More about Tape recording
High-frequency response limited by size of gap and size of domains: smaller gap means more particles per inch of tape. Finer particles mean more particles per square inch of tape. Also, speed of tape big factor = number of particles per unit of time. Professional tape speeds: 15 and 30 inches per second on 1/4" tape. Consumer (old): 3-3/4 and 7-1/2 ips. Analog cassette: 1-7/8 ips on 1/8" tape.
Dynamic response is limited: inherent noise caused by random orientation of particles ("hiss") means sounds cannot go much lower than noise level. Top end limited by "saturation": if magnetic particles are pushed too much, they resist, and the waveform will be distorted. Under controlled conditions, this can actually add to the "warmth" or immediacy of the sound, but often it just makes it sound nasty.
Dolby noise reduction is a scheme for boosting certain frequencies on record and reducing them on playback to lower noise and increase dynamic range. Best we can ask for in analog tape is about 70 dB of dynamic range. As tape widths and speeds went down, needed it more. Four types: A, B (consumer), C (better consumer), SR. New Dolby formats refer to film and transmission codecs.
Competing system: dbx. Still used in stereo broadcast TV.
Overdubbing: Les Paul invented method of overdubbing on one tape recorder, called it "Sound on Sound". Then went to multitrack.
Multitrack recording needed wider tape: 1/2", 1", 2". Fostex and Tascam bucked trend.
Nady Ribbon mics: bi-directional, equal pickup front and back. Off-axis frequency response doesn't change. Good for drum overheads if ceiling isn't too low. Do NOT use phantom power (although you probably won't harm them). Ribbons are delicate: don't drop them!
Plug-ins: instruments or signal processing modules, provided with a DAW or by third parties.
Formats: application native, VST, AU, TDM, RTAS, MAS, DirectX, etc.
Since dry recordings and synthetic sounds have no sense of space.
Two ways to use it:
1) direct insertion in an audio track.
2) create an Aux track with Bus 1-2 as input, insert Reverb plug-in, add Send (to Bus 1-2) on all tracks you want to apply reverb to. Sends allow different amounts of reverb to be applied to different tracks. Aux channel level is “return”: amount of overall reverb.
Advantage of aux track method is it uses less CPU power, allows similar reverbs to be used across multiple tracks, making more cohesive sound.
parameters: Frequency, gain, bandwidth ("Q")
Unless you are trying to eliminate a specific frequency, or boost it, eq is generally used to manage formants, not individual notes. Formants of an instrument remain much the same regardless of what pitch you're playing, and include all noise components and harmonics.
Often 200-800 Hz range is where "mud" is. Backing this off on some instruments can make tracks clearer.
To emphasize a bass drum, don't boost at 125 Hz, boost at 2 kHz to bring out transient "ictus".
Boosting any part of the spectrum by a large amount will boost the overall level of the signal. You should compenstate with attenuator just before EQ stage to keep any part of the system from overloading. Cutting a part of the signal doesn't affect the level as much. Cutting a small part of the signal is often inaudible, unless you are dealing with a single rogue frequency.
Use judiciously. A) Correct problems--except try to do that at session with mic placement and room treatment. B) Keep instruments from interfering with each other.
Two ways to do eq.
• The first is to start with a very low Q tuned to the approximate frequency range you think you want to change, and then to very gently boost or cut until you think you’ve got approximately the right tonal color, and then finish off by trimming the bandwidth down to just the width you want.
• The second approach is to start with a high Q frequency band boosted grossly (even if you ultimately want to cut it) and then sweep across the frequency range until you find the exact frequency that your ear says to change. Once you’ve located it, then start trimming the amount of boost (or start cutting if that’s what you want to do), and at the same time start decreasing the Q (increase the bandwidth) until you’ve got just the right timbral quality.
I prefer the latter technique for working on individual tracks and the former for more generalized eq on submixes or stereo recordings.
Analog recording history
Cylinder, wax/lacquer/vinyl discs
78s, 4-5 minutes on a side.
