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Sound Engineering Documentation Dec-11-2010 Table of Contents Active ............................................................................................................................................... 5 AFL .................................................................................................................................................. 5 Alternating Current (AC) .................................................................................................................. 5 Amplifier........................................................................................................................................... 5 Amplitude......................................................................................................................................... 6 Attenuate ......................................................................................................................................... 6 Attenuator ........................................................................................................................................ 6 Aux-send.......................................................................................................................................... 6 Aux-return ........................................................................................................................................ 6 Balanced Connection ...................................................................................................................... 7 Cables & Connectors....................................................................................................................... 7 Cables .......................................................................................................................................... 7 Cable Connectors ........................................................................................................................ 7 1/4" Connector.......................................................................................................................... 7 1/8” Connector.......................................................................................................................... 7 Banana (MDP).......................................................................................................................... 7 Optical (ADAT) ......................................................................................................................... 7 TS ............................................................................................................................................. 7 TRS .......................................................................................................................................... 8 RCA .......................................................................................................................................... 8 XLR........................................................................................................................................... 8 MIDI .......................................................................................................................................... 8 Capacitor ......................................................................................................................................... 9 Comb Filter ...................................................................................................................................... 9 Compressor ..................................................................................................................................... 9 Threshold ..................................................................................................................................... 9 Ratio............................................................................................................................................. 9 Attack ......................................................................................................................................... 10 Release ...................................................................................................................................... 10 Knee........................................................................................................................................... 10 Makeup Gain.............................................................................................................................. 10 Connection (signal) types .............................................................................................................. 11 Balanced .................................................................................................................................... 11 Unbalanced ................................................................................................................................ 11 Decibel........................................................................................................................................... 11 Delay Effects.................................................................................................................................. 12 Delay .......................................................................................................................................... 12 Flange ........................................................................................................................................ 12 Chorus........................................................................................................................................ 12 Phasing ...................................................................................................................................... 12 DI (Direct Injection) Box................................................................................................................. 13 Passive DI .................................................................................................................................. 13 Active DI..................................................................................................................................... 14 Direct Current (DC)........................................................................................................................ 14 Dynamic Range ............................................................................................................................. 14 Effects Loop................................................................................................................................... 14 Effects Unit .................................................................................................................................... 15 Equalizer........................................................................................................................................ 15 Frequency .................................................................................................................................. 15 Q ................................................................................................................................................ 15 Gain (Boost/Cut) ........................................................................................................................ 15 Fader ............................................................................................................................................. 17 Feedback ....................................................................................................................................... 17 Controlling Feedback ................................................................................................................. 17 2 Fletcher-Munson Curves ............................................................................................................... 18 Frequency...................................................................................................................................... 18 Audible Frequencies .................................................................................................................. 18 Infrasonic Frequencies............................................................................................................... 19 Ultrasonic Frequencies .............................................................................................................. 19 Frequency Range ...................................................................................................................... 21 Frequency Response................................................................................................................. 21 Gain ............................................................................................................................................... 21 Gain Knob...................................................................................................................................... 21 Gain Structure (Gain Staging) ....................................................................................................... 21 Gate ............................................................................................................................................... 21 Threshold ................................................................................................................................... 22 Attack ......................................................................................................................................... 22 Release ...................................................................................................................................... 22 Range......................................................................................................................................... 22 Hold............................................................................................................................................ 22 Graphic EQ .................................................................................................................................... 22 Ground Lift ..................................................................................................................................... 22 Ground Loop.................................................................................................................................. 22 Headroom ...................................................................................................................................... 23 Hertz .............................................................................................................................................. 23 Hi-Z ................................................................................................................................................ 23 Impedance ..................................................................................................................................... 23 Input............................................................................................................................................... 23 Input Level types............................................................................................................................ 23 Microphone level ........................................................................................................................ 23 Line Level................................................................................................................................... 23 Instrument Level ........................................................................................................................ 24 Limiter ............................................................................................................................................ 24 Loudspeaker .................................................................................................................................. 24 Lo-Z ............................................................................................................................................... 24 Microphone .................................................................................................................................... 24 Condenser Mic ........................................................................................................................... 24 Electret Condenser Mic.............................................................................................................. 25 Dynamic Mic............................................................................................................................... 25 Ribbon Mic ................................................................................................................................. 25 Microphone polar patterns ......................................................................................................... 25 Omnidirectional....................................................................................................................... 26 Subcardioid............................................................................................................................. 27 Cardioid .................................................................................................................................. 27 Supercardioid ......................................................................................................................... 28 Hypercardioid ......................................................................................................................... 28 Bi-directional (figure of 8) ....................................................................................................... 29 Shotgun .................................................................................................................................. 29 Directional Response................................................................................................................. 30 Off-Axis Response ..................................................................................................................... 30 Microphone Placement .............................................................................................................. 30 MIDI - Musical Instrument Digital Interface ................................................................................... 30 Mixing Console .............................................................................................................................. 32 Modulation ..................................................................................................................................... 32 Monitors (“foldback” cabinets) ....................................................................................................... 32 Noise Floor .................................................................................................................................... 32 Noise Gate..................................................................................................................................... 32 Ohm’s Law..................................................................................................................................... 33 Output ............................................................................................................................................ 33 PAD ............................................................................................................................................... 33 3 PA System ..................................................................................................................................... 33 Parametric EQ ............................................................................................................................... 34 Passive .......................................................................................................................................... 34 Phantom Power ............................................................................................................................. 34 Phase............................................................................................................................................. 34 Phase Shift................................................................................................................................. 34 PFL ................................................................................................................................................ 34 Potentiometer ................................................................................................................................ 35 Power............................................................................................................................................. 35 Preamplifier.................................................................................................................................... 35 Proximity Effect.............................................................................................................................. 35 Resistance ..................................................................................................................................... 35 Reverberation ................................................................................................................................ 36 Ringing Out a System.................................................................................................................... 36 Signal Path .................................................................................................................................... 36 Signal to Noise Ratio (SNR) .......................................................................................................... 36 Sound Reinforcement System....................................................................................................... 36 Sound Wave .................................................................................................................................. 36 Combining Waves...................................................................................................................... 37 Speaker ......................................................................................................................................... 38 Speed of Sound ............................................................................................................................. 38 Sound Formulas......................................................................................................................... 38 Transducer..................................................................................................................................... 39 Transformer ................................................................................................................................... 39 Transients ...................................................................................................................................... 39 Transient Response ...................................................................................................................... 40 Trimpot........................................................................................................................................... 40 Unbalanced Connection ................................................................................................................ 40 Wavelength.................................................................................................................................... 40 Wedge ........................................................................................................................................... 40 4 Active – In the context of audio, active equipment (DI box, loudspeaker, guitar pick-up, etc.) does not require an external power supply to operate the unit. For example, an active monitor has a built-in amplifier and an active DI box operates on battery power or via a power plug. AFL – An acronym for ‘After Fader Listen’, AFL is used on mixing consoles to override the normal monitoring path to allow the operator to monitor a specific signal at a predefined point within the mixer. The AFL signal is taken after the fader of a channel or group buss and therefore, the level of the fader will affect the level heard in the AFL monitor circuit. AFL is often taken after the pan pots, allowing the sound engineer to monitor the signal with the channel/ bus pan position as it is set within the mix. It’s a useful way to monitor groups of related instruments or vocals solo with EQ, gain, and pan reproduced as in the overall mix. An activated AFL on an auxiliary send, for example, would allow the sound engineer to monitor the signal level and listen to the mix at the point in the signal path directly after the specified aux trim pot. If the aux send is being used for a stage monitor wedge, the sound engineer would be able to listen to the mix being sent to that stage monitor. Alternating Current (AC) – (also see Direct Current (DC)) An electrical current flow that alternates its direction on a periodic basis. This is the way household electricity operates. Alternating the current flow makes it easier for electrical power to be efficiently distributed over vast distances. In North America the rate at which the electrical current alternates is 60 times/ second (or 60Hz). Similarly, audio signals also alternate with their frequencies corresponding to the frequencies of the sound wave present (e.g., 440 Hz = 440 cycles/ second). Amplifier – A device for increasing the power of a signal. An amplifier takes power from a power supply and shapes the output to match the (relatively low power) input signal. The process is not 100% efficient and does introduce some noise and distortion into the signal path. Different designs of amplifier are used for different types of signals and applications. Amplifiers can be generally categorized into three types: small signal amplifiers, low frequency power amplifiers, and RF (radio frequency) power amplifiers. Each calls for a slightly different design approach due to the physical limitations of the internal components used to implement the amplifier and the efficiencies that can be realized from them. 5 Amplitude – In physics and electronics, amplitude is literally the maximum absolute value of a periodically varying quantity. Simply put, it is the strength of a signal or sound without regard to its content. The amplitude of audio signals generally references the signal voltage. Amplitude alone does not determine power (or loudness in audio), but does affect it. In the physical world the amplitude of a sound is measured in dB of SPL (Sound Pressure Level), which also does not truly define the power or intensity of sound – only the sound level at a point in time. RMS (root mean square) amplitude is used in electrical engineering. The peak-topeak voltage of a sine wave is nearly 3 times the RMS value (Peak-to-Peak = 2.8 x RMS). Attenuate – (also see Attenuator) To reduce the intensity of a flux through a medium. For the purposes of audio, attenuation affects the propagation of waves and signals in electrical circuits as well as in air. Simply, to attenuate a signal is to turn it down. A -20dB PAD on a mixer is an attenuator which reduces the amplitude of a signal by 20dB. Attenuator – (also see Attenuate) An attenuator is the opposite of an amplifier. An amplifier provides gain while an attenuator provides loss. Aux-send – An output on a mixing console supplying an independently routed mix to an external device such as a monitoring system, an effects processor, or to recording equipment. Sending a signal through the aux-send to an external effects unit and then back via the aux-return input jack creates an effects loop (effects loops are also created for individual channels using the “insert I/O” of the channel strip). The auxsends from a group of inputs can also be routed to an amplifier and then sent to a monitoring system. Aux-return – The input compliment of an aux-send output on a mixing console. It is primarily used for the return signal from a single effects unit or chain of effects units creating an effects loop (also see Aux-send). 6 Balanced Connection – see Connection (signal) Types Cables & Connectors: Cables – A cable, patch cord, or speaker wire is an electrical or optical cable used to connect one electronic or optical device to another for the purpose of signal routing. Cables between components are also called interconnects. An electrical wire consists of a signal conductor and an electrical shield. The shield connects to the ground of a circuit and absorbs (catches) any unwanted external interfering signals before they get into the signal chain. For a descriptive list of some of the most common types of terminations used on cables, see “Cable Connectors” below. Cable Connectors – An electrical connector fastened to the end of a wire or cable is an electrical conductive device used for joining electrical circuits together. There are hundreds of connector types so for the purpose of the audio world, here are the most common found connectors… 1/4" Connector These are 1/4" diameter connectors used in professional audio equipment setups and instrument patch cords. See TS and TRS below for the photo examples. 1/8” Connector 1/8" diameter connectors are generally used in smaller audio visual interconnects. Usually the size of headphone jacks on most portable media players and may be TRS (balanced) or TS (unbalanced). See TS and TRS below for descriptions of these two types of plugs. Banana (MDP) A banana connector (usually referred to as a banana plug for the male and a banana jack for the female) is a single-wire (one conductor) electrical connector used for joining wires to equipment. Banana plugs are also sometimes called "MDP" plugs and they inserted into the open end of "binding posts". In the world of pro audio and live sound they are sometimes used as connectors on cables that connect the power amplifier to the loudspeakers. Optical (ADAT) Contain optical fiber which carries far more information than conventional copper wire and usually isn’t subject to electromagnetic interference. These cables are for compatible 2-channel S/PDIF and ADAT lightpipe connections as seen on digital audio equipment such as A/D D/A converters and audio interfaces. TS A 1/4" cable similar to the TRS connector but without the bottom "ring". TS is an acronym for “Tip & Sleeve”. With only one divider ring, there are only a "Tip" and a "Sleeve". The "Tip" is the carrier of the signal and the "Sleeve" is the grounding wire. Guitar patch cables or cables for line-level instruments are typically TS type. These are known as unbalanced. 7 TRS Tip Ring Sleeve TRS stands for "Tip Ring & Sleeve" and it was originally invented for telephone switchboards. TRS connectors can be 1/8” or 1/4" sizes. They look like a 1/4" TS plug with an extra "ring" around the connector. These two rings form three isolated connection points along the plug. TRS connectors are used for connecting equipment such as microphones or line level signals or wherever you may need to have two conductors plus a ground in one plug. Unbal. Mono I/O Signal Ground or no connect Ground Unbal. Mono Insert Send or Return sig. Return or Send sig. Ground Bal. Mono I/O Positive (Hot) Negative (Cold) Ground Unbal. Stereo Left channel Right channel Ground RCA These phono connectors are named after the RCA Corporation which developed products that used these connectors. Usually there is one white (or grey) connector for the left channel and one red connector for the right channel. There may be a composite video RCA connector as well - typically yellow. RCA connectors are popular with mixers, CD Players, televisions and DVD players. XLR XLR connectors are commonly used for transmitting balanced mic and line level signals to speakers and mixers. They were originally developed by Cannon for the "Cannon XL" series which later added a Rubber compound surrounding the contacts - eventually leading to the abbreviation "XLR". Many manufacturers now produce 3-poin XLR cables and connectors. one for a positive, one for a negative, and one for a ground connection. The XLR connector is designed so that the pin for the ground connection is made first, which aids in preventing unwanted sounds and pops in the audio chain. Standard Pin Designations 1. Chassis ground (cable shield) 2. Positive polarity (hot) + 3. Return terminal (cold) - MIDI Musical Instrument Digital Interface. It is an industry-standard protocol that allows electronic musical instruments, such as synthesizers and drum machines, to communicate and synchronize with one another. The original physical MIDI connection uses DIN 5/180° connectors. 8 Capacitor – A passive electronic component capable of storing energy in the form of a static electric field. They are widely used in electronic circuits for blocking direct current while allowing alternating current to pass, in filter networks, for smoothing the output of power supplies, in resonant circuits that tune radios to specific frequencies, and many other purposes. An ideal capacitor is characterized by a single constant value, capacitance, measured in farads (ratio of electric charge on each conductor to the potential difference between them). Comb Filter – A filter that has a series of deep notches in its frequency response with all notches being at multiples of the frequency of the lowest notch. This means that all the notched frequencies are harmonically related. When graphically shown, the notched response graph looks like a comb, hence the name. The comb filter is produced when a signal is time delayed and added back to itself and therefore, some frequencies will cancel one another out while others become reinforced. Modulating a comb filter will create a flange effect. Compressor - A dynamics device whose function is to attenuate the signal above a certain threshold. When the signal goes above a set threshold, the audio level will be compressed at the set ratio. Compressors are often used to decrease the dynamic range (reduce the difference between high and low audio levels) of the overall signal by pulling the high transients of an offending signal down (squishing the waveform from the top-end). This allows increased gain of the overall signal if desired using the makeup gain setting. Compressors are also advantageous in evening out vocals or bass guitars, fattening up sounds, increasing sustain of guitars etc. See the diagram below for a visual representation of what a compressor does to an audio signal. Common control settings found on a compressor are: Threshold – sets the level at which the compressor will begin to attenuate a signal based on the compression ratio. Ratio - sets the input/output ratio at which the compressor will compress a signal. For example, a 3:1 ratio means that a signal overshooting the threshold by 4 dB will leave the compressor 1 dB above the threshold. Some different compression ratios are graphically represented below. 9 Attack – Measured in a unit of time (e.g., milliseconds), the period when the compressor begins to attenuate the signal once it has been detected as going beyond the threshold limit – similar to a fade-in. Release – Measured in a unit of time (e.g., milliseconds), the period in which the compressor begins to increase the gain back to the level set by the ratio, or to 0 dB, once the signal level has fallen below the threshold – similar to a fade-out. Knee – determines whether the response curve bend is sharp or gradual. A soft knee slowly increases the compression ratio as the signal level increases until it reaches the set ratio. Soft knees help to reduce the audible change created by the compression process (especially for higher ratio settings). Makeup Gain – increases overall output gain after compression has been applied. 