Master'sElectronics, Embedded Systems & Telecom Applications open

Master’s in DSP and Real-Time Signal Processing Engineering

Online master's programme of 1,900 hours in DSP and real-time signal processing engineering. 12 modules covering discrete signals and systems, spectral analysis, FIR and IIR filter design, DSP processor architectures, real-time fixed-point implementation, multirate and adaptive filtering, digital communications, audio and radar processing, and algorithm deployment, with an assigned technical tutor.

Duration1900 h
ModeOnline
LanguageEnglish
Start datesSet by the faculty
AwardSEIUM Institute diploma
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Learning outcomes

What you will be able to do

01

Analyse discrete signals in the time and frequency domains.

02

Design FIR and IIR digital filters for real applications.

03

Implement algorithms on DSP processors and microcontrollers.

04

Manage fixed-point arithmetic and quantisation effects.

05

Develop multirate and adaptive processing systems.

06

Apply signal processing to communications, audio and radar.

07

Optimise algorithms to meet real-time constraints.

Profile

Who it's for

  • Signal processing engineers.
  • Embedded DSP engineers.
  • Digital audio engineers.
  • Digital communications engineers.
  • Algorithm engineers.
  • Radar systems engineers.
  • Engineering or science graduates looking to enter the telecommunications sector.
Standards & tools

What you'll work with

  • IEEE 754 (floating-point arithmetic)
  • ITU-T G.711 (pulse code modulation of voice frequencies)
  • ITU-T G.168 (digital network echo cancellers)
  • ETSI TS 138 211 (5G NR physical channels and modulation)
  • IEEE 802.11 (wireless LAN)
  • AES67 (audio-over-IP interoperability)
  • IEC 61000-4-7 (measurement of harmonics and interharmonics)
  • ISO 26262 (functional safety of road vehicles)
  • MATLAB and Signal Processing Toolbox
  • Simulink and Embedded Coder
  • Python with SciPy and NumPy
  • Code Composer Studio
  • GNU Radio
Syllabus

12 modules · 1900 h

  1. 1.1Sampling, the Nyquist theorem and aliasing
  2. 1.2Quantisation and signal-to-noise ratio
  3. 1.3Linear time-invariant systems
  4. 1.4Discrete convolution and impulse response
  5. 1.5The Z-transform and region of convergence
  6. 1.6Frequency response and Bode plots
  7. 1.7Stability, causality and poles
  8. 1.8Difference equations
  9. 1.9System simulation with SciPy
  10. 1.10Practical exercise: characterising a discrete system in Python
  1. 2.1The discrete Fourier transform
  2. 2.2Cooley-Tukey FFT algorithms
  3. 2.3Windows and spectral leakage
  4. 2.4Frequency resolution and zero padding
  5. 2.5Welch spectral estimation
  6. 2.6Spectrograms and the short-time Fourier transform
  7. 2.7Parametric estimation with AR models
  8. 2.8Harmonic and distortion analysis
  9. 2.9Wavelet transform for non-stationary signals
  10. 2.10Lab: spectral analysis of machine vibrations
  1. 3.1Properties of FIR filters
  2. 3.2Linear phase conditions
  3. 3.3Window method design
  4. 3.4Parks-McClellan optimal design
  5. 3.5Least-squares design
  6. 3.6Direct, transposed and symmetric structures
  7. 3.7Hilbert filters and differentiators
  8. 3.8Fixed-point coefficients
  9. 3.9Filter design tools
  10. 3.10Workshop: band-pass FIR filter in MATLAB
  1. 4.1Butterworth and Chebyshev analogue prototypes
  2. 4.2Elliptic and Bessel filters
  3. 4.3Bilinear transform and pre-warping
  4. 4.4Cascaded second-order sections
  5. 4.5Coefficient sensitivity
  6. 4.6Round-off noise and scaling
  7. 4.7Limit cycles and numerical stability
  8. 4.8Comparison of FIR and IIR
  9. 4.9Parametric equalisation filters
  10. 4.10Practical exercise: fixed-point IIR in biquad sections
  1. 5.1Modified Harvard architecture
  2. 5.2MAC units and extended accumulators
  3. 5.3Circular and bit-reversed addressing
  4. 5.4VLIW and SIMD architectures
  5. 5.5Texas Instruments DSP families
  6. 5.6DSP and Helium extensions in ARM
  7. 5.7Memory hierarchy and DMA
  8. 5.8DSP versus FPGA and GPU
  9. 5.9Platform selection criteria
  10. 5.10Case study: processor selection for audio
  1. 6.1Q format and fixed-point arithmetic
  2. 6.2Scaling, saturation and overflow
  3. 6.3Sample-based and block-based processing
  4. 6.4Ping-pong and circular buffers
  5. 6.5Input-to-output latency
  6. 6.6Optimisation with compiler intrinsics
  7. 6.7Optimised CMSIS-DSP libraries
  8. 6.8Cycle and CPU load measurement
  9. 6.9Debugging real-time algorithms
  10. 6.10Lab: real-time filter on Cortex-M7
  1. 7.1Decimation and anti-aliasing filtering
  2. 7.2Interpolation and spectral images
  3. 7.3Efficient polyphase structures
  4. 7.4Analysis and synthesis filter banks
  5. 7.5CIC integrator-comb filters
  6. 7.6Fractional rate conversion
  7. 7.7Digital frequency conversion in radio
  8. 7.8Resampling in audio systems
  9. 7.9Computational cost and efficiency
  10. 7.10Practical exercise: 48 to 44.1 kHz rate converter
  1. 8.1The Wiener optimal filter
  2. 8.2The LMS algorithm and its variants
  3. 8.3The RLS algorithm and its cost
  4. 8.4Acoustic echo cancellation
  5. 8.5Active noise cancellation
  6. 8.6Adaptive channel equalisation
  7. 8.7Convergence speed and misadjustment
  8. 8.8Efficient implementation on DSPs
  9. 8.9The Kalman filter as an estimator
  10. 8.10Workshop: noise canceller with normalised LMS
  1. 9.1Digital modulations and constellations
  2. 9.2Pulse-shaping filters
  3. 9.3Carrier and symbol synchronisation
  4. 9.4Channel equalisation
  5. 9.5Orthogonal frequency-division multiplexing (OFDM)
  6. 9.6Software-defined radio
  7. 9.75G NR physical layer
  8. 9.8Pilot-based channel estimation
  9. 9.9EVM and BER measurement
  10. 9.10Lab: OFDM receiver in GNU Radio
  1. 10.1Audio processing and equalisers
  2. 10.2Dynamic range compression
  3. 10.3Speech and audio coding
  4. 10.4FMCW radar and range processing
  5. 10.5Doppler estimation and range-velocity maps
  6. 10.6Constant false alarm rate (CFAR) detection
  7. 10.7Beamforming with arrays
  8. 10.8Direction-of-arrival estimation
  9. 10.9Fusion with inertial sensors
  10. 10.10Case study: processing for an automotive FMCW radar
  1. 11.1From MATLAB prototype to C code
  2. 11.2Automatic code generation
  3. 11.3Bit-exact verification against the reference
  4. 11.4Processor-in-the-loop testing
  5. 11.5FPGA implementation with HDL Coder
  6. 11.6Acceleration on embedded GPUs
  7. 11.7Energy consumption of processing
  8. 11.8Functional safety requirements
  9. 11.9Documentation and traceability
  10. 11.10Practical exercise: code generation from Simulink
  1. 12.1Defining the processing application
  2. 12.2Analysis of requirements and signals
  3. 12.3Floating-point algorithm design
  4. 12.4Conversion to fixed point
  5. 12.5Implementation on a real-time platform
  6. 12.6Performance optimisation
  7. 12.7Experimental validation with real signals
  8. 12.8Conclusions and future work
  9. 12.9Master's Thesis report
  10. 12.10Master's Thesis: defence before a panel
Methodology, software & labs

