• 한국음향학회지
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• 제21권4호
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• pp.394-400
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• 2002

디지털 음성 통신에서 송신하는 음성 패킷의 전송 에러율을 알면 송신 채널 상황에 적합한 압축 동작을 통하여 전체 통신의 품질을 향상시킬 수 있다. 그러나 현재의 이동통신과 인터넷 통신에서는 음성 패킷의 전송 에러정보를 알려주는 프로토콜이 지원되지 않는다. 본 논문에서는 이를 해결하기 위하여 채널의 전송 에러 정보를 음성 패킷에 삽입하여 실시간으로 전달하는 방법을 제안한다. 제안하는 채널 에러 정보 삽입 방법은 ACELP (algebraic code-excited linear predictin) 코드벡터의 펄스 위치의 상관 관계를 이용하며, 이를 통하여 추가정보 삽입에 의한 음질 저하를 막고 오인식율을 줄일 수 있다. 다양한 음성 데이터를 이용하여 제안한 방법의 성능을 측정하였으며 음질의 저하가 거의 발생하지 않고 정보의 검출 능력과 오인식율에서 만족할 만한 성능을 가지는 것을 확인하였다.

• 한국음향학회지
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• 제26권1호
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• pp.8-15
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• 2007

음향 공동-유리창 연성계를 대상으로 도청음의 음성 명료도를 검토한다. MLS신호를 음원으로 유리창의 가속도와 속도 응답을 가속도계와 레이저 도플러 진동계를 사용하여 측정한다. 변조전송함수 (MTF)를 사용하여 공동-유리창 진동계의 음성전달특성을 규명한다. 변조전송함수에 근거하여 음성전송지수 (STI)를 구하고, 유리창 진동음의 음성명료도를 평가한다. 가속도음과 속도음의 음성명료도를 비교하고, 최종적으로 대화음의 명료도를 주관적 평가로 확인한다.

• 한국음향학회:학술대회논문집
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• 한국음향학회 2002년도 하계학술발표대회 논문집 제21권 1호
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• pp.15-18
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• 2002

Prosodic features of speech are known to play an important role in the transmission of linguistic information in human conversation. Their roles in the transmission of para- and non- linguistic information are even much more. In spite of their importance in human conversation, from engineering viewpoint, research focuses are mainly placed on segmental features, and not so much on prosodic features. With the aim of promoting research works on prosody, a research project 'Prosody and Speech Processing' is now going on. A rough sketch of the project is first given in the paper. Then, the paper introduces several prosody-related research works, which are going on in our laboratory. They include, corpus-based fundamental frequency contour generation, speech rate control for dialogue-like speech synthesis, analysis of prosodic features of emotional speech, reply speech generation in spoken dialogue systems, and language modeling with prosodic boundaries.

• 한국통신학회논문지
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• 제16권6호
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• pp.509-516
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• 1991

In this paper, we propose on objective evaluation model of telephone transmission qulity in order to estimate a satisfaction score regarding speech quality in a relephone network. As the degradantion factors of telephone transmission quality, this model takes into account transmission loss, noise, distortion, talker echo and sidetone. A performance index[PI] is introduced for five psychological factors affecting telephone speech qualty, and a Mean Opinion Score(MOS) is estimated from the sum of all Pis. The simulation results indicate theat the MOS obtained from the objective evaluation model is in good agreement with that of subjective evaluation.

• 대한전기학회:학술대회논문집
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• 대한전기학회 2000년도 하계학술대회 논문집 D
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• pp.3028-3030
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• 2000

In general. a speech coding is dedicated to the compression performance or the speech quality. But. the speech coding in this paper is focused on the performance of flexible transmission to the, network speed. For this. the subbanding coding is needed. which is used the wavelet packet concept in the signal analysis. The extraction of each frequency-band is difficult to general signal analysis methods, after coding each band, the reconstruction of these is also a difficult problem. But. with the wavelet packet concept(perfect reconstruction) and its fast computation algorithm. the extraction of each band and the reconstruction are more natural. Also, this paper describes a direct solution of the voice transmission on network and implement this algorithm at the TCP/IP network environment of PC.

• 음성과학
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• 제10권3호
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• pp.241-250
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• 2003

In this paper, we implemented the vocoder of variable rate by applying the SOLA-B algorithm to the G.729A to the TMS320C5416 in real-time. This method using the SOLA-B algorithm is that it is reduced the duration of the speech in encoding and is played at the speed of normal by extending the duration of the speech in decoding. But the method applied to the existed G.729A and SOLA-B algorithm is caused the loss of speech quality in G.729A which is not reflected about length variation of speech. Therefore the proposed method is encoded according as it is modified the structure of LSP quantization table about the length of speech is reduced by using the SOLA-B algorithm. The vocoder of variable rate by applying the G.729A and SOLA-B algorithm is represented the maximum complexity of 10.2MIPS about encoder and 2.8MIPS about decoder in 8kbps transmission rate. Also it is evaluated 17.3MIPS about encoder, 9.9MIPS about decoder in 6kbps and 18.5MIPS about encoder, 11.1MIPS about decoder in 4kbps according to the transmission rate. The used memory is about program ROM 9.7kwords, table ROM 4.69kwords, RAM 5.2kwords. The waveform of output is showed by the result of C simulator and Bit Exact. Also, the result of MOS test for evaluation of speech quality of the vocoder of variable rate which is implemented in real-time, it is estimated about 3.68 in 4kbps.