10 Connection (signal) types: Balanced - Balanced audio is a method of interconnecting audio equipment using impedance-balanced lines. This type of connection is very important in sound recording and production because it allows for the use of long cable runs while reducing the susceptibility to external noise. They use three-conductor connectors like those found on the XLR or TRS connectors. XLR connectors are usually used with microphones because of their durable construction, while TRS jacks are typically used for mixer inputs and outputs because of their small profile. A typical balanced cable contains two identical wires twisted together and then wrapped with a third conductor (foil or braid) that acts as a shield. The term "balanced" comes from the method of connecting each wire to identical impedances at the source and load. This means that much of the electromagnetic interference will put an equal noise voltage in each wire. Because the amplifier at the far end measures the difference in voltage between the two signal lines, noise that is identical on both wires is rejected. The noise received in the second, inverted line is applied against the first, upright signal, and cancels it out when the two signals are subtracted. The wires are also twisted together, to reduce interference from electromagnetic induction. A twisted pair makes the loop area between the conductors as small as possible, and ensures that a magnetic field that passes equally through adjacent loops will induce equal levels of noise on both lines, which is cancelled out by the differential amplifier. If the noise source is extremely close to the cable, then it is possible it will be induced on one of the lines more than the other, hence it won't be cancelled as well. Cancellation will still occur to the extent of the amount of noise that is equal on both wires. The separate shield of a balanced audio connection also yields a noise rejection advantage over unbalanced two-conductor cables where the shield must also act as the signal return wire. Any noise currents introduced into a balanced audio shield will not therefore be directly modulated onto the signal, whereas in a two-conductor system they will be. A typical balanced line level output is +4 dBv (referenced to 0.775 volts) or about 1.2 Volts. Unbalanced – An audio signal where there is one signal conductor and one return ground line (or shield). An unbalanced connection has unequal impedances and is much more subject to hum due to noise interference and ground looping issues. Cable runs in an unbalanced system should be short. A typical consumer unbalanced audio device would have an output around -10 dBv (referenced to 1 volt) or 316 mVolts. Decibel - The decibel (dB) is a logarithmic unit that indicates the ratio of a physical quantity (usually power or intensity) relative to a specified or implied reference level. With that said, the decibel (a tenth of a bel) means nothing in of itself and requires a reference level to be useful. +3 dB SPL is meaningful as it references SPL (Sound Pressure Level). The decibel is widely known as a measure of sound pressure level, but is also used for a wide variety of other measurements in science and engineering, most 11 prominently in acoustics, electronics, and control theory. In electronics, the gain of amplifiers, attenuation of signals, and signal to noise ratios are often expressed in decibels. It confers a number of advantages, such as the ability to conveniently represent very large or small numbers, a logarithmic scaling that roughly corresponds to the human perception of sound and light. The decibel symbol is often qualified with a suffix, that indicates which reference quantity has been used such as dBu, referencing 0.775 volts RMS. Delay Effects: Delay – An audio effect that records an input signal to a storage medium and then plays it back after a period of time. The delayed signal can be played back multiple times or can be played back into the recording again to create the effect of a repeating, decaying echo. Flange – Delay times of 10ms or less, the flange effect creates a series of peaks and dips in the signal’s frequency response (comb filter). A flange is the metal rim or the reel part of a reel to reel tape (as opposed to the hub). Originally, tape machines were used to create this delay effect in audio production in a process dubbed as flanging. It consisted of recording identical signals on two separate tape reels and then playing them together. Using pressure to one of the reel flanges, the one tape reel would briefly slow down on one of the tape machines. The short timing discrepancies that resulted produced a very pronounced comb filter effect which was often modulated by alternating pressure to each of the machine's reels. One machine would slow down relative to the other, and then the second machine would be slowed beyond the first. It was also possible to route some of the signal being played back into the recording circuit to provide regeneration and resonance effects. Later, electronic flange effect units were invented that incorporate a modulated analog or digital delay line which is mixed back with the dry signal. Although the electronic flange effect units are much more convenient than the original open reel approach many sound engineers uphold that the electronic units do not sound as good as the original open reel flange. Chorus – When two individual signals that have roughly the same timbre and almost (but never exactly) the same pitch are combined. They are often slightly delayed and are perceived as one signal. Chorus can be electronically simulated using effects units, but can occur naturally as with choirs and orchestras. A piano, synth, or 12 string guitar can produce this effect on their own. Phasing – Similar to the flange effect, phasing creates a combing effect that is automatically varied to create phasing. Phasing creates a series of peaks and valleys in the frequency spectrum. The placement of these peaks and valleys is generally modulated so they vary over time and create a sweeping effect. The process splits an audio signal into two parts and treats one with an all-pass filter (which preserves the original signal’s amplitude) and alters the phase of the other signal path. Phase change depends on the frequency and when the two paths are mixed, the frequencies that are out of phase are cancelled out, creating the notches in the frequency spectrum. Changing the mix ratio of a phaser changes the depth of the notches. The deepest notches (valleys) occur at a mix ratio of 50%. 12 DI (Direct Injection) Box - A device typically used in recording studios to connect a high impedance, line level, unbalanced output signal to a low impedance mic level balanced input, usually via XLR connector. DIs are frequently used to connect an electric guitar or electric bass to a mixing console's microphone input. The DI performs level matching, balancing, and either active buffering or passive impedance matching/impedance bridging to minimise noise, distortion, and ground loops. On a DI box there is often a second output connector called a pass-through connector, sometimes simply paralleled to the input connector, that delivers the input signal unchanged to allow the DI unit to be inserted into a signal path without interrupting the signal. Pass-through is also commonly referred to as a bypass. True bypass occurs when the signal goes straight from the input jack to the output jack with no circuitry involved and no loading of the source impedance. False bypass (or simply 'bypass') occurs when the signal is routed through the device circuitry with no intentional change to the signal. However, due to the nature of electrical designs there is almost always some slight change in the signal. The extent of change and how noticeable it may be can vary from unit to unit. DI boxes come in passive and active models. Passive DI - A passive DI unit typically consists of an audio transformer. The turns ratio is typically chosen to match a nominal 50 kΩ signal source (such as the magnetic pickup of an electric guitar) to a 100-200 Ω input of an audio mixer. Typical turns ratios are in the range of 10:1 to 20:1 Cheaper units are more susceptible to hum and passive units tend to be less versatile than active units, but higher quality units are extremely reliable when used as designed. Passive DIs are simpler and do not require batteries. There is usually a ground lift switch on all DI boxes to eliminate ground loop hum where applicable. 13 Active DI - An active DI unit contains a preamplifier and can therefore provide gain, and are inherently more complex and versatile than passive units. They require a power source, which is normally provided by batteries or a standard AC outlet connection and may contain the option for phantom power use. Most active DI units provide switches to enhance their versatility. These may include gain or level adjustment, ground lift, power source selection, and mono or stereo mode. Ground lift switches often disconnect phantom power on an active unit. Direct Current (DC) – (also see Alternating Current(AC))The unidirectional flow of electric charge in a constant direction (distinguishing it from alternating current). Direct current is produced by sources such as batteries, thermocouples and solar cells. It may flow in conductors such as wires, semiconductors and insulators. A rectifier can be used to convert AC to DC. Similarly, an inverter is used to convert DC to AC. Dynamic Range – Measured in decibels (dB), in audio engineering, it is the ratio between the amplitude of the loudest possible undistorted sine wave of an audio signal and its RMS (root mean square) noise amplitude (noise floor). Dynamic range in analog audio is the difference between low-level thermal noise in the electronic circuitry and high-level signal saturation resulting in increased distortion and, if pushed higher, clipping. Multiple noise processes determine the noise floor of a system as noise can be picked up from multiple sources such as microphone self-noise, preamp noise, wiring and interconnection noise, media noise, etc. In music, dynamic range is the difference between the quietest and loudest volume of an instrument, part or piece of music. Effects Loop – A signal path from a piece of equipment such as a mixing console routed through an effects unit and then back into the first device> This creates an insertion point within the signal path for inserting special audio equipment such as effects processors (delay, reverb, chorus, phase shifters, etc.). 14 Effects Unit – Electronic device used to alter the sound of a musical instrument, voice or other source. Some effects are subtle, only lightly coloring the sound while others give a very dramatic effect. Common examples are: wah pedals, fuzzboxes, delay units, and reverb units. The most common formats of effect units are “stompbox” (or pedal) types operated by foot, and “rack mount” equipment. Effects generally fall into the following classifications: dynamic, time-based, tonal, filter, pitch, and feedback/ sustain. Equalizer – (also see Parametric EQ, Graphic EQ) An EQ is a filter, usually adjustable, designed to compensate for the unequal frequency response of some other signal processing circuit or system. In audio engineering, the EQ filter is more often used creatively to alter the frequency response characteristics of a musical source or a sound mix. Typically consisting of more than one adjustable parameter, it is generally used to improve the fidelity of sound, remove unwanted noises and feedback, emphasize certain instruments, or shape timbres. They are designed with many different parameters and features such as: peaking filters, shelving filters, band-pass filters, high and low-pass filters. The 2 main types of peaking equalizers are: parametric and graphic EQ. A notch filter is an EQ that can be tuned to a particular frequency and has a very narrow bandwidth (Q). Frequency - All equalizers built on peaking filters use a bell curve which allows the equalizer to operate smoothly across a range of frequencies. The center frequency occurs at the top of the bell curve and is the frequency most affected by equalization. It is often notated as f and is measured in Hz and kHz. Q - This is a variable (quality factor) which refers to the width of the bell curve mentioned above. The higher the ‘Q’ setting, the narrower the bandwidth (number of frequencies above and below the selected frequency) that is affected. A high ‘Q’ means that only a narrow frequency range around the center frequency is affected, whereas a low ‘Q’ affects a wide frequency range. Gain (Boost/Cut) - This determines how much the selected frequencies should be increased or decreased. A boost means that those frequencies will be louder after being equalized and a cut will attenuate them. The amount of boost or cut is measured in decibels, such as +3 dB or -6 dB. A boost of +3 dB will double the sound power after equalization, but an increase of +10dB is required for the perceived loudness to be twice as loud to the human ear. A good starting point for setting the EQ on specific instruments or vocals may be found in the Chris Lord Alge EQ Cheat Sheet shown below. The sheet is intended for the recording studio, but can be implemented into any live situation as a jump-off point to keep sound sources from interfering with one another. When sources interfere with one another within the frequency spectrum, they combine to create a “muddy” mix where instruments and vocals blend or become lost. For a list of fundamental & harmonic frequencies of common instruments as well as the properties of specific frequencies for adjusting EQ, see the chart under “frequency”. 15 16 Fader – A variable attenuator, volume control or potentiometer. Found on almost all mixers, the fader control is a potentiometer that works by sliding the fader control vertically rather than via rotation. The fader-type control helps to provide a graphic representation of the relative levels of multiple channels on a mixing console, but can be used for adjusting other parameters as well, depending on the manufacturer of the equipment implementing it. Feedback - The return of a portion of the output of a process or system to the input of that same system. In audio this usually occurs when a microphone picks up the sound output from a nearby loudspeaker reinforcing the signal being input through that microphone. The signal is repeatedly input by the microphone and amplified by the system in a phenomenon known as feedback loop. If too much of this "feedback" occurs the signal will quickly degrade into an oscillation at some problematic frequency. The produced "squeal" from this frequency is what we generally call feedback although technically feedback occurred well before it was identified. Audio quality and the amount of potential feedback depends on the sound system, the room's acoustic characteristics, the number of audience members, placement of microphones, and a number of other factors. Controlling Feedback – In indoor venues, room modes substantially affect a sound reinforcement system’s susceptibility to feedback. The feedback will lock into the room’s resonant frequencies and will generally favour a particular frequency (although feedback can occur at any frequency) depending on room dimensions and microphone placement. A well trained vocalist holding their microphone correctly (close to the sound source) is a huge factor in controlling feedback in any situation because the gain can be appropriately set by the sound engineer and not have to be turned up too high (hot) to get a good signal level. A “hot” microphone becomes very sensitive and picks up the return signal from a nearby loudspeaker very easily. EQ filters can be a very effective way of controlling feedback by notching out the offending frequencies. This is called “ringing out” the system and is done by inserting individual graphic Eqs (see Graphic EQ) into the signal chain before each monitor loudspeaker amplifier and then using the procedure of ringing out the system to find and turn down the problematic frequency bands. This must be done carefully in order not to damage system components and not to destroy the sound quality by over equalizing. Ringing out the monitors can improve the amount of headroom in the monitoring system, allowing for greater gain settings (if required) without feedback. 17 Fletcher-Munson Curves – An “equal loudness contour” graphically depicting the way that the human ear hears frequencies of the same amplitude. Equal amplitude frequencies are perceived as not being of the same loudness by the human ear and therefore, the human ear does not possess a flat frequency response. We hear higher frequencies as being louder than lower frequencies. Frequency – (also see Hertz, Speed of Sound, Sound Wave & Wavelength) Frequency is the number of times something occurs in a unit of time. In the audio world the frequency of sound vibrations are directly related to what we hear as pitch (440 Hz = A above middle C, Middle C (C4) = 261.63 Hz), though the relationship is NOT linear. It is also inversely related to wavelength. We use the word frequency, and the values associated with it, as an objective way to speak about sound characteristics. Frequencies are measured in Hz (hertz). For formulas to calculate the wavelengths of specific frequencies, see “Speed of Sound & Wavelength”. Audible Frequencies – frequencies between 20Hz – 20,000Hz. They are heard by the human ear. Some sources claim the range to be between 16Hz – 16,000Hz. The audio spectrum can be divided into four main frequency bands: Band Low Low-Mid Hi-Mid Hi Frequency Range 20 Hz – 200 Hz 200 Hz – 1000 Hz 1000 Hz – 5000 Hz 5000 Hz – 20000 Hz 18 Sample wavelengths of audio frequencies: Piano Note Frequency (f) Key# (Hz) 20 Hz 27.5 Hz 1 (1st key) A0 261.63 Hz 40 C4 (mid. C) 440 Hz 49 A4 64 C6 (soprano C) 1046.5 Hz 4186.01 Hz 88 (last key) C8 20,000 Hz Wavelength (l) (m) 17.2 m 12.5 m 1.31 m 0.78 m 0.329 m (32.9 cm) 0.082 m (8.2 cm) 0.0172 m (1.72 cm) Just for fun, a tube of ½l (half the wavelength) of sound will resonate that note or pitch when blown across the top. This is the principal of an organ. Infrasonic Frequencies – low frequencies between 16Hz – 0.001Hz. They cannot be heard by the human ear. Sample wavelengths of infrasonic frequencies: Frequency (f) 16 Hz 10 Hz 0.1 Hz 0.001 Hz Wavelength (l) 21.5 m 34.4 m 3400 m 344 km Ultrasonic Frequencies – high frequencies above 20kHz. They cannot be heard by the human ear. Sample wavelengths of ultrasonic frequencies: Frequency (f) 20 kHz 34.4 kHz 1000 kHz (1 MHz) 10 MHz Wavelength (l) 1.72 cm 1.0 cm 0.34 mm 0.034 mm The chart shown below indicates the fundamental and harmonic frequencies of many sound sources as well as the properties that the frequency ranges contain. 19 20 Frequency Range – The range that an audio device can reproduce a signal without reference to the accuracy or “flatness” of the actual response across the frequency range. The human ear has a frequency range of 20 – 20,000 Hz (although the range varies between individuals). The frequency range of a Shure SM57 microphone can be seen below in the frequency response curve (range = 40 – 15,000 Hz). Frequency Response – The measure of any system’s output spectrum in response to an input signal. It is usually in reference to electronic amplifiers, loudspeakers, and microphones. A frequency response curve is used to indicate the accuracy of a component in the audio system or the system as a whole over a given range. If a component can reproduce a signal without any attenuation or emphasis on any frequency band across it’s frequency range, it is said to have a “flat” response. Any frequency response that is not “flat” colors the sound in some way. The response curve shown below is for a Shure SM57 dynamic microphone. Gain – (also see Trimpot & Gain Structure) a measure of the ability of a circuit (often an amplifier) to increase the power or amplitude of a signal from the input to the output. Gain is measured in decibels (dB). Gain Knob – see Trim Pot Gain Structure (Gain Staging) – Refers to the aligning of all units making up a sound reinforcement system so that their gain levels match and will potentially all reach distortion simultaneously. A properly set gain structure will greatly reduce noise and/or distortion as well as maximize the potential of the PA system. If a mixing console does not feature a level meter on each individual channel, then appropriate use of the PFL and PFL metering is required to set the gain structure. An incorrectly set gain structure anywhere along the signal path containing a signal that is too weak or too strong will produce poor sound quality. A weak signal introduces noise while a strong signal causes clipping and distortion. A noisy, distorted signal can potentially damage system components. Gate - A dynamics device whose function is to remove unwanted audio material below a certain threshold. When the signal falls below a set the gate will quickly drop the audio level down to a predetermined output level. This level is usually very low, or completely off. Gates are often used on drums to prevent bleed from other nearby drum 21 microphones, and they may be used on noisy sound sources so when the desired audio signal stops, the unwanted noise is attenuated or shut off. Common control settings found on a noise gate are: Threshold – sets the gain level at which the gate will open to allow a signal pass. Attack – sets the amount of time it takes for a gate to go from closed (or the attenuated level set by the range control) to open – similar to a fade-in. Release - sets the amount of time taken for a gate to go from open to closed (or amount of attenuation set by the range control) – similar to a fade-out. Range – sets the amount of attenuation applied when the gate is closed. Hold – sets the amount of time the gate will remain open once the signal falls below the threshold. Graphic EQ - A graphic equalizer uses predetermined Q and frequency ranges which are equally spaced according to the musical intervals, such as the octave (12-band graphic EQ) or one third of an octave (36-band graphic EQ). These frequency ranges can then be independently boosted or cut. This type of EQ is often used for live applications, such as concerts and especially for ringing out a monitoring system to reduce feedback potential. Ground Lift – Found on many pieces of audio equipment, it is a switch used to defeat a signal’s path to ground. Breaking the ground path can be advantageous in eliminating ground loop hum. The ground lift switch effectively disconnects audio signal ground from earth or chassis ground. Ground Loop – An unwanted current in a conductor that connects two points that are supposed to be at the same potential (often ground), but are at different potentials. Simply, it is a phenomenon that occurs when audio systems have multiple paths and path lengths to ground. Ground loops that are caused by incorrectly designed or installed equipment are one of the major causes of noise and interference in an audio system. The hum which is heard is at 60Hz or a multiple of 60Hz because in the US and Canada that is the typical line frequency. One of the only ways to effectively get rid 22 of ground loop hum is to be sure that all of the audio equipment within an audio system has only one path to ground. It is not recommended to ground lift an AC plug on offending pieces of equipment as this can be unsafe. Headroom – Is the difference between the normal operating level of a device, and the maximum level that device can pass without distortion. If there is not enough headroom the equipment may clip (distort) during brief high-level transients. Hertz – (also see Frequency) Equivalent to cycles per second and designated as the symbol Hz, is most commonly used as a description of the sine wave, particularly in radio and audio applications. Hi-Z – High output impedance (a.k.a., tri-stated or floating). Devices with impedances ranging into the 1000’s of ohms are said to have high impedance. They have a high voltage and a low current. Hi-Z is useful with tube amps because tubes naturally have high impedance inputs. Quality guitar amps are still made with tubes and because guitar patches are usually short, they don't have problems with line loss, but they are more prone to hum and buzz. A long run of cable with a high impedance signal would have problems with line loss and signal degradation. Hi-Z outputs can be converted to Lo-Z by utilizing a transformer such as those found in Direct Injection boxes. Typically, guitars require a Hi-Z input although they are usually connected to a DI box and then into a mic input using phantom power. Impedance – (also see Resistance) Measured in ohms (Ω), impedance refers to the resistance of a circuit or device on AC power. Input – A port or peripheral device for receiving signals or data to be received from any source to any information processing system. An input device may be a microphone which has an output plug for connection to an XLR cable. The cable is then plugged into a mixer (information processing device) via its XLR or 1/4" inputs. Input Level types: Microphone level – A normally weak signal level transmitted by the transducer within a microphone (typically around 2mV). The signal is too weak to be processed by other equipment or to be transmitted over long distances without signal loss or introducing noise into and distorting the signal. The signal from a microphone requires a preamplifier to boost it to line level, the level of signal strength required by devices such as mixing consoles. Line Level – A term used to denote the strength of analogue sound transmissions between components in an audio system. Such components include: CD and DVD players, TVs, and mixing consoles. In contrast, weaker signals come from pickups in instruments and microphones while stronger signals may be from equipment such as those used to drive loudspeakers. Signal strength does not necessarily correlate with the output voltage of the equipment or device and usually changing the volume setting on the source equipment does not vary the strength of the line out signal. A line level describes a line's nominal signal level as a ratio (in decibels) against a standard reference voltage. The nominal level and the reference voltage depend on the line level being used. While nominal levels vary, there are only two common reference voltages: one is for consumer applications – measured in decibel volts (dBV), and one is for professional applications – measured in decibels unloaded (dBu). The reference voltage for the decibel volt (0 dBV) is 1 VRMS 23 In the US and Canada, the nominal voltage levels for a professional audio line level (+4dBu) signal is approximately 1.228 VRMS with a peak amplitude of approximately 1.737 VPK. Line level signals use AC current. Acoustic sounds like voices and guitars are often recorded through transducers (microphones and pickups). These transducers produce a weak signal which requires amplification to a line level signal where they are more easily manipulated by downstream equipment such as mixing consoles. Preamplifiers are typically used for this task. Instrument Level – The signal level put out by an instrument such as an electric bass guitar, electric guitar, or the pickup in an acoustic guitar. There is no set standard for an instrument level signal, but it generally falls somewhere between the voltage of a mic level and a line level signal. A passive or piezo pickup in an acoustic guitar can be only a few millivolts where instruments with active pickups or built-in preamplifiers can be line level in their strength. Limiter – A dynamic processor with a circuit that allows signals below a determined input power to pass unaffected while attenuating the signal peaks that surpass the predetermined input power. It is similar to a compressor and many compressors can act as limiters when set correctly. The main difference is the ratio that is used in reducing the signal’s gain. The ratio in a limiter is set up to be as close to infinity : 1 as possible so that no matter how much the input signal changes, the output level will remain constant. In theory, a limiter establishes a maximum gain setting and prevents signals from surpassing that setting. Loudspeaker - A loudspeaker (or "speaker") is an electro-acoustic transducer that converts an electrical signal into sound. The speaker cone moves in accordance with the variations of that electrical signal and causes sound waves to propagate through a medium such as air. Lo-Z – Low output impedance. Devices with impedances up through 600 ohms are said to have low impedance. They have a low voltage and a high current and don't lose high frequencies over long cable runs. Lo-Z devices are not overly sensitive to hum and buzz and can be converted to Hi-Z utilizing a transformer. Microphone – A microphone is a transducer, or instrument by which sound generates an electric current usually for the purpose of transmitting or recording that sound. All microphones translate sound waves into mechanical vibrations via a thin, flexible diaphragm. These vibrations are then converted by various methods into an electrical signal and sent to a signal processing device. All microphones have sonic characteristics which color the sound source in some way. Always check the microphones frequency response graphs and understand what microphones are useful for what sound sources (vocals, guitar amps, etc.) and which ones require phantom power or may be damaged by phantom power. The word “microphone” originates from the Greek words micro, meaning ‘small’, and phone, meaning ‘voice.’ Condenser Mic – Also known as a capacitor or electrostatic microphone, the diaphragm acts as one plate of a capacitor while the other plate is fixed. The capacitor is capable of storing an electrical charge and as the plates charge by an internal DC power supply, the vibrations made by sound pressure variances against the diaphragm produce changes in the distance between the plates which in turn, produces a voltage change across the capacitor. The change in voltage is the output signal. 24 The signal from the diaphragm has a very high impedance and is fed through an impedance conversion amplifier. The amp is very close to the diaphragm to prevent noise pickup and signal losses. The amplifier requires a DC power supply to operate which is why phantom power is required for the microphones to operate. Electret Condenser Mic – Similar to a condenser microphone except that the polarizing charge required for operation is internally stored within the diaphragm. Electret microphones do not require DC phantom power to operate. Modern mics use FET (field effect transistors) to reduce capsule impedance while older and newer versions of vintage mics use vacuum tubes. Tubes exhibit sonically favourable colouration and harmonic distortion that is missing from mics using FET. Dynamic Mic – Works by electromagnetic induction. The dynamic mic is very robust, inexpensive and resistant to moisture. They posses high gain potential before feedback and are ideal for on-stage use. They use a moving coil on the same principal as a loudspeaker, but in reverse. The small movable induction coil, positioned in the magnetic field of a permanent magnet, is attached to the diaphragm. Sound waves move the microphone’s diaphragm and the coil moves in the magnetic field. This movement produces a varying current in the coil via electromagnetic induction. A single dynamic membrane does not respond linearly to all audio frequencies and therefore, some microphones utilize multiple membranes for the different parts of the audio spectrum. Combining multiple signals correctly is difficult and designs that do this are often very expensive. Dynamic mics require a lot of gain (around 60 dB) to operate optimally. Ribbon Mic – A dynamic microphone that uses a thin aluminum, duraluminum, or nanofilm ribbon set between the two poles of a magnet to produce voltages by electromagnetic induction. Ribbon microphones are usually bidirectional which means that they will pick up sounds equally from either side of the microphone. Microphone polar patterns – A microphone's directionality or polar pattern indicates how sensitive it is to sounds arriving at different angles about its central axis. An omnidirectional microphone’s response is considered a perfect sphere in 3 dimensions. The polar pattern is a function of frequency. Because the microphone is not infinitely small it tends to get in the way of itself with respect to sounds arriving from the rear, causing a slight flattening of the polar response. This flattening increases as the diameter of the microphone (assuming it's cylindrical) reaches the wavelength of the frequency in question. Therefore, the smallest diameter microphone gives the best omnidirectional characteristics at high frequencies. The wavelength of sound at 10 kHz is just over 1” (3.4 cm) so the smallest measuring microphones are often 1/4" (6 mm) in diameter, which practically eliminates directionality even up to the highest frequencies. They do not employ resonant cavities as delays (like cardioids do), and so can be considered the "purest" microphones in terms of low coloration (they add very little to the original sound). Because omnidirectional mics are pressure-sensitive they can have a very flat low frequency response down to 20 Hz or lower. Pressure-sensitive microphones also respond much less to wind noise and plosives than directional microphones which are velocity sensitive. A unidirectional microphone is sensitive to sounds from only one direction. The diagrams below illustrate a number of unidirectional patterns and provide an overview of typical pattern shapes and their names. 25 The most common unidirectional microphone is the cardioid and is named for its heart-shaped pattern. Hypercardioids are similar except that they have a tighter area of front sensitivity and a smaller lobe of rear sensitivity. Supercardioids are similar to hypercardioids except that they have more front pickup and less rear pickup. These three polar patterns are commonly used on vocal/ speech microphones because they are good at rejecting sound sources from other directions. Bidirectional or “figure of 8” microphones receive sound from both the front and the back. Most ribbon microphones have bidirectional patterns. Omnidirectional 26 Subcardioid Cardioid 27 Supercardioid Hypercardioid 28 Bi-directional (figure of 8) Shotgun 29 Directional Response – It is the variation in sensitivity at different angles with respect to the on-axis response of a given microphone. Off-Axis Response – The frequency response of a microphone does not take into account the off-axis response. Some microphone designs have erratic response curves off-axis. This can create coloration when off-axis sounds such as leakage reach the microphone’s diaphragm. This curve is occasionally put on the same graph as the on-axis directional response curves within the microphones specification sheets. Microphone Placement – See the Shure documents entitled “Microphone Technique for Recording” and “Microphone Technique for Live Sound”. MIDI - Musical Instrument Digital Interface – MIDI is an industry-standard protocol that allows electronic musical instruments, such as synthesizers and drum machines, to communicate and synchronize with one another. MIDI devices used to trigger musical sounds are often called "controllers", because with most MIDI set-ups, the keyboard or other device does not make any sounds by itself. MIDI controllers need to be connected to a voice bank or sound module to produce musical tones or sounds. The keyboard or other MIDI device "controls" the voice bank or sound module by acting as a trigger. The most common MIDI controller is the piano-style keyboard. 