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Signal & Image Processing : An International Journal (SIPIJ) Vol.5, No.5, October 2014 
A NOVEL UNCERTAINTY PARAMETER SR 
(SIGNAL TO RESIDUAL SPECTRUM RATIO) 
EVALUATION APPROACH FOR SPEECH 
ENHANCEMENT 
M. Ravichandra Kumar1 and B. Ravi Teja2 
1Department of Electronics and Communication, 
M-tech, Gudlavalleru Engineering College, A.P, India 
2Department of Electronics and Communication Engineering, Assistant professor, 
Gudlavalleru Engineering College, A.P, India 
ABSTRACT 
Usually, hearing impaired people use hearing aids which are implemented with speech enhancement 
algorithms. Estimation of speech and estimation of nose are the components in single channel speech 
enhancement system. The main objective of any speech enhancement algorithm is estimation of noise power 
spectrum for non stationary environment. VAD (Voice Activity Detector) is used to identify speech pauses 
and during these pauses only estimation of noise. MMSE (Minimum Mean Square Error) speech 
enhancement algorithm did not enhance the intelligibility, quality and listener fatigues are the perceptual 
aspects of speech. Novel evaluation approach SR (Signal to Residual spectrum ratio) based on uncertainty 
parameter introduced for the benefits of hearing impaired people in non stationary environments to control 
distortions. By estimation and updating of noise based on division of original pure signal into three parts 
such as pure speech, quasi speech and non speech frames based on multiple threshold conditions. Different 
values of SR and LLR demonstrate the amount of attenuation and amplification distortions. The proposed 
method will compared with any one method WAT(Weighted Average Technique) Hence by using 
parameters SR (signal to residual spectrum ratio) and LLR (log like hood ratio), MMSE (Minim Mean 
Square Error) in terms of segmented SNR and LLR. 
KEYWORDS 
Noise Estimation, Voice Activity Detector (VAD), Speech Enhancement, SR (Signal to Residual spectrum 
ratio) parameter, Speech Intelligibility Improvement. 
1. INTRODUCTION 
The major problem arises in speech enhancement background noise and it is affected by speech 
signal. There are many applications which are speech recognition, hearing aid, VOIP (Voice over 
Internet Protocol), teleconferencing systems and mobile phones of reduces background noise. The 
noise present in the both analogy and digital systems. An unwanted signal as noise and it 
degrades the speech intelligibility and speech quality. Vehicle noise and background noise are the 
different types of noises. In speech enhancement mainly considered as noise estimation it requires 
to estimate of noise from noisy speech signal. Speech enhancement main objective is to give 
better performance of speech quality and speech intelligibility by using various algorithms and 
based on these algorithms to minimise the MSE (Man Square Error) [5]. The effect of various 
distortions (attenuation and amplification distortions) present in the speech signal so these 
distortions are proper control to improve the speech intelligibility. The negative difference 
DOI : 10.5121/sipij.2014.5501 1
Signal & Image Processing : An International Journal (SIPIJ) Vol.5, No.5, October 2014 
between clean and enhanced spectrum would be amplification distortion, while a positive 
difference would be attenuation distortion. Speech enhancement for noise reduction can be 
categorised into three fundamental classes and those are model based, spectral restoration and 
filtering technique methods. All the methods are common feature is clean speech power spectrum 
estimation from noisy environment spectrum. 
The presence or absence of human speech detected is called VAD (Voice Activity Detector). In 
speech processing technique used VAD and also called as speech detection or speech active 
detector as well as VAD used in noise reduction also. Multimedia application VAD allows 
simultaneously voice and data. Here consider another application cellular based system (GSM, 
CDMA) in discontinuous transmission mode. Speech intelligibility and speech quality both are 
correlated highly by measuring frequency domain of segmental SNR so for this measure is refer 
to residual spectrum ratio [14]. 
2 
2. RELATED WORK 
In obtainable algorithms are not suitable for estimate of background noise but VAD (voice 
activity detector) good background noise estimation algorithm for stationary environment [13]. 
Speech presence or speech absence of human speech is detected by VAD (voice activity detector) 
by using this algorithm to estimate noise in speech pauses only. Every algorithm makes to give 
speech quality but not speech intelligibility and this drawback occurred in present existing 
algorithms [3]. Wiener and MMSE (minimum mean square error) algorithms are used to 
minimize the error in between of enhanced and clean spectrum so these algorithms are based in 
spectral principals. 
Most of the algorithms were proposed speech recognized application to estimate the noise in non 
stationary environments VAD did not estimate the noise accurately. The lack of intelligibility in 
present algorithms is not proper to estimation of noise. These problems can be reduces by using 
the propose algorithm SR (signal to residual spectrum ratio) for improve speech quality and 
speech intelligibility in noisy environment. 
