The notion of perceptual coding, which is based on the concept of distortion masking by the signal being compressed, is developed. Progress in this field as a result of advances in classical coding theory, modeling of human perception, and digital signal processing, is described. It is proposed that fundamental limits in the science can be expressed by the semiquantitative concepts of perceptual entropy and the perceptual distortion-rate function, and current compression technology is examined in that framework. Problems and future research directions are summarized. >

Signal compression based on models of human perception

The past decade has witnessed substantial progress towards the application of low-rate speech coders to civilian and military communications as well as computer-related voice applications. Central to this progress has been the development of new speech coders capable of producing high-quality speech at low data rates. Most of these coders incorporate mechanisms to: represent the spectral properties of speech, provide for speech waveform matching, and "optimize" the coder's performance for the human ear. A number of these coders have already been adopted in national and international cellular telephony standards. The objective of this paper is to provide a tutorial overview of speech coding methodologies with emphasis on those algorithms that are part of the recent low-rate standards for cellular communications. Although the emphasis is on the new low-rate coders, we attempt to provide a comprehensive survey by covering some of the traditional methodologies as well. We feel that this approach will not only point out key references but will also provide valuable background to the beginner. The paper starts with a historical perspective and continues with a brief discussion on the speech properties and performance measures. We then proceed with descriptions of waveform coders, sinusoidal transform coders, linear predictive vocoders, and analysis-by-synthesis linear predictive coders. Finally, we present concluding remarks followed by a discussion of opportunities for future research. >

Speech coding: a tutorial review

Since even before the time of Alexander Graham Bell's revolutionary invention, engineers and scientists have studied the phenomenon of speech communication with an eye on creating more efficient and effective systems of human-to-human and human-to-machine communication. Starting in the 1960s, digital signal processing (DSP), assumed a central role in speech studies, and today DSP is the key to realizing the fruits of the knowledge that has been gained through decades of research. Concomitant advances in integrated circuit technology and computer architecture have aligned to create a technological environment with virtually limitless opportunities for innovation in speech communication applications. In this text, we highlight the central role of DSP techniques in modern speech communication research and applications. We present a comprehensive overview of digital speech processing that ranges from the basic nature of the speech signal, through a variety of methods of representing speech in digital form, to applications in voice communication and automatic synthesis and recognition of speech. The breadth of this subject does not allow us to discuss any aspect of speech processing to great depth; hence our goal is to provide a useful introduction to the wide range of important concepts that comprise the field of digital speech processing. A more comprehensive treatment will appear in the forthcoming book, Theory and Application of Digital Speech Processing [101].

https://research.iaun.ac.ir/pd/mahmoodian/pdfs/UploadFile_2643.pdf

An Introduction to Digital Speech Processing

Providing voice and data communications to people away from their wireline telephones has become a major communication frontier. This frontier is being penetrated by evolving approaches to portable communications, e.g., cordless telephones, mobile radiotelephone, and radio paging. However, these approaches have many limitations; none can provide universal portable communications services. This paper discusses limitations of the evolving approaches and considers objectives and approaches for providing more universal digital portable communications as an integrated part of telephone exchange networks. These more universal communications could be accomplished by using demand-assigned radio links for the last thousand feet or so of telephone loops and sharing the remainder of the fixed distribution facilities. Fixed radio ports as integrated parts of telephone distribution networks could be placed throughout service regions. Efficient use of the radio spectrum could be insured by the planned reuse of radio frequencies throughout the regions. The severe multipath radio propagation environment within and around buildings that strongly influences the design of portable communications systems is described in the paper. System configurations and radio link techniques, that can provide reliable communications in the multipath environment, are discussed. Radio system calculations are illustrated for radio ports with 30-ft-high antennas in residential areas. The calculations indicate that radio link availability would be greater than 99 percent for 2000-ft port separations and 5-mW portable transmitters. Reuse of frequencies would require dividing the allocated frequency band into segments for use at adjacent ports. Calculations suggest that link availability in the cochannel interference environment would be greater than 99 percent, if 25 to 35 segments were used in residential areas.

Bishnu S. Atal

Papers

Speech coding