Voice Compression and Communications: Principles and Applications for Fixed and Wireless Channels
Up-to-date, expert coverage of topics in wireless voice communications Voice communication is the most important facet of mobile radio service. Even when the predicted surge of wireless data and Internet services becomes a reality, voice will remain the most natural means of human communication. Voice Compression and Communications details issues in wireless voice communications and treats compression, channel coding, and wireless transmission as a joint subject. Part I covers background material, whereas Part II provides detailed information on both proprietary and standardized analysis-by-synthesis codecs, including the speech codecs of virtually all existing wireline-based and wireless systems. Parts III and IV discuss mainly research-based wideband, audio, as well as very low-rate schemes likely to find their way into future standards. Voice Compression and Communications describes fundamental concepts in a non-mathematical way early in the book for those with only a background knowledge of signal processing and communications. More advanced readers will find detailed discussions of theoretical principles, future concepts, and solutions to various specific wireless voice communications problems.
Why Read This Book
You should read this book if you need a single reference that links speech-compression algorithms with the realities of wired and wireless transmission — including standards and perceptual evaluation. It walks you through both codec algorithms and the system-level trade-offs (error concealment, joint source-channel design) that matter in deployed voice systems.
Who Will Benefit
Practicing DSP and communications engineers or graduate students who design or evaluate voice codecs and voice transmission systems for fixed or mobile networks.
Level: Advanced — Prerequisites: Solid background in digital signal processing (LPC, linear systems, FFT), basic probability/communications theory, and familiarity with speech-production fundamentals.
Key Takeaways
- Explain the principles of analysis-by-synthesis speech coding and compare major codec families (LPC, CELP, waveform, vocoders).
- Evaluate and select standardized codecs (GSM, AMR, MELP, ITU-T) for different bit-rate and channel conditions.
- Apply joint source-channel design concepts and error-concealment strategies for wireless voice transmission.
- Assess perceptual quality and objective measures used in speech codec evaluation and system trade-offs.
- Characterize wideband and low-bit-rate voice coding techniques and their impact on bandwidth and complexity.
- Relate codec algorithm choices to practical wireless constraints (packet loss, fading, delay, and signaling).
Topics Covered
- Part I — Background: Speech production, perceptual aspects and basic signal representations
- Linear predictive coding and analysis methods
- Perceptual models and objective quality measures
- Part II — Analysis-by-synthesis codecs: theory and algorithmic building blocks
- Standardized narrowband codecs: GSM, RPE-LTP, AMR and related designs
- Low-rate and very low-rate codecs (MELP, waveform & hybrid approaches)
- Wideband and audio codecs for enhanced voice quality
- Part III — Channel aspects: wireless channel models, errors, and packet networks
- Joint source-channel coding and unequal error protection for voice
- Error concealment, packet loss mitigation and quality adaptation
- Implementation considerations, complexity and power trade-offs
- Standards, benchmarking and future directions
Languages, Platforms & Tools
How It Compares
Compared with Spanias/Painter/Atti's Speech Coding (a broad survey of codecs and perceptual coding), Hanzo places greater emphasis on wireless transmission, channel effects and joint source-channel considerations; Quatieri's Discrete-Time Processing of Speech Signals is more focused on in-depth signal modeling and theory rather than codec standards and system-level issues.
