Metering, Levels, and Phase

Types of Meters

Peak Meters

Show the instantaneous maximum level of the signal. Fast response — they catch transients and brief spikes that your ears might not register as loud. Most DAW meters are peak meters. When the meter hits 0 dBFS (decibels full scale), you’re clipping. In digital audio, clipping is absolute — there’s nothing above 0 dBFS, and anything that tries to go there gets chopped flat. It sounds harsh and it’s permanent in the recording.

RMS Meters

Show the average power of the signal over a short time window. Closer to how your ears actually perceive loudness — a sustained chord feels louder than a sharp snare hit at the same peak level, and an RMS meter reflects that. These meters move more slowly and smoothly than peak meters, which makes them better for gauging how “loud” something feels rather than how high its peaks spike.

LUFS Meters

Loudness Units Full Scale — the current standard for measuring perceived loudness. LUFS weights the frequency spectrum to match human hearing sensitivity (we’re most sensitive to midrange), then averages over time. Streaming platforms normalize to LUFS targets: roughly -14 for Spotify, -16 for Apple Music. Essential for mastering, less relevant during tracking and recording.

VU Meters

The analog-era meter. Slow response time, designed to approximate how loud something sounds rather than how high it peaks. Calibrated so that 0 VU corresponds to +4 dBu (professional line level). Still found on outboard hardware and loved by engineers who came up on analog consoles. Not common in software, but useful to understand because the concept — metering that tracks perceived loudness — is the ancestor of LUFS.

Gain Structure Through the Chain

Every stage of the signal chain has an ideal operating level. Too quiet and the noise floor rises relative to your signal; too loud and you get distortion. Gain staging is about keeping the signal comfortably within the usable range at every point:

1. Mic → Preamp: Set gain so the loudest peaks hit -12 to -6 dBFS. This leaves headroom for unexpected transients.
2. A/D converter: Already handled if the preamp gain is set correctly — the preamp feeds the converter directly.
3. DAW channels → plugins: Most plugins are modeled to behave best with input levels around -18 dBFS (roughly equivalent to 0 VU in the analog world). Hitting them much harder can cause them to clip or behave unpredictably. Too quiet and you’re wasting their usable range.
4. Mix bus → master output: Leave headroom for mastering. Peaks around -6 to -3 dBFS on your mix bus give a mastering engineer room to work.

None of these numbers are sacred — they’re starting points. The principle is to leave yourself room at every stage so problems don’t compound downstream.

Phase

Any time two microphones pick up the same source, you have a phase relationship to manage. When two waveforms are perfectly in phase, their peaks and troughs line up — they reinforce each other and the combined signal is louder. When one is inverted (180° out of phase), the peaks of one align with the troughs of the other — they cancel.

Complete cancellation is rare in practice. More common is partial phase cancellation — certain frequencies cancel while others reinforce, producing a hollow, thin, flanged character. This happens whenever two mics pick up the same source from different distances.

Polarity vs. Phase

These get used interchangeably, but they’re different:

Polarity is a binary flip — the signal is inverted. Positive becomes negative, negative becomes positive. The polarity button on your preamp or DAW does this. It’s either flipped or it’s not.

Phase is a continuous relationship — how far one signal is shifted in time relative to another. Moving a microphone one inch changes the phase relationship differently at every frequency. Low frequencies (long wavelengths) are barely affected; high frequencies (short wavelengths) shift dramatically.

Polarity problems are easy to fix — press a button. Phase problems require either repositioning microphones or using time-alignment tools in the DAW.

Comb Filtering

When two signals of the same source arrive with a small time offset, they create alternating bands of cancellation and reinforcement across the frequency spectrum. Plot the resulting frequency response and it looks like the teeth of a comb — hence the name. Comb filtering is the most common phase artifact in multi-mic recordings.

The 3:1 Rule

Every textbook calls it a rule. It’s more like a napkin sketch — a starting point that keeps you out of trouble when you’re setting up multiple mics.

The idea: when two microphones are recording two separate sources, place the mics at least three times farther from each other than each mic is from its own source. If a vocal mic is 6 inches from the singer and a guitar mic is 6 inches from the amp, the two mics should be at least 18 inches apart. The goal is to reduce bleed — when one source leaks into the other’s microphone at a level and delay that causes comb filtering.

Why does distance help? Because sound loses energy as it travels. If the bleed arriving at a mic is significantly quieter than the direct source, any phase cancellation it causes will be too faint to matter. The 3:1 ratio creates roughly a 10 dB level difference between direct sound and bleed — enough that the interference becomes negligible in most situations.

Talk to enough engineers and you’ll hear three or four versions of this guideline. Sometimes it’s applied to two mics on one source (like a snare top and bottom). Sometimes the ratio changes to 4:1 or 5:1 for picky situations. The math is less important than the principle: more distance between mics means less destructive interference. And no ratio replaces checking with your ears — flip polarity on one channel, listen for the fuller sound, and adjust mic positions until the phase relationship works. That polarity check from Chapter 10 should be automatic on every multi-mic session.

What to Practice

1. Watch your meters while recording. Pay attention to peak levels as you track. Get comfortable with signals peaking around -12 to -6 dBFS. Resist the urge to push them higher — headroom is free in 24-bit.
2. Compare peak and RMS. Pull up a peak meter and an RMS meter on the same track. Play a snare-heavy drum loop — notice the peak meter jumps much higher than the RMS. Now play a sustained pad — the two meters read closer together. That gap between peak and RMS is the difference between transient level and perceived loudness.
3. Hear comb filtering. Set up two mics on the same source (two mics on a guitar amp, or two mics on a voice). Record both simultaneously. In the DAW, nudge one track forward or backward in time by small increments — 1 ms, 2 ms, 5 ms. Listen to how the tone changes. That hollow, phased sound is comb filtering.
4. Use the polarity button. With two mics on the same source summed to one output, flip the polarity on one channel. If it sounds fuller, leave it flipped — the mics were partially out of phase. If it sounds thinner, flip it back. This should be automatic on every multi-mic session.