27,149 research outputs found
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Mobile Audiovisual Terminal: System Design and Subjective Testing in DECT and UMTS networks
It is anticipated that there will shortly be a requirement
for multimedia terminals that operate via mobile
communications systems. This paper presents a functional specification
for such a terminal operating at 32 kb/s in a digital
European cordless telecommunications (DECT) and universal
mobile telecommunications system (UMTS) radio network. A terminal
has been built, based on a PC with digital signal processor
(DSP) boards for audio and video coding and decoding. Speech
coding is by a phonetically driven code-excited linear prediction
(CELP) speech coder and video coding by a block-oriented hybrid
discrete cosine transform (DCT) coder. Separate channel coding
is provided for the audio and video data. The paper describes the
techniques used for audio and video coding, channel coding, and
synchronization. Methods of subjective testing in a DECT network
and in a UMTS network are also described. These consisted of
subjective tests of first impressions of the mobile audioâvisual
terminal (MAVT) quality, interactive tests, and the completion
of an exit questionnaire. The test results showed that the quality
of the audio was sufficiently good for comprehension and the
video was sufficiently good for following and repeating simple
mechanical tasks. However, the quality of the MAVT was not
good enough for general use where high-quality audio and video
was needed, especially when transmission was in a noisy radio
environment
A study and experiment plan for digital mobile communication via satellite
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
Near-Instantaneously Adaptive HSDPA-Style OFDM Versus MC-CDMA Transceivers for WIFI, WIMAX, and Next-Generation Cellular Systems
Burts-by-burst (BbB) adaptive high-speed downlink packet access (HSDPA) style multicarrier systems are reviewed, identifying their most critical design aspects. These systems exhibit numerous attractive features, rendering them eminently eligible for employment in next-generation wireless systems. It is argued that BbB-adaptive or symbol-by-symbol adaptive orthogonal frequency division multiplex (OFDM) modems counteract the near instantaneous channel quality variations and hence attain an increased throughput or robustness in comparison to their fixed-mode counterparts. Although they act quite differently, various diversity techniques, such as Rake receivers and space-time block coding (STBC) are also capable of mitigating the channel quality variations in their effort to reduce the bit error ratio (BER), provided that the individual antenna elements experience independent fading. By contrast, in the presence of correlated fading imposed by shadowing or time-variant multiuser interference, the benefits of space-time coding erode and it is unrealistic to expect that a fixed-mode space-time coded system remains capable of maintaining a near-constant BER
Proceedings of the Second International Mobile Satellite Conference (IMSC 1990)
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
Deconstructing Speech: new tools for speech manipulation
My research at the London College of Communication is concerned with archives of recorded speech, what new tools
need to be devised for its manipulation and how to go about
this process of invention. Research into available forms of
analysis of speech is discussed below with regard to two
specific areas, feature vectors from linear predictive coding (LPC) analysis and hidden Markov-model-based automatic speech recognition (ASR) systems. These are discussed in order to demonstrate that whilst aspects of each may be useful in devising a system of speech-archive manipulation for artistic use. Their drawbacks and deficiencies for use in art â consequent of the reasons for their invention â necessitate the creation of tools with artistic, rather than engineering agendas in mind. It is through the initial process of devising conceptual tools for understanding speech as sound objects that I have been confronted with issues of semiotics and semantics of the voice and of the relationship between sound and meaning in speech, and of the role of analysis in mediating existing methods of communication. This is discussed with reference to Jean-Jacques Nattiezâs Music and Discourse: Towards a Semiology of Music (Nattiez 1987). The âtraceâ â a neutral level of semiotic analysis proposed by Nattiez, far from being hypothetical as suggested by Hatten (1992: 88â98) and others, is present by analogy to many forms of mediation in modern spoken communication and the reproduction of music, and it is precisely this neutrality with regards to meaning that tools for manipulation of speech must possess, since the relationships between the sound of speech and its meaning are âintenseâ (after Deleuze 1968
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Error resilient video transcoding for robust inter-network communications using GPRS
A novel fully comprehensive mobile video communications
system is proposed in this paper. This system exploits
the useful rate management features of the video transcoders and
combines them with error resilience for transmissions of coded
video streams over general packet radio service (GPRS) mobileaccess
networks. The error-resilient video transcoding operation
takes place at a centralized point, referred to as a video proxy,
which provides the necessary output transmission rates with the
required amount of robustness. With the use of this proposed
algorithm, error resilience can be added to an already compressed
video stream at an intermediate stage at the edge of two or more
different networks through two resilience schemes, namely the
adaptive intra refresh (AIR) and feedback control signaling (FCS)
methods. Both resilience tools impose an output rate increase
which can also be prevented with the proposed novel technique in
this paper. Thus, an error-resilient video transcoding scheme is
presented to give robust video outputs at near target transmission
rates that only require the same number of GPRS timeslots as
the nonresilient schemes. Moreover, an ultimate robustness is
also accomplished with the combination of the two resilience
algorithms at the video proxy. Extensive computer simulations
demonstrate the effectiveness of the proposed system
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