Details

Voice and Audio Compression for Wireless Communications


Voice and Audio Compression for Wireless Communications


IEEE Press 2. Aufl.

von: Lajos Hanzo, F. Clare A. Somerville, Jason Woodard

197,99 €

Verlag: Wiley
Format: PDF
Veröffentl.: 05.06.2008
ISBN/EAN: 9780470516027
Sprache: englisch
Anzahl Seiten: 880

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Beschreibungen

Voice communications remains the most important facet of mobile radio services, which may be delivered over conventional fixed links, the Internet or wireless channels. This all-encompassing volume reports on the entire 50-year history of voice compression, on recent audio compression techniques and the protection as well as transmission of these signals in hostile wireless propagation environments. <p><b><i>Audio and Voice Compression for Wireless and Wireline Communications, Second Edition</i></b> is divided into four parts with Part I covering the basics, while Part II outlines the design of analysis-by-synthesis coding, including a 100-page chapter on virtually all existing standardised speech codecs.  The focus of Part III is on wideband and audio coding as well as transmission. Finally, Part IV concludes the book with a range of very low rate encoding techniques, scanning a range of research-oriented topics.</p> <ul> <li>Fully updated and revised second edition of “Voice Compression and Communications”, expanded to cover Audio features</li> <li>Includes two new chapters, on narrowband and wideband AMR coding, and MPEG audio coding</li> <li>Addresses the new developments in the field of wideband speech and audio compression</li> <li>Covers compression, error resilience and error correction coding, as well as transmission aspects, including cutting-edge turbo transceivers</li> <li>Presents both the historic and current view of speech compression and communications.</li> </ul> <p>Covering fundamental concepts in a non-mathematical way before moving to detailed discussions of theoretical principles, future concepts and solutions to various specific wireless voice communication problems, this book will appeal to both advanced readers and those with a background knowledge of signal processing and communications. </p>
<p>About the Authors xxi</p> <p>Other Wiley and IEEE Press Books on Related Topics xxiii</p> <p>Preface and Motivation xxv</p> <p>Acknowledgements xxxv</p> <p>I Speech Signals and Waveform Coding 1</p> <p>1 Speech Signals and an Introduction to Speech Coding 3</p> <p>1.1 Motivation of Speech Compression 3</p> <p>1.2 Basic Characterisation of Speech Signals 4</p> <p>1.3 Classification of Speech Codecs 8</p> <p>1.4 Waveform Coding 11</p> <p>1.5 Chapter Summary 28</p> <p>2 Predictive Coding 29</p> <p>2.1 Forward-Predictive Coding 29</p> <p>2.2 DPCMCodec Schematic 30</p> <p>2.3 Predictor Design 31</p> <p>2.4 Adaptive One-word-memory Quantisation 39</p> <p>2.5 DPCM Performance 40</p> <p>2.6 Backward-adaptive Prediction 42</p> <p>2.7 The 32 kbps G.721 ADPCM Codec 47</p> <p>2.8 Subjective and Objective Speech Quality 53</p> <p>2.9 Variable-rateG.726andEmbeddedG.727ADPC54</p> <p>2.10 Rate-distortion in Predictive Coding 62</p> <p>2.11 Chapter Summary 67</p> <p>II Analysis-by-Synthesis Coding 69</p> <p>3 Analysis-by-Synthesis Principles 71</p> <p>3.1 Motivation 71</p> <p>3.2 Analysis-by-Synthesis Codec Structure 72</p> <p>3.3 The Short-term Synthesis Filter 73</p> <p>3.4 Long-term Prediction 76</p> <p>3.5 Excitation Models 85</p> <p>3.6 Adaptive Short-term and Long-term Post-Filtering 88</p> <p>3.7 Lattice-based Linear Prediction 90</p> <p>3.8 Chapter Summary 97</p> <p>4 Speech Spectral Quantisation 99</p> <p>4.1 Log-area Ratios 99</p> <p>4.2 Line Spectral Frequencies 103</p> <p>4.3 Vector