ADC/DAC: resolution, LSB and SNR
Quantization and LSB size, where 6.02·N+1.76 comes from, SINAD and ENOB as the real figure of merit, dynamic range and the Nyquist frequency.
An analog-to-digital (ADC) or digital-to-analog (DAC) converter is described by a handful of figures that are easily confused: resolution, SNR, dynamic range, ENOB. This tool computes them from resolution, reference and sample rate; this page explains where they come from and which one to actually watch.
Quantization and the LSB
An N-bit converter splits the full-scale input range V_ref into 2ᴺ equal steps. The smallest step is the Least Significant Bit (LSB): the smallest voltage change the converter can resolve. It is the grid step onto which every sample is rounded.
Each extra bit halves the LSB: at the same V_ref, going from 12 to 16 bits shrinks it 16×. A 16-bit ADC with V_ref = 3.3 V has an LSB of about 50 µV — below that level you need clean references and a low noise floor, otherwise the last bits carry no information.
Ideal SNR: where 6.02·N + 1.76 comes from
Rounding a continuous signal to the grid introduces a quantization error. If the signal is “lively enough” relative to the LSB, this error is well modelled as white noise uniform over ±½ LSB. The variance (power) of a uniform distribution of width LSB is:
A full-scale sine (peak amplitude V_ref/2) has power P_s = (V_ref/2)²/2. The ratio of the two powers, in dB, is the ideal SNR:
The two terms have a precise meaning:
- 6.02 dB/bit = 20·log₁₀(2): each added bit doubles the number of levels and improves the SNR by about 6 dB.
- +1.76 dB = 10·log₁₀(1.5): the shape ratio between a full-scale sine and uniform noise.
It is a theoretical ceiling: no real converter beats it. Known datasheet values: an ideal 12-bit ADC has SNR = 74.0 dB, a 16-bit one = 98.1 dB.
Typical resolutions compared (V_ref = 3.3 V)
| N bits | LSB | ideal SNR |
|---|---|---|
| 8 | 12.9 mV | 49.9 dB |
| 12 | 806 µV | 74.0 dB |
| 16 | 50.4 µV | 98.1 dB |
| 24 | 197 nV | 146.2 dB |
Every 4 bits the ideal SNR rises by ~24 dB and the LSB shrinks 16-fold. The LSB column explains why beyond 16–18 bits the real limit is set by reference and analog front-end noise, not quantization.
Dynamic range
For an ideal converter, limited only by quantization noise, the dynamic range (ratio of the largest representable signal to the noise floor) numerically equals the full-scale SNR. In real devices the two diverge: distortion and spurious tones lower the SNR but not necessarily the dynamic range measured with a small signal. Here we model the ideal case, so dynamic range = ideal SNR.
ENOB and SINAD
A real converter does not reach its theoretical SNR: thermal noise, clock jitter, non-linearity (INL/DNL) and harmonic distortion erode it. The representative measurement is the SINAD (Signal-to-Noise-And-Distortion), whose denominator includes not just noise but all harmonics and spurs.
Inverting the ideal-SNR formula and substituting the measured SINAD yields the effective number of bits, the ENOB:
A “16-bit” ADC with a 90 dB SINAD has an ENOB of about 14.7 bits: the last bits are noise, not information. That is why ENOB — not the nominal resolution — is the figure of merit by which converters are really compared. The difference N − ENOB tells how many bits are “lost” relative to the data sheet.
Aperture jitter, the limit resolution does not tell you about
Among the factors that erode real SNR, aperture jitter — the uncertainty in the instant the sampler closes — deserves a place of its own, because it does not depend on the converter but on the clock, and because its law is surprisingly severe. Sampling a moment too early or too late produces an amplitude error equal to the signal’s slope times the timing error; on a sinusoid of frequency f_in with rms jitter t_j the resulting signal-to-noise ratio is
and it contains no N. It does not depend on resolution: it is a ceiling the clock imposes on any converter. Since f_in enters linearly, the ceiling falls 20 dB per decade of input frequency:
| Jitter | 1 kHz | 10 kHz | 100 kHz | 1 MHz | 10 MHz | 100 MHz |
|---|---|---|---|---|---|---|
| 1 ps | 164 dB | 144 dB | 124 dB | 104 dB | 84 dB | 64 dB |
| 10 ps | 144 dB | 124 dB | 104 dB | 84 dB | 64 dB | 44 dB |
| 100 ps | 124 dB | 104 dB | 84 dB | 64 dB | 44 dB | 24 dB |
| 1 ns | 104 dB | 84 dB | 64 dB | 44 dB | 24 dB | 4 dB |
The chart reads directly: wherever your jitter line passes below the horizontal line of your resolution, the extra bits buy nothing. A 100 ps clock — not a bad clock, for many digital systems — caps you at 64 dB at 1 MHz: less than an 11-bit ADC would give.
