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Communications, IEEE Transactions on

Issue 4 • Date April 1982

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Displaying Results 1 - 25 of 29
  • [Front cover and table of contents]

    Publication Year: 1982 , Page(s): 0
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    Freely Available from IEEE
  • Guest Editorial: In a Second--How Many Bits?

    Publication Year: 1982 , Page(s): 565 - 566
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    Freely Available from IEEE
  • [Back cover]

    Publication Year: 1982 , Page(s): 0
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    Freely Available from IEEE
  • An Approach to the Implementation of a Discrete Cosine Transform

    Publication Year: 1982 , Page(s): 635 - 641
    Cited by:  Papers (3)  |  Patents (1)
    Save to Project icon | Request Permissions | Click to expandAbstract | PDF file iconPDF (752 KB)  

    An approach to the implementation of a discrete cosine transform (DCT) for application to coding speech is described. The approach is oriented toward single speech channel encoding. In addition, a detailed computer simulation of an adaptive transform coder is described. The purpose of the computer simulation is to determine the internal precision at various points in the implementation required to... View full abstract»

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  • Subjective Evaluation of Several Efficient Speech Coders

    Publication Year: 1982 , Page(s): 655 - 662
    Cited by:  Papers (24)
    Save to Project icon | Request Permissions | Click to expandAbstract | PDF file iconPDF (944 KB)  

    The voice quality of several 9.6 - 32 kbit/s coders is determined with an extensive set of subjective listening tests. Single encodings of μ255 PCM, adaptive differential PCM (ADPCM), subband coding (SBC), vocoder-driven adaptive transform coding (ATC), adaptive predictive coding (APC), and time domain harmonic scaling combined with SBC are compared in an idealized situation, that is, no adde... View full abstract»

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  • Efficient Coding and Speech Interpolation: Principles and Performance Characterization

    Publication Year: 1982 , Page(s): 769 - 779
    Cited by:  Papers (1)  |  Patents (4)
    Save to Project icon | Request Permissions | Click to expandAbstract | PDF file iconPDF (1168 KB)  

    The BPA (block priority assignment) system combines the principle of speech interpolation with waveform coding methods for the active speech signal. Coding with variable wordlength permits adapting the demand of the N speech sources to the capacity of the channel. The system emphasized in this paper not only varies the bits per source according to the number of... View full abstract»

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  • A 60 Channel PCM-ADPCM Converter

    Publication Year: 1982 , Page(s): 567 - 573
    Cited by:  Papers (14)  |  Patents (14)
    Save to Project icon | Request Permissions | Click to expandAbstract | PDF file iconPDF (760 KB)  

    Coding techniques based on adaptive linear prediction and quantization are well suited to signals carried by telephone channels and can provide, with a per channel rate of 32 kbits/s, a level of quality compatible with the specifications of the conventional 64 kbit/s rate. The ADPCM technique described in this paper features a simple adaptive quantization scheme and a tenth-order linear prediction... View full abstract»

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  • A Multirate Voice Digitizer Based Upon Vector Quantization

    Publication Year: 1982 , Page(s): 721 - 727
    Cited by:  Papers (9)  |  Patents (6)
    Save to Project icon | Request Permissions | Click to expandAbstract | PDF file iconPDF (848 KB)  

    The importance of integrating voice and data over digital networks has increased during the last few years primarily because of the growing popularity of such networks. Of particular interest are efficient voice digitizing terminals, capable of operating at various data rates in both circuit-switched and packet-switched data networks. Several such terminals, including two or more speech compressio... View full abstract»

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  • A Comparison of Measured and Calculated Speech Temporal Parameters Relevant to Speech Activity Detection

    Publication Year: 1982 , Page(s): 728 - 738
    Cited by:  Papers (54)  |  Patents (1)
    Save to Project icon | Request Permissions | Click to expandAbstract | PDF file iconPDF (872 KB)  

    This paper deals with the measurement and calculation of various speech temporal parameters of interest in an environment where speech activity detection is employed. In particular it is shown that, based on either a measurement or model of the probability density function (pdf) for silence durations for the case of zero talkspurt "hangover" or "fill-in," that the following temporal parameters can... View full abstract»

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  • TASI-E Communications System

    Publication Year: 1982 , Page(s): 803 - 807
    Cited by:  Papers (28)  |  Patents (13)
    Save to Project icon | Request Permissions | Click to expandAbstract | PDF file iconPDF (648 KB)  

    This paper describes a Bell System electronic circuit multiplier designated TASI-E. The paper covers operation, features, restrictions, and performance of this system. View full abstract»

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  • Real-Time Speech Coder Implementation on an Array Processor

    Publication Year: 1982 , Page(s): 615 - 620
    Cited by:  Papers (5)
    Save to Project icon | Request Permissions | Click to expandAbstract | PDF file iconPDF (712 KB)  

