• Proceedings of the Acoustical Society of Korea Conference
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• spring
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• pp.179-182
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• 2002

본 논문에서는 음성 압축기를 사용하는 디지털 이동통신 환경에서 한글 음성 인식기의 성능을 분석하기 위하여 다양한 표준 음성 압축기를 이용하여 음성 압축기의 구조, 전송률, 전송 채널의 에러율에 대한 성능을 측정하여 비교하였다. 동일한 구조의 음성 압축기에 대하여 전송률의 증가에 따라 음성 인식률이 증가하지만, 음성 압축기의 구조에 따라 동일 전송률에서도 많은 성능 차이가 발생하는 것을 확인하였다. 특히 IS-127 EVRC의 인식 성능이 매우 떨어지는 것을 알 수 있고, EVRC의 잡음 제거기와 가변 전송률에 의하여 음성 인식 성능이 저하되는 것을 확인하였다. 이를 통하여 청취 음질과 음성 인식 성능 사이의 상관 관계가 높지 않는 것을 알 수 있다. 모든 음성 압축기에 대하여 채널 에러율과 음성 인식기의 성능은 매우 밀접한 관계가 있음을 확인하였고, 평균적으로 채널 에러율 $1.0\%$에서 인식률이 $0.6\%$ 감소하고, 에러 $5.0\%$에서 인식률이 $1.8\%$ 감소한다.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.8 no.3
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• pp.563-568
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• 2007

We propose the computation reduction method of real root method that is used in the EVRC(Enhanced Variable Rate Codec) system. The real root method is that if polynomial equations have the real roots, we are able to find those and transform them into LSP. However, this method takes much time to compute, because the root searching is processed sequentially in frequency region. But, the important characteristic of LSP is that most of coefficients are occurred in specific frequency region. So, to reduce the computation time of real root, we used the met scale that is linear below 1kHz and logarithmic above. In order to compare real root method with proposed method, we measured the following two. First, we compared the position of transformed LSP(Line Spectrum Pairs) parameters in the proposed method with these of real root method. Second, we measured how long computation time is reduced. The experimental result is that the searching time was reduced by about 48% in average without the change of LSP parameters.

• Proceedings of the IEEK Conference
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• 1998.10a
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• pp.1203-1206
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• 1998

Considerable research in the last three decades has examined the problem of enhancement of speech degraded by additive background noise. We compare traditional methods such as spectral subtraction and Wiener filter, recently proposed psychoacoustic model based methods such as perceptual filter and noise suppression in EVRC in terms of performance and complexity.

• The Journal of the Acoustical Society of Korea
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• v.24 no.7
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• pp.416-422
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• 2005

Speech recognition in digital communication systems has very low performance due to the spectral distortion caused by speech codecs. In this paper, the spectral distortion by speech codecs is analyzed and a pre-processing method which compensates for the spectral distortion is proposed for performance enhancement of speech recognition. Three standard speech codecs. IS-127 EVRC. ITU G.729 CS-ACELP and IS-96 QCELP. are considered for algorithm development and evaluation, and a single method which can be applied commonly to all codecs is developed. The performance of the proposed method is evaluated for three codecs, and by using the speech features extracted from the compensated spectrum. the recognition rate is improved by the maximum of $15.6\%$ compared with that using the degraded speech features.

• Proceedings of the Korea Information Processing Society Conference
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• 2002.11b
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• pp.1357-1360
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• 2002

무선 데이터 통신이 하루가 다르게 증가하면서 무선망에서 핸드폰을 이용해 멀티미디어 데이터를 스트리밍하는 기술들이 연구되며 소개되어지고 있다. 이러한 기술을 연구하는데 있어서는 첫째로 데이터 전송 실험을 통한 무선 망에 대한 이해 및 특성 파악이 필요하고, 둘째로 이런 망 특성에 따른 적합한 스트리밍 기법 연구가 필요하다. 본 논문에서는 이러한 방법에 따라 망의 특성 파악에 관한 데이터를 기반으로 실험을 통해 EVRC 오디오 코텍을 이용한 핸드폰에서의 오디오 스트리밍에 있어 좀 더 나은 에러 대처 방법과 트랜스포트 프로토콜의 사용 기법을 제안하고 있다.

