Beyond VoIP protocols [[electronic resource] ] : understanding voice technology and networking techniques for IP telephony / / Olivier Hersent, Jean-Pierre Petit, and David Gurle |
Autore | Hersent Olivier |
Pubbl/distr/stampa | Hoboken, NJ, : John Wiley, 2005 |
Descrizione fisica | 1 online resource (285 p.) |
Disciplina |
004.62
621.385 |
Altri autori (Persone) |
PetitJean-Pierre
GurleDavid |
Soggetto topico |
Internet telephony
Speech processing systems |
ISBN |
1-280-27632-0
9786610276325 0-470-02364-3 0-470-02363-5 |
Formato | Materiale a stampa ![]() |
Livello bibliografico | Monografia |
Lingua di pubblicazione | eng |
Nota di contenuto |
Beyond VoIP Protocols Understanding Voice Technology and Networking Techniques for IP Telephony; Contents; Glossary; List of Abbreviations; 1 Introduction; 1.1 The rebirth of VoIP; 1.2 Why beyond VoIP protocols?; 1.2.1 Selecting a voice coder; 1.2.2 Providing 'toll quality' . . . and more; 1.2.3 Controlling IP quality of service; 1.2.4 Dimensioning the network; 1.2.5 Unleashing the potential of multicast; 1.3 Scope of this book; 1.4 Intended audience; 1.5 Conclusion; 1.6 References; 2 Introduction to Speech-coding Techniques; 2.1 A primer on digital signal processing; 2.1.1 Introduction
2.1.2 Sampling and quantization2.1.3 The sampling theorem; 2.1.4 Quantization; 2.1.5 ITU G.711 A-law or μ-law, a basic coder at 64 kbit/s; 2.2 The basic tools of digital signal processing; 2.2.1 Why digital technology simplifies signal processing; 2.2.2 The Z transform and the transfer function; 2.2.3 Linear prediction for speech-coding schemes; 2.3 Overview of speech signals; 2.3.1 Narrow-band and wide-band encoding of audio signals; 2.3.2 Speech production: voiced, unvoiced, and plosive sounds; 2.3.3 A basic LPC vocoder: DOD LPC 10 2.3.4 Auditory perception used for speech and audio bitrate reduction2.4 Advanced voice coder algorithms; 2.4.1 Adaptive quantizers. NICAM and ADPCM coders; 2.4.2 Differential predictive quantization; 2.4.3 Long-term prediction for speech signal; 2.4.4 Vector quantization; 2.4.5 Entropy coding; 2.5 Waveform coders. ADPCM ITU-T G.726; 2.5.1 Coder specification . . . from digital test sequences to C code; 2.5.2 Embedded version of the G.726 ADPCM coder G.727; 2.5.3 Wide-band speech coding using a waveform-type coder; 2.6 Hybrids and analysis by synthesis (ABS) speech coders; 2.6.1 Principle 2.6.2 The GSM full-rate RPE-LTP speech coder (GSM 06.10)2.7 Codebook-excited linear predictive (CELP) coders; 2.7.1 ITU-T 8-kbit/s CS-ACELP G.729; 2.7.2 ITU-T G.723.1: dual-rate speech coder for multimedia communications transmitting at 5.3 kbit/s and 6.3 kbit/s; 2.7.3 The low-delay CELP coding scheme: ITU-T G.728; 2.7.4 The AMR and AMR-WB coders; 2.8 Quality of speech coders; 2.8.1 Speech quality assessment; 2.8.2 ACR subjective test, mean opinion score (MOS); 2.8.3 Other methods of assessing speech quality; 2.8.4 Usage of MOS; 2.9 Conclusion on speech-coding techniques and their near future 2.9.1 The race for low-bitrate coders2.9.2 Optimization of source encoding and channel encoding; 2.9.3 The future; 2.10 References; 2.10.1 Articles; 2.10.2 Books; 2.11 Annexes; 2.11.1 Main characteristics of ITU-T standardized speech coders; 2.11.2 Main characteristics of cellular mobile standardized speech coders; 3 Voice Quality; 3.1 Introduction; 3.2 Reference VoIP media path; 3.3 Echo in a telephone network; 3.3.1 Talker echo, listener echo; 3.3.2 Electric echo; 3.3.3 Acoustic echo; 3.3.4 How to limit echo; 3.4 Delay; 3.4.1 Influence of the operating system 3.4.2 The influence of the jitter buffer policy on delay |
Record Nr. | UNINA-9910143578103321 |
Hersent Olivier
![]() |
||
Hoboken, NJ, : John Wiley, 2005 | ||
![]() | ||
Lo trovi qui: Univ. Federico II | ||
