Diploma in Digital Filters, FFT and Spectral
Online postgraduate diploma of 800 hours in Digital Filters, FFT and Spectral Analysis: 8 modules covering discrete signals and systems, FIR and IIR filter design, the DFT and FFT, spectral estimation, multirate and adaptive processing, and engineering applications, with an assigned technical tutor.
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What you will be able to do
Analyse discrete systems using the Z-transform and frequency response.
Design FIR filters with window and optimal methods.
Design stable IIR filters in second-order structures.
Implement the FFT and fast convolution efficiently.
Estimate power spectra with non-parametric and parametric methods.
Apply multirate and adaptive techniques to signal processing.
Solve real measurement and diagnostic problems with spectral analysis.
Who it's for
- Signal processing engineers who want a solid grounding in filter design and spectral methods.
- Instrumentation engineers measuring and characterising signals and converters.
- Communications systems engineers working on channelisation and sample rate conversion.
- Predictive maintenance specialists analysing machine vibration.
- Digital audio engineers designing filters and rate converters.
- DSP algorithm developers implementing processing chains on DSP or FPGA.
- Engineering or science graduates looking to enter the telecommunications sector.
What you'll work with
- IEEE 754-2019 (floating-point arithmetic)
- IEEE 1241-2010 (terminology and test methods for analogue-to-digital converters)
- IEEE 1057-2017 (digitising waveform recorders)
- IEC 61000-4-7:2002 (harmonics and interharmonics measurement)
- IEC 61260-1:2014 (octave-band and fractional-octave-band filters)
- ISO 18431-2:2004 (signal processing for mechanical vibration and shock: windows for Fourier transform analysis)
- IEC 61672-1:2013 (sound level meters)
- MATLAB (Signal Processing Toolbox)
- Python (SciPy and NumPy)
- GNU Octave
- pyFDA
- LabVIEW
8 modules
1.1Sampling, aliasing and the Nyquist theorem1.2Sequences and LTI systems1.3Discrete convolution and correlation1.4Z-transform and regions of convergence1.5Poles, zeros and stability1.6Frequency response and the DTFT1.7Difference equations1.8Quantisation and quantisation noise1.9Fixed-point and floating-point representation1.10Lab: analysis of LTI systems in Python
2.1Linear phase and coefficient symmetry2.2Window method: Hamming, Kaiser and Blackman2.3Parks-McClellan optimal design2.4Least squares and multiband filters2.5Differentiator and Hilbert filters2.6Ripple and attenuation specifications2.7Direct and transposed structures2.8Effects of coefficient quantisation2.9Moving-average and CIC filters2.10Practical: band-pass filter with Parks-McClellan
3.1Butterworth, Chebyshev and elliptic prototypes3.2Bilinear transform and prewarping3.3Impulse invariance method3.4Cascaded second-order sections3.5Stability and numerical sensitivity3.6Limit cycles and overflow3.7Notch filters and resonators3.8Comparison of FIR and IIR3.9Minimum phase and group delay3.10Lab: elliptic filter in biquad sections
4.1DFT: definition and properties4.2Radix-2 decimation-in-time algorithm4.3Radix-4, split-radix and arbitrary sizes4.4Real FFT and data packing4.5Fast convolution: overlap-add and overlap-save4.6Computational cost and memory4.7FFTW and CMSIS-DSP libraries4.8Numerical precision of the FFT4.9Goertzel algorithm for tone detection4.10Practical: fast convolution with overlap-save
5.1Spectral leakage and windows5.2Frequency resolution and zero-padding5.3Periodogram and Welch's method5.4Power spectral density5.5Spectrogram and STFT5.6Parametric estimation: AR and Burg5.7Subspace methods: MUSIC and ESPRIT5.8Measurement of SNR, SINAD and THD5.9Amplitude calibration and scaling5.10Lab: vibration spectrum of a machine
6.1Decimation and interpolation6.2Polyphase filters and their efficiency6.3Fractional sample rate conversion6.4Filter banks and QMF6.5Discrete wavelet transform6.6LMS and RLS adaptive filters6.7Envelope detection and demodulation6.8Real-time block processing6.9Latency and buffer size6.10Workshop: rate conversion from 48 to 44.1 kHz
7.1Harmonic analysis according to IEC 61000-4-77.2Vibration-based predictive maintenance7.3Biomedical signal processing7.4Radar and Doppler frequency estimation7.5Acoustics: octave and third-octave bands7.6Characterisation of ADCs and instrumentation7.7Digital channelisation in communications7.8Implementation on DSP and FPGA7.9Validation with reference signals7.10Case study: bearing fault detection
8.1Definition of the application and requirements8.2Signal acquisition and characterisation8.3Filter design and comparison8.4Spectral analysis and metrics8.5Real-time implementation8.6Verification and testing8.7Documentation of results8.8Discussion of limitations and improvements8.9Report and presentation8.10Final project: final review with your tutor
How you'll learn
The programme is delivered 100% online on the SEIUM campus, with downloadable material, recorded classes and an assigned technical tutor. There are no fixed timetables: you progress at your own pace within the intake, so you can combine it with your job.
Each module combines theory, guided reading of the original standard and applied exercises. The working software is MATLAB (Signal Processing Toolbox), Python (SciPy and NumPy), GNU Octave, pyFDA and LabVIEW, in free, academic or evaluation versions. Testing is studied through real procedures: filter design and simulation in Python, machinery vibration analysis and harmonic measurement in power networks. Assessment is continuous, module by module, plus the final project.
Standards covered: IEEE 754-2019, IEEE 1241-2010, IEEE 1057-2017, IEC 61000-4-7:2002, IEC 61260-1:2014, ISO 18431-2:2004 and IEC 61672-1:2013.
Build it end to end
Your final project follows the real engineering workflow, step by step, reviewed by your tutor.
Where this takes you
Requirements
- Prior technical training or professional experience in the sector. No university degree is required.
- An internet connection and a computer capable of running the programme software.
- English sufficient to read technical standards.
Ready to start?
Send your application today. Our admissions team replies within 72 hours with your study plan and funding options.
Frequently asked questions
The programme totals 800 hours over 6–8 months. You set your own pace within each intake, and your tutor helps you plan the workload around your job.
You work with MATLAB (Signal Processing Toolbox), Python (SciPy and NumPy), GNU Octave, pyFDA and LabVIEW, using free, academic or evaluation versions. Your tutor indicates the free alternative wherever one exists.
A SEIUM Institute diploma, awarded on passing the continuous assessment and the final project. It is SEIUM's own qualification, not an official state degree; its recognition depends on the programme completed and the regulations of each country.
Yes. Modules passed are recognised in a higher programme in the same area (postgraduate diploma or master's). Ask about the recognition policy when you enrol.
Prior technical training or professional experience in the sector (no university degree required), a computer able to run the programme software and English sufficient to read technical standards.
Start dates are set by SEIUM's faculty for each intake. Our admissions team will confirm the next available date when you apply.
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