Putting a sound somewhere
Three things that get conflated constantly: polarity is not phase, a waveform is a recipe rather than a shape, and the ways of placing a sound in a room are not variations of one idea — they fail differently, and the failure is how you choose between them.
The confusion
Polarity is not phase
These two words are used interchangeably by people who know better, and the distinction is worth being pedantic about because the fix is different.
Polarity is a sign flip. Every positive becomes negative and every negative positive, identically at every frequency, instantly. It is what the ø button on a preamp does, what swapping pins 2 and 3 does, and what wiring a loudspeaker backwards does. It is completely reversible: press the button again and you are exactly where you were.
Phase is a time relationship, and here is the part that matters — a fixed delay is a different number of degrees at every frequency. Three metres of air is about 8.7 ms, which is a full cycle at 115 Hz, half a cycle at 57 Hz, and dozens of cycles at 4 kHz. So two mics on one source are not “out of phase” by some amount you can dial out. They are in a different phase relationship at every single frequency, and the comb filtering that results has peaks and nulls spaced by the reciprocal of that delay.
Which is why the polarity button sometimes helps and never fixes it. Flipping polarity turns every cancellation into a reinforcement and vice versa — it moves the comb rather than removing it, and whether that is an improvement depends entirely on where the nulls land relative to what you wanted to hear. The actual fix is time: move the microphone, or delay the other one. The speaker delay calculator is the same arithmetic pointed at loudspeakers.
Absolute polarity
The third thing, and the one worth the least argument
There is a further question hiding under the first one, and it gets more heat than it deserves: not whether two sources agree with each other, but whether the whole chain agrees with the world. If a drum skin moving towards the microphone ends with a loudspeaker cone moving towards the audience, the system has correct absolute polarity. If everything is flipped together, it does not.
It is largely inaudible on most material, for a good reason: a symmetrical waveform sounds the same either way up, and most sustained musical sound is close to symmetrical. Where it becomes audible is on asymmetrical waveforms — a kick drum, a bass guitar, brass, the human voice on plosives — where the pressure excursion is genuinely larger in one direction than the other. There the flip changes which way the loudspeaker moves first, and on a system with real low-frequency authority some people reliably hear the difference on those sources and nothing else.
The practical position is the honest one. It is worth getting right because it costs nothing to get right, it is worth checking with a polarity popper when a system is commissioned, and it is not worth an argument at eleven at night. What is worth an argument at eleven at night is relative polarity between two boxes in the same array, because that is not subtle, it cancels real output, and it reads as a room problem rather than a wiring one.
- Reaching for ø on a comb filter. It moves the peaks and nulls; it does not remove them. Sometimes that is an improvement and it is never a fix.
- A polarity error on one loudspeaker in an array. The low end largely cancels where the coverage overlaps, and it reads as a room problem rather than a wiring one.
- The 3:1 rule, misremembered. It is about keeping the second microphone at least three times as far from a source as the first, so what leaks in is low enough that the comb filtering does not matter. It is a level rule dressed as a distance one.
- Assuming a DI and a mic on the same instrument agree. They are metres apart in time. That is phase, and it wants delay, not a button.
The recipe
What a square, a saw and a triangle actually are
Every periodic wave is a sum of sines at whole multiples of its fundamental. That is not an approximation or a way of thinking about it — it is what the wave is, and the four classic synthesiser shapes have famously tidy recipes.
A sine is the fundamental and nothing else, which is why it is featureless and why it is the only shape that survives any linear system unchanged. A sawtooth has every harmonic, falling off as 1/n, which makes it the brightest and the most useful raw material for subtractive synthesis — there is plenty there for a filter to remove. A square has odd harmonics only, also at 1/n: no octaves at all, which is exactly what makes it sound hollow and clarinet-ish rather than bright. And a triangle has odd harmonics at 1/n², which falls away so fast that the third is already down at a ninth and the shape is very nearly a sine.
The connection worth carrying off this page: those odd harmonics on a square include the third, which is a triplen — the same harmonic that arrives in phase on all three legs of a supply and fills up a neutral conductor. An oscillator generating a square wave and a switch-mode power supply drawing a spiky current are not analogous, they are the same fact pointed at different jobs, and the same arithmetic describes both.
