Digital speech transmission [[electronic resource] ] : enhancement, coding and error concealment / / Peter Vary, Rainer Martin
| Digital speech transmission [[electronic resource] ] : enhancement, coding and error concealment / / Peter Vary, Rainer Martin |
| Autore | Vary Peter |
| Pubbl/distr/stampa | Chichester, England ; ; Hoboken, NJ, : John Wiley, c2006 |
| Descrizione fisica | 1 online resource (645 p.) |
| Disciplina | 621.399 |
| Altri autori (Persone) | MartinRainer |
| Soggetto topico |
Speech processing systems
Signal processing - Digital techniques Error-correcting codes (Information theory) |
| ISBN |
1-280-60611-8
9786610606115 0-470-03174-3 0-470-03175-1 |
| Formato | Materiale a stampa |
| Livello bibliografico | Monografia |
| Lingua di pubblicazione | eng |
| Nota di contenuto |
Digital Speech Transmission; Contents; Preface; 1 Introduction; 2 Models of Speech Production and Hearing; 2.1 Organs of Speech Production; 2.2 Characteristics of Speech Signals; 2.3 Model of Speech Production; 2.3.1 Acoustic Tube Model of the Vocal Tract; 2.3.2 Digital All-Pole Model of the Vocal Tract; 2.4 Anatomy of Hearing; 2.5 Psychoacoustic Properties of the Auditory Organ; 2.5.1 Hearing and Loudness; 2.5.2 Spectral Resolution; 2.5.3 Masking; Bibliography; 3 Spectral Transformations; 3.1 Fourier Transform of Continuous Signals; 3.2 Fourier Transform of Discrete Signals
3.3 Linear Shift Invariant Systems3.3.1 Frequency Response of LSI Systems; 3.4 The z-transform; 3.4.1 Relation to FT; 3.4.2 Properties of the ROC; 3.4.3 Inverse z-transform; 3.4.4 z-transform Analysis of LSI Systems; 3.5 The Discrete Fourier Transform; 3.5.1 Linear and Cyclic Convolution; 3.5.2 The DFT of Windowed Sequences; 3.5.3 Spectral Resolution and Zero Padding; 3.5.4 Fast Computation of the DFT: The FFT; 3.5.5 Radix-2 Decimation-in-Time FFT; 3.6 Fast Convolution; 3.6.1 Fast Convolution of Long Sequences; 3.6.2 Fast Convolution by Overlap-Add; 3.6.3 Fast Convolution by Overlap-Save 3.7 Cepstral Analysis3.7.1 Complex Cepstrum; 3.7.2 Real Cepstrum; 3.7.3 Applications of the Cepstrum; Bibliography; 4 Filter Banks for Spectral Analysis and Synthesis; 4.1 Spectral Analysis Using Narrowband Filters; 4.1.1 Short-Term Spectral Analyzer; 4.1.2 Prototype Filter Design for the Analysis Filter Bank; 4.1.3 Short-Term Spectral Synthesizer; 4.1.4 Short-Term Spectral Analysis and Synthesis; 4.1.5 Prototype Filter Design for the Analysis-Synthesis Filter Bank; 4.1.6 Filter Bank Interpretation of the DFT; 4.2 Polyphase Network Filter Banks; 4.2.1 PPN Analysis Filter Bank 4.2.2 PPN Synthesis Filter Bank4.3 Quadrature Mirror Filter Banks; 4.3.1 Analysis-Synthesis Filter Bank; 4.3.2 Compensation of Aliasing and Signal Reconstruction; 4.3.3 Efficient Implementation; Bibliography; 5 Stochastic Signals and Estimation; 5.1 Basic Concepts; 5.1.1 Random Events and Probability; 5.1.2 Conditional Probabilities; 5.1.3 Random Variables; 5.1.4 Probability Distributions and Probability Density Functions; 5.1.5 Conditional PDFs; 5.2 Expectations and Moments; 5.2.1 Conditional Expectations and Moments; 5.2.2 Examples; 5.2.3 