• 대한전자공학회논문지
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• 제15권5호
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• pp.1-6
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• 1978

본 논문은 음성신호의 디지탈화와 대역식 축소에 관한 일부1)에 이은 이부 논문이다. 몇가지 근래에 개발된 Vocoding 방법, 즉 linear predictive coding (LPC), formant vocoding, residual excited linear prediction (RELP) vocoding,그리고 adaptive predictive coding(APC)에 관하여 논하였다. 본 논문에서는 음성전송에 있어서의 대역 제한 방법 중 지금 가장 효과가 있는 LPC방법을 중점적으로 취급하였다. 또한 현재 처하고 있는 문제점들과 해결책을 토의하였다.

• 말소리와 음성과학
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• 제6권3호
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• pp.91-99
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• 2014

Although it is well documented that patients with cochlear implant experience hearing difficulties when processing their first language, very little is known whether or not and to what extent cochlear implant patients recognize segments in a second language. This preliminary study examines how Korean learners of English identify English segments in a normal hearing and cochlear implant simulation conditions. Participants heard English vowels and consonants in the following three conditions: normal hearing condition, 12-channel noise vocoding with 0mm spectral shift, and 12-channel noise vocoding with 3mm spectral shift. Results confirmed that nonnative listeners could also retrieve spectral information from vocoded speech signal, as they recognized vowel features fairly accurately despite the vocoding. In contrast, the intelligibility of manner and place features of consonants was significantly decreased by vocoding. In addition, we found that spectral shift affected listeners' vowel recognition, probably because information regarding F1 is diminished by spectral shifting. Results suggest that patients with cochlear implant and normal hearing second language learners would experience different patterns of listening errors when processing their second language(s).

• 대한전자공학회논문지
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• 제15권6호
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• pp.1-7
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• 1978

本 論文에서는 음성신호의 디지털化와 대역폭축소의 한 방법으로 예측부호化 원리를 사용하는 adaptive differential pulse code modulation(ADPCM)과 adaptive delta modulation(ADM)에 관하여 고찰하였다. ADPCM에서 사용되는 대표적인 적응양자기의 원리를 설명하고 적응예측기의 계수를 얻는 두 방법, 즉 브록해석과 연차해석 방법을 검토하였다. 또한 ADM에서 사용되는 세가지 壓伸方法(instantaneous, syllabic, hybrid companding)을 구체적으로 설명하고 그의 성능을 비교하였다. 마지막으로 ADPCM과 ADM을 음성신호의 부호化器로 쓸 때의 성능과 장단점들을 비교 검토하였다.

• 한국음향학회:학술대회논문집
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• 한국음향학회 1994년도 FIFTH WESTERN PACIFIC REGIONAL ACOUSTICS CONFERENCE SEOUL KOREA
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• pp.899-903
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• 1994

This paper describes two methods to assess the output speech, quality of vocoders for telephone conferencing in digital mobile communication networks. The proposed methods are the sentence discrimiantion method and the modified degraded mean opinion score (MDMOS) test. We apply these two methods to Qualcomm code excited linear prediction (QCELP), vector sum excited linear prediction (VSELP) and regular pulse excited-long term predictin (RPE-LTD) voceders to evaluate which vocoding algorithm can process mixed voice signal from two speakers better for telephone conferencing. From the experiments we obtain that the VSELP vocoding algorithm reveals superior output speech quality to the other two.

• 한국음향학회:학술대회논문집
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• 한국음향학회 1985년도 학술발표회 논문집
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• pp.79-86
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• 1985

Residual-ecited linear prediction vocoding is known to be one of the best approaches to speech coding in the range of 4.8 to 9.6 kbits/s. One problem associated with the RELP vocoder is that it often produces some roughness and tonal noise as the transmission rate becomes lower. In this paper, we investigate three methods to improve its quality. These include the multiband spectral folding method, the method of using both the spectrally folded signal and the pulsed ecitation signal, and the method of using both the multiband spectrally folded signal and the pulsed ecitation signal. It has been found that, among the three methods, the last one yields the best performance. It produces no roughness and little tonal noise.

