• Proceedings of the IEEK Conference
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• 2001.09a
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• pp.809-812
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• 2001

본 논문에서는 AMR(Adaptive Multi-Rate)과 MLT (Modulated Lapped Transform) 벡터 양자화 방법을 이용하여 광대역 음성부호화기를 설계하였다. 제안한 음성부호화 알고리즘은 split-band 구조를 가지고 있으며 16kHz로 샘플링 된 신호를 입력받아 QMF 필터에 의해 두 개의 대역으로 나누어, 각각 8kHz 샘플링 신호로 변환시킨 후 저대역(0Hz-3400Hz)의 신호와 고대역(3400Hz -7000Hz)의 신호로 나누어 각각 부호화한다. 나누어진 두 개의 협대역 음성신호는 AMR(Adaptive Multi-Rate)부호화기와 MLT (Modulated Lapped Transform)벡터 양자화 방법을 사용하여 각각 부호화되어 전송된다. 수신단에서는 각 대역을 AMR과 IMLT(Inverse MLT) 벡터 양자화 방법으로 역부호화하여 음성신호를 합성한다. 제안한 음성부호화기는 20.2kbps에서 12.15kbps까지의 다전송률로 동작된다. 설계된 광대역 음성부호화기는 MOS시험 결과로부터 G.722의 56 kbps 음성이 설계된 코더의 20.2 kbps와 비슷한 음질을 갖음을 확인할 수 있었다.

• Journal of the Korean Institute of Telematics and Electronics B
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• v.33B no.3
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• pp.126-132
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• 1996

In this paper, the complexity reduction method for QCELP speech coder (IS-96) without any perfomrance degradation is proposed for the vecoder of CDMA digital cellular system. The energy terms in pitch parameter search and codebook search routines that require large computations are calculated recursively by utilizing the overlapped structure of code vectors in adaptive codebook and excitation codebook. The additional complexity reduction in the codebook search routine can be achieved by using a simple form in calculation of the energy term when the initial codebook value is zero. In the case of lower transmission rates such as 4,2,1 kbps, the complexity reduction by recursive calulations of energy term is increased.

• The Journal of the Acoustical Society of Korea
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• v.27 no.1E
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• pp.11-16
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• 2008

In this paper, a pitch cycle waveform (PCW) is extracted, quantized, and interpolated in a time domain to synthesize high-quality speech at low bit rates. The pre-alignment technique is proposed for the accurate and efficient PCW extraction, which predicts the current PCW position from the previous PCW position assuming that pitch periods evolve slowly. Since the pitch periods are different frame by frame, the original PCW is converted into the fixed-dimension PCW using the dimension-conversion method, and subsequently quantized by code-excited linear predictive (CELP) coding. The excitation signal for the linear predictive coding (LPC) synthesis filter is generated using the time-domain interpolation and interlink of the quantized PCW's. The coder operates at 4.2 kbit/s and 3.2 kbit/s depending on the pitch period. Informal listening test demonstrates the effectiveness of the proposed coding scheme.

• Proceedings of the IEEK Conference
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• 2003.07e
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• pp.2228-2231
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• 2003

In this paper, a transcoding algorithm for the Selectable Mode Vocoder (SMV) and the G.723.1 speech coder via direct parameter transformation is proposed. In contrast to the conventional tandem transcoding algorithm, the proposed algorithm converts the parameters of one coder to the Other Without going through the decoding md encoding process. The proposed algorithm is composed of four parts: the parameter decoding, line spectral pair (LSP) conversion, pitch period conversion, excitation conversion and rate selection. The evaluation results show that the proposed algorithm achieves equivalent speech quality to that of tandem transcoding with reduced computational complexity and delay.

