Diploma in Audio/Comms: Codecs, VoC and Echo-cancelling
Online postgraduate diploma of 800 hours in Audio/Comms: Codecs, VoC and Echo Cancelling: 8 modules covering the speech signal and auditory perception, speech and audio codecs, voice transport over networks, acoustic echo cancellation, speech enhancement, real-time audio implementation and terminal testing, with an assigned technical tutor.
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What you will be able to do
Characterise the speech signal and perceived quality metrics.
Compare speech and audio codecs by quality, bitrate and complexity.
Configure voice transport over IP and mobile networks.
Design acoustic echo cancellers with adaptive filters.
Apply noise suppression and beamforming techniques.
Implement real-time audio processing chains.
Evaluate terminals against voice quality standards.
Who it's for
- Digital audio engineers working on voice products and conferencing devices.
- Speech processing engineers developing enhancement and echo control algorithms.
- VoIP engineers responsible for call quality across IP and mobile networks.
- Codec developers comparing and integrating speech and audio codecs.
- Acoustic test engineers who measure and tune terminals.
- Communications systems engineers dealing with end-to-end voice quality.
- Engineering or science graduates looking to enter the telecommunications sector.
What you'll work with
- ITU-T G.168 (digital network echo cancellers)
- ITU-T P.863 – POLQA (perceptual objective listening quality analysis)
- ITU-T G.711 (PCM of voice frequencies)
- ITU-T G.722 (7 kHz wideband audio coding)
- IETF RFC 6716 – Opus (interactive speech and audio codec)
- 3GPP TS 26.441 – EVS (Enhanced Voice Services codec)
- ITU-T P.862 – PESQ (perceptual evaluation of speech quality)
- ITU-T G.107 – E-model (transmission planning model)
- MATLAB (Audio Toolbox)
- Audacity
- WebRTC (AEC3)
- libopus
- Wireshark
8 modules
1.1Voice production and the source-filter model1.2Spectral characteristics and formants1.3Psychoacoustics and masking1.4Narrowband, wideband and super-wideband1.5Objective metrics: PESQ and POLQA1.6Subjective tests and MOS1.7Noise, reverberation and acoustic environments1.8Latency and conversational perception1.9The ITU-T G.107 E-model1.10Lab: formant analysis and spectrogram
2.1PCM and A-law/µ-law according to G.7112.2ADPCM coding and G.7222.3Linear prediction and CELP2.4AMR, AMR-WB and EVS in mobile networks2.5Opus: SILK and CELT modes2.6Perceptual codecs: AAC and MDCT2.7Packet loss concealment2.8Bitrate, quality and complexity compared2.9Transcoding and its impact2.10Practical: evaluating Opus at different bitrates
3.1RTP, RTCP and synchronisation3.2SIP and SDP signalling3.3VoLTE, VoNR and IMS3.4Adaptive jitter buffer3.5Packet loss and FEC3.6Quality of service and DSCP marking3.7WebRTC and secure transport with SRTP3.8End-to-end quality measurement3.9Stream capture and analysis3.10Lab: RTP call analysis with Wireshark
4.1Line echo and acoustic echo4.2NLMS adaptive filters4.3Frequency-domain and subband algorithms4.4Double-talk detection4.5Non-linear residual echo suppression4.6Reference delay estimation4.7ITU-T G.168 requirements4.8WebRTC AEC3 architecture4.9ERLE metrics and convergence4.10Workshop: NLMS canceller in Python
5.1Spectral and Wiener noise suppression5.2Noise estimation and VAD detectors5.3Automatic gain control5.4Beamforming with microphone arrays5.5Speech dereverberation techniques5.6Noise suppression with neural networks5.7Equalisation and output processing5.8Complete processing chain5.9Objective and subjective evaluation5.10Practical: delay-and-sum beamformer with two microphones
6.1Platforms: DSP, ARM and audio SoCs6.2I2S, TDM and PDM interfaces6.3Frame-based processing and latency6.4Fixed point versus floating point in audio6.5Linux audio stacks: ALSA and PipeWire6.6Clock synchronisation and drift6.7Memory management and CPU load6.8Integration into an audio RTOS6.9Debugging audible artefacts6.10Lab: AEC chain on an ARM board with ALSA
7.1Acoustic measurements according to 3GPP TS 26.131 and 26.1327.2Head and torso simulators7.3Frequency response measurement7.4Echo and double-talk tests7.5Quality tests with POLQA7.6Codec interoperability7.7Testing on live networks and in the laboratory7.8Conformance report7.9Iterative tuning improvement7.10Case study: acoustic tuning of a hands-free device
8.1Definition of the terminal or service8.2Codec and transport selection8.3Design of the processing chain8.4Implementation of the echo canceller8.5Speech enhancement and gain control8.6Objective quality measurement8.7Interoperability tests8.8Technical documentation of the prototype8.9Report 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 (Audio Toolbox), Audacity, WebRTC (AEC3), libopus and Wireshark, in free, academic or evaluation versions. Testing is studied through real procedures: voice quality measurements with POLQA, echo cancellation tests with double talk and RTP stream analysis. Assessment is continuous, module by module, plus the final project.
Standards covered: ITU-T G.168, ITU-T P.863 (POLQA), ITU-T G.711, ITU-T G.722, IETF RFC 6716 (Opus), 3GPP TS 26.441 (EVS), ITU-T P.862 (PESQ) and ITU-T G.107 (E-model).
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 (Audio Toolbox), Audacity, WebRTC (AEC3), libopus and Wireshark, 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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