• 음성과학
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• 제7권1호
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• pp.17-29
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• 2000

The STC(Sinusoidal Transform Coding) is a vocoding technique that uses a sinusoidal speech model to obtain high- quality speech at low data rate. It models and synthesizes the speech signal with fundamental frequency and its harmonic elements in frequency domain. To reduce the data rate, it is necessary to represent the sinusoidal amplitudes and phases with as small number of peaks as possible while maintaining the speech quality. As a basic research to develop a low-rate speech coding algorithm using the sinusoidal model, in this paper, we investigate the speech quality depending on the number of sinusoids. By varying the number of spectral peaks from 5 to 40 speech signals are reconstructed, and then their qualities are evaluated using spectral envelope distortion measure and MOS(Mean Opinion Score). Two approaches are used to obtain the spectral peaks: one is a conventional STFT (Short-Time Fourier Transform), and the other is a multiresolutional analysis method.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2003년도 신호처리소사이어티 추계학술대회 논문집
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• pp.45-48
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• 2003

In general, speech coding methods are classified into the following three categories: the waveform coding, the source coding and the hybrid coding. Fast speaking is possible to encode with a few information compared with slow speaking rate. In case of speaking rate, low frequency band is more important than high frequency band while listening. Speech vocoding technique is developing to way with low bit rate and complexity and high sound quality. the CELP type of vocoder support very good sound quality with low bit rate but these vocoders don't consider about the speaking rate. When we consider speaking rate and encode the frame depending on the speaking rate, the bit rate is able to reduce the bit rate than the conventional vocoder. We propose the technique to estimate the speaking rate and applied PSOLA technique in case of the frame of slow speaking rate. As a result of simulation bit rate can be reduced about 300 bps.

• ETRI Journal
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• 제19권3호
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• pp.242-258
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• 1997

The Qualcomm code excited linear prediction (QCELP) speech coder was adopted to increase the capacity of the CDMA Mobile System (CMS). In this paper, we implemented the QCELP speech coding algorithm by using TMS320C50 fixed point DSP chip. Also the fixed point simulation was done with C language. The computation complexity of QCELP on TMS320C50 was 10k words and data memory was 4k words. In the normal call test on the CMS, where mobile to mobile call test was done in the bypass mode without double vocoding, mean opinion score for the speech quality was he Qualcomm code excited linear prediction (QCELP) speech quality was 3.11.

• 대한전기학회:학술대회논문집
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• 대한전기학회 1999년도 하계학술대회 논문집 G
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• pp.3246-3248
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• 1999

The APC is a coding algorithm which has the middle property of both Wave Coding(ex ADPCM) and Vocoding(ex CELP) and can decode a proper quality of sound by using scalar quantizer instead of vector quantizer at computation a low calculation. So, the APC required for Voice Codec of INMARSAT-B could be successfully implemented by full duplex using TMS32OC30(DSP).

• 대한의용생체공학회:의공학회지
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• 제20권4호
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• pp.427-433
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• 1999

본 논문에서는 고음질 음성부호화 및 압축, 음성인식 및 합성등의 성능개선에 있어서 중요한 파라메터인 피치정보를 실시간으로 검출하기 위해 연구를 수행하였다. 이를 위하여 변복조 스폿(spot) 전극을 적용한 새로운 EGG(Electroglottograph) 측정 시스템을 개발하여 이용한 안정된 피치검출 알고리즘을 연구하였다. 구체적으로 EGG 신호에 의한 성문 닫힘시점을 실시간으로 결정하여 EGG 기반의 피치궤적 알고리즘을 개발하였고, 음성신호만의 피치궤적 추적기와 성능비교를 수행하여 우월한 성능을 가짐을 보였다. 또한, EGG 신호를 이용한 음성분석을 수행하여 한국인에 있어서 성문의 다양한 진동신호 패턴의 측정 및 분석을 통해 한국인 음원의 모델과 성문신호 패턴에 대한 정량적인 해석을 하였다.