• 대한전자공학회:학술대회논문집
• /
• 대한전자공학회 2002년도 ITC-CSCC -1
• /
• pp.478-481
• /
• 2002

In this paper, the zerotrees of DWT is proposed to be used in a speech based digital watermarking for digital images. Since in this research work the raw speech and its content are used as a watermark signal, in the watermarking scheme, the PCM coded speech signal is embedded into a sequence of images. The performance of the scheme is evaluated by the PSNR obtained from the watermarked images and the strength of attacks the embedded speech signal can survive. Moreover, since in this research work the contents contained in the speech is used to identify the specific information hidden in the embedded signal. The speech signal after being extracted from the watermarked images is played back to the listeners to determine whether its content is intelligible or not. The experimental results show impressive performance of the scheme implementing our proposed technique, judged by the higher robustness obtained form the embedded signal against various types of attack, including brightness/contrast enhancement, Twirling, highpass filtering and JPEG compression standard.

• 대한전자공학회:학술대회논문집
• /
• 대한전자공학회 2002년도 ITC-CSCC -1
• /
• pp.466-469
• /
• 2002

Using intelligent information contained within the speech to identify the specific hidden data in the watermarked multimedia data is considered to be an efficient method to achieve the speech digital watermarking. This paper presents the performance comparison between various types of speech codec in order to determine an appropriate one to be used in digital watermarking applications. In the experiments, the speech signal encoded by four different types of speech codec, namely CELP, GSM, SBC and G.723.1codecs is embedded into a grayscale image, and theirs performance in term of speech recognition are compared. The method for embedding the speech signal into the host data is borrowed from a watermarking method based on the zerotrees of wavelet packet coefficients. To evaluate efficiency of the speech codec used in watermarking applications, the speech signal after being extracted from the attacked watermarked image will be played back to the listeners, and then be justified whether its content is intelligible or not.

• 한국음향학회지
• /
• 제15권1호
• /
• pp.93-101
• /
• 1996

본 논문에서는 전화가입자에게 보다 향상된 여러 가지 음성 서비스를 제공하기 위한 음성 처리 시스템을 구현하였다. 음성 신호처리만을 수행하는 전용 보드를 개발하고 하나의 마스터 보드가 여러 장의 DSP(Digital Signal Processing) 보드를 제어하여 음성의 저장과 재생기능을 수행하는 시스템을 다중 보드 구성에 적합한 방식인 VME버스를 사용하여 하드웨어를 구성하였다. 마스터 보드로서는 CPU30 보드를 사용하였고 DSP 보드로는 음성 입출력을 위한 전용 하드웨어인 SPM(Signal Processing Module) 보드를 제작하여 시스템 성능 평가를 하였다.

• 대한전기학회:학술대회논문집
• /
• 대한전기학회 1987년도 전기.전자공학 학술대회 논문집(II)
• /
• pp.1583-1586
• /
• 1987

In the study for effective speech control we designed a personal computer system with A/D converter in which the speech signal is transformed by digital data displayed graphically on the moniter and with a D/A converter in which the digital data is transformed into speech signal which people can hear. We analyzed the character of the speech signal produced by the system. We designed the adaptive noise cancel algorithm so that noise and Interference are cancelled whenever the speech signal is recognized by the computer system. This is a basic system for artificial Intelligence.

• 전자공학회논문지B
• /
• 제30B권10호
• /
• pp.35-41
• /
• 1993

In this paper, a new speech enhancement method for speech signal corrupted by environmental noise is proposed. Two signals are obtained from the microphone and from the accelerometer attached to the neck, respectively. Since two signals are generated from same source signal, both signals are closely correlated. And environmental noise has no effect on the accelerometer signal. The speech enhancement system identifies the optimum linear system between two signals on the basis of the dependence between the signals. The enhanced speech can be obtained by filtering the noise-free accelerometer signal. Since the characteristcs of the speech signal and environmental noise are changing with time, adaptive filtering system has to be used for characterizing the time-varing system. Simulation results show 7dB enhancement with 0dB speech signal level relative to the white noise.

• 대한전자공학회논문지SP
• /
• 제42권3호
• /
• pp.121-128
• /
• 2005

음성신호를 대상으로 하는 연구 분야에서 신경회로망은 주로 음성인식 등의 카테고리 분류의 목적으로 사용되며 신호처리의 응용에도 유망하다. 따라서 본 논문에서는 신경회로망에 시간구조를 취한 시간지연 신경회로망을 이용하여 잡음이 중첩된 음성신호의 공간으로부터 잡음이 없는 음성신호의 공간으로 사상을 실행함으로써 잡음을 제거하는 것을 목적으로 한다. 본 논문은 푸리에 변환의 진폭성분을 복원하는 잡음제거의 알고리즘을 사용하여 백색잡음 및 유색잡음에 대해서 본 수법의 유효성을 확인한다.

