Principles of Speech Coding (Hardcover)
Tokunbo Ogunfunmi, Madihally Narasimha
- 出版商: CRC
- 出版日期: 2010-04-21
- 定價: $3,600
- 售價: 6.0 折 $2,160
- 語言: 英文
- 頁數: 381
- 裝訂: Hardcover
- ISBN: 0849374286
- ISBN-13: 9780849374289
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相關分類:
通訊系統 Communication-systems
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商品描述
It is becoming increasingly apparent that all forms of communication—including voice—will be transmitted through packet-switched networks based on the Internet Protocol (IP). Therefore, the design of modern devices that rely on speech interfaces, such as cell phones and PDAs, requires a complete and up-to-date understanding of the basics of speech coding.
Outlines key signal processing algorithms used to mitigate impairments to speech quality in VoIP networks
Offering a detailed yet easily accessible introduction to the field, Principles of Speech Coding provides an in-depth examination of the underlying signal processing techniques used in speech coding. The authors present coding standards from various organizations, including the International Telecommunication Union (ITU). With a focus on applications such as Voice-over-IP telephony, this comprehensive text covers the most recent research findings on topics including:
- A general introduction to speech processing
- Digital signal processing concepts
- Sampling theory and related topics
- Principles of pulse code modulation (PCM) and adaptive differential pulse code modulation (ADPCM) standards
- Linear prediction (LP) and use of the linear predictive coding (LPC) model
- Vector quantization and its applications in speech coding
- Case studies of practical speech coders from ITU and others
- The Internet low-bit-rate coder (ILBC)
Developed from the authors’ combined teachings, this book also illustrates its contents by providing a real-time implementation of a speech coder on a digital signal processing chip. With its balance of theory and practical coverage, it is ideal for senior-level undergraduate and graduate students in electrical and computer engineering. It is also suitable for engineers and researchers designing or using speech coding systems in their work.
商品描述(中文翻譯)
越來越明顯的是,包括語音在內的所有形式的通信將通過基於互聯網協議(IP)的分組交換網絡進行傳輸。因此,依賴語音界面的現代設備(如手機和個人數字助理)的設計,需要對語音編碼的基礎有全面且最新的理解。
《語音編碼原理》提供了對語音編碼中使用的底層信號處理技術進行深入研究的詳細且易於理解的介紹。作者介紹了包括國際電信聯盟(ITU)在內的各種組織的編碼標準。本書重點關注語音互聯網電話等應用,涵蓋了最新的研究成果,包括以下主題:
- 語音處理的一般介紹
- 數字信號處理概念
- 取樣理論及相關主題
- 脈衝編碼調變(PCM)和自適應差分脈衝編碼調變(ADPCM)標準的原理
- 線性預測(LP)和線性預測編碼(LPC)模型的應用
- 向量量化及其在語音編碼中的應用
- ITU和其他組織的實用語音編碼器案例研究
- 互聯網低比特率編碼器(ILBC)
本書是作者們結合教學經驗而開發的,並通過在數字信號處理芯片上實時實現語音編碼器來說明其內容。由於理論和實踐兼顧,本書非常適合電氣工程和計算機工程的高年級本科生和研究生。同時,對於在工作中設計或使用語音編碼系統的工程師和研究人員也很適用。