Making one
Subtractive, additive, FM, and why one of them won
Subtractive synthesis starts with something harmonically rich — a saw, usually — and takes away. A filter with a cutoff and a resonance removes the top, an envelope moves the cutoff over time, and that is most of the sound of electronic music. It won because it maps onto how the ear hears change: a filter sweep is a single gesture that alters an enormous number of harmonics at once, and it is one knob.
Additive synthesis goes the other way, building a sound by summing sines with individual envelopes. It can theoretically make anything, and it is almost unusable by hand: a convincing sound needs dozens of partials each with their own amplitude curve, which is a lot of parameters for one note and no gesture that moves them together meaningfully.
FM uses one oscillator to modulate the frequency of another. The result is a sideband spectrum that is not obvious from the controls at all — small changes in the modulation index reorganise the harmonic content wholesale, which is why FM is famous for metallic and bell-like sounds and for being difficult to steer. It is enormously efficient, which is why it appeared in hardware when memory and processing were expensive.
All three are still around, and the distinction blurs in software where nothing costs what it used to. What survives is the ergonomic point: subtractive won not because it sounds better but because one knob moves the whole spectrum in a way a person can predict.
The two knobs
What an envelope and a filter are actually doing
Subtractive synthesis is two ideas wearing a lot of front panel. Everything else is a variation.
The envelope describes how something changes from the moment a key goes down to after it comes up, and the standard four stages are attack, decay, sustain and release — ADSR, which is what the four knobs are labelled on almost every instrument ever made. Three of those are times and one is not: sustain is a level. Attack is how long to reach full, decay is how long to fall from full to the sustain level, sustain is the level it holds at while the key is held, and release is how long to fall from wherever it is to silence once the key is let go. Almost every confused synth patch is somebody turning sustain expecting a duration. A plucked sound is a fast attack, a short decay and a sustain of zero — the note ends while your finger is still down, and it is the sustain level that ends it.
The filter is usually a low-pass with two controls. Cutoff is the frequency above which it starts removing, and it is the single most expressive control on the instrument because moving it changes every harmonic at once in a way the ear reads as a sound opening or closing. Resonance is feedback around the filter that boosts a narrow band right at the cutoff, which makes the sweep audible as a distinct tone travelling through the sound rather than as a dulling. Turn it far enough and the loop sustains itself: the filter oscillates with no input, which is a sine source you get for free and a very effective way to make an unpleasant noise at full level.
Then the two are joined, and that join is the whole instrument: an envelope routed to the filter cutoff rather than to volume. Now the sound is bright at its start and darkens as it decays, which is what nearly every struck or plucked physical object does, and it is why that patch sounds like something happened rather than like a tone was switched on.
- Sustain is a level, the other three are times. If a sound will not stop while the key is down, you want sustain at zero and a decay, not a shorter release.
- Resonance at high cutoff plus full level is how monitors get damaged. A self-oscillating filter has no dynamics and no headroom of its own.
- Filter envelope amount can be negative. A sound that opens rather than closes is the same patch with the sign flipped, and it reads as unnatural for exactly that reason.
- Envelope times are not linear on most instruments. They are exponential, which is why the last third of an attack knob does most of the audible work.
Placing it
Three ways, and they fail differently
Putting a sound somewhere in a room is not one problem with three solutions. It is three different bargains.
Share one signal between speakers by level, so the ear infers a direction between them. VBAP does it geometrically from a triangle of speakers; DBAP does it from distance without assuming where the listener is.
Fails by moving. The image is correct at the point the geometry was drawn around and follows the listener everywhere else — a sweet spot rather than a location.
Does not fake a direction. It rebuilds the wavefront a real source would have made, using an array of loudspeakers as a discrete sampling of a continuous surface, so the image really is at a place and stays there as you walk.
Fails by frequency. The array is a spatial sampler, so it has a Nyquist limit: above c / 2d it aliases, and halving the spacing doubles both the limit and the loudspeaker count.
Stores direction as spherical harmonics rather than as speaker feeds, so the same file decodes to any layout — a dome, a cube, or a pair of headphones. Higher orders carry more directional detail.
