• Proceedings of the KSPS conference
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• 2003.05a
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• pp.141-144
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• 2003

In this paper, we compared formant frequency extraction algorithms with various conditions, and show their performances. The formant frequency is the resonance frequency which is decided by the vocal tract characteristics. It is related with phonemes, or characteristics of the physical condition of the vocal track. Since the speech signal is influenced by both the sound source and the vocal tract, it is difficult to calculate the exact formant frequencies. Many studies on the formant frequency extraction had been executed already Besides, any new formant frequency extraction algorithm is hardly found recently.

• Proceedings of the Korean Society for Emotion and Sensibility Conference
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• 2003.11a
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• pp.131-132
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• 2003

사람의 음성이 나오기까지 화자가 전달하고자 하는 생각이 언어학적 구조로 바뀌고 이 과정에서 생각을 나타내는 적절한 단어나 구가 선택된다. 또 특정언어의 문법규칙에 의해 어순을 배열하고, 전체 의미에서 중요한 면을 강조하기 위해 피치ⅰ), 억양이나 강세와 같은 특성들을 첨가하는 등의 처리 절차를 통하게 된다. 음성은 기본적으로 여기ⅱ) 성분과 성도ⅲ) 성분으로 구분할 수 있다. 성도는 인두강과 구강을 합쳐서 일컫는다. 따라서 입 모양을 어떻게 하느냐에 따라서도 같은 말이라도 명료성에 영향을 미치게 되고 이러한 특성은 자신감이 넘치고 외향적인 모습으로 비춰지게 된다. 본 논문에서는 입의 모양에 따른 음성의 특징과 발성습관을 통해서 나타나는 사람의 성격을 알아보았다.

• The Journal of the Acoustical Society of Korea
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• v.28 no.4
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• pp.383-391
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• 2009

In speech signal processing, speech signal corrupted by noise should be enhanced to improve quality. Usually noise estimation methods need flexibility for variable environment. Noise profile is renewed on silence region to avoid effects of speech properties. So we have to preprocess finding voice region before noise estimation. However, if received signal does not have silence region, we cannot apply that method. In this paper, we proposed SNR estimation method for continuous speech signal. The waveform which is stationary region of voiced speech is very correlated by pitch period. So we can estimate the SNR by correlation of near waveform after dividing a frame for each pitch. For unvoiced speech signal, vocal track characteristic is reflected by noise, so we can estimate SNR by using spectral distance between spectrum of received signal and estimated vocal track. Lastly, energy of speech signal is mostly distributed on voiced region, so we can estimate SNR by the ratio of voiced region energy to unvoiced.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.19 no.12
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• pp.569-573
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• 2018

Music is composed of the sound of many instruments. Among them, the sound of the human voice naturally stands out to us and immediately connects with the listener. However, A lot of different steps go into perfectly mixing a vocal, but I'm going to focus on the most important step, equalization. In this paper, starting with the concept and the type of EQ for the requirements associated with the EQ's work and will know about when and how to use subtractive EQ, additive EQ during the recording and mixing process. EQ is one if the most important tools for mixing, especially when dealing with vocals. The control that EQ's offer allows you work, boosting and cutting to fit the vocal perfectly into the mix. The key to get a professional sounding vocal every time is to always keep in mind what you're trying to achieve stylistically and for it, using reference track is very effective. In addition to EQing, there are a variety of complex working steps such as compression, reverb, chorus, delay, adjusted for the effects of the work and harmonies of backing vocals and that are also very important task. The work of EQing is the beginning of the mixing process, among other things, need to be a detailed work throughout the consideration of the above points to its importance is greater relationship.

• Proceedings of the IEEK Conference
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• 2000.09a
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• pp.787-790
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• 2000

This paper introduces a new type of representation of the LSPs as a promising alternative used for transmitting the LPC parameters. Major contribution in this paper is that the vocal track information embedded on the spectral envelope can be represented in terms of the reduced number of LSF compared tn the conventional. Hence, it provides a possibility that LPC parameters could be quantized at a reduced bit rate without causing any major spectral distortion. The simulation result illustrates the capability of the proposed LSPs representation as an efficient quantization method via a proper rejection of the redundant pairs of pole and zero along the unit circle.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.8 no.1
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• pp.213-219
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• 2004

This paper proposes speaker recognition algorithm which includes the pitch parameter for the peak and valley. The time-frequency hybrid method for pitch extraction is valuable in that it can improve resolution in the time domain and accuracy in the frequency domain at the same time. It makes reference pattern using membership function and performs vocal track recognition of common character using fuzzy pattern matching in order to include time variation width for non-linear utterance for proposed method, speaker recognition experiments are carried out using vowels and number sounds.

• Proceedings of the Acoustical Society of Korea Conference
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• spring
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• pp.171-174
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• 2002

음성신호는 성문특성으로 인해 고주파 특성이 약화되는 경향이 있다. 이를 보상하기 위해 Pre-emphasis filter를 사용한다. 수식으로 표현하면 y(n)=s(n)-As(n-1) 와 같이 차분방정식으로 나타낼 수 있다. 여기서 A값은 보통 0.9에서 1사이의 값을 주로 사용한다. 그러나 Pre-emphasis filter는 고주파 특성을 보상하는 과정에서 극점과 같이 영점도 왜곡된다. 본 논문에서는 음성특성에 따른 LSP(Line Spectrum Pairs) 분포특성을 이용하여 영점을 보존하고 vocoder 및 coding에 필연적인 고주파 특성 혹은 저주파 특성을 강조한다.

• The Journal of the Acoustical Society of Korea
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• v.16 no.2
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• pp.101-106
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• 1997

In this paper, sample selective RLS(SSRLS) method is proposed, which aims to eliminate the influence of pitch bias. Its basic concepts are as follows. First it extracts the open glottis interval by using the residual signals, then estimates the formant values from the selected speech samples excluding above open glottis interval. This method has some analogy with the SSLPS, the simulation is conducted upon the synthetic and real speech. From these results, we find more usefulness of the proposed method than the conventional ones.

• Journal of the Korea Society of Computer and Information
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• v.8 no.3
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• pp.73-79
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• 2003

This paper proposes speaker recognition algorithm which includes both the pitch detection and the fuzzified pattern matching. This study utilizes pitch pattern using a pitch and speech parameter uses binary spectrum. In this paper. makes reference pattern using fuzzy membership function in order to include time variation width for non-utterance time and performs vocal track recognition of common character using fuzzified pattern matching.

• Journal of the Korea Society of Computer and Information
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• v.7 no.3
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• pp.49-55
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• 2002

This paper proposes speaker recognition algorithm which includes both the pitch parameter and the fuzzy. This study proposes a pitch detection method for the peak and valley pitch detection function by means of comparing spectra which utilizes the transform characteristics between time and frequency. It measures the similarity to the original spectrum while arbitrarily varying the period in the time domain. It heavily weights the error due to the changing characteristics of the phonemes, while it is strong against noise. In this paper, makes reference pattern using membership function and performs vocal track recognition of common character using fuzzy pattern matching in odor to include time variation width for non-linear utterance time.