Calculate ADC resolution, LSB voltage, quantization steps, and dynamic range for analog-to-digital converters.
This ADC calculator computes every key parameter of an analog-to-digital converter — LSB step size, digital output code (decimal, binary, hex), quantization error, SNR, ENOB, and dynamic range — for any bit width from 4 to 32 bits and any reference voltage. A live transfer characteristic curve updates in real time as you adjust parameters, making it easy to visualise how resolution, range, and error trade off in your design.
| Parameter | Formula | Unit |
|---|---|---|
| LSB step size | LSB = Vref / (2N − 1) | V (or mV) |
| Quantization levels | 2N | — |
| Digital output code | Dout = round(Vin / LSB) | integer |
| Quantization error | ±0.5 × LSB (max) | V |
| SNR (ideal) | 6.02 × N + 1.76 | dB |
| ENOB | (SINAD − 1.76) / 6.02 | bits |
| Dynamic range | 20 × log10(2N − 1) | dB |
| Bits (N) | Levels | LSB @ 3.3 V | LSB @ 5 V | Ideal SNR | Typical Use |
|---|---|---|---|---|---|
| 8 | 256 | 12.94 mV | 19.61 mV | 49.9 dB | Audio, basic sensors |
| 10 | 1 024 | 3.23 mV | 4.89 mV | 62.0 dB | Arduino, general MCU |
| 12 | 4 096 | 0.806 mV | 1.22 mV | 74.0 dB | STM32, ESP32, precision |
| 14 | 16 384 | 0.201 mV | 0.305 mV | 86.1 dB | Instrumentation |
| 16 | 65 536 | 50.4 µV | 76.3 µV | 98.1 dB | Audio DAQ, medical |
| 24 | 16.7M | 0.197 µV | 0.298 µV | 146.2 dB | Precision measurement |
| Type | Speed | Resolution | Typical Use |
|---|---|---|---|
| Flash | Very fast (> 1 GSPS) | 4–8 bits | Oscilloscopes, radar |
| SAR | Medium (10 kSPS–10 MSPS) | 8–18 bits | MCU built-in, DAQ |
| Delta-Sigma (ΔΣ) | Slow (10–10k SPS) | 16–32 bits | Precision measurement, audio |
| Pipeline | Fast (10–500 MSPS) | 10–16 bits | Communications, video |
| Dual-Slope | Very slow | 12–22 bits | DMM, integrating meters |
It means the ADC can distinguish 4096 (212) voltage levels. With 3.3 V Vref, each step is ~0.8 mV — enough for most sensor and control applications.
Use a clean Vref, minimize noise on the analog input, add an anti-aliasing filter, oversample and average, and calibrate offset and gain errors.
Half the sampling rate: fNyquist = fsample / 2. Signals above this frequency alias into the baseband. Always filter before sampling.