• 음성과학
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• 제8권4호
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• pp.45-58
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• 2001

Modulated components of a speech signal are frequently used for speech coding, speech recognition, and speech synthesis. Time-frequency representation (TFR) reveals some information about instantaneous frequency, instantaneous bandwidth and boundary of each component of the considering speech signal. In many cases, the extraction of AM-FM components corresponding to instantaneous frequencies is difficult since the Fourier spectra of the components with time-varying instantaneous frequency are overlapped each other in Fourier frequency domain. In this paper, an efficient method decomposing speech signal into AM-FM components is proposed. A variable bandwidth filter is developed for the decomposition of speech signals with time-varying instantaneous frequencies. The variable bandwidth filter can extract AM-FM components of a speech signal whose TFRs are not overlapped in timefrequency domain. Also, amplitude and instantaneous frequency of the decomposed components are estimated by using Hilbert transform.

• 한국정보통신학회:학술대회논문집
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• 한국정보통신학회 2018년도 추계학술대회
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• pp.431-433
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• 2018

본 논문은 광대역 음성의 음질 향상을 위해 시간 영역에서 인공대역 확장 기술을 사용하여 초광대역 음성신호를 출력하여 사용자에게 개선된 음질의 음성을 제공하는 시스템을 제안한다. 시간 영역에서 소스필터 모델에 기반하여 광대역 여기신호 및 LSP를 추출하고, 각각의 대역폭 확장 알고리즘을 적용였고, 초광대역 여기신호 및 LSP를 추론하여 초광대역 음성신호를 합성한다. 주관적인 테스트를 통해 광대역 음성신호보다 초광대역 음성신호의 음질을 더 선호하는 결과를 도출하였다.

• 음성과학
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• 제12권2호
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• pp.153-170
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• 2005

The sub-Nyquist nonuniform sampling (SNNS) and the perfect reconstruction (PR) formula are proposed for the development of a systematic method to obtain minimal representation of a speech signal. In the proposed method, the instantaneous sampling frequency (ISF) varies, depending on the least upper boundary of spectral support of a speech signal in time-frequency domain (TFD). The definition of the instantaneous bandwidth (IB), which determines the ISF and is used for generating the set of samples that represent continuous-time signals perfectly, is given. Also, the spectral characteristics of the sampled data generated by the sub-Nyquist nonuniform sampling method is analyzed. The proposed method doesn't generate the redundant samples due to the time-varying property of the instantaneous bandwidth of a speech signal.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2003년도 하계종합학술대회 논문집 Ⅳ
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• pp.2208-2211
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• 2003

In this paper, a variable bandwidth filter(VBF) is realized with the purpose of the decomposition of speech signals with time-varying instantaneous of frequencies. The proposed VBF can extract AM-FM components of a speech signal whose time-frequency representations(TFRs) are not overlapped in time-frequency domain

• 한국음향학회지
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• 제24권7호
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• pp.395-401
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• 2005

본 논문에서는 대역폭 계층 구조의 광대역 음성 부호화기를 위한 8kbps 상위 대역 부호화기를 제안한다. 광대역 입력 음성 신호는 하위 대역 신호와 상위 대역 신호로 분리되고, 하위 대역 신호는 표준 협대역 음성 부호화기로 처리하고, 상위 대역 신호는 제안하는 부호화기로 처리하는 구조를 가진다. 제안한 상위 대역 부호화기는 입력 신호를 부프레임 단위로 MLT 변환하고 MLT 계수를 크기와 부호로 분리하여 각각의 특성에 적합하도록 양자화 한다. MLT 계수 크기는 다수의 시간과 주파수 성분을 함께 가지는 밴드로 분할하고, 각 밴드 신호는 2차원 DCT 변환하여 양자화하며, 하위 대역의 에너지 정보를 이용하여 양자화 성능을 향상시킨다. MLT 계수 부호는 각각의 중요도를 판정하여 일부 부호만 선택적으로 양자화 하는 방법을 사용한다. 제안한 상위 대역 부호화기를 포함하는 19.8kbps 광대역 부호화기의 객관적 성능과 주관적 성능을 측정하였으며, 32kbps G.722.1보다 우수한 성능을 가지는 것을 확인하였다.