3. PROPOSED WORK 
Consider, here P(n) and Q(n) are the clean speech, noise and then noisy speech denoted as 
follows, 
X(n) = P(n) + Q(n) (1) 
Time domain of noisy speech is segmented by frames by using of windowed technique let it be 
consider hamming window and represented equation as follows 
2  (n-1) 
Nw-1 
W[n,] = 0.54 – 0.4 cos for 0nNw-1 (2) 
The short time Fourier transforms is give equation for wave form of windowed speech signal 
,  =  	
,
−
 (3) 
Where  represents centre time at window
Signal  Image Processing : An International Journal (SIPIJ) Vol.5, No.5, October 2014 
3.1 Determination of Threshold Condition 
Speech intelligibility and speech quality both are correlated highly so to measure using segmental 
SNR (Signal to Noise Ratio) in consider frequency domain version and this measure to mention 
as signal to residual spectrum. 
SNRESI(k) = (4) 
S2(k) 
(S(k)-S(k))2 
S(k) is speech enhancement algorithm of estimated spectrum and S(k) is clean speech magnitude 
spectrum. To improve the speech intelligibility by proper control of distortions using regions are 
constraint and it has follows 
3 
a) S(k)  S(k), suggested only attenuation distortion 
b) S(k)  2. S(k), suggested greater or 6.02 db of amplification distortion 
c) S(k)  S(k)  2. S(k) , suggested up to 6.02 db of amplification distortion 
Reason (a) and Reason (b) from that we constraint to this reason S(k)  S(k) and it is used in 
speech enhancement algorithms. 
S(k)  2. S(k) (5) 
This after squaring on both sides becomes 
S2(k)  4. S2(k) (6) 
So assume S(k)= X(k) it is not enhance noisy speech by algorithms and then 
S2(k) = X2(k) = S2(k) + Q2(k) and reduces to S2(k)  1/3 Q2(k) 
SNR(k) 1/3 (7) 
3.2. SR (Signal to Residual spectrum ratio) 
Figure.1 represents the SR algorithm and the noisy signal is segmented using windowed 
technique eq.(2) later FFT is performed on the segmented frames with the help of with the help 
of eq.(3). Noisy speech has different frames so we can calculate SNR (Signal To Noise Ratio) 
based on threshold determination. 
3.3. Noise power estimation method 
Here focused on noise estimation and it has different approaches so the fundamental component 
of speech enhancement is noise power estimation. It required estimating of noise from noisy 
speech spectrum by using different algorithms based on classification of speech into quasi speech, 
original speech and noise speech [11]. 
3.3.1. Non-Speech 
It has to be occurred in speech absence or speech pauses only and to estimate noise power of 
these frames for the following proposed condition 
if S(k)2.S(k) then 
A(m,k) =  Â(m-1,k)+(1-) |Â(m ,k)|2 (8) 
Where  is called as smoothing factor and typically set to =0.98 and lies in 01. 
3.3.2. Quasi-Speech 
To estimate noise power for quasi speech is both noise and speech on each frame and the 
proposed condition
Signal  Image Processing : An International Journal (SIPIJ) Vol.5, No.5, October 2014 
4 
It S(k)  S(k)  2. S(k) then 
Â(m, k) = B(m, k)Â(m-1, k)+(1-B(m, k)) (9) 
Where Â(m, k) is non speech frame of noise spectrum estimation 
Figure 1: SR Algorithm. 
3.4. Tracking the Minimum of Noisy Speech 
To tracking of noisy speech by regularly averaging precedent spectral values, here used rule non 
linear in different approach [10] 
if Bmin (m − 1, k)  B (m, k) then 
1- 
1- 
Bmin (m, k )=  Bmin (m − 1, k) + ( (B (m, k) –  B( m − 1, k )) (10) 
If Bmin( m − 1, k )  B (m, k) then 
Bmin (m, k) = B(m, k) (11) 
To determine the values of ,  and 	 by experiment, in practical implementation smoothing 
parameter in (11) whose maximum value is 0.98 to avoid deadlock for r(m,k)=1. 
3.5. Speech Presence Probability 
To measure how much speech present probability in noisy speech by following equation 
|A(m,k)|2 
Bmin(m,k) 
Bsp(m,k) = (12)
Signal  Image Processing : An International Journal (SIPIJ) 
Vol.5, No.5, October 2014 
where Bmin(m,k) and |A(m,k)|2 are represented as local minimum and power spectrum of noisy 
speech. Speech present and speech absent are dependent on the ratio of speech present probability 
if it is grater to threshold then consider as speech present otherwise it 
gives speech absent. 
