ethernet hardware and cabling. A single RAVE network can now replace hundreds of analog audio cables, dramatically reducing installation time, effort and cabling costs while improving routing flexibility and audio performance.    Rate this link


    Microphones just convert a real sound wave into an electrical audio signal. In order to do so, they have a small, light material in them called the diaphragm. When the sound vibrations through the air reach the diaphragm, they cause the diaphragm to vibrate. This in turns will somehow cause an electrical current in the microphone to vary, whereupon it is sent out to a mixer, preamplifier or amplifier for use. There are a wide variety of microphones available, each with differing construction and response.

    Each type has its own characteristics, and hence its ideal applications. Microphones are typically classified according to how the diaphragms produce sound. Dynamic microphones (some call those also "dinamic microphones") typically use moving-coil technology. As the diaphragm vibrates, the coil connected to diaphragm vibrates, and its changing position relative to the magnet causes a varying current to flow through the coil. Dynamic (so called) mics have magnetic transducers and are generally in therange of 100 to 600 Ohms impedance. They should ideally see a load impedance of grater 5 times their own source impedance (typically few kilo-ohms). Sometimes a transformer is included to step up the impedance (and level) for use with older designs of guitar amplifiers and cheaper PA amplifiers. These transformers, if external to the mic, must be connected at the amplifier end of the cable. Well-made dynamic mics are generally regarded as the most robust microphonesare are very often used in stage by musicians on the stage. Dynamic microphones available with different directional characteristics (known as polar patterns). Most popular ones are unidirectional, cardoid and hypercardoid.

    Robbon microphones consist of a thin ribbon of a metallic foil suspended in front of a metal plate. Sound waves cause the foil to vibrate, causing fluctuations in the electrical current. Thus, an electrical audio signal is created. Ribbon microphones are very low impedance signal sources.Ribbon mics generally have transformers to raise their very low impedance tothat of a dynamic mic (100-600 ohms). Ribbon mics are rarely encountered in domestic equipment.

    In condenser microphones, a static charge is impressed on the diaphragm or on a back-plate to the diaphragm. As the diaphragm vibrates, the distance from the back-plate to the diaphragm vibrates, altering the capacitance of the diaphragm and the back-plate. This fluctuating capacitance results in a fluctuating electric current. Condenser microphones need a source of power to impress the charge on the capacitor. This can be provided via an internal battery, or by phantom powering. Phantom power is conventionally understood to mean that thepower is supplied over the same wires as the audio signalitself. This technique uses the audio wire and screen of the connecting microphone cable to send a 48Vdc power supply (can vary from 9V to 52V depending on the application). Capacitor mics (which require external or 'phantom' power supplies) are rarely encountered in domestic equipment, but are videly used for professional sound recording. A typical phantom powered microphone can take up to around 7 mA current from 48V power source and many microphones take considerably less power. According IEC specifications the maximum allowed phantom power current is 10 mA.

    Electret microphones are a variant of condenser microphones that mostly utilise a permanently charged diaphragm over a conductive metal back-plate. Electret mics are also capacitive sources requiring very high impedance loads but are invariably provided with in-built FET impedance converters.Their output impedance is typically in the range of 500 to 5000 Ohms and they like to be loaded with at least their own impedance. Widely used electret microphone capsules tend to be small, even minuscule, cheap and light. Electret capsules operate typically at 1 to 10V power and consumer power from fractions of milliamperes up to few milliamperes. Back-electret microphones use a charged back-plate instead of a charged diaphragm. These may or may not be phantom powered. Most typically electret capsules are powered from around 5 volt "bias voltage" power supplied by wireless mic transmitter, portable recorder (like DAT & MD), or computer sound cards. There are also electret capsule based microphones that use internal battery or phantom power as their operating power.

    Electret and back-electret microphones have special preference for voice communication, where clarity of speech is essential at the sacrifice of perfect sound reproduction. For low cost applications, electrets offer the highest sound quality andhave quite a high output, easing the need for a high quality preamp. They can be obtained in omnidirectional or directional models for different applications. You can find electret microphone inside many modern electronics devices which have built-in microphone (video cameras,telephones, cellular phones, voice recorders, computer microphones). In professional audio work electret capsules are used in some(semi) professional microphones and in small lavalier microphones(those which clip to the clothes are are taped to the skin of an actorin theatre). Electret microphones are also used in many audio measurement applications, because there exist electret microphones which have very good response (both frequency and impulse response) and are physically small (small microphone does not distort the sound field that is beign measured too much).

    An electret MIC is usually the best value for money omnidirectional microphone you can buy. Those microphones are used in many applications where small und inexspensive microphones with good performance characteristics are used. These microphones have typically a very naturally crisp sound, providing deep bass, smooth mid-range and clean high-range with a very flat frequency response. Electret microphone is a modification of the classic condensor microphone. Whereas a condensor mic needs an applied phantom power, the electret condensor has a build in charge. The bias voltage of arround 1-10V is needed to supply the build-in FET buffer and should be applied using a 1-10 kOhm resistor. Normally an electret capsule is a 2 terminal device that works like a current source when biased. The bias voltage should be kept clean, because the noise in this will get to the microphone output. This bias voltage to microphone is most often applied as "plug-in" power. Electret microphones can usually be easily plugged into any minidisc, dat or analog recorder that supplies a bias voltage of between 1.5 to 10 volts D.C. (also known as plug in power, typically 3 to 4 volts DC) at the microphone input jack. Typical PC soundcards also supply bias voltage (usually around 5 volts DC) to to microphone, but in them then power supplying is wired differently than in devices like minidisc recorders (this means that electret microphones need wired differently when connected to PC soundcard). In professional applications electret microphones are powered through phantom power from the mixer (those microphones have some extra electronics in them which converts the phantom power to voltage needed by electret capsule and balances the unbalanced signal which electret capsule sends out).

