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Abstract:
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This thesis investigated the potential use of Linear Predictive
Coding in speech communication applications. A Modified Block Adaptive
Predictive Coder is developed, which reduces the computational burden and
complexity without sacrificing the speech quality, as compared to the
conventional adaptive predictive coding (APC) system. For this, changes in
the evaluation methods have been evolved. This method is as different from
the usual APC system in that the difference between the true and the
predicted value is not transmitted. This allows the replacement of the high
order predictor in the transmitter section of a predictive coding system, by
a simple delay unit, which makes the transmitter quite simple. Also, the
block length used in the processing of the speech signal is adjusted
relative to the pitch period of the signal being processed rather than
choosing a constant length as hitherto done by other researchers. The
efficiency of the newly proposed coder has been supported with results of
computer simulation using real speech data.
Three methods for voiced/unvoiced/silent/transition
classification have been presented. The first one is based on energy,
zerocrossing rate and the periodicity of the waveform. The second method
uses normalised correlation coefficient as the main parameter, while the
third method utilizes a pitch-dependent correlation factor. The third
algorithm which gives the minimum error probability has been chosen in a
later chapter to design the modified coder The thesis also presents a comparazive study beh-cm the
autocorrelation and the covariance methods used in the evaluaiicn of the
predictor parameters. It has been proved that the azztocorrelation method is
superior to the covariance method with respect to the filter stabf-it)‘ and
also in an SNR sense, though the increase in gain is only small. The
Modified Block Adaptive Coder applies a switching from pitch precitzion to
spectrum prediction when the speech segment changes from a voiced or
transition region to an unvoiced region. The experiments cont;-:ted in
coding, transmission and simulation, used speech samples from .\£=_‘ajr2_1a:r1
and English phrases. Proposal for a speaker reecgnifion syste: and a
phoneme identification system has also been outlized towards the end of
the thesis. |