Music 65 • Prof. Lehrman's notes
Micing—what you need to consider:
• What's the instrument?
• What's the performance style?
• Is the room sound good? Is it quiet?
• Are there other instruments playing at the same time?
• How much room sound do you want?
• What mics do you have?
• Do you want stereo or mono? How much stereo?
Good positioning is always better than trying to eq later. Good positioning means phasing is favorable: hard to fix with eq!
Mics need to be closer than our ears, since we don't have the visual cues to tell us what to look for, and mics can't distinguish between direct and reflected sound—we always want more direct sound in the recording. Can add reflections (echo/reverb) later, but (virtually) impossible to remove them!
Listening to the instruments in the space: finding the right spot to record. Get the room balance in your ear, then take two steps forward and put the mic there.
3-to-1 rule: when using multiple microphones, mics need to be at least three times as far away from each other as they are from their individual sources.
Winds & Strings: at least 3 ft away from source, if possible, except when it would violate 3-to-1 rule! String sections: mic in stereo as an ensemble, not meant to be a bunch of soloists. Horn sections, can go either way: mic individually or if there is enough isolation from other instruments, as section.
Guitar: exception since we are used to hearing close-miked guitars. But there is no one good spot on the guitar, since sound comes from all over the instrument: soundhole (too boomy by itself), body, top, neck, headstock. Best to use 2 mics, or if room is quiet, from a distance.
Piano: exception since pianists like the sound of the instrument close up--doesn’t really need the room to expand. Different philosophies for pop and classical. 3:1 rule on soundboard, or even better, 5:1 since reflections are very loud and phase relationships very complex. Can use spaced cardioids, spaced omnis, or coincident cardioids, in which case you want to reposition them for the best balance within the instrument (bass/treble).
Drums: first of all, make them sound good! Tune them, dampen rattles, dampen heads so they don’t ring as much (blanket in kick drum).
Three philosophies--Choice will depend on spill, room sound, and how much power and immediacy you want in the drums.
1) stereo pair overhead (cardioid or omni); good for jazz, if you don’t mind some spill, or if they’re in a good-sounding isolation room.
2) add kick (dynamic or high-level condensor) and snare mics for extra punch and flexibility
3) add mics to everything. Complicates things because of spill, may have to add noise gates later.
Glyn Johns technique—three cardioids in an isoceles triangle.
Mic techniques for stereo: XY (coincident and spaced: ORTF, DIN, etc.), MS, spaced omni, Decca tree (uses three omnis).
Microphone techniques: respect the historical use of instruments!
for Vocals: pop filters, monitors
for Piano: stereo image?
for strings: not on top of the bridge. Too close, loses resonance and high frequencies (data from Michigan Tech, using DPA 4011 cardioid).
Cables: Balanced vs. Unbalanced:
Balanced = two conductor and surrounding shield or ground. Two conductors are in electrical opposition to each other — when one has positive voltage the other has negative. At receiving end, one leg is flipped in polarity—also called phase—and the two are added. If noise is introduced, it affects each conductor the same. If you flip any signal and add it to itself, the result is zero. Because it is flipped at the receiving end, the noise cancels out. This means there is little noise over long lengths of cable. Best for microphones, which have low signal levels, but also for long lengths of line level.
Unbalanced = single conductor and shield. Cheaper and easier to wire, but open to noise as well as signal loss over long length, particularly high frequencies due to capacitance (of interest to EEs only). Okay for line-level signals over short distances (like hi-fi rigs or electronic instruments), or microphones over very short distances (cheap recorders and PA systems).
Connectors: Balanced: XLR (as on microphone cable), 1/4” tip-ring-sleeve.
Unbalanced: RCA (“phono”), 1/4” (“phone”), mini (cassette deck or computer).
Mini comes in stereo version also (tip-ring-sleeve), for computers and Walkman headphones (both channels share a common ground). 1/4” TRS is also used as a stereo cable for headphones = two unbalanced channels with a common ground.
Guitar pickups = two kinds: piezo (mechanical vibration to electric current) and magnetic (metal vibration to electric current using fixed magnetic field: humbucker is special type)
DI boxes = transformers to match level and impedance of instrument to that of mic input on console and to balance unbalanced instrument cables.