Transformation of a Random Variable 5.2.4 Relative Frequencies and Histograms5.3 Bivariate Statistics; 5.3.1 Marginal Densities; 5.3.2 Expectations and Moments; 5.3.3 Uncorrelatedness and Statistical Independence; 5.3.4 Examples of Bivariate PDFs; 5.3.5 Functions of Two Random Variables; 5.4 Probability and Information; 5.4.1 Entropy; 5.4.2 Kullback-Leibler Divergence; 5.4.3 Mutual Information; 5.5 Multivariate Statistics; 5.5.1 MultivariateGaussian Distribution; 5.5.2 χ2-distribution; 5.6 Stochastic Processes; 5.6.1 Stationary Processes; 5.6.2 Auto-correlation and Auto-covariance Functions 5.6.3 Cross-correlation and Cross-covariance Functions |
| Record Nr. | UNINA-9910143561303321 |
Vary Peter
|
||
| Chichester, England ; ; Hoboken, NJ, : John Wiley, c2006 | ||
| Lo trovi qui: Univ. Federico II | ||
| ||
Digital speech transmission [[electronic resource] ] : enhancement, coding and error concealment / / Peter Vary, Rainer Martin
| Digital speech transmission [[electronic resource] ] : enhancement, coding and error concealment / / Peter Vary, Rainer Martin |
| Autore | Vary Peter |
| Pubbl/distr/stampa | Chichester, England ; ; Hoboken, NJ, : John Wiley, c2006 |
| Descrizione fisica | 1 online resource (645 p.) |
| Disciplina | 621.399 |
| Altri autori (Persone) | MartinRainer |
| Soggetto topico |
Speech processing systems
Signal processing - Digital techniques Error-correcting codes (Information theory) |
| ISBN |
1-280-60611-8
9786610606115 0-470-03174-3 0-470-03175-1 |
| Formato | Materiale a stampa |
| Livello bibliografico | Monografia |
| Lingua di pubblicazione | eng |
| Nota di contenuto |
Digital Speech Transmission; Contents; Preface; 1 Introduction; 2 Models of Speech Production and Hearing; 2.1 Organs of Speech Production; 2.2 Characteristics of Speech Signals; 2.3 Model of Speech Production; 2.3.1 Acoustic Tube Model of the Vocal Tract; 2.3.2 Digital All-Pole Model of the Vocal Tract; 2.4 Anatomy of Hearing; 2.5 Psychoacoustic Properties of the Auditory Organ; 2.5.1 Hearing and Loudness; 2.5.2 Spectral Resolution; 2.5.3 Masking; Bibliography; 3 Spectral Transformations; 3.1 Fourier Transform of Continuous Signals; 3.2 Fourier Transform of Discrete Signals
3.3 Linear Shift Invariant Systems3.3.1 Frequency Response of LSI Systems; 3.4 The z-transform; 3.4.1 Relation to FT; 3.4.2 Properties of the ROC; 3.4.3 Inverse z-transform; 3.4.4 z-transform Analysis of LSI Systems; 3.5 The Discrete Fourier Transform; 3.5.1 Linear and Cyclic Convolution; 3.5.2 The DFT of Windowed Sequences; 3.5.3 Spectral Resolution and Zero Padding; 3.5.4 Fast Computation of the DFT: The FFT; 3.5.5 Radix-2 Decimation-in-Time FFT; 3.6 Fast Convolution; 3.6.1 Fast Convolution of Long Sequences; 3.6.2 Fast Convolution by Overlap-Add; 3.6.3 Fast Convolution by Overlap-Save 3.7 Cepstral Analysis3.7.1 Complex Cepstrum; 3.7.2 Real Cepstrum; 3.7.3 Applications of the Cepstrum; Bibliography; 4 Filter Banks for Spectral Analysis and Synthesis; 4.1 Spectral Analysis Using Narrowband Filters; 4.1.1 Short-Term Spectral Analyzer; 4.1.2 Prototype Filter Design for the Analysis Filter Bank; 