    Ceramic and crystal mics are high impedance and appear to be capacitive sources. Some of the signal level is lost due to the shunt capacitance ofthe cable. The input impedance of the equipment needs to be at least 1Megohm to avoid loss of bass. They are rarely supplied with new equipment nowadays. In some applications small piezoelectric pickups are used to pick up the sound of some instruments directly from the case (for example acoustic guitar). Those piezoelectric pickups are a type of crystal microphones that are directly glued/taped to the instrument. Crystal microphones and piezo picups are designed to be connected to a high input impedance (1 Mohm or so) microphone amplifier (to match the very output impedance of the pickup). The effect of this high impeance amplifier is to eliminate much of the "tinny" sound, which you get if you connect crystal microphone to a normal microphone input.

    Carbon granule microphones are found in many older telephones and some communications radio applications. The vibration of the diaphragm alters the resistance of current passing through the microphone, creating an audio signal. The sould quality of this kind of microphone is poor and it has only been used widely in telephone handsets and similat voice applications. The audio signal from this type of microphone can be picked up by looking at the modulated current from the element or the voltage over the microphone element when some known low current (typically few milliamperes) is fed through it.

    Please remember that all microphones are made with certain applications in mind. Microphones are not always expected to pick up sound universally and from all directions. The way that a microphone picks up sound from various directions is known as its pickup pattern. There are a few standard pickup patterns: Omnidirectional, Unidirectional, Bidirectional and Cardioid. Pickup patterns are usually depicted as polar diagrams, a circular graph of sensitivity of a microphone from various directions.Omnidirectional picks sounds from all directions equally well (or almost equally well). Other microphone types have more or less directional pick-up pattern (they pick sound better from some directions than other).

    Most microphones are not "flat" in frequency response and some are better suited for certain jobs than others. Some microphones have intentionally non-flat frequency response to get the wanted sound from the microphones. Examples of such microphones are some vocal microphones that add "precense" and "power" to the sound then used compared to a "flat response" microphone. With directional microphones keep in mind that the distance of sound source from microphone can affect the frequency response: the frequency response can be greatly affected when the sound source is very near to the microphones (for example the low frequency response of vocal microphone can change considerably when microphone is moved very near to sound source).

    There is difference of the microphone inputs used in different equipment. For stage and studio use, balanced wiring is preferred to minimise interference in long cable runs. Connectors are usually 'XLR-type', 3-pole. Mic impedance on professional mics is normally specified as 200 to 300 Ohmsand the input impedance of the equipment (usually a mixer) should be atleast 1500 Ohms with transformer or electronic input balancing. Usually this kind of microphone inputs are included in the professional audio mixers. This type of professional microphones with balanced wiring have been known to do 300 meters of cable without problems.

    The majority of domestic equipment will have 'medium input impedance' which will accept the output from either electrets or dynamics, subject to the gain range available. Domestic equipment generally has an unbalanced micinput on a 1/4" 2-pole jack or miniature jack. Those devices which use miniature jacks are their own story, and it is best to check the manual of the equipment on what type of microphone do they take. Consumer, single-ended, mic-level inputs tend to be around the range of 1-5K in impedance, frequently with bias voltage to operate electret condenser capsules(i.e. computer sound-cards, consumer camcorders, etc.)

    In some applications information on microphone sensitivity is needed. A microphone sensitivity specification tells how much electrical output (in thousandths of a volt or "millivolts") a microphone produces for a certain sound pressure input (in dB SPL). If two microphones are subject to the same sound pressure level and one puts out a stronger signal (higher voltage), that microphone is said to have higher sensitivity. "Open circuit" means the microphone is not connected to anything. That is, there is no electrical load on the microphone. The open circuit voltage rating indicates how much voltage appears at the microphone output when a certain SPL is introduced to the microphone diaphragm. A value for a typical dynamic mic is -75 dBV/microbar. The "V" in dBV indicates the microphone output level is referenced to 1 Volt. If there was a microphone with an output voltage of 1 volt, its level would be given as 0 dBV. Microphone manufacturers normally specify one of two dB SPL input levels: 74 dB SPL or 94 dB SPL ("dB SPL " is a measurement of Sound Pressure Level).A value for a typical dynamic mic is -75 dBV/microbar. (-75 dBV converts to .00018 volts). The "microbar" part indicatesthe microphone was tested with an input of 74 dB SPL. To compare this typical dynamic microphone's sensitivity with a different microphone that was tested at 94 dB SPL or 1 Pascal, simply add 20 dB to the rating: -75 + 20 = -55 dBV/Pascal. Remember, to compare specifications from different manufacturers, make certain that each has been converted to the same input dB SPL level. Sometimes microphone sensitivity is expressed in decibels where 0 dB = 1V/Pa at 1 kHz frequency. I have also seen figures where decibels are referred to 0dB=1V/0.1Pa @ 1KHz (this was seen on one electret microphone that had sensitivity rating of 64 dB on that scale).

    There are microphone types that need electrical power to operate. The most common way to power microphones connected to professional mixers is to use a technology called phantom power. Phantom power technology is not fixed. The 1966 DIN 45596 standard well defined phantom power as: 48V, 6k8 resistors and a maximum current of 2mA. That worked well, until some manufacurers started to build microphones that needed tripple that current or more (those worked well with most pre-amps but not all). There came a new standard IEC 61938. Phantom powering as it is known today is defined by the international standard IEC 61938 for 12-Volt, 24-Volt and 48-Volt implementations. In Europe, it is known as EN 61938. The 24-Volt implementation has never been widely adopted by equipment manufacturers due to a chicken and egg problem (why build consoles and preamps to work with microphones that don't exist, and vice versa?), leaving just the 12 and 48 Volt versions ("P12" and "P48") in practice. The standard has a long history and is well established. Many modern microphones (but not all) are designed to work well with a wide range of voltages e.g. 9-52V.