3.6. Computing Logistic Function 
Logistic function is one of special case in the form of mathematical and it is also called as 
sigmoid function or sigmoid curve as given function 
g(x)=1/(1-e-x) 
Figure 2: sigmoid curve 
3.7. Calculating Frequency Dependent Smoothing Constant 
To compute smoothing factor need the time 
B (m, k) = s+ (1- s) Bsp (m, k) 
Where, s is denotes as constant. 
time-frequency domain as to follow this equation 
To updating of minimum noisy spectrum is B 
B( m, k) = 	B(m − 1, k) + (1 − 	) 
Bmin (m, k) and given equation 
Where, B (l, k) is average noise spectrum and 
,  is known as smoothing factor 
b(, k) = b b(
-1,k)+(1- b) I (, k) 
Where b (, k) is a smoothing constant, the above recursive absolutely utilize the correlation for 
speech presence in adjacent frames. 
For r (m, k) = 1. 
r(m, k) = N (m − 1, k)/N 
2(m, k) 
Posterior SNR of smoothed version is represented by eq. (16) 
The Wiener filter solves the signal estimation problem for stationary signals. A major 
contribution was the use of a statistical model for the estimated signal the filter is best in the 
intellect of the MMSE [16]. . We shall focus here on the discrete-discrete 
time version of the Wiener filter 
and it is used to generate estimated pure signal from a given noise speech signal. 
5 
(13) 
[9]. 
(14) 
(15) 
(16) 
(17) 
he
Signal  Image Processing : An International Journal (SIPIJ) Vol.5, No.5, October 2014 
6 
4. IMPLEMENTATION AND RESULTS 
Speech enhancement algorithms are tested on MATLAB for Non-stationary environment of 
speech database [2]. The unvoiced speech regions to detected correctly by observed the results of 
proposed algorithm and speech activity region also accurately measured even noise is present. 
The table gives classification of results and performance of the algorithms in which segmental 
SNR and LLR (Log Like hood Ratio) are compared to proposed algorithm SR (Signal to Residual 
spectrum ratio). Spectrogram is way to visualize the speech signal in the domain time-frequency 
representation. In speech signal through several intermediate levels which are linguistic message 
and paralinguistic information including emotion is effectively visualized based on the 
spectrogram. Now we can concludes the variations in the noisy speech signal of Spectrogram 
represented in different areas those are trains, cars, and airport. 
Table1: comparison of weighted average technique and proposed SR technique using LLR 
and segmental SNR methods 
(i) 
Type of Noise 
LLR Segmental SNR 
SNR in 
db 
Weighted 
Average 
Technique 
Proposed 
(SR) 
Technique 
Weighted 
Average 
Technique 
Proposed 
(SR) 
Technique 
CAR 
0 1.687827 1.500914 -6.806391 -6.716270 
5 1.842711 1.596159 -5.668619 5.485975 
10 1.976017 1.602708 -4.866237 -3.861581 
15 1.831509 1.580956 -4.335797 -3.537122 
AIRPORT 
0 1.237398 1.057377 -3.802483 -3.440414 
5 1.124488 0.934859 -2.781458 -2.526855 
10 0.919158 0.736983 -0.731036 -0.083965 
15 0.910468 0.549913 1.310788 3.080826 
TRAIN 
0 2.091845 1.798190 -6.486321 -6.296185 
5 2.322675 1.845213 -5.559169 -4.970945 
10 2.036162 1.759774 -5.251629 -4.206358 
15 2.230337 1.827800 -3.211548 4.284449
Signal  Image Processing : An International Journal (SIPIJ) Vol.5, No.5, October 2014 
7 
(ii) 
(iii) 
(iv) 
Figure 3: timing wave form and spectrogram of (i) pure speech signal (ii) noisy speech signal and enhanced 
signal with (iii) weighted average technique (iv) proposed SR technique in car noise with different SNR 
levels. 
(i)
Signal  Image Processing : An International Journal (SIPIJ) Vol.5, No.5, October 2014 
8 
(ii) 
(iii) 
(iv) 
Figure 4: timing wave form and spectrogram of (i) pure speech signal (ii) noisy speech signal and enhanced 
signal with (iii) weighted average technique (iv) proposed SR technique in airport noise with different SNR 
levels. 
(i)
Signal  Image Processing : An International Journal (SIPIJ) Vol.5, No.5, October 2014 
9 
(ii) 
(iii) 
(iv) 
Figure 5: Timing wave form and spectrogram of (i) pure speech signal (ii) noisy speech signal and 
enhanced signal with (iii) weighted average technique (iv) proposed SR technique in Train noise with 
different SNR levels. 