• Proceedings of the Acoustical Society of Korea Conference
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• 1994.06c
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• pp.199-204
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• 1994

In this paper, we researched abut CELP speech coding algorithm using efficlent pseudo-stochastic block codes, adaptive-codebook and improved fixed-gain codebook. The pseudo-stochastic block codes refer to stochastically populated block codes in which the adjacent codewords in an innovation codebook are non-independent. The adaptive-codebook was made with previous prediction speech data by storage-shift register. This CELP coding algorithm enables the coding of toll quality speech at bit rates from 4.8kbits/s to 9.6 kbits/s. This algorithm was realized TMS320C30 microprocessor in realtime.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.38 no.5
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• pp.576-582
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• 2001

The Multi-Band Excitation(MBE) speech coder uses a different approach for the representation of the excitation signal. It replaces the frame-based single voiced/unvoiced classification of a classical speech coder with a set of such decision over harmonic intervals in the frequency domain. This enables each speech segment to be a mixture of voiced and unvoiced, and improves the synthetic speech quality by reducing decision errors that might occur on the frame-based single voiced and unvoiced decision process when input speech is degraded with noise. The IMBE-LP, improved version of MBE with linear prediction, represents the spectral information of MBE model with linear prediction coefficients to obtain low bit rate of 2.4 kbps. In this Paper, we proposed a variable rate IMBE-LP vocoder that has lower bit rate than IMBE-LP without degrading the synthetic speech quality. To determine the LP order, it uses the spectral band information of the MBE model that has something to do with he input speech's characteristics. Experimental results are riven with our findings and discussions.

• The Journal of the Acoustical Society of Korea
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• v.23 no.5
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• pp.420-427
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• 2004

This paper presents real time implementation of 4kbps EHSX (Enhanced Harmonic Stochastic Excitation) speech coder that combines the harmonic vector excitation coding with time-separated transition coding. The harmonic vector excitation coding uses the harmonic excitation coding for voiced frames and used the vector excitation coding with the structure of analysis-by-synthesis for unvoiced frames, respectively. For transition frames mixed with voiced and unvoiced signal, we use the time-separated transition coding. In this paper. we present the optimization methods of implementation speech coder on the EMS320C6701/sup (R)/ DSP. To reduce the complex for real-time implementation. we perform the optimization method in algorithm by replacing the complex sinusoidal synthesis method with IFFT. and we apply fully pipelines hand assembly coding after converting it from floating source to fixed source. To generate a more efficient code. we also make use or the available EMS320C6701/sup (R)/ resources such as Fastest67x library and memory organization.

• Proceedings of the IEEK Conference
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• 2003.11a
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• pp.57-60
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• 2003

In this paper, we propose a new excitation enhancement technique to improve the speech quality of low bit rate speech coders. The proposed technique is based on a harmonic model and it is employed only in the decoding process of speech coders without any additional bits. We develop the procedure of harmonic model parameters estimation and harmonic generation. and apply the technique to a current state of the art low bit rate speech coder, ITU-T G.729 Annex D. Also its performance is measured by using the ITU-T P.862 PESQ score and compared to those of the phase dispersion filter and the long-term postfilter applied to the decoded excitation. It is shown that the proposed excitation enhancement technique can improve the quality of decoded speech and provide better quality for male speech than other techniques.

• Proceedings of the KSPS conference
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• 2003.05a
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• pp.203-206
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• 2003

In this paper, we employ the adaptive comb filtering for effective noise reduction in mobile communication environment. Adaptive comb filtering is a well- known method for noise reduction, but requires the correct pitch period and must be applied just in voiced speech frames. To satisfy these requirements we use two kinds of information extracted from speech packets, one of which is the pitch period information measured precisely by a speech coder and the other is the frame rate information related to a decision on speech or silence frame. Experiments on speech recognition system confirm the efficiency of this method. Feature parameters employing this method give superior performance in noise environment to those extracted directly from output speech.

• The Journal of the Acoustical Society of Korea
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• v.31 no.7
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• pp.472-479
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• 2012

This paper is aimed to design 9.2 kbps low bits late transform excitation coder that target to voice and audio signal. To set up low bit rate, we used Band-selection in frequency domain and gain-shape quantization and AbS structure. To decrease lots of calculation from ABS structure, we used each band IDFT and synthesis. And we designed non-transfer band for performance by inserting comfort noise. We propose coder that has low bit rate and similar performance comparing with original 10.4 kbps AMR-WB+ TCX mode.