30 31 Mixing Console - In audio, the mixing console (mixing board, audio mixer, mixing desk) is an electronic device for combining, routing, or changing the level, tone, and dynamics of a single or multiple audio signals. Mixing consoles are used in professional recording studios, PA and sound reinforcement systems, broadcasting, and film postproduction. Modulation – Modulation is change. It is the process of varying one or more properties of a high frequency periodic waveform (carrier signal) with respect to a modulating signal. The three key parameters of a periodic waveform are: amplitude (volume), phase (timing), frequency (pitch). All three can be modified within the carrier signal with a control signal such as a low frequency signal to obtain the modulated signal. Many synthesis sounds are based around using the frequency of one signal to change the frequency of another. AM radio works because of amplitude modulation. Monitors (“foldback” cabinets) - In a recording environment, monitors are the loudspeakers used to play back the live signals and recorded tracks of a project. In live applications using a sound reinforcement system, monitors are used to provide the stage talent with an individual customized reference mix of the live performance (a.k.a., cue mix). The monitors can be wedge speakers, headphones, in-ear monitoring systems or a mixture of any of these. Noise Floor - The noise floor is the measure of the signal created from the sum of all the noise sources and unwanted, interfering signals within a measurement system with no intended signal present. The noise floor is measured in decibels and all electronic devices will generate a certain amount of noise. Minimizing the noise floor increases dynamic range and produces cleaner sound. A common way to lower the noise floor is to cool the system, reducing thermal noise, which is usually the major noise source. The noise floor can also be artificially lowered with digital signal processing techniques. Noise Gate – see Gate 32 Ohm’s Law – States that the current through a conductor between two points is directly proportional to the potential difference or voltage across the two points, and inversely proportional to the resistance between them (I = E / R). Quantity Current Voltage Resistance Ohm’s Law Unit Wheel Symbol I E or V R Unit of Measurement Ampere (amp) Volt Ohm Unit Abbreviation A V Ω Formula Solver Triangle Sample Circuit Output - A port from any device which allows a signal or data to be passed from it to any other device. An output may be an XLR jack, 1/4” jack, RCA outputs or a speaker (among others). PAD – (also see Attenuate & Attenuator) An attenuator which reduces the amplitude of a signal. The PAD function is found on many audio devices including mixing consoles, select DI boxes, and some microphones. It helps to reduce the signal-to-noise ratios and should only be used during high sound pressure levels (mics) or to PAD hot line level signals (e.g., -20dB pad on a mixing console for a channel with a CD player input). PA System – (also see Sound Reinforcement System) PA is an acronym for Personal Address. The PA System is an electronic amplification system utilizing a signal mixer, amplifier(s) and loudspeaker(s) to reinforce a sound source such as the human voice and then distribute that sound throughout a venue or building (e.g. school, institute, commercial building or office). There is disagreement over when to call these type of audio systems ‘sound reinforcement’ systems or ‘PA’ systems. Some audio engineers distinguish between them via the technology used and their capability, while others by their intended use (e.g. sound reinforcement systems are for live music and PA systems are for the reproduction of speech and recorded music in buildings and institutions). The terms are often used interchangeably. 33 Parametric EQ - A parametric equalizer uses independent parameters for Q, center frequency, and boost/cut. Any range of frequencies can be selected and then processed. This is the most powerful EQ because it allows control over all three variables and is predominantly used in recording and mixing. Passive – In the context of audio, passive equipment (DI box, loudspeaker, guitar pick-up, etc.) requires an external power supply to operate the unit. Phantom Power – Best known for a convenient power source for condenser microphones, in the context of professional audio equipment, phantom power is a method for transmitting DC electric power through microphone cables (usually XLR) to operate microphones that contain active electronic circuitry. It consists of direct current applied equally through the two signal lines of a balanced audio connector. The supply voltage is referenced to the ground pin of the connector (pin 1 of an XLR), which normally is connected to the cable shield or a ground wire in the cable or both. With phantom power, the supply voltage is effectively invisible to balanced microphones that do not use it, which includes most dynamic microphones. A balanced signal consists only of the differences in voltage between two signal lines; phantom powering places the same DC voltage on both signal lines of a balanced connection. Many DI (direct injection) boxes use phantom power and it is built in to mixers, microphone preamplifiers and other audio equipment. Traditional condenser microphones also use phantom power for polarizing the microphone's transducer element. Phantom powering can cause equipment malfunction or even damage if used with cables or adapters that connect one side of the input to ground, or if certain equipment other than microphones are connected to it. If uncertain, always check the equipment specifications to see if phantom power may damage the unit in question and be sure of the cables you’re using. Phase – The fraction of a wave cycle which has elapsed relative to an arbitrary point. Phase Shift – in audio, a phase shift is a change that occurs in the phase of a sinusoidal wave over time. An example of a phase shift is shown below where the horizontal axis represents an angle (phase) which increases with time.  represents the angle of shift and a 180 shift in phase results in a complete cancellation of amplitude of a signal (phase cancellation). PFL – acronym for ‘Pre Fade Listen’ provides individual channel metering and allows the engineer to listen to the input signal of a channel via headphones or a speaker prior to sending the signal to the house mix and auxiliary outputs (monitors or effects). 34 Potentiometer - a potentiometer (also known as a "pot") is a three-terminal resistor with a sliding contact that forms an adjustable voltage divider. If only two terminals are used (one side and the wiper), it acts as a variable resistor or rheostat. Potentiometers are commonly used to control electrical devices such as volume controls on audio equipment. Power – (also see Ohm’s Law)Is the measure of the amount of work that can be done in a given amount of time. Electrical power is almost always measured in “watts”. Power is equal to current x voltage. 1 horsepower = 745.7 watts. Preamplifier – (also see Amplifier) An amplifier designed to amplify very weak signals before they are sent to downstream gain stages or devices. They often “color” the sound of the audio signal. The preamp is most commonly used to amplify the input signal from microphones and guitar pick-ups to line level. They are a critical component of the audio chain and usually the first stage in setting the gain structure. If not set properly, as in any device, the preamplifier can introduce noise into the signal chain. The most common preamplifier is that found on the channel strip of a mixing console. Proximity Effect – In audio, proximity effect is an increase in bass or low frequency response when a sound source becomes close to a microphone. It is cause by the microphones’ ports which are designed to create directional polar patterns. Therefore, omnidirectional microphones are not affected by it. In general, the effect is focused on frequencies below 100Hz. Resistance – (see Ohm’s Law) A measurement of the opposition of a body or substance to current passing through it. Resistance (measured in ohms Ω) causes a change of electrical energy into heat or another form of energy. All electrical circuits and wires have some amount of resistance to the passing of electrical current. The ohm was named after George Ohm, who formulated Ohm's Law. The resistance of a given resistor or conductor grows with the length of conductor and decreases for larger cross-sectional areas. The resistance R of a conductor of uniform cross section, therefore, can be computed as where is the length of the conductor, measured in meters (m), A is the crosssectional area of the conductor measured in square meters (m²), and ρ (Greek: rho) is the electrical resistivity (also called specific electrical resistance) of the material measured in ohm-metres (Ωm). Resistivity is a measure of the material's ability to oppose an electrical current. The resistivity of copper at 20°C = 1.67 × 10-8 Ωm (0.0000000168 Ωm) The resistivity of metals increases with temperature. 35 For practical reasons, any connections to a real conductor will almost certainly mean the current density is not totally uniform. However, this formula still provides a good approximation for long thin conductors such as wires. Reverberation – The presence of sound in an enclosed space after the original sound is removed. Basically, reverb is a large number of closely spaced echoes that slowly decay as they are absorbed by the surroundings of the space (people, furniture, walls, air, etc.). Some common reverb effects for recording and live music are: chamber, room, hall, plate, spring and digital. Digital reverbs use signal processing algorithms to create the reverb effect. The first early reflections that reflect back to the listener in a given space is what gives the listener the perception of size and space. The later reflections, which are so close together that the listener cannot discern each individual reflection, is perceived as a single decaying signal (reverb). Ringing Out a System – see Feedback Signal Path – The path in which electrical signals flow through the various components of the sound system. The signal always begins with a sound source (e.g., human voice, instrument). As an example, although a human voice may be the sound source, the electrical signal source of that sound begins with the microphone (transducer) where the sound waves are converted into voltage. A mixer will receive the relatively weak signal from the microphone and amplify it through various stages of its circuitry. The path continues through the mixer outputs into an amplifier which in turn sends the signal to a loudspeaker to be converted back into sound waves. Signal to Noise Ratio (SNR) – The measurement of noise power in a device compared to signal power within a signal path. Technically, it measures how much a signal has been corrupted by noise. Higher signal to noise ratios are better as the difference between the noise floor and the desired signal is greater (the noise is less obtrusive). The concepts of signal-to-noise ratio and dynamic range are closely related. Sound Reinforcement System – (also see PA System) A system utilizing the combination of microphones, signal processors, amplifiers, loudspeakers, and other technologies to make live or pre-recorded sounds louder and to even enhance them. They are often intended to also distribute those sounds to a larger or more distant audience such as those found at a concert. The sound reinforcement system can be very simple (using only a single microphone and self-powered loudspeaker system) or immensely complex. By some, the sound reinforcement system is interchangeably called a PA system. Sound Wave – (also see Speed of Sound & Wavelength) A series of mechanical compressions (hi-pressure) and rarefactions (low-pressure) that successively propagate through air or any medium that is at least a little compressible or elastic (solid, liquid, or gas, but not vacuum). They are created by the back and forth vibration of the particles of that medium. See the below diagram for a visual example of a tuning fork propagating sound waves through the air in an open tube. 36 Elevation, air temperature and humidity affect sound waves. Why elevation? Because at higher elevations the temperature decreases and there is less oxygen so the composition of air changes. See “Speed of Sound & Wavelength” for specific formulas to calculate velocity, wavelength and frequency. When a detection device (e.g., human ear or microphone) detects a sound wave, it detects the fluctuations in pressure that the sound wave impinges upon it. Combining Waves – When multiple waves meet together they combine. The two or more sound waves may be from the same source or different sources, whether a direct sound combining with a reflected wave from the same sound source or waves of the same frequency from independent sources such as a bass guitar and a bass drum coming together. Combining waves with the same frequency and polarity will lead to a new combined sound wave that is equivalent to the sum of each wave’s amplitudes and therefore, stronger, but with the same frequency. This is because both sound waves are considered to be in phase with one another. The example below shows two sound waves of equal amplitude being combined and so their sum amplitude is 2x the original strength of the sound wave. Combining waves that are of the same frequency and amplitude, but have their polarity opposite one another when they meet will cancel one another out because they are considered completely out of phase. The example below shows two out of phase sound waves of equal amplitude being combined and so their sum amplitude is zero. In live sound reinforcement, phase cancellation is often what causes “dead spots” within a room. Two combining waves that are in different parts of their cycle, but not completely out of phase (see Phase, Phase Shift) will subtract from each other and create a thinner 37 sound rather than reinforcing the sound as in the first example. This is called “phase cancellation”. Speaker – see Loudspeaker Speed of Sound – The speed of sound is a measurement of the distance traveled in a unit of time by a sound wave through an elastic medium. . In air, the speed of sound is 1126 feet (or 343.2 meters) per second at sea level at a temperature of 20°C. The speed of sound varies from substance to substance and travels faster through liquids and non-porous solids than it does through air. The speed of sound in water is about 4.3 times faster than in air and approximately 15 times faster in iron with the air temperature = 20°C. Sound Formulas: 1) Speed of sound based on frequency and wavelength v=fxl v = velocity (m/s or ft/s) f = frequency (Hz – cycles/sec.) l = wavelength (m or ft) 2) Speed of sound in DRY air based on temperature v = 331.4 + 0.6Tc ** The 331.4m/s is based on Tc = 0°C. This is an approximate formula for DRY air. v = velocity of sound (m/s) Tc = temperature (Celsius) 3) Theoretical speed of sound in an ideal gas based on density v = √ (γP / p) v = speed of sound (m/s) P = frequency (Hz – cycles/sec.) p = density γ = adiabatic constant of the specific gas (γ of DRY air = 1.4) 4) Temperature dependent speed of sound for a gas (very accurate for DRY air) Vsound = √ (γRT / M) R = gas constant (8.314 J/mol K) γ = adiabatic constant of the specific gas (γ of DRY air = 1.4) M = molecular mass of the gas (Kg/mol) (M of air = 0.02895 Kg/mol) T = absolute temperature (kelvin) where Tkelvin = Tcelsius + 273.15 The speed of sound in a gas is a function of temperature, molecular structure, and molecular mass. The molecular mass is the atomic weight of the molecule / 1000. 38 The speed of various gases at 0°C is: Gas DRY Air Carbon dioxide Oxygen Helium Hydrogen Speed (m/s) 331.4 259 316 965 1290 Using formulas 2 or 4 above, examples of the speed of sound in DRY air for a given temperature are: Tc (°C) 0 20 30 21 v (m/s) 331.4 343.4 349.4 344 (ft/s) 1087.27 1126.64 1146.325 1128.61 All the above formulas with respect to air assumed DRY air. Humidity (moisture) affects the density of air and hence, using formula 3 above, the speed of sound. Moist air is not as dense as dry air and therefore, “p” (density) becomes smaller which causes an increase in the speed of sound. Moisture also causes the specificheat ratio (adiabatic constant γ) to decrease, which would cause the speed of sound to decrease, but the decrease in density is dominating and so the speed of sound will increase with the increase of moisture in the air. To accurately include the effects of moisture on the speed of sound, two variables need to be modified from formula #4: γ (specific-heat ratio = 1.4 for dry air) and M (molecular weight of molecules in the air). We will not go into the mathematical formulas for determining the velocity of sound with regard to humidity, but need to understand that it does affect the speed of sound. Transducer – A transducer is an electronic component that transforms one type of energy into another. Some examples are: microphones (sound waves into electric current), loudspeaker (electric current into sound waves), piezo guitar pick-ups, and tape heads. Transformer – A device that transfers electrical energy from one circuit to another through inductively coupled conductors (transformer’s coils). Two conductors are referred to as being inductively coupled when they are configured so that any change in current flow through one wire will induce a voltage across the ends of the other wire via electromagnetic induction (production of voltage across a conductor moving through a magnetic field). Transients – A transient is a short-duration signal that represents a non-harmonic attack phase of a musical sound or spoken word. Or more simply, it is a non-repeating waveform, 39 usually of much higher level than the surrounding sounds or the average level. Examples include: the attack of many percussive instruments, the "pluck" of a guitar note, consonants within human speech (e.g., a "T" sound), etc. Transients are difficult to record and reproduce, eating up precious headroom, and often resulting in overload distortion (signal clipping). Careful use of compression can help in taming transients and raise the average audio level, but over-compression can result in a dull, squashed, flat sound. Transient Response – The time it takes for a microphone’s diaphragm to respond to a waveform. The transient response varies greatly between microphones and is a leading factor in the sound qualities of the different mic types. For instance, condenser mics have very thin diaphragms that allow them to react quickly to changes in sound pressure levels and transfer the waveform more accurately over their frequency range. Trimpot - in 1952, Marlan Bourns patented the world's first trimming potentiometer, trademarked "Trimpot", a name now commonly used to refer to any trimming potentiometer. Unbalanced Connection – see Connection (signal) Types Wavelength – (also see Speed of Sound & Sound Wave) The distance between one peak or crest of a sine wave and the next corresponding peak or crest. The symbol for wavelength is the Greek letter lambda (l). The wavelength of any frequency is determined by dividing the speed of sound by the frequency (see the formula below). In air, the speed of sound is 1126 feet (or 343.2 meters) per second at sea level at a temperature of 20°C. The wavelength of a 60Hz sine wave would be approximately 18.8 ft. Knowing wavelengths is very important when designing and working with acoustical spaces (e.g., studios, control rooms, speaker enclosures). Wavelength formula: l=v/f l = wavelength (m or ft) f = frequency (Hz – cycles/sec.) v = velocity (m/s or ft/s)** ** the speed of sound for different temperatures can be calculated using the formula shown under “Speed of Sound” Wedge – a speaker, usually referring to a stage monitor due to their “wedge-like” shape. 40 A Shure Educational Publication MICROPHONE TECHNIQUES LIVE SOUND REINFORCEMENT Microphone Techniques for LIVE SOUND Ta b l e o f C o n t e n t s Introduction ........................................................................... 4 Microphone Characteristics .................................................. 5 Musical Instrument Characteristics..................................... 11 Acoustic Characteristics ..................................................... 14 Microphone Placement....................................................... 22 Stereo Microphone Techniques.......................................... 32 Microphone Selection Guide ............................................. 34 Glossary ............................................................................. 35 Live Sound 3 Microphone Techniques for LIVE SOUND Introduction Microphone techniques (the selection and placement of microphones) have a major influence on the audio quality of a sound reinforcement system. For reinforcement of musical instruments, there are several main objectives of microphone techniques: to maximize pick-up of suitable sound from the desired instrument, to minimize pick-up of undesired sound from instruments or other sound sources, and to provide sufficient gain-before-feedback. “Suitable” sound from the desired instrument may mean either the natural sound of the instrument or some particular sound quality which is appropriate for the application. “Undesired” sound may mean the direct or ambient sound from other nearby instruments or just stage and background noise. “Sufficient” gain-before-feedback means that the desired instrument is reinforced at the required level without ringing or feedback in the sound system. Obtaining the proper balance of these factors may involve a bit of give-and-take with each. In this guide, Shure application and development engineers suggest a variety of microphone techniques for musical instruments to achieve these objectives. In order to provide some background for these techniques it is useful to understand some of the important characteristics of microphones, musical instruments and acoustics. Introduction 4 Microphone Techniques for LIVE SOUND Microphone Characteristics The most important characteristics of microphones for live sound applications are their operating principle, frequency response and directionality. Secondary characteristics are their electrical output and actual physical design. Operating principle - The type of transducer inside the microphone, that is, how the microphone picks up sound and converts it into an electrical signal. A transducer is a device that changes energy from one form into another, in this case, acoustic energy into electrical energy. The operating principle determines some of the basic capabilities of the microphone. The two most common types are Dynamic and Condenser. Dynamic microphones employ a diaphragm/ voice coil/magnet assembly which forms a miniature sounddriven electrical generator. Sound waves strike a thin plastic membrane (diaphragm) which vibrates in response. A small coil of wire (voice coil) is attached to the rear of the diaphragm and vibrates with it. The voice coil itself is surrounded by a magnetic field created by a small permanent magnet. It is the motion of the voice coil in this magnetic field which generates the electrical signal corresponding to the sound picked up by a dynamic microphone. Dynamic microphones have relatively simple construction and are therefore economical and rugged. They can provide excellent sound quality and good specifications in all areas of microphone performance. In particular, they can handle extremely high sound levels: it is almost impossible to overload a dynamic microphone. In addition, dynamic microphones are relatively unaffected by extremes of temperature or humidity. Dynamics are the type most widely used in general sound reinforcement. Condenser microphones are based on an electricallycharged diaphragm/backplate assembly which forms a sound-sensitive capacitor. Here, sound waves vibrate a very thin metal or metal-coated-plastic diaphragm. The diaphragm is mounted just in front of a rigid metal or metal-coated-ceramic backplate. In electrical terms this assembly or element is known as a capacitor (historically called a “condenser”), which has the ability to store a charge or voltage. When the element is charged, an electric field is created between the diaphragm and the backplate, proportional to the spacing between them. It is the variation of this spacing, due to the motion of the diaphragm relative to the backplate, that produces the electrical signal corresponding to the sound picked up by a condenser microphone. The construction of a condenser microphone must include some provision for maintaining the electrical charge or polarizing voltage. An electret condenser microphone has a permanent charge, maintained by a special material deposited on the backplate or on the diaphragm. Non-electret types are charged (polarized) by means of an external power source. The majority of condenser microphones for sound reinforcement are of the electret type. All condensers contain additional active circuitry to allow the electrical output of the element to be used with typical microphone inputs. This requires that all condenser microphones be powered: either by batteries or by phantom power (a method of supplying power to a microphone through the microphone cable itself). There are two potential limitations of condenser microphones due to the additional circuitry: first, the electronics produce a small amount of noise; second, there is a limit to the maximum signal level that the electronics can handle. For this reason, condenser microphone specifications always include a noise figure and a maximum sound level. Good designs, however, have very low noise levels and are also capable of very wide dynamic range. Condenser microphones are more complex than dynamics and tend to be somewhat more costly. Also, condensers may be adversely affected by extremes of temperature and humidity which can cause them to become noisy or fail temporarily. However, condensers can readily be made with higher sensitivity and can provide a smoother, more natural sound, particularly at high frequencies. Flat frequency response and extended frequency range are much easier to obtain in a condenser. In addition, condenser microphones can be made very small without significant loss of performance. 5 Microphone Techniques for LIVE SOUND Phantom Power Phantom power is a DC voltage (usually 12-48 volts) used to power the electronics of a condenser microphone. For some (non-electret) condensers it may also be used to provide the polariziing voltage for the element tself. This voltage is supplied through the microphone cable by a mixer equipped with phantom power or by some type of in-line external source. The voltage is equal on Pin 2 and Pin 3 of a typical balanced, XLR-type connector. For a 48 volt phantom souorce, for example, Pin 2 is 48 VDC and Pin 3 is 48 VDC, both with respect to Pin 1 which is ground (shield). Because the voltage is exactly the same on Pin 2 and Pin 3, phantom power will have no effect on balanced dynamic microphones: no current will flow since there is no voltage difference across the output. In fact, phantom power supplies have current limiting which will prevent damage to a dynamic microphone even if it is shorted or miswired. In general, balanced dynamic microphones can be connected to phantom powered mixer inputs with no problem. lightweight condenser diaphragm. It also takes longer for the dynamic diaphragm to stop moving in comparison to the condenser diaphragm. Thus, the dynamic transient response is not as good as the condenser transient response. This is similar to two vehicles in traffic: a truck and a sports car. They may have equal power engines but the truck weighs much more than the car. As traffic flow changes, the sports car can accelerate and brake very quickly, while the semi accelerates and brakes very slowly due to its greater weight. Both vehicles follow the overall traffic flow but the sports car responds better to sudden changes. Pictured here are two studio microphones responding to the sound impulse produced by an electric spark: condenser mic on top, dynamic mic on bottom. It is evident that it takes almost twice as long for the dynamic microphone to respond to the sound. It also takes longer for the dynamic to stop moving after the impulse has passed (notice the ripple on the second half of the graph). Since condenser microphones generally have better transient response then dynamics, they are better suited for instruments that have very sharp attack or extended high frequency output such as cymbals. It is this transient response difference that causes condenser mics to have a more crisp, detailed sound and dynamic mics to have a more mellow, rounded sound. Transient Response Transient response refers to the ability of a microphone to respond to a rapidly changing sound wave. A good way to understand why dynamic and condenser mics sound different is to understand the differences in their transient response. In order for a microphone to convert sound energy into electrical energy, the sound wave must physically move the diaphragm of the microphone. The amount of time it takes for this movement to occur depends on the weight (or mass) of the diaphragm. For instance, the diaphragm and voice coil assembly of a dynamic microphone may weigh up to 1000 times more than the diaphragm of a condenser microphone. It takes longer for the heavy dynamic diaphragm to begin moving than for the 6 Condenser/dynamic scope photo Microphone Techniques for LIVE SOUND The decision to use a condenser or dynamic microphone depends not only on the sound source and the sound reinforcement system but on the physical setting as well. From a practical standpoint, if the microphone will be used in a severe environment such as a rock and roll club or for outdoor sound, dynamic types would be a good choice. In a more controlled environment such as a concert hall or theatrical setting, a condenser microphone might be preferred for many sound sources, especially when the highest sound quality is desired. Frequency response - The output level or sensitivity of the microphone over its operating range from lowest to highest frequency. Virtually all microphone manufacturers list the frequency response of their microphones over a range, for example 50 - 15,000 Hz. This usually corresponds with a graph that indicates output level relative to frequency. The graph has frequency in Hertz (Hz) on the x-axis and relative response in decibels (dB) on the y-axis. A microphone whose output is equal at all frequencies has a flat frequency response. Flat response microphones typically have an extended frequency range. They reproduce a variety of sound sources without changing or coloring the original sound. A microphone whose response has peaks or dips in certain frequency areas exhibits a shaped response. A shaped response is usually designed to enhance a sound source in a particular application. For instance, a microphone may have a peak in the 2 - 8 kHz range to increase intelligibility for live vocals. This shape is called a presence peak or rise. A microphone may also be designed to be less sensitive to certain other frequencies. One example is reduced low frequency response (low end roll-off) to minimize unwanted “boominess” or stage rumble. The Decibel The decibel (dB) is an expression often used in electrical and acoustic measurements. The decibel is a number that represents a ratio of two values of a quantity such as voltage. It is actually a logarithmic ratio whose main purpose is to scale a large measurement range down to a much smaller and more useable range. The form of the decibel relationship for voltage is: dB = 20 x log(V1/V2) where 20 is a constant, V1 is one voltage, V2 is the other voltage, and log is logarithm base 10. Examples: What is the relationship in decibels between 100 volts and 1 volt? dB = 20 x log(100/1) dB = 20 x log(100) dB = 20 x 2 (the log of 100 is 2) dB = 40 That is, 100 volts is 40dB greater than 1 volt. What is the relationship in decibels between 0.001 volt and 1 volt? Flat frequency response dB = 20 x log(0.001/1) dB = 20 x log(0.001) dB = 20 x (-3) (the log of .001 is -3) dB = -60 That is, 0.001 volt is 60dB less that 1 volt. Similarly: if one voltage is equal to the other they are 0dB different if one voltage is twice the other they are 6dB different if one voltage is ten times the other they are 20dB different Shaped frequency response 7 Microphone Techniques for LIVE SOUND Since the decibel is a ratio of two values, there must be an explicit or implicit reference value for any measurement given in dB. This is usually indicated by a suffix on the decibel value such as: dBV (reference to 1 volt which is 0dBV) or dB SPL (reference to 0.0002 microbar which is 0dB Sound Pressure Level) 1. Compare 2. Compress 0 b a b/a 10 =1 101=10 102=100 103=1000 104=10,000 105=100,000 106=1,000,000 3. scale (x 20) 0 20 40 60 80 100 120 Decibel scale for dBV or dB SPL One reason that the decibel is so useful in certain audio measurements is that this scaling function closely approximates the behavior of human hearing sensitivity. For example, a change of 1dB SPL is about the smallest difference in loudness that can be perceived while a 3dB SPL change is generally noticeable. A 6dB SPL change is quite noticeable and finally, a 10dB SPL change is perceived as “twice as loud.” The choice of flat or shaped response microphones again depends on the sound source, the sound system and the environment. Flat response microphones are usually desirable to reproduce instruments such as acoustic guitars or pianos, especially with high quality sound systems. They are also common in stereo miking and distant pickup applications where the microphone is more than a few feet from the sound source: the absence of response peaks minimizes feedback and contributes to a more natural sound. On the other hand, shaped response microphones are preferred for closeup vocal use and for certain instruments such as drums and guitar amplifiers which may benefit from response enhancements for presence or punch. They are also useful for reducing pickup of unwanted sound and noise outside the frequency range of an instrument. 