5. CONCLUSION 
This paper focused on the issue of noise estimation for enhancement of noisy speech. The noise 
estimate was updated continuously in every frame using time–frequency smoothing factors 
calculated based on speech-presence probability in each frequency bin of the noisy speech 
spectrum [1].The main achievements of speech enhancement algorithms are speech intelligibility 
and speech quality. Here to reduce the amplification distortion and attenuation distortion by using 
proposed method SR (Signal to Residual spectrum ratio) [5]. The proper control of these 
distortions to improve speech intelligibility it is main drawback of speech enhancement 
algorithms. The proposed method SR it gives better performance when compared to the previous 
existing methods are LLR (log like hood ratio) and segmental SNR.
Signal  Image Processing : An International Journal (SIPIJ) Vol.5, No.5, October 2014 
REFERENCES 
[1] Anuradha R. Fukane1, Shashikant L. Sahare, “Noise estimation Algorithms for Speech 
Enhancement in highly non-stationary Environments”, IJCSI International Journal of Computer 
Science Issues, Vol. 8, Issue 2, March 2011. 
[2] D. Malah, R V Cox, and A J Accardi, “Tracking speech-presence uncertainty to improve speech 
enhancement in non-stationary noise environments, Proc.IEEE Int. Conf. Acoustics, Speech, Signal 
Processing, pp. 1153-1516, 2010. 
[3] Speech Enhancement for Non-stationary Noise Environments 978-1-4244-4994-1/09/$25.00 ©2009 
10 
IEEE. 
[4] K.Nakayama, H.Suzuki and A.Hirano, “Improved methods for noise spectral estimation and adaptive 
spectral gain control in noise spectral suppressor”., Proc. ISPACS‘ 07, Xiamen, China, pp.97-100, 
Dec. 2007. 
[5] S. Rangachari and P. C. Loizou, “A noise-estimation algorithm for highly non-stationary 
environments,” Speech Communication 48, 2006, pp. 220-231. 
[6] T.Lotter and P.Vary, “Noise reduction by joint maximum a posteriori spectral amplitude and phase 
estimation with super-gaussian speech modeling”., Proc. EUSIPCO-04, pp.1447-60, Sep. 2004. 
[7] Cohen.I.,“Noise spectrum estimation in adverse environments improved minima controlled recursive 
averaging”, IEEE Trans. Speech Audio Process., 11(5), pp. 466-475, 2003. 
[8] Cohen.I., “Noise spectrum estimation in adverse environments: improved minima controlled recursive 
averaging”, IEEE Trans. Speech Audio Process., 11(5), pp. 466-475, 2003. 
[9] Cohen.I, “Noise estimation by minima controlled recursive averaging for robust speech 
enhancement”, IEEE Signal Process. Lett., 9(1), pp. 12-15, 2002. 
[10] R. Martin, “Noise power spectral density estimation based on optimal smoothing and minimum 
statistics,” IEEE Trans. Speech Audio Process. vol. 9, no. 5, July 2001, pp. 504-512. 
[11] Martin.R, “Noise power spectral density estimation based on optimal smoothing and minimum 
statistics”, IEEE Tran. Speech Audio Process., 9(5), pp. 504-512,2001. 
[12] I. Cohen, “On speech enhancement under signal presence uncertainty,” in Proc. 26th IEEE Int. Conf. 
Acoust. Speech Signal Process.(ICASSP’2001), Salt Lake City, UT, May 7–11, 2001, pp. 167–170. 
[13] Sohn. J, Kim. N, “Statistical model-based voice activity detection”, IEEE Signal Process. Lett. 6(1), 
pp. 1-3, 1999. 
[14] Malah.D, Cox.R, Accardi.A, “Tracking speech-presence uncertainty to improve speech enhancement 
in non-stationary environments”, Proc. IEEE Internat. On Conf. Acoust. Speech Signal Process., pp. 
789-792, 1999. 
[15] Doblinger.G, “Computationally efficient speech enhancement by spectral minima tracking in 
subbands”, Proc. Euro speech, pp.1513-1516, 1995. 
[16] N. Fan, “Low distortion speech denoising using an adaptive parametric Wiener filter”, In 
ICASSP2004, Montreal, Canada, pp.309-312, 2004. 
[17] J.H.L. Hansen, and B.L. Pellom, “An effective quality evaluation protocol for speech enhancement 
algorithm,” In Inter. Conf. on Spoken Language Processing, vol.7, pp. 2819-2822, Sydney, Australia, 
December 1998. 
[18] Ephraim, Y., Malah, D.,” Speech enhancement using a minimum mean-square error short-time 
spectral amplitude”, estimator. IEEE Trans. Acoust. Speech Signal Process. ASSP 32 (6), 1109–1121 
1984. 
[19] Ephraim, Y., Van Trees, H.L., 1993. “A signal subspace approach for speech enhancement”. Proc. 
IEEE Internat. Conf. on Acoust. Speech, Signal Process. II, 355–358. 