    There are also other powering methods used in comsumer devices for powering electret microphones. The most common approaches are "PC multimedia microphones" and "plug-in-power approaches.

    Sound Blaster soundcards (SB16,AWE32,SB32,AWE64) from Creative Labs started the "PC soundcard multimedia microphone" trend by using a 3.5 mm stereo jack for the electret microphones. The connector carried audio signal on the connetor tip and the microphone power on the connector ring contact. The microphone power was voltage source current limited with 2.2 kohm resistor. Many other manufacturers copied this idea and make quite similar systems to their sound cards. There has also been standardizing work going on this connector also. For example PC99 standard mentions the PC soundcard microphone interface details: Three-conductor 1/8 inch (3.5 mm) tip/ring/sleeve microphone jack where the mic signal is on the tip, bias is on the ring, and the sleeve is grounded. This design is optimized for electret microphones with three-conductor plugs, but will also support dynamic microphones with two-conductor (ring and sleeve shorted together) plugs. Minimum AC input impedance between tip and ground: minimum, 4 kOhm; recommended 10 kOhm. Input voltages of 10.100 mV deliver full-scale digital input, using software-programmable .20 dB gain for low output microphones.Bias should be less than 5.5V when no input and at least 2V with 0.8mA load. Minimum bias impedance between bias voltage source and ring: 2 kOhm. AC-coupled tip to implement analog (external to ADC) 3 dB rolloffs at 60 Hz and 15 kHz. Most sound card inputs require a minimum signal level of at least 10 millivolts. Sound Blasters and some older 8-bit cards need 100 millivolts.

    Many small video cameras and Minidisc recoders use 3.5 mm stereo microphone connector for attaching stereo microphone to the system. The ones which supply power though the connector are usually called to have "plug-in power". That system supplies low voltage low current DC to the microphone through the same signal wire that carries the audio signal. The "plug-in-power" is few volts and the value of current limiting resistor isn series with microphone power supply is typically few kilo-ohms. Both channels on the stereo microphone have their own current limiting resistors.

    Diffent microphone type give different performance on different conditions. Directional microphones suffer bass lift up close to the sound source(usually singer mouth). That's because the directionality is a direct result of rearcancellation on the diaphragm, and such cancellation requires aleak with a pretty short time constant. And that means a relatively high low frequency cutoff. Above a certain frequency,the path length difference between the front and rear of thediaphragm restores some of the gain. In this type of microphonefrequency response is dependent on the soundpressure at the front of the diapgrapgm and the entrance to the rear being equal. If they are not, the frequency response changes. And one common instance of when this happens is when very-close mic'ing a singer. Because there is now a real difference in SPL between the two points, the frequency response is perturbed, most often showing a significant boost in the bass. This is the so-called proximity effect that cardiodssuffer from.The physical reason for above is that the velocity of an air particle in a sphericalfield is frequency-dependent, while the pressure isn't (OK, I know itis, but velocity is more so). So a pressure-sensitive omni mic will have a flat frequency response however close it is.

    Directional mics mix pressure and velocity responses to varying degrees. Figure eight microphones are pure velocity, hypercardiods are mostly velocity, with some pressure. Cardiods are equal parts pressure and velocity. Because the velocity of particles in a spherical field rises with reducing frequency, in a directional mic, the velocity response in the bass region dominates. Two things happen as a result. One is that theresponse tips up in the bass, and the other is that close up a cardiodmicrophone has a hypercardioid response at low frequencies.

    A Pressure Zone Microphone (PZM) is a type of microphone which sits on a flat surface and has a wide pick up pattern. The PZm microphones are typically jsut a small unit that you put in the middle of a flast surface (for example wall or a suitable around half square meter board). A PZM is nothing but a small electret mic capsule positioned with its diaphragm as close as possible to a boundary. Any decent electret circuits will work fine. Any solid wall you put the electret against will seem to disappear from the sound, with the caveat that it needs to be really close - 1mm is good, maybe a little further is still OK, but don't go too far or you'll get comb filter artifacts. To build a PZM you need to find the best reasonably small electret capsule that you mount absolutely flush to the surface. You can for example take a piece of plywood (for example eight inches square) where you install your microphone element in the middle (make hole in the middle and install microphone to hole absolutely flush to the surface). Similar idea can also be built by using 300x300x8mm acrylic sheet, with the edges heavily chamfered. The principle is still the same - sensing right at the boundary. The PZM idea is not the only boundary microphone in use. There are two boundary microphone technologies in use:

    • PZM (Pressure Zone Microphone): transducer against the boundary, separated by some mm's of air. Emispherical pattern
    • PCC (Phase Coherent Cardioid microphone): transducer inside the boundary, raised some mm's over the boundary. Slightly directional pattern (should be a half-dipole pattern)
    In both kinds of microphone, distance between diaphragm and boundary is a function of reinforcement frequency and diaphragm size. You are practically bound to use small size transducers (small electret capsules).

    All microphones have their limitations of low and high frequency response. Some of those limitations come from the physical construction and some are put into microphones intentionally. Microphones are purposely designed to not respond to frequency band much below hearable sound sound frequencies (20 Hz) because it would impair their normal functionality. The high frequency response limitations generaly just come from the microphone construction and is not usually intentionally limited. Many microphones have also maximum input osund pressure level, above which they do not work as they should (for example excessive distortion, even damage to microphone if exceeded greatly). For example some eletret microphone capsules list that figure in 100-120 dB range, but there are microphoen types that can measure higher sound levels.

    When using microphones with amplifiers, quite often you can get into contact with noise caused by the microphone itself and the microphone preamplifier. To avoid too much noise in the recording, you should select a suitable microphone for the applcation, use a good amplifier and make sure that you get enough sound to microphone so that there is more sounnd than noise.There are many sources for noise. Typically the microphone self noise is one contributor. That depends on the microphone and itsbuilt-in active element(s), if any. Whatever loss elements exist in theraw transducer will induce noise dueto thermal agitation (This is dueto the 2nd law of thermodynamics andthe inherent bidirectionality of rawtransducers.) To the extent that the tranducer iscoupled to air externally, there will be thermal noise arising from motionof medium's molecules. If there is an active amplifier builtinto the microphone, that device will have thermal noise (if it is a FET) or shot noise (if it is a BJT).