• KSII Transactions on Internet and Information Systems (TIIS)
• /
• 제13권5호
• /
• pp.2588-2609
• /
• 2019

Speech homomorphic encryption has become one of the key components in secure speech storing in the public cloud computing. The major problem of speech homomorphic encryption is the huge data expansion of speech cipher-text. To address the issue, this paper presents a speech homomorphic encryption scheme with less data expansion, which is a probabilistic statistics and addition homomorphic cryptosystem. In the proposed scheme, the original digital speech with some random numbers selected is firstly grouped to form a series of speech matrix. Then, a proposed matrix encryption method is employed to encrypt that speech matrix. After that, mutual information in sample speech cipher-texts is reduced to limit the data expansion. Performance analysis and experimental results show that the proposed scheme is addition homomorphic, and it not only resists statistical analysis attacks but also eliminates some signal characteristics of original speech. In addition, comparing with Paillier homomorphic cryptosystem, the proposed scheme has less data expansion and lower computational complexity. Furthermore, the time consumption of the proposed scheme is almost the same on the smartphone and the PC. Thus, the proposed scheme is extremely suitable for secure speech storing in public cloud computing.

• 디지털융복합연구
• /
• 제11권7호
• /
• pp.209-214
• /
• 2013

음성 인식 시스템에서는 인식 성능 향상을 위한 방법으로 인간의 청취 능력을 인식 시스템에 접목하였으며 잡음 환경에서 음성 신호와 잡음을 분리하여 원하는 음성 신호만을 선택할 수 있도록 구성되었다. 하지만 실용적 측면에서 음성 인식 시스템의 성능 저하 요인으로 인식 환경 변화에 따른 잡음으로 인한 음성 검출이 정확하지 못하여 일어나는 것과 학습 모델이 일치하지 않는 것을 들 수 있다. 따라서 본 논문에서는 음성 인식 향상을 위해 감마톤을 이용하여 특징을 추출하고 음향 모델을 이용한 학습 모델을 제안하였다. 제안한 방법은 청각 장면 분석을 이용한 특징을 추출을 통해 인간의 청각 인지 능력을 반영하였으며 인식을 위한 학습 모델 과정에서 음향 모델을 이용하여 인식 성능을 향상시켰다. 성능 평가를 위해 잡음 환경의 -10dB, -5dB 신호에서 잡음 제거를 수행하여 SNR을 측정한 결과 3.12dB, 2.04dB의 성능이 향상됨을 확인하였다.

• ETRI Journal
• /
• 제35권4호
• /
• pp.697-705
• /
• 2013

Emerging trends in the area of digital very large scale integration (VLSI) signal processing can lead to a reduction in the cost of the cochlear implant. Digital signal processing algorithms are repetitively used in speech processors for filtering and encoding operations. The critical paths in these algorithms limit the performance of the speech processors. These algorithms must be transformed to accommodate processors designed to be high speed and have less area and low power. This can be realized by basing the design of the auditory filter banks for the processors on digital VLSI signal processing concepts. By applying a folding algorithm to the second-order digital gammatone filter (GTF), the number of multipliers is reduced from five to one and the number of adders is reduced from three to one, without changing the characteristics of the filter. Folded second-order filter sections are cascaded with three similar structures to realize the eighth-order digital GTF whose response is a close match to the human cochlea response. The silicon area is reduced from twenty to four multipliers and from twelve to four adders by using the folding architecture.

• 방송공학회논문지
• /
• 제23권1호
• /
• pp.86-92
• /
• 2018

본 논문에서는 디지털 음성 신호의 부호화 이력을 검출하는 방법을 제안한다. 음성 신호를 디지털 방식으로 전송 또는 저장할 때 데이터양을 줄이기 위해 부호화한다. 따라서 음성 신호 파형이 주어질 때, 해당 신호가 원본인지 부호화된 신호인지 판단하고, 만일 부호화 되었다면 부호화 횟수를 검출하는 부호화 이력 검출 과정이 필요하다. 본 논문에서는 12.2kbps 비트율의 AMR 부호화기에 대하여 원본, 단일 부호화, 이중 부호화 여부를 판단하는 부호화 이력 검출 방법을 제안한다. 제안한 방법은 입력 음성 신호에서 음성 고유의 특성 벡터를 추출하고, 해당 특성 벡터를 심층 신경망으로 모델링 하는 방법을 사용한다. 본 논문에서 제안하는 특성 벡터가 일반적인 스펙트로그램으로부터 추출한 특성 벡터보다 우수한 부호화 이력 검출 성능을 제공하는 것을 확인하였다.