[20] Hirsch, H., Ehrlicher, C., 1995. “Noise estimation techniques for robust speech recognition”. Proc. 
IEEE Internat. Conf. on Acoust. Speech Signal Process., 153–156.

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A Novel Uncertainty Parameter SR ( Signal to Residual Spectrum Ratio ) Evaluation Approach for Speech Enhancement

  • 1. Signal & Image Processing : An International Journal (SIPIJ) Vol.5, No.5, October 2014 A NOVEL UNCERTAINTY PARAMETER SR (SIGNAL TO RESIDUAL SPECTRUM RATIO) EVALUATION APPROACH FOR SPEECH ENHANCEMENT M. Ravichandra Kumar1 and B. Ravi Teja2 1Department of Electronics and Communication, M-tech, Gudlavalleru Engineering College, A.P, India 2Department of Electronics and Communication Engineering, Assistant professor, Gudlavalleru Engineering College, A.P, India ABSTRACT Usually, hearing impaired people use hearing aids which are implemented with speech enhancement algorithms. Estimation of speech and estimation of nose are the components in single channel speech enhancement system. The main objective of any speech enhancement algorithm is estimation of noise power spectrum for non stationary environment. VAD (Voice Activity Detector) is used to identify speech pauses and during these pauses only estimation of noise. MMSE (Minimum Mean Square Error) speech enhancement algorithm did not enhance the intelligibility, quality and listener fatigues are the perceptual aspects of speech. Novel evaluation approach SR (Signal to Residual spectrum ratio) based on uncertainty parameter introduced for the benefits of hearing impaired people in non stationary environments to control distortions. By estimation and updating of noise based on division of original pure signal into three parts such as pure speech, quasi speech and non speech frames based on multiple threshold conditions. Different values of SR and LLR demonstrate the amount of attenuation and amplification distortions. The proposed method will compared with any one method WAT(Weighted Average Technique) Hence by using parameters SR (signal to residual spectrum ratio) and LLR (log like hood ratio), MMSE (Minim Mean Square Error) in terms of segmented SNR and LLR. KEYWORDS Noise Estimation, Voice Activity Detector (VAD), Speech Enhancement, SR (Signal to Residual spectrum ratio) parameter, Speech Intelligibility Improvement. 1. INTRODUCTION The major problem arises in speech enhancement background noise and it is affected by speech signal. There are many applications which are speech recognition, hearing aid, VOIP (Voice over Internet Protocol), teleconferencing systems and mobile phones of reduces background noise. The noise present in the both analogy and digital systems. An unwanted signal as noise and it degrades the speech intelligibility and speech quality. Vehicle noise and background noise are the different types of noises. In speech enhancement mainly considered as noise estimation it requires to estimate of noise from noisy speech signal. Speech enhancement main objective is to give better performance of speech quality and speech intelligibility by using various algorithms and based on these algorithms to minimise the MSE (Man Square Error) [5]. The effect of various distortions (attenuation and amplification distortions) present in the speech signal so these distortions are proper control to improve the speech intelligibility. The negative difference DOI : 10.5121/sipij.2014.5501 1
  • 2. Signal & Image Processing : An International Journal (SIPIJ) Vol.5, No.5, October 2014 between clean and enhanced spectrum would be amplification distortion, while a positive difference would be attenuation distortion. Speech enhancement for noise reduction can be categorised into three fundamental classes and those are model based, spectral restoration and filtering technique methods. All the methods are common feature is clean speech power spectrum estimation from noisy environment spectrum. The presence or absence of human speech detected is called VAD (Voice Activity Detector). In speech processing technique used VAD and also called as speech detection or speech active detector as well as VAD used in noise reduction also. Multimedia application VAD allows simultaneously voice and data. Here consider another application cellular based system (GSM, CDMA) in discontinuous transmission mode. Speech intelligibility and speech quality both are correlated highly by measuring frequency domain of segmental SNR so for this measure is refer to residual spectrum ratio [14]. 