Music 65 lecture notes
Near-field monitors: small speakers up close to minimize room effects.
In a studio, use multiple speakers to monitor recording and especially mix: high-end and low-end. Auratones, Yamaha NS-10s popular for simulating home hi-fi, television, car. Engineers would put tissue paper in front of the NS-10 tweeter to reduce the high end a little.
Power Amplifiers: matching to speakers, impedance (= resistance at audio frequencies, in ohms), damping factor: ratio of speaker impedance to source impedance. How well it controls mechanical resonances: high damping factor acts as a "brake" on the cone; low damping factor means it can ring. So you want output impedance low (typically 0-1Ω), speaker impedance high (8Ω down to 2Ω).
Many amplifier manufacturers state power levels going into a low-impedance load, makes them look more powerful.
Real world studios:
Studio layout: control room, live room, iso booth
Studio wiring: input panels, monitor outputs, cue systems
Mixing console channel strip:
• Input selector—mic and tape inputs usually hard-wired through patchbay (normalling)
Mic and line inputs, balanced
Mic level: ~2 mv
Line levels: -10 dBV= 0.316 V or +4 dBu=1.2276 V
0 dBV=1 V RMS without impedance reference (usually high)
0 dBu=0.775 V RMS (corresponds to dBm, which is across 600Ω load)
Pro consoles usually +4, semi-pro -10 or switchable.
ª Input trim for adapting to input level, pad (-20dB)
Important to have all amplifier stages operating in optimum range! Avoid noise pickup and distortion: “Proper gain staging”
• Mic preamps (virtual consoles use outboard interface including mic preamps and A-to-D converters)
Some prefer outboard mic preamps to built-in preamps; convert to line level or to digital.
• Hi and low cut filters, for room noise, hiss, sibilance, mic proximity effect
EQ, simple or parametric, in/out switch
• Compressor/Gate. Smoothing out levels on vocals, basses, drums; gate for isolating drums, other track leakage.
• Output assignments: to tape or hard-disk tracks. Select an output bus or “direct”. Output buses can also be used for sub-mixes: e.g., group all the drums to two faders.
• Aux sends for processing. Aux buses allow multiple tracks to go through a single processor, so all tracks get same reverb for example, but you can adjust how much reverb is added to each track. Pre/Post-fader switch usually set to Post.
• Monitor mix: control room, studio playback. Solo button isolates source in monitors, but doesn’t change assignments or mix. “Solo in place” keeps pan position, otherwise comes up in mono.
• Cue sends for musicians on headphones. Might be the same as aux sends. Pre/Post switch usually set to Pre.
Of all the components in an audio system, these have by far the worst frequency response and distortion. Physics of moving air is difficult. The perfect speaker would weigh nothing and have infinite rigidity. The spider which holds the cone against the magnet would weigh nothing and have infinite flexibility. The space inside the cabinet would be infinite so that nothing impedes the movement of the cone.
Break up the spectrum into components that work best over a linited range: Woofers, tweeters, midrange, Sub-woofers.
Directivity: low frequencies spread out more, high frequencies are localized, “beamed”.
Crossovers: filters to divide sound between different drivers.
Time-aligned: tweeter is delayed or set back to compensate for depth of woofer cone. Theory says this preserves transients, prevents phase interference between drivers at overlapping frequencies.
Concentric drivers sometimes used for time/space alignment.
Passive vs. active speakers: where does crossover go? In active speaker, before the amplifiers: Bi-amplification.
Sensitivity: output SPL per watt input. To be meaningful, needs distance (e.g. 1 meter) and frequency e.g. 1 kHz).
Other specs: frequency response +/- x dB), THD, maximum power. All of these are often miselading.
Choosing your mic pattern: use the results of last week’s experiment. How much “center” do you need? Is tonal balance or spatial placement more important. If recording instruments of very different volumes, not necessary to stick with a stereo pattern, just make it sound good in stereo.
If you use M-S, when you are playing back, duplicate the Side channel onto another track. Phase-reverse it using using Trim plug-in, and then group both Side faders. Level of side faders determines width of stereo image.