4.1.3 Short-Term Spectral Synthesizer; 4.1.4 Short-Term Spectral Analysis and Synthesis; 4.1.5 Prototype Filter Design for the Analysis-Synthesis Filter Bank; 4.1.6 Filter Bank Interpretation of the DFT; 4.2 Polyphase Network Filter Banks; 4.2.1 PPN Analysis Filter Bank 4.2.2 PPN Synthesis Filter Bank4.3 Quadrature Mirror Filter Banks; 4.3.1 Analysis-Synthesis Filter Bank; 4.3.2 Compensation of Aliasing and Signal Reconstruction; 4.3.3 Efficient Implementation; Bibliography; 5 Stochastic Signals and Estimation; 5.1 Basic Concepts; 5.1.1 Random Events and Probability; 5.1.2 Conditional Probabilities; 5.1.3 Random Variables; 5.1.4 Probability Distributions and Probability Density Functions; 5.1.5 Conditional PDFs; 5.2 Expectations and Moments; 5.2.1 Conditional Expectations and Moments; 5.2.2 Examples; 5.2.3 Transformation of a Random Variable 5.2.4 Relative Frequencies and Histograms5.3 Bivariate Statistics; 5.3.1 Marginal Densities; 5.3.2 Expectations and Moments; 5.3.3 Uncorrelatedness and Statistical Independence; 5.3.4 Examples of Bivariate PDFs; 5.3.5 Functions of Two Random Variables; 5.4 Probability and Information; 5.4.1 Entropy; 5.4.2 Kullback-Leibler Divergence; 5.4.3 Mutual Information; 5.5 Multivariate Statistics; 5.5.1 MultivariateGaussian Distribution; 5.5.2 χ2-distribution; 5.6 Stochastic Processes; 5.6.1 Stationary Processes; 5.6.2 Auto-correlation and Auto-covariance Functions 5.6.3 Cross-correlation and Cross-covariance Functions |
| Record Nr. | UNINA-9910830553603321 |
Vary Peter
|
||
| Chichester, England ; ; Hoboken, NJ, : John Wiley, c2006 | ||
| Lo trovi qui: Univ. Federico II | ||
| ||
Digital speech transmission : enhancement, coding and error concealment / / Peter Vary, Rainer Martin
| Digital speech transmission : enhancement, coding and error concealment / / Peter Vary, Rainer Martin |
| Autore | Vary Peter |
| Pubbl/distr/stampa | Chichester, England ; ; Hoboken, NJ, : John Wiley, c2006 |
| Descrizione fisica | 1 online resource (645 p.) |
| Disciplina | 621.39/9 |
| Altri autori (Persone) | MartinRainer |
| Soggetto topico |
Speech processing systems
Signal processing - Digital techniques Error-correcting codes (Information theory) |
| ISBN |
9786610606115
9781280606113 1280606118 9780470031742 0470031743 9780470031759 0470031751 |
| Formato | Materiale a stampa |
| Livello bibliografico | Monografia |
| Lingua di pubblicazione | eng |
| Nota di contenuto |
Digital Speech Transmission; Contents; Preface; 1 Introduction; 2 Models of Speech Production and Hearing; 2.1 Organs of Speech Production; 2.2 Characteristics of Speech Signals; 2.3 Model of Speech Production; 2.3.1 Acoustic Tube Model of the Vocal Tract; 2.3.2 Digital All-Pole Model of the Vocal Tract; 2.4 Anatomy of Hearing; 2.5 Psychoacoustic Properties of the Auditory Organ; 2.5.1 Hearing and Loudness; 2.5.2 Spectral Resolution; 2.5.3 Masking; Bibliography; 3 Spectral Transformations; 3.1 Fourier Transform of Continuous Signals; 3.2 Fourier Transform of Discrete Signals