2 2. RELATED WORK In obtainable algorithms are not suitable for estimate of background noise but VAD (voice activity detector) good background noise estimation algorithm for stationary environment [13]. Speech presence or speech absence of human speech is detected by VAD (voice activity detector) by using this algorithm to estimate noise in speech pauses only. Every algorithm makes to give speech quality but not speech intelligibility and this drawback occurred in present existing algorithms [3]. Wiener and MMSE (minimum mean square error) algorithms are used to minimize the error in between of enhanced and clean spectrum so these algorithms are based in spectral principals. Most of the algorithms were proposed speech recognized application to estimate the noise in non stationary environments VAD did not estimate the noise accurately. The lack of intelligibility in present algorithms is not proper to estimation of noise. These problems can be reduces by using the propose algorithm SR (signal to residual spectrum ratio) for improve speech quality and speech intelligibility in noisy environment. 3. PROPOSED WORK Consider, here P(n) and Q(n) are the clean speech, noise and then noisy speech denoted as follows, X(n) = P(n) + Q(n) (1) Time domain of noisy speech is segmented by frames by using of windowed technique let it be consider hamming window and represented equation as follows 2 (n-1) Nw-1 W[n,] = 0.54 – 0.4 cos for 0nNw-1 (2) The short time Fourier transforms is give equation for wave form of windowed speech signal , = ,
  • 3. − (3) Where represents centre time at window
  • 4. Signal Image Processing : An International Journal (SIPIJ) Vol.5, No.5, October 2014 3.1 Determination of Threshold Condition Speech intelligibility and speech quality both are correlated highly so to measure using segmental SNR (Signal to Noise Ratio) in consider frequency domain version and this measure to mention as signal to residual spectrum. SNRESI(k) = (4) S2(k) (S(k)-S(k))2 S(k) is speech enhancement algorithm of estimated spectrum and S(k) is clean speech magnitude spectrum. To improve the speech intelligibility by proper control of distortions using regions are constraint and it has follows 3 a) S(k) S(k), suggested only attenuation distortion b) S(k) 2. S(k), suggested greater or 6.02 db of amplification distortion c) S(k) S(k) 2. S(k) , suggested up to 6.02 db of amplification distortion Reason (a) and Reason (b) from that we constraint to this reason S(k) S(k) and it is used in speech enhancement algorithms. S(k) 2. S(k) (5) This after squaring on both sides becomes S2(k) 4. S2(k) (6) So assume S(k)= X(k) it is not enhance noisy speech by algorithms and then S2(k) = X2(k) = S2(k) + Q2(k) and reduces to S2(k) 1/3 Q2(k) SNR(k) 1/3 (7) 3.2. SR (Signal to Residual spectrum ratio) Figure.1 represents the SR algorithm and the noisy signal is segmented using windowed technique eq.(2) later FFT is performed on the segmented frames with the help of with the help of eq.(3). Noisy speech has different frames so we can calculate SNR (Signal To Noise Ratio) based on threshold determination. 3.3. Noise power estimation method Here focused on noise estimation and it has different approaches so the fundamental component of speech enhancement is noise power estimation. It required estimating of noise from noisy speech spectrum by using different algorithms based on classification of speech into quasi speech, original speech and noise speech [11]. 3.3.1. Non-Speech It has to be occurred in speech absence or speech pauses only and to estimate noise power of these frames for the following proposed condition if S(k)2.S(k) then A(m,k) = Â(m-1,k)+(1-) |Â(m ,k)|2 (8) Where is called as smoothing factor and typically set to =0.98 and lies in 01. 3.3.2. Quasi-Speech To estimate noise power for quasi speech is both noise and speech on each frame and the proposed condition
  • 5. Signal Image Processing : An International Journal (SIPIJ) Vol.5, No.5, October 2014 4 It S(k) S(k) 2. S(k) then Â(m, k) = B(m, k)Â(m-1, k)+(1-B(m, k)) (9) Where Â(m, k) is non speech frame of noise spectrum estimation Figure 1: SR Algorithm. 3.4. Tracking the Minimum of Noisy Speech To tracking of noisy speech by regularly averaging precedent spectral values, here used rule non linear in different approach [10] if Bmin (m − 1, k) B (m, k) then 1- 1- Bmin (m, k )= Bmin (m − 1, k) + ( (B (m, k) – B( m − 1, k )) (10) If Bmin( m − 1, k ) B (m, k) then Bmin (m, k) = B(m, k) (11) To determine the values of , and by experiment, in practical implementation smoothing parameter in (11) whose maximum value is 0.98 to avoid deadlock for r(m,k)=1. 3.5. Speech Presence Probability To measure how much speech present probability in noisy speech by following equation |A(m,k)|2 Bmin(m,k) Bsp(m,k) = (12)