8 Directionality - A microphone’s sensitivity to sound relative to the direction or angle from which the sound arrives. There are a number of different directional patterns found in microphone design. These are typically plotted in a polar pattern to graphically display the directionality of the microphone. The polar pattern shows the variation in sensitivity 360 degrees around the microphone, assuming that the microphone is in the center and that 0 degrees represents the front of the microphone. The three basic directional types of microphones are omnidirectional, unidirectional, and bidirectional. The omnidirectional microphone has equal output or sensitivity at all angles. Its coverage angle is a full 360 degrees. An omnidirectional microphone will pick up the maximum amount of ambient sound. In live sound situations an omni should be placed very close to the sound source to pick up a useable balance between direct sound and ambient sound. In addition, an omni cannot be aimed away from undesired sources such as PA speakers which may cause feedback. Omnidirectional The unidirectional microphone is most sensitive to sound arriving from one particular direction and is less sensitive at other directions. The most common type is a cardioid (heart-shaped) response. This has the most sensitivity at 0 degrees (on-axis) and is least sensitive at 180 degrees (off-axis). The effective coverage or pickup angle of a cardioid is about 130 degrees, that is up to about 65 degrees off axis at the front of the microphone. In addition, the cardioid mic picks up only about one-third as much ambient sound as an omni. Unidirectional microphones isolate the desired on-axis sound from both unwanted off-axis sound and from ambient noise. Microphone Techniques for LIVE SOUND Cardioid Supercardioid For example, the use of a cardioid microphone for a guitar amplifier which is near the drum set is one way to reduce bleed-through of drums into the reinforced guitar sound. The bidirectional microphone has maximum sensitivity at both 0 degrees (front) and at 180 degrees (back). It has the least amount of output at 90 degree angles (sides). The coverage or pickup angle is only about 90 degrees at both the front and the rear. It has the same amount of ambient pickup as the cardioid. This mic could be used for picking up two opposing sound sources, such as a vocal duet. Though rarely found in sound reinforcement they are used in certain stereo techniques, such as M-S (mid-side). Unidirectional microphones have several variations on the cardioid pattern. Two of these are the supercardioid and hypercardioid. Both patterns offer narrower front pickup angles than the cardioid (115 degrees for the supercardioid and 105 degrees for the hypercardioid) and also greater rejection of ambient sound. While the cardioid is least sensitive at the rear (180 degrees off-axis) the least sensitive direction is at 126 degrees off-axis for the supercardioid and 110 degrees for the hypercardioid. When placed properly they can provide more focused pickup and less ambient noise than the cardioid pattern, but they have some pickup directly at the rear, called a rear lobe. The rejection at the rear is -12 dB for the supercardioid and only -6 dB for the hypercardioid. A good cardioid type has at least 15-20 dB of rear rejection. Microphone Polar Patterns Compared 9 Microphone Techniques for LIVE SOUND Using Directional Patterns to Reject Unwanted Sources In sound reinforcement, microphones must often be located in positions where they may pick up unintended instrument or other sounds. Some examples are: individual drum mics picking up adjacent drums, vocal mics picking up overall stage noise, and vocal mics picking up monitor speakers. In each case there is a desired sound source and one or more undesired sound sources. Choosing the appropriate directional pattern can help to maximize the desired sound and minimize the undesired sound. Although the direction for maximum pickup is usually obvious (on-axis) the direction for least pickup varies with microphone type. In particular, the cardioid is least sensitive at the rear (180 degrees off-axis) while the supercardioid and hypercardioid types actually have some rear pickup. They are least sensitive at 125 degrees off-axis and 110 degrees off axis respectively. For example, when using floor monitors with vocal mics, the monitor should be aimed directly at the rear axis of a cardioid microphone for maximum gain-before-feedback. When using a supercardioid, however, the monitor should be positioned somewhat off to the side (55 degrees off the rear axis) for best results. Likewise, when using supercardioid or hypercardioid types on drum kits be aware of the rear pickup of these mics and angle them accordingly to avoid pickup of other drums or cymbals. Other directional related microphone characteristics: Ambient sound rejection - Since unidirectional microphones are less sensitive to off-axis sound than omnidirectional types they pick up less overall ambient or stage sound. Unidirectional mics should be used to control ambient noise pickup to get a cleaner mix. Distance factor - Because directional microphones pick up less ambient sound than omnidirectional types they may be used at somewhat greater distances from a sound source and still achieve the same balance between the direct sound and background or ambient sound. An omni should be placed closer to the sound source than a uni—about half the distance—to pick up the same balance between direct sound and ambient sound. Off-axis coloration - Change in a microphone’s frequency response that usually gets progressively more noticeable as the arrival angle of sound increases. High frequencies tend to be lost first, often resulting in “muddy” off-axis sound. Proximity effect - With unidirectional microphones, bass response increases as the mic is moved closer (within 2 feet) to the sound source. With close-up unidirectional microphones (less than 1 foot), be aware of proximity effect and roll off the bass until you obtain a more natural sound. You can (1) roll off low frequencies on the mixer, or (2) use a microphone designed to minimize proximity effect, or (3) use a microphone with a bass rolloff switch, or (4) use an omnidirectional microphone (which does not exhibit proximity effect). Proximity effect graph Monitor speaker placement for maximum rejection: cardioid and supercardioid 10 Microphone Techniques for LIVE SOUND Unidirectional microphones can not only help to isolate one voice or instrument from other singers or instruments, but can also minimize feedback, allowing higher gain. For these reasons, unidirectional microphones are preferred over omnidirectional microphones in almost all sound reinforcement applications. The electrical output of a microphone is usually specified by level, impedance and wiring configuration. Output level or sensitivity is the level of the electrical signal from the microphone for a given input sound level. In general, condenser microphones have higher sensitivity than dynamic types. For weak or distant sounds a high sensitivity microphone is desirable while loud or close-up sounds can be picked up well by lower-sensitivity models. The output impedance of a microphone is roughly equal to the electrical resistance of its output: 150-600 ohms for low impedance (low-Z) and 10,000 ohms or more for high impedance.(high-Z). The practical concern is that low impedance microphones can be used with cable lengths of 1000 feet or more with no loss of quality while high impedance types exhibit noticeable high frequency loss with cable lengths greater than about 20 feet. Finally, the wiring configuration of a microphone may be balanced or unbalanced. A balanced output carries the signal on two conductors (plus shield). The signals on each conductor are the same level but opposite polarity (one signal is positive when the other is negative). A balanced microphone input amplifies only the difference between the two signals and rejects any part of the signal which is the same in each conductor. Any electrical noise or hum picked up by a balanced (two-conductor) cable tends to be identical in the two conductors and is therefore rejected by the balanced input while the equal but opposite polarity original signals are amplified. On the other hand, an unbalanced microphone output carries its signal on a single conductor (plus shield) and an unbalanced microphone input amplifies any signal on that conductor. Such a combination will be unable to reject any electrical noise which has been picked up by the cable. Balanced, low-impedance microphones are therefore recommended for nearly all sound reinforcement applications. The physical design of a microphone is its mechanical and operational design. Types used in sound reinforcement include: handheld, headworn, lavaliere, overhead, stand-mounted, instrument-mounted and surface-mounted designs. Most of these are available in a choice of operating principle, frequency response, directional pattern and electrical output. Often the physical design is the first choice made for an application. Understanding and choosing the other characteristics can assist in producing the maximum quality microphone signal and delivering it to the sound system with the highest fidelity. Musical Instrument Characteristics Some background information on characteristics of musical instruments may be helpful. Instruments and other sound sources are characterized by their frequency output, by their directional output and by their dynamic range. Frequency output - the span of fundamental and harmonic frequencies produced by an instrument, and the balance or relative level of those frequencies. Musical instruments have overall frequency ranges as found in the chart below. The dark section of each line indicates the range of fundamental frequencies and the shaded section represents the range of the highest harmonics or overtones of the instrument. The fundamental frequency establishes the basic pitch of a note played by an instrument while the harmonics produce the timbre or characteristic tone. 11 Microphone Techniques for LIVE SOUND A microphone which responds evenly to the full range of an instrument will reproduce the most natural sound from an instrument. A microphone which responds unevenly or to less than the full range will alter the sound of the instrument, though this effect may be desirable in some cases. Directional output - the three-dimensional pattern of sound waves radiated by an instrument. Instrument frequency ranges It is this timbre that distinguishes the sound of one instrument from another. In this manner, we can tell whether a piano or a trumpet just played that C note. The following graphs show the levels of the fundamental and harmonics associated with a trumpet and an oboe each playing the same note. A musical instrument radiates a different tone quality (timbre) in every direction, and each part of the instrument produces a different timbre. Most musical instruments are designed to sound best at a distance, typically two or more feet away. At this distance, the sounds of the various parts of the instrument combine into a pleasing composite. In addition, many instruments produce this balanced sound only in a particular direction. A microphone placed at such distance and direction tends to pick up a natural or well-balanced tone quality. On the other hand, a microphone placed close to the instrument tends to emphasize the part of the instrument that the microphone is near. The resulting sound may not be representative of the instrument as a whole. Thus, the reinforced tonal balance of an instrument is strongly affected by the microphone position relative to the instrument. oboe trumpet in Bb 200 500 1000 2000 3000 4000 5000 frequency Instrument spectra comparison The number of harmonics along with the relative level of the harmonics is noticeably different between these two instruments and provides each instrument with its own unique sound. 12 Unfortunately, it is difficult, if not impossible, to place a microphone at the “natural sounding” distance from an instrument in a sound reinforcement situation without picking up other (undesired) sounds and/or acoustic feedback. Close microphone placement is usually the only practical way to achieve sufficient isolation and gain-before-feedback. But since the sound picked up close to a source can vary significantly with small changes in microphone position, it is very useful to experiment with microphone location and orientation. In some cases more than one microphone may be required to get a good sound from a large instrument such as a piano. Microphone Techniques for LIVE SOUND Instrument Loudspeakers Another instrument with a wide range of characteristics is the loudspeaker. Anytime you are placing microphones to pick up the sound of a guitar or bass cabinet you are confronted with the acoustic nature of loudspeakers. Each individual loudspeaker type is directional and displays different frequency characteristics at different angles and distances. The sound from a loudspeaker tends to be almost omnidirectional at low frequencies but becomes very directional at high frequencies. Thus, the sound on-axis at the center of a speaker usually has the most “bite” or high-end, while the sound produced off-axis or at the edge of the speaker is more “mellow” or bassy. A cabinet with multiple loudspeakers has an even more complex output, especially if it has different speakers for bass and treble. As with most acoustic instruments the desired sound only develops at some distance from the speaker. Sound reinforcement situations typically require a close-mic approach. A unidirectional dynamic microphone is a good first choice here: it can handle the high level and provide good sound and isolation. Keep in mind the proximity effect when using a uni close to the speaker: some bass boost will be likely. If the cabinet has only one speaker a single microphone should pick up a suitable sound with a little experimentation. If the cabinet has multiple speakers of the same type it is typically easiest to place the microphone to pick up just one speaker. Placing the microphone between speakers can result in strong phase effects though this may be desirable to achieve a particular tone. However, if the cabinet is stereo or has separate bass and treble speakers multiple microphones may be required. Placement of loudspeaker cabinets can also have a significant effect on their sound. Putting cabinets on carpets can reduce brightness, while raising them off the floor can reduce low end. Open-back cabinets can be miked from behind as well as from the front. The distance from the cabinet to walls or other objects can also vary the sound. Again, experiment with the microphone(s) and placement until you have the sound that you like! Dynamic range - the range of volume of an instrument from its softest to its loudest level. The dynamic range of an instrument determines the specifications for sensitivity and maximum input capability of the intended microphone. Loud instruments such as drums, brass and amplified guitars are handled well by dynamic microphones which can withstand high sound levels and have moderate sensitivity. Softer instruments such as flutes and harpsichords can benefit from the higher sensitivity of condensers. Of course, the farther the microphone is placed from the instrument the lower the level of sound reaching the microphone. In the context of a live performance, the relative dynamic range of each instrument determines how much sound reinforcement may be required. If all of the instruments are fairly loud, and the venue is of moderate size with good acoustics, no reinforcement may be necessary. On the other hand, if the performance is in a very large hall or outdoors, even amplified instruments may need to be further reinforced. Finally, if there is a substantial difference in dynamic range among the instruments, such as an acoustic guitar in a loud rock band, the microphone techniques (and the sound system) must accommodate those differences. Often, the maximum volume of the overall sound system is limited by the maximum gain-before-feedback of the softest instrument. An understanding of the frequency output, directional output, and dynamic range characteristics of musical instruments can help significantly in choosing suitable microphones, placing them for best pickup of the desired sound and minimizing feedback or other undesired sounds. VIOLIN PIA NO GUI TAR SAX OPHONE HAR MONICA TRUMPET MALE VOICE FEMALE VOICE BASS DRUM SNARE DRUM CYM BAL 0 20 40 60 80 100 120 Intensity Level in Decibels (at distance of 10 feet) 13 Microphone Techniques for LIVE SOUND Acoustic Characteristics Approximate wavelengths of common frequencies: 100 Hz: about 10 feet 1000 Hz: about 1 foot 10,000 Hz: about 1 inch Sound Waves Sound moves through the air like waves in water. Sound waves consist of pressure variations traveling through the air. When the sound wave travels, it compresses air molecules together at one point. This is called the high pressure zone or positive component(+). After the compression, an expansion of molecules occurs. This is the low pressure zone or negative component(-). This process continues along the path of the sound wave until its energy becomes too weak to hear. The sound wave of a pure tone traveling through air would appear as a smooth, regular variation of pressure that could be drawn as a sine wave. 140 130 120 110 100 90 80 70 60 50 40 30 20 10 0 1 CYCLE ▲ ▲ Frequency, wavelength and the speed of sound ▲ PRESSURE ▲ The frequency 1 /2 CYCLE of a sound wave indicates the rate of + ▲ pressure 0 AMPLITUDE _ variations WAVELENGTH DISTANCE or cycles. One cycle is Schematic of sound wave a change from high pressure to low pressure and back to high pressure. The number of cycles per second is called Hertz, abbreviated “Hz.” So, a 1,000 Hz tone has 1,000 cycles per second. ▲ ▲ ▲ The wavelength of a sound is the physical distance from the start of one cycle to the start of the next cycle. Wavelength is related to frequency by the speed of sound. The speed of sound in air is about 1130 feet per second or 344 meters/second. The speed of sound is constant no matter what the frequency. The wavelength of a sound wave of any frequency can be determined by these relationships: The Wave Equation: c = f • l speed of sound = frequency • wavelength or wavelength = Ambient sounds speed of sound frequency Loudness The fluctuation of air pressure created by sound is a change above and below normal atmospheric pressure. This is what the human ear responds to. The varying amount of pressure of the air molecules compressing and expanding is related to the apparent loudness at thehuman ear. The greater the pressure change, the louder the sound. Under ideal conditions the human ear can sense a pressure change as small as 0.0002 microbars (1 microbar = 1/1,000,000 atmospheric pressure). The threshold of pain is about 200 microbars, one million times greater! Obviously the human ear responds to a wide range of amplitude of sound. This amplitude range is more commonly measured in decibels Sound Pressure Level (dB SPL), relative to 0.0002 microbars (0 dB SPL). 0 dB SPL is the threshold of hearing Lp and 120 dB SPL is the threshold of pain. 1dB is about the smallest change in SPL that can be heard. A 3dB change is generally noticeable while a 6dB change is very noticeable. A 10dB SPL increase is perceived to be twice as loud! for a 500Hz sound wave: 1,130 feet per second wavelength = 500Hz wavelength = 2.26 feet 14 Sound Propagation There are four basic ways in which sound can be altered by its environment as it travels or propagates: reflection, absorption, diffraction and refraction. Microphone Techniques for LIVE SOUND 1. Reflection - A sound wave can be reflected by a surface or other object if the object is physically as large or larger than the wavelength of the sound. Because low frequency sounds have long wavelengths they can only be reflected by large objects. Higher frequencies can be reflected by smaller objects and surfaces as well as large. The reflected sound will have a different frequency characteristic than the direct sound if all frequencies are not reflected equally. Reflection is also the source of echo, reverb, and standing waves: Echo occurs when a reflected sound is delayed long enough (by a distant reflective surface) to be heard by the listener as a distinct repetition of the direct sound. Reverberation consists of many reflections of a sound, maintaining the sound in a reflective space for a time even after the direct sound has stopped. Standing waves in a room occur for certain frequencies related to the distance between parallel walls. The original sound and the reflected sound will begin to reinforce each other when the distance between two opposite walls is equal to a multiple of half the wavelength of the sound. This happens primarily at low frequencies due to their longer wavelengths and relatively high energy. 2. Absorption - Some materials absorb sound rather than reflect it. Again, the efficiency of absorption is dependent on the wavelength. Thin absorbers like carpet and acoustic ceiling tiles can affect high frequencies only, while thick absorbers such as drapes, padded furniture and specially designed bass traps are required to attenuate low frequencies. Reverberation in a room can be controlled by adding absorption: the more absorption the less reverberation. Clothed humans absorb mid and high frequencies well, so the presence or absence of an audience has a significant effect on the sound in an otherwise reverberant venue. 3. Diffraction - A sound wave will typically bend around obstacles in its path which are smaller than its wavelength. Because a low frequency sound wave is much longer than a high frequency wave, low frequencies will bend around objects that high frequencies cannot. The effect is that high frequencies tend to have a higher directivity and are more easily blocked while low frequencies are essentially omnidirectional. In sound reinforcement, it is difficult to get good directional control at low frequencies for both microphones and loudspeakers. 4. Refraction - The bending of a sound wave as it passes through some change in the density of the environment. This effect is primarily noticeable outdoors at large distances from loudspeakers due to atmospheric effects such as wind or temperature gradients. The sound will appear to bend in a certain direction due to these effects. Direct vs. Ambient Sound A very important property of direct sound is that it becomes weaker as it travels away from the sound source. The amount of change is controlled by the inverse-square law which states that the level change is inversely proportional to the square of the distance change. When the distance from a sound source doubles, the sound level decreases by 6dB. This is a noticeable decrease. For example, if the sound from a guitar amplifier is 100 dB SPL at 1 ft. from the cabinet it will be 94 dB at 2 ft., 88 dB at 4 ft., 82 dB at 8 ft., etc. Conversely, when the distance is cut in half the sound level increases by 6dB: It will be 106 dB at 6 inches and 112 dB at 3 inches! On the other hand, the ambient sound in a room is at nearly the same level throughout the room. This is because the ambient sound has been reflected many times within the room until it is essentially nondirectional. Reverberation is an example of nondirectional sound. For this reason the ambient sound of the room will become increasingly apparent as a microphone is placed further away from the direct sound source. In every room, there is a distance (measured from the sound source) where the direct sound and the reflected (or reverberant) sound become equal in intensity. In acoustics, this is known as the Critical Distance. If a microphone is placed at the Critical Distance or farther, the sound quality picked up may be very poor. This sound is often described as “echoey”, reverberant, or “bottom of the barrel”. The reflected sound overlaps and blurs the direct sound. Critical distance may be estimated by listening to a sound source at a very short distance, then moving away until the sound level no longer decreases but seems to be constant. That distance is critical distance. 15 Microphone Techniques for LIVE SOUND A unidirectional microphone should be positioned no farther than 50% of the Critical Distance, e.g. if the Critical Distance is 10 feet, a unidirectional mic may be placed up to 5 feet from the sound source. Highly reverberant rooms may require very close microphone placement. The amount of direct sound relative to ambient sound is controlled primarily by the distance of the microphone to the sound source and to a lesser degree by the directional pattern of the mic. +1 -1 “in-phase” a ▲ -2 -1 +1 0 ”1800 out of phase” -1 + = +1 0 0 ▲ 16 0 0 b The phase of a single frequency sound wave is always described relative to the starting point of the wave or 0 degrees. The pressure change is also zero at this point. The 00 900 1800 2700 3600 peak of the high pressure zone is at 90 degrees, the Sound pressure wave pressure change falls to zero again at 180 degrees, the peak of the low pressure zone is at 270 degrees, and the pressure change rises to zero at 360 degrees for the start of the next cycle. + = +1 Phase relationships and interference effects one cycle or one period +2 0 -1 +1 +2 0 -1 “phase shifts” +1 0 c -1 +1 + = 0 -1 -2 Phase relationships Two identical sound waves starting at the same point in time are called “in-phase” and will sum together creating a single wave with double the amplitude but otherwise identical to the original waves. Two identical sound waves with one wave’s starting point occurring at the 180 degree point of the other wave are said to be “out of phase” and the two waves will cancel each other completely. When two sound waves of the same single frequency but different starting points are combined the resulting wave is said to have “phase shift” or an apparent starting point somewhere between the original starting points. This new wave will have the same frequency as the original waves but will have increased or decreased amplitude depending on the degree of phase difference. Phase shift, in this case, indicates that the 0 degree points of two identical waves are not the same. The first case is the basis for the increased sensitivity of boundary or surface-mount microphones. When a microphone element is placed very close to an acoustically reflective surface both the incident and reflected sound waves are in phase at the microphone. This results in a 6dB increase (doubling) in sensitivity, compared to the same microphone in free space. This occurs for reflected frequencies whose wavelength is greater than the distance from the microphone to the surface: if the distance is less than one-quarter inch this will be the case for frequencies up to at least 18 kHz. However, this 6dB increase will not occur for frequencies that are not reflected, that is, frequencies that are either absorbed by the surface or that diffract around the surface. High frequencies may be absorbed by surface materials such as carpeting or other acoustic treatments. Low frequencies will diffract around the surface if their wavelength is much greater than the dimensions of the surface: the boundary must be at least 5 ft. square to reflect frequencies down to 100 Hz. Most soundwaves are not a single frequency but are made up of many frequencies. When identical multiplefrequency soundwaves combine there are three possibilities for the resulting wave: a doubling of amplitude at all frequencies if the waves are in phase, a complete cancellation at all frequencies if the waves are 180 degrees out of phase, or partial cancellation and partial reinforcement at various frequencies if the waves have intermediate phase relationship. The results may be heard as interference effects. The second case occurs when two closely spaced microphones are wired out of phase, that is, with reverse polarity. This usually only happens by accident, due to miswired microphones or cables but the effect is also used as the basis for certain noise-canceling microphones. In this technique, two identical microphones are placed very close to each other (sometimes within the same housing) and wired with opposite polarity. Sound waves from distant sources which arrive equally at the two microphones are effectively canceled when the outputs are mixed. Microphone Techniques for LIVE SOUND Polarity reversal Multi-mic comb filtering However, sound from a source which is much closer to one element than to other will be heard. Such close-talk microphones, which must literally have the lips of the talker touching the grille, are used in high-noise environments such as aircraft and industrial paging but rarely with musical instruments due to their limited frequency response. than one speaker or multiple loudspeaker cabinets for a single instrument would be examples. The delayed sound travels a longer distance (longer time) to the mic and thus has a phase difference relative to the direct sound. When these sounds combine (acoustically) at the microphone, comb filtering results. This time the effect of the comb filtering depends on the distance between the microphone and the source of the reflection or the distance between the multiple sources. It is the last case which is most likely in musical sound reinforcement, and the audible result is a degraded frequency response called “comb filtering.” The pattern of peaks and dips resembles the teeth of a comb and the depth and location of these notches depend on the degree of phase shift. With microphones this effect can occur in two ways. The first is when two (or more) mics pick up the same sound source at different distances. Because it takes longer for the sound to arrive at the more distant microphone there is effectively a phase difference between the signals from the mics when they are combined (electrically) in the mixer. The resulting comb filtering depends on the sound arrival time difference between the microphones: a large time difference (long distance) causes comb filtering to begin at low frequencies, while a small time difference (short distance) moves the comb filtering to higher frequencies. The second way for this effect to occur is when a single microphone picks up a direct sound and also a delayed version of the same sound. The delay may be due to an acoustic reflection of the original sound or to multiple sources of the original sound. A guitar cabinet with more Reflection comb filtering 17 Microphone Techniques for LIVE SOUND The 3-to-1 Rule Microphone Phase Effects When it is necessary to use multiple microphones or to use microphones near reflective surfaces the resulting interference effects may be minimized by using the 3-to-1 rule. For multiple microphones the rule states that the distance between microphones should be at least three times the distance from each microphone to its intended sound source. The sound picked up by the more distant microphone is then at least 12dB less than the sound picked up by the closer one. This insures that the audible effects of comb filtering are reduced by at least that much. For reflective surfaces, the microphone should be at least 11/2 times as far from that surface as it is from its intended sound source. Again, this insures minimum audibility of interference effects. One effect often heard in sound reinforcement occurs when two microphones are placed in close proximity to the same sound source, such as a drum kit or instrument amplifier. Many times this is due to the phase relationship of the sounds arriving at the microphones. If two microphones are picking up the same sound source from different locations, some phase cancellation or summing may be occurring. Phase cancellation happens when two microphones are receiving the same soundwave but with opposite pressure zones (that is,180 degrees out of phase). This is usually not desired. A mic with a different polar pattern may reduce the pickup of unwanted sound and reduce the effect or physical isolation can be used. With a drum kit, physical isolation of the individual drums is not possible. In this situation the choice of microphones may be more dependent on the off-axis rejection characteristic of the mic. 3-to-1 rule Strictly speaking, the 3-to-1 rule is based on the behavior of omnidirectional microphones. It can be relaxed slightly if unidirectional microphones are used and they are aimed appropriately, but should still be regarded as a basic rule of thumb for worst case situations. Another possibility is phase reversal. If there is cancellation occurring, a 180 degree phase flip will create phase summing of the same frequencies. A common approach to the snare drum is to place one mic on the top head and one on the bottom head. Because the mics are picking up relatively similar sound sources at different points in the sound wave, you may experience some phase cancellations. Inverting the phase of one mic will sum any frequencies being canceled. This may sometimes achieve a “fatter“ snare drum sound. This effect will change dependent on mic locations. The phase inversion can be done with an in-line phase reverse adapter or by a phase invert switch found on many mixers inputs. Potential Acoustic Gain vs. Needed Acoustic Gain The basic purpose of a sound reinforcement system is to deliver sufficient sound level to the audience so that they can hear and enjoy the performance throughout the listening area. As mentioned earlier, the amount of reinforcement needed depends on the loudness of the instruments or performers themselves and the size and acoustic nature of the venue. This Needed Acoustic Gain (NAG) is the amplification factor necessary so that the furthest listeners can hear as if they were close enough to hear the performers directly. 18 Microphone Techniques for LIVE SOUND To calculate NAG: NAG = 20 x log (Df/Dn) The simplified PAG equation is: Where: Df = distance from sound source to furthest listener PAG = 20 (log D1 - log D2 + log D0 - log Ds) -10 log NOM -6 Where: PAG = Potential Acoustic Gain (in dB) Dn = distance from sound source to nearest listener Ds = distance from sound source to microphone log = logarithm to base 10 D0 = distance from sound source to listener Note: the sound source may be a musical instrument, a vocalist or perhaps a loudspeaker. The equation for NAG is based on the inverse-square law, which says that the sound level decreases by 6dB each time the distance to the source doubles. For example, the sound level (without a sound system) at the first row of the audience (10 feet from the stage) might be a comfortable 85dB. At the last row of the audience (80 feet from the stage) the level will be 18dB less or 67dB. In this case the sound system needs to provide 18dB of gain so that the last row can hear at the same level as the first row. The limitation in real-world sound systems is not how loud the system can get with a recorded sound source but rather how loud it can get with a microphone as its input. The maximum loudness is ultimately limited by acoustic feedback. The amount of gain-before-feedback that a sound reinforcement system can provide may be estimated mathematically. This Potential Acoustic Gain involves the distances between sound system components, the number of open mics, and other variables. The system will be sufficient if the calculated Potential Acoustic Gain (PAG) is equal to or greater than the Needed Acoustic Gain (NAG). Below is an illustration showing the key distances. D1 D2 Ds D0 D1 = distance from microphone to loudspeaker D2 = distance from loudspeaker to listener NOM = the number of open microphones -6 = a 6 dB feedback stability margin log = logarithm to base 10 In order to make PAG as large as possible, that is, to provide the maximum gain-before-feedback, the following rules should be observed: 1) Place the microphone as close to the sound source as practical. 