33 (Columbia) and 45 (RCA) came out at the same time, 45 ended up being used for singles.
Stereo discs, cutter head with two coils at 45° angle to vertical. Had to restrict dynamic range or playback stylus might jump out of groove. Had to put bass in the center of the stereo image.
Four-step process: master lacquer or acetate (positive), metal master or father or matrix (negative), mother (positive), stamper (negative), disk.
To maximize playing time, grooves could change distance between them (“pitch”) so softer passages could be closer together. When tape became the mastering medium, you could put a separate playback head to look ahead and determine pitch.
Wire, tape, multitrack tape
Tape recorder basics: tape formulas, heads, transport, bias, noise reduction
= records waveform voltages by aligning magnetic particles or “domains” on tape in step with the changing voltage. "Head" is transducer between AC voltage and fluctuating magnetic field. Tape is iron oxide or similar on plastic. Actual magnet is called "gap". As voltage changes, orientation of particles changes.
Multiple mic setups
Use single mics on most instruments. First, place the instruments in the room so that they sound good. Then place the mics where they sound good.
Observe the 3-to-1 rule: Mic must be at least three times further away from something it’s not picking up than what it is picking up. Sometimes phase switch (press Ø/insert/delay button on mixer to bring up phase switches on screen) can help with leakage problems.
Use stereo micing (pattern of your choice) for piano, drums, marimba, large instruments. Also consider using stereo room mics as well. Mono room mics don't often work.
Musicians should balance themselves before you set balances. They must able to hear each other.
On all channels except kick or bass: Set Yamaha low-band Q to "HPF”, roll off below 71 Hz. Eliminates room noise, some hum.
Live mixing: Pan individual instruments where you want them. They should be more or less in the same place as they are in the room. If that's not possible (instruments are against two walls or in a circle around the mics) then adjust intelligently.
Use pop filter on all vocals.
Soloing: in place (meaning in their pan position). Use to analyze individual mics. Only affects monitor and headphone outputs, doesn't affect stereo bus, so you can do it during a take.
New scene (#2) on mixer: 2-track to ADAT1-2. Mix will be in the Yamaha mixer, just sending two tracks to Pro Tools.
Dither: noise injected so that lowest bit is never dropped. Originally broadband white noise, then they found they could use high-frequency noise--essentially inaudible, has the same effect: known as Noise-shaping.
Limitation on early digital audio formats
Sony PCM-F1, PCM-1610, JVC, etc: used video tape, either 1/2" or 1/4", so
could only edit on frame boundaries i.e., 33.3 msec resolution.
Transmitting digital audio—Stereo
AES/EBU = AES Type I Balanced – 3-conductor, 110-ohm twisted pair cabling with an XLR connector, 5 volt signal level
S/PDIF = AES Type II Unbalanced – 2-conductor, 75-ohm coaxial cable with an RCA connector, used in consumer audio, 0.5v
The two data sets are almost identical. You can easily convert from one to the other with a simple voltage gain or drop.
TOSLINK = AES Type II Optical – optical fiber, usually plastic but occasionally glass, with an F05 connector.
ADAT Lightpipe, 8 channels on optical fiber. Same cables as TOSLINK, but not compatible.
Tascam TDIF, 8 channels on DB25 connector, same as original SCSI spec.
MADI coaxial (BNC connnector) or optical (wider than TOSLINK), 48 or more channels, used in older multitrack decks and high-end installations. Now making a comeback with digital sound reinforcement consoles.
Newest format: over Ethernet. Right now mostly for live sound: allows single cable between stage and mix position. Ethersound, CobraNet, Aviom, all incompatible with each other. Dante probably going to catch on: can use ordinary switchers. Many manufacturers signing up for licenses.
Clocking & Jitter
When using multiple digital sources, they must have a common clock, or else there will be clicks where the clocks are out of sync and samples are dropped. So there must always be one master.
Word clock signal can be generated by one device, and fed through the others, or fanned out to the others.
Or, if all devices are capable of syncing to incoming digital audio stream, you can daisy-chain them.
