Open any audio tool and you see decibels: on a mixer's faders, on a spectrogram's colour bar, in a codec's SNR, in a phone's “prediction gain”. The decibel is not a unit of anything. It is a way of writing ratios, a logarithmic ruler. This page explains why that ruler fits audio so well, from three points of view: the ear (loudness), the spectrum (seeing quiet detail next to loud peaks), and the engineer (errors, gains and budgets). It ends with the traps that catch everybody once.
Ask “how much louder is a rock concert than a whisper?” and the honest answer is a ratio: the concert's sound pressure is about 10 000 times larger, and its power about 100 million times larger. Ratios like these are awkward to write, compare and add. The decibel takes the logarithm of the ratio, so that multiplying becomes adding:
Audio signals are usually amplitudes (a pressure, a voltage, a sample value), not powers. Power grows with the square of amplitude, \(P \propto A^2\), so the same ratio written with amplitudes needs a 20:
| change | power ratio | amplitude ratio |
|---|---|---|
| +3 dB | \(\times 2\) | \(\times 1.41\) |
| +6 dB | \(\times 4\) | \(\times 2\) |
| +10 dB | \(\times 10\) | \(\times 3.16\) |
| +20 dB | \(\times 100\) | \(\times 10\) |
| +60 dB | \(\times 1\,000\,000\) | \(\times 1000\) |
| 0 dB | \(\times 1\) (no change) | \(\times 1\) |
| −6 dB | \(\times \tfrac14\) | \(\times \tfrac12\) |
Three properties make this ruler useful, and each of the next three sections leans on one of them:
A factor of a trillion in power is just 120 dB. Tiny and enormous values fit on one axis.
\(\log(ab) = \log a + \log b\). Gains in a chain, and source \(\times\) filter, simply add.
Going from 1 to 2 and from 100 to 200 are both +3 dB (in power). That matches how we perceive change.
The quietest sound a young listener can hear at 1 kHz has a pressure of about \(20\ \mu\text{Pa}\), a fluctuation of two ten-billionths of atmospheric pressure. The threshold of pain is around \(20\ \text{Pa}\). That is a factor of a million in pressure (a million million in power), and the ear handles all of it. On a linear axis almost every everyday sound is crushed against zero:

In the 1830s Ernst Weber noticed that the smallest change in a stimulus that people can detect is a fixed fraction of the stimulus, not a fixed amount. Gustav Fechner turned this into a law: if every just-noticeable step is the same ratio, then perceived magnitude grows with the logarithm of the stimulus. For sound intensity, the just-noticeable change is roughly 1 dB (about 0.5–1 dB for broadband sounds at comfortable levels) whether the sound is quiet or loud. A decibel scale therefore makes equal steps on paper close to equal steps for the ear.
Later, Stanley Smith Stevens measured loudness directly by asking listeners to judge “twice as loud”. He found loudness grows roughly as a power law of intensity, \(N \propto I^{0.3}\). In decibels that has a neat consequence:
Start with your volume low and raise it until the 0 dB version is comfortable. All the other buttons are quieter.
Each staircase plays five short bursts of the phrase, getting louder. Linear adds the same amount of amplitude each time (0.2, 0.4, 0.6, 0.8, 1.0); decibel multiplies by the same factor each time (+6 dB, so each burst is twice the amplitude of the last). Which one sounds like an even climb?
The linear staircase makes a big jump at the start (+6 dB) and then almost stalls (+1.9 dB at the end), while the decibel staircase climbs evenly. This is the practical reason why every volume knob, fader and gain control in audio is marked in dB, and why loudness is quoted on a decibel scale.
A decibel value needs a reference. In audio you will meet several, and the suffix tells you which:
| written | reference (0 dB) | where you see it |
|---|---|---|
| dB SPL | \(20\ \mu\text{Pa}\) sound pressure, roughly the threshold of hearing | acoustics, noise regulations, hearing safety |
| dB(A) | dB SPL after an “A-weighting” filter that reduces low and very high frequencies, as the ear does at moderate levels | noise meters, workplace and traffic limits |
| dB HL | the average normal hearing threshold at each frequency | audiograms (hearing tests) |
| dBFS | digital full scale: the largest value the samples can hold | recording software, codecs; always \(\le 0\) for a clean signal |
| dB (plain) | another signal: the input, the noise, the original | gain, attenuation, SNR, prediction gain |
A voiced speech spectrum has strong low harmonics and weak high formants. The difference is easily 40–60 dB, a factor of 100–1000 in amplitude. On a linear axis the weak parts are invisible, even though the ear hears them clearly and they carry much of the information (F2 and F3 tell vowels apart, and fricatives live up there).