    Here are some standard wirings used in microphones. The standard wiring for mono microphones using 3-pin XLR connector is:

    • 1 - Ground/shield/screen
    • 2 - "Hot" side (+)
    • 3 - "Cold" side (-)
    The standard wiring for stereo microphones using 5-pin XLR connector is:
    • 1 - Ground/shield/screen for both channels
    • 2 - "Cold" side (-), Left channel
    • 3 - "Hot" side (+), Left channel
    • 4 - "Cold" side (-), Right channel
    • 5 - "Hot" side (+), Right channel
    Consumer and prosumer 6.3 mm mono microphone connector:
    • Tip - audio signal
    • Shield - Signal and power ground

    PC soundcard 3.5 mm electret microphone connector for electret capsule:

    • Tip - audio signal
    • Ring - bias voltage power (low current around 5V)
    • Shield - Signal and power ground

    3.5 mm stereo jack microphone connector many DAT recorders and camcorders:

    • Tip - left channel audio signal and bias voltage power
    • Ring - right channel audio signal and bias voltage power
    • Shield - Signal and power ground

    There are also many other microphone wirings in use,

      Microphone guides and tutorials

      • An Expert?s Guide to Wireless Set-up and Operation - With the advent of more consumer-friendly wireless technology, performers now have the flexibility and freedom to move about the congregation. With a little bit of knowledge and some helpful tips, you too can take advantage of the benefits of wireless technology and better sound production.    Rate this link
      • Bluffer's Micrpohone Guide - introduction to different microphone types    Rate this link
      • Crown Boundary Microphone Application Guide - A boundary microphone is a miniature microphone designed to be used on a surface such as a piano lid, wall, stage floor, table, or panel. The Pressure Zone is the region next to the boundary where the direct and reflected waves are in-phase (or nearly so).    Rate this link
      • Microphones Primer - how they work, characteristics and placement    Rate this link
      • Microphone Techniques: tips from the experts - Uni or Omni? Don?t be Afraid to Experiment! Conventional wisdom says that, in a sound reinforcement situation, a unidirectional microphone will be more feedback-stable than an omnidirectional microphone. This may not necessarily be true in all situations.    Rate this link
      • Microphone University - information about microphone technology, microphone and stereo techniques as well as suggestions of how to use microphones in different applications    Rate this link
      • Red Dotting Microphones - convernser microphones require a DC voltage in order to operate and there are various ways how it is fed to different microphone types    Rate this link
      • Shockmounts & Windscreens - tools to get rid of rumble and wind noise when recording    Rate this link
      • What You Need to Know About Microphones - The microphone handles the most-crucial conversion of energy in the whole sound system, where sound energy becomes an electrical signal. If you don?t have the right microphone positioned in the right place, no amount of after-the-fact processing will give you optimum sound.    Rate this link
      • Shockmounts & Windscreens - Two of the worst problems that plague location sound recording are RUMBLE and WIND NOISE. The solution to rumble lies in isolating the microphone from these vibrations by some means of free-floating suspension or non-conductive insulation... which is the role of a good shockmount. Contact wind noise, on the other hand, is that blast of distortion and audio breakup caused from wind physically striking the sensitive diaphragm of the microphone capsule. Contact wind noise can be prevented. That's what a windscreen does.    Rate this link

      Wireless microphones

      While today?s wireless mics can be quite good, no wireless mic is going to match the sound quality or reliability of a high-quality wired mic. In order to get the best performance from your wireless mics, and to help you decide which wireless mic is right for your use, you need to understand some of the technology and how it impacts your use of these mics. Any wireless mic, monitor or intercom system includes a very-low-power radio transmitter. This low-power transmitter sends out a very weak radio signal which must then be picked up by a special radio receiver tuned to the same frequency as the transmitter. Since the signal from the wireless mic transmitter is so weak, it is easily susceptible to interference.

      One commmon interference is "Multipath" interference. "Multipath" interference occurs when radio waves bounce off of metal objects or other surfaces and the receiver "hears" more than one signal. In this situation, the signal at the receiver's antenna fades in and out as the transmitter moves around the room. When the signal fades out, you will hear noise and static. Diversity receivers minimize this sort of interference. Diversity receivers use two (or more) antennas, and the signal is always received from the antenna that gives the best signal. Antenna switching as needed is done automatically by the receiver electronics.

      Other radio transmitters can interfere with wireless microphones. Most wireless mics are not licensed, and must accept any interference they get from other source operating at the same frequency (like sometimes TV stations, other radio systems etc.). Even the best receiver will not reject interference if it is on the same frequency as the wireless mic. If you have two wireless mics operating on the same frequency, neither one will work correctly. When buying a new wireless mic, it is smart to survey nearby environement what they use and at what frequency. If you use multiple wireless micrphones in your audio system, you need to do careful frequency planning to guarantee interference free operation of all of them (all of them operate at their own frequency that is far anough from the other frequencies used). Multi-channel wireless systems work reliably, however, after much time is spent in frequency coordination and optimum onsite antenna placements are made. The reliability factor improves dramatically if you use only high quality receivers designed for multi-channel environments. The performance specs on a receiver can be a bit nebulous, but among the most important specs for multi-channel capability are selectivity and third order intercept.