ProTools 10 on the recording cart
Use template! Saving on external drive will create folder and PTX file.
Opening and logging into MacBook Pro.
Recording into ProTools. 44.1 kHz, 16-bit. Make sure clock is set to Optical.
After recording, use a flash or portable drive to move entire folder onto lab computer to edit and/or mix.
Two Rode multipattern and two Electro-Voice n/d267a dynamic cardioids with stands and cables are in the front of the booth.
You can use Fisher (but not when there’s a musical event going on in Distler), room 24, 27 (only when 24 is booked by someone else), 155, 251, or 271. Reserve the room and the recording cart with the music office at least 24 hours ahead of time. When you are ready to go into the room, find the practice room monitor to open the closet and the room for you.
01V: use scene 1
Go through the mixer channels 1 & 2, panning hard left and hard right, comes up in ProTools as inputs 1 and 2.
Simple Yamaha 01V mixer controls: Mic trims, panning, setting levels, phantom power. Get a "green " signal without a red light. Level should top out between -12 and -18.
Fader positions don’t matter.
Do not use any eq or processing. Try different instrument and mic positions. Edit if you need to. Goal is to make something that sounds realistic, and good. Try different instrument and mic positions.
Pro Tools basics:
Smart tool for trimming, selecting region (command-E), fading in or out (command-F), cross-fading between adjacent regions.
To automate fader movements: put track in auto record, not Record! Afterwards, use drop-down in Edit window to show movements, edit.
Multiple takes: use adjacent tracks, comp together by selecting regions and moving vertically (Hold shift key to lock in time).
Grouping stereo or multiple tracks for editing and/or mixing. .
Editing modes: Slip: move freely. Grid: move in quantized intervals. Shuffle: move a region and other regions jump around to fill in.
Bouncing to AIFF or WAV: File>Bounce to Disk. 16-bit, 44.1 kHz, interleaved. Mix is done in real time. Mix will be stored in Audio Folder of session.
1st Project Teams:
John, Ben, Peter
Anjali, Tim, Adam
Nathan, Eamon, Joe P
Joe C, Tom, Walker
Micing—how to decide
• 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 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. https://www.youtube.com/watch?v=8KRDhxkz4sQ
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 for drums: three large cardioid condensors in an isoceles triangle.
How pickup patterns are designed into mics:
Omni, picks up only from the front. Slight shadow effect on sound from the rear, but otherwise truly omni at all frequencies.
Cardioid. Sound from rear arrives at rear before it arrives at the front. So this mic has ports that feed sound from the rear with a slight delay, created by labyrinth or material that slows down the sound, so that sound arrives at front and back of the diaphragm at the same time, cancelling each other out. Sounds arriving from the front cause the greatest difference in pressure between front and back; sounds from sides and rear least pressure.
Figure 8 has one diaphragm open to both sides. Sounds from side arrive on both sides at the same time, cancel each other out.
Hypercardioid, supercardioid, shotgun: variations using different types of ports.
Frequency response changes off axis, not linear. Causes coloration in off-axis instruments. When using multiple mics in a setup, have to be careful with this. Experiment: need powered speaker hooked up to another computer (1/8’ cable), generating complex sound, microphone on stand hooked up to Mbox 2 (XLR cable), feeding through spectrum analyzer. Move it around.
Proximity effect: Sound entering a microphone has two components: phase difference and amplitude
Directional mic blocks sound from the rear and sides. Sound from the front goes to the front of the diaphragm. Sound from the rear goes to the front and through a labyrinth to the rear. Difference of arrival time/phase moves the diaphragm. As frequency rises, phase difference goes up—fixed arrival time means that it becomes a greater proportion of the waveform. Slope is 6 dB/octave. Electronics built in to compensate: bring down high end @ 6 dB/octave.
Omnidirectional mics don’t block sound from any angle, so the frequency response is flat—don’t need compensation!
