Speech Quality of VoIP: Assessment and Prediction
暫譯: VoIP 語音品質:評估與預測

Alexander Raake

  • 出版商: Wiley
  • 出版日期: 2006-11-01
  • 售價: $5,120
  • 貴賓價: 9.5$4,864
  • 語言: 英文
  • 頁數: 336
  • 裝訂: Hardcover
  • ISBN: 0470030607
  • ISBN-13: 9780470030608
  • 海外代購書籍(需單獨結帳)

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Description 

Finally a comprehensive overview of speech quality in VoIP from the user’s perspective!

Speech Quality of VoIP is an essential guide to assessing the speech quality of VoIP networks, whilst addressing the implications for the design of VoIP networks and systems. This book bridges the gap between the technical network-world and the psychoacoustic world of quality perception.  Alexander Raake’s unique perspective combines awareness of the technical characteristics of VoIP networks and original research concerning the perception of speech transmitted across them.

Starting from the network designer’s point of view, the different characteristics of the network are addressed, and then linked to features perceived by users. This book provides an overview of the available knowledge on the principal, relevant aspects of speech and speech quality perception, of speech quality assessment, and of transmission properties of telephone and VoIP networks, and of the related perceptual features and resulting speech quality. Discussing new research into the specific time-varying degradations VoIP brings along, but also the considerable potential of quality improvement to be achieved with wideband speech transmission, Alexander Raake demonstrates how network and service characteristics impact on the users perception of quality.

Speech Quality of VoIP:

  • Offers an insight into speech quality of VoIP from a user's perspective.
  • Presents an overview of different modelling approaches and a parametric network-planning model for quality prediction in VoIP networks.
  • Draws on innovative new research on the quality degradation characteristic of VoIP.
  • Explains in detail how telephone speech quality can be greatly enhanced with VoIP’s wideband speech transmission capability.
  • Assesses the vast collection of references into the technical and scientific literature related to VoIP quality.
  • Illustrates concepts throughout with mathematical models, algorithms and simulations.

Speech Quality of VoIP is the definitive guide for researchers, engineers and network planners working in the field of VoIP, Quality of Service, and speech communication processing in telecommunications. Advanced undergraduate and graduate students on telecommunication and networking courses will also find this text an invaluable resource.

 

Table of Contents

Preface.

List of Abbreviations.

Introduction.