      Nowadays wireless microphones use analogue techniques. Wireless microphones typically use quite a bit of processing before the signal from microphone enters the radio and some more processing on the receiving end. Virtually all wireless microphone systems use some form of companding to reduce noise. The term "companding" is a combination of the words "compressing" and "expanding" ("compansion" is a combination of "compression" and "expansion"). In a wireless system with companding, the audio signal is compressed in the transmitter and expanded in the receiver. Without companding, sometimes also referred to as compansion, the audio signal-to-noise ratio of wireless systems would be only 60-80 dB, too low for most professional applications. When companding is employed, SNRs of 100 dB or more are possible. Typical companding is 2:1 companding, that makes original 100 dB reduced to 50 dB. The typical modulation on radio band is frequency modulation. Typical transmission power from microphone is few milliwatts to tens of milliwatts.

      Many wireless system manufacturers are working on new digital models, but the mainstream is still analogue. Wireless mics will never replace all wired mics. However, they have become an extremely important part of many live audioproductions.

      There are wireless microphone systems operating at different frequency ranges. Typically the microphones operate at VHF or UHF frequencies at the frequency range allocated for this kind of applications. Please note that the frequencies that can be used those applications vary somewhat from country to country, as well as the licensing policies (what frequencies need licenses and what are license free).

      Here are few details of some frequency bands for wireless microphone applications in Europe (mostly based on Finnnish frequency allocation list).

      • VHF 174 - 230 MHz: Wireless microhones according EN 300 422-1 standard, bnadwidth maximum 200 kHz, maximum power 10 mW ERP, locational limits on use; typical frequencies used 175 kHz, 192.25 kHz, 197.10 kHz
      • UHF 433.05 MHz: frequency used by some wireless microphones in Germany
      • UHF 863 - 865 MHz: Wireless speakers, in-ear monitor systems, helmet intercoms etc. according EN 301 357-1, maximum bandwidth 200 kHz, maximum power 10 mW ERP, SRD recommendation ERC/REC/70-03, ERC ruling ERC/DEC/(01)18.
      • UHF 863 - 865 MHz: License free radio microphones, maximum bandwidth 200 kHz, maximum power 10 mW ERP, standard EN 300 422-1, SRD recommendation ERC/REC/70-03
      • UHF 854 - 862: Licensed wireless microphones, maximum bandwidth 200 kHz, maximum power 50 mW ERP, standard EN 300 422-1, SDR recommendation ERC/REC/70-03; frequencies free of third order cross-modulation are 855.500, 856.000, 857.250, 860.375, 861.500 and 861.875 MHz
      • UHF 790.100 - 821.900 MHz: Licensed wireless microphones, maximum bandwidth 200 kHz, maximum power 50 mW ERP, standard EN 300 422-1, SDR recommendation ERC/REC/70-03
      • UHF 869,700 - 870,000 MHz: License free short range applications, maximum power 5 mW ERP, standard EN 300 220-1
      Please note your local regulations befroe using any of those freuqencies. The freuquency allocations on other countries could be different. For example USA uses entirely different frequency allocations. In USA wireless mic systems generally operate in several bands from 150 MHz to 216 MHz, which includes the VHF TV channels 7 through 13, or in the 470 MHz to 806 MHz UHF band (TV channels 14 through 69). Since multi-channel wireless mic systems often utilize inactive TV channels, one of your first considerations in operating a multi-channel system in a particular area often involves identifying the local TV stations. If you try to operate the wireless mic system on the same frequency as the local TV transmission, there is little hope that your battery powered transmitter signal will have a chance of overcoming a local TV station signal.

      When considering buying or renting a wireless microphone system you need to consider what are your need and available budget. If you want broadcast quality, you can easily spend several hundred dollars on just the microphone, plus the transmitter, plus the receiver. These systems work reliably at distances of 50 to 75 feet from the mike to the receiver. They are also available in a number of frequencies, so multiple mikes can be used at the same time. If you need only a couple of mikes, the mike to receiver distance will be under 20 feet, and/or your budget is tiny, you might try some of the lower-priced wireless mikes ($70 to $180 for a pair of handheld mikes with receiver).

      Microphone preamplifiers

      Using microphones

        Stereo miking techniques

        Stereo mic pickups may be divided into coincident, semi-coincident, spaced and multiple microphone techniques The object is to produce a stereo image in the playback that conveys the placement of sound sources in the acoustic space that the producer intends...that might be "recreative" when the reproduced image generally sounds like the original acoustics, and "creative" where a different image is created. Within these two approaches, many engineers divide the result into "they are here" or "you are there" images, depending on whether the performers sound like they're in the playback room, or if it sounds like the listener is in the performance room. There are three ways that the ears and brain can make a stereo illusion: differences in loudness from one ear to the other, differences in time of arrival of the sound waves, and differences in frequency response. While there are fairly narrow ranges of timing and frequency response that the ears can detect, the brain is extremely powerful in detecting minute differences in level and timing between the two ears' signals, and this contributes to precision in imaging. Different stereo microphone techniques use different ways to make stereo illusion. They can use one method or combine more than one.

        • Coincident stereo techniques such as M-S or X-Y use only loudness or intensity differences.
        • Semi-coincident stereo techniques use at least two mics, spaced up to about 50 cm (18 inches) apart. This technique gets both intensity and time of arrival cues.
        • Spaced techniques primarily use time of arrival differences to produce a wide, spread-out stereo image.
        • In multiple microphone techniques sounds from different microphones are delayed and panned into the stereo image.
        There are also special techniques which combine diffent main approaches.


