Sound Capture and Processing: Practical Approaches
Provides state-of-the-art algorithms for sound capture, processing and enhancement Sound Capture and Processing: Practical Approaches covers the digital signal processing algorithms and devices for capturing sounds, mostly human speech. It explores the devices and technologies used to capture, enhance and process sound for the needs of communication and speech recognition in modern computers and communication devices. This book gives a comprehensive introduction to basic acoustics and microphones, with coverage of algorithms for noise reduction, acoustic echo cancellation, dereverberation and microphone arrays; charting the progress of such technologies from their evolution to present day standard. Sound Capture and Processing: Practical Approaches * Brings together the state-of-the-art algorithms for sound capture, processing and enhancement in one easily accessible volume* Provides invaluable implementation techniques required to process algorithms for real life applications and devices* Covers a number of advanced sound processing techniques, such as multichannel acoustic echo cancellation, dereverberation and source separation* Generously illustrated with figures and charts to demonstrate how sound capture and audio processing systems work* An accompanying website containing Matlab code to illustrate the algorithms This invaluable guide will provide audio, R&D and software engineers in the industry of building systems or computer peripherals for speech enhancement with a comprehensive overview of the technologies, devices and algorithms required for modern computers and communication devices. Graduate students studying electrical engineering and computer science, and researchers in multimedia, cell-phones, interactive systems and acousticians will also benefit from this book.
Why Read This Book
You will get a practical, engineering-oriented treatment of modern sound capture and front-end processing: how microphones and arrays work, and how to implement noise reduction, echo cancellation and dereverberation that matter in real systems. The book blends signal‑processing algorithms with hardware and implementation considerations so you can move from theory to working prototypes.
Who Will Benefit
Engineers and graduate students building audio front-ends, speech-recognition pre-processing, hands‑on developers working on acoustic capture, enhancement, or microphone-array products.
Level: Intermediate — Prerequisites: Basic DSP (Fourier transforms, filtering), linear algebra and probability; familiarity with discrete-time systems and FFTs; MATLAB experience is helpful but not mandatory.
Key Takeaways
- Implement practical noise reduction methods (spectral subtraction, Wiener filtering, MMSE-based estimators) for speech
- Develop and tune adaptive acoustic echo cancellation algorithms and their frequency-domain implementations
- Design and apply dereverberation techniques (inverse filtering, early-late tradeoffs, WPE-style approaches)
- Configure and implement microphone-array beamforming and spatial filtering for capture and source separation
- Characterize microphone and acoustic front-end hardware and account for real-world constraints (latency, computational cost, nonidealities)
- Evaluate enhancement algorithms with appropriate objective metrics and considerations for ASR and human perception
Topics Covered
- 1. Introduction: Sound Capture in Modern Devices
- 2. Basics of Acoustics and Room Propagation
- 3. Microphones and Transducer Characteristics
- 4. Front-End Hardware and ADC Considerations
- 5. Signal Models for Speech and Noise
- 6. Single-Channel Noise Reduction Techniques
- 7. Adaptive Filtering and Acoustic Echo Cancellation
- 8. Dereverberation and Late-Reflection Suppression
- 9. Microphone Arrays and Beamforming Techniques
- 10. Blind Source Separation and Spatial Filtering
- 11. Implementation Issues: Real-Time, Complexity, and Calibration
- 12. Evaluation Metrics and Testing for ASR/Perceptual Quality
- 13. Case Studies and Applications
- Appendices: Useful Transforms, Algorithms, and Implementation Notes
Languages, Platforms & Tools
How It Compares
Compared with P. Loizou's Speech Enhancement (which focuses deeper on single-channel denoising algorithms), Tashev's book is broader in covering capture hardware and array processing; Brandstein & Ward's Microphone Arrays is more academic on array theory, while Tashev is more implementation-oriented.