|
Beyond VoIP protocols [[electronic resource] ] : understanding voice technology and networking techniques for IP telephony / / Olivier Hersent, Jean-Pierre Petit, and David Gurle |
Autore | Hersent Olivier |
Pubbl/distr/stampa | Hoboken, NJ, : John Wiley, 2005 |
Descrizione fisica | 1 online resource (285 p.) |
Disciplina |
004.62
621.385 |
Altri autori (Persone) |
PetitJean-Pierre
GurleDavid |
Soggetto topico |
Internet telephony
Speech processing systems |
ISBN |
1-280-27632-0
9786610276325 0-470-02364-3 0-470-02363-5 |
Formato | Materiale a stampa ![]() |
Livello bibliografico | Monografia |
Lingua di pubblicazione | eng |
Nota di contenuto |
Beyond VoIP Protocols Understanding Voice Technology and Networking Techniques for IP Telephony; Contents; Glossary; List of Abbreviations; 1 Introduction; 1.1 The rebirth of VoIP; 1.2 Why beyond VoIP protocols?; 1.2.1 Selecting a voice coder; 1.2.2 Providing 'toll quality' . . . and more; 1.2.3 Controlling IP quality of service; 1.2.4 Dimensioning the network; 1.2.5 Unleashing the potential of multicast; 1.3 Scope of this book; 1.4 Intended audience; 1.5 Conclusion; 1.6 References; 2 Introduction to Speech-coding Techniques; 2.1 A primer on digital signal processing; 2.1.1 Introduction
2.1.2 Sampling and quantization2.1.3 The sampling theorem; 2.1.4 Quantization; 2.1.5 ITU G.711 A-law or μ-law, a basic coder at 64 kbit/s; 2.2 The basic tools of digital signal processing; 2.2.1 Why digital technology simplifies signal processing; 2.2.2 The Z transform and the transfer function; 2.2.3 Linear prediction for speech-coding schemes; 2.3 Overview of speech signals; 2.3.1 Narrow-band and wide-band encoding of audio signals; 2.3.2 Speech production: voiced, unvoiced, and plosive sounds; 2.3.3 A basic LPC vocoder: DOD LPC 10 2.3.4 Auditory perception used for speech and audio bitrate reduction2.4 Advanced voice coder algorithms; 2.4.1 Adaptive quantizers. NICAM and ADPCM coders; 2.4.2 Differential predictive quantization; 2.4.3 Long-term prediction for speech signal; 2.4.4 Vector quantization; 2.4.5 Entropy coding; 2.5 Waveform coders. ADPCM ITU-T G.726; 2.5.1 Coder specification . . . from digital test sequences to C code; 2.5.2 Embedded version of the G.726 ADPCM coder G.727; 2.5.3 Wide-band speech coding using a waveform-type coder; 2.6 Hybrids and analysis by synthesis (ABS) speech coders; 2.6.1 Principle 2.6.2 The GSM full-rate RPE-LTP speech coder (GSM 06.10)2.7 Codebook-excited linear predictive (CELP) coders; 2.7.1 ITU-T 8-kbit/s CS-ACELP G.729; 2.7.2 ITU-T G.723.1: dual-rate speech coder for multimedia communications transmitting at 5.3 kbit/s and 6.3 kbit/s; 2.7.3 The low-delay CELP coding scheme: ITU-T G.728; 2.7.4 The AMR and AMR-WB coders; 2.8 Quality of speech coders; 2.8.1 Speech quality assessment; 2.8.2 ACR subjective test, mean opinion score (MOS); 2.8.3 Other methods of assessing speech quality; 2.8.4 Usage of MOS; 2.9 Conclusion on speech-coding techniques and their near future 2.9.1 The race for low-bitrate coders2.9.2 Optimization of source encoding and channel encoding; 2.9.3 The future; 2.10 References; 2.10.1 Articles; 2.10.2 Books; 2.11 Annexes; 2.11.1 Main characteristics of ITU-T standardized speech coders; 2.11.2 Main characteristics of cellular mobile standardized speech coders; 3 Voice Quality; 3.1 Introduction; 3.2 Reference VoIP media path; 3.3 Echo in a telephone network; 3.3.1 Talker echo, listener echo; 3.3.2 Electric echo; 3.3.3 Acoustic echo; 3.3.4 How to limit echo; 3.4 Delay; 3.4.1 Influence of the operating system 3.4.2 The influence of the jitter buffer policy on delay |
Record Nr. | UNINA-9910831181203321 |
Hersent Olivier
![]() |
||
Hoboken, NJ, : John Wiley, 2005 | ||
![]() | ||
Lo trovi qui: Univ. Federico II | ||