• The Journal of the Acoustical Society of Korea
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• v.20 no.7
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• pp.37-46
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• 2001

This paper proposes a new noise suppression method using the Wavelet transform analysis. The noise suppressor using the Wavelet transform shows the more effective advantages in a babble noise than one using the short-time Fourier transform. We designed a new channel structure based on spectral subtraction of Wavelet transform coefficients and used the Wavelet mask pattern with more higher time resolution in high frequency. It showed a good adaptation capability for babble noise with a non-stationary property. To evaluate the performance of proposed noise canceller, the informal subjective listening tests (Mos tests) were performed in background noise environments (car noise, street noise, babble noise) of mobile communication. The proposed noise suppression algorithm showed about MOS 0.2 performance improvements than the suppression algorithm of EVRC in informal listening tests. The noise reduction by the proposed method was shown in spectrogram of speech signal.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.1C
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• pp.81-89
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• 2003

For improving the performance of noise suppression in tactical communication environments, an enhanced MELP vocoder is suggested, in which an acoustic noise suppressor is integrated into the front end of the MELP algorithm, and an FEC code into the channel side of the MELP algorithm. The acoustic noise suppressor is the modified IS-127 EVRC noise suppressor which is adapted for the MELP vocoder. As for FEC, the turbo code, which consists of rate-113 encoding and BCJR-MAP decoding algorithm, is utilized. In acoustic noise environments, the lower the SNR becomes, the more the effects of noise suppression is increased. Moreover, The suggested system has greater noise suppression effects in stationary noise than in non-stationary noise, and shows its superiority by 0.24 in MOS test to the original MELP vocoder. When the interleave size is one MELP frame, BER 10-6 is accomplished at channel bit SNR 4.2 ㏈. The iteration of decoding at 3 times is suboptimal in its complexity vs. performance. Synthetic quality is realized as more than MOS 2.5 at channel bit SNR 2 ㏈ in subjective voice quality test, when the interleave size is one MELP frame and the iteration of decoding is more than 3 times.

• Journal of the Korea Institute of Information Security & Cryptology
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• v.22 no.1
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• pp.57-65
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• 2012

People leave voicemails or record phone conversations in their daily cell phone use. Sometimes important voice data is deleted by the user accidently, or purposely to cover up criminal activity. In these cases, deleted voice data must be able to be recovered for forensics, since the voice data can be used as evidence in a criminal case. Because cell phones store data that is easily fragmented in flash memory, voice data recovery is very difficult. However, if there are identifiable patterns for the deleted voice data, we can recover a significant amount of it by researching images of it. There are several types of voice data, such as QCP, AMR, MP4, etc.. This study researches the data recovery solutions for EVRC codec and AMR codec in QCP file, Qualcumm's voice data format in cell phone.

• Proceedings of the IEEK Conference
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• 2000.09a
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• pp.19-22
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• 2000

최근 이동전화의 사용이 급격히 확산됨에 따라 핸즈프리 단말기를 이용한 전화통신의 필요성이 대두되고 있다. 차량내 핸즈프리 통신상황의 경우 근거리에 위치한 스피커와 마이크로폰의 커플링에 의해 발생하는 음향반향과 차량내에 존재하는 배경잡음은 통화 품질을 크게 저하시킨다. 본 논문에서는 이동통신에 적합한 음향반향제거기와 잡음제거기의 결합시스템을 제안하고, 이를 고정 소수점 DSP를 이용하여 실시간 구현하였다. 실시간 구현을 위하여 음향반향제거기에는 NLMS 알고리즘에 의해 구동되는 제한된 차수의 적응반향제거기법을 사용하였다. 잔여반향 및 배경잡음제거를 위해 CDMA방식의 셀룰라 이동통신에 사용되는IS-127 EVRC음성 부호화기의 표준안에 포함된 잡음제거방식을 사용하였다. 제안된 시스템을 16 비트 고정소수점DSP인 OAK DSP Core를 이용하여 약 18.6MIPS의 연산량으로 실시간 구현되었다.

• MALSORI
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• no.64
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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.