Fails by order. First order is a smear; useful sharpness needs third order or above, and the channel count grows as (n+1)² — 16 channels for third order, 36 for fifth.
Applies the head-related transfer function (HRTF) — the filtering a real head, torso and outer ear impose on a sound before it reaches the eardrum. Timing and level differences only ever say left or right; it is the notches the pinna carves into the spectrum that say up, down and behind, which is why a sphere needs the whole filter and not just a delay.
Fails by head. Those filters are individual — someone else’s ears give you their localisation, not yours — and without head tracking the world turns when you do.
- Equal-power panning read as equal-voltage. Dead centre is −3 dB in each speaker, not −6. Get that wrong and the middle of the image dips.
- A WFS array specified without its aliasing frequency. It is the number that decides the cost, and it is one division.
- First-order ambisonics sold as immersive. It is four channels and it is genuinely blurry. That is not a criticism of the format, it is what first order is.
- Binaural on speakers. The filters assume the sound arrives at each ear separately. Without crosstalk cancellation, played on speakers, it is just an odd-sounding stereo mix.
Where this goes next
The calculators on this page
Waveform harmonics shows the recipe for each shape and hands it to the distortion maths. VBAP and DBAP does both panning laws, including the rolloff trade in DBAP. Wave field synthesis gives the aliasing frequency and what a target would cost in loudspeakers. And how each sense tells things apart is why any of it works — two ears, a delay of a few hundred microseconds, and a great deal of inference.
4 questions on the mechanism rather than the trivia. Nothing is scored and you can answer again — the explanation under a wrong answer is the point of it.
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1Two microphones on one guitar cabinet sound thin and hollow together. Somebody reaches for the polarity button. Will it fix it?
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It will change the sound, and it will not fix it. Flipping polarity turns every cancellation into a reinforcement and vice versa, which moves the comb rather than removing it.
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Right. A fixed delay is a different number of degrees at every frequency, so there is no single flip that aligns them. Move the microphone or delay the other one. The button may improve it by moving where the nulls land, which is not the same as fixing it.
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Microphone type has nothing to do with it. The distance between them does.
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Matching mics does not remove the path length difference, which is what is causing the comb filtering.
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2Why does a square wave sound hollow rather than bright, when it has the same 1/n harmonic rolloff as a sawtooth?
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Right. A square is odd harmonics only. The even ones are the octaves and their relatives, and a spectrum with those missing is what "hollow" describes. A saw has every harmonic, which is why it sounds full and bright.
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Both fall off as 1/n. That is exactly what makes this a useful comparison.
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A digital oscillator may be, but an ideal square is hollow for a reason that is in the recipe rather than in the implementation.
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It has half as many, but that is the mechanism rather than the description. Which half is missing is the point.
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3A wave field synthesis array has loudspeakers every 25 cm. Above roughly what frequency does it stop reconstructing the wavefront?
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That is off by the factor of two that Nyquist puts in. Half a wavelength is what has to fit, not a whole one.
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Unlike amplitude panning, the aliasing limit is a property of the array rather than of where anybody is standing.
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It is a spatial sampler, so it has a Nyquist limit in space exactly as sampling has one in time.
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Right. 343 / (2 × 0.25). Above that the array can no longer represent the wavefront and aliases instead. It is the one number that decides what a WFS system costs, because halving the spacing doubles the limit and doubles the loudspeaker count.
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4You need an image that stays put as the audience walks around. Which method is the wrong one to reach for?
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Not elegant, and completely correct. A physical source is at a place for everybody.
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Right. Amplitude panning produces a sweet spot rather than a location: the image is correct where the geometry was drawn around and follows the listener everywhere else. That is precisely the failure mode you cannot accept here.
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WFS reconstructs the wavefront, so the image genuinely is at a place and stays there as you move. It fails by frequency instead, not by position.
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It has its own limits — order, and channel count growing as (n+1)² — but position stability is not the first thing it gives up.
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Answers are kept in this browser on this device. Nothing is sent anywhere.
- Polarity or phase?
- What is a square wave made of?
- VBAP, WFS or ambisonics?
Where to go next
These pages are a chain rather than a library — the next one along usually answers the question this one raised.