3.3 Linear Shift Invariant Systems3.3.1 Frequency Response of LSI Systems; 3.4 The z-transform; 3.4.1 Relation to FT; 3.4.2 Properties of the ROC; 3.4.3 Inverse z-transform; 3.4.4 z-transform Analysis of LSI Systems; 3.5 The Discrete Fourier Transform; 3.5.1 Linear and Cyclic Convolution; 3.5.2 The DFT of Windowed Sequences; 3.5.3 Spectral Resolution and Zero Padding; 3.5.4 Fast Computation of the DFT: The FFT; 3.5.5 Radix-2 Decimation-in-Time FFT; 3.6 Fast Convolution; 3.6.1 Fast Convolution of Long Sequences; 3.6.2 Fast Convolution by Overlap-Add; 3.6.3 Fast Convolution by Overlap-Save 3.7 Cepstral Analysis3.7.1 Complex Cepstrum; 3.7.2 Real Cepstrum; 3.7.3 Applications of the Cepstrum; Bibliography; 4 Filter Banks for Spectral Analysis and Synthesis; 4.1 Spectral Analysis Using Narrowband Filters; 4.1.1 Short-Term Spectral Analyzer; 4.1.2 Prototype Filter Design for the Analysis Filter Bank; 4.1.3 Short-Term Spectral Synthesizer; 4.1.4 Short-Term Spectral Analysis and Synthesis; 4.1.5 Prototype Filter Design for the Analysis-Synthesis Filter Bank; 4.1.6 Filter Bank Interpretation of the DFT; 4.2 Polyphase Network Filter Banks; 4.2.1 PPN Analysis Filter Bank 4.2.2 PPN Synthesis Filter Bank4.3 Quadrature Mirror Filter Banks; 4.3.1 Analysis-Synthesis Filter Bank; 4.3.2 Compensation of Aliasing and Signal Reconstruction; 4.3.3 Efficient Implementation; Bibliography; 5 Stochastic Signals and Estimation; 5.1 Basic Concepts; 5.1.1 Random Events and Probability; 5.1.2 Conditional Probabilities; 5.1.3 Random Variables; 5.1.4 Probability Distributions and Probability Density Functions; 5.1.5 Conditional PDFs; 5.2 Expectations and Moments; 5.2.1 Conditional Expectations and Moments; 5.2.2 Examples; 5.2.3 Transformation of a Random Variable 5.2.4 Relative Frequencies and Histograms5.3 Bivariate Statistics; 5.3.1 Marginal Densities; 5.3.2 Expectations and Moments; 5.3.3 Uncorrelatedness and Statistical Independence; 5.3.4 Examples of Bivariate PDFs; 5.3.5 Functions of Two Random Variables; 5.4 Probability and Information; 5.4.1 Entropy; 5.4.2 Kullback-Leibler Divergence; 5.4.3 Mutual Information; 5.5 Multivariate Statistics; 5.5.1 MultivariateGaussian Distribution; 5.5.2 χ2-distribution; 5.6 Stochastic Processes; 5.6.1 Stationary Processes; 5.6.2 Auto-correlation and Auto-covariance Functions 5.6.3 Cross-correlation and Cross-covariance Functions |
| Record Nr. | UNINA-9911019682603321 |
Vary Peter
|
||
| Chichester, England ; ; Hoboken, NJ, : John Wiley, c2006 | ||
| Lo trovi qui: Univ. Federico II | ||
| ||
Digital Speech Transmission and Enhancement
| Digital Speech Transmission and Enhancement |
| Autore | Vary Peter |
| Edizione | [2nd ed.] |
| Pubbl/distr/stampa | Newark : , : John Wiley & Sons, Incorporated, , 2023 |
| Descrizione fisica | 1 online resource (595 pages) |
| Disciplina | 006.454 |
| Altri autori (Persone) | MartinRainer |
| Collana | IEEE Press Series |
| ISBN |
1-119-06099-0
1-119-06097-4 |
| Formato | Materiale a stampa |
| Livello bibliografico | Monografia |
| Lingua di pubblicazione | eng |
| Nota di contenuto |