Envelope = Change over time, applicable to any of the above:
volume envelope: fast = snare drum. slow = trombone with crescendo and diminuendo
frequency envelope: fast = Simmons tom. slow = siren
timbral envelope. fast = marimba. slow = gong.
Perception: dynamic and frequency range of human hearing
Characteristics of the ear as transducer.
Ear converts sound waves to nerve impulses.
Frequency response of cochlea
Each hair or cilium responds to a certain frequency, like a tuning fork. Frequencies in between get interpolated. As we get older, hairs stiffen, break off, and high-frequency sensitivity goes down. Also can be broken by prolonged or repeated exposure to loud sound.
Frequency sensitivity changes at different loudness levels: at low levels, we hear low frequencies poorly, and high frequencies too, although the effect isn’t as dramatic. Fletcher-Munson curve: ear is more sensitive to midrange frequencies at low levels, less sensitive to lows and extreme highs. In other words, the frequency response of the ear changes depending on the volume or intensity of the sound. When you monitor a recording loud, it sounds different (better?) than when soft.
Loudness sensitivity: Just Noticeable Difference (JND)--about 1 dB--changes with frequency and loudness level. We can often hear much smaller differences under some conditions, and not hear larger ones under different conditions.
Also, JND changes with duration--short sounds (<a few tenths of a second) seem softer than long sounds of the same intensity
Fidelity: what is it and what can get in the way? What goes in = what goes out.
Ideal amplifier=A straight wire with gain (signal is louder)
Coloration: Frequency response is limited.
Frequency response curve is not linear.
Distortion: extra harmonics are produced, either even or odd.
Aliasing, a by-product of digital conversion.
Dynamic range limitations
• Distortion caused by clipping or non-linearity: adds odd harmonics, particularly nasty=harmonic distortion
• Crossover distortion= certain types of amplifiers, where different power supplies work on the negative and positive parts of the signal (“push-pull”). If they’re not balanced perfectly, you get a glitch when the signal swings from + to - and vice versa.
• Intermodulation distortion=frequencies interacting with each other.
• Noise, hum, extraneous signals, electromagnetic interference (static, RFI)
Using filters/eq to change frequency response. EQ used to solve problems, and to be creative.
Haas effect: precedence of first-arriving signal. <35 ms later, second sound is blended. 35<50 ms, second sound is heard as ambience. >50 ms, distinct sounds. Lower values with transient sounds like drums.
Transducer = converts one type of energy to another
Microphone = converts sound waves in air to Alternating Current (AC) voltages. Dynamic Microphone has a magnetic metal diaphragm mounted inside a coil of wire. Diaphragm vibrates with sound waves, induces current into coil, which is analog (stress the term!) of sound wave. This travels down a wire as an alternating current: positive voltage with compression, negative voltage with rarefaction.
• Dynamic/moving coil (pressure-gradient mic)
• Condensor/capacitor=charged plate, uncharged plate, acts as capacitor, one plate moves, capacitance changes.
Charge comes from battery, or permanently-charged plate (electret), or dedicated power supply (old tube mics), or phantom power: 48v DC provided by mixer (doesn’t get into signal, because input transformer removes it).
• Ribbon (velocity mic) Metal ribbon is suspended between strong magnets, as it vibrates it generates a small current. High sensitivity, good freq response, a little delicate, figure-8 pattern.
• Boundary (pressure zone) Owned by Crown. Mic element is very close to wall. Hemispherical pickup, reflections off of wall are very short, essentially non-existent, prevents comb-filtering caused by usual reflections, even frequency response. Not good for singing, but good for grand piano (against soundboard), conference rooms, theatrical (put on the stage, pad against foot noises).
Polar patterns: omnidirectional, cardioid, hypercardioid, shotgun, bi-directional
Characteristics of a sound:
Frequency in Hz: how many vibrations or changes in pressure per second.
Loudness in dB SPL: how much air is displaced by the pressure wave.
Waveforms = simple and complex
Simple waveform is a sine wave, has just the fundamental frequency. Other forms have harmonics, which are integer multiples of the fundamental. [Thompson labels harmonics wrong: 1st harmonic is fundamental!] Fourier analysis theory says that any complex waveform can be broken down into a series of sine waves.