2) Keep the microphone as far away from the loudspeaker as practical. 3) Place the loudspeaker as close to the audience as practical. 4) Keep the number of microphones to a minimum. In particular, the logarithmic relationship means that to make a 6dB change in the value of PAG the corresponding distance must be doubled or halved. For example, if a microphone is 1 ft. from an instrument, moving it to 2 ft. away will decrease the gain-before-feedback by 6dB while moving it to 4 ft. away will decrease it by 12dB. On the other hand, moving it to 6 in. away increases gain-beforefeedback by 6dB while moving it to only 3 in. away will increase it by 12dB. This is why the single most significant factor in maximizing gain-before-feedback is to place the microphone as close as practical to the sound source. PAG 19 Microphone Techniques for LIVE SOUND The NOM term in the PAG equation reflects the fact that gain-before-feedback decreases by 3dB every time the number of open (active) microphones doubles. For example, if a system has a PAG of 20dB with a single microphone, adding a second microphone will decrease PAG to 17dB and adding a third and fourth mic will decrease PAG to 14dB. This is why the number of microphones should be kept to a minimum and why unused microphones should be turned off or attenuated. Essentially, the gain-before-feedback of a sound system can be evaluated strictly on the relative location of sources, microphones, loudspeakers, and audience, as well as the number of microphones, but without regard to the actual type of component. Though quite simple, the results are very useful as a best case estimate. Understanding principles of basic acoustics can help to create an awareness of potential influences on reinforced sound and to provide some insight into controlling them. When effects of this sort are encountered and are undesirable, it may be possible to adjust the sound source, use a microphone with a different directional characteristic, reposition the microphone or use fewer microphones, or possibly use acoustic treatment to improve the situation. Keep in mind that in most cases, acoustic problems can best be solved acoustically, not strictly by electronic devices. General Rules Microphone technique is largely a matter of personal taste—whatever method sounds right for the particular instrument, musician, and song is right. There is no one ideal microphone to use on any particular instrument. There is also no one ideal way to place a microphone. Choose and place the microphone to get the sound you want. We recommend experimenting with a variety of microphones and positions until you create your desired sound. However, the desired sound can often be achieved more quickly and consistently by understanding basic microphone characteristics, sound-radiation properties of musical instruments, and acoustic fundamentals as presented above. 20 Here are some suggestions to follow when miking musical instruments for sound reinforcement. • Try to get the sound source (instrument, voice, or amplifier) to sound good acoustically (“live”) before miking it. • Use a microphone with a frequency response that is limited to the frequency range of the instrument, if possible, or filter out frequencies below the lowest fundamental frequency of the instrument. • To determine a good starting microphone position, try closing one ear with your finger. Listen to the sound source with the other ear and move around until you find a spot that sounds good. Put the microphone there. However, this may not be practical (or healthy) for extremely close placement near loud sources. • The closer a microphone is to a sound source, the louder the sound source is compared to reverberation and ambient noise. Also, the Potential Acoustic Gain is increased—that is, the system can produce more level before feedback occurs. Each time the distance between the microphone and sound source is halved, the sound pressure level at the microphone (and hence the system) will increase by 6 dB. (Inverse Square Law) • Place the microphone only as close as necessary. Too close a placement can color the sound source’s tone quality (timbre), by picking up only one part of the instrument. Be aware of Proximity Effect with unidirectional microphones and use bass rolloff if necessary. • Use as few microphones as are necessary to get a good sound. To do that, you can often pick up two or more sound sources with one microphone. Remember: every time the number of microphones doubles, the Potential Acoustic Gain of the sound system decreases by 3 dB. This means that the volume level of the system must be turned down for every extra mic added in order to prevent feedback. In addition, the amount of noise picked up increases as does the likelihood of interference effects such as comb-filtering. Microphone Techniques for LIVE SOUND • When multiple microphones are used, the distance between microphones should be at least three times the distance from each microphone to its intended sound source. This will help eliminate phase cancellation. For example, if two microphones are each placed one foot from their sound sources, the distance between the microphones should be at least three feet. (3 to 1 Rule) • To reduce feedback and pickup of unwanted sounds: • To reduce “pop” (explosive breath sounds occurring with the letters “p,” “b,” and “t”): 1) mic either closer or farther than 3 inches from the mouth (because the 3-inch distance is worst) 2) place the microphone out of the path of pop travel (to the side, above, or below the mouth) 1) place microphone as close as practical to desired sound source 3) use an omnidirectional microphone 2) place microphone as far as practical from unwanted sound sources such as loudspeakers and other instruments 4) use a microphone with a pop filter. This pop filter can be a ball-type grille or an external foam windscreen 3) aim unidirectional microphone toward desired sound source (on-axis) • If the sound from your loudspeakers is distorted the microphone signal may be overloading your mixer’s input. To correct this situation, use an in-line attenuator (such as the Shure A15AS), or use the input attenuator on your mixer to reduce the signal level from the microphone. 4) aim unidirectional microphone away from undesired sound source (180 degrees off-axis for cardioid, 126 degrees off-axis for supercardioid) 5) use minimum number of microphones • To reduce handling noise and stand thumps: 1) use an accessory shock mount (such as the Shure A55M) Seasoned sound engineers have developed favorite microphone techniques through years of experience. If you lack this experience, the suggestions listed on the following pages should help you find a good starting point. These suggestions are not the only possibilities; other microphones and positions may work as well or better for your intended application. Remember—Experiment and Listen! 2) use an omnidirectional microphone 3) use a unidirectional microphone with a specially designed internal shock mount 21 Microphone Techniques for LIVE SOUND Microphone Placement Tonal Balance Comments Bassy, robust (unless an omni is used) Minimizes feedback and leakage. Roll off bass if desired for more natural sound. Bassy, robust (unless an omni is used) Minimizes feedback and leakage. Allows engineer control of voice balances. Roll off bass if necessary for more natural sound when using cardioids. 1 to 3 feet above and 2 to 4 feet in front of the first row of the choir, aimed toward the middle row(s) of the choir, approximately 1 microphone per 15-20 people Full range, good blend, semi-distant Use flat-response unidirectional microphones, Use minimum number of microphones needed to avoid overlapping pickup areas. Miniature microphone clipped outside of sound hole Natural, well-balanced Good isolation. Allows freedom of movement. Miniature microphone clipped inside sound hole Bassy, less string noise Reduces feedback. 8 inches from sound hole Bassy Good starting placement when leakage or feedback is a problem. Roll off bass for a more natural sound (more for a uni than an omni). 3 inches from sound hole Very bassy, boomy, muddy, full Very good isolation. Bass rolloff needed for a natural sound. 4 to 8 inches from bridge Woody, warm, mellow. Midbasy, lacks detail Reduces pick and string noise. 6 inches above the side, over the bridge, and even with the front soundboard Natural, well-balanced, slightly bright Less pickup of ambience and leakage than 3 feet from sound hole. miniature microphone clipped outside of sound hole Natural, well-balanced Good isolation. Allows freedom of movement. miniature microphone clipped inside sound hole Bassy, less string noise Reduces feedback. Lead vocal: Handheld or on stand, microphone windscreen touching lips or just a few inches away Backup vocals: One microphone per singer Handheld near chin or stand-mounted Touching lips or a few inches away Choral groups: Acoustic guitar: 22 Microphone Techniques for LIVE SOUND Microphone Placement Tonal Balance Comments 3 inches from center of head Bassy, thumpy Rejects feedback and leakage. Roll off bass for natural sound. 3 inches from edge of head Bright Rejects feedback and leakage. Miniature microphone clipped to tailpiece aiming at bridge Natural Rejects feedback and leakage. Allows freedom of movement. A few inches from side Natural Well-balanced sound. Miniature lavalier microphone mounted on strings between bridge and tailpiece Full, bright Use string mount. Listen for vibrations, adjust mount position. Well-defined Well-balanced sound, but little isolation. Banjo: Violin (fiddle): Cello: 1 foot from bridge General string instruments (mandolin, dobro and dulcimer): Miniature microphone attached to strings between bridge and tailpiece Bright Minimizes feedback and leakage. Allows freedom of movement. Acoustic bass (upright bass, string bass, bass violin): 6 inches to 1 foot out front, just above bridge Well-defined Natural sound. A few inches from f-hole Full Roll off bass if sound is too boomy. Wrap microphone in foam padding (except for grille) and put behind bridge or between tailpiece and body Full, “tight” Minimizes feedback and leakage. Miniature lavalier microphone mounted on strings between bridge and tailpiece Full, bright Use string mount. Listen for vibrations, adjust mount position. Aiming toward player at part of soundboard, about 2 feet away Natural See “Stereo Microphone Techniques” section for other possibilities. Tape miniature microphone to soundboard Somewhat constricted Minimizes feedback and leakage. Harp: 23 Microphone Techniques for LIVE SOUND Microphone Placement Tonal Balance Comments 12 inches above middle strings, 8 inches horizontally from hammers with lid off or at full stick Natural, well-balanced Less pickup of ambience and leakage. Move microphone(s) farther from hammers to reduce attack and mechanical noises. Good coincident-stereo placement. See “Stereo Microphone Techniques” section. 8 inches above treble strings, as above Natural, well-balanced, slightly bright Place one microphone over bass strings and one over treble strings for stereo. Phase cancellations may occur if the recording is heard in mono. Aiming into sound holes Thin, dull, hard, constricted Very good isolation. Sometimes sounds good for rock music. Boost mid-bass and treble for more natural sound. 6 inches over middle strings, 8 inches from hammers, with lid on short stick Muddy, boomy, dull, lacks attack Improves isolation. Bass rolloff and some treble boost required for more natural sound. Next to the underside of raised lid, centered on lid Bassy, full Unobtrusive placement. Underneath the piano, aiming up at the soundboard Bassy, dull, full Unobtrusive placement. Surface-mount microphone mounted on underside of lid over lower treble strings, horizontally close to hammers for brighter sound, further from hammers for more mellow sound Bright, wellbalanced Excellent isolation. Experiment with lid height and microphone placement on piano lid for desired sounds. Two surface-mount microphones positioned on the closed lid, under the edge at its keyboard edge, approximately 2/3 of the distance from middle A to each end of the keyboard Bright, well-balanced, strong attack Excellent isolation. Moving “low” mic away from keyboard six inches provides truer reproduction of the bass strings while reducing damper noise. By splaying these two mics outward slightly, the overlap in the middle registers can be minimized. Surface-mount microphone placed vertically on the inside of the frame, or rim, of the piano, at or near the apex of the piano’s curved wall Full, natural Excellent isolation. Minimizes hammer and damper noise. Best if used in conjunction with two surface-mount microphones mounted to closed lid, as above. Grand piano: 24 Microphone Techniques for LIVE SOUND Microphone Placement Tonal Balance Comments Just over open top, above treble strings Natural (but lacks deep bass), picks up hammer attack Good placement when only one microphone is used. Just over open top, above bass strings Slightly full or tubby, picks up hammer attack Mike bass and treble strings for stereo. Inside top near the bass and treble stings Natural, picks up hammer attack Minimizes feedback and leakage. Use two microphones for stereo. 8 inches from bass side of soundboard Full, slightly tubby, no hammer attack Use this placement with the following placement for stereo. 8 inches from treble side of soundboard Thin, constricted, no hammer attack Use this placement with the preceding placement for stereo. 1 foot from center of soundboard on hard floor or one-foot-square plate on carpeted floor, aiming at piano. Soundboard should face into room Natural, good presence Minimize pickup of floor vibrations by mounting microphone in low-profile shock-mounted microphone stand. Aiming at hammers from front, several inches away (remove front panel) Bright, picks up hammer attack Mike bass and treble strings for stereo. Upright piano: Brass (trumpet, cornet, trombone, tuba): The sound from these instruments is very directional. Placing the mic off axis with the bell of the instrument will result in less pickup of high frequencies. 1 to 2 feet from bell. A couple of instruments can play into one microphone On-axis to bell sounds bright; to one side sounds natural or mellow Close miking sounds “tight” and minimizes feedback and leakage. More distant placement gives fuller, more dramatic sound. Miniature microphone mounted on bell Bright Maximum isolation. 25 Microphone Techniques for LIVE SOUND Microphone Placement Tonal Balance Comments Natural Watch out for extreme fluctuations on VU meter. French horn: Microphone aiming toward bell Saxophone: With the saxophone, the sound is fairly well distributed between the finger holes and the bell. Miking close to the finger holes will result in key noise. The soprano sax must be considered separately because its bell does not curve upward. This means that, unlike all other saxophones, placing a microphone toward the middle of the instrument will not pick-up the sound from the key holes and the bell simultaneously. The saxophone has sound characteristics similar to the human voice. Thus, a shaped response microphone designed for voice works well. A few inches from and aiming into bell Bright Minimizes feedback and leakage. A few inches from sound holes Warm, full Picks up fingering noise. A few inches above bell and aiming at sound holes Natural Good recording technique. Miniature microphone mounted on bell Bright, punchy Maximum isolation, up-front sound. Flute: The sound energy from a flute is projected both by the embouchure and by the first open fingerhole. For good pickup, place the mic as close as possible to the instrument. However, if the mic is too close to the mouth, breath noise will be apparent. Use a windscreen on the mic to overcome this difficulty. A few inches from area between mouthpiece and first set of finger holes Natural, breathy Pop filter or windscreen may be required on microphone. A few inches behind player’s head, aiming at finger holes Natural Reduces breath noise. About 1 foot from sound holes Natural Provides well-balanced sound. A few inches from bell Bright Minimizes feedback and leakage. Woodwinds (Oboe, bassoon, etc): 26 Microphone Techniques for LIVE SOUND Microphone Placement Tonal Balance Comments Full, bright Minimizes feedback and leakage. Microphone may be cupped in hands. Emphasized midrange Minimizes feedback and leakage. Allows freedom of movement. Harmonica: Very close to instrument Accordion: Miniature microphone mounted internally Electric guitar amplifier/speaker: The electric guitar has sound characteristics similar to the human voice. Thus, a shaped response microphone designed for voice works well. 4 inches from grille cloth at center of speaker cone Natural, well-balanced Small microphone desk stand may be used if loudspeaker is close to floor. 1 inch from grille cloth at center of speaker cone Bassy Minimizes feedback and leakage. Off-center with respect to speaker cone Dull or mellow Microphone closer to edge of speaker cone results in duller sound. Reduces amplifier hiss noise. 3 feet from center of speaker cone Thin, reduced bass Picks up more room ambience and leakage. Miniature microphone draped over amp in front of speaker Emphasized midrange Easy setup, minimizes leakage. Microphone placed behind open back cabinet Depends on position Can be combined with mic in front of cabinet, but be careful of phase cancellation. Depends on placement Improve clarity by cutting frequencies around 250 Hz and boosting around 1,500 Hz. Bass guitar amplifier/speaker: Mike speaker as described in Electric Guitar Amplifier section Electric keyboard amplifier/speakers: Mike speaker as described in Electric Guitar Amplifier section Depends on brand of piano Roll off bass for clarity, roll off highs to reduce hiss. 27 Microphone Techniques for LIVE SOUND Microphone Placement Tonal Balance Comments Aim one microphone into top louvers 3 inches to 1 foot away Natural, lacks deep bass Good one-mike pickup. Mike top louvers and bottom bass speaker 3 inches to 1 foot away Natural, well-balanced Excellent overall sound. Mike top louvers with two microphones, one close to each side. Pan to left and right. Mike bottom bass speaker 3 inches to 1 foot away and pan its signal to center Natural, well-balanced Stereo effect. Leslie organ speaker: Front View Top View Drum kit: In most sound reinforcement systems, the drum set is miked with each drum having its own mic. Using microphones with tight polar patterns on toms helps to isolate the sound from each drum. It is possible to share one mic with two toms, but then, a microphone with a wider polar pattern should be used. The snare requires a mic that can handle very high SPL, so a dynamic mic is usually chosen. To avoid picking up the hi-hat in the snare mic, aim the null of the snare mic towards the hi-hat. The brilliance and high frequencies of cymbals are picked up best by a flat response condenser mic. 1. Overhead-Cymbals: One microphone over center of drum set, about 1 foot above drummer’s head (Position A); or use two spaced or crossed microphones for stereo (Positions A or B). See “Stereo Microphone Techniques” section 28 Natural; sounds like Picks up ambience and leakage. For cymbal drummer hears set pickup only, roll off low frequencies. Boost at 10,000 Hz for added sizzle. To reduce excessive cymbal ringing, apply masking tape in radial strips from bell to rim. Microphone Techniques for LIVE SOUND Microphone Placement Tonal Balance Comments Full, smooth Tape gauze pad or handkerchief on top head to tighten sound. Boost at 5,000 Hz for attack, if necessary. 2. Snare drum: Just above top head at edge of drum, aiming at top head. Coming in from front of set on boom (Position C); or miniature microphone mounted directly on drum 3. Bass drum (kick drum): Placing a pad of paper towels where the beater hits the drum will lessen boominess. If you get rattling or buzzing problems with the drum, put masking tape across the drum head to damp out these nuisances. Placing the mic off center will pick up more overtones. Remove front head if necessary. Mount microphone on boom arm inside drum a few inches from beater head, about 1/3 of way in from edge of head (Position D); or place surfacemount microphone inside drum, on damping material, with microphone element facing beater head Full, good impact Put pillow or blanket on bottom of drum against beater head to tighten beat. Use wooden beater, or loosen head, or boost around 2,500 Hz for more impact and punch. Full, good impact Inside drum gives best isolation. Boost at 5,000 Hz for attack, if necessary. Natural, bright Place microphone or adjust cymbal height so that puff of air from closing hi-hat cymbals misses mike. Roll off bass to reduce low-frequency leakage. To reduce hi-hate leakage into snare-drum microphone, use small cymbals vertically spaced 1/2” apart. 4. Tom-toms: One microphone between every two tom-toms, close to top heads (Position E); or one microphone just above each tom-tom rim, aiming at top head (Position F); or one microphone inside each tom-tom with bottom head removed; or miniature microphone mounted directly on drum 5. Hi-hat: Aim microphone down towards the cymbals, a few inches over edge away from drummer (Position G). Or angle snare drum microphone slightly toward hi-hat to pick up both snare and hi-hat 29 Microphone Techniques for LIVE SOUND Microphone Placement Tonal Balance Comments 6. Snare, hi-hat and high tom: Place single microphone a few inches from Natural snare drum edge, next to high tom, just above top head of tom. Microphone comes in from front of the set on a boom (Position H) In combination with Placements 3 and 7, provides good pickup with minimum number of microphones. Tight sound with little leakage. 7. Cymbals, floor tom and high tom: Using single microphone, place its grille just above floor tom, aiming up toward cymbals and one of high tomes (Position I) Natural In combination with Placements 3 and 6, provides good pickup with minimum number of microphones. Tight sound with little leakage. One microphone: Use Placement 1. Placement 6 may work if the drummer limits playing to one side of the drum set. Two microphones: Placements 1 and 3; or 3 and 6. Three microphones: Placements 1, 2, and 3; or 3, 6, and 7. Four microphones: Placements 1, 2, 3, and 4. Five microphones: Placements 1, 2, 3, 4, and 5. More microphones: Increase number of tom-tom microphones as needed. Use a small microphone mixer to submix multiple drum microphones into one channel. Timbales, congas, bongos: One microphone aiming down between pair of drums, just above top heads Natural Provides full sound with good attack. Natural Experiment with distance and angles if sound is too bright. Tambourine: One microphone placed 6 to 12 inches from instrument 30 Microphone Techniques for LIVE SOUND Microphone Placement Tonal Balance Comments Bright, with plenty of attack Allow clearance for movement of pan. Steel Drums: Tenor, Second Pan, Guitar One microphone placed 4 inches above each pan Decent if used for tenor or second pans. Too boomy with lower voiced pans. Microphone placed underneath pan Cello, Bass One microphone placed 4 - 6 inches above each pan Natural Can double up pans to a single microphone. Natural Pan two microphones to left and right for stereo. See “Stereo Microphone Techniques” section. Bright, with lots of attack For less attack, use rubber mallets instead of metal mallets. Plastic mallets will give a medium attack. Tonal Balance Comments Voice range, semi-distant Use flat response, unidirectional microphones. Use minimum number of microphones needed to avoid overlapping pickup area. Use shock mount if needed. Voice range, semi-distant Use flat response, unidirectional microphones. Use minimum number of microphones needed to avoid overlapping pickup area. Voice range, on mic Multiple wireless systems must utilize different frequencies. Use lavaliere or handheld microphones as appropriate. Xylophone, marimba, vibraphone: Two microphones aiming down toward instrument, about 1 1/2 feet above it, spaced 2 feet apart, or angled 135 º apart with grilles touching Glockenspiel: One microphone placed 4 - 6 inches above bars Stage area miking Downstage: Surface-mount microphones along front of stage aimed upstage, one microphone center stage; use stage left and stage right mics as needed, approximately 1 per 10-15 feet Upstage: Microphones suspended 8 -10 feet above stage aimed upstage, one microphone center stage; use stage left and stage right mics as needed, approximately 1 per 10-15 feet Spot pickup: Use wireless microphones on principal actors; mics concealed in set; “shotgun” microphones from above or below 31 Microphone Techniques for LIVE SOUND Stereo Microphone Techniques These methods are recommended for pickup of orchestras, bands, choirs, pipe organs, quartets, soloists. They also may work for jazz ensembles, and are often used on overhead drums and close-miked piano. Use two microphones mounted on a single stand with a stereo microphone stand adapter (such as the Shure A27M). Or mount 2 or 3 microphones on separate stands. Set the microphones in the desired stereo pickup arrangement (see below). Coincident Techniques Comments Microphone diaphragms close together and aligned vertically; microphones angled apart. Example: 1350 angling (X-Y). Tends to provide a narrow stereo spread (the reproduced ensemble does not always spread all the way between the pair of playback loud-speakers). Good imaging. Mono-compatible. MS (Mid-Side) Comments A front-facing cardioid cartridge and a side-facing bidirectional cartridge are mounted in a single housing. Their outputs are combined in a matrix circuit to yield discrete left and right outputs. Provides good stereo spread, excellent stereo imaging and localization. Some types allow adjustable stereo control. Mono-compatible. Near-Coincident Techniques Microphones angled and spaced apart 6 to 10 inches between grilles. Examples: 110 0 angled, 7-inch spacing 32 For sound reinforcement, stereo mic techniques are only warranted for a stereo sound system and even then, they are generally only effective for large individual instruments, such as piano or miramba, or small instrument groups, such as drum kit, string section or vocal chorus. Relatively close placement is necessary to achieve useable gainbefore-feedback. Comments Tends to provide accurate image localization. Microphone Techniques for LIVE SOUND Spaced Techniques Comments Two microphones spaced several feet apart horizontally, both aiming straight ahead toward ensemble. Example: Microphones 3 to 10 feet apart. Tends to provide exaggerated separation unless microphone spacing is 3 feet. However, spacing the microphones 10 feet apart improves overall coverage. Produces vague imaging for off-center sound sources. Provides a “warm” sense of ambience. Musical Ensemble (Top View) Musical Ensemble Three microphones spaced several feet apart horizontally, aiming straight ahead toward ensemble. Center microphone signal is split equally to both channels. Example: Microphones 5 feet apart. Improved localization compared to two spaced microphones. (Top View) 33 Microphone Techniques for LIVE SOUND Selection Guide Shure Microphone Selection Guide Instrument Vocal Live Vocals Live Choirs KSM9 Beta58A SM58 Beta54 Beta87A Beta87C SM87A SM86 PG58 55SH Series II WH30 MX202 EZO SM81 SM94 PG81 Karaoke SM58S SM48S 565 PG58 PG48 Studio Vocals KSM44 KSM32 KSM27 KSM9 SM7A Beta87A Beta87C SM87A SM86 Studio Ensemble Vocals KSM44 KSM32 KSM27 KSM141 KSM137 KSM109 Spoken Word Beta53 SM48 PG48 Studio Instrument Brass / Saxophone KSM141 KSM137 KSM109 KSM44 KSM32 KSM27 SM81 Beta98H/C KSM44 KSM32 KSM27 Beta57A Beta98 S Beta56 A SM57 PG56 PG57 Orchestra KSM141 KSM137 KSM109 KSM44 KSM32 KSM27 SM81 SM94 PG81 Strings KSM141 KSM137 KSM109 KSM44 KSM32 KSM27 SM81 SM94 PG81 Beta98S Woodwinds KSM44 KSM32 KSM27 KSM141 KSM137 KSM109 SM81 Beta98H/C Beta98 S1 PG81 34 1 Bell mounted with A98KCS clamp. Drum 2 A56D enables microphone to mount on rim. Acoustic Guitar KSM141 KSM137 KSM109 KSM44 KSM32 KSM27 SM81 Beta57A SM57 PG81 PG57 Acoustic Bass KSM44 KSM32 KSM27 KSM141 KSM137 KSM109 Beta52A SM81 SM94 PG81 Guitar Amp Other Bass Amp Kick Drum Mallets Beta52A SM7B Beta57A Beta56A SM57 PG52 SM94 PG57 PG81 Beta52A Beta91 PG52 Beta57A SM57 KSM44 KSM32 KSM27 KSM141 KSM137 KSM109 SM81 SM94 PG81 Snare Drum2 Leslie Speaker KSM44 KSM32 KSM27 Beta91 Beta57A Beta56A SM57 Piano / Organ KSM44 KSM32 KSM27 KSM141 KSM137 KSM109 SM81 Beta91 PG81 SM94 MX202 Harmonica 520DX SM57 SM58 PG57 Beta56A Beta57A SM57 KSM141 KSM137 KSM109 KSM44 KSM32 KSM27 SM94 PG57 3 A50D enables microphone to mount on rim. Beta57A2,3 Beta56A2 SM572,3 PG56 PG572,3 Rack / Floor Toms Beta98D/S Beta57A2,3 Beta56A2 SM572,3 PG56 PG572,3 (Percussion) KSM141 KSM137 KSM109 KSM44 KSM32 KSM27 SM81 Beta57A SM57 PG57 Congas Beta98D/S Beta56A2 Beta57A2 SM572 PG56 PG572 Cymbals KSM141 KSM137 KSM109 KSM44 KSM32 KSM27 SM81 SM94 PG81 4 For optimum flexibility, use A27M stereo microphone mount. Sampling / Effects KSM44 KSM32 KSM27 KSM141 KSM137 KSM109 VP88 SM81 SM94 Live Stereo Recording KSM141(pair)4 KSM137(pair)4 KSM109(pair)4 KSM44(pair)4 KSM32(pair)4 KSM27(pair)4 SM81(pair)4 SM94(pair)4 VP88 (M-S stereo) Voice-Over KSM44 KSM32 KSM27 SM7B Beta58A SM58 SM815 Beta87C Beta87A 5 With A81G. Glossary 3-to-1 Rule-When using multiple microphones, the distance between microphones should be at least 3 times the distance from each microphone to its intended sound source. Absorption-The dissipation of sound energy by losses due to sound absorbent materials. Active Circuitry-Electrical circuitry which requires power to operate, such as transistors and vacuum tubes. Ambience-Room acoustics or natural reverberation. Amplitude-The strength or level of sound pressure or voltage. Audio Chain-The series of interconnected audio equipment used for recording or PA. Backplate-The solid conductive disk that forms the fixed half of a condenser element. Balanced-A circuit that carries information by means of two equal but opposite polarity signals, on two conductors. Bidirectional Microphone-A microphone that picks up equally from two opposite directions. The angle of best rejection is 90 deg. from the front (or rear) of the microphone, that is, directly at the sides. Boundary/Surface Microphone-A microphone designed to be mounted on an acoustically reflective surface. Cardioid Microphone-A unidirectional microphone with moderately wide front pickup (131 deg.). Angle of best rejection is 180 deg. from the front of the microphone, that is, directly at the rear. Cartridge (Transducer)-The element in a microphone that converts acoustical energy (sound) into electrical energy (the signal). Close Pickup-Microphone placement within 2 feet of a sound source. Comb Filtering-An interference effect in which the frequency response exhibits regular deep notches. Condenser Microphone-A microphone that generates an electrical signal when sound waves vary the spacing between two charged surfaces: the diaphragm and the backplate. Microphone Techniques for LIVE SOUND Critical Distance-In acoustics, the distance from a sound source in a room at which the direct sound level is equal to the reverberant sound level. Current-Charge flowing in an electrical circuit. Analogous to the amount of a fluid flowing in a pipe. Decibel (dB)-A number used to express relative output sensitivity. It is a logarithmic ratio. Diaphragm-The thin membrane in a microphone which moves in response to sound waves. Diffraction-The bending of sound waves around an object which is physically smaller than the wavelength of the sound. Direct Sound-Sound which travels by a straight path from a sound source to a microphone or listener. Distance Factor-The equivalent operating distance of a directional microphone compared to an omnidirectional microphone to achieve the same ratio of direct to reverberant sound. Distant Pickup-Microphone placement farther than 2 feet from the sound source. Dynamic Microphone-A microphone that generates an electrical signal when sound waves cause a conductor to vibrate in a magnetic field. In a moving-coil microphone, the conductor is a coil of wire attached to the diaphragm. Dynamic Range-The range of amplitude of a sound source or the range of sound level that a microphone can successfully pick up. Echo-Reflection of sound that is delayed long enough (more than about 50 msec.) to be heard as a distinct repetition of the original sound. Electret-A material (such as Teflon) that can retain a permanent electric charge. EQ-Equalization or tone control to shape frequency response in some desired way. Feedback-In a PA system consisting of a microphone, amplifier, and loudspeaker feedback is the ringing or howling sound caused by amplified sound from the loudspeaker entering the microphone and being re-amplified. 35 Microphone Techniques for LIVE SOUND Flat Response-A frequency response that is uniform and equal at all frequencies. Frequency-The rate of repetition of a cyclic phenomenon such as a sound wave. Frequency Response Tailoring Switch-A switch on a microphone that affects the tone quality reproduced by the microphone by means of an equalization circuit. (Similar to a bass or treble control on a hi-fi receiver.) Frequency Response-A graph showing how a microphone responds to various sound frequencies. It is a plot of electrical output (in decibels) vs. frequency (in Hertz). Fundamental-The lowest frequency component of a complex waveform such as musical note. It establishes the basic pitch of the note. Gain-Amplification of sound level or voltage. Gain-Before-Feedback-The amount of gain that can be achieved in a sound system before feedback or ringing occurs. Harmonic-Frequency components above the fundamental of a complex waveform. They are generally multiples of the fundamental which establish the timbre or tone of the note. Hypercardioid-A unidirectional microphone with tighter front pickup (105 deg.) than a supercardioid, but with more rear pickup. Angle of best rejection is about 110 deg. from the front of the microphone. Impedance-In an electrical circuit, opposition to the flow of alternating current, measured in ohms. A high impedance microphone has an impedance of 10,000 ohms or more. A low impedance microphone has an impedance of 50 to 600 ohms. Interference-Destructive combining of sound waves or electrical signals due to phase differences. Inverse Square Law-States that direct sound levels increase (or decrease) by an amount proportional to the square of the change in distance. Isolation-Freedom from leakage; ability to reject unwanted sounds. 