1 Speech Quality in Telephony.

1.1 Speech.

1.1.1 Speech Acoustics.

1.1.2 Speech Perception.

1.2 Speech Quality.

1.2.1 Definition of Quality.

1.2.2 Speech Quality Assessment.

1.2.3 Quality Elements.

1.2.4 Speech Quality and Quality of Service.

2 Speech Quality Measurement Methods.

2.1 Auditory Methods.

2.1.1 Utilitarian Methods.

2.1.2 Analytical Methods.

2.2 Instrumental Methods.

2.2.1 Signal-based Models.

2.2.2 Parameter-based Models.

2.2.3 Monitoring Models.

2.3 Speech Quality Measurement Methods: Summary.

3 Quality Elements and Quality Features of VoIP.

3.1 Speech Transmission Using Internet Protocol.

3.1.1 VoIP Applications .

3.1.2 VoIP Quality of Service.

3.2 Overview of Quality Elements.

3.3 Quality Elements and Related Features.

3.3.1 Voice Activity Detection (VAD).

3.3.2 Codecs.

3.3.3 Jitter.

3.3.4 Delay.

3.3.5 Packet Loss.

3.3.6 Bit Errors.

3.3.7 Talker Echo.

3.3.8 Listener Echo.

3.3.9 Echo Cancellers.

3.3.10 Loudness.

3.3.11 Noise.

3.3.12 Noise Suppression.

3.3.13 Linear Distortion and Wideband Transmission.

3.3.14 User Interfaces.

3.4 Quality Dimensions.

3.5 Combined Elements and Combined Features.

3.6 Listening and Conversational Features.

3.7 Desired Nature.

3.7.1 Context.

3.7.2 Cost.

3.7.3 Service Options.

3.8 Open Questions.

3.9 From Elements to Features: Modeling VoIP Speech Quality.

3.9.1 Signal-based Measures and VoIP.

3.9.2 ParametricModels and VoIP.

3.10 Quality Elements and Quality Features of VoIP: Summary.

4 Time-Varying Distortion: Quality Features and Modeling.

4.1 Microscopic Loss Behavior.

4.1.1 Random Packet Loss.

4.1.2 Dependent Packet Loss.

4.2 Macroscopic Loss Behavior.

4.2.1 Macroscopic Loss with Segments of Random Loss.

4.2.2 Macroscopic Loss with Segments of Dependent Loss.

4.3 Interactivity.

4.4 Packet Loss and Combined Impairments.

4.4.1 Additivity and Multidimensional Feature Space.

4.4.2 Microscopic Loss Behavior: Random Packet Loss.

4.4.3 Macroscopic Loss Behavior: 3-State Markov Loss.

4.4.4 Model: Packet Loss and Combined Impairments.

4.5 Time-Varying Distortion: Summary.

5 Wideband Speech, Linear and Non Linear Distortion: Quality Features and Modeling.

5.1 Wideband Speech: Improvement Over Narrowband.

5.1.1 Summary.

5.2 Bandpass-Filtered Speech.

5.2.1 Listening Test.

5.2.2 Parametric Impairment Model.

5.3 Wideband Codecs.

5.4 Desired Nature.

5.4.1 User Interfaces.

5.4.2 Content.

5.4.3 Summary.

6 From Elements to Features: Extensions of the E-model.

6.1 E-model: Packet Loss.

6.2 E-model: Additivity.

6.3 E-model: Wideband, Linear and Non-Linear Distortion.

7 Summary and Conclusions.

8 Outlook.

A Aspects of a Parametric Description of Time-Varying Distortion.

A.1 4-state Markov Chain: Sojourn in Bad and Good State.

A.2 Impact of Other Quality Elements on Packet Loss.

A.2.1 Forward Error Correction.

A.2.2 Jitter and Jitter Buffer.

A.3 Impairment under GSM Bit Errors.

B Simulation of Quality Elements.

B.1 PSTN/ISDN.

B.1.1 Instrumental Verification.

B.2 Mobile Networks.

B.3 VoIP.

B.3.1 Analog to ISDN Conversion.

B.3.2 VoIP Gateways.

B.3.3 IP Network Simulation.

B.3.4 Instrumental Verification.

B.4 Wideband Transmission.

B.5 User Interfaces.

B.5.1 Handset Telephones.

B.5.2 Headsets.

B.5.3 Wideband Handset and Hi-fi Phone.

B.5.4 Bandpass Filters.

B.5.5 Hands-free Terminals (HFTs)

B.6 Test Rooms.

B.7 Simulation of Quality Elements: Summary.

C Frequency Responses.

C.1 Transmission Bandpass.

C.1.1 Narrowband.

C.1.2 Wideband.

C.2 User Interfaces.

C.2.1 Headsets.

C.2.2 Wideband Handsets.

D Test Data Normalization and Transformation.

D.1 Equivalent-QMethod.

D.2 Method According to ITU-T Rec. P.833.

D.3 Linear Transformation.

D.4 Note onMOS-terminology.

E E-model Algorithm.

F Interactive Short Conversation Test Scenarios (iSCTs).

F.1 Example.

G Auditory Test Settings and Results.

G.1 Global System for Mobile (GSM): Short Conversation and Listening Only Test.

G.2 2-stateMarkov Loss: Listening Only Test.

G.3 Random Loss: Conversation Test.

G.3.1 Test Setup.

G.3.2 Test Procedure.

G.3.3 Subjects.

G.3.4 Details on Selected Results.

G.4 3-state Markov Loss: Conversation Test.

G.4.1 Test Setup.

G.4.2 Test Procedure and Test Subjects.

G.4.3 Details on Selected Results.

G.5 Speech Sound Quality and Content.

H Modeling Details.

H.1 Time-Varying Distortions.

H.1.1 Macroscopic Loss Behavior.

I Glossary.

Bibliography.

Index.

商品描述(中文翻譯)

最後,從使用者的角度提供了對VoIP語音品質的全面概述!

《VoIP語音品質》是評估VoIP網路語音品質的重要指南,同時考慮到VoIP網路和系統設計的影響。本書彌合了技術網路世界與質量感知的心理聲學世界之間的差距。亞歷山大·拉克(Alexander Raake)獨特的視角結合了對VoIP網路技術特性的認識以及有關通過這些網路傳輸的語音感知的原創研究。

從網路設計者的角度出發,探討了網路的不同特性,然後將其與使用者感知的特徵聯繫起來。本書提供了有關語音及語音品質感知的主要相關方面、語音品質評估以及電話和VoIP網路的傳輸特性及相關感知特徵和最終語音品質的可用知識概述。亞歷山大·拉克討論了VoIP所帶來的特定時間變化的劣化的新研究,以及通過寬頻語音傳輸實現的顯著質量改善潛力,展示了網路和服務特性如何影響使用者對品質的感知。

《VoIP語音品質》:
- 提供了從使用者的角度對VoIP語音品質的見解。
- 概述了不同的建模方法以及用於VoIP網路品質預測的參數化網路規劃模型。
- 借鑒了有關VoIP品質劣化特徵的創新研究。
- 詳細解釋了如何通過VoIP的寬頻語音傳輸能力大幅提升電話語音品質。
- 評估了與VoIP品質相關的技術和科學文獻的廣泛參考資料。
- 通過數學模型、算法和模擬全書貫穿說明概念。

《VoIP語音品質》是研究人員、工程師和網路規劃者在VoIP、服務品質和電信語音通信處理領域的權威指南。高年級本科生和研究生在電信和網路課程中也會發現這本書是無價的資源。

