• Proceedings of the IEEK Conference
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• 2000.09a
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• pp.131-135
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• 2000

본 논문에서는 고성능 고정 소수점 DSP (Digital Signal Processor) 코어인 Vincent6 코어 [1]를 이용하여 ITU-T C.728 음성 부호화기를 실시간으로 구현하였다 G.728 은 16 kb/s전송률의 ITU-T표준 음성 부호화기이며, 입력신호는 8 kHz로 샘플링되며 샘플 당 16 bit 로 양자화된 PCM 신호이다. G.728 은 LD-CELP(Low Delay Code Excited Linear Prediction)라고도 하며, 알고리 듬 delay는 0.625ms 이다. Vincent6 DSP core 는 VLIW (Very-Long Instruction Word) 특성을 가지므로 다중 명령 (multiple instruction)을 수행할 수 있다 이를 위해서 G.728 annex G를 이용하여 고정 소숫점 연산으로 코드를 작성한 후, 이를 vincent6 어셈블리 코드로 구현하였다. 최종적으로 구현된 코드는 ITU-T 의 test vector 에 대 해 bit exact 한 결과를 보이며 34 MCPS (Million Cycles Per Second)의 계산량을 가지며 사용 메모리크기는 데이터 메모리가 약 9KByte, 프로그램 메모리가 약 57 KByte 이다.

• Proceedings of the KIEE Conference
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• 2005.07d
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• pp.2762-2764
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• 2005

In this paper, we study the robot controller design using the voice and data communication via CDMA(Code Division Multiple Access) mobile communication network. We design the robot remote controller using the three methods, telephone call speech recognition, DTMF (Dual Tone Multiple Frequency) realization, SMS(Short Message Service) transmission/reception way via CDMA mobile communication network. We investigate the validity and effectiveness of the proposed remote controller which applied to the mobile robot.

• Proceedings of the IEEK Conference
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• 1998.06a
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• pp.19-22
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• 1998

In this study, the efficient channel coding method combined with CS-ACELP is proposed. The same convolutional coder and Viterbi decoder of COMA mobile communication system is used as channel coder. To make the best available use of limited channel coding redundancy, unequal error protection of punctured convolutional coder is used for variable reate allocation. But, the overall code rate is given by 2. The performance of proposed coder is analyzed and simulated in a Rayleigh fading channel. Experimental results show that the objective and subjective speech quality of variable rate channel coding methods are superior to those of non-variable channel coding method.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.21 no.10
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• pp.2634-2644
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• 1996

Two different low rate speech coders using one of four types of lattice vector quantizers(LVQ's) with fairly low complexity were investigated for an application to mobile communications. More specifically, two-stage vector quantizer-lattic vector quantizer(VQ-LVQ) systems and vector differenctial pulse code modulation(VDPCM)systems with lattice vector quantizers simulated to encode the line spectrum frequencies of various sentences at the rate 22 to 39 bits per 20 msec frame. The simulation results showed that the VDPCM system with the lattice VQ can save up to 10 bits/fram compared to the quantization scheme used in QCELP system. For the VQ-LVQ system, the spherical quasi-uniform LVQ below 36 bits/frame outperformed the other 3 types of LVQ's and the pyramidal quasi-uniform LVQ at 37 bits/frame outperformed the other 3 types of LVQ's with the spectral distortion 0.97.

• The Transactions of the Korean Institute of Electrical Engineers
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• v.45 no.2
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• pp.325-328
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• 1996

In this paper, we proposed an estimation method of a posterior probability and PDF(Probability density function) using a feed forward neural network and code books of VQ(vector quantization). In this study, We estimates a posterior probability and probability density function, which compose a new parameter with well-known Mel cepstrum and verificate the performance for the five vowels taking from syllables by NN(neural network) and PNN(probabilistic neural network). In case of new parameter, showed the best result by probabilistic neural network and recognition rates are average 83.02%.

• Proceedings of the Acoustical Society of Korea Conference
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• 1994.06a
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• pp.1104-1111
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• 1994

Code Excited Linear Prediction(CELP) as a speech coder exhibits good performance at data rates below 4.8 kbps. The major drawback to CELP type coders is their large amount of computation. In this paper, we propose a new pitch search method that preserves the quality of the CELP vocoder with reduced complexity. The basic idea is to restrict the pitch searching range by estimating the preliminary pitches. Applying the proposed method to the CELP vocoder, we can get approximately 87% complexity reduction in the pitch search.

• Journal of the Korean Institute of Telematics and Electronics
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• v.23 no.5
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• pp.597-604
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• 1986

This paper describes various mismatch phenomena in differential pulse code modulation (DPCM) coding, such as the mismatch effects of probability density functin(pdf), signal variance, and correlation. At a high transmission rate(i.e., above 32 kbits/s), the performance of DPCM can be improved by matching the pdf shape between the input signal and the quantizer. However, the same gain cannot be obtained at a lower transmission rate. Also, it is shown that the gamma quantizer is realtively robust to the variation of pdf shaper and signal variance. Moreover, as the transmission rate increases, the performance of DPCM for the input signal with large variance is worse than that of DPCM for the signal with small variance due to the increase of overload noise. According to our simuladiton results, the mismatch effects of pdf shape and variance appear to yield more degradatin than that of correlation in a DPCM system.

• 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.

• Journal of the Korean Institute of Telematics and Electronics
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• v.21 no.3
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• pp.73-83
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• 1984

The performances of embedded adaptive differential pulse code modulation (ADPCM), embedded adaptive delta modulation (ADM), and the same systems with a delayedfecision scheme have been studied with real speech over a wide dynamic range. The embedded ADPCM and ADM coders have been obtained by modifying the conventional ADPCM and ADM coders. The basic scheme of the embedded ADPCM coder is based on the ADPCM originally proposed by Cummiskey et at. For embedded ADM systems, we have modified continuously variable slope DM (CVSD) and hybrid commanding DM (HCDM) systems. Among these embedded coders, the performance of the embedded HCDM is superior to the other coders over a wide range of transmission rate from 16 to 64 kbits/s, When the delayedtecision scheme is applied to the embedded ADPCM the performance is improved significantly at all transmission rates. But, in the embedded ADM systems with 16 kHz sampling rate, the performance improvement resulting from delayed decision is not drastic as is in the embedded ADPCM with the same number of delayed samples.

• 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.