Cover -- Title Page -- Copyright -- Contents -- Preface -- Chapter 1 Introduction -- Chapter 2 Models of Speech Production and Hearing -- 2.1 Sound Waves -- 2.2 Organs of Speech Production -- 2.3 Characteristics of Speech Signals -- 2.4 Model of Speech Production -- 2.4.1 Acoustic Tube Model of the Vocal Tract -- 2.4.2 Discrete Time All‐Pole Model of the Vocal Tract -- 2.5 Anatomy of Hearing -- 2.6 Psychoacoustic Properties of the Auditory System -- 2.6.1 Hearing and Loudness -- 2.6.2 Spectral Resolution -- 2.6.3 Masking -- 2.6.4 Spatial Hearing -- 2.6.4.1 Head‐Related Impulse Responses and Transfer Functions -- 2.6.4.2 Law of The First Wavefront -- References -- Chapter 3 Spectral Transformations -- 3.1 Fourier Transform of Continuous Signals -- 3.2 Fourier Transform of Discrete Signals -- 3.3 Linear Shift Invariant Systems -- 3.3.1 Frequency Response of LSI Systems -- 3.4 The z‐transform -- 3.4.1 Relation to Fourier Transform -- 3.4.2 Properties of the ROC -- 3.4.3 Inverse z‐Transform -- 3.4.4 z‐Transform Analysis of LSI Systems -- 3.5 The Discrete Fourier Transform -- 3.5.1 Linear and Cyclic Convolution -- 3.5.2 The DFT of Windowed Sequences -- 3.5.3 Spectral Resolution and Zero Padding -- 3.5.4 The Spectrogram -- 3.5.5 Fast Computation of the DFT: The FFT -- 3.5.6 Radix‐2 Decimation‐in‐Time FFT -- 3.6 Fast Convolution -- 3.6.1 Fast Convolution of Long Sequences -- 3.6.2 Fast Convolution by Overlap‐Add -- 3.6.3 Fast Convolution by Overlap‐Save -- 3.7 Analysis-Modification-Synthesis Systems -- 3.8 Cepstral Analysis -- 3.8.1 Complex Cepstrum -- 3.8.2 Real Cepstrum -- 3.8.3 Applications of the Cepstrum -- 3.8.3.1 Construction of Minimum‐Phase Sequences -- 3.8.3.2 Deconvolution by Cepstral Mean Subtraction -- 3.8.3.3 Computation of the Spectral Distortion Measure -- 3.8.3.4 Fundamental Frequency Estimation -- References.
Chapter 4 Filter Banks for Spectral Analysis and Synthesis -- 4.1 Spectral Analysis Using Narrowband Filters -- 4.1.1 Short‐Term Spectral Analyzer -- 4.1.2 Prototype Filter Design for the Analysis Filter Bank -- 4.1.3 Short‐Term Spectral Synthesizer -- 4.1.4 Short‐Term Spectral Analysis and Synthesis -- 4.1.5 Prototype Filter Design for the Analysis-Synthesis filter bank -- 4.1.6 Filter Bank Interpretation of the DFT -- 4.2 Polyphase Network Filter Banks -- 4.2.1 PPN Analysis Filter Bank -- 4.2.2 PPN Synthesis Filter Bank -- 4.3 Quadrature Mirror Filter Banks -- 4.3.1 Analysis-Synthesis Filter Bank -- 4.3.2 Compensation of Aliasing and Signal Reconstruction -- 4.3.3 Efficient Implementation -- 4.4 Filter Bank Equalizer -- 4.4.1 The Reference Filter Bank -- 4.4.2 Uniform Frequency Resolution -- 4.4.3 Adaptive Filter Bank Equalizer: Gain Computation -- 4.4.3.1 Conventional Spectral Subtraction -- 4.4.3.2 Filter Bank Equalizer -- 4.4.4 Non‐uniform Frequency Resolution -- 4.4.5 Design Aspects & -- Implementation -- References -- Chapter 5 Stochastic Signals and Estimation -- 5.1 Basic Concepts -- 5.1.1 Random Events and Probability -- 5.1.2 Conditional Probabilities -- 5.1.3 Random Variables -- 5.1.4 Probability Distributions and Probability Density Functions -- 5.1.5 Conditional PDFs -- 5.2 Expectations and Moments -- 5.2.1 Conditional Expectations and Moments -- 5.2.2 Examples -- 5.2.2.1 The Uniform Distribution -- 5.2.2.2 The Gaussian Density -- 5.2.2.3 The Exponential Density -- 5.2.2.4 The Laplace Density -- 5.2.2.5 The Gamma Density -- 5.2.2.6 χ2‐Distribution -- 5.2.3 Transformation of a Random Variable -- 5.2.4 Relative Frequencies and Histograms -- 5.3 Bivariate Statistics -- 5.3.1 Marginal Densities -- 5.3.2 Expectations and Moments -- 5.3.3 Uncorrelatedness and Statistical Independence -- 5.3.4 Examples of Bivariate PDFs. 5.3.4.1 The Bivariate Uniform Density -- 5.3.4.2 The Bivariate Gaussian Density -- 5.3.5 Functions of Two Random Variables -- 5.4 Probability and Information -- 5.4.1 Entropy -- 5.4.2 Kullback-Leibler Divergence -- 5.4.3 Cross‐Entropy -- 5.4.4 Mutual Information -- 5.5 Multivariate Statistics -- 5.5.1 Multivariate Gaussian Distribution -- 5.5.2 Gaussian Mixture Models -- 5.6 Stochastic Processes -- 5.6.1 Stationary Processes -- 5.6.2 Auto‐Correlation and Auto‐Covariance Functions -- 5.6.3 Cross‐Correlation and Cross‐Covariance Functions -- 5.6.4 Markov Processes -- 5.6.5 Multivariate Stochastic Processes -- 5.7 Estimation of Statistical Quantities by Time Averages -- 5.7.1 Ergodic Processes -- 5.7.2 Short‐Time Stationary Processes -- 5.8 Power Spectrum and its Estimation -- 5.8.1 White Noise -- 5.8.2 The Periodogram -- 5.8.3 Smoothed Periodograms -- 5.8.3.1 Non Recursive Smoothing in Time -- 5.8.3.2 Recursive Smoothing in Time -- 5.8.3.3 Log‐Mel Filter Bank Features -- 5.8.4 Power Spectra and Linear Shift‐Invariant Systems -- 5.9 Statistical Properties of Speech Signals -- 5.10 Statistical Properties of DFT Coefficients -- 5.10.1 Asymptotic Statistical Properties -- 5.10.2 Signal‐Plus‐Noise Model -- 5.10.3 Statistics of DFT Coefficients for Finite Frame Lengths -- 5.11 Optimal Estimation -- 5.11.1 MMSE Estimation -- 5.11.2 Estimation of Discrete Random Variables -- 5.11.3 Optimal Linear Estimator -- 5.11.4 The Gaussian Case -- 5.11.5 Joint Detection and Estimation -- 5.12 Non‐Linear Estimation with Deep Neural Networks -- 5.12.1 Basic Network Components -- 5.12.1.1 The Perceptron -- 5.12.1.2 Convolutional Neural Network -- 5.12.2 Basic DNN Structures -- 5.12.2.1 Fully‐Connected Feed‐Forward Network -- 5.12.2.2 Autoencoder Networks -- 5.12.2.3 Recurrent Neural Networks -- 5.12.2.4 Time Delay, Wavenet, and Transformer Networks. 5.12.2.5 Training of Neural Networks -- 5.12.2.6 Stochastic Gradient Descent (SGD) -- 5.12.2.7 Adaptive Moment Estimation Method (ADAM) -- References -- Chapter 6 Linear Prediction -- 6.1 Vocal Tract Models and Short‐Term Prediction -- 6.1.1 All‐Zero Model -- 6.1.2 All‐Pole Model -- 6.1.3 Pole‐Zero Model -- 6.2 Optimal Prediction Coefficients for Stationary Signals -- 6.2.1 Optimum Prediction -- 6.2.2 Spectral Flatness Measure -- 6.3 Predictor Adaptation -- 6.3.1 Block‐Oriented Adaptation -- 6.3.1.1 Auto‐Correlation Method -- 6.3.1.2 Covariance Method -- 6.3.1.3 Levinson-Durbin Algorithm -- 6.3.2 Sequential Adaptation -- 6.4 Long‐Term Prediction -- References -- Chapter 7 Quantization -- 7.1 Analog Samples and Digital Representation -- 7.2 Uniform Quantization -- 7.3 Non‐uniform