    The feasibility of a speech coding algorithm is most effectively indicated by its successful real-time implementation. The implementation effort presents issues distinct from those related to the development of the algorithm. Problems of real-time software design and debugging, system reliability, and implementation correctness must be addressed. In addition, the power of a general purpose array p... View full abstract»

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  • Multipath Search Coding of Stationary Signals with Applications to Speech

    Publication Year: 1982 , Page(s): 687 - 701
    Cited by:  Papers (17)
    Save to Project icon | Request Permissions | Click to expandAbstract | PDF file iconPDF (1632 KB)  

    This paper deals with the application of multipath search coding (MSC) concepts to the coding of stationary memoryless and correlated sources and of speech signals at a rate of one bit per sample. We have made use of three MSC classes: 1) codebook coding (vector quantization), 2) tree coding, and 3) trellis coding. This paper explains the performances of these coders and compares them both with th... View full abstract»

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  • Digital Speech Interpolation with Predicted Wordlength Assignment (PWA)

    Publication Year: 1982 , Page(s): 762 - 769
    Cited by:  Papers (1)
    Save to Project icon | Request Permissions | Click to expandAbstract | PDF file iconPDF (776 KB)  

    The predicted wordlength assignment system (PWA) is a digital speech interpolation method which avoids speech clipping and "freeze-out" distortion. Inactive sources are excluded by a speech detector. The active speech signals are coded with variable wordlengths (3-8 bits) at a sampling rate of 8 kHz. In an overload case, all active sources are still served, but at reduced wordlength. The required ... View full abstract»

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  • The Design of Trellis Waveform Coders

    Publication Year: 1982 , Page(s): 702 - 710
    Cited by:  Papers (69)
    Save to Project icon | Request Permissions | Click to expandAbstract | PDF file iconPDF (1040 KB)  

    New algorithms for the design of trellis encoding data compression systems are described. The mare algorithm uses a training sequence of actual data from a source to improve an initial trellis decoder. An additional algorithm extends the constraint length of a given decoder. Combined, these algorithms allow the automatic design of a trellis encoding system for a particular source. The algorithms' ... View full abstract»

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  • CELTIC Field Trial Results

    Publication Year: 1982 , Page(s): 808 - 814
    Cited by:  Papers (4)
    Save to Project icon | Request Permissions | Click to expandAbstract | PDF file iconPDF (600 KB)  

    The CELTIC system (Concentrateur Exploitant les Temps d'Inactivité des Circuits [1], designed to increase the capacity of submarine cable links, was developed in France by CIT-Alcatel under contract with the Centre National d'Etudes des Télécommunications (CNET). CELTIC uses the inactive periods of the circuits to carry additional active circuits.The development of CELTIC led to fie... View full abstract»

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  • Variable Frame Rate Transmission: A Review of Methodology and Application to Narrow-Band LPC Speech Coding

    Publication Year: 1982 , Page(s): 674 - 686
    Cited by:  Papers (20)  |  Patents (12)
    Save to Project icon | Request Permissions | Click to expandAbstract | PDF file iconPDF (1432 KB)  

    We review the variable frame rate (VFR) transmission methodology that we developed, implemented, and tested during the period 1973-1978 for efficiently transmitting LPC vocoder parameters extracted from the input speech at a fixed frame rate. In the VFR method, parameters are transmitted only when their values have changed sufficiently over the interval since their preceding transmission. We explo... View full abstract»

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  • High-Gain Digital Speech Interpolation with Adaptive Differential PCM Encoding

    Publication Year: 1982 , Page(s): 750 - 761
    Cited by:  Papers (13)  |  Patents (1)
    Save to Project icon | Request Permissions | Click to expandAbstract | PDF file iconPDF (1104 KB)  

    This paper describes a digital speech interpolationadaptive differential PCM bit reduction technique in which digital speech interpolation (DSI) is combined with ADPCM encoding. A highly sensitive speech detector, a voiceband data discriminator, and a variable rate ADPCM encoding are used to achieve a high compression ratio. The speech detector proposed in [1] detects speech signals above -51 dBm ... View full abstract»

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  • PLC-1: A TASI System for Small Trunk Groups

    Publication Year: 1982 , Page(s): 786 - 791
    Cited by:  Papers (3)  |  Patents (10)
    Save to Project icon | Request Permissions | Click to expandAbstract | PDF file iconPDF (480 KB)  

    The PLC-1 Private Line Voice Concentrator is introduced. This equipment uses the well-known TASI (time assignment speech interpolation) technique to reduce the number of transmission facilities required to transmit a number of trunks by a factor of up to two. In addition to reducing transmission facility requirements, the PLC-1 provides a very high level of management and diagnostic tools to the t... View full abstract»