  • 6. Signal Image Processing : An International Journal (SIPIJ) Vol.5, No.5, October 2014 where Bmin(m,k) and |A(m,k)|2 are represented as local minimum and power spectrum of noisy speech. Speech present and speech absent are dependent on the ratio of speech present probability if it is grater to threshold then consider as speech present otherwise it gives speech absent. 3.6. Computing Logistic Function Logistic function is one of special case in the form of mathematical and it is also called as sigmoid function or sigmoid curve as given function g(x)=1/(1-e-x) Figure 2: sigmoid curve 3.7. Calculating Frequency Dependent Smoothing Constant To compute smoothing factor need the time B (m, k) = s+ (1- s) Bsp (m, k) Where, s is denotes as constant. time-frequency domain as to follow this equation To updating of minimum noisy spectrum is B B( m, k) = B(m − 1, k) + (1 − ) Bmin (m, k) and given equation Where, B (l, k) is average noise spectrum and , is known as smoothing factor b(, k) = b b( -1,k)+(1- b) I (, k) Where b (, k) is a smoothing constant, the above recursive absolutely utilize the correlation for speech presence in adjacent frames. For r (m, k) = 1. r(m, k) = N (m − 1, k)/N 2(m, k) Posterior SNR of smoothed version is represented by eq. (16) The Wiener filter solves the signal estimation problem for stationary signals. A major contribution was the use of a statistical model for the estimated signal the filter is best in the intellect of the MMSE [16]. . We shall focus here on the discrete-discrete time version of the Wiener filter and it is used to generate estimated pure signal from a given noise speech signal. 5 (13) [9]. (14) (15) (16) (17) he
  • 7. Signal Image Processing : An International Journal (SIPIJ) Vol.5, No.5, October 2014 6 4. IMPLEMENTATION AND RESULTS Speech enhancement algorithms are tested on MATLAB for Non-stationary environment of speech database [2]. The unvoiced speech regions to detected correctly by observed the results of proposed algorithm and speech activity region also accurately measured even noise is present. The table gives classification of results and performance of the algorithms in which segmental SNR and LLR (Log Like hood Ratio) are compared to proposed algorithm SR (Signal to Residual spectrum ratio). Spectrogram is way to visualize the speech signal in the domain time-frequency representation. In speech signal through several intermediate levels which are linguistic message and paralinguistic information including emotion is effectively visualized based on the spectrogram. Now we can concludes the variations in the noisy speech signal of Spectrogram represented in different areas those are trains, cars, and airport. Table1: comparison of weighted average technique and proposed SR technique using LLR and segmental SNR methods (i) Type of Noise LLR Segmental SNR SNR in db Weighted Average Technique Proposed (SR) Technique Weighted Average Technique Proposed (SR) Technique CAR 0 1.687827 1.500914 -6.806391 -6.716270 5 1.842711 1.596159 -5.668619 5.485975 10 1.976017 1.602708 -4.866237 -3.861581 15 1.831509 1.580956 -4.335797 -3.537122 AIRPORT 0 1.237398 1.057377 -3.802483 -3.440414 5 1.124488 0.934859 -2.781458 -2.526855 10 0.919158 0.736983 -0.731036 -0.083965 15 0.910468 0.549913 1.310788 3.080826 TRAIN 0 2.091845 1.798190 -6.486321 -6.296185 5 2.322675 1.845213 -5.559169 -4.970945 10 2.036162 1.759774 -5.251629 -4.206358 15 2.230337 1.827800 -3.211548 4.284449
  • 8. Signal Image Processing : An International Journal (SIPIJ) Vol.5, No.5, October 2014 7 (ii) (iii) (iv) Figure 3: timing wave form and spectrogram of (i) pure speech signal (ii) noisy speech signal and enhanced signal with (iii) weighted average technique (iv) proposed SR technique in car noise with different SNR levels. (i)
  • 9. Signal Image Processing : An International Journal (SIPIJ) Vol.5, No.5, October 2014 8 (ii) (iii) (iv) Figure 4: timing wave form and spectrogram of (i) pure speech signal (ii) noisy speech signal and enhanced signal with (iii) weighted average technique (iv) proposed SR technique in airport noise with different SNR levels. (i)
  • 10. Signal Image Processing : An International Journal (SIPIJ) Vol.5, No.5, October 2014 9 (ii) (iii) (iv) Figure 5: Timing wave form and spectrogram of (i) pure speech signal (ii) noisy speech signal and enhanced signal with (iii) weighted average technique (iv) proposed SR technique in Train noise with different SNR levels. 5. CONCLUSION This paper focused on the issue of noise estimation for enhancement of noisy speech. The noise estimate was updated continuously in every frame using time–frequency smoothing factors calculated based on speech-presence probability in each frequency bin of the noisy speech spectrum [1].The main achievements of speech enhancement algorithms are speech intelligibility and speech quality. Here to reduce the amplification distortion and attenuation distortion by using proposed method SR (Signal to Residual spectrum ratio) [5]. The proper control of these distortions to improve speech intelligibility it is main drawback of speech enhancement algorithms. The proposed method SR it gives better performance when compared to the previous existing methods are LLR (log like hood ratio) and segmental SNR.