Tom says: Master clock when you're recording should be the device that is doing the analog-to-digital conversion, so that jitter is minimized. In our setup, mixer is both A-to-D converter and master clock.
DAT: was killed in the consumer market by RIAA lobbying for law requiring Serial Copy Management System (SCMS) chip in consumer units, so no one made any.
Digital disc vs. digital tape
Tape is sequential, disc is random access.
With tape there is a direct correlation between the number of physical inputs, the number of tracks, and the number of physical outputs.
With disc, there is no correlation: inputs and outputs are determined by the audio interfaces, and tracks can be much higher: determined by the speed of the CPU and the throughput of the disc.
Pro Tools Basics:
Smart tool for trimming, selecting region (command-E), fading in or out, 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. Regions.
Editing modes: Slip-move freely. Grid-move in quantized intervals. Shuffle-move a region and other regions jump around to fill in.
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.
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; they 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. Pre-fader for monitors, post-fader for effects.
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.
Technical overview: Take a sample of the signal voltage and write it down as a number.
Issues: how often (sample rate), how accurate is the number (word length).
A-D converter does this.
Nyquist theorem: highest frequency sampleable is 1/2 the sampling rate. If you go too high, you get aliasing.
Word length: number is binary, so number of bits determine the range. With 10 bits, you get 0-1023. With 16, 0-65335. Difference between analog input and digitized signal is called Quantization noise.
Dynamic range=highest level possible/quantization noise level=6.02 number of bits + 1.76dB
Aliasing: difference between over-Nyquist frequency and Nyquist frequency. Not harmonically related to anything--like a ring modulator. Sampling process creates sidebands above and below the sampling frequency. Normally these are not a problem since they are above the Nyquist frequency and are filtered out either digitally or at the D-A stage. But if the input signal is higher than half the sampling frequency, the sideband is lower than half the sampling frequency (the Nyquist frequency), and it goes into the audible range. Created by bad filtering, or distortion from overload at the analog stage of the A-D converter.
D-A converter: creates signal voltages from samples; uses sharp filters (decimation) to round off the edges of the waveforms.
First popular system: Sony PCM-F1, PCM-1610: used video tape, either 1/2" or 3/4" -- could only edit on frame boundaries, 33.3 msec resolution.
Multitrack: Mitsubishi, Studer, Tascam, Sony PCM-3324 and -3348. Most of them long gone. Replaced by ADAT, and to some extent by Tascam DA-88 (didn't do as well: price point was higher and introduction was a few months later).
ADATs could easily be combined, and controlled by a single controller which acted as if it was a 32-track deck.
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 separate out frequencies for 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? Bi-amplification.
Sensitivity: output SPL per watt input at specific distance at specific frequency. Without all these parameters, spec is meaningless.
Other specs: freq response, THD, maximum power, often miselading.
Damage: causing woofer cone to go too far can tear it or pull it off its mount. Too much current can melt it. Tweeters suffer from under-powering: an amp that doesn't have enough power clips the signal, sending loud high-frequency distortion products to tweeter can damage it.
Near-field: small speakers up close to minimize room effects.
How to use speakers in practical situations? Get used to them! Listen to music that you know on them, so your ear can make comparisons.
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.
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.
Headphones: open (foam), closed (Koss), semi-closed (lighter plastic), noise-cancelling (Bose).
Can be more accurate, move much less air so elements are lighter, no room effects.
Problem: interaural bleed is gone, so stereo image is very different from speakers. Processors now available (we have Focusrite VRM) that simulate speakers in headphones.
Ear buds: no isolation, low dynamic range, less low-freq response. Getting better! Watch out for exaggerated LF response. Watch SPL!!
In-ear monitors: Isolated, advantage is less sound on stage getting into FOH system. For drummers often combined with speakers or throne drivers, e.g. “Buttkicker”
Using speakers in the lab. Only when you or your team are the only ones in there! Remember to turn off mixer inputs when you're done.
