50 research outputs found

    A study and experiment plan for digital mobile communication via satellite

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    The viability of mobile communications is examined within the context of a frequency division multiple access, single channel per carrier satellite system emphasizing digital techniques to serve a large population of users. The intent is to provide the mobile users with a grade of service consistant with the requirements for remote, rural (perhaps emergency) voice communications, but which approaches toll quality speech. A traffic model is derived on which to base the determination of the required maximum number of satellite channels to provide the anticipated level of service. Various voice digitalization and digital modulation schemes are reviewed along with a general link analysis of the mobile system. Demand assignment multiple access considerations and analysis tradeoffs are presented. Finally, a completed configuration is described

    Time and frequency domain algorithms for speech coding

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    The promise of digital hardware economies (due to recent advances in VLSI technology), has focussed much attention on more complex and sophisticated speech coding algorithms which offer improved quality at relatively low bit rates. This thesis describes the results (obtained from computer simulations) of research into various efficient (time and frequency domain) speech encoders operating at a transmission bit rate of 16 Kbps. In the time domain, Adaptive Differential Pulse Code Modulation (ADPCM) systems employing both forward and backward adaptive prediction were examined. A number of algorithms were proposed and evaluated, including several variants of the Stochastic Approximation Predictor (SAP). A Backward Block Adaptive (BBA) predictor was also developed and found to outperform the conventional stochastic methods, even though its complexity in terms of signal processing requirements is lower. A simplified Adaptive Predictive Coder (APC) employing a single tap pitch predictor considered next provided a slight improvement in performance over ADPCM, but with rather greater complexity. The ultimate test of any speech coding system is the perceptual performance of the received speech. Recent research has indicated that this may be enhanced by suitable control of the noise spectrum according to the theory of auditory masking. Various noise shaping ADPCM configurations were examined, and it was demonstrated that a proposed pre-/post-filtering arrangement which exploits advantageously the predictor-quantizer interaction, leads to the best subjective performance in both forward and backward prediction systems. Adaptive quantization is instrumental to the performance of ADPCM systems. Both the forward adaptive quantizer (AQF) and the backward oneword memory adaptation (AQJ) were examined. In addition, a novel method of decreasing quantization noise in ADPCM-AQJ coders, which involves the application of correction to the decoded speech samples, provided reduced output noise across the spectrum, with considerable high frequency noise suppression. More powerful (and inevitably more complex) frequency domain speech coders such as the Adaptive Transform Coder (ATC) and the Sub-band Coder (SBC) offer good quality speech at 16 Kbps. To reduce complexity and coding delay, whilst retaining the advantage of sub-band coding, a novel transform based split-band coder (TSBC) was developed and found to compare closely in performance with the SBC. To prevent the heavy side information requirement associated with a large number of bands in split-band coding schemes from impairing coding accuracy, without forgoing the efficiency provided by adaptive bit allocation, a method employing AQJs to code the sub-band signals together with vector quantization of the bit allocation patterns was also proposed. Finally, 'pipeline' methods of bit allocation and step size estimation (using the Fast Fourier Transform (FFT) on the input signal) were examined. Such methods, although less accurate, are nevertheless useful in limiting coding delay associated with SRC schemes employing Quadrature Mirror Filters (QMF)

    A Hybrid voice/text electronic mail system: an application of the integrated services digital network

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    The objective of this thesis is to present a useful application for the Integrated Services Digital Network (ISDN) that is expected to one day replace the analog phone system in use today. ISDN itself and its continuing evolution are detailed. The system developed as a part of this thesis involved the creation of an inexpensive phone terminal that can serve as an ISDN terminal and also as a bridge to a Local Area Network (LAN). The phone terminal provides a hybrid electronic mail system that allows the attachment of speech to text within a message. Messages created with this phone terminal could theoretically be sent locally using the LAN interface and globally using ISDN to other users with either phone terminals or multimedia personal computers. For this project, the two phone terminals created were interconnected via an Ethernet and using an 80486 PC to act as a Central Office System. This Central Office System provides speech/message storage for the phone terminals. It makes use of speech compression techniques to minimize the storage requirements. The speech compression techniques used as well as the field of speech coding in general are discussed

    Table of Contents

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    Contains the table of contents

    Proceedings of the Second International Mobile Satellite Conference (IMSC 1990)

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    Presented here are the proceedings of the Second International Mobile Satellite Conference (IMSC), held June 17-20, 1990 in Ottawa, Canada. Topics covered include future mobile satellite communications concepts, aeronautical applications, modulation and coding, propagation and experimental systems, mobile terminal equipment, network architecture and control, regulatory and policy considerations, vehicle antennas, and speech compression