Quantisation of Spectral Parameters 115</p> <p>4.4 Spectral Quantisers for Wideband Speech Coding 123</p> <p>4.5 Chapter Summary 138</p> <p>5 Regular Pulse Excited Coding 139</p> <p>5.1 Theoretical Background 139</p> <p>5.2 The 13 kbps RPE-LTP GSM Speech Encoder 146</p> <p>5.3 The 13 kbps RPE-LTP GSM Speech Decoder 151</p> <p>5.4 Bit-sensitivity of the 13 kbps GSM RPE-LTP Codec 153</p> <p>5.5 Application Example: A Tool-box Based Speech Transceiver 154</p> <p>5.6 Chapter Summary 157</p> <p>6 Forward-Adaptive Code Excited Linear Prediction 159</p> <p>6.1 Background 159</p> <p>6.2 The Original CELP Approach 160</p> <p>6.3 Fixed Codebook Search 163</p> <p>6.4 CELP Excitation Models 165</p> <p>6.5 Optimisation of  the CELP Codec Parameters 174</p> <p>6.6 The Error Sensitivity of CELP Codecs 192</p> <p>6.7 Application Example: A Dual-mode 3.1 kBd Speech Transceiver 204</p> <p>6.8 Multi-slot PRMA Transceiver 218</p> <p>6.9 Chapter Summary 223</p> <p>7 Standard Speech Codecs 225</p> <p>7.1 Background 225</p> <p>7.2 The US DoD FS-1016 4.8 kbps CELP Codec 225</p> <p>7.3 The 7.95 kbps Pan-American Speech Codec – Known as IS-54 DAMPS Codec 231</p> <p>7.4 The 6.7 kbps Japanese Digital Cellular System’s Speech Codec 235</p> <p>7.5 The Qualcomm Variable Rate CELPCodec 237</p> <p>7.6 Japanese Half-rate Speech Codec 245</p> <p>7.7 The Half-rate GSM Speech Codec 253</p> <p>7.8 The 8 kbps G.729 Codec 257</p> <p>7.9 The Reduced Complexity G.729 Annex A Codec 278</p> <p>7.10 The 12.2 kbps Enhanced Full-rate GSM Speech Codec 282</p> <p>7.11 The Enhanced Full-rate 7.4 kbps IS-136 Speech Codec 287</p> <p>7.12 The ITU G.723.1 Dual-rate Codec 292</p> <p>7.13 Advanced Multirate JD-CDMA Transceiver 302</p> <p>7.14 Chapter Summary 327</p> <p>8 Backward-adaptive Code Excited Linear Prediction 331</p> <p>8.1 Introduction 331</p> <p>8.2 Motivation and Background 331</p> <p>8.3 Backward-adaptiveG728CodecSchematic 334</p> <p>8.4 Backward-adaptiveG728CodingAlgorithm 336</p> <p>8.5 Reduced-rate G728-like Codec: Variable-length Excitation Vector 351</p> <p>8.6 The Effects of Long-term Prediction 354</p> <p>8.7 Closed-loop Codebook Training 359</p> <p>8.8 Reduced-rate G728-like Codec: Constant-length Excitation Vector 364</p> <p>8.9 Programmable-rate 8–4 kbps Low-delay CELP Codecs 365</p> <p>8.10 Backward-adaptive Error Sensitivity Issues 381</p> <p>8.11 A Low-delay Multimode Speech Transceiver 388</p> <p>8.12 Chapter Summary 392</p> <p>III Wideband Speech, MPEG-4 Audio and Their Transmission 393</p> <p>9 Wideband Speech Coding 395</p> <p>9.1 Sub-band-ADPCM Wideband Coding at 64 kbps 395</p> <p>9.2 Wideband Transform-coding at 32 kbps 413</p> <p>9.3 Sub-band-split Wideband CELPCodecs 416</p> <p>9.4 Fullband Wideband A CELPCoding 420</p> <p>9.5 A Turbo-coded Burst-by-burst Adaptive Wideband Speech Transceiver 425</p> <p>9.6 Turbo-detected Unequal Error Protection Irregular Convolutional Coded AMR-WB Transceivers 442</p> <p>9.7 The AMR-WB+AudioCodec 454</p> <p>9.8 Chapter Summary 466</p> <p>10 MPEG-4 Audio Compression and Transmission 469</p> <p>10.1 OverviewofMPEG-4Audio 469</p> <p>10.2 General Audio Coding 471</p> <p>10.3 Speech Coding in MPEG-4 Audio 495</p> <p>10.4 MPEG-4CodecPerformance 503</p> <p>10.5 MPEG-4 Space–time Block Coded OFDM Audio Transceiver 505</p> <p>10.6 Turbo-detected Space–time Trellis Coded MPEG-4 Audio Transceivers 516</p> <p>10.7 Turbo-detected Space–time Trellis Coded MPEG-4 Versus AMR-WB Speech Transceivers 525</p> <p>10.8 Chapter Summary 534</p> <p>IV Very Low-rate Coding and Transmission 537</p> <p>11 Overview of Low-rate Speech Coding 539</p> <p>11.1 Low-bitrate Speech Coding 539</p> <p>11.2 Linear Predictive Coding Model 553</p> <p>11.3 Speech Quality Measurements 557</p> <p>11.4 Speech