Plural Publishing
   
Digital Hearing Aids
April 2008

Digital Hearing Aids

James M. Kates, E.E.
464 pages. Softcover. 6x9 in
ISBN10: 1-59756-316-X. ISBN13: 978-1-59756-317-8.
US$89.95 CAN$101.00 £57.00 AU$140.00
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ABOUT THE BOOK
The first book available on the subject, Digital Hearing Aids provides an overview of the signal-processing strategies implemented in modern digital hearing aids. Algorithms ranging from dynamic-range compression and directional microphones to sound classification and binaural noise suppression are clearly explained. The basic equations describing the signal-processing algorithms are presented along with full explanations for those less comfortable with the mathematics, and each processing strategy is accompanied by a summary of its effectiveness. The text is intended for a graduate audiology course in hearing aids and hearing-aid technology.

CONTENTS
  • Preface
    • Hearing-Aid Technology
        Types of Hearing Aids
          Canal Aids
          In-the-Ear Aids
          Behind-the-Ear Aids
          Open Fitting BTE Aids
          Body-Worn Aids
        From Analog to Digital
          Moore’s Law
          Digital Current Drain
          Analog and Digital Power Consumption
        Digital Circuit Components
          Analog-to-Digital Converter
          Digital Signal Processor
          Memory
          Clock
        Batteries
        Concluding Remarks
        References
      Signal Processing Basics
        Signal and System Properties
          Sequences
          Linear Time Invariance
          Convolution
          Correlation
          Transfer Functions
          Feedback
          Group Delay
          Amplitude Modulation
          Distortion
        Discrete Fourier Transform
        Filters and Filter Banks
          Filter Definitions
          Recursive Filters
          Non-Recursive Filters
          Frequency Resolution and Group Delay
        Block Processing
        Digital System Concerns
          Integer Arithmetic
          Quantization Noise
          Aliasing Distortion
          Temporal Aliasing
          Algorithm Complexity
          Algorithm Interaction
        Concluding Remarks
        References
      The Electroacoustic System
        Hearing Aid System
        Head and Ear
        Microphone and Receiver
        Vent Acoustics
          Acoustic Elements
          Vent Size
          Residual Ear-Canal Volume
          Open BTE Fitting
        Occlusion Effect
        Concluding Remarks
        References
      Directional Microphones
        Hearing-Aid Microphones
          Microphone Construction
          Directional Microphones and Spatial Signal Processing
        Directional Response Patterns
          Microphone Spacing and Time Delay
          Microphone Mismatch
        Frequency Response
        Magnitude Frequency Response
        Microphone Mismatch
        Interaction with Vents
        Microphone Noise
        Microphones on the Head
        Microphone Performance Indices
          Front-to-Back Ratio
          Directivity Index
          Unidirectional Index The AI-DI
        Rooms and Reverberation
          Critical Distance
          Early Reflections
          Combining Noise and Reverberation
        Benefit in the Real World
        Concluding Remarks
        References
      Adaptive and Multi-Microphone Arrays
        Two-Microphone Adaptive Array
          Adaptive Delay
          Adaptive Gain
          Adaptive Filter
        Delay-And-Sum Beamforming
        Adaptive Arrays
        Superdirective Arrays
        Widely-Spaced Arrays
        Array Benefits
        Concluding Remarks
        References
      Wind Noise
        Turbulence
        Hearing-Aid Measurements
          Air Flow and Turbulence
          Sound Pressure
        Signal Characteristics
          Spectrum
          Temporal Fluctuations
          Correlation
        Wind-Noise Reduction
          Directional Microphones
          Wind Screens
          Spectrum Algorithms
          Correlation Algorithms
          Adaptive Arrays
        Concluding Remarks
        References
      Feedback Cancellation
        The Feedback System
          System Equations
          System Response
        Gain-Reduction Solutions
        Adaptive Feedback Cancellation
          System Equations
          LMS Adaptation
          Constrained Adaptation
            Adaptation Equations
            Initialization
            Simulation Results
          Decorrelation Techniques
            Interrupted Adaptation
            Delay
            Filtered-X Algorithm
          Frequency-Domain Adaptation
        Processing Limitations
          Room Reflections
            Measurement Procedure
            Measurement Results
            Maximum Stable Gain
          Non-Linear Distortion
        Concluding Remarks
        References
      Dynamic-Range Compression
        Does Compression Help?
        Algorithm Design Concerns
          Frequency Resolution
          Processing Delay
        Single-Channel Compression
          Input/Output Rules
          Volume Control
          Envelope Detection
        Multi-Channel Compression
          Temporal Response
          Swept Frequency Response
        Frequency-Domain Compression
          Ideal FFT System
          Practical FFT System
          Side-Branch Structure
        Frequency Warping
          Digital Frequency Warping
          Warped Compressor System
          System Delay Comparison
        Concluding Remarks
        References
      Single-Microphone Noise Suppression
        Properties of Speech and Noise Signals
        Low-Level Expansion
        Envelope Valley Tracking
        Bandwidth Reduction
          Adaptive High-Pass Filter
          Adaptive Low-Pass Filter
          “Zeta Noise Blocker”
        Envelope Modulation Filters
        Concluding Remarks
        References
      Spectral Subtraction
        Noise Estimation
          Valley Detection
          Minima Statistics
          Histogram
        Wiener Filter
        Spectral Subtraction
          Classical Approaches
          General Equation
          Nonlinear Expansion
          Ephraim-Malah Algorithm
          Auditory Masking
          A Fundamental Compromise
        Algorithm Effectiveness
        Concluding Remarks
        References
      Spectral Contrast Enhancement
        Auditory Filters in the Damaged Cochlea
          Physiological Measurements
            Tuning Curves
            Neural Firing Patterns
          Perceptual Measurements
          Processing Strategies
        Spectral Valley Suppression
        Spectral Contrast Modification
          Raise Spectrum to a Power
          Spectral Filtering
        Excess Upward Spread of Masking
        F2/F1 Ratio
        Processing Comparison
        Combining Spectral Contrast Enhancement with Compression
        Concluding Remarks
        References
      Sound Classification

