• 한국HCI학회:학술대회논문집
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• 한국HCI학회 2008년도 학술대회 1부
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• pp.230-235
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• 2008

본 논문에서는 ARM 프로세서로 설계된 VoWiFi 단말기에서 광대역 음성 서비스를 가능하게 하기 위한 방법으로 ITU-T 표준 코텍인 G.729.1을 실시간으로 구현하고 그 성능을 평가한다. 실시간 G.729.1 코덱 구현은 C 코드 최적화 및 코덱 알고리즘의 고속화를 근간으로 한다. 이렇게 최적화된 코덱의 성능은 VoWiFi 단말기내에서 ARM 프로세서가 요구하는CPU 동작 시간으로 평가된다. 실험 결과, ARM926EJ를 사용하여 최적화된 G.729.1 코덱이 실시간으로 동작함을 확인할 수 있으며, 기존의 G.729에 비해 넓은 대역폭의 음성 전송이 가능함을 보일 수 있다.

• 대한음성학회지:말소리
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• 제64호
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• pp.105-119
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• 2007

Noise reduction technologies are indispensable to achieve acceptable speech quality in VoIP systems. This paper proposes a Wiener filter optimized to the estimated SNR of noisy speech for the noise reduction in VoIP environments. The proposed noise canceller is applied as a pre-processor before speech encoding. The performance of the proposed method is evaluated by the PESQ in various noisy conditions. In this paper, the proposed algorithm is applied to G.711, G.723.1, and G.729A which are all VoIP speech codecs. The PESQ results show that the performance of our proposed noise reduction scheme outperforms those of the noise suppression in the IS-127 EVRC and the ETSI standard for the advanced distributed speech recognition front-end.

• 대한음성학회지:말소리
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• 제57호
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• pp.99-112
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• 2006

In this paper, we introduce two retrieval methods for photos with speech documents. We compare the pattern of speech query with those of speech documents recorded in digital cameras, and measure the similarities, and retrieve photos corresponding to the speech documents which have high similarity scores. As the first approach, a phoneme recognition scheme is used as the pre-processor for the pattern matching, and in the second one, the vector quantization (VQ) and the dynamic time warping (DTW) are applied to match the speech query with the documents in signal domain itself. Experimental results show that the performance of the first approach is highly dependent on that of phoneme recognition while the processing time is short. The second method provides a great improvement of performance. While the processing time is longer than that of the first method due to DTW, but we can reduce it by taking approximated methods.

• 대한음성학회지:말소리
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• 제65호
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• pp.143-152
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• 2008

This paper focuses on the real time implementation of an adaptive multi-rate (AMR) speech codec, that is a standard speech codec of IMT-2000, using the TMS320VC5510. The series of TMS320VC55x is a 16-bit fixed-point digital signal processor (DSP) having low power consumption for the use of mobile communications by Texas Instruments (TI) corporation. After we analyze the AMR algorithm and source code as well as the structure and I/O of 7MS320VC55x, we carry out optimizing the programs for real time implementation. The implemented AMR speech codec uses 55.2 kbyte for the program memory and 98.3 kbyte for the data memory, and it requires 709,878 clocks, i.e. about 3.5 ms, for processing a frame of 20 ms speech signal.

• 한국방송∙미디어공학회:학술대회논문집
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• 한국방송공학회 2011년도 하계학술대회
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• pp.85-88
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• 2011

This paper describes a single channel speech enhancement as the pre-processor of automatic speech recognition system. The improvements are based on using optimally modified log-spectra (OM-LSA) gain function with a non-causal a priori signal-to-noise ratio (SNR) estimation. Experimental results show that the proposed method gives better perceptual evaluation of speech quality score (PESQ) and lower log-spectral distance, and also better word accuracy. In the enhancement system, parameters was turned for automatic speech recognition.

• 대한전자공학회논문지
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• 제24권3호
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• pp.394-399
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• 1987

It is necessary for the speech samples to be stored in memory buffer before speech analyzers without a real time processor process them. In this paper, we propose an algorithm that uses the buffer efficiently, when the analog speech signal is converted to the digital samples by the analog to digital converter. In order to implement this method in real time, the buffer is divided into the starting buffer and the remaining buffer. Until a voiced speech is found, the converted samples are sequentially stored in the starting buffer, and then the buffer is shifted. When a voiced speech is found, the next samples are sequentally recorded in the remaining buffer.

• 대한의용생체공학회:의공학회지
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• 제28권5호
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• pp.694-702
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• 2007

The adaptation effect originating from the chemical synapse between auditory nerve and inner hair cell gives advantage in accurate representation of temporal cues of incoming speech such as speech onset. Thus it is expected that the modification of conventional speech processing strategies of cochlear implant(CI) by incorporating the adaptation effect will result in considerable improvement of speech perception performance such as consonant perception score. Our purpose in this paper was to evaluate our new CI speech processing strategy incorporating the adaptation effect by the observation of auditory nerve responses. By classifying the presence or absence of speech from the auditory nerve responses, i. e. spike trains, we could quantitatively compare speech onset detection performances of conventional and improved strategies. We could verify the effectiveness of the adaptation effect in improving the speech onset representation characteristics.

• 한국음향학회지
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• 제19권5호
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• pp.9-12
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• 2000

음성 다이얼링 시스템은 화자의 음성을 인식하여 원하는 전화번호로 자동으로 전화를 걸어주는 시스템으로 주로 이동 전화나 휴대형 통신 장비에 유용하게 사용된다. 개인 음성 다이얼링 시스템의 경우, 다이얼링에 사용되는 모든 구문은 사용자가 선택하고 사용자의 음성을 사용하여 학습되어 음성 인식을 위한 HMM을 생성한다. 이러한 시스템은 화자독립 시스템 보다 매우 적은 메모리 공간과 계산량으로 구현이 가능하다. 그러나 이러한 시스템은 학습시 각 단어당 2-3개의 음성만을 사용하므로 음성인식 시스템의 성능을 개선하기 위한 각 상태에서의 상태지속분포을 추정하기는 매우 어렵다. 따라서 본 논문에서는 성능개선을 위한 후처리기를 제안하였다. 전화선을 통하여 구성된 데이터베이스를 이용한 실험에서 제안된 후처리기가 인식 시스템의 성능을 향상시킴을 확인하였다.

• 대한의용생체공학회:학술대회논문집
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• 대한의용생체공학회 1998년도 추계학술대회
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• pp.161-162
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• 1998

In this paper, a speech processor which is the most important part of the cochlear implant is developed, to recover auditory ability for the sensorineural disorders who have damaged for their inner ear. This system consists of the analog and digital signal processing part, of which functions is the pre-processing and the main processing, respectively. The main processing is peformed in DSP processor (TMS320C31-40) by using S/W. Because the program is used in this system, it is possible to cope with the individual status of the patients, very easily.

• 대한의용생체공학회:학술대회논문집
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• 대한의용생체공학회 1992년도 추계학술대회
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• pp.69-70
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• 1992

We have designed the speech signal processor for cochlear prosthesis using quadrature mirror filter (QMF). And it is real-time implemented using DSP macros TMS 320C30.