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Signal Theory

Quantisation: noise, 6 dB per bit and sigma-delta converters

The uniform quantiser, quantisation noise power and SNR with proof, effective number of bits, μ-law and A-law, dithering, oversampling and noise shaping.

Sampling discretises time; to store a sample in a finite number of bits the amplitude must be discretised too. This is quantisation, and unlike sampling it always loses something: the rounding error. In this lesson we see how large this error is, why each extra bit gains 6 dB and how modern converters do better with processing tricks.

The uniform quantiser

An N-bit analogue-to-digital converter (ADC) divides the input range, of total width V_{FS} (full scale), into 2^N levels spaced

    \[ \Delta = \frac{V_{FS}}{2^N} \]

and assigns each sample the nearest level. The input-output characteristic is a staircase, and the quantisation error e = Q(x) - x is a sawtooth between -\Delta/2 and \Delta/2.

Characteristic of a uniform quantiser and quantisation error
On the left the quantiser output as a function of the input: a staircase with step Δ following the dashed line. On the right the error, the difference between output and input: a sawtooth swinging between minus Δ/2 and plus Δ/2.

Example. A 12-bit ADC with a 3.3 V full scale has \Delta = 3.3/4096 \approx 0.8 mV: this is the resolution with which a microcontroller reads a sensor.

The error power

If the signal is complex enough and spans many levels, the error behaves like noise with a uniform distribution between -\Delta/2 and \Delta/2, independent of the signal.

Theorem. The quantisation noise power is

    \[ \sigma_e^2 = \frac{\Delta^2}{12} \]

Proof. With uniform density 1/\Delta over the interval, the mean is zero and

    \[ \sigma_e^2 = \int_{-\Delta/2}^{\Delta/2} e^2\,\frac{1}{\Delta}\,de = \frac{1}{\Delta}\left[\frac{e^3}{3}\right]_{-\Delta/2}^{\Delta/2} = \frac{1}{\Delta}\cdot\frac{2}{3}\cdot\frac{\Delta^3}{8} = \frac{\Delta^2}{12} \]

∎

Signal-to-noise ratio

Theorem. For a sinusoid spanning the whole full scale, the quantisation signal-to-noise ratio of an N-bit converter is

    \[ \mathrm{SNR} = 6.02\,N + 1.76\ \text{dB} \]

Proof. The sinusoid has amplitude A = V_{FS}/2 = 2^N\Delta/2 and power A^2/2 = 2^{2N}\Delta^2/8. Dividing by \Delta^2/12:

    \[ \mathrm{SNR} = \frac{2^{2N}\Delta^2/8}{\Delta^2/12} = \frac32\cdot 2^{2N} \]

In decibels, 10\log_{10}\frac32 + 2N\cdot 10\log_{10}2 = 1.76 + 6.02\,N. ∎

Each extra bit is worth about 6 dB, that is it halves the error amplitude.

BitsTheoretical SNRTypical use
849.9 dBvideo, simple sensors
1274.0 dBmicrocontrollers, instruments
1698.1 dBaudio CD
24146.2 dBprofessional recording

Real converters do not reach these values, because the noise and nonlinearities of the analogue circuits add up. So the effective number of bits (ENOB) is defined from the measured SNR: a “24-bit” converter with 90 dB of measured SNR has about (90 - 1.76)/6.02 \approx 14.7 effective bits.

The signal must also use the full scale: a sinusoid using only a tenth of the range loses 20 dB of SNR. Exceeding full scale, on the other hand, causes clipping, far more annoying than noise.

Non-uniform quantisation

Speech has a widely varying amplitude: soft syllables would fall into few levels and be very noisy. Digital telephony therefore uses logarithmic quantisation (μ-law in America and Japan, A-law in Europe): small steps for weak signals, large ones for strong signals, so that SNR stays roughly constant. With 8 bits at 8 kHz you get the classic 64 kbit/s telephone channel.

Dithering

With very weak signals, varying by less than one level, the error is no longer random but follows the signal: it shows up as harmonic distortion, very audible. Adding a small random noise before quantising (the dither, about one level in amplitude), the error becomes uniform, independent noise again. It seems paradoxical to add noise to improve the sound, but a steady hiss is much less annoying than distortion.

Oversampling and noise shaping

The quantisation noise power \Delta^2/12 is spread uniformly between 0 and f_s/2. Sampling at a rate M times higher than necessary and then digitally filtering only the useful band removes a fraction (M - 1)/M of the noise: SNR improves by 10\log_{10}M dB. With M = 4 you gain 6 dB, one bit.

Sigma-delta converters take the idea to the extreme: they use a quantiser of a few bits (even just one) at a very high rate, inside a feedback loop that “pushes” quantisation noise towards high frequencies (noise shaping), where the digital filter removes it. This gives 20 bits and more with simple analogue circuits: it is the technique of almost all audio and precision measurement converters. For fast signals successive approximation (SAR) converters are used instead, up to a few tens of megasamples per second, and flash or pipeline converters for the highest frequencies.

How much data. A CD holds 44\,100 samples per second, at 16 bits, for two channels: 1.41 Mbit/s. Compressed formats such as MP3 and AAC exist to reduce this stream, by removing what the ear does not perceive.

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