36 Glossary Leakage-Pickup of an instrument by a microphone intended to pick up another instrument. Creative leakage is artistically favorable leakage that adds a “loose” or “live” feel to a recording. NAG-Needed Acoustic Gain is the amount of gain that a sound system must provide for a distant listener to hear as if he or she was close to the unamplified sound source. Noise-Unwanted electrical or acoustic interference. Noise Canceling-A microphone that rejects ambient or distant sound. NOM-Number of open microphones in a sound system. Decreases gain-before-feedback by 3dB everytime NOM doubles. Omnidirectional Microphone-A microphone that picks up sound equally well from all directions. Overload-Exceeding the signal level capability of a microphone or electrical circuit. PAG-Potential Acoustic Gain is the calculated gain that a sound system can achieve at or just below the point of feedback. Phantom Power-A method of providing power to the electronics of a condenser microphone through the microphone cable. Phase-The “time” relationship between cycles of different waves. Pickup Angle / Coverage Angle-The effective arc of coverage of a microphone, usually taken to be within the 3dB down points in its directional response. Pitch-The fundamental or basic frequency of a musical note. Polar Pattern (Directional Pattern, Polar Response)A graph showing how the sensitivity of a microphone varies with the angle of the sound source, at a particular frequency. Examples of polar patterns are unidirectional and omnidirectional. Polarization-The charge or voltage on a condenser microphone element. Glossary Pop Filter-An acoustically transparent shield around a microphone cartridge that reduces popping sounds. Often a ball-shaped grille, foam cover or fabric barrier. Pop-A thump of explosive breath sound produced when a puff of air from the mouth strikes the microphone diaphragm. Occurs most often with “p,” “t,” and “b” sounds. Presence Peak-An increase in microphone output in the “presence” frequency range of 2000 Hz to 10,000 Hz. A presence peak increases clarity, articulation, apparent closeness, and “punch.” Proximity Effect-The increase in bass occurring with most unidirectional microphones when they are placed close to an instrument or vocalist (within 1 ft.). Does not occur with omnidirectional microphones. Rear Lobe-A region of pickup at the rear of a supercardioid or hypercardioid microphone polar pattern. A bidirectional microphone has a rear lobe equal to its front pickup. Reflection-The bouncing of sound waves back from an object or surface which is physically larger than the wavelength of the sound. Refraction-The bending of sound waves by a change in the density of the transmission medium, such as temperature gradients in air due to wind. Microphone Techniques for LIVE SOUND Sound Chain-The series of interconnected audio equipment used for recording or PA. Sound Reinforcement-Amplification of live sound sources. Speed of Sound-The speed of sound waves, about 1130 feet per second in air. SPL-Sound Pressure Level is the loudness of sound relative to a reference level of 0.0002 microbars. Standing Wave-A stationary sound wave that is reinforced by reflection between two parallel surfaces that are spaced a wavelength apart. Supercardioid Microphone-A unidirectional microphone with tighter front pickup angle (115 deg.) than a cardioid, but with some rear pickup. Angle of best rejection is 126 deg. from the front of the microphone, that is, 54 deg. from the rear. Timbre-The characteristic tone of a voice or instrument; a function of harmonics. Transducer-A device that converts one form of energy to another. A microphone transducer (cartridge) converts acoustical energy (sound) into electrical energy (the audio signal). Transient Response-The ability of a device to respond to a rapidly changing input. Resistance-The opposition to the flow of current in an electrical circuit. It is analogous to the friction of fluid flowing in a pipe. Unbalanced-A circuit that carries information by means of one signal on a single conductor. Reverberation-The reflection of a sound a sufficient number of times that it becomes non-directional and persists for some time after the source has stopped. The amount of reverberation depends on the relative amount of sound reflection and absorption in the room. Unidirectional Microphone-A microphone that is most sensitive to sound coming from a single directionin front of the microphone. Cardioid, supercardioid, and hypercardioid microphones are examples of unidirectional microphones. Rolloff-A gradual decrease in response below or above some specified frequency. Voice Coil-Small coil of wire attached to the diaphragm of a dynamic microphone. Sensitivity-The electrical output that a microphone produces for a given sound pressure level. Voltage-The potential difference in an electric circuit. Analogous to the pressure on fluid flowing in a pipe. Shaped Response-A frequency response that exhibits significant variation from flat within its range. It is usually designed to enhance the sound for a particular application. Wavelength-The physical distance between the start and end of one cycle of a soundwave. 37 Microphone Techniques for LIVE SOUND Rick Waller Tim Vear Now residing in the Chicago area, Rick grew up near Peoria, Tim is a native of Chicago who has come to the audio field Illinois. An interest in the technical and musical aspects of audio as a way of combining a lifelong interest in both entertainment has led him to pursue a career as both engineer and musician. and science. He has worked as an engineer in live sound, He received a BS degree in Electrical Engineering from the recording and broadcast, has operated his own recording studio University of Illinois at Urbana/Champaign, where he specialized and sound company, and has played music professionally since in acoustics, audio synthesis and radio frequency theory. Rick is high school. an avid keyboardist, drummer and home theater hobbyist and At the University of Illinois, Urbana-Champaign, Tim earned has also worked as a sound engineer and disc jockey. Currently a BS in Aeronautical and Astronautical Engineering with a minor he is an associate in the Applications Engineering Group at Shure in Electrical Engineering. During this time he also worked as chief Incorporated. In this capacity Rick provides technical support to technician for both the Speech and Hearing Science and domestic and international customers, writing and conducting Linguistics departments. seminars on wired and wireless microphones, mixers and other audio topics. In his tenure at Shure Incorporated, Tim has served in a technical support role for the sales and marketing departments, providing product and applications training for Shure customers, John Boudreau John, a lifelong Chicago native, has had extensive experience seminars for a variety of domestic and international audiences, as a musician, a recording engineer, and a composer. His desire including the National Systems contractors Association, the Audio to better combine the artistic and technical aspects of music led Engineering Society and the Society of Broadcast Engineers. him to a career in the audio field. Tim has authored several publications for Shure Incorporated and Having received a BS degree in Music Business from Elmhurst College, John performed and composed for both a Jazz and a Rock band prior to joining Shure Incorporated in 1994 as an associate in the Applications Engineering group. At Shure, John led many audio product training seminars and clinics, with an eye to helping musicians and others affiliated with the field use technology to better fulfill their artistic interpretations. No longer a Shure associate, John continues to pursue his interests as a live and recorded sound engineer for local bands and venues, as well as writing and recording for his own band. 38 dealers, installers, and company staff. He has presented his articles have appeared in Recording Engineer/Producer, Live Sound Engineering, Creator, and other publications. Additional Shure Publications Available: Printed or electronic versions of the following guides are available free of charge. To obtain your complimentary copies, call one of the phone numbers listed below or visit www.shure.com. • Selection and Operation of Wireless Microphone Systems • Audio Systems Guide for Video Production • Audio Systems Guide for Houses of Worship • Microphone Techniques for Studio Recording Our Dedication to Quality Products Shure offers a complete line of microphones and wireless microphone systems for everyone from first-time users to professionals in the music industry–for nearly every possible application. For over eight decades, the Shure name has been synonymous with quality audio. All Shure products are designed to provide consistent, high-quality performance under the most extreme real-life operating conditions. www.shure.com United States: Shure Incorporated 5800 West Touhy Avenue Niles, IL 60714-4608 USA Europe, Middle East, Africa: Shure Europe GmbH Wannenäckerstr. 28, 74078 Heilbronn, Germany Phone: 847-600-2000 Fax: 847-600-1212 Email: [email protected] Phone: 49-7131-72140 Fax: 49-7131-721414 Email: [email protected] ©2007 ©2007Shure ShureIncorporated Incorporated AL1266H AL0000 10M 12/07 Asia, Pacific: Shure Asia Limited 3/F, Citicorp Centre 18 Whitfield Road Causeway Bay, Hong Kong Canada, Latin America, Caribbean: Shure Incorporated 5800 West Touhy Avenue Niles, IL 60714-4608 USA Phone: 852-2893-4290 Fax: 852-2893-4055 Email: [email protected] Phone: 847-600-2000 Fax: 847-600-6446 Email: [email protected] A Shure Educational Publication MICROPHONE TECHNIQUES RECORDING Microphone Techniques for RECORDING Ta b l e o f C o n t e n t s Introduction: Selection and Placement of Microphones ............. 4 Section One .................................................................................. 5 Microphone Techniques ........................................................ 5 Vocal Microphone Techniques ............................................... 5 Spoken Word/“Podcasting” ................................................... 7 Acoustic String and Fretted Instruments ................................ 8 Woodwinds .......................................................................... 13 Brass ................................................................................... 14 Amplified Instruments .......................................................... 15 Drums and Percussion ........................................................ 18 Stereo .................................................................................. 21 Introduction: Fundamentals of Microphones, Instruments, and Acoustics ....................................................... 23 Section Two ................................................................................ 24 Microphone Characteristics ................................................. 24 Instrument Characteristics ................................................... 27 Acoustic Characteristics ....................................................... 28 Shure Microphone Selection Guide ..................................... 32 Shure Recording Microphone Lockers ................................ 33 Glossary ............................................................................... 34 On the cover: Shure’s Performance Listening Center featuring state-of-the-art recording and product testing capabilities. Photo by Frank Dina/Shure Inc. Appendix A: The Decibel ..................................................... 37 Internal application photography by Cris Tapia/Shure Inc. About the Authors ................................................................ 39 Appendix B: Transient Response ......................................... 38 Recording 3 Microphone Techniques for RECORDING Introduction The selection and placement of microphones can have a major influence on the sound of an acoustic recording. It is a common view in the recording industry that the music played by a skilled musician with a quality instrument properly miked can be sent directly to the recorder with little or no modification. This simple approach can often sound better than an instrument that has been reshaped by a multitude of signal processing gear. In this guide, Shure Application Engineers describe particular microphone techniques and placement: techniques to pick up a natural tonal balance, techniques to help reject unwanted sounds, and even techniques to create special effects. Following this, some fundamentals of microphones, instruments, and acoustics are presented. Section One 4 Microphone Techniques for RECORDING SECTION ONE Microphone Techniques Here is a very basic, general procedure to keep in mind when miking something that makes sound: 1) Use a microphone with a frequency response that is suited to the frequency range of the sound, if possible, or filter out frequencies above and/or below the highest and lowest frequencies of the sound. 2) Place the microphone at various distances and positions until you find a spot where you hear from the studio monitors the desired tonal balance and the desired amount of room acoustics. If you don’t like it, try another position, try another microphone, try isolating the instrument further, or change the sound of the instrument itself. For example, replacing worn out strings will change the sound of a guitar. 3) Often you will encounter poor room acoustics, or pickup of unwanted sounds. In these cases, place the microphone very close to the loudest part of the instrument or isolate the instrument. Again, experiment with microphone choice, placement and isolation, to minimize the undesirable and accentuate the desirable direct and ambient acoustics. Microphone technique is largely a matter of personal taste. Whatever method sounds right for the particular sound, instrument, musician, and song is right. There is no one ideal way to place a microphone. There is also no one ideal microphone to use on any particular instrument. Choose and place the microphone to get the sound you want. We recommend experimenting with all sorts of microphones and positions until you create your desired sound. However, the desired sound can often be achieved more quickly by understanding basic microphone characteristics, sound-radiation properties of musical instruments, and basic room acoustics. Vocal Microphone Techniques Individual Vocals Microphones with various polar patterns can be used in vocal recording techniques. Consider recording a choral group or vocal ensemble. Having the vocalists circle around an omnidirectional mic allows well trained singers to perform as they would live: creating a blend of voices by changing their individual singing levels and timbres. Two cardioid mics, positioned back to back could be used for this same application. An omnidirectional mic may be used for a single vocalist as well. If the singer is in a room with ambience and reverb that add to the desired effect, the omnidirectional mic will capture the room sound as well as the singer’s direct voice. By changing the distance of the vocalist to the microphone, you can adjust the balance of the direct voice to the ambience. The closer the vocalist is to the mic, the more direct sound is picked up relative to the ambience. The standard vocal recording environment usually captures the voice only. This typically requires isolation and the use of a unidirectional mic. Isolation can be achieved with baffles surrounding the vocalist like a “shell” or some other method of reducing reflected sound from the room. Remember even a music stand can cause reflections back to the mic. The axis of the microphone should usually be pointed somewhere between the nose and mouth to pick up the complete sound of the voice. Though the mic is usually directly in front of the singer’s mouth, a slightly off-axis placement may help to avoid explosive sounds from breath blasts or certain consonant sounds such as “p”, “b”, “d”, or “t”. Placing the mic even further off-axis, or the use of an accessory pop filter, may be necessary to fully eliminate this problem. While many vocals are recorded professionally in an isolation booth with a cardioid condenser microphone, other methods of vocal recording are practiced. For instance, a rock band’s singers may be uncomfortable in the isolated environment described earlier. They may be used to singing in a loud environment with a monitor loudspeaker as the reference. This is a typical performance situation and forces them to sing louder and push their voices in order to hear themselves. This is a difficult situation to recreate with headphones. A technique that has been used successfully in this situation is to bring the singers into the control room to perform. This would be especially convenient for project studios that exist in only one room. Once in that environment, a supercardioid dynamic microphone could be used in conjunction with the studio monitors. The singer faces the monitors to hear a mix of music and voice together. The supercardioid mic rejects a large amount of the sound projected from the speakers if the rear axis of the microphone is aimed between the speakers and the speakers are aimed at the null angle of the mic (about 65 degrees on either side of its rear axis). Just as in live sound, you are using 5 Microphone Techniques for RECORDING 0.6 - 1m (2 - 3 ft) 1.8 - 3m (6 - 9 ft) Choir microphone positions - top view the polar pattern of the mic to improve gain-before-feedback and create an environment that is familiar and encouraging to the vocalists. Now the vocalist can scream into the late hours of the night until that vocal track is right. Ensemble Vocals A condenser is the type of microphone most often used for choir applications. They are generally more capable of flat, wide-range frequency response. The most appropriate directional type is a unidirectional, usually a cardioid. A supercardioid or a hypercardioid microphone may be used for a slightly greater reach or for more ambient sound rejection. Balanced low-impedance output is used exclusively, and the sensitivity of a condenser microphone is desirable because of the greater distance between the sound source and the microphone. Application of choir microphones falls into the category known as “area” coverage. Rather than one microphone per sound source, the object is to pick up multiple sound sources (or a “large” sound source) with one (or more) microphone(s). Obviously, this introduces the possibility of interference effects unless certain basic principles (such as the “3-to-1 rule”) are followed, as discussed below. For one microphone picking up a typical choir, the suggested placement is a few feet in front of, and a few feet above, the heads of the first row. It should be centered in front of the choir and aimed at the last row. In this configuration, a cardioid microphone can “cover” up to 15-20 voices, arranged in a rectangular or wedge-shaped section. For larger or unusually shaped choirs, it may be necessary to use more than one microphone. Since the pickup angle of a microphone is a function of its directionality (approximately 130 degrees for a cardioid), broader coverage requires more distant placement. 6 In order to determine the placement of multiple microphones for choir pickup, remember the following rules: observe the 3-to-1 rule (see glossary); avoid picking up the same sound source with more than one microphone; and finally, use the minimum number of microphones. For multiple microphones, the objective is to divide the choir into sections that can each be covered by a single microphone. If the choir has any existing physical divisions (aisles or boxes), use these to define basic sections. If the choir is grouped according to vocal range (soprano, alto, tenor, bass), these may serve as sections. If the choir is a single, large entity, and it becomes necessary to choose sections based solely on the coverage of the individual microphones, use the following spacing: one microphone for each lateral section of approximately 6 to 9 feet. If the choir is unusually deep (more than 6 or 8 rows), it may be divided into two vertical sections of several rows each, with 0.6 - 1m aiming angles (2 - 3 ft) adjusted accordingly. In any case, it is better to use too few microphones 0.6 - 1m than too many. (2 - 3 ft) In a goodsounding space, a pair of microphones in a stereo configuration can provide realistic reproduction. (See page 22.) Microphone positions - side view Microphone Techniques for RECORDING Spoken Word/ “Podcasting” Countless “how-to” articles have been written on podcasting, which is essentially a current trend in spoken word distribution, but few offer many tips on how to properly record the human voice. Below are some suggestions: 1. Keep the microphone 6 –12” from your mouth. Generally, keep the microphone as close as possible to your mouth to avoid picking up unwanted room reflections and reverberation. Do not get too close either. Proximity effect, which is an increase in low frequency response that occurs as you get closer to a directional microphone, can cause your voice to sound “muddy” or overly bassy. 2. Aim the microphone toward your mouth from below or above. This placement minimizes “popping” caused by plosive consonants (e.g. “p” or “t”). 3. Use an external pop filter. Though most microphones have some sort of builtin windscreen, an additional filter will provide extra insurance against “p” pops. The pop filter can also serve as a reference to help you maintain a consistent distance from the microphone. (See Image 1.) 4. Keep the microphone away from reflective surfaces. Reflections caused by hard surfaces, such as tabletops or music stands, can adversely affect the sound quality captured by the microphone. (See the section “Phase relationships and interference effects” page 30.) Image 1: Example of an external pop filter 5. Speak directly into the microphone. High frequencies are very directional, and if you turn your head away from the microphone, the sound captured by the microphone will get noticeably duller. 7 Microphone Techniques for RECORDING Acoustic String and Fretted Instruments Experimentation with mic placement provides the ability to achieve accurate and pleasing sound reproduction on these complex sound sources. It is also an opportunity for exploring sound manipulation, giving the studio engineer many paths to the final mix. Whether you are involved in a music studio, a commercial studio, or a project studio, you should continue to explore different methods of achieving the desired results. The possibilities are limited only by time and curiosity. Acoustic Guitar (Also Dobro, Dulcimer, Mandolin, Ukelele) When recording an acoustic guitar, try placing one mic three to six inches away, directly in front of the sound hole. Then put another microphone, of the same type, four feet away. This will allow you to hear the instrument and an element of room ambience. Record both mics dry and flat (no effects or EQ), each to its own track. These two tracks will sound vastly different. Combining them may provide an open sound with the addition of the distant mic. Giving the effect of two completely different instruments or one in a stereo hallway may be achieved by enhancing each signal with EQ and effects unique to the sound you want to hear. Try the previously mentioned mic technique on any acoustic instrument. Attempt to position the mic in different areas over the instruments, listening for changes in timbre. You will find different areas offer different tonal characteristics. Soon you should develop “an ear” for finding instruments’ sweet spots. In addition, the artist and style of music should blend with your experiences and knowledge to generate the desired effect. Above Front 4 6” 3 2 1 Various microphone positions for acoustic guitar 8 Microphone Techniques for RECORDING Microphone Placement Tonal Balance Comments Bassy Good starting placement when leakage is a problem. Roll off bass for a more natural sound (more for a uni than an omni). 2 3 inches from sound hole Very bassy, boomy, muddy, full Very good isolation. Bass roll-off needed for a natural sound. 3 4 to 8 inches from bridge Woody, warm, mellow. Mid-bassy, lacks detail Reduces pick and string noise. Natural, well-balanced, slightly bright Less pickup of ambiance and leakage than 3 feet from sound hole. Miniature microphone clipped outside of sound hole Natural, well-balanced Good isolation. Allows freedom of movement. Miniature microphone clipped inside sound hole Bassy, less string noise Reduces leakage. Test positions to find each guitar’s sweet spot. 3 inches from center of head Bassy, thumpy Limits leakage. Roll off bass for natural sound. 3 inches from edge of head Bright Limits leakage. Miniature microphone clipped to tailpiece aiming at bridge Natural Limits leakage. Allows freedom of movement. Natural Well-balanced sound. Well-defined Well-balanced sound, but little isolation. Bright Minimizes feedback and leakage. Allows freedom of movement. Acoustic Guitar: 1 8 inches from sound hole (see image 2) (see image 3) 4 6 inches above the side, over the bridge, and even with the front soundboard Banjo: Violin (Fiddle): A few inches from side Cello: 1 foot from bridge All String Instruments: Miniature microphone attached to strings between bridge and tailpiece Image 2: Acoustic guitar position 1 Image 3: Acoustic guitar position 3 9 Microphone Techniques for RECORDING Microphone Placement Tonal Balance Comments Acoustic Bass: (Upright Bass, String Bass, Bass Violin) 6 inches to 1 foot out front, just above bridge Well-defined Natural sound. A few inches from f-hole Full Roll off bass if sound is too boomy. Wrap microphone in foam padding (except for grille) and put behind bridge or between tailpiece and body Full, “tight” Minimizes feedback and leakage. Aiming toward player at part of soundboard, about 2 feet away Natural See “Stereo Microphone Techniques” section for other possibilities. Tape miniature microphone to soundboard Somewhat constricted Minimizes feedback and leakage. Harp: Grand Piano 8 9 5 1 4 7 2 8 1 4 Hammers 6”-12” 8” 6 10 Microphone Techniques for RECORDING Microphone Placement Tonal Balance Comments Natural, well-balanced Less pickup of ambience and leakage than 3 feet out front. Move microphone(s) farther from hammers to reduce attack and mechanical noises. Good coincident-stereo placement. See “Stereo Microphone Techniques” section. Natural, well-balanced, slightly bright Place one microphone over bass strings and one over treble strings for stereo. Phase cancellations may occur if the recording is heard in mono. 3 Aiming into sound holes Thin, dull, hard, constricted Very good isolation. Sometimes sounds good for rock music. Boost mid-bass and treble for more natural sound. 4 6 inches over middle strings, Muddy, boomy, dull, lacks attack Improves isolation. Bass roll-off and some treble boost required for more natural sound. Bassy, full Unobtrusive placement. Bassy, dull, full Unobtrusive placement. Grand Piano: 1 12 inches above middle strings, 8 inches horizontally from hammers with lid off or at full stick 2 8 inches above treble strings, as above (see image 4) (see image 5) 8 inches from hammers, with lid on short stick 5 Next to the underside of raised lid, centered on lid 6 Underneath the piano, aiming up at the soundboard 7 Surface-mount microphone mounted Bright, on underside of lid over lower treble strings, horizontally, close to hammers for brighter sound, further from hammers for more mellow sound 8 Two surface-mount microphones positioned on the closed lid, under the edge at its keyboard edge, approximately 2/3 of the distance from middle A to each end of the keyboard 9 Surface-mount microphone placed vertically on the inside of the frame, or rim, of the piano, at or near the apex of the piano’s curved wall Image 4: Grand piano position 2 well-balanced Excellent isolation. Experiment with lid height and microphone placement on piano lid for desired sounds. Bright, well-balanced, strong attack Excellent isolation. Moving “low” mic away from keyboard six inches provides truer reproduction of the bass strings while reducing damper noise. By splaying these two mics outward slightly, the overlap in the middle registers can be minimized. Full, natural Excellent isolation. Minimizes hammer and damper noise. Best if used in conjunction with two surface-mount microphones mounted to closed lid, as above. Image 5: Grand piano position 3 11 Microphone Techniques for RECORDING Microphone Placement Tonal Balance Comments Upright Piano: 1 Just over open top, above treble Natural (but lacks Good placement when only one deep bass), picks up microphone is used. hammer attack strings 2 Just over open top, above bass strings 3 Inside top near the bass and treble stings 4 8 inches from bass side of soundboard 5 8 inches from treble side of soundboard 6 Aiming at hammers from front, several inches away (remove front panel) 1 foot from center of soundboard on hard floor or one-foot-square plate on carpeted floor, aiming at piano (soundboard should face into room) Slightly full or tubby, picks up hammer attack Mike bass and treble strings for stereo. Natural, picks up hammer attack Minimizes feedback and leakage. Use two microphones for stereo. Full, slightly tubby, no hammer attack Use this placement with the following placement for stereo. Thin, constricted, no hammer attack Use this placement with the preceding placement for stereo. Bright, picks up hammer attack Mike bass and treble strings for stereo. Natural, good presence Minimize pickup of floor vibrations by mounting microphone in low-profile shock-mounted microphone stand. Open 2 3 12 Open 1 3 Mic Mic 4 5 1 2 Mics Open 6 Microphone Techniques for RECORDING Woodwinds Microphone Placement Tonal Balance Comments Saxophone: With the saxophone, the sound is fairly well distributed between the finger holes and the bell. Miking close to the finger holes will result in key noise. The soprano sax must be considered separately because its bell does not curve upward. This means that, unlike all other saxophones, placing a microphone toward the middle of the instrument will not pick-up the sound from the key holes and the bell simultaneously. The saxophone has sound characteristics similar to the human voice. Thus, a shaped response microphone designed for voice works well. Image 6: Example of saxophone mic placement for natural sound A few inches from and aiming into bell Bright Minimizes feedback and leakage. A few inches from sound holes Warm, full Picks up fingering noise. A few inches above bell and aiming at sound holes (see image 6) Natural Good recording technique. Miniature microphone mounted on bell Bright, punchy Maximum isolation, up-front sound. Flute: The sound energy from a flute is projected both by the embouchure and by the first open fingerhole. For good pickup, place the mic as close as possible to the instrument. However, if the mic is too close to the mouth, breath noise will be apparent. Use a windscreen on the mic to overcome this difficulty. A few inches from area between mouthpiece and first set of finger holes Natural, breathy Pop filter or windscreen may be required on microphone. A few inches behind player’s head, aiming at finger holes Natural Reduces breath noise. About 1 foot from sound holes Natural Provides well-balanced sound. A few inches from bell Bright Minimizes feedback and leakage. Oboe, Bassoon, Etc.: 13 Microphone Techniques for RECORDING Woodwinds (continued) Microphone Placement Tonal Balance Comments Full, bright Minimizes feedback and leakage. Microphone may be cupped in hands. One or two feet in front of instrument, centered Full range, natural sound Use two microphones for stereo or to pick up bass and treble sides separately. Miniature microphone mounted internally Emphasizes midrange Minimizes leakage. Allows freedom of movement. Tonal Balance Comments Harmonica: Very close to instrument Accordion: Brass Microphone Placement Trumpet, Cornet Trombone, Tuba: The sound from most brass instruments is very directional. Placing the mic off axis with the bell of the instrument will result in less pickup of high frequencies. 1 to 2 feet from bell (a couple of instruments can play into one microphone) On-axis to bell sounds bright; to one side sounds natural or mellow Close miking sounds “tight” and minimizes feedback and leakage. More distant placement gives fuller, more dramatic sound. Miniature microphone mounted on bell Bright Maximum isolation. Natural Watch out for extreme fluctuations on VU meter. French Horn: Microphone aiming toward bell 14 Microphone Techniques for RECORDING Amplified Instruments Another “instrument” with a wide range of characteristics is the loudspeaker. Anytime you are recording a guitar or bass cabinet, you are confronted with the acoustic nature of loudspeakers. A single loudspeaker is directional and displays different frequency characteristics at different angles and distances. On-axis at the center of a speaker tends to produce the most “bite”, while off-axis or edge placement of the microphone produces a more “mellow” sound. A cabinet with multiple loudspeakers has an even more complex output, especially if it has different speakers for bass and treble. As with most acoustic instruments, the desired sound develops at some distance away from the speaker. The most common approach is to close-mic an individual speaker. This is a habit people develop from viewing or doing live sound. In the live sound environment, most audio sources are close-miked to achieve the highest direct to ambient pickup ratios. Using unidirectional mics for close miking maximizes off-axis sound rejection as well. These elements lead to reduction of potential feedback opportunities. In the recording environment, the loudspeaker cabinet can be isolated and distant-mic techniques can be used to capture a more representative sound. Often, by using both a close and a distant (more than a few feet) mic placement at the same time, it is possible to record a sound which has a controllable balance between “presence” and “ambience”. Placement of loudspeaker cabinets can also have a significant effect on their sound. Putting cabinets on carpets can reduce brightness, while raising them off the floor can reduce low end. Open-back cabinets can be miked from behind as well as from the front. The distance from the cabinet to walls or other objects can also vary the sound. Again, move the instrument and the mic(s) around until you achieve something that you like! Top 3 Side 2 1 4 1 2 4 See page 16 for placement key. 15 Microphone Techniques for RECORDING Microphone Placement Tonal Balance Comments Electric Guitar: The electric guitar has sound characteristics similar to the human voice. Thus, a shaped response microphone designed for voice works well. 1 4 inches from grille cloth at Natural, well-balanced Small microphone desk stand may be used if loudspeaker is close to floor. Bassy Minimizes feedback and leakage. 