• Proceedings of the IEEK Conference
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• 2001.06d
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• pp.115-118
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• 2001

This paper presents dispersed-pulse and random codebook for CELP coder. This coder operates on speech frames of 20ms and generates an excitation vector by convoluting dispersion vectors with signed pulses in an algebraic codevector. The improvement of pulse-based fixed codebook is performed at a low bit rate. A high performance fixed-codebook consists of a partial algebraic codebook and a random codebook in unvoiced and stationary noise regions. The proposed CELP coder is quantized with 4kb/s and is compared with G.729 (Bkb/s CS-ACELP). Subjective testing shows better quality than reference coders under some background noise conditions

• Journal of IKEEE
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• v.15 no.2
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• pp.171-178
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• 2011

In this paper, a novel pulse repetition interval (PRI) jittering coder based on quasi-random M-sequence codes is proposed for improvement of counter-countermeasure capability in a radar. Each of the proposed jittered 256 PRI codes has a unique code chip combination with 256 code chips, such that any set of three consequent code chips (4 pulses) from any code appears only once among the entire code chip sequences of the codes. This indicates that only 4 of received pulses are enough to determine uniquely the exact timing position of the incoming pulse train (or code chip sequence) required for counter-countermeasure, as well as the identity of the transmitted code. To prove the proposed idea experimentally, the jittered PRI coder is implemented and demonstrated.

• Proceedings of the Acoustical Society of Korea Conference
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• 1995.06a
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• pp.146-149
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• 1995

This paper proposes a SM-CELP speech coder which applies different excitation signal according to the characteristic of speech segment at bit-rate below 4 kbps. Speech signal is divided with 2 modes such as stationary voice and etc. using the parameters of average energy of the short-time speech and the residual signal after long term prediction. Structured multi-pulse method is used for the excitation of mode-A and gaussian or pulse-like codebook for mode-B. 4.8kbps DoD-CELP are used to evaluate the performance of the proposed coder. As a result, the propose method shows 1~2 dB higher segmental signal to noise ratio and better subjectional quality without increasing the computational amount.

• The Journal of the Acoustical Society of Korea
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• v.20 no.5
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• pp.35-47
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• 2001

This paper describes an efficient harmonic-CELP speech coder by taking advantages of harmonic and CELP coders into account. According to frame voicing decision, the proposed harmonic-CELP coder adopts the RP-VSELP coder as a fast CELP in case of an unvoiced frame, or an improved harmonic coder in case of a voiced frame. The proposed coder has main features as follows: simple pitch detection, fast harmonic estimation, variable dimension harmonic vector quantization, perceptual weighting reflecting frequency resolution, fast harmonic synthesis, naturalness control using band voicing, and multi-mode. These features make the proposed coder require very low complexity, compared with HVXC coder To demonstrate the performance of the proposed coder, a 2.4 kbps coder has been implemented and compared with reference coders. From results of informal listening tests, the proposed coder showed good quality while requiring low delay and complexity.

• Proceedings of the Acoustical Society of Korea Conference
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• 1995.06a
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• pp.246-249
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• 1995

4.16kbps의 전송률을 갖는 음성 부호화기 구조에 관하여 기술한다. 제안된 음성 부호화기는 개방 회로 피치 검출기와 이로부터 생성된 pulse train을 코드북으로 갖는 CELP 부호화기이다. Pulse-Train codebook은 분석 프레임별로 부호화 및 복호화 양단에서 생성되며 음성의 피치 및 포만트 정보를 내포하고 있다. 구현된 PT-CELP는 random codebook 방식의 CELP에 비해 적은 크기로 codebook을 만들 수 있으며 음성의 특징을 충분히 반영하므로 합성된 음성의 음질을 향상시킬 수 있다.