    Digital Signal Processing

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    Contains table of contents for Part III, table of contents for Section 1, an introduction and reports on seventeen research projects.National Science Foundation FellowshipNational Science Foundation (Grant ECS 84-07285)National Science Foundation (Grant MIP 87-14969)U.S. Navy - Office of Naval Research (Contract N00014-81-K-0742)Scholarship from the Federative Republic of BrazilU.S. Air Force - Electronic Systems Division (Contract F19628-85-K-0028)AT&T Bell Laboratories Doctoral Support ProgramCanada, Bell Northern Research ScholarshipCanada, Fonds pour la Formation de Chercheurs et I'Aide a la Recherche Postgraduate FellowshipSanders Associates, Inc.OKI Semiconductor, Inc.Tel Aviv University, Department of Electronic SystemsU.S. Navy - Office of Naval Research (Contract N00014-85-K-0272)Natural Sciences and Engineering Research Council of Canada, Science and Engineering Scholarshi

    Delta modulation techniques for low bit-rate digital speech encoding

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    Includes bibliography.Two new hybrid companding delta modulators for speech encoding are presented here. These modulators differ from the Hybrid Companding Delta Modulator (HCDM) proposed by Un et al in that the two new encoders employ Song Voice Adaptation as the basis of the instantaneous compandor, rather than Constant Factor adaptation. A detailed analysis of the performance, both objective and subjective, of these hybrid codecs has been carried out. Results show that overall the two codecs developed as part of this project are better than the HCDM codec. In addition the new codecs offer simpler implementation in digital hardware than the HCDM. A Computer Aided Test (CAT) system has been developed to simplify the design and test processes for speech codecs

    Design of a digital voice data compression technique for orbiter voice channels

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    Candidate techniques were investigated for digital voice compression to a transmission rate of 8 kbps. Good voice quality, speaker recognition, and robustness in the presence of error bursts were considered. The technique of delayed-decision adaptive predictive coding is described and compared with conventional adaptive predictive coding. Results include a set of experimental simulations recorded on analog tape. The two FM broadcast segments produced show the delayed-decision technique to be virtually undegraded or minimally degraded at .001 and .01 Viterbi decoder bit error rates. Preliminary estimates of the hardware complexity of this technique indicate potential for implementation in space shuttle orbiters

    Multipath/modulation study for the tracking and data relay satellite system Final report, 14 Apr. 1969 - 12 Jan. 1970

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    Multipath modulation study of tracking and data relay satellite syste

    Channel Fading Statistics For Real-Time Data Transmission In Emergency Call Systems And Unmanned Aerial Systems

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    The Third Generation Partnership Project (3GPP) selected an in-band modem to transmit emergency data over cellular voice channel for the European Union emergency call (eCall) system. However, the road test results presented by the Harmonized eCall European Pilot project showed that the success rate of data delivery was only 71%, indicating that there is significant potential to improve its performance. In this dissertation, a testbed is designed for the eCall system that satisfies the 3GPP TS 26.267/268/269 standards. A method is proposed to measure the power of the received signal that passes through the in-band channel. Experiments are performed with the in-vehicle system testbed in a laboratory or a car travelling in city, suburb, country- side, or freeway. Fading statistics of the received signal after power control are found and discussed, together with cumulative distribution function (CDF), level crossing rate (LCR), and average fade duration (AFD). It is found that with probability less than or equal to 0.1%, fading and attenuation can vary from -19 dB for the continuous wave (CW) signal at 500 Hz to -9.5 dB for the CW signal at 2000 Hz. This dissertation recommends moving the CW signals at 500 Hz and 800 Hz for detection and synchronization in the 3GPP standard to 1500 Hz and 2000 Hz, respectively. This will give 9.5 dB improvement in detection and synchronization. The fading results are used to calculate the bit error rate (BER) performance for the eCall in-band modem. Synchronization detection probability are obtained by transmitting the synchronization preamble through various adaptive multi-rate vocoders and an additive white Gaussian noise channel. The testbed and proposed method are also used to measure the power of signals received by an unmanned aerial systems (UAS) and by the receiver in the operation center, respectively. Field experiments are carried out by flying the UAS above different locations. Statistics, including CDF, LCR, and AFD, are calculated for the six test-sites. The results of the fading statistics, synchronization detection probability, and BER can be directly applied to design real-time communication systems, including detection, delay estimation, modulation and coding
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