3 Off-center with respect to Dull or mellow Microphone closer to edge of speaker cone results in duller sound. Reduces amplifier hiss noise. 4 3 feet from center of speaker cone Thin, reduced bass Picks up more room ambiance and leakage. 3 & 4 Good two-mic technique Natural Use condenser microphone for position 4 – adjust spacing to minimize phase issues. Miniature microphone draped over amp in front of speaker Emphasized midrange Easy setup, minimizes leakage. Microphone placed behind open back cabinet Depends on position Can be combined with mic in front of cabinet, but be careful of phase cancellation. center of speaker cone 2 1 inch from grille cloth at center of speaker cone speaker cone (see image 7) 3 16 4 Image 7: Example of a good “two mic technique” for electric guitar amp Microphone Techniques for RECORDING Bass Guitar: If the cabinet has only one speaker a single microphone should pick up a suitable sound with a little experimentation. If the cabinet has multiple speakers of the same type it is typically easiest to place the microphone to pick up just one speaker. Microphone Placement Placing the microphone between speakers can result in strong phase effects though this may be desirable to achieve a particular tone. However, if the cabinet is stereo or has separate bass and treble speakers multiple microphones may be required. Tonal Balance Comments Depends on brand of piano Roll off bass for clarity, roll off high frequencies to reduce hiss. Aim one microphone into top louvers 3 inches to 1 foot away Natural, lacks deep bass Good one-microphone pickup. Mike top louvers and bottom bass speaker 3 inches to 1 foot away Natural, well-balanced Excellent overall sound. Mike top louvers with two microphones, one close to each side; pan to left and right; mike bottom bass speaker 3 inches to 1 foot away and pan its signal to center Natural, well-balanced Stereo effect. Electric Keyboard Amp: Aim microphone at speaker as described in Electric Guitar Amplifier section Leslie Organ Speaker: 17 Microphone Techniques for RECORDING Drums and Percussion Drum Kit Miking – The drum kit is one of the most complicated sound sources to record. Although there are many different methods, some common techniques and principles should be understood. Since the different parts of the drum kit have widely varying sound they should be considered as individual instruments, or at least a small group of instrument types: Kick, Snare, Toms, Cymbals, and Percussion. Certain mic characteristics are extremely critical for drum usage. Dynamic Range – A drum can produce very high Sound Pressure Levels (SPLs). The microphone must be able to handle these levels. A dynamic microphone will usually handle high SPLs better than a condenser. Check the Maximum SPL in condenser microphone specifications. It should be at least 130 dB for closeup drum use. Directionality – Because we want to consider each part of the kit an individual instrument; each drum may have its own mic. Interference effects may occur due to the close proximity of the mics to each other and to the various drums. Choosing mics that can reject sound at certain angles and placing them properly can be pivotal in achieving an overall drum mix with minimal phase problems. Proximity Effect – Unidirectional mics may have excessive low frequency response when placed very close to the drums. A low frequency roll-off either on the microphone or at the mixer will help cure a “muddied” sound. However, proximity effect may also enhance low frequency response if desired. It can also be used to effectively reduce pickup of distant low frequency sources by the amount of lowrolloff used to control the closeup source. Typically, drums are isolated in their own room to prevent bleed through to microphones on other instruments. In professional studios it is common for the drums to be raised above the floor. This helps reduce low frequency transmission through the floor. Image 8: Example of bass (kick) drum mic placement A microphone for this use should have good low frequency response and possibly a boost in the attack range, although this can be done easily with EQ. The mic should be placed in the drum, in close proximity (1 - 6 inches), facing the beater head. (See position D in diagram on the following page.) Or for less “slap” just inside the hole. (See image 8.) 2 Snare Drum – This is the most piercing drum in the kit and almost always establishes tempo. In modern music it usually indicates when to clap your hands! This is an extremely transient drum with little or no sustain to it. Its attack energy is focused in the 4 - 6kHz range. Typically, the drum is miked on the top head at the edge of the drum with a cardioid or supercardioid microphone. (See position C in diagram on the following page; see image 9.) Here is a basic individual drum miking technique: 1 Bass (Kick) Drums – This drum’s purpose in most music is to provide transient, low-frequency energy bursts that help establish the primary rhythmic pattern of a song. The kick drum’s energy is primarily focused in two areas: very low-end timbre and “attack”. Although this varies by individual drum, the attack tends to be in the 2.55kHz range. Image 9: Example of snare drum mic placement 18 Microphone Techniques for RECORDING 3 Hi-Hats – These cymbals are primarily short, high frequency bursts used for time keeping, although the cymbals can be opened for a more loose sound. Many times the overhead mics will provide enough response to the high hat to eliminate the need for a separate hi-hat microphone. If necessary, a mic placed away from the puff of air that happens when hi-hats close and within four inches to the cymbals should be a good starting point. (See position G in diagram to right; see image 10.) Simpler methods of drum miking are used for jazz and any application where open, natural kit sounds are desired. Using fewer mics over sections of the drums is common. Also, one high quality mic placed at a distance facing the whole kit may Image 10: Example of capture the sounds of kit mic placement for hi-hats and room acoustics in an enjoyable balance. Additional mics may be added to reinforce certain parts of the kit that are used more frequently. 5 Overheads – The cymbals perform a variety of sonic duties from sibilant transient exclamation points to high frequency time keeping. In any case, the energy is mostly of a high-frequency content. Flat frequency response condenser microphones will give accurate reproduction of these sounds. Having microphones with low frequency roll-off will help to reject some of the sound of the rest of the kit which may otherwise cause phase problems when the drum channels are being mixed. The common approach to capturing the array of cymbals that a drummer may use is an overhead stereo pair of microphones. (positions A and B) A B G H F C I 4 Tom Toms – While the kick and snare establish the D low and high rhythmic functions, the toms are multiple drums that will be tuned from high to low between the snare and kick. They are primarily used for fills, but may also be consistent parts of the rhythmic structure. The attack range is similar to the snare drum, but often with more sustain. An individual directional mic on the top head near the edge can be used on each drum and panned to create some spatial imaging. A simpler setup is to place one mic slightly above and directly between two toms. (See position E in diagram to right; see image 11. ) B E Front view I I G A CH F E Image 11: Example of “simpler” mic set-up for tom toms Top view 19 Microphone Techniques for RECORDING When there are limited microphones available to record a drum kit use the following guidelines: Number of microphones Positioning Alternative (Positioning reference) One Two Three Four Five Use as “overhead” ( 5 ) Kick drum and overhead ( 1 and 5 ) Kick drum, snare, and overhead or kick drum ( 1 , 2 , and 5 ) Kick drum, snare, high hat, and overhead ( 1 , 2 , 3 , and 5 ) Kick drum, snare, high hat, tom-toms, and overhead ( 1 , 2 , 3 , 4 , and 5 ) Microphone Placement Tonal Balance Comments Natural Provides full sound with good attack. Natural Experiment with distance and angles if sound is too bright. Bright, with plenty of attack Allow clearance for movement of pan. Timbales, Congas, Bongos: One microphone aiming down between pair of drums, just above top heads Tambourine: One microphone placed 6 to 12 inches from instrument Steel Drums: Tenor Pan, Second Pan, Guitar Pan One microphone placed 4 inches above each pan Microphone placed underneath pan Cello Pan, Bass Pan One microphone placed 4 - 6 inches above each pan Decent if used for tenor or second pans. Too boomy with lower voiced pans. Natural Can double up pans to a single microphone. Natural Pan two microphones to left and right for stereo. See “Stereo Microphone Techniques” section. Bright, with lots of attack For less attack, use rubber mallets instead of metal mallets. Plastic mallets will give a medium attack. Xylophone, Marimba, Vibraphone: Two microphones aiming down toward instrument, about 1 1/2 feet above it, spaced 2 feet apart, or angled 1350 apart with grilles touching Glockenspiel: One microphone placed 4 - 6 inches above bars 20 Microphone Techniques for RECORDING Stereo Stereo Microphone Techniques – One of the most popular specialized microphone techniques is stereo miking. This use of two or more microphones to create a stereo image will often give depth and spatial placement to an instrument or overall recording. There are a number of different methods for stereo. Three of the most popular are the spaced pair (A/B), the coincident or near-coincident pair (X-Y configuration), and the Mid-Side (M-S) technique. The spaced pair sound (A/B) technique source uses two cardioid or omni directional microphones spaced 3 - 10 feet apart from each other panned in left/right configuration to capture the stereo image of an ensemble or instrument. Effective stereo A/B top view separation is very wide. The distance between the two microphones is dependent on the physical size of the sound source. For instance, if two mics are placed ten feet apart to record an acoustic guitar; the guitar will appear in the center of the stereo image. This is probably too much spacing for such a small sound source. A closer, narrower mic placement should be used in this situation. The drawback to A/B stereo is the potential for undesirable phase cancellation of the signals from the microphones. Due to the relatively large distance between the microphones and the resulting difference of sound arrival times at the microphones, phase cancellations and summing may be occurring. A mono reference source can be used to check for phase problems. When the program is switched to mono and frequencies jump out or fall out of the sound, you can assume that there is phase problem. This may be a serious problem if your recording is going to be heard in mono as is typical in broadcast or soundtrack playback. X-Y top view The X-Y technique uses two cardioid microphones of the same type and manufacture with the two mic capsules placed either as close as possible (coincident) or within 12 inches of each other (near-coincident) and facing each other at an angle ranging from 90 - 135 degrees, depending on the size of the sound source and the particular sound desired. The pair is placed with the center of the two mics facing directly at the sound source and panned left and right. Due to the small distance between the microphones, sound arrives at the mics at nearly the same time, reducing (near coincident) or eliminating (coincident) the possible phase problems of the A/B techniques. The stereo separation of this technique is good but may be limited if the sound source is extremely wide. Mono compatibility is fair (near-coincident) to excellent (coincident). The M-S or Mid-Side stereo technique involves a cardioid mic element and a bi-directional mic element, usually housed in a single case, mounted in a coincident arrangement. The cardioid (mid) faces directly at the source and picks up primarily on-axis sound while the bi-directional (side) faces left and right and picks up off-axis sound. The two signals are combined via the M-S matrix to give a variable controlled stereo image. By adjusting the level of mid versus side signals, a narrower or wider image can be created without moving the microphone. This technique is completely mono-compatible and is widely used in broadcast and film applications. 21 Microphone Techniques for RECORDING (see image 12) Stereo Microphone Techniques 22 Image 12: Example of “X-Y” stereo miking technique using Shure A27M stereo microphone adapter Microphone Techniques for RECORDING Introduction The world of studio recording is much different from that of live sound reinforcement, but the fundamental characteristics of the microphones and sound are the same. It is the ability to isolate individual instruments that gives a greater element of control and freedom for creativity in the studio. Since there are no live loudspeakers, feedback is not an issue. The natural sound of the instrument may be the desired effect, or the sound source can be manipulated into a sound never heard in the natural acoustic world. In order to achieve the desired result it is useful to understand some of the important characteristics of microphones, musical instruments, and acoustics. Section Two 23 Microphone Techniques for RECORDING SECTION TWO Microphone Characteristics There are three main considerations when choosing a microphone for recording applications: operating principle, frequency response, and directionality. Operating Principle – A microphone is an example of a transducer, a device which changes energy from one form into another, in this case from acoustic into electrical. The type of transducer is defined by the operating principle. In the current era of recording, the two primary operating principles used in microphone design are the dynamic and the condenser. Dynamic microphone elements are made up of a diaphragm, voice coil, and magnet which form a sound-driven electrical generator. Sound waves move the diaphragm/voice coil in a magnetic field to generate the electrical equivalent of the acoustic sound wave. The signal from the dynamic element can be used directly, without the need for additional circuitry. This design is extremely rugged, has good sensitivity and can handle the loudest possible sound pressure levels without distortion. The dynamic has some limitations at extreme high and low frequencies. To compensate, small resonant chambers are often used to extend the frequency range of dynamic microphones. Ribbon microphone elements, a variation of the dynamic microphone operating principle, consist of a thin piece of metal, typically corrugated aluminum, suspended between two magnetic pole pieces. As with moving-coil dynamics, no additional circuitry or powering is necessary for operation, however, the output of ribbon microphones tends to be quite low. Depending on the gain of the mixer or recording device to which the microphone is connected, additional pre-amplification may be necessary. Note that ribbon microphones are not as rugged as moving-coil dynamic microphones. The ribbon element itself is typically no more than a few microns thick, and can be deformed by a strong blast of air, or by blowing into the 24 microphone. Also, phantom power applied to the ribbon microphone could be harmful. Ribbon microphones are highly regarded in studio recording for their “warmth” and good low frequency response. Condenser microphone elements use a conductive diaphragm and an electrically charged backplate to form a sound-sensitive “condenser” (capacitor). Sound waves move the diaphragm in an electric field to create the electrical signal. In order to use this signal from the element, all condensers have active electronic circuitry, (often referred to as the “preamp”) either built into the microphone or in a separate pack. This means that condenser microphones require phantom power or a battery to operate. (For a detailed explanation of “phantom power”, see the sidebar.) However, the condenser design allows for smaller mic elements, higher sensitivity and is inherently capable of smooth response across a very wide frequency range. The main limitations of a condenser microphone relate to its electronics. These circuits can handle a specified maximum signal level from the condenser element, so a condenser mic has a maximum sound level before its output starts to be distorted. Some condensers have switchable pads or attenuators between the element and the electronics to allow them to handle higher sound levels. If you hear distortion when using a condenser microphone close to a very loud sound source, first make sure that the mixer input itself is not being overloaded. If not, switch in the attenuator in the mic (if equipped), move the mic farther away, or use a mic that can handle a higher level. In any case, the microphone will not be damaged by excess level. A second side effect of the condenser/electronics design is that it generates a certain amount of electrical noise (self-noise) which may be heard as “hiss” when recording very quiet sources at high gain settings. Higher quality condenser mics have very low self-noise, a desirable characteristic for this type of recording application. Most modern condenser microphones use solid state components for the internal circuitry, but older designs employed vacuum tubes (also known as “valves”) for this purpose. The subjective qualities imparted by vacuum Microphone Techniques for RECORDING tube electronics, often described as “warmth” or “smoothness,” have led to a resurgence in the popularity of vacuum tube-based condenser microphones. These sonic advantages come at the expense of higher self-noise and fragility. Vacuum tubes typically have a limited life span, and eventually need to be replaced. Most vacuum tube microphones require an external power supply, as standard 48V phantom power is not sufficient. Some power supplies offer the ability to switch polar patterns remotely on microphones that feature dual-diaphragms (see Directionality for a discussion of microphone polar patterns). Frequency response – The variation in output level or sensitivity of a microphone over its useable range from lowest to highest frequency. Virtually all microphone manufacturers will list the frequency response of their microphones as a range, for example 20 - 20,000Hz. This is usually illustrated with a graph that indicates relative amplitude at each frequency. The graph has the frequency in Hz on the x-axis and relative response in decibels on the y-axis. A microphone whose response is equal at all frequencies is said to have a “flat” frequency response. These microphones typically have a wide frequency range. Flat response microphones tend to be used to reproduce sound sources without coloring the original source. This is usually desired in reproducing instruments such as acoustic guitars or pianos. It is also common for stereo miking techniques and distant miking techniques. A microphone whose response has peaks or dips in certain frequency areas is said to have a “shaped” response. This response is designed to enhance a frequency range that is specific to a given sound source. For instance, a microphone may have a peak in the 2-10Khz range to enhance the intelligibility or presence of vocals. This shape is said to have a “presence peak”. A microphone’s response may also be reduced at other frequencies. One example of this is a low frequency roll-off to reduce unwanted “boominess”. Although dynamic microphones and condenser microphones may have similar published frequency response specifications their sound qualities can be quite different. A primary aspect of this difference is in their transient response. See the appendix for an explanation of this characteristic. Directionality is usually plotted on a graph referred to as a polar pattern. The polar pattern shows the variation in sensitivity 360 degrees around the microphone, assuming that the microphone is in the center and 0 degrees represents the front or on-axis direction of the microphone. There are a number of different directional patterns designed into microphones. The three basic patterns are omnidirectional, unidirectional, and bidirectional. The omnidirectional microphone has equal response at all angles. Its “coverage” or pickup angle is a full 360 degrees. This type of microphone can be used if more room ambience is desired. For example, when using an “omni”, the balance of direct and ambient sound depends on the distance of the microphone from the instrument, and can be adjusted to the desired effect. Phantom Power Phantom power is a DC voltage (usually 12-48 volts) used to power the electronics of a condenser microphone. For some (non-electret) condensers it may also be used to provide the polarizing voltage for the element itself. This voltage is supplied through the microphone cable by a mixer equipped with phantom power or by some type of in-line external source. The voltage is equal on Pin 2 and Pin 3 of a typical balanced, XLR-type connector. For a 48 volt phantom source, for example, Pin 2 is 48 VDC and Pin 3 is 48 VDC, both with respect to Pin 1 which is ground (shield). Because the voltage is exactly the same on Pin 2 and Pin 3, phantom power will have no effect on balanced dynamic microphones: no current will flow since there is no voltage difference across the output. In fact, phantom power supplies have current limiting which will prevent damage to a dynamic microphone even if it is shorted or miswired. In general, balanced dynamic microphones can be connected to phantom powered mixer inputs with no problem. Directionality – The sensitivity to sound relative to the direction or angle of arrival at the microphone. 25 Microphone Techniques for RECORDING 125 degrees for the supercardioid and 110 degrees for the hypercardioid. When placed properly they can provide more “focused” pickup and less room ambience than the cardioid pattern, but they have less rejection at the rear: -12 dB for the supercardioid and only -6 dB for the hypercardioid. Flat frequency response drawing The bidirectional microphone has full response at both 0 degrees (front) and at 180 degrees (back). It has its least response at the sides. The coverage or pickup angle is only about 90 degrees at the front (or the rear). It has the same amount of ambient pickup as the cardioid. This mic could be used for picking up two sound sources such as two vocalists facing each other. It is also used in certain stereo techniques. Shaped frequency response drawing The unidirectional microphone is most sensitive to sound arriving from one particular direction and is less sensitive at other directions. The most common type is a cardioid (heart-shaped) response. This has full sensitivity at 0 degrees (on-axis) and is least sensitive at 180 degrees (off-axis). Unidirectional microphones are used to isolate the desired on-axis sound from unwanted off-axis sound. In addition, the cardioid mic picks up only about one-third as much ambient sound as an omni. For example, the use of a cardioid microphone for a guitar amplifier, which is in the same room as the drum set, is one way to reduce the bleed-through of drums on to the recorded guitar track. The mic is aimed toward the amplifier and away from the drums. If the undesired sound source is extremely loud (as drums often are), other isolation techniques may be necessary. Omnidirectional Cardioid (unidirectional) Unidirectional microphones are available with several variations of the cardioid pattern. Two of these are the supercardioid and hypercardioid. Both patterns offer narrower front pickup angles than the cardioid (115 degrees for the supercardioid and 105 degrees for the hypercardioid) and also greater rejection of ambient sound. While the cardioid is least sensitive at the rear (180 degrees off-axis), the least sensitive direction is at 26 Supercardioid Microphone Techniques for RECORDING Understanding and choosing the frequency response and directionality of microphones are selective factors which can improve pickup of desired sound and reduce pickup of unwanted sound. This can greatly assist in achieving both natural sounding recordings and unique sounds for special applications. Instrument Characteristics Microphone polar patterns compared Other directional-related microphone characteristics: First, let’s present a bit of background information about how instruments radiate sound. The sound from a musical instrument has a frequency output which is the range of frequencies produced and their relative amplitudes. The fundamental frequencies establish the basic pitch, while the harmonic frequencies produce the timbre or characteristic tone of the instrument. Here are frequency ranges for some commonly known instruments: Ambient sound sensitivity – Since unidirectional microphones are less sensitive to off-axis sound than omnidirectional types, they pick up less overall ambient or room sound. Unidirectional mics should be used to control ambient noise pickup to get a “cleaner” recording. Distance factor – Since directional microphones have more rejection of off-axis sound than omnidirectional types, they may be used at greater distances from a sound source and still achieve the same balance between the direct sound and background or ambient sound. An omnidirectional microphone will pick up more room (ambient) sound than a unidirectional microphone at the same distance. An omni should be placed closer to the sound source than a “uni”– about half the distance – to pick up the same balance between direct sound and room sound. Off-axis coloration – A microphone’s frequency response may not be uniform at all angles. Typically, high frequencies are most affected, which may result in an unnatural sound for off-axis instruments or room ambience. Proximity effect – For most unidirectional types, bass response increases as the microphone is moved closer to the sound source. When miking close with unidirectional microphones (less than 1 foot), be aware of proximity effect: it may help to roll off the bass until you obtain a more natural sound. You can (1) roll off low frequencies at the mixer, (2) use a microphone designed to minimize proximity effect, (3) use a microphone with a bass roll-off switch, or (4) use an omnidirectional microphone (which does not exhibit proximity effect). Chart of instrument frequency ranges Also, an instrument radiates different frequencies at different levels in every direction, and each part of an instrument produces a different timbre. This is the directional output of an instrument. You can partly control the recorded tonal balance of an instrument by adjusting the microphone position relative to it. The fact that low frequencies tend to be omnidirectional while higher frequencies tend to be more directional is a basic audio principle to keep in mind. Most acoustic instruments are designed to sound best at a distance (say, two or more feet away). The sounds of the various parts of the instrument combine into a complete audio picture at some distance from the instrument. So, a microphone placed at that distance will pick up a “natural” or well-balanced tone quality. On the other hand, a microphone placed close to the instrument emphasizes the part of the instrument that the microphone is near. The sound picked up very close may or may not be the sound you wish to capture in the recording. 27 Microphone Techniques for RECORDING Acoustic Characteristics Since room acoustics have been mentioned repeatedly, here is a brief introduction to some basic factors involved in acoustics. Sound Waves – Sound waves consist of pressure variations traveling through the air. When the sound wave travels, it compresses air molecules together at one point. This is called the high pressure zone or positive component(+). After the compression, an expansion of molecules occurs. This is the low pressure zone or negative component(-). This process continues along the path of the sound wave until its energy becomes too weak to hear. If you could view the sound wave of a pure tone traveling through air, it would appear as a smooth, regular variation of pressure that could be drawn as a sine wave. The diagram shows the relationship of the air molecules and a sine wave. compression rarefaction wave motion Frequency, Wavelength, and the Speed of Sound – The frequency of a sound wave indicates the rate of pressure variations or cycles. One cycle is a change from high pressure one cycle or one period to low pressure and back to high rms pressure. The peak number of cycles per second is called Hertz, peak-to-peak abbreviated “Hz.” So, a 1,000Hz tone has 1,000 Wave amplitude cycles per second. The wavelength of a sound is the physical distance from the start of one cycle to the start of the next cycle. Wavelength is related to frequency by the speed of sound. The speed of sound in air is 1130 feet per second or 344 meters/second. The speed of sound is constant no matter what the frequency. You can determine the wavelength of a sound wave of any frequency if you understand these relationships: 28 for a 500Hz sound wave: 1,130 feet per second wavelength = 500Hz wavelength = 2.26 feet Approximate wavelengths of common frequencies: 100 Hz: about 10 feet 1000 Hz: about 1 foot 10,000 Hz: about 1 inch compression ............................ ............................ ............................ ............................ ............................ ............................ ............................ ............................ ............................ ............................ ............................ ............................ ............................ ............................ ............................ .. .. .. .. .. .. .. .. .. .. .. .. .. .. .. .. .. .. .. .. .. .. .. .. .. .. .. ............................ ............................ ............................ ... ... ... ... ... ... ... ... ... ... ... ... ... ... ... ... ... ... ... ... ... ... ... ... ... ... ... ............................ ............................ ............................ ............................ ............................ ............................ ............................ ............................ ............................ ............................ ............................ ............................ ............................ ............................ ............................ ............................ ............................ ............................ ............................ ............................ ............................ .. .. .. .. .. .. .. .. .. .. .. .. .. .. .. .. .. .. .. .. .. .. .. .. .. .. .. ............................ ............................ ............................ ... ... ... ... ... ... ... ... ... ... ... ... ... ... ... ... ... ... ... ... ... ... ... ... ... ... ... ............................ ............................ ............................ ............................ ............................ ............................ ............................ rarefaction The Wave Equation: c = f • l speed of sound = frequency • wavelength or speed of sound wavelength = frequency Loudness – The fluctuation 140 of air pressure 130 created by sound 120 is a change above 110 100 and below normal 90 atmospheric 80 pressure. This is 70 60 what the human 50 ear responds to. 40 The varying amount 30 20 of pressure of the 10 air molecules 0 compressing and expanding is related Ambient sounds to the apparent loudness at the human ear. The greater the pressure change, the louder the sound. Under ideal conditions the human ear can sense a pressure change as small as .0002 microbar. One microbar is equal to one millionth of atmospheric pressure. The threshold of pain is about 200 microbar. Obviously, the human ear responds to a wide range of amplitude of sound. This amplitude range is more commonly referred to in decibels. Sound Pressure Level (dB SPL), relative to .0002 microbar (0dB SPL). 0 dB SPL is the threshold of hearing and 120 dB SPL is the threshold of pain. 1 dB is about the smallest change in SPL that can be heard. A 3 dB change is generally noticeable, while a 6 dB change is very noticeable. A 10 dB SPL increase is perceived to be twice as loud! Microphone Techniques for RECORDING Sound Transmission – It is important to remember that sound transmission does not normally happen in a completely controlled environment. In a recording studio, though, it is possible to separate or isolate the sounds being recorded. The best way to do this is to put the different sound sources in different rooms. This provides almost complete isolation and control of the sound from the voice or instrument. Unfortunately, multiple rooms are not always an option in studios, and even one sound source in a room by itself is subject to the effects of the walls, floor, ceiling and various isolation barriers. All of these effects can alter the sound before it actually arrives at the microphone. 