        The Rationale for Classification
        Signal Features
        Feature Selection
          Mutual Information
          Feature Selection Using Mutual Information
        Classifier Algorithms
          Training the Classifier
          Types of Classifiers
          Clustering
          Linear Discriminant
          Support Vector Machine
          Neural Network
          Hidden Markov Model
          Bayes Classifier
          Personalizing the Classifier
        Classification Examples
          Number of Features
          Isolated Signals vs. Mixtures
        Concluding Remarks
        References
      Binaural Signal Processing
        The “Cocktail Party” Problem
        Signal Transmission
        Binaural Compression
          Compression Using Binaural Loudness Summation
          Auditory Efferents
          Compression Using Auditory Efferents
        Binaural Noise Suppression
          Interaural Cross-Correlation
          Binaural Wiener Filter
          Binaural Signal Difference
          Binaural Spectral Subtraction
          Directional Cues
          Localization Model
        Dichotic Band Splitting
        Concluding Remarks
        References
      Index


    ABOUT THE AUTHOR
    James M. Kates, E.E. holds the position of Research Fellow with hearing-aid manufacturer GN ReSound. He is also an Adjunct Professor in the Department of Speech Language and Hearing Sciences at University of Colorado in Boulder. Prior to moving to Colorado, he was Research Scientist in the laboratory of Harry Levitt at the City University of New York. He received the BSEE and MSEE degrees from the Massachusetts Institute of Technology in 1971, and the professional degree of Electrical Engineer from M.I.T. in 1972. He has developed many of the signal-processing algorithms implemented in the GN family of hearing aids, including the DFS feedback cancellation algorithm and the Warp-17 frequency-warped dynamic-range compression algorithm. He is a Fellow of the Acoustical Society of America and a Fellow of the Audio Engineering Society, and is the author or co-author of 47 technical papers and holds 17 US patents with seven patents pending. A widower, he resides in Niwot, Colorado.


    AUDIENCE
    Primary: Audiology

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