Saw: each harmonic at level 1/n. Square, only odd harmonics at 1/n. Triangle, only odd harmonics at 1/n2.
If there are lots of non-harmonic components, we hear it as noise.
White noise: equal energy per cycle (arithmetic scale)
Pink noise: equal energy per octave (logarithmic scale-more suited for ears)
So timbre = complexity of waveform, number and strength of harmonics. We can change timbre with filters or equalizers. Main types of filters: low-pass, high-pass, bandpass.
Stereo = since we have two ears. Simplest and best high-fidelity system is walking around with two mics clipped to your ears, and then listening over headphones: this is called binaural. Binaural recordings are commercially available: they use a dummy head with microphones in the earholes.
Systems with speakers are an approximation of stereo. The stereo field is the area between the speakers, and the “image” is what appears between the two speakers. If you sit too far from the center, you won’t hear a stereo image.
Multi-channel surround can do more to simulate "real" environments. Quad, 5.1 (".1"=Low Frequency Energy [LFE] since low frequencies are heard less directionally), 7.1, 10.1, 22.2, etc.
Position in the stereo or surround field = L/R, Front/Back, Up/Down. Determined by relative amplitude, arrival time, and phase.
The Nature of Sound
waves = pressure waves through a medium = compression (more molecules per cubic inch) and rarefaction (fewer molecules per cubic inch) of air. A vibrating object sets the waves in motion, your ear decodes them. Sound also travels through other media, like water and metal. No sound in a vacuum, because there’s nothing to carry it.
Speed of sound in air: about 1100 feet per second. That’s why you count seconds after a lightning strike to see how far the lightning is: 5 seconds = one mile. Conversely, 1 millsecond = about 1 foot.
Sound travels a little faster in warmer air, about 0.1% per degree F, and in a more solid medium: in water, 4000-5000+ fps, in metal, 9500-16000 fps.
When we turn sound into electricity, the electrical waveform represents the pressure wave in the form of alternating current. The electrical waveform is therefore an analog of the sound wave, Electricity travels at close to the speed of light, much faster than sound, so transmission of audio in electrical form is instantaneous.
Characteristics of a sound:
Frequency = pitch (look at pitch chart: http://www.psbspeakers.com/Images/Audiotopics/fChart.gif).
Expressed in cycles per second, or Hertz (Hz).
The mathematical basis of the musical scale: go up an octave = 2x the frequency.
Each half-step is the twelfth root of 2 higher than the one below it. = approx. 1.063
The limits of human hearing = approximately 20 Hz to 20,000 Hz or 20 k(ilo)Hz.
Fundamentals vs. harmonics = fundamental pitch is predominant pitch, harmonics are multiples (sometimes not exactly even) of the fundamental, that give the sound character, or timbre.
Period = 1/frequency
Wavelength = velocity of sound in units per second/frequency
Loudness (volume, amplitude) = measured in decibels (dB) above threshold of audibility (look at chart). The decibel is actually a ratio, not an absolute, and when you use it to state an absolute value, you need a reference. “dB SPL” (as in chart in course pack) is also referenced to the perception threshold of human hearing. Obviously subjective, so set at 0.0002 dyne/cm2, or 0.00002 Newtons/m2. That is called 0 dB SPL. By contrast, atmospheric pressure is 100,000 Newtons/m2
dB often used to denote a change in level. A minimum perceptible change in loudness (Just Noticeable Difference) is about 1 dB. Something we hear as being twice as loud is about 10 dB louder. So we talk about “3 dB higher level on the drums” in a mix, or a “96 dB signal-to noise-ratio” as being the difference between the highest volume a system is capable of and the residual noise it generates.
“dBV” is referenced to something, so it is an absolute measurement. “0 dBV” means a signal referenced to a specific electrical voltage in a wire, which is 1 volt. “0 dBu” is referenced to 0.775 volts, but it also specifies an impedance of 600 ohms. We’ll deal with impedance later. Common signal levels in audio are referenced to that: -10 dBV (consumer gear), +4 dBu (pro gear)
The threshold of pain is about 130 dB SPL, so the total volume or “dynamic” range of human hearing is about 130 dB.
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