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CHAPTER 7

CONCLUSION AND SUGGESTIONS

FOR FUTURE WORK

7.1 CONCLUSION

In this research, dual channel and single channel speech

enhancement algorithms are discussed and the performances of the
proposed algorithms are analyzed in detail based on the objective and
subjective quality measures. From the discussion, it is identified that the
proposed algorithms provide improvement in quality and intelligibility of
the enhanced speech signal compared to the conventional methods.

In the proposed dual channel speech enhancement methods,

Hadamard transform are utilized to separate the cosine transformed input
signal in terms of its frequency components. Further the conventional
adaptive algorithms are used to enhance the noisy speech signal.
These adaptive filtering is to minimize the mean square error between the
desired signal and the processed signal. From the objective and subjective
measures of the proposed methods it is identified that signals operated in
frequency domain gives better results than signals operated in time domain.
The results show that proposed DCT-Hadamard-RLS algorithm provides
better performance compared to other methods.

It is unproblematic to enhance signal using dual channel

algorithms because of the availability of the reference signal. Most of the
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speech processing systems will not have reference signal when it is used in
real time environment. In such case noise reduction is a challenging
problem by having noisy speech alone. In Chapter 4 and Chapter 5, single
channel speech enhancement algorithms are proposed and examined to
enhance noisy speech signal without the reference signal.

In Chapter 4, a basic speech enhancement technique using

partial differential equation is proposed for stationary noisy environment.
The next proposed algorithm is using variance and modified gain function.
The above two algorithms process the speech signal in single band.
In Chapter 5, single channel speech enhancement algorithms are proposed
and discussed which utilizes subband approach. The subband spectral
subtraction method of speech enhancement is performed in which noise
estimation algorithm used is of adaptive in nature. This frequency
dependent subband spectral subtraction method provides a definite
improvement over the conventional power spectral subtraction method and
does not suffer from musical noise.

The improvement can be attributed to the fact that the subband

approach takes into account the non uniform effect of non-stationary noise
on the spectrum of speech. The added computational complexity of the
algorithm is minimal and it adapts with non-stationary noise environments.
Further the quality and intelligibility of the enhanced signal is improvised
using SBTSDD approach with adaptive weighting factor and perceptual
gain factor. The performances of the proposed algorithms are analyzed in
the corresponding chapters and concluded that SBTSDD approach with
adaptive weighting factor and perceptual gain factor performs well in all
the noisy environment and gives better quality and intelligibility.
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Further the application of the proposed methods in digital

hearing aid is discussed in Chapter 6. From the results, it is identified that
the proposed methods can be used as a preprocessing technique for any of
the speech processing applications to improve its performance in all types
of noisy environments.

7.2 SUGGESTIONS FOR FUTURE WORK

The dual channel algorithms proposed in this research utilizes

the reference signal for the enhancement of noisy speech signal; in this
case the reference signal is of desired signal. In future an algorithm can be
developed considering that no desired signal is available. In such case two
different noisy speech signals from different directions and their coherence
function can be used for enhancement.

In future the improvement in the performance of the subband

approach method can be obtained with adaptive subband, where the
separation of frequency bands is based on the noise spectrum and the
performance of noise in different frequency levels. This may give a better
signal to noise ratio because of the adaptive separation of bands and better
intelligibility for the speech processing system, for that a detailed analysis
of the noise and its characteristics are required. Further dereverberation can
also be considered for enhancement.

This proposed SBTSDD approach algorithm can be

implemented in real time on a fixed point Digital signal processor for
evaluation in real world conditions.