Principles of Speech Coding (Hardcover)
暫譯: 語音編碼原理 (精裝版)
Tokunbo Ogunfunmi, Madihally Narasimha
- 出版商: CRC
- 出版日期: 2010-04-21
- 售價: $3,600
- 貴賓價: 9.5 折 $3,420
- 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)
隨著所有形式的通信——包括語音——越來越明顯地將通過基於網際網路協議(IP)的封包交換網路傳輸,因此,依賴語音介面的現代設備(如手機和個人數位助理)的設計需要對語音編碼的基本原理有全面且最新的理解。
**概述在VoIP網路中用於減輕語音品質損害的關鍵信號處理演算法**
《語音編碼原理》提供了一個詳細但易於理解的領域介紹,深入探討語音編碼中使用的基本信號處理技術。作者介紹了來自各種組織的編碼標準,包括國際電信聯盟(ITU)。本書專注於如語音傳輸網路(Voice-over-IP)等應用,涵蓋了最近的研究成果,主題包括:
- 語音處理的一般介紹
- 數位信號處理概念
- 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)及線性預測編碼(LPC)模型的使用
- 向量量化及其在語音編碼中的應用
- 來自ITU及其他機構的實用語音編碼器案例研究
- 網際網路低位元率編碼器(ILBC)
本書基於作者的共同教學經驗,還通過提供在數位信號處理晶片上實時實現語音編碼器的示例來說明其內容。理論與實務的平衡使其成為電機與計算機工程的高年級本科生和研究生的理想選擇,也適合在工作中設計或使用語音編碼系統的工程師和研究人員。