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The principles of packet speech systems are discussed, and how the structure allows the use of redundancy to design viable solu- tions to the problem. The paper proposes the use of synthetic speech coding algorithms vocoders to pro- vide redundancy, since the algorithms produce a very low bit-rate stream, which only adds a small over- head to a packet.
Preliminary experiments show that normal speech repaired with synthetic quality speech is intelligible, even at very high loss rates. If multimedia conferencing is to become widely used in the Internet community, user must perceive the quality to be sufficiently good for most applica- tions.
Experience has shown that audio is almost al- ways the most important component of multimedia conferencing. Whilst we have identified a number of problems which impair the quality of audio, the ma- jor one with audio over the MBONE is packet loss .
Packet loss can occur for a number of reasons: Packet loss is a persistent problem, particularly given the increasing popularity, and therefore in- creasing lead, of the Internet.
Possible ways of com- batting congestion include bandwidth reservation and moves toward an integrated service management on the Internet. These would require wide-scale changed to be agreed and implemented, so these solutions will be available in the short to medium term.
Yet, the disruption of speech intelligibility even at low loss rates which we currently experience may convince a whole generation of users that multimedia conferenc- ing over the Internet is not viable. Martina angela sasse thesis therefore pro- pose a solution which renders the speech intelligible under current network conditions, and can be de- ployed in the short term.
Such a solution will have to be at the application level, i.
Current audio applications repair lost packets with silence, which leads to the speech clipping effects currently experienced by many users. Since compara- tively large packets are used, even the loss of individ- ual packet loss has a serious impact on the intelligibility of speech.
We propose a method of repairing damaged speech using cheap redundancy within the packets sent from the transmitter. The redundancy is synthetic speech, which, when split into packets, only adds a very small amount of overhead, and therefore does not add to the congestion at the network level.
The re- dundancy for any given packet of speech is piggy- backed onto a later packet. This mechanism means that when the receiver suffers the loss of the primary speech information, it still has the possibility of sub- stituting something sensible in the output stream of speech, provided that the redundancy can be received.
In order to establish the effectiveness of this solu- tion, we have performed experiments into user per- ception of speech repaired with a synthetic substitute. The paper also describes how the pro- posed solution scales in the multicast environment.
Background Speech coding schemes have been standardised for use over telephone networks; a variety of speech coding algorithms exist for a single target quality of service QoSand at a very few discrete bit-rates: An- other standard speech coding algorithm is Groupe Speciale Mobile GSMwhich was designed for use over cellular telephone networks.
The target QoS is consequently slightly less than toll, but the algorithm is popular, since it operates at the same bit-rate as CELP, but is much less complex. A fuller discussion of toll quality speech coding algorithms can be found in .
These algorithms operate at very low bit-rates ap- prox. A fuller description of which can be found in . Packet Speech Systems Packet speech systems usually employ the stand- ard speech coding algorithms, and group the emerg- ing stream of codewords into packets for transmission over the network.
At the receiver, the packets may be delivered: Consequently, a reconstruction delay must be used at the receiver to repair the network effects; this enables sample play-out to be smoothed.
In a packet speech system, the end-to-end delay is always a critical factor in the usability of a real-time voice system, and should be kept below ms in the absence of echoes The figure may be in fact be less than this - ms , if conversation patterns are not to break down.
The size of the packets in ms chosen for a packet speech system directly impacts the end- to-end delay. A delay equal to the size of one packet is incurred at the transmitter, since the samples in the packet have to be collected before a packet can be sent.
At the receiver, a rough estimate of the recon- struction delay required to smooth out packet arrival times is two packets worth in ms  , although the true value may be substantially in excess of this rule of thumb.
Consequently, a minimum of three packets worth of delay is incurred on an end-to-end basis, be- fore the network propagation delay has been taken into account.
Furthermore, the network itself may lose packets. The presence of the dummy packet is usually discernible to the listen- er, and unfortunately, the perceptibility of the loss in- creases with increasing packet size, as well as with increasing loss rate.Results for 'Martina Angela Sasse' (try it on Scholar) + found.
Order: Order. Options 1 filter applied. A Framework for the Lived Experience of Identity. Adrian Rahaman & Martina Angela Sasse - - Identity in focusing on their criticisms of the awareness of awareness thesis.
Sasse, Martina Angela Awarding Body: University of Birmingham Current Institution: University of Birmingham The thesis commences with an examination of the history and current state of the discipline of human-computer interaction to establish the context and determine the appropriate methods for conducting research on users' models.
. The Compliance Budget should be understood and managed in the same way as any financial budget, as compliance directly affects, and can place a cap on, effectiveness of organisational security measures.
Persuasive Password Security Dirk Weirich The thesis then presents a discourse-analytic investigation of the interpretative Professor Martina Angela Sasse, for her support and advice throughout the time this research and the subsequent write-up have taken.
Reliable Audio for Use over the Internet Abstract This paper describes current problems found with audio applications over the MBONE (Multicast Backbone), and investigates possible solutions to the most common one - packet loss.
BIBLIOGRAPHY [AS99] Anne Adams and Martina Angela Sasse. Users are not the enemy. Communica-tions of the ACM, –46, [Ass05] Association for India’s Development Austin.