How you'll learn

Training is 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 each intake, so the programme is compatible with work.

Each module combines theory, guided reading of the original standard and applied exercises. The working software is MATLAB and Signal Processing Toolbox, Simulink and Embedded Coder, Python with SciPy and NumPy, Code Composer Studio and GNU Radio, in free, academic or evaluation versions. Exercises are based on real procedures for designing and implementing digital filters, real-time processing on DSPs and Cortex-M, and spectral analysis of vibration and audio signals. Assessment is continuous per module, plus the final Master's Thesis.

Standards covered: IEEE 754, ITU-T G.711, ITU-T G.168, ETSI TS 138 211, IEEE 802.11, AES67, IEC 61000-4-7 and ISO 26262.

Assigned technical tutor:every module ends with a deliverable that your tutor marks individually.
Standards-based learning:guided reading of IEEE 754, ITU-T G.711 and ITU-T G.168 applied to real cases, not as a theoretical annex.
Industry software:MATLAB and Signal Processing Toolbox, Simulink and Embedded Coder, Python with SciPy and NumPy, Code Composer Studio and GNU Radio, in free, academic or evaluation versions.
Master's Thesis with defence:a complete real-time signal processing application, from floating-point design and fixed-point conversion to implementation and validation with real signals, defended before a panel.
Flexible format:100% online, no fixed timetables, compatible with full-time work.
Capstone project

Build it end to end

Your final project follows the real engineering workflow, step by step, reviewed by your tutor.

Spectral analysis
Analyse machine vibration signals with FFT, windows and Welch estimation.
Digital filter design
Design FIR and fixed-point IIR filters and compare their performance.
Real-time implementation
Implement a filter in real time on a Cortex-M7 using CMSIS-DSP and measure CPU load.
Adaptive noise canceller
Build a noise canceller with normalised LMS and evaluate its convergence.
Master's Thesis
A complete real-time DSP application validated with real signals, defended before a panel.
Career opportunities

Where this takes you

Signal processing engineerEmbedded DSP engineerDigital audio engineerDigital communications engineerAlgorithm engineerRadar systems engineer
Admissions & scholarships

Requirements

  • University degree in engineering, science or architecture, or equivalent demonstrable professional experience.
  • An internet connection and a computer able to run 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.

FAQ

Frequently asked questions

The programme totals 1,900 hours over 12–14 months. You set your own pace within each intake, and your tutor helps you plan the workload around your job.

You work with MATLAB and Signal Processing Toolbox, Simulink and Embedded Coder, Python with SciPy and NumPy, Code Composer Studio and GNU Radio, 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 Master's Thesis. 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 in a course or postgraduate diploma in the same area are recognised in the higher programme. Ask about the recognition policy when you enrol.

A university degree in engineering, science or architecture (or equivalent professional experience), a computer able to run the 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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