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Signal Processing & DSP

Turning signals into data: sampling, Nyquist, FFT bins, SNR, filters and quantisation

Signal Processing & DSP formulas help you turn real-world signals into usable data, from sampling and Nyquist limits to FFT bins, filters, quantisation, and decibels. This category is useful for students, exam revision, and engineers working with audio, communications, and electronics, because these equations explain how signals are captured, analysed, distorted, and improved in practical systems.

Nyquist Rate

NEWIntermediate
fs≥2fmaxf_s \ge 2 f_{max}

Sample at least twice the highest frequency to avoid aliasing.

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FFT Bin Frequency

NEWAdvanced
fk=kfsNf_k = \frac{k f_s}{N}

Frequency represented by the k-th bin of an N-point FFT.

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Frequency Resolution

NEWIntermediate
Δf=fsN\Delta f = \frac{f_s}{N}

Spacing between adjacent FFT bins.

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Signal-to-Noise Ratio

NEWBasic
SNR=10log⁡10 ⁣(PsPn)SNR = 10\log_{10}\!\left(\frac{P_s}{P_n}\right)

Ratio of signal power to noise power, in decibels.

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Quantisation SNR

NEWAdvanced
SNR≈6.02b+1.76SNR \approx 6.02 b + 1.76

Best possible SNR (dB) of an ideal b-bit converter.

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Decibel (Power Ratio)

NEWBasic
L=10log⁡10 ⁣(P2P1)L = 10\log_{10}\!\left(\frac{P_2}{P_1}\right)

Express a power ratio in decibels.

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Decibel (Amplitude Ratio)

NEWBasic
L=20log⁡10 ⁣(A2A1)L = 20\log_{10}\!\left(\frac{A_2}{A_1}\right)

Express a voltage/amplitude ratio in decibels.

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RC Filter Cutoff

NEWIntermediate
fc=12πRCf_c = \frac{1}{2\pi RC}

Corner frequency of a simple RC low/high-pass filter.

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RC Time Constant

NEWBasic
τ=RC\tau = RC

Time for an RC circuit to charge to ~63% of final value.

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Normalised Sinc Function

NEWAdvanced
sinc(x)=sin⁡(πx)πx\text{sinc}(x) = \frac{\sin(\pi x)}{\pi x}

The ideal interpolation kernel for reconstructing sampled signals.

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Digital Audio Bit Rate

NEWBasic
R=fs×b×cR = f_s \times b \times c

Raw data rate from sample rate, bit depth and channels.

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Aliased Frequency

NEWAdvanced
fa=∣f−nfs∣f_a = |f - n f_s|

Frequency an under-sampled tone masquerades as.

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Discrete Convolution

NEWAdvanced
y[n]=∑kx[k]h[n−k]y[n] = \sum_k x[k] h[n-k]

Output of a filter is the convolution of input and impulse response.

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Decibels to Linear Gain

NEWBasic
G=10L/20G = 10^{L/20}

Convert a decibel gain back to a linear amplitude factor.

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Total Harmonic Distortion

NEWAdvanced
THD=∑n≥2Vn2V1THD = \frac{\sqrt{\sum_{n\ge2} V_n^2}}{V_1}

Ratio of harmonic energy to the fundamental.

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