Database 560</p> <p>11.5 Chapter Summary 563</p> <p>12 Linear Predictive Vocoder 565</p> <p>12.1 Overview of a Linear Predictive Vocoder 565</p> <p>12.2 Line Spectrum Frequencies Quantisation 566</p> <p>12.3 Pitch Detection 571</p> <p>12.4 Unvoiced Frames 583</p> <p>12.5 Voiced Frames 584</p> <p>12.6 Adaptive Postfilter 585</p> <p>12.7 Pulse Dispersion Filter 588</p> <p>12.8 Results for Linear Predictive Vocoder 592</p> <p>12.9 Chapter Summary 597</p> <p>13 Wavelets and Pitch Detection 599</p> <p>13.1 Conceptual Introduction to Wavelets 599</p> <p>13.2 Introduction to Wavelet Mathematics 602</p> <p>13.3 Preprocessing the Wavelet Transform Signal 607</p> <p>13.4 Voiced–unvoiced Decision 610</p> <p>13.5 Wavelet-based Pitch Detector 612</p> <p>13.6 Chapter Summary 619</p> <p>14 Zinc Function Excitation 621</p> <p>14.1 Introduction 621</p> <p>14.2 Overview of Prototype Waveform Interpolation Zinc Function Excitation 622</p> <p>14.3 Zinc Function Modelling 627</p> <p>14.4 Pitch Detection 631</p> <p>14.5 Voiced Speech 635</p> <p>14.6 Excitation Interpolation Between Prototype Segments 639</p> <p>14.7 Unvoiced Speech 645</p> <p>14.8 Adaptive Postfilter 645</p> <p>14.9 Results for Single Zinc Function Excitation 646</p> <p>14.10 Error Sensitivity of the 1.9 kbps PWI-ZFE Coder 649</p> <p>14.11 Multiple Zinc Function Excitation 654</p> <p>14.12 A Sixth-rate, 3.8 kbps GSM-like Speech Transceiver 661</p> <p>14.13 Chapter Summary 665</p> <p>15 Mixed-multiband Excitation 667</p> <p>15.1 Introduction 667</p> <p>15.2 Overview of Mixed-multiband Excitation 668</p> <p>15.3 Finite Impulse Response Filter 671</p> <p>15.4 Mixed-multiband Excitation Encoder 673</p> <p>15.5 Mixed-multiband Excitation Decoder 676</p> <p>15.6 Performance of the Mixed-multiband Excitation Coder 680</p> <p>15.7 A Higher Rate 3.85 kbps Mixed-multiband Excitation Scheme 686</p> <p>15.8 A 2.35 kbps Joint-detection-based CDMA Speech Transceiver 691</p> <p>15.9 Chapter Summary 699</p> <p>16 Sinusoidal Transform Coding Below 4 kbps 701</p> <p>16.1 Introduction 701</p> <p>16.2 Sinusoidal Analysis of Speech Signals 702</p> <p>16.3 Sinusoidal Synthesis of Speech Signals 704</p> <p>16.4 Low-bitrate Sinusoidal Coders 705</p> <p>16.5 Incorporating Prototype Waveform Interpolation 709</p> <p>16.6 Encoding the Sinusoidal Frequency Component 710</p> <p>16.7 Determining the Excitation Components 712</p> <p>16.8 Quantising the Excitation Parameters 720</p> <p>16.9 Sinusoidal Transform Decoder 728</p> <p>16.10 Speech Coder Performance 730</p> <p>16.11 Chapter Summary 736</p> <p>17 Conclusions on Low-rate Coding 737</p> <p>17.1 Summary 737</p> <p>17.2 Listening Tests 738</p> <p>17.3 Summary of Very-low-rate Coding 739</p> <p>17.4 Further Research 741</p> <p>18 Comparison of Speech Codecs and Transceivers 743</p> <p>18.1 Background to Speech Quality Evaluation 743</p> <p>18.2 Objective Speech Quality Measures 744</p> <p>18.3 Subjective Measures 752</p> <p>18.4 Comparison of Subjective and Objective Measures 753</p> <p>18.5 Subjective Speech Quality of Various Codecs 755</p> <p>18.6 Error Sensitivity Comparison of Various Codecs 757</p> <p>18.7 Objective Speech Performance of Various Transceivers 757</p> <p>18.8 Chapter Summary 764</p> <p>19 The Voice over Internet Protocol 765</p> <p>19.1 Introduction 765</p> <p>19.2 Session Initiation Protocol 766</p> <p>19.3 H.323Standards 774</p> <p>19.4 Real-time Transport Protocol 778</p> <p>19.5 Conclusion 781</p> <p>A Constructing the Quadratic Spline Wavelets 783</p> <p>B Zinc Function Excitation 787</p> <p>C Probability Density Function for Amplitudes 793</p> <p>Bibliography 797</p> <p>Index 825</p> <p>Author Index 834</p>
"I recommend this book as a useful reference on voice and audio compression for wireless communication." (<i>Computing Reviews</i>, June 10, 2008)