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  • Subjective Quality of the Same Speech Transmission Conditions in Seven Different Countries

    Publication Year: 1982 , Page(s): 642 - 654
    Cited by:  Papers (13)
    Save to Project icon | Request Permissions | Click to expandAbstract | PDF file iconPDF (1208 KB)  

    Because of the diversity of subjective testing methods, it is not possible, in general, to compare speech quality measurements from different laboratories. A recent experiment had the aim of determining whether comparable results can be obtained when the same test is performed in several different countries. Tape recordings of speech samples in the native languages of Britain, Canada, France, Ital... View full abstract»

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  • Hybrid Companding Delta Modulation with Variable-Rate Sampling

    Publication Year: 1982 , Page(s): 593 - 599
    Cited by:  Papers (3)
    Save to Project icon | Request Permissions | Click to expandAbstract | PDF file iconPDF (672 KB)  

    Hybrid companding delta modulation (HCDM) is known to be superior in performance to other instantaneous or syllabic companding delta modulation systems [1]. To improve its performance or to reduce the bit rate further in coding speech, we propose to use a variable-rate sampling scheme in the HCDM system. The proposed system employs several different sampling rates but transmits the output binary s... View full abstract»

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  • Design of a Robust Baseband LPC Coder for Speech Transmission Over 9.6 Kbit/s Noisy Channels

    Publication Year: 1982 , Page(s): 663 - 673
    Cited by:  Papers (18)  |  Patents (16)
    Save to Project icon | Request Permissions | Click to expandAbstract | PDF file iconPDF (1320 KB)  

    This paper describes the design of a baseband LPC coder that transmits speech over 9.6 kbit/s (kilobit/second) synchronous channels with random bit errors of up to 1 percent. Presented are the results of our investigation of a number of aspects of the baseband LPC coder with the goal of maximizing the quality of the transmitted speech. Important among these aspects are: bandwidth of the baseband, ... View full abstract»

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  • A New Channel Bank with Block Companding

    Publication Year: 1982 , Page(s): 574 - 580
    Cited by:  Papers (3)  |  Patents (6)
    Save to Project icon | Request Permissions | Click to expandAbstract | PDF file iconPDF (840 KB)  

    Performance improvement of the nearly instantaneous companding PCM (or block companding PCM) is discussed. A simplified algorithm is proposed. The design and performance of a channel bank adopting the proposed algorithm is described. The channel bank can transmit 44 voiceband signals on a standard T1 line. It provides commercially acceptable speech quality as well as sufficient transparency for vo... View full abstract»

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  • Operational Evaluation of a Voice Concentrator Over AUTOVON Interswitch Trunks

    Publication Year: 1982 , Page(s): 792 - 802
    Cited by:  Papers (2)
    Save to Project icon | Request Permissions | Click to expandAbstract | PDF file iconPDF (856 KB)  

    This paper describes results of test and evaluation of a commercial voice concentrator applied to AUTOVON interswitch trunks between Feldberg, Germany and Ft. Detrick, MD. By virtue of time assignment speech interpolation (TASI) techniques, the voice concentrator provided approximately a 2-to-1 compression of voice channels onto trunks, with a configuration of 17 channels onto nine trunks selected... View full abstract»

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  • Analysis of a TASI System Employing Speech Storage

    Publication Year: 1982 , Page(s): 780 - 785
    Cited by:  Papers (6)  |  Patents (9)
    Save to Project icon | Request Permissions | Click to expandAbstract | PDF file iconPDF (544 KB)  

    This paper presents a performance analysis of a PLC-1 private line voice concentrator which uses speech interpolation to increase the capacity of transmission facilities. The PLC-1 employs speech storage. As a result it can be applied to relatively small trunk groups where statistics of loading patterns are particularly unfavorable. Speech impairments can be categorized as delay, gap modulation, a... View full abstract»

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  • Voice Coding and Tree Encoding Speech Compression Systems Based Upon Inverse Filter Matching

    Publication Year: 1982 , Page(s): 711 - 720
    Cited by:  Papers (2)
    Save to Project icon | Request Permissions | Click to expandAbstract | PDF file iconPDF (1056 KB)  

    Two speech compression systems based on codebooks of inverse filters produced by off-line linear predictive coding (LPC) and vector quantization (VQ) techniques are considered. The first system is a pitch excited vocoder that is a variation on a speech coding system based upon vector quantization. The encoder selects an LPC reverse filter from a finite codebook that best "matches" an observed fram... View full abstract»

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Aims & Scope

IEEE Transactions on Communications focuses on all telecommunications including telephone, telegraphy, facsimile, and point-to-point television by electromagnetic propagation.

 

 

Full Aims & Scope

Meet Our Editors

Editor-in-Chief
Robert Schober
University of British Columbia