Who we are

Babcock Training Academy

We deliver world-class training in complex systems to AUS/NZ defence, government and high tech markets

Office Address

Babcock Training Academy
Level 10 70 Franklin Street Adelaide  SA  5000

Contact Details

Training Project Support
T:   +61 (0)8 8440 9507
F:   +61 (0)8 8312 3227
This email address is being protected from spambots. You need JavaScript enabled to view it.


Advanced Topics in Digital and Multirate Signal Processing for Comms Systems

Two popular Digital Processing courses, condensed and combined into an intensive 5 day program.

Advanced Topics in Digital Signal Processing

Multirate Signal Processing for Communications Systems

Target Audience

The course assumes students have a basic understanding of maths and awareness of digital signal processing and communication systems. Concepts are reviewed as a refresher for all students to support the transition towards new concepts introduced in the course. Whilst mathematics are involved they are integrated into the course intuitively; allowing students to learn how theory is applied in a practical context.

Course Outline

Advanced Topics in Digital Signal Processing

This 4 day course delivered by Professor Fred Harris is designed for communication systems engineers, programmers, implementers and managers who need to understand current practice and next generation Digital Signal Processing (DSP) techniques for upcoming communication systems. DSP is more than simply mapping legacy analogue designs to a DSP implementation. To avoid compromised solutions based on analogue limitations from an earlier time period, this course returns to first principles to demonstrate how to apply new technology capabilities to the design of next generation communication systems.

Course Outline

An examination of past, present, and future digital modulation systems

Digital Filters

FIR filters, re-sampling filters, interpolators and decimators, half-band filters, cascade integrator-comb (CIC) filters, Hogenauer filters, multi-rate IIR filters


Analysis and Synthesis Channelisers, Perfect Reconstruction filter banks, modulation and demodulation, design techniques, workload advantages

Filter Design Techniques

Window designs and performance considerations, equiripple designs, system considerations, options to improve system performance, finite arithmetic Digital Baseband Transmission: Nyquist filter, excess bandwidth, matched filters, square-root Nyquist filter, shaping and up-sampling filters

Pre- and Post-Signal Conditioning

Analogue filters, timing jitter, direct digital synthesizers, CORDIC processors, direct digital synthesizers and oscillators, interpolating and decimating filters in A-to-D and D-to-A, AGC, DC cancelling, I-Q balancing, AGC loops, SNR estimators

Sigma-delta Converters

A-to-D, D-to-A, D-to-D, multi-loop and wide-band converters, system considerations

Carrier Centred Modulation and Demodulation

Shaping and interpolation, QPSK, QAM, Digital IF options, OFDM, legacy analogue modulation and demodulation in DSP. FM modulation and demodulation


Phase locked loop, proportional plus integral loops, phase recovery, band edge filters in frequency recovery, timing recovery, polyphase filters in timing recovery

Adaptive Filters

LMS algorithm, RLS algorithm, lattice filters, linear and adaptive equalization, decision feedback equalizers, constant modulus (blind) equalizers

Modem Structures

Wireline, cable, satellite, and terrestrial modems and considerations

Multirate Signal Processing for Communications Systems

A multirate filter has embedded within its structure a mechanism to implement one or more sample rate changes as part of its signal processing task. One might ask why one would want to change the sample rate. It was once suggested to me that ‘We change sample rate because we didn’t do it right the first time.’ Cute, but not true! We change sample rate to reduce the cost or improve the performance of a signal processing task. One obvious driver for changing sample rate is the Nyquist criterion that directs us to select a sample rate that exceeds the signal’s two sided bandwidth. Thus when we use a filter to reduce the bandwidth of a signal we should also reduce the output sample rate in proportion to the bandwidth reduction. A pleasant surprise is that we can embed the re-sampling operation within the filtering operation to obtain a realization that simultaneously exhibits minimum work load and minimum hardware requirements. Filters containing embedded sample rate changes are linear time varying filters.

These filters offer an amazing list of signal processing options not available to the standard filter structures. The list appears to contain elements of magic. It does!

The traditional structures that accomplish sample rate changes are the polyphase FIR filter, the dyadic half-band filter, and the cascade Integrator and comb filter. Other structures include the polyphase recursive all-pass filter, iterated filters, the Farrow filter, and variants of Taylor series filters. A new architecture, cascade analysis and synthesis filter banks, offers green, minimum energy implementation of filter functions. Filter structures are but the tip of the iceberg. What we need is an understanding how these structures can be modified and applied to various processing tasks. This course addresses both filter structures and filter applications. Material presented in the short course is listed in the following table.


First course in DSP: Course in Modulation Techniques, FIR Filters, Sampling Theorem and Spectrum Analysis Techniques

Course Outline

Practical Design of Finite Impulse Response Filters

Spectral and time domain characteristics Transition bandwidth, side-lobe levels and spectral envelope
In-band ripple level and paired echoes Remez Algorithm revisited, square-Root Nyquist filter

The Resampling Process

The Nobel Identity I-F sampling Nyquist zone filtering Intentional aliasing and spectral translation

Multirate FIR Filters

Rational ratio and arbitrary ratio interpolators (up and down-resampling)

Polyphase Partitions

Base-band filtering Multi-channel filtering Cascade filters,
cascade Perfect reconstruction analysis-synthesis filter banks

Half Band Filters

Quadrature mirror filters Hilbert transform filters

Cascade Integrator-Comb Filters

Hogenauer filter structures

Recursive Multirate Filters

Polyphase all-pass structures Base-band filtering Multi-channelfiltering Cascade filters

Recursive Half Band Filters

Quadrature mirror filters, Hilbert transform filters

MODEM Applications

Shaping filters with up-sampling Matched filters with up-samplingfor timing recovery
Multi-channel modulation and demodulation, Narrow-band signal and noise generators
Re-sampling for ADC and DAC applications Multirate considerations in equalizer filters

Course Details

Course start 19 March 2018
Course end 23 March 2018
Individual price $3500.00
Extra Information To secure your seat please submit a Registration Form. If you are waiting for financial/supervisor approval you are welcome to reserve a seat without commitment. Please note that the course fee is ex GST. We thank you for your interest and look forward to hearing from you.
Course location Adelaide (Mawson Lakes)
Please register your interest in this course by contacting Babcock on +61 (0)8 8440 1498

Register Here!

Registering for one of our courses is easy! Simply fill out one of the below Course Registration Forms and return via email to training@babcock.com.au

Course Registration Form - Individual Registration Form

Course Registration Form - Group Registration Form

Babcock Terms and Conditions


GST Terms

All prices shown are Exclusive of GST. GST of 10% will be added to all invoices