  • 11. Signal Image Processing : An International Journal (SIPIJ) Vol.5, No.5, October 2014 REFERENCES [1] Anuradha R. Fukane1, Shashikant L. Sahare, “Noise estimation Algorithms for Speech Enhancement in highly non-stationary Environments”, IJCSI International Journal of Computer Science Issues, Vol. 8, Issue 2, March 2011. [2] D. Malah, R V Cox, and A J Accardi, “Tracking speech-presence uncertainty to improve speech enhancement in non-stationary noise environments, Proc.IEEE Int. Conf. Acoustics, Speech, Signal Processing, pp. 1153-1516, 2010. [3] Speech Enhancement for Non-stationary Noise Environments 978-1-4244-4994-1/09/$25.00 ©2009 10 IEEE. [4] K.Nakayama, H.Suzuki and A.Hirano, “Improved methods for noise spectral estimation and adaptive spectral gain control in noise spectral suppressor”., Proc. ISPACS‘ 07, Xiamen, China, pp.97-100, Dec. 2007. [5] S. Rangachari and P. C. Loizou, “A noise-estimation algorithm for highly non-stationary environments,” Speech Communication 48, 2006, pp. 220-231. [6] T.Lotter and P.Vary, “Noise reduction by joint maximum a posteriori spectral amplitude and phase estimation with super-gaussian speech modeling”., Proc. EUSIPCO-04, pp.1447-60, Sep. 2004. [7] Cohen.I.,“Noise spectrum estimation in adverse environments improved minima controlled recursive averaging”, IEEE Trans. Speech Audio Process., 11(5), pp. 466-475, 2003. [8] Cohen.I., “Noise spectrum estimation in adverse environments: improved minima controlled recursive averaging”, IEEE Trans. Speech Audio Process., 11(5), pp. 466-475, 2003. [9] Cohen.I, “Noise estimation by minima controlled recursive averaging for robust speech enhancement”, IEEE Signal Process. Lett., 9(1), pp. 12-15, 2002. [10] R. Martin, “Noise power spectral density estimation based on optimal smoothing and minimum statistics,” IEEE Trans. Speech Audio Process. vol. 9, no. 5, July 2001, pp. 504-512. [11] Martin.R, “Noise power spectral density estimation based on optimal smoothing and minimum statistics”, IEEE Tran. Speech Audio Process., 9(5), pp. 504-512,2001. [12] I. Cohen, “On speech enhancement under signal presence uncertainty,” in Proc. 26th IEEE Int. Conf. Acoust. Speech Signal Process.(ICASSP’2001), Salt Lake City, UT, May 7–11, 2001, pp. 167–170. [13] Sohn. J, Kim. N, “Statistical model-based voice activity detection”, IEEE Signal Process. Lett. 6(1), pp. 1-3, 1999. [14] Malah.D, Cox.R, Accardi.A, “Tracking speech-presence uncertainty to improve speech enhancement in non-stationary environments”, Proc. IEEE Internat. On Conf. Acoust. Speech Signal Process., pp. 789-792, 1999. [15] Doblinger.G, “Computationally efficient speech enhancement by spectral minima tracking in subbands”, Proc. Euro speech, pp.1513-1516, 1995. [16] N. Fan, “Low distortion speech denoising using an adaptive parametric Wiener filter”, In ICASSP2004, Montreal, Canada, pp.309-312, 2004. [17] J.H.L. Hansen, and B.L. Pellom, “An effective quality evaluation protocol for speech enhancement algorithm,” In Inter. Conf. on Spoken Language Processing, vol.7, pp. 2819-2822, Sydney, Australia, December 1998. [18] Ephraim, Y., Malah, D.,” Speech enhancement using a minimum mean-square error short-time spectral amplitude”, estimator. IEEE Trans. Acoust. Speech Signal Process. ASSP 32 (6), 1109–1121 1984. [19] Ephraim, Y., Van Trees, H.L., 1993. “A signal subspace approach for speech enhancement”. Proc. IEEE Internat. Conf. on Acoust. Speech, Signal Process. II, 355–358. [20] Hirsch, H., Ehrlicher, C., 1995. “Noise estimation techniques for robust speech recognition”. Proc. IEEE Internat. Conf. on Acoust. Speech Signal Process., 153–156.
  • 12. Signal Image Processing : An International Journal (SIPIJ) Vol.5, No.5, October 2014 11 Authors Ravichandra Kumar Manike pursuing M.Tech in the branch of Digital Electronics and Communication Systems at Gudlavalleru Engineering College and B.Tech degree in Electronics and Communication Engineering received from Prakasam Engineering College in the year of 2011.Gate qualified in the year 2012 13. Ravi Teja Ballikura received the B.Tech and M.Tech degree in Electronics and Communication Engineering in 2010 from Bapatla Engineering College, Digital electronics and communication systems in 2012 from Gudlavalleru Engineering College affiliated by JNTUK, Kakinada respectively. Working as a assistant professor in Gudlavalleru Engineering College from 2012 to till date. Research interests in speech processing and more especially in enhancement of speech signal.