Quantization -- 7.4 Optimal Quantization -- 7.5 Adaptive Quantization -- 7.6 Vector Quantization -- 7.6.1 Principle -- 7.6.2 The Complexity Problem -- 7.6.3 Lattice Quantization -- 7.6.4 Design of Optimal Vector Code Books -- 7.6.5 Gain-Shape Vector Quantization -- 7.7 Quantization of the Predictor Coefficients -- 7.7.1 Scalar Quantization of the LPC Coefficients -- 7.7.2 Scalar Quantization of the Reflection Coefficients -- 7.7.3 Scalar Quantization of the LSF Coefficients -- References -- Chapter 8 Speech Coding -- 8.1 Speech‐Coding Categories -- 8.2 Model‐Based Predictive Coding -- 8.3 Linear Predictive Waveform Coding -- 8.3.1 First‐Order DPCM -- 8.3.2 Open‐Loop and Closed‐Loop Prediction -- 8.3.3 Quantization of the Residual Signal -- 8.3.3.1 Quantization with Open‐Loop Prediction -- 8.3.3.2 Quantization with Closed‐Loop Prediction -- 8.3.3.3 Spectral Shaping of the Quantization Error -- 8.3.4 ADPCM with Sequential Adaptation -- 8.4 Parametric Coding -- 8.4.1 Vocoder Structures -- 8.4.2 LPC Vocoder -- 8.5 Hybrid Coding -- 8.5.1 Basic Codec Concepts. 8.5.1.1 Scalar Quantization of the Residual Signal -- 8.5.1.2 Vector Quantization of the Residual Signal -- 8.5.2 Residual Signal Coding: RELP -- 8.5.3 Analysis by Synthesis: CELP -- 8.5.3.1 Principle -- 8.5.3.2 Fixed Code Book -- 8.5.3.3 Long‐Term Prediction, Adaptive Code Book -- 8.6 Adaptive Postfiltering -- 8.7 Speech Codec Standards: Selected Examples -- 8.7.1 GSM Full‐Rate Codec -- 8.7.2 EFR Codec -- 8.7.3 Adaptive Multi‐Rate Narrowband Codec (AMR‐NB) -- 8.7.4 ITU‐T/G.722: 7 kHz Audio Coding within 64 kbit/s -- 8.7.5 Adaptive Multi‐Rate Wideband Codec (AMR‐WB) -- 8.7.6 Codec for Enhanced Voice Services (EVS) -- 8.7.7 Opus Codec IETF RFC 6716 -- References -- Chapter 9 Concealment of Erroneous or Lost Frames -- 9.1 Concepts for Error Concealment -- 9.1.1 Error Concealment by Hard Decision Decoding -- 9.1.2 Error Concealment by Soft Decision Decoding -- 9.1.3 Parameter Estimation -- 9.1.3.1 MAP Estimation -- 9.1.3.2 MS Estimation -- 9.1.4 The A Posteriori Probabilities -- 9.1.4.1 The A Priori Knowledge -- 9.1.4.2 The Parameter Distortion Probabilities -- 9.1.5 Example: Hard Decision vs. Soft Decision -- 9.2 Examples of Error Concealment Standards -- 9.2.1 Substitution and Muting of Lost Frames -- 9.2.2 AMR Codec: Substitution and Muting of Lost Frames -- 9.2.3 EVS Codec: Concealment of Lost Packets -- 9.3 Further Improvements -- References -- Chapter 10 Bandwidth Extension of Speech Signals -- 10.1 BWE Concepts -- 10.2 BWE using the Model of Speech Production -- 10.2.1 Extension of the Excitation Signal -- 10.2.2 Spectral Envelope Estimation -- 10.2.2.1 Minimum Mean Square Error Estimation -- 10.2.2.2 Conditional Maximum A Posteriori Estimation -- 10.2.2.3 Extensions -- 10.2.2.4 Simplifications -- 10.2.3 Energy Envelope Estimation -- 10.3 Speech Codecs with Integrated BWE -- 10.3.1 BWE in the GSM Full‐Rate Codec. 10.3.2 BWE in the AMR Wideband Codec. |
| Record Nr. | UNINA-9910829806303321 |
Vary Peter
|
||
| Newark : , : John Wiley & Sons, Incorporated, , 2023 | ||
| Lo trovi qui: Univ. Federico II | ||
| ||