    The goal of the process of audio recording is to accurately reproduce sounds recorded at one time and place in a different time and place. We want this reproduction to sound exactly like the original. While this seems pretty simple in concept, it is not so easily accomplished. Recording is a wide topic. There are only two ways to get your sonic information onto the tape, through a microphone or directly from an electronic output. In general, the quality of what comes back is affected by the quality of the equipment the signal passes through. The signal created by the microphone is very small and it needs po be amplified to "line level" to be able to be recorded (unless the recorder has built-in microphone input). The microphone signals are generally amplified using microphone preamplifier or a mixing board with built-in microphone preamplifiers. Everyone has their favorite microphones and pre-amps for different situations and most do color the sound. The important thing is whether you like that color and if it's appropriate for the particular situation at hand. Analog recording devices use a plastic tape coated with magnetic particles moving across a magnetic recording head at a constant speed to record and playback. There is a limit to the intensity of the signal that the tape particles can actually absorb and reproduce. Tape recording is a complicated process and there are lots of things to go wrong. When everythign goes right or almost right the sound quality is quite acceptable. Analogue tape, if recorded at too low level will sound noisy and if recorded at too high level will cause compression/distortion to the sound. Many sound engineers like to slam high levels on to analog tape to get the natural "tape compression" sound When doing recording, you need to be careful on the tape deck connection. When recording loud sound sources, sensitive microphones can put out a level that overloads a recorder's microphone pre-amp. Knowledgable tapers would then switch the mics to feed the line inputs, which can typically accept a signal 26 dB higher. However, some devices (e.g., video camcorders and lower priced analog cassette recorders) don't have line inputs. For these devices, our attenuator cables are worthwhile accessories. And when recording from a soundboard's pro-level outputs into a recorder's line inputs (designed to operate at lower level), attenuator cables prevent overloading. Sometimes special attenuators are needed when you want to connect a line level source to recorder microphone input. Most professional or semi-professional music recordings are done using multi-track recorders. Multi-track recorders are simply tape machines that allow you to record tracks and then overdub additional tracks in any order. Multi-track recorder allows you to record different sound intruments and sound sources separatery and later mix those sounds to the final product (usually stereo sound to tape or CD). Until recently, "Analog" was the only kind of recording available to most musicians. The wide availability of digital recording options (DAT, ADAT, har didsk recorders, computer recording hardware/software, CD recorders, MiniDisc). The digital recording process is far simpler mechanically, but much more involved electronically. Digital tape machines use mechanical transports and plastic tape as a storage medium for the digital information. Computers and hard disk recorder use hard disk. Digital recorders nowadays have very good sound quality and work well. When the sound is in digital format, it can be reproduced (copied) without any signal quality loss and processed with computer if needed. One thing to remeber on digital recorders is that they are not as forgiving on sound levels as analogue recorders, so "slamming high levels" on digital recorder will not give that "tape compression", but instead a very bad sounding distortion. The mixing console is the center of a recording studio. We use it to organize our signals going to the tape machines, to organize what we need to hear back from the tape machines, to monitor playback from our mixdown sources, to add effects (with aid of effect inits wired to mixer effect bus) to whatever is needed. In short, it is the heart of the multi-track studio. Practically all mixers provide some kind of EQ, usually switchable on or off, in the signal path. There are many types of equalizers and they get used in many different ways by different people.When connecting different devices you need to be careful on connectors and signal levels.Semi-pro and home recording gear operates at a -10 dBm level while professional equipment operates at a + dBm level. Without getting too technical, this means you have to pay attention to the particular input and output levels of your boxes and how you interconnect them.General tips on recording:

    • Voices, horns and acoustic piano are recorded with microphones. There are different microphone types which suit to different situations.
    • String instruments can be recorded acoustically with microphones or directly if they have pickups. There is a different sound to each and in different situations, one may be more appropriate than the other.
    • Samplers, synthesizers and drum machines have direct outputs and can be connected straight to your mixer.
    • Real drums and real drummers are their own topic for recorder. First you need a well tuned drum set and a goo drummer. The basic approach to drum miking involves a seperate mike for the kick, the snare, the hat, the toms and one or two overheads to get the cymbals and the room sound if there is one. This is the most common setup, but keep in mind that some of the great drum sounds from classic rock records were recorded with only two mikes on the whole kit. Compression can be a big help when recording drums.
    • Avoid using digital audio formats which use data compression to make your initial recording which you plan to process later. Every time sound is compressed in this way the quality goes worse.

      Other on-location recording

      • Recording Your Christmas Events - Recording a Christmas program can be both exciting and intimidating. Your equipment setup can be as simple as an inexpensive set of microphones and cassette deck monitored with headphones. At the other end of the spectrum, you can use multiple studio-quality microphones, a large recording console and a multi-channel tape deck or digital audio workstation (DAW) that allows you to edit and remix the program to your hearts desire.    Rate this link


      Mastering is simply a music producer's last chance to make his or her music as good as it can be. After all the songs are mixed and (ideally) some time has passed, a mastering engineer gives the music a fresh listen and does fine-tuning if needed. The mastering process usually works something like this: Finished mixes are re-recorded onto another tape, or transferred into a computer. To go from tape to tape (digital or otherwise), perform your sonic tweaks during the transfer. A lot of modern pop music uses heavy compression on their sound nowadays. Usually more compression is used than what is needed for a good sounding recording. Using to much compression on recording is primarily a social problem. There are strong incentives for music to be processed in such a waythat it's "louder" and more attention-getting, when played over FMradio. There are a bunch of reasons for this, having to do withcompetition between stations (the station with the loudest-soundingsignal is believed to have a better chance of "grabbing" a listenerthan one with a quieter-sounding signal) and the conditions underwhich popular music is often played back (in cars, boomboxes, Walkman-and MP3-players with cheap headphones, etc. in conditions with highambient noise levels). The same attitude then tends to reflect down into the studio... thebands and producers often want their music to sound "louder" or"catchier", and over-process it to death as a result. A song that is constantly nudging the maximum level will sound louderthan one with a full dynamic range, only occasionally hitting themaximum. In effect, it'san arms race... the band which uses more/newer/sexier compressiontechnology is believed to have an 'edge' over bands with a lesserarsenal of processors. A number of folks in the pro-audio industry are trying to buck thistrend, but at the moment it seems that "over compressed" seems to be the sound trend on most pop music. On some applications some compression is a good idea. You can understand the need for compression if you listen towide-dynamic-range classical music in a noisy car or in a quiet "open plan"office. In the car the quiet passages disappear from audibility below roadnoise, and in the office, if you set levels so you can hear the quietpassages, people six cubicles in all directions will be disturbed by thechreschendoes. So adding some compression makes music more attractive to people who listen in environments that can'tsupport wide dynamic range.