目錄
- 前言
- 縮寫詞表
- 介紹
- 1 電話中的語音品質
- 1.1 語音
- 1.1.1 語音聲學
- 1.1.2 語音感知
- 1.2 語音品質
- 1.2.1 品質定義
- 1.2.2 語音品質評估
- 1.2.3 品質要素
- 1.2.4 語音品質與服務品質
- 2 語音品質測量方法
- 2.1 聽覺方法
- 2.1.1 實用方法
- 2.1.2 分析方法
- 2.2 儀器方法
- 2.2.1 基於信號的模型
- 2.2.2 基於參數的模型
- 2.2.3 監控模型
- 2.3 語音品質測量方法:總結
- 3 VoIP的品質要素和品質特徵
- 3.1 使用網際網路協議的語音傳輸
- 3.1.1 VoIP應用
- 3.1.2 VoIP服務品質
- 3.2 品質要素概述
- 3.3 品質要素及相關特徵
- 3.3.1 語音活動檢測(VAD)
- 3.3.2 編解碼器
- 3.3.3 抖動
- 3.3.4 延遲
- 3.3.5 封包丟失
- 3.3.6 位元錯誤
- 3.3.7 說話者回聲
- 3.3.8 聽者回聲
- 3.3.9 回聲消除器
- 3.3.10 音量
- 3.3.11 噪音
- 3.3.12 噪音抑制
- 3.3.13 線性失真和寬頻傳輸
- 3.3.14 使用者介面
- 3.4 品質維度
- 3.5 組合要素和組合特徵
- 3.6 聆聽和對話特徵
- 3.7 期望特性
- 3.7.1 上下文
- 3.7.2 成本
- 3.7.3 服務選項
- 3.8 開放問題
- 3.9 從要素到特徵:建模VoIP語音品質
- 3.9.1 基於信號的測量和VoIP
- 3.9.2 參數模型和VoIP
- 3.10 VoIP的品質要素和品質特徵:總結
- 4 時變失真:品質特徵和建模
- 4.1 微觀損失行為
- 4.1.1 隨機封包損失
- 4.1.2 依賴性封包損失
- 4.2 宏觀損失行為
- 4.2.1 隨機損失段的宏觀損失
- 4.2.2 依賴性損失段的宏觀損失
- 4.3 互動性
- 4.4 封包損失和組合損害
- 4.4.1 可加性和多維特徵空間
- 4.4.2 微觀損失行為:隨機封包損失
- 4.4.3 宏觀損失行為:三狀態馬可夫損失
- 4.4.4 模型:封包損失和組合損害
- 4.5 時變失真:總結
- 5 寬頻語音、線性和非線性失真:品質特徵和建模
- 5.1 寬頻語音:相對於窄頻的改進
- 5.1.1 總結
- 5.2 帶通濾波的語音
- 5.2.1 聆聽測試
- 5.2.2 參數損害模型
- 5.3 寬頻編解碼器
- 5.4 期望特性
- 5.4.1 使用者介面
- 5.4.2 內容
- 5.4.3 總結
- 6 從要素到特徵:E模型的擴展
- 6.1 E模型:封包損失
- 6.2 E模型:可加性
- 6.3 E模型:寬頻、線性和非線性失真
- 7 總結與結論
- 8 展望
- A 時變失真的參數描述的各個方面
- A.1 四狀態馬可夫鏈:在壞狀態和好狀態的停留
- A.2 其他品質要素對封包損失的影響
- A.2.1 向前錯誤更正
- A.2.2 抖動和抖動緩衝區
- A.3 在GSM位元錯誤下的損害
- B 品質要素的模擬
- B.1 PSTN/ISDN
- B.1.1 儀器驗證
- B.2 行動網路
- B.3 VoIP
- B.3.1 類比到ISDN轉換
- B.3.2 VoIP閘道
- B.3.3 IP網路模擬
- B.3.4 儀器驗證
- B.4 寬頻傳輸
- B.5 使用者介面
- B.5.1 手機電話
- B.5.2 耳機
- B.5.3 寬頻手機和高保真電話
- B.5.4 帶通濾波器
- B.5.5 免持終端(HFT)
- B.6 測試室
- B.7 品質要素的模擬:總結
- C 頻率響應
- C.1 傳輸帶通
- C.1.1 窄頻
- C.1.2 寬頻
- C.2 使用者介面
- C.2.1 耳機
- C.2.2 寬頻耳機
- D 測試數據的標準化和轉換
- D.1 等效-Q方法
- D.2 根據ITU-T Rec. P.833的方法
- D.3 線性轉換
- D.4 關於MOS術語的說明
- E E模型算法
- F 互動短對話測試場景(iSCTs)
- F.1 範例
- G 聽覺測試設置和結果
- G.1 全球行動系統(GSM):短對話和僅聆聽測試
- G.2 二狀態馬可夫損失:僅聆聽測試
- G.3 隨機損失:對話測試
- G.3.1 測試設置
- G.3.2 測試程序
- G.3.3 受試者
- G.3.4 選定結果的詳細信息
- G.4 三狀態馬可夫損失:對話測試
- G.4.1 測試設置
- G.4.2 測試程序和受試者
- G.4.3 選定結果的詳細信息
- G.5 語音音質和內容
- H 建模細節
- H.1 時變失真
- H.1.1 宏觀損失行為
- I 詞彙表
- 參考文獻
- 索引