The same applies to a spectrogram, which is just many spectra side by side. Every spectrogram you have seen in this course was drawn in dB:

In module 3 speech was a source passed through a filter, and in the frequency domain a filter multiplies: \(X(f) = G(f)\,H(f)\). Take \(20\log_{10}\) of the magnitudes and the multiplication becomes an addition:

This additive picture is behind several things you have met or will meet:
When a signal passes through a microphone, an amplifier, a codec and a network, every stage adds a little noise and scales the signal up or down. Engineers describe both with decibels.
Reading it from the error's point of view: an SNR of \(S\) dB means the error's amplitude is \(10^{-S/20}\) of the signal's.

White noise is added over the whole recording, scaled so that \(10\log_{10}(\text{total speech energy} / \text{total noise energy})\) equals the chosen SNR. The noise is louder than it sounds on paper: start with your volume low.
One more bit per sample doubles the number of quantiser levels, which halves the step and so halves the error amplitude. Halving an amplitude is always \(20\log_{10}2 = 6.02\) dB, whatever the signal level, so every bit is worth the same 6 dB of SNR (module 6). For the same reason a predictor with gain \(G_p\) dB makes the values to be coded \(G_p\) dB smaller, which saves about \(G_p/6\) bits. In dB, both statements are simple additions; written as ratios they would be powers of 2 and 10.
A typical recording chain, as ratios and in dB:
| stage | amplitude ratio | in dB |
|---|---|---|
| microphone preamplifier | \(\times 100\) | +40 dB |
| long cable | \(\times 0.89\) | −1 dB |
| anti-aliasing filter (pass-band) | \(\times 0.71\) | −3 dB |
| digital gain | \(\times 2\) | +6 dB |
| total | \(100 \times 0.89 \times 0.71 \times 2 \approx 126\) | +42 dB |
Mental arithmetic in the right-hand column, a calculator in the middle one. This is why loss and gain specifications, filter responses (“−3 dB cut-off”, “60 dB stop-band attenuation”) and link budgets are all written in dB.
Speech is loud in vowels and near silent in pauses, so the same steady noise is far below the speech in some places and above it in others. A single global SNR hides that, which is why module 6 reported the segmental SNR, the average of per-frame SNRs in dB:

A bare decibel number is a ratio with its reference missing. 60 dB SPL is conversation; 60 dB of SNR is a very clean recording; −60 dBFS is a very quiet digital signal. Always say, or ask, what the reference is.
Two people each talking at 60 dB SPL do not make 120 dB. Decibels are logarithms, so you must convert back, add, and convert again. For two independent (uncorrelated) sounds the powers add:
A useful consequence: a source 10 dB weaker than another raises the total by only 0.4 dB, and one 20 dB weaker by only 0.04 dB. The loudest source dominates, which is why reducing a weak noise source when a strong one is present achieves almost nothing.
Averaging dB values gives the log of the geometric mean, not of the arithmetic mean. Take two frames with the same noise power \(N\), one with SNR 40 dB (signal \(10^4 N\)) and one with 0 dB (signal \(N\)):
Neither is wrong: they answer different questions (Figure 6). But you must say which you computed, and never average decibels when you meant to average power.
\(\log_{10} 0 = -\infty\). Digital silence, an empty FFT bin or a perfect prediction gives \(-\infty\) dB, which breaks plots and averages. Code therefore always adds a floor, for example
20*log10(abs(X) + 1e-10), or clips at a chosen range (the “80 dB” of Figure 3). Similarly, there is no decibel of a negative number: take the magnitude or the energy first.
Amplitude: \(10^{-12/20} \approx 0.25\) (a quarter). Power: \(10^{-12/10} \approx 0.063\) (a sixteenth). Same physical change, same 12 dB.
−18 dB is about three halvings of amplitude (\(3 \times 6.02\) dB), so the top 3 bits are never used: the recording behaves like a 13-bit one, about 18 dB worse SNR than it could have had.
53 dB(A) (independent sources: powers add, +3 dB). By the “+10 dB \(\approx\) twice as loud” rule, +3 dB is a clearly noticeable but modest increase, not a doubling of loudness.
Because in dB the speech spectrum is the source spectrum plus the envelope (Figure 4), so the envelope's shape can be read off directly. On a linear axis the higher formants of the envelope would be invisible, as in Figure 2.
The loud frames are coded much more accurately (relative to their level) than the quiet ones. The global figure is dominated by the loud vowels; the segmental figure gives the quiet frames an equal vote and exposes their poorer SNR.