• ETRI Journal
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• 제32권3호
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• pp.395-405
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• 2010

A novel multiclassifier system (MCS) strategy is proposed and applied to a text-dependent speaker verification task. The presented scheme optimizes the linear combination of classifiers on an on-line basis. In contrast to ordinary MCS approaches, neither a priori distributions nor pre-tuned parameters are required. The idea is to improve the most accurate classifier by making use of the incremental information provided by the second classifier. The on-line multiclassifier optimization approach is applicable to any pattern recognition problem. The proposed method needs neither a priori distributions nor pre-estimated weights, and does not make use of any consideration about training/testing matching conditions. Results with Yoho database show that the presented approach can lead to reductions in equal error rate as high as 28%, when compared with the most accurate classifier, and 11% against a standard method for the optimization of linear combination of classifiers.

• 대한전자공학회논문지
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• 제19권6호
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• pp.84-90
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• 1982

본 논물에서는 HCDM(hybrid companding delta modulation)을 사용하여 음성을 부호화할 때, 음성의 간헐성을 이용하여 전송속도를 줄이거나 잡음에 대한 신호비(SQNR)을 증가시키는 연구를 하였다. 음성부분과 묵음(silence)부분을 식별하는 판별기를 이용하여 음성의 묵음부분을 검출하며, 이때 음성부분에 대해서는 HCDM 부호화를 행한다. 음성을 5msec 간격으로 검사하여, 그때 검출되는 묵음부분에 대해서는 그 구간이 묵음이라는 정도만을 전송하며, 수신단에서는 이 정보를 이용하여 묵음부불을 재생한다. 그런데 HCDM 부호기는 2진 신호를 일정한 속도로 또 동기적으로 전송하기 때문에, 버퍼 (buffer)를 사용해야 하며 또한 그것을 효율적으로 제어해야 한다. 음성을 부호화할 때, 묵음검출 기능을 이용하는 HCDM 부호기를 사용하면, 재래의 HCDM 보다 잡음에 대한 신호비를 6dB 만큼 증가시킬 수 있거나, 전송속도를 1/3가량 줄일 수 있다.In this paper we exploit the use of the intermittent property of speech to reduce the transmission rate or to increase signal-to-quantization noise ratio (SQNR) in coding speech by hybrid companding data modulation (HCDM). In this scheme we detect silence in speech by a speech/silence discriminator. HCDM coding is done only for speech portion. For silence that is detected in evert block of 5 ms, only the information indicating that the Since the HCDM coder transmits bina교 signal synchronously at a fixed rate, the use of a buffer and its efficient control is essential. By using the HCDM with silence detection in coding speech, we could improve SONR by as much as 6 dB over the conventional HCDM or reduce the transmission rate by one third of the HCDM rate.

• 한국소음진동공학회:학술대회논문집
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• 한국소음진동공학회 2007년도 춘계학술대회논문집
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• pp.649-654
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• 2007

Eavesdropping prevention performance is evaluated on a commercial window shaker, which is used to prevent a glass window from eavesdropping. Speech transmission index (STI) is introduced in order to estimate quantitatively the speech intelligibility of the sound detected on the glass window. Objective test by IEC standard using modulation transfer function (MTF) is performed to determine STI. Using Maximum Length Sequency (MLS) signal as a sound source, MTF is measured by accelerometers and laser doppler vibrometer. STI under different level of disturbing wave are compared to confirm the disturbing effect on the speech intelligibility.

• 대한청각학회지
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• 제18권3호
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• pp.105-111
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• 2014

Assistive listening devices (ALDs) refer to various types of amplification equipment designed to improve the communication of individuals with hard of hearing to enhance the accessibility to speech signal when individual hearing instruments are not sufficient. There are many types of ALDs to overcome a triangle of speech to noise ratio (SNR) problems, noise, distance, and reverberation. ALDs vary in their internal electronic mechanisms ranging from simple hard-wire microphone-amplifier units to more sophisticated broadcasting systems. They usually use microphones to capture an audio source and broadcast it wirelessly over a frequency modulation (FM), infra-red, induction loop, or other transmission techniques. The seven types of ALDs are introduced including hardwire devices, FM sound system, infra-red sound system, induction loop system, telephone listening devices, television, and alert/alarm system. Further development of digital wireless technology in hearing instruments will make possible direct communication with ALDs without any accessories in the near future. There are two technology solutions for digital wireless hearing instruments improving SNR and convenience. One is near-field magnetic induction combined with Bluetooth radio frequency (RF) transmission or proprietary RF transmission and the other is proprietary RF transmission alone. Recently launched digital wireless hearing aid applying this new technology can communicate from the hearing instrument to personal computer, phones, Wi-Fi, alert systems, and ALDs via iPhone, iPad, and iPod. However, it comes with its own iOS application offering a range of features but there is no option for Android users as of this moment.