      Tape deck topics

      The tape recorder is the principal instrument of the classic electronic music studio. The technical quality of the composition is limited by the decks used, and may be further compromised by how the decks are used. Analog tape decks use a plastic tape coated with magnetic particles moving across a magnetic recording head at a constant speed to record and playback. Most magnetic tapes have a mylar or polyester base with a thin coat of magnetic material, usually gamma ferric oxide or chromium dioxide, but newer tapes are double-layered which combine the good low-frequency response of ferric oxide and good high-frequency response and low noise of chromium dioxide; the oxide is cured onto the base and the tape is calandered. The metal particles have a random orientation in unmagnetized tape, but they are aligned into definite magnetic patterns by the magnetic field produced by the recording head. If all other factors are the same, the wider the track, the greater the S/N ratio.Professional analog tape recorders are available with tape widths up to 2" and up to 24 tracks. There is a thin guardband of uncoated base tape between the tracks to, yield improvedproviding channel separation, reduceing crosstalk, and provideing some tolerance for differences in head/track alignment among machines.There is a limit to the intensity of the signal that the tape particles can actually absorb and reproduce. Tape recording is a complicated process and there are lots of things to go wrong. When everythign goes right or almost right the sound quality is quite acceptable. Analogue tape, if recorded at too low level will sound noisy and if recorded at too high level will cause compression/distortion to the sound. Most professional or semi-professional music recordings are done using multi-track recorders. Multi-track recorders are simply tape machines that allow you to record tracks and then overdub additional tracks in any order. For a 4-track home studio, -10dB line signal levels are generally used for interfacing to tape deck. Many home C-cassette decks have a switch labelled and has the settings LOW, MED(I/III) and HIGH (II). Sometimes second is labelled EQ and has the settings TYPE I, TYPE II and TYPE III. Those switches (sometimes only one) are ment to be set to match the cassette type you have (whould be written to the tape package). The available tape types are:

      • Type I = ordinary ferric tape
      • Type II = chrome tape
      • Type IV = metal tape
      Type III was a Ferrichrome combination tape made by Sony. It is no longer available.Each of those tape types require a different combination of bias and EQ settings to generate optimal results. "Bias" refers a large amplitude high-frequency current that issuperimposed on the recording current in order to drive it outside thehighly non-linear portion of the magnetic transfer characteristic ofthe tape, and thereby reduce distortion. Slight variation in frequencyresponse is a side-effect of adjusting this. "EQ" is a modification to the frequency response of the audioamplifiers (both record and replay) designed to ensure a standardisedrecording that can be played back correctly on different machines.The combinations of EQ and bias settings for different cassette types:
      • Type 1 (Normal): Low bias current with 120microsecond EQ. Sell has no bias holes.
      • Type II (High or Chrome): Highish bias current with 70 Microsecond EQ .Shell has bias holes near the erase prevent tabs.
      • Type IV (Metal): Use a very high bias current with 70 microsecond EQ. Shell has bias holes near the center of the spine.
      Select the tape type according to the cassette in use and set the matchingBias current (it's usually written on the packing). Types II and IV are thesame (70usec). Type I is 120 usec.Make sure you get all the setting right or it will seriously degrade the hign frequency response of your recordings. For example under biasing results in more top and more distortion. Bias setting has only effect on what the tape recorder does when it records sound to a tape. EQ is used on record and playback. There weren't separate switches for bias and EQ on all decks, some had switched with just tape types and did adjustments for both parameters with a one switch. The better decks allow fullyvariable adjustment of bias, with manual and/or auto EQ setting.NOTE: Lots of commercial pre-recorded tapes were recoreded on chrome tape,but with type 1 (120 usec) equalisation, so people with basic cassetteplayers could play them with 'correct' equalisation. Type II tape can be recorded using the high bias setting and a 120 microsecondEQ. This will net almost 4dB higher recording level, plus the tape will becompatible with most standard players.The mass duplicators of commercial cassettes used to do this all the time andthe tapes were marked Chrome tape, 120 microsecond plus the tapes do not havethe chrome tabs in the shell. Because analogue tape recording is a complicated process, how the player responds to the recorder machine, is very eratic. If you want to played back a tape from another machine that souded good, unless it is calibrated it first. Studio tape users generallyuse calibaration tones in the beginnign of tape for this. Home casette recorder userssometimes use calibration tapes to adjust all tape decks tomatch the one calibration tape. To mkae good recording yourmust keep your tape deck in mechanically good condition, cleanit as needed and calibrate it. Dolby Noise Reduction is a compression (during recording) and expansion (during playback) method to reduce noise, particularly in the higher frequencies where tapehiss is prevalent. For correct Dolby operation, the tape machine must be matched to the tapebeing used. When the machine was properly set up for sensitivity, bias and EQ, the Dolbynoise reduction give very good performance.If you record with Dolby on, you must play with Dolby onfor it to work as designed. Dolby B is what most units use and all pre recorded tapes are. Dolby C is multiband noise reduction, that is slightly better than B (and is much more critical in it's calibration between machines). If you play atape on a different machine than recorded on, and the playback machine onlyspecifies "Dolby," then it is probably Dolby B, which is not compatible withDolby C. Many inexpensive tape decks may not play back Dolby recorded tapesvery well, especially if recorded on a different machine.

    Linking telephone lines to audio systems

    Almost every audio engineer has had to deal with telephone lines at one time or another. The reason for this is that the dial-up "plain old telephone system" or POTS, remains the least-expensive system for transmission or reception of audio, albeit with limited audio frequency response. With last-minute remotes, breaking news stories, increased program demands and budgetary pressures, some radio users still depend on POTS for transmission and reception. If the application is in your studio, you can take your time and make some calls to determine the right equipment to buy and the correct phone line configuration to order. Out on the road it's usually a different story, because not all telephone connectors on the world are the same.