3. Diffraction – A sound wave will typically bend around obstacles in its path which are smaller than its wavelength. Because a low frequency sound wave is much longer than a high frequency wave, low frequencies will bend around objects that high frequencies cannot. The effect is that high frequencies are more easily blocked or absorbed while low frequencies are essentially omnidirectional. When isolating two instruments in one room with a gobo as an acoustic barrier, it is possible to notice the individual instruments are “muddy” in the low end response. This may be due to diffraction of low frequencies around the acoustic barrier. Applications Tip: In the study of acoustics there are three basic ways in which sound is altered by its environment: 1. Reflection – A sound wave can be reflected by a surface or other object if the object is physically as large or larger than the wavelength of the sound. Because low-frequency sounds have long wavelengths, they can only be reflected by large objects. Higher frequencies can be reflected by smaller objects and surfaces. The reflected sound will have a different frequency characteristic than the direct sound if all sounds are not reflected equally. Reflection is also the source of echo, reverb, and standing waves: Echo occurs when an indirect sound is delayed long enough (by a distant reflective surface) to be heard by the listener as a distinct repetition of the direct sound. Reverberation consists of many reflections of a sound, maintaining the sound in a room for a time even after the direct sound has stopped. Standing waves in a room occur for certain frequencies related to the distance between parallel walls. The original sound and the reflected sound will begin to reinforce each other when the wavelength is equal to the distance between two walls. Typically, this happens at low frequencies due to their longer wavelengths and the difficulty of absorbing them. 2. Refraction – The bending of a sound wave as it passes through some change in the density of the transmission environment. This change may be due to physical objects, such as blankets hung for isolation or thin gobos, or it may be due to atmospheric effects such as wind or temperature gradients. These effects are not noticeable in a studio environment. Absorption (beware of carpets!) When building a project studio or small commercial studio, it is usually necessary to do some sound treatment to the walls and possibly build some isolating gobos for recording purposes. Many small studios assume they can save money and achieve the desired absorption effect by using inexpensive carpet. This is a bad assumption. Absorption is the changing of sound energy into heat as it tries to pass through some material. Different materials have different absorption effects at multiple frequencies. Each material is measured with an absorption coefficient ranging between 0-1 (sabins). This can be thought of as the percentage of sound that will be absorbed. For instance: a material may have an absorption coefficient of .67 at 1,000 Hz. This would mean the material absorbs 67% of the 1,000 Hz frequencies applied to it. Here is a chart showing the advantages of acoustic foam over bare walls or carpeting. 29 Microphone Techniques for RECORDING +1 Direct vs. Ambient Sound – A very important property of direct sound is that it becomes weaker as it travels away from the sound source, at a rate controlled by the inverse-square law. When the distance from a sound source doubles, the sound level decreases by 6dB. This is a noticeable audible decrease. For example, if the sound from a guitar amplifier is 100 dB SPL at 1 ft. from the cabinet it will be 94 dB at 2 ft., 88 dB at 4 ft., 82 dB at 8 ft., etc. When the distance is cut in half the sound level increases by 6dB: It will be 106 dB at 6 inches and 112 dB at 3 inches. On the other hand, the ambient sound in a room is at nearly the same level throughout the room. This is because the ambient sound has been reflected many times within the room until it is essentially non-directional. Reverberation is an example of non-directional sound. This is why the ambient sound of the room will become increasingly apparent as a microphone is placed further away from the direct sound source. The amount of direct sound relative to ambient sound can be controlled by the distance of the microphone to the sound source and to a lesser degree by the polar pattern of the mic. However, if the microphone is placed beyond a certain distance from the sound source, the ambient sound will begin to dominate the recording and the desired balance may not be possible to achieve, no matter what type of mic is used. This is called the “critical distance” and becomes shorter as the ambient noise and reverberation increase, forcing closer placement of the microphone to the source. ▲ 0 30 -1 “in-phase” +1 + = 0 0 a -1 -2 +1 0 ”1800 out of phase” -1 +1 + = 0 0 b -1 +1 +2 0 “phase shifts” -1 +1 +1 + = 0 c -1 0 -1 -2 Phase relationships Two identical sound waves starting at the same point in time are called “in-phase” and will sum together creating a single wave with double the amplitude but otherwise identical to the original waves. Two identical sound waves with one wave’s starting point occurring at the 180degree point of the other wave are said to be “out of phase”, and the two waves will cancel each other completely. When two sound waves of the same single frequency but different starting points are combined, the resulting wave as said to have “phase shift” or an apparent starting point somewhere between the original starting points. This new wave will have the same frequency as the original waves but will have increased or decreased amplitude depending on the degree of phase difference. Phase shift, in this case, indicates that the 0 degree points of two identical waves are not the same. ▲ Phase relationships and interference effects – The phase of a single frequency sound wave is always described relative to the starting point of the wave or 0 degrees. The pressure change is also one cycle or one period zero at this point. The peak of the high pressure zone is at 90 degrees, and the pressure change falls to zero again at 180 degrees. The peak of the low pressure zone is at 270 0 90 180 270 360 degrees, and the pressure change rises to zero at 360 Sound pressure wave degrees for the start of the next cycle. +2 0 0 0 0 0 Most soundwaves are not a single frequency but are made up of many frequencies. When identical multiplefrequency soundwaves combine, there are three possibilities for the resulting wave: a doubling of amplitude at all frequencies if the waves are “in phase”, a complete cancellation at all frequencies if the waves are 180 degrees “out of phase”, or partial cancellation and partial reinforcement at various frequencies if the waves have intermediate phase relationship. Microphone Techniques for RECORDING The last case is the most likely, and the audible result is a degraded frequency response called “comb filtering.” The pattern of peaks and dips resembles the teeth of a comb and the depth and location of these notches depend on the degree of phase shift. With microphones this effect can occur in two ways. The first is when two (or more) mics pick up the same sound source at different distances. Because it takes longer for the sound to arrive at the more distant microphone, there is effectively a phase difference between the signals from the mics when they are combined (electrically) in the mixer. The resulting comb filtering depends on the sound arrival time difference between the microphones: a large time difference (long distance) causes comb filtering to begin at low frequencies, while a small time difference (short distance) moves the comb filtering to higher frequencies. The second way for this effect to occur is when a single microphone picks up a direct sound and also a delayed version of the same sound. The delay may be due to an acoustic reflection of the original sound or Multi-mic comb filtering to multiple sources of the original sound. A guitar cabinet with more than one speaker or multiple cabinets for the same instrument would be an example. The delayed sound travels a longer distance (longer time) to the mic and thus has a phase difference relative to the direct sound. When these sounds combine (acoustically) at the microphone, comb filtering results. This time the effect of the comb filtering depends on the distance between the microphone and the source of the reflection or the distance between the Reflection comb filtering multiple sources. The goal here is to create an awareness of the sources of these potential influences on recorded sound and to provide insight into controlling them. When an effect of this sort is heard, and is undesirable, it is usually possible to move the sound source, use a microphone with a different directional characteristic, or physically isolate the sound source further to improve the situation. Applications Tip: Microphone phase One of the strangest effects that can happen in the recording process is apparent when two microphones are placed in close proximity to the same sound source. Many times this is due to the phase relationship of the sounds arriving at the microphones. If two microphones are picking up the same sound source from different locations, some phase cancellation or summing may be occurring. Phase cancellation happens when two microphones are receiving the same soundwave but with opposite pressure zones (that is, more than 180 degrees out of phase). This is usually not desired. A mic with a different polar pattern may reduce the pickup of unwanted sound and reduce the effect, or physical isolation can be used. With a drum kit, physical isolation of the individual drums is not possible. In this situation your choice of microphones may be more dependent on the off-axis rejection of the mic. Another possibility is phase reversal. If there is cancellation occurring, a 180 degree phase flip will create phase summing of the same frequencies. A common approach to the snare drum is to place one mic on the top head and one on the bottom head. Because the mics are picking up relatively similar sound sources at different points in the sound wave, you are probably experiencing some phase cancellations. Inverting the phase of one mic will sum any frequencies being canceled. This may sometimes achieve a “fatter” snare drum sound. This effect will change dependent on mic locations. The phase inversion can be done with an in-line phase reverse adapter or by a phase invert switch found on many mixer inputs. 31 Microphone Techniques for RECORDING Selection Guide Shure Microphone Selection Guide Vocal Instrument Solo Vocal KSM44 SM27 SM7B SM58 PG42 Guitar Amplifier KSM32 BETA 56A/57A SM57 Ensemble/Choir KSM32 KSM141 KSM137 Podcasting/ Voice-Over PG42 SM27 SM7B SM58 55SH Series II Acoustic Guitar KSM32 KSM141 KSM137 SM57 Bass Amplifier BETA 52A SM7B SM57 Acoustic Bass KSM32 KSM44 KSM137 SM137 Piano KSM44 KSM32 KSM137 BETA91 (under lid) VP88 Orchestra/Ensemble KSM141 KSM137 KSM44 KSM32 SM137 Strings KSM32 KSM137 SM137 MC50B Snare Drum (top) BETA 57A SM57 BETA 98D/S Snare Drum (bottom) KSM137 SM137 Rack/Floor Toms BETA 56A/57A SM57 BETA 98D/S 32 Harmonica 520DX BETA 58A SM58 Woodwinds KSM32 SM27 KSM137 BETA 98H/C Brass/Saxophone KSM32 BETA 56A BETA 98H/C Drums Kick Drum BETA 52A BETA91 SM57 Leslie Cabinet Top: SM57 Top: KSM32 Bottom: BETA 52A Stereo Recording Overheads KSM32 SM27 KSM137 SM137 Congas BETA 56A/57A SM57 BETA 98D/S Mallets KSM32 SM27 KSM137 SM137 Auxiliary Percussion KSM32 KSM137 SM137 SM57 X-Y KSM137 SM137 KSM32 M-S VP88 KSM44 (pair) Spaced Pair KSM44 KSM141 KSM137 Microphone Techniques for RECORDING Selection Guide Shure Recording Microphone Lockers: If you are just getting started, and need a basic selection of microphones to get your studio up and running, select the studio situation below that most closely resembles the type of recording you will be doing. Home Studio Basic (overdubs, vocals, acoustic guitar): PG27 2 – SM57 1 – PG27 (multi purpose) 1 – PG42 (vocals) Home Studio Advanced (tracking, overdubs, drums, guitars, vocals): 1 – Beta 52A* 3 – SM57* 2 – SM137 1 – SM27 PG42 Project Studio Commercial (tracking, overdubs, professional voice-overs, larger ensembles, drums, piano): 1 – Beta 52A 4 – SM57 2 – KSM137 2 – KSM32 1 – KSM44 1 – SM7B KSM137 *Available as model number DMK57-52, which includes all four mics, plus three A56D drum mounts. SM137 SM57 SM7B Beta 52A SM27 KSM32 KSM44 33 Microphone Techniques for RECORDING 3-to-1 Rule - When using multiple microphones, the distance between microphones should be at least 3 times the distance from each microphone to its intended sound source. Absorption - The dissipation of sound energy by losses due to sound absorbent materials. Active Circuitry - Electrical circuitry which requires power to operate, such as transistors and vacuum tubes. Ambience - Room acoustics or natural reverberation. Amplitude - The strength or level of sound pressure or voltage. Audio Chain - The series of interconnected audio equipment used for recording or PA. Backplate - The solid conductive disk that forms the fixed half of a condenser element. Balanced - A circuit that carries information by means of two equal but opposite polarity signals, on two conductors. Bidirectional Microphone - A microphone that picks up equally from two opposite directions. The angle of best rejection is 90 degrees from the front (or rear) of the microphone, that is, directly at the sides. Boundary/Surface Microphone - A microphone designed to be mounted on an acoustically reflective surface. Cardioid Microphone - A unidirectional microphone with moderately wide front pickup (131 degrees). Angle of best rejection is 180 degrees from the front of the microphone, that is, directly at the rear. Cartridge (Transducer) - The element in a microphone that converts acoustical energy (sound) into electrical energy (the signal). Clipping Level - The maximum electrical output signal level (dBV or dBu) that the microphone can produce before the output becomes distorted. 34 Glossary Current - Charge flowing in an electrical circuit. Analogous to the amount of a fluid flowing in a pipe. Decibel (dB) - A number used to express relative output sensitivity. It is a logarithmic ratio. Diaphragm - The thin membrane in a microphone which moves in response to sound waves. Diffraction - The bending of sound waves around an object which is physically smaller than the wavelength of the sound. Direct Sound - Sound which travels by a straight path from a sound source to a microphone or listener. Distance Factor - The equivalent operating distance of a directional microphone compared to an omnidirectional microphone to achieve the same ratio of direct to reverberant sound. Distant Pickup - Microphone placement farther than 2 feet from the sound source. Dynamic Microphone - A microphone that generates an electrical signal when sound waves cause a conductor to vibrate in a magnetic field. In a moving-coil microphone, the conductor is a coil of wire attached to the diaphragm. In a ribbon microphone, the diaphragm is the conductor. Dynamic Range - The range of amplitude of a sound source. Also, the range of sound level that a microphone can successfully pick up. Echo - Reflection of sound that is delayed long enough (more than about 50 msec.) to be heard as a distinct repetition of the original sound. Electret - A material (such as Teflon) that can retain a permanent electric charge. EQ - Equalization or tone control to shape frequency response in some desired way. Close Pickup - Microphone placement within 2 feet of a sound source. Feedback - In a PA system consisting of a microphone, amplifier, and loudspeaker, feedback is the ringing or howling sound caused by amplified sound from the loudspeaker entering the microphone and being re-amplified. Comb Filtering - An interference effect in which the frequency response exhibits regular deep notches. Flat Response - A frequency response that is uniform and equal at all frequencies. Condenser Microphone - A microphone that generates an electrical signal when sound waves vary the spacing between two charged surfaces: the diaphragm and the backplate. Frequency - The rate of repetition of a cyclic phenomenon such as a sound wave. Critical Distance - In acoustics, the distance from a sound source in a room at which the direct sound level is equal to the reverberant sound level. Frequency Response Tailoring Switch - A switch on a microphone that affects the tone quality reproduced by the microphone by means of an equalization circuit. (Similar to a bass or treble control on a hi-fi receiver.) Glossary Microphone Techniques for RECORDING Frequency Response - A graph showing how a microphone responds to various sound frequencies. It is a plot of electrical output (in decibels) vs. frequency (in Hertz). NAG - Needed Acoustic Gain is the amount of gain that a sound system must provide for a distant listener to hear as if he or she was close to the unamplified sound source. Fundamental - The lowest frequency component of a complex waveform such as musical note. It establishes the basic pitch of the note. Noise - Unwanted electrical or acoustic interference. Noise Cancelling - A microphone that rejects ambient or distant sound. Gain - Amplification of sound level or voltage. Gain-Before-Feedback - The amount of gain that can be achieved in a sound system before feedback or ringing occurs. NOM - Number of open microphones in a sound system. Decreases gain-before-feedback by 3dB everytime NOM doubles. Gobos - Movable panels used to reduce reflected sound in the recording environment. Omnidirectional Microphone - A microphone that picks up sound equally well from all directions. Harmonic - Frequency components above the fundamental of a complex waveform. They are generally multiples of the fundamental which establish the timbre or tone of the note. Output Noise (Self-Noise) - The amount of residual noise (dB SPL) generated by the electronics of a condenser microphone. Hypercardioid - A unidirectional microphone with tighter front pickup (105 degrees) than a supercardioid, but with more rear pickup. Angle of best rejection is about 110 degrees from the front of the microphone. Overload - Exceeding the signal level capability of a microphone or electrical circuit. PAG - Potential Acoustic Gain is the calculated gain that a sound system can achieve at or just below the point of feedback. Impedance - In an electrical circuit, opposition to the flow of alternating current, measured in ohms. A high-impedance microphone has an impedance of 10,000 ohms or more. A lowimpedance microphone has an impedance of 50 to 600 ohms. Phantom Power - A method of providing power to the electronics of a condenser microphone through the microphone cable. Interference - Destructive combining of sound waves or electrical signals due to phase differences. Pickup Angle/Coverage Angle - The effective arc of coverage of a microphone, usually taken to be within the 3dB down points in its directional response. Phase - The “time” relationship between cycles of different waves. Inverse Square Law - States that direct sound levels increase (or decrease) by an amount proportional to the square of the change in distance. Pitch - The fundamental or basic frequency of a musical note. Isolation - Freedom from leakage; the ability to reject unwanted sounds. Polar Pattern (Directional Pattern, Polar Response) - A graph showing how the sensitivity of a microphone varies with the angle of the sound source, at a particular frequency. Examples of polar patterns are unidirectional and omnidirectional. Leakage - Pickup of an instrument by a microphone intended to pick up another instrument. Creative leakage is artistically favorable leakage that adds a “loose” or “live” feel to a recording. Polarization - The charge or voltage on a condenser microphone element. Maximum Sound Pressure Level - The maximum acoustic input signal level (dB SPL) that the microphone can accept before clipping occurs. Pop Filter - An acoustically transparent shield around a microphone cartridge that reduces popping sounds. Often a ball-shaped grille, foam cover or fabric barrier. Microphone Sensitivity - A rating given in dBV to express how “hot” the microphone is by exposing the microphone to a specified sound field level (typically either 94 dB SPL or 74 dB SPL). This specification can be confusing because manufacturers designate the sound level different ways. Here is an easy reference guide: 94 dB SPL = 1 Pascal = 10 microbars. To compare a microphone that has been measured at 74 dB SPL with one that has been measured at 94 dB SPL, simply add 20 to the dBV rating. Pop - A thump of explosive breath sound produced when a puff of air from the mouth strikes the microphone diaphragm. Occurs most often with “p”, “t”, and “b” sounds. Presence Peak - An increase in microphone output in the “presence” frequency range of 2,000 Hz to 10,000 Hz. A presence peak increases clarity, articulation, apparent closeness, and “punch.” 35 Microphone Techniques for RECORDING Proximity Effect - The increase in bass occurring with most unidirectional microphones when they are placed close to an instrument or vocalist (within 1 foot). Does not occur with omnidirectional microphones. Supercardioid Microphone - A unidirectional microphone with tighter front pickup angle (115 degrees) than a cardioid, but with some rear pickup. Angle of best rejection is 126 degrees from the front of the microphone, that is, 54 degrees from the rear. Rear Lobe - A region of pickup at the rear of a supercardioid or hypercardioid microphone polar pattern. A bidirectional microphone has a rear lobe equal to its front pickup. 3-to-1 Rule - (See top of page 34.) Reflection - The bouncing of sound waves back from an object or surface which is physically larger than the wavelength of the sound. Timbre - The characteristic tone of a voice or instrument; a function of harmonics. Transducer - A device that converts one form of energy to another. A microphone transducer (cartridge) converts acoustical energy (sound) into electrical energy (the audio signal). Refraction - The bending of sound waves by a change in the density of the transmission medium, such as temperature gradients in air due to wind. Transient Response - The ability of a device to respond to a rapidly changing input. Resistance - The opposition to the flow of current in an electrical circuit. It is analogous to the friction of fluid flowing in a pipe. Unbalanced - A circuit that carries information by means of one signal on a single conductor. Reverberation - The reflection of a sound a sufficient number of times that it becomes non-directional and persists for some time after the source has stopped. The amount of reverberation depends on the relative amount of sound reflection and absorption in the room. Unidirectional Microphone - A microphone that is most sensitive to sound coming from a single direction-in front of the microphone. Cardioid, supercardioid, and hypercardioid microphones are examples of unidirectional microphones. Rolloff - A gradual decrease in response below or above some specified frequency. Sensitivity - The electrical output that a microphone produces for a given sound pressure level. Shaped Response - A frequency response that exhibits significant variation from flat within its range. It is usually designed to enhance the sound for a particular application. Signal to Noise Ratio - The amount of signal (dBV) above the noise floor when a specified sound pressure level is applied to the microphone (usually 94 dB SPL). Sound Chain - The series of interconnected audio equipment used for recording or PA. Sound Reinforcement - Amplification of live sound sources. Speed of Sound - The speed of sound waves, about 1130 feet per second in air. SPL - Sound Pressure Level is the loudness of sound relative to a reference level of 0.0002 microbars. Standing Wave - A stationary sound wave that is reinforced by reflection between two parallel surfaces that are spaced a wavelength apart. 36 Glossary Vacuum Tube (valve) - An electric device generally used to amplify a signal by controlling the movement of electrons in a vacuum. Vacuum tubes were widely used in the early part of the 20th century, but have largely been replaced by transistors. Voice Coil - Small coil of wire attached to the diaphragm of a dynamic microphone. Voltage - The potential difference in an electric circuit. Analogous to the pressure on fluid flowing in a pipe. Wavelength - The physical distance between the start and end of one cycle of a soundwave. Appendix A Microphone Techniques for RECORDING Appendix A: The Decibel The decibel (dB) is an expression often used in electrical and acoustic measurements. The decibel is a number that represents a ratio of two values of a quantity such as voltage. It is actually a logarithmic ratio whose main purpose is to scale a large measurement range down to a much smaller and more useable range. The form of the decibel relationship for voltage is: Since the decibel is a ratio of two values, there must be an explicit or implicit reference value for any measurement given in dB. This is usually indicated by a suffix on the dB. Some devices are measured in dBV (reference to 1 Volt = 0 dBV), while others may be specified in dBu or dBm (reference to .775V = 0dBu/dBm). Here is a chart that makes conversion for comparison easy: dB = 20 x log(V1/V2) where 20 is a constant, V1 is one voltage, V2 is a reference voltage, and log is logarithm base 10. Examples: What is the relationship in decibels between 100 volts and 1 volt? (dbV) dB = 20 x log(100/1) dB = 20 x log(100) dB = 20 x 2 (the log of 100 is 2) dB = 40 That is, 100 volts is 40dB greater than 1 volt. What is the relationship in decibels between .0001 volt and 1 volt? (dbV) dB = 20 x log(.001/1) dB = 20 x log(.001) dB = 20 x (-3) (the log of .001 is -3) dB = -60 That is, .001 volt is 60dB less than 1 volt. Audio equipment signal levels are generally broken into 3 main categories: Mic, Line, or Speaker Level. Aux level resides within the lower half of line level. The chart also shows at what voltages these categories exist. Similarly: If one voltage is equal to the other, they are 0dB different. If one voltage is twice the other, they are 6dB different. If one voltage is ten times the other, they are 20dB different. One reason that the decibel is so useful in certain audio measurements is that this scaling function closely approximates the behavior of human hearing sensitivity. For example, a change of 1dB SPL is about the smallest difference in loudness that can be perceived while a 3dB SPL change is generally noticeable. A 6dB SPL change is quite noticeable and finally, a 10dB SPL change is perceived as “twice as loud.” 37 Microphone Techniques for RECORDING Appendix B Appendix B: Transient Response The ability of a microphone to respond to a rapidly changing sound wave. A good way to understand why dynamic and condenser mics sound different is to understand the differences in their transient response. In order for a microphone to convert sound energy into electrical energy, the sound wave must physically move the diaphragm of the microphone. The speed of this movement depends on the weight or mass of the diaphragm. For instance, the diaphragm and voice coil assembly of a dynamic microphone may have up to 1000 times the mass of the diaphragm of a condenser microphone. The lightweight condenser diaphragm starts moving much more quickly than the dynamic’s diaphragm. It also takes longer for the dynamic’s diaphragm to stop moving in comparison to the condenser’s diaphragm. Thus, the dynamic’s transient response is not as good as the condenser’s transient response. This is similar to two vehicles in traffic: a truck and a sports car. They may have engines of equal power, but the truck weighs much more than the car. As traffic flow changes, the sports car can accelerate and brake very quickly, while the semi accelerates and brakes very slowly due to its greater weight. Both vehicles follow the overall traffic flow but the sports car responds better to sudden changes. The picture below is of two studio microphones responding to the sound impulse produced by an electric spark: condenser mic on top, dynamic mic on bottom. It is evident that it takes almost twice as long for the dynamic microphone to respond to the sound. It also takes longer for the dynamic to stop moving after the impulse has passed (notice the ripple on the second half of the graph). Since condenser microphones generally have better transient response then dynamics, they are better suited for instruments that have very sharp attacks or extended high frequency output such as cymbals. It is this transient response difference that causes condenser mics to have a more crisp, detailed sound and dynamic mics to have a more mellow, rounded sound. Condenser/dynamic scope photo 38 Microphone Techniques for RECORDING About the Authors John Boudreau Tim Vear John has had extensive experience as a musician, a recording engineer, and a composer. His desire to better combine the artistic and technical aspects of music led him to a career in the audio field. Having received a BS degree in Music Business from Elmhurst College, John performed and composed for both a Jazz and a Rock band prior to joining Shure in 1994 as an associate in the Applications Engineering group. While at Shure, John led many audio product training seminars and clinics, with an eye to helping musicians and others affiliated with the field use technology to better fulfill their artistic interpretations. No longer a Shure associate, John continues to pursue his interests as a live and recorded sound engineer for local bands and venues, as well as writing and recording for his own band. Tim is a native of Chicago who has come to the audio field as a way of combining a lifelong interest in both entertainment and science. He has worked as an engineer in live sound, recording and broadcast, has operated his own recording studio and sound company, and has played music professionally since high school. In his tenure at Shure, Tim has served in a technical support role for the sales and marketing departments, providing product and applications training for Shure customers, dealers, installers, and company staff. He has presented seminars for a variety of domestic and international audiences, including the National Systems contractors Association, the Audio Engineering Society and the Society of Broadcast Engineers. Tim has authored several publications for Shure and his articles have appeared in several trade publications. Rick Frank Over his career, Rick has been involved in a wide variety of music and recording activities including composing, teaching, performing, and producing popular music, jazz and commercial jingles. He has spent his life in Illinois where he received his BS in English and his MBA from the University of Illinois, Urbana-Champaign. While in downstate Illinois he also operated a successful retail musical instrument business and teaching program that coincided with working as a professional guitarist and electric bassist. Rick was Shure’s Marketing Director for Wired Microphones, responsible for Music Industry products. No longer a Shure associate, he continues to perform music professionally. Gino Sigismondi Gino, a Chicago native and Shure Applications Specialist since 1997, has been active in the music and audio industry for nearly ten years. In addition to his work as a live sound and recording engineer, Gino’s experience also includes performing and composing. Gino earned his BS degree in Music Business from Elmhurst College, where he was a member of the Jazz Band, as both guitar player and sound technician. As a member of Applications Engineering, Gino brings his years of practical experience to the product training seminars he conducts for Shure customers, dealers, distribution centers, and internal staff. Gino continues to remain active as a sound engineer, expanding his horizons beyond live music to include sound design for modern dance and church sound. Rick Waller An interest in the technical and musical aspects of audio has led Rick to pursue a career as both engineer and musician. He received a BS degree in Electrical Engineering from the University of Illinois at Urbana/Champaign, where he specialized in acoustics, audio synthesis and radio frequency theory. Rick is an avid keyboardist, drummer and home theater hobbyist and has also worked as a sound engineer and disc jockey. Currently he is an associate in the Applications Engineering Group at Shure. In this capacity Rick provides technical support to customers, writing and conducting seminars on wired and wireless microphones, mixers and other audio topics. Additional Shure Publications Available: Printed and electronic versions of the following guides are available free of charge. To obtain your complimentary copies, call one of the phone numbers listed below or visit www.shure.com/literature. • Selection and Operation of Personal Monitor Systems • Selection and Operation of Wireless Microphone Systems • Microphone Techniques for Live Sound Reinforcement Other Sources of Information: There are books written about acoustics and how to mathematically determine their effects. Here are a few: • FUNDAMENTALS OF MUSICAL ACOUSTICS by Arthur H. Benade • ACOUSTICS SOURCE BOOK by Sybil P. Parker • MODERN RECORDING TECHNIQUES by Huber & Runstein • THE MASTER HANDBOOK OF ACOUSTICS by F. Alton Everest Our Dedication to Quality Products Shure offers a complete line of microphones and wireless microphone systems for everyone from first-time users to professionals in the music industry–for nearly every possible application. For over eight decades, the Shure name has been synonymous with quality audio. All Shure products are designed to provide consistent, high-quality performance under the most extreme real-life operating conditions. www.shure.com United States: Shure Incorporated 5800 West Touhy Avenue Niles, IL 60714-4608 USA Europe, Middle East, Africa: Shure Europe GmbH Wannenäckerstr. 28, 74078 Heilbronn, Germany Phone: 847-600-2000 Fax: 847-600-1212 Email: [email protected] Phone: 49-7131-72140 Fax: 49-7131-721414 Email: [email protected] ©2009 ©2007Shure ShureIncorporated Incorporated AL1265J AL0000 10M 4/09 Asia, Pacific: Shure Asia Limited 3/F, Citicorp Centre 18 Whitfield Road Causeway Bay, Hong Kong Canada, Latin America, Caribbean: Shure Incorporated 5800 West Touhy Avenue Niles, IL 60714-4608 USA Phone: 852-2893-4290 Fax: 852-2893-4055 Email: [email protected] Phone: 847-600-2000 Fax: 847-600-6446 Email: [email protected]