Focusrite VRM box: simulates multiple speaker types in three different rooms when used with headphones. Plugs into USB port on Mac. When using with ProTools, select Playback Engine>VRM Box to send output to it, increase sample buffer to 1024.
Using server: Music65 storage. Make sure you back up entire session folder. Don't run session from server--always copy it back to your local computer.
Real world studios:
Studio layout: control room, live room, iso booth(s), gobos
Studio wiring: input panels, monitor outputs, cue systems
1st teams: *has swipe access
Greg, Anecia, Danny*: two acoustic guitars
Eriks, Brett, Scott*: piano and acoustic guitar
Andrew*, Cooper, Maeve: vocal and guitar
Kurt*, Stephen, Jackson: piano and melodica
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 levels, don't need to stick with one of the stereo patterns.
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.
on the recording cart
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 when sending signal to ProTools.
Recording into ProTools. 44.1 kHz, 16-bit. Make sure playback engine is MOTU 828x, clock is set to Optical.
Create a folder with your names on it on the laptop EXTERNAL drive, put session in there.
After recording, use a flash drive to move entire folder onto lab computer to edit and/or mix.
Two A/T multipattern and two Shure SM57 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.
Go through the mixer channels 1 & 2, panning hard left and hard right, comes up in ProTools as inputs 1 and 2. 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.
Micing—how do you decide what and where?
• 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 fix 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
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 (acoustic): 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.
Guitar amp: Experiment to see what sounds best, maybe close (dynamic usually just fine—not a lot of highs and lows), maybe far, maybe both (condensor for room sound).
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 used 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 (isolating individual drums) later.
Mic techniques for stereo: none of them are perfect! XY, ORTF, DIN, NOS, MS, spaced omni, spaced cardioid, Decca tree.
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.
Proximity effect: Sound entering a microphone has two components: phase difference and amplitude. A 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 increases, 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.
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
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.
Cables: Balanced vs.
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.
Phase: where in
the waveform you are at any moment. Hearing absolute phase is difficult,
but relative phase between two signals is easy. Localization in the human
hearing system uses amplitude, time (phase), and frequency response. It is
especially 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
Head acts as bafffle for high frequencies, so relative amplitude used more for localization at high frequencies. Low frequencies bend around the head, so phase is more important. (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, which happens in a reflection in a room or the pinna (earlobe), the various harmonics will cancel or reinforce each other based on the delay period. This is comb filtering.
Also, tiny reflections within the pinna clue us in on directionality, because spectrum changes.It means that the frequency spectrum of what you’re hearing changes if you move, even very slightly. So 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.
NIH study: People who lose their earlobes and are given new ones have a lot of trouble re-learning how to localize sounds.If you change the delay period, 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, "traps".
More a problem at low frequencies, since specific frequencies stand out. At higher frequencies, they blend together, not nearly as obvious.Three types of room modes: Axial, tangential (-3 dB. 1/2 level), oblique (-6dB, 1/4 level)
Calculate them with this utility: http://www.mcsquared.com/modecalc.htm
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.
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 back plate + diaphragm 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 mics (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).
Pickup patterns: Omndirectional, Cardioid, Figure 8 (bi-directional), Hypercardioid, Shotgun.
Characteristics of a
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.
Timbre = complexity of waveform, number and strength of harmonics. We can change timbre with filters or equalizers.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. 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)
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.
and frequency range of human hearing
Ear converts sound waves to nerve impulses.
Each hair or cilium responds to a certain frequency. 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.How 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.
Using filters/eq to
change frequency response. graphic, parametric, High Pass, Low Pass, BandPass,
EQ used to solve problems, and to be creative.The smallest difference we can hear in a level of sound--the “Just Noticeable Difference (JND)”--is 1 dB. This 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.
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.
• Bandwidth limitations
• Frequency response anomalies=like a filter or eq
• Dynamic range limitiations
• Distortion caused by clipping or non-linearity: adds odd harmonics, particularly nasty (show in Reason)=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
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.
©2014 Paul D. Lehrman, all rights reserved