    In the POTS universe there are generally three phone interface methods in use:

    • An analog hybrid for direct interface to a phone line via RJ-11
    • A "universal" interface for connecting to the handset port on PBX, ISDN or key systems via RJ-22
    • A headset interface to wireless or other telephones via a mini-phone (1/8-inch) TRS plug
    Most telephone line connections are done to POTS lines. The POTS line consists of two wires called tip and ring. These two wires provide: DC current to power the telephone electronics, AC current to ring the telephone bell or electronic ringer, full duplex balanced voice path. POTS line is not hifi medium at all. POTS Line Characteristics is that it has bandwidth of around 180 Hz to 3.2 kHz and signal to noise ratio is approximatelly 45 dB normally (worse on bad line conditions). The phone company decided years ago that this performance would be sufficient for speech intelligibility while allowing them to multiplex many calls over coax and twisted pair. The low end is rolled off early to stay away from the 60 Hz region and allow the use of small telephone transformer. The high end cut off is caused by the telephone transmission system (nowadays theaudio is digitized at 8 kHz). The high end cut off is critical. Voice on the telephone network is digitized at 8 kHz sampling rate which means that any signal above 4 kHz will be aliased back as noise in the voice band. Most voice CODECs roll off at about -25dB at 4 kHz with a -3dB down point around 3.2 kHz. The noise on the line comes in many forms such as electrical interference from fluorescent fixtures or hiss from the many amplifier stages in the voice path. Speech correlated noise can be introduced from non-linear speech digitizing and compression methods. Crosstalk from other conversations is another form of noise. The bottom line is that you can never count on more than 45 dB signal to noise ratio.

    Signal Levels on telephone line are -9 dBm average speech (at tip/ring) Speech peaks out to +4 dBm are common but will start to clip. The FCC requires that all telephone audio interconnect equipment limit speech to -9dBm, averaged over 3 seconds. Consult telephone regulations requirements (FCC Part 68 in USA) for all the details. The voice on a tip/ring pair is full duplex balanced audio which requires a two wire to four wire hybrid circuit or transformer to convert it into separate transmit and receive audio paths. Bulky and expensive hybrid transformers have been replaced in most telephones by ICs which perform the same function. Whether it is a transformer or IC, the hybrid must also provide 1500 volt isolation and surge suppression from lightning strikes.

    In typical telephone equipment , the biggest contributor to poor audio quality is the handset microphone (has bo be cheap and has to withstand hard use for years). You can usually get somewhat better audio quality when you feed sound directly to the line.

    A telephone hybrid is a relatively simple electronic device used to connect a telephone system to a regular audio circuits. These are normally used in radio stations to connect callers in airchain, so that conversations may be broadcast. The main principle at work is impedance matching, blocking of 48V POTS line DC and isolate the audio side from the line electrically. Some inexpensive designs connect to the telephone handset cord, with a button to activate either the handset or the hybrid. These only cost around $100 need no power. More expensive versions can cost thousands of dollars or more, may include better isolation between caller and studio voice, may provide signal compression and even ful signal processing to make the telephone to sound better on the broadcast. There are also models that support many telephone lines to allow many callers to be on the show.

    The hybrid functionality is not perfect. You can get something like 20-30 dB of isolation between signals going to different directions, usually not much more with general purpose "fits for all" hybrids. Usually in normal voice applications this does not cause problems that you hear some of your voice back, but in very long distance calls this can be irrating (in those cases special adaptive echo cancellers are used). Commercial hybrid couplers provide familiar audio connections for full duplex transmit and receive audio. The primary difference between couplers is the amount of trans-hybrid loss or echo from the hybrid. When you send audio into a hybrid, some of the audio leaks back into the receive audio mixed with the caller's voice. The amount of return leakage depends on the type of hybrid and how well it matches the characteristics of the phone line. Many good professional audio hybrids have manual settings to minimuze the signal leakage from input to output (needs to be tuned to current line conditions). With a well tuned hybirid you might get something like 30-40 dB isolation.

    In addition to hybrid interfaces that plug directly to telephone line there are other type of audio interfaces. Handset replacement devices plug up or into the handset port on a POTS telephone. They replace the telephone handset microphone, and sometimes the receiver, either with an enhanced microphone or with a provision that lets you connect a broadcast mic and headphones to the telephone instrument. The replacement devices do not act like a telephone hybrid, because they just rely on telephone device electronics, where in handset, by design, allows leakage between the send and receive paths ("side tone"). If you connect this kind of adapter to your audio system, be sure that it has the same 1500 volt isolation from line. Without isolation you generally get lots of noise to line, risk damaging your equipment and create electrical safety risk. There are some speakerphones and such devices that have external audio connectors (input or output). When usign this kind of connection with any audio system be very careful in making of the connection. It is best to make sure that this particular output is properly isolated from incoming telephone line, or otherwise you get into problems. The safest bet in all connections is to have and audio isolation transformer between the telephone equipment and your audio system. Because signal is from telephone has pretty limited signal quality, you don't need the highest quality transformers you can find (for exmaple the line transformer from an old modem should do the job nicely here, it passes telephone frequencies nicely and proves safe isolation level from line).

    Digital hybrids are even used for broadcasting over standard telephone systems, using a special unit with DSP audio data compression and decompression at each end. Audio bandwidths up to 15 kHz can be chieved this way, along with slow auxiliary data (for example for remote triggering of relays or other audio devices). Compression is often via MPEG standards, particulary now MPEG-4. Depending on the design the digital hybrid may be designed to be connected through a normal PSTN telephone line or through ISDN line.

      Linking telephone conversation to studio systems


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