Music 65 • Prof. Lehrman's notes
Analog recording: theory & history
Cylinder, wax: vertical “hill-and-dale” recording.
Lacquer/vinyl discs: lateral recording (mostly)
78 rpms, 4-5 minutes on a side. Records came in multi-disc “albums” so you could have an entire symphony.
33 (LP) (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.
Five-step process: master lacquer or acetate (positive), metal master or father or matrix (negative), mother (positive), stamper (negative), disk. Could use three-step process, but stampers and mothers wear out. Five-step allows more copies to be made.
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. Particles have intertia that will prevent them from changing orientation fast enough, which introduces distortion.
Bias (invented by the Germans—this was their big breakthrough) is high-frequency signal that keeps domains moving at all times, eliminates inertia.
3 parameters: Frequency, gain, bandwidth ("Q"). Also peaking/notching (select frequencies around a center frequency) or shelving (all frequencies above or below cutoff point)
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 an attenuator just before EQ stage to keep any part of the system from overloading. (ProTools EQ has input and output levels for proper gain-staging of the eq.) 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 determine where to 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.
Jitter is slight variation—at the nanosecond level—in the clock rate of a digital audio system. Result is noise similar to quantization noise, but in the frequency domain as opposed to the amplitude domain. Very hard to find a digital clock that has audible jitter! You can spend a lot of money on dedicated clocks, but listening tests show that they make no difference.
Second recording project: multiple microphones to two-track: no overdubbing, can only record on two tracks. Mix will be in the Yamaha mixer, sending Stereo L+R to Pro Tools.
New scene (#1) on mixer: 8 inputs to 2-track via ADAT1-2. Set levels with trims and faders; set pans using Select buttons. Hipass filter at 75Hz has been added to all channels; you can defeat it on bass or kick (also using Select buttons). Don't let the main stereo meter show red!
Monitor your recording from the headphone output of the mixer, through the splitter box if you need to. Listen to playback from the headphone out of the MOTU 828.
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 composite video, 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. Making a bit of a comeback with multichannel digital consoles.
Newest way: over Ethernet CAT-5e or CAT-6. Right now mostly for live sound: allows single cable between stage and mix position. Can carry audio, video, and control data.
Several formats: Ethersound, CobraNet, Aviom, MOTU AVB
Dante owned by Audinate most promising: can use ordinary switchers. About 200 manufacturers now have licenses.
Advantages: possible frequency response, dynamic range increases? Better marketing!)
Disadvantages: more storage; more bandwidth through mutlitrack interfaces; more CPU power needed, therefore harder on DAW
Is it necessary? Hard to justify higher sample rates; longer word lengths make some sense.
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.
Disc systems can do processing on the fly, non-destructive editing (signal is processed as it plays).
In early days when space was scarce, sometimes used destructive editing. Now "constructive": do a file edit, and it creates a new file.
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.
Master clock when you're recording should be the device that is doing the analog-to-digital conversion.
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), how accurate is the sample clock (jitter).
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
D-A converter: creates signal voltages from samples; uses sharp filters (decimation) to round off the edges of the waveforms.
Digital Recording 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.
DAT: was killed in the consumer market by RIAA lobbying for law requiring SCMS chip in consumer units, so no one made any.
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: after amplifier (in passive systems) or before (in active systems). Bi-amplification: separate amplifier for each driver.
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.
Sensitivity: output SPL per watt input. But they should also specify a distance (e.g. 1m) and a frequency (e.g. 1 kHz).
Other specs: frequency response (watch out for ±n dB!), THD, maximum power, often misleading.
Near-field: 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. Tissue paper in front of the NS-10 tweeter?
Damage: causing woofer cone to go too far can tear it or pull it off its mount. Sending high-frequency distortion products to tweeter can damage it.
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.
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 beginning to appear that simulate speakers in headphones.
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.
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 bass players and drummers, often combined with speakers or throne drivers, e.g. “Buttkicker”.
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.
Using speakers in the lab: Starting with the next project. Not when anyone besides you is in the lab. If anyone comes in to work, you must turn off the speakers immediately. And don't forget to turn off the input channel at the mixer when you are done.
Bouncing to AIFF: 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.
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
Studio wiring: input panels, monitor outputs, cue systems
Mixing consoles: see Prof. Swanson's notes here
Evening class notes:
Most consoles have balanced mic AND line inputs (XLR and TRS)
Mic level: ~2 mv
Line levels can be either -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.
Important to have all amplifier stages operating in optimum range! Avoid noise pickup and distortion: “Proper gain staging” of preamp, faders, aux sends and returns, etc.
Mic preamps: Built into consoles. Some people prefer outboard mic preamps to built-in preamps; they convert to line level or to digital. They can get very expensive! Virtual consoles in DAWs use outboard interfaces which include mic preamps and A-to-D converters.
Phantom power. Use capacitors to block DC from power supply from reaching mic preamp. Use Zener diodes to reduce power surge and "pop" when connecting a mic while phantom power is on.
First project proposals. Finalize by Thursday. Due Oct 6
2 mics to Pro Tools.
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 room for you.
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.
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.
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.
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. 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 splits the region), 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. Crossfades
Editing modes: Slip: move freely. Grid: move in quantized intervals. Shuffle: move a region and other regions jump around to fill in.
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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