|
Beyond VoIP protocols : understanding voice technology and networking techniques for IP telephony / / Olivier Hersent, Jean-Pierre Petit, and David Gurle |
Autore | Hersent Olivier |
Pubbl/distr/stampa | Hoboken, NJ, : John Wiley, 2005 |
Descrizione fisica | 1 online resource (285 p.) |
Disciplina | 621.382/12 |
Altri autori (Persone) |
PetitJean-Pierre
GurleDavid |
Soggetto topico |
Internet telephony
Speech processing systems |
ISBN |
1-280-27632-0
9786610276325 0-470-02364-3 0-470-02363-5 |
Formato | Materiale a stampa ![]() |
Livello bibliografico | Monografia |
Lingua di pubblicazione | eng |
Nota di contenuto |
Beyond VoIP Protocols Understanding Voice Technology and Networking Techniques for IP Telephony; Contents; Glossary; List of Abbreviations; 1 Introduction; 1.1 The rebirth of VoIP; 1.2 Why beyond VoIP protocols?; 1.2.1 Selecting a voice coder; 1.2.2 Providing 'toll quality' . . . and more; 1.2.3 Controlling IP quality of service; 1.2.4 Dimensioning the network; 1.2.5 Unleashing the potential of multicast; 1.3 Scope of this book; 1.4 Intended audience; 1.5 Conclusion; 1.6 References; 2 Introduction to Speech-coding Techniques; 2.1 A primer on digital signal processing; 2.1.1 Introduction
2.1.2 Sampling and quantization2.1.3 The sampling theorem; 2.1.4 Quantization; 2.1.5 ITU G.711 A-law or μ-law, a basic coder at 64 kbit/s; 2.2 The basic tools of digital signal processing; 2.2.1 Why digital technology simplifies signal processing; 2.2.2 The Z transform and the transfer function; 2.2.3 Linear prediction for speech-coding schemes; 2.3 Overview of speech signals; 2.3.1 Narrow-band and wide-band encoding of audio signals; 2.3.2 Speech production: voiced, unvoiced, and plosive sounds; 2.3.3 A basic LPC vocoder: DOD LPC 10 2.3.4 Auditory perception used for speech and audio bitrate reduction2.4 Advanced voice coder algorithms; 2.4.1 Adaptive quantizers. NICAM and ADPCM coders; 2.4.2 Differential predictive quantization; 2.4.3 Long-term prediction for speech signal; 2.4.4 Vector quantization; 2.4.5 Entropy coding; 2.5 Waveform coders. ADPCM ITU-T G.726; 2.5.1 Coder specification . . . from digital test sequences to C code; 2.5.2 Embedded version of the G.726 ADPCM coder G.727; 2.5.3 Wide-band speech coding using a waveform-type coder; 2.6 Hybrids and analysis by synthesis (ABS) speech coders; 2.6.1 Principle 2.6.2 The GSM full-rate RPE-LTP speech coder (GSM 06.10)2.7 Codebook-excited linear predictive (CELP) coders; 2.7.1 ITU-T 8-kbit/s CS-ACELP G.729; 2.7.2 ITU-T G.723.1: dual-rate speech coder for multimedia communications transmitting at 5.3 kbit/s and 6.3 kbit/s; 2.7.3 The low-delay CELP coding scheme: ITU-T G.728; 2.7.4 The AMR and AMR-WB coders; 2.8 Quality of speech coders; 2.8.1 Speech quality assessment; 2.8.2 ACR subjective test, mean opinion score (MOS); 2.8.3 Other methods of assessing speech quality; 2.8.4 Usage of MOS; 2.9 Conclusion on speech-coding techniques and their near future 2.9.1 The race for low-bitrate coders2.9.2 Optimization of source encoding and channel encoding; 2.9.3 The future; 2.10 References; 2.10.1 Articles; 2.10.2 Books; 2.11 Annexes; 2.11.1 Main characteristics of ITU-T standardized speech coders; 2.11.2 Main characteristics of cellular mobile standardized speech coders; 3 Voice Quality; 3.1 Introduction; 3.2 Reference VoIP media path; 3.3 Echo in a telephone network; 3.3.1 Talker echo, listener echo; 3.3.2 Electric echo; 3.3.3 Acoustic echo; 3.3.4 How to limit echo; 3.4 Delay; 3.4.1 Influence of the operating system 3.4.2 The influence of the jitter buffer policy on delay |
Record Nr. | UNINA-9910877872903321 |
Hersent Olivier
![]() |
||
Hoboken, NJ, : John Wiley, 2005 | ||
![]() | ||
Lo trovi qui: Univ. Federico II | ||
|