• Journal of the Korean Institute of Telematics and Electronics
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• v.27 no.5
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• pp.784-792
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• 1990

In this paper, attempts have been made to compare the subjective performance of sub-band coding (SBC) techniques incorporating four representative modulation transfer functions (MTF's) of th human visual system(HVS): Sakrison, Nill, Ngan, and Rao. In SBC, the frequency band of input signal is split into 16 equal sub-bands. In addition, 28 sub-band splitting schemd which splits more sub-bands in low spatial frequency is considered to compare the 4 MTF's effectively. In encoding of each sub-band, the weight of each sub-band obtained from the MTF of HVS is applied to bit allocation process which minimize the weighted mean square error (WMSE). The differential pulse code modulation(DPCM) coder is used to encode the lowest sub-band and the pulse code modulation(PCM) coder is used for the rest of sub-bands. It is found that the images incorporating the MTF of Rao yields the best results in subjective criteria, followed by Ngan, Nill, Sakrison, and the images not employing the HVS.

• Proceedings of the Acoustical Society of Korea Conference
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• 1998.08a
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• pp.136-141
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• 1998

In order to enhance the performance of a CELP coder at low bit rates, it would be necessary to make the CELP excitation have the peaky pulse characteristic. In this paper we introduce an excitation signal with peaky pulse characteristic. It is obtained by using a two-tap pitch predictor. Samples of the signal have different gains according to their amplitudes by the predictor. In voiced sound the signal has the desirable peaky pulse characteristic, and its periodicity is well reproduced. Particularly, peaky pulses at voiced onset and a burst of plosive sound are clearly reconstructed.

• International Journal of Fuzzy Logic and Intelligent Systems
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• v.16 no.4
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• pp.276-280
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• 2016

A fixed rate speech coder based on the filter bank and the non-uniform sampling technique is proposed. The non-uniform sampling is achieved by the detection of inflection points (IPs). A speech block is band passed by the filter bank, and the subband signals are processed by the IP detector, and the detected IP patterns are compared with entries of the IP database. For each subband signal, the address of the closest member of the database and the energy of the IP pattern are transmitted through channel. In the receiver, the decoder recovers the subband signals using the received addresses and the energy information, and reconstructs the speech via the filter bank summation. As results, the coder shows fixed data rate contrary to the existing speech coders based on the non-uniform sampling. Through computer simulation, the usefulness of the proposed technique is confirmed. The signal-to-noise ratio (SNR) performance of the proposed method is comparable to that of the uniform sampled pulse code modulation (PCM) below 20 kbps data rate.

• Proceedings of the IEEK Conference
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• 1998.10a
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• pp.1219-1222
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• 1998

This paper analyzes the dynamic spars algebraic codebook used to model a residual signal and proposes a new algebraic codebook structure as well as a searching process with improved performance. The proposed algorithm improves the disadvantage of algebraic codebook without increased computation. First, this paper makes it possibel to select various pulse amplitudes differently from the conventional method which looks up the sign bit simply. In addition, two pulses are made to be selected on the same track. For speech quality on the telephone line 5.6kbps speech coder using the proposed algorithm was equivalent to the 6.3kbps MP-MLQ in the viewpoint of subjective speech quality. However, speech degradation was caused a little compared to the MP-MLQ where MNRU 1=15dB.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.42 no.2 s.302
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• pp.131-142
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• 2005

The low rate speech coders under 4 kbit/s are based on sinusoidal transform coding (STC) or multiband excitation (MBE). Since the harmonic coders are not efficient to reconstruct the transient segments of speech signals such as onsets, offsets, non-periodic signals, etc, the coders do not provide a natural speech quality. This paper proposes method of a efficient transient model :d a multi-mode low rate coder at 2.4 kbit/s that uses harmonic model for the voiced speech, stochastic model for the unvoiced speech and a model using aperiodic pulse location tracking (APPT) for the transient segments, respectively. The APPT utilizes the harmonic model. The proposed method uses different models depending on the characteristics of LPC residual signals. In addition, it can combine synthesized excitation in CELP coding at time domain with that in harmonic coding at frequency domain efficiently. The proposed coder shows a better speech quality than 2.4 kbit/s version of the mixed excitation linear prediction (MELP) coder that is a U.S. Federal Standard for speech coder.