• The Journal of the Acoustical Society of Korea
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• v.14 no.4
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• pp.38-45
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• 1995

In this paper a way to improve the performance of the long term prediction is proposed, which adopts the Multi-band Excitation (MBE) method in addition to the Code-Excited Linear Prediction (CELP) method at low bit rates below 4.8 kbps. In the proposed method, the multiband long term prediction is performed on the periodic components which still remain after the long term prediction of the conventional CELP method. At this point, the whole frequency region is divided into subbands whose size is equal to the spacing between the harmonics of the fundamental frequency, and the periodic multiband excitation signals. are represented as the sum of sine waves approximately as large as the spectrum of the excitation signals, so that the actual characteristics of the excitation signals can be better taken into account. To evaluate the performance of the proposed method, computer simulation is performed at 4.8 kbps. The 4.8 kbps DoD CELP and the 4.4 kbps IMBE were chosen as the reference vocoders for the speech quality measure. The result of the perceptual speech quality measure showed that the performance of the proposed method is better than that of the 4.8 kbps DoD CELP vocoder, and similar to that of the 4.4 kbps IMBE vocoder.

• The Journal of the Acoustical Society of Korea
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• v.13 no.2E
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• pp.20-25
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• 1994

The major drawback to the Code Excited Linear Prediction(CELP) type vocoders is their large computational requirements. In this paper, a simple method is proposed to reduce the pitch searching time in the pitch filter almost without degradation of quality. Bease upon the observational regularity of the correlation function of speech, the searching range can be restricted to the positive side in pitch search. This is done by skipping the negative side with the width which is estimated from the previous positive envelope. In addition to that, the maximum number of available lags can be limited by the threshold, $L_T$, which is set on 58 empirically. So, only the limited numbers of lags are considered in pitch search, which is less than a half of that of the full search method. By using the proposed method in pitch search, its required computations are greatly reduced. Experimental result shows 51% time reduction almost without lowering the speech quality in segmental SNR measure.

• Proceedings of the IEEK Conference
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• 2000.06d
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• pp.98-101
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• 2000

On CELP type Vocoders G.723.1 6.3kbps/5.3kbps Dual Rate Speech Codec, which is developed for Internet Phone and videoconferencing, uses VAD(Voice Activity Detection)/CNG (Comfort Noise Generator) in order to reduce the bit rate in a silence period. In order to reduce the bit rate effectively in this paper, we first set the boundary condition of the energy threshold to prevent the consumption of unnecessary processing time, and use three decision rules to detect an active frame by energy, pitch gain and LSP distance. To evaluate the performance of the proposed algorithm we use silence-inserted speech data with 0, 5, 10, 20dB of SNR. As a result when SNR is over 5dB, the bit rate is reduced up to about 40% without speech degradation and the processing time is additionally decreased.

• Proceedings of the Acoustical Society of Korea Conference
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• 1994.06c
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• pp.214-217
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• 1994

The major drawback in the Code Excited Linear Prediction type vocoders is their large computational requirements. In this paper, a simple method is proposed to reduce the pitch searching time in the pitch filter almost without degradation of quality. On the basis of the observational regularity of the correlation function of speech, only the limited numbers of pitch lags are considered to be an optimum pitch. This is done by skipping the negative envelope side of the correlation function and limiting the maximum number of lags to be considered preliminarily. By doing so, we can reduce the computational time of pitch searching more than 51% with negligible quality degradation. In addition to that, by combining that method with the conventional delta search technique, we can reduce the computational time requirements more than 60% without serious lowering the speech quality in segmental SNR measure compared to the conventional full search method.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.13 no.2
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• pp.962-977
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• 2019

The human voice is a convenient method of information transfer between different objects such as between men, men and machine, between machines. The development of information and communication technology, the voice has been able to transfer farther than before. The way to communicate, it is to convert the voice to another form, transmit it, and then reconvert it back to sound. In such a communication process, a vocoder is a method of converting and re-converting a voice and sound. The CELP (Code-Excited Linear Prediction) type vocoder, one of the voice codecs, is adapted as a standard codec since it provides high quality sound even though its transmission speed is relatively low. The EVRC (Enhanced Variable Rate CODEC) and QCELP (Qualcomm Code-Excited Linear Prediction), variable bit rate vocoders, are used for mobile phones in 3G environment. For the real-time implementation of a vocoder, the reduction of sound quality is a typical problem. To improve the sound quality, that is important to know the size and shape of noise. In the existing sound quality improvement method, the voice activated is detected or used, or statistical methods are used by the large mount of data. However, there is a disadvantage in that no noise can be detected, when there is a continuous signal or when a change in noise is large.This paper focused on finding a better way to decrease the reduction of sound quality in lower bit transmission environments. Based on simulation results, this study proposed a preprocessor application that estimates the SNR (Signal to Noise Ratio) using the spectral SNR estimation method. The SNR estimation method adopted the IMBE (Improved Multi-Band Excitation) instead of using the SNR, which is a continuous speech signal. Finally, this application improves the quality of the vocoder by enhancing sound quality adaptively.