How much jitter you can afford
Inverting the formula gives the jitter at which clock noise equals quantisation noise, that is the threshold beyond which the converter starts being clock-limited:
| Resolution | Quantisation SNR | at 100 kHz | at 1 MHz | at 10 MHz |
|---|---|---|---|---|
| 12 bit | 74.0 dB | 317.6 ps | 31.8 ps | 3.2 ps |
| 14 bit | 86.0 dB | 79.8 ps | 8.0 ps | 0.8 ps |
| 16 bit | 98.1 dB | 19.8 ps | 2.0 ps | 0.2 ps |
| 18 bit | 110.1 dB | 5.0 ps | 0.5 ps | 50 fs |
| 24 bit | 146.2 dB | 78 fs | 8 fs | 1 fs |
The 10 MHz column is why 18- and 24-bit converters live at low frequency: demanding fifty femtoseconds of jitter, or worse one, is not a matter of budget but of the physics of the generator and the clock path. A 24-bit part is a DC-instrumentation and audio component, not an RF one — and the formula says so before any data sheet does.
When the two contributions are comparable they add in power. On a 16-bit ADC with a 10 ps clock:
| f_in | Jitter SNR | Total SNR | ENOB |
|---|---|---|---|
| 10 kHz | 124.0 dB | 98.1 dB | 16.0 bit |
| 100 kHz | 104.0 dB | 97.1 dB | 15.8 bit |
| 1 MHz | 84.0 dB | 83.9 dB | 13.6 bit |
| 10 MHz | 64.0 dB | 64.0 dB | 10.3 bit |
At 10 kHz jitter is irrelevant and ENOB stays at 16.0. At 1 MHz it has already eaten 2.4 bits, and at 10 MHz the converter performs like a 10-bit part. None of these figures depends on the quality of the silicon.
Oversampling and the real signal
Two practical corrections to the theoretical ceiling, pulling in opposite directions.
How many bits oversampling buys
Sampling faster than necessary does not reduce total quantisation noise, it spreads it over a wider band: filtering afterwards to the useful band throws the out-of-band part away. The gain is 10·log₁₀(OSR), that is 3 dB — half a bit — per doubling:
| OSR | SNR gain | Equivalent bits |
|---|---|---|
| 2× | +3.0 dB | +0.5 |
| 4× | +6.0 dB | +1.0 |
| 16× | +12.0 dB | +2.0 |
| 64× | +18.1 dB | +3.0 |
| 256× | +24.1 dB | +4.0 |
| 1024× | +30.1 dB | +5.0 |
Four free bits cost a factor of 256 on the sampling rate: a good bargain in the audio band, untenable at high frequencies. What makes it practical is the noise shaping of a sigma-delta modulator, which does not merely spread the noise but pushes it out of band: a first-order modulator gains 9 dB per OSR doubling (1.5 bits), a second-order one 15 dB (2.5 bits), a third-order one 21 dB (3.5 bits). It is why a 1-bit sigma-delta at OSR 64 and second order comfortably beats a 16-bit SAR over the same band.
And if the signal is not full scale
The 6.02·N + 1.76 holds for a sinusoid using all of the scale. Every decibel of headroom left at the top is a decibel of SNR lost, one for one. On a 16-bit ADC:
| Signal level | SNR | ENOB |
|---|---|---|
| +0 dBFS | 98.1 dB | 16.0 bit |
| -6 dBFS | 92.1 dB | 15.0 bit |
| -12 dBFS | 86.1 dB | 14.0 bit |
| -20 dBFS | 78.1 dB | 12.7 bit |
| -40 dBFS | 58.1 dB | 9.4 bit |
| -60 dBFS | 38.1 dB | 6.0 bit |
Twenty decibels of headroom — a prudent and very common choice to never clip — cost 3.3 bits: a 16-bit part used at −20 dBFS performs like a 12.7-bit one at full scale. It is the most underrated trade-off in designing an acquisition chain, and it explains why the analogue front-end’s gain matters as much as the choice of converter.
Nyquist and aliasing
Sampling at rate f_s, only components below half of f_s — the Nyquist frequency — are represented without ambiguity:
Any component — useful signal or noise — above Nyquist folds back (aliases) into the useful band, where it becomes indistinguishable from a legitimate frequency. That is why an anti-aliasing filter always precedes an ADC, attenuating everything above Nyquist before sampling, and oversampling converters use a very high f_s to push the quantization noise out of band.
Limitations
- Ideal-converter model: uniform quantization noise only, no distortion, INL/DNL or jitter. The computed SNR is the theoretical ceiling, not the real performance.
- ENOB requires a measured SINAD on the device: without that figure the tool does not estimate it.
- Full-scale sine test signal: with smaller or non-sinusoidal signals the numbers change.
- Demonstrative tool — it does not replace the characterization of a real converter.
References
- ADC/DAC vendor application notes on the relationship between resolution, SNR, SINAD and ENOB and on converter measurement (IEEE Std 1241 definitions).
- Related tool — the ADC/DAC Resolution & SNR calculator puts this page into practice: enter N, V_ref and f_s and read the LSB, ideal SNR, dynamic range and Nyquist; add a measured SNR/SINAD and get the ENOB.
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