Amplitude is actual air pushing against the diaphragm. At distances, because of the inverse square law, the effect on the diaphragm is negligible. But as distance decreases, amplitude differences between front and back get higher, and at very short distance, amplitude component overwhelms the phase component. The amplitude component does not change with frequency—result is that the highs are attenuated but the lows are not. Hence, bass boost.
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).
Impedance: electrical characteristic, has to be matched to effect efficient energy transfer. In audio, best to have a low-impedance source feeding a high-impedance input. If impedances are not matched correctly, signal reflects back along the line, causing loss. At high frequencies (above 1 MHz) this also causes frequency anomalies, but not at audio frequencies.
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).
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.
EQ and filters: graphic, parametric, High Pass, Low Pass, BandPass
Phase: where in the waveform you are at any moment. Hearing absolute phase is difficult, but relative phase between two signals is easy.
How we localize: Amplitude, time, phase, frequency spectrum
The human hearing system is very sensitive to phase: interaural effect. It tells us quickly about the location of a sound as it arrives at our two ears, along with relative amplitude.
Head acts as bafffle for high frequencies, so relative amplitude used more for localization. Low frequencies bend around the head, so phase is more important for them than localization. (It’s why you need only one channel of subwoofer in a surround system.)
When waves coincide, energy is increased. When waves are in opposition, they cancel each other. If you take two identical complex signals and delay one, the various harmonics will cancel or reinforce each other based on the delay period. This is comb filtering. It means the spectral content of the sound changes. It happens in any room and also within the pinna. Tiny reflections within the pinna clue us in on directionality, because spectrum changes.
So if you move your head, even very slightly, the frequency spectrum you’re hearing changes. Therefore turning your head to localize a sound changes the phase, the timing, and the spectrum. We learn how to use this very well early in life.
If you change the delay period artifiically, the phase cancellations move, creating the phasing or flanging effect. Sound isn't moving, but seems like it should be.
The role of the room: standing waves or “room modes” caused by phase reinforcements due to the reflections in the room. Based on dimensions of the room. The more reflective the walls, the greater the problem. Lots of techniques for minimizing these including absorbers, diffusors.
More a problem at low frequencies, since specific frequencies stand out. At higher frequencies, they blend together, not nearly as obvious.
Effects of speaker placement on frequency response: in a corner or against a wall, bass is emphasized. Some speakers are designed to go in corners--their low-end response is tailored to compensate.
How to avoid room modes: traps
In the studio: four different types of rooms
Control room—very tight, flat
Live room—may have different areas with different acoustics
Drum room—also may be variable
Iso booth—for voices, amps, usually dead
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).
Omndirectional, Cardioid, Figure 8 (bi-directional), Hypercardioid, Shotgun.
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=LFE since low frequencies are heard less directionally), 7.1, 10.1, etc. Will do a little with it in this course.
Position in the stereo or surround field = L/R, F/B, U/D. Determined by relative amplitude, arrival time, and phase.
Haas effect: precedence of first-arriving signal. If two sounds are equal in amplitude, the source of the first to arrive will be the apparent location. If second sound is <35 ms (approx.) later, second sound is blended. 35<50 ms, second sound is heard as ambience. >50 ms, heard as two distinct sounds. Lower values with transient sounds like drums.
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.
Vibrato envelope = change speed or depth of volume, pitch, or timbre
Perception: dynamic and frequency range of human hearing
Characteristics of the ear as transducer.
Ear converts sound waves to nerve impulses.
Morphology of ear
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 or function of frequency response curve is not linear.
Distortion is introduced, certain extra harmonics are produced, either even or odd.
Aliasing, a by-product of digital conversion.
Noise is introduced.
Dynamic range limitations
Distortion caused by clipping or non-linearity: adds odd harmonics, particularly nasty
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)
EQ used to solve problems, and to be creative.
Basic audio principles:
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, 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 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.
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.
Timbre = complexity of waveform, number and strength of harmonics.
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
Change timbre with filters or equalizers by boosting or cutting parts of spectrum.
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