• 대한음성학회:학술대회논문집
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• 대한음성학회 2002년도 11월 학술대회지
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• pp.175-178
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• 2002

Most existing telephone networks transmit narrowband speech witch has been bandlimited below 4 kHz. Compared with wideband speech up to 8 kHz, narrowband speech shows reduced intelligibility and a muffled quality. Bandwidth extension is a technique to generate wideband speech by reconstructing 4-8 kHz highband speech without any additional information. This paper presents experimental results of the bandwidth extension adopted for 4800 bps CELP speech coder. In this experiment, we examine various methods for reconstruction of wideband spectrum and excitation signal, compare and analyze their performance by performing the subjective preference test and measuring the cepstral distortion.

• 대한음성학회지:말소리
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• 제61호
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• pp.75-86
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• 2007

In this paper, we propose a scalable audio coder, which has a variable bandwidth from the narrowband speech bandwidth to the audio bandwidth and also has a bit-rate from 8 to 320 kbits/s, in order to cope with the quality of service(QoS) according to the network load. First of all, the proposed scalable coder splits bandwidth of the input audio into narrowband up to around 4 kHz and above. Next, the narrowband signals are compressed by a speech coding method compatible to an existing standard speech coder such as G.729, and the other signals whose bandwidth is above the narrowband are compressed on the basis of a psychoacoustic model. It is shown from the objective quality tests using the signal-to-noise ratio(SNR) and the perceptual evaluation of audio quality(PEAQ) that the proposed scalable audio coder provides a comparable quality to the MPEG-1 Layer III (MP3) audio coder.

• 한국음향학회지
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• 제4권1호
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• pp.16-21
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• 1985

Both shifiting the input signal's frequencies by a fixed frequency and compressing the input signal's bandwidth have been known to be effective in improving the stability margin of public adress systems operating in reverberant spaces. This paper describes the effect of an alternative approach of improving the acoustic-feedback stability and yet maintaining speech inteligibility by bandwidth compression and expansion. Conditions are derived for this technizue to be realized and an experimental system has been made - up. A series of experiments has been performed in small spaces and the results have shown that more than 5dB improvement can be obtained in the stability margin.

• 한국음향학회지
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• 제23권6호
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• pp.481-487
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• 2004

본 논문에서는 대역폭 계층 구조의 광대역 음성 부호화기 구조와 이를 위한 상위 대역 부호화기를 제안한다. 상위 대역 부호화기는 상위 대역 신호를 다수의 주파수 밴드로 분할하고, 각 밴드 신호를 DCT 변환하여 양자화한다. DCT 계수는 크기와 부호로 분리하여 독립적으로 처리하며, 각각의 특성에 적합한 양자화 방법을 개발하여 성능을 향상시킨다. 또한, 하위 대역의 부호화된 이득 파라미터를 상위 대역의 양자화에 이용함으로써 부호화 성능을 향상시키는 방법을 제안한다. G.729E로 하위 대역을 부호화하고 제안한 부호화기로 상위 대역을 처리하는 대역폭 계층 구조의 광대역 음성 부호화기를 개발하였고, 주관적 음질을 측정하여 24kbps G.722.1보다 우수한 음질을 가지는 것을 확인하였다.

• 음성과학
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• 제12권4호
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• pp.17-29
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• 2005

Time-frequency representation reveals some useful information about instantaneous frequency, instantaneous bandwidth and boundary of each AM-FM component of a speech signal. In many cases, the instantaneous frequency of each component is not constant. The variability of instantaneous frequency causes degradation of resolution in time-frequency representation. This paper presents a method of adaptively adjusting the transform kernel for preventing degradation of resolution due to time-varying instantaneous frequency. The transform kernel is the form of frequency modulated function. The modulation function in the transform kernel is determined by the estimate of instantaneous frequency which is approximated by first order polynomial at each time instance. Also, the window function is modulated by the estimated instantaneous. frequency for mitigation of fringing. effect. In the proposed method, not only the transform kernel but also the shape and the length of. the window function are adaptively adjusted by the instantaneous frequency of a speech signal.