<b>Lajos Hanzo</b> has been with the Department of Electronics and Computer Science, University of Southampton, UK, since 1986 where he holds the Chair in Telecommunications. He received his degree in electronics in 1976 and his doctorate in 1983. During his career in telecommunications he has held various research and academic posts in Hungary, Germany and the UK. Lajos is a Fellow of the IEEE and the IEE as well as an IEEE Distinguished Lecturer. He has co-authored 11 books on mobile radio communications, published over 500 research papers and been awarded a number of distinctions. <p><b>Clare Somerville</b> received M.Eng and Ph.D degrees in Electronic Engineering in 1995 and 1999, respectively, from the University of Southampton. From 1995 to 1998 she performed research into low bit rate speech coders for wireless communications. Following a spell of research with the Global Wireless Systems Research department, Bell Laboratories, Swindon, U.K, she moved to PicoChip in Bath, UK. Her current research involves real-time services over wireless networks with a packet air interface.</p> <p><b>Jason Woodard</b> received an MA in Physics from the University of Oxford in 1991, then an MSc in Electronics and a PhD in Speech Coding from the University of Southampton in 1992 and 1995 respectively. From 1995 to 1998 he worked as a Research Fellow at the University of Southampton, researching error correction coding, especially turbo codes. In 1998 he joined the Wireless Technology Practice at PA Consulting Group in Cambridge. At the start of 1999 he was transferred from PA to UbiNetics Ltd, where he is responsible for the development and implementation of bit rate processing algorithms, including turbo codes, for 3rd Generation mobile communications products.</p>
Voice communications remains the most important facet of mobile radio services, which may be delivered over conventional fixed links, the Internet or wireless channels. This all-encompassing volume reports on the entire 50-year history of voice compression, on recent audio compression techniques and the protection as well as transmission of these signals in hostile wireless propagation environments. <p><i>Audio and Voice Compression for Wireless and Wireline Communications</i>, Second Edition is divided into four parts with Part I covering the basics, while Part II outlines the design of analysis-by-synthesis coding, including a 100-page chapter on virtually all existing standardised speech codecs.  The focus of Part III is on wideband and audio coding as well as transmission. Finally, Part IV concludes the book with a range of very low rate encoding techniques, scanning a range of research-oriented topics.</p> <p>Covering fundamental concepts in a non-mathematical way before moving to detailed discussions of theoretical principles, future concepts and solutions to various specific wireless voice communication problems, this book will appeal to both advanced readers and those with a background knowledge of signal processing and communications.</p> <ul> <li>Fully updated and revised second edition of "Voice Compression and Communications", expanded to cover Audio features</li> <li>Includes two new chapters, on narrowband and wideband AMR coding, and MPEG audio coding</li> <li>Addresses the new developments in the field of wideband speech and audio compression</li> <li>Covers compression, error resilience and error correction coding, as well as transmission aspects, including cutting-edge turbo transceivers</li> <li>Presents both the historic and current view of speech compression and communications.</li> </ul>

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