• 한국정보처리학회:학술대회논문집
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• 한국정보처리학회 2009년도 추계학술발표대회
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• pp.243-244
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• 2009

So far, many researches have been conducted in the area of audio based context recognition. Nevertheless, most of them are based on existing feature extraction techniques derived from linear signal processing such as Fourier transform, wavelet transform, linear prediction... Meanwhile, environmental audio signal may potentially contains non-linear dynamic properties. Therefore, it is a big potential to utilize non-linear dynamic signal processing techniques in audio based context recognition.

• 대한전자공학회논문지
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• 제24권4호
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• pp.618-624
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• 1987

This paper describes the design and fabrication of a signal processing integrated circuit required for the recording and playback of VTR audio signal. The integrated circuit was designed using 8\ulcorner design rule and its chip size is 2.5x2.5mm\ulcorner It was fabricated using SST bipolar standard process technology. The measurement analysis of the fabricated circuit proves the satisfactory DC characteristics and its proper audio signal processing funcstion.

• ETRI Journal
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• 제22권1호
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• pp.20-29
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• 2000

In this paper, we present a design of video and audio single chip encoder/decoder for portable multimedia application. The single-chip called as video audio signal processor (VASP) consists of a video signal processing block and an audio single processing block. This chip has mixed hardware/software architecture to combine performance and flexibility. We designed the chip by partitioning between video and audio block. The video signal processing block was designed to implement hardware solution of pixel input/output, full pixel motion estimation, half pixel motion estimation, discrete cosine transform, quantization, run length coding, host interface, and 16 bits RISC type internal controller. The audio signal processing block is implemented with software solution using a 16 bits fixed point DSP. This chip contains 142,300 gates, 22 Kbits FIFO, 107 kbits SRAM, and 556 kbits ROM, and the chip size is $9.02mm{\times}9.06mm$ which is fabricated using 0.5 micron 3-layer metal CMOS technology.

• ETRI Journal
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• 제40권1호
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• pp.133-145
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• 2018

Robustness, payload, and imperceptibility of audio watermarking algorithms are contradictory design issues with high-level security of the watermark. In this study, the major issue in achieving high payload along with adequate robustness against challenging signal-processing attacks is addressed. Moreover, a security code has been strategically used for secure transmission of data, providing tamper detection at the receiver end. The high watermark payload in this work has been achieved by using the complementary features of third-level detailed coefficients of discrete wavelet transform where the human auditory system is not sensitive to alterations in the audio signal. To counter the watermark loss under challenging attacks at high payload, Daubechies wavelets that have an orthogonal property and provide smoother frequencies have been used, which can protect the data from loss under signal-processing attacks. Experimental results indicate that the proposed algorithm has demonstrated adequate robustness against signal processing attacks at 4,884.1 bps. Among the evaluators, 87% have rated the proposed algorithm to be remarkable in terms of transparency.

• 한국음향학회지
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• 제24권7호
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• pp.387-394
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• 2005

본 논문에서는 다채널 오디오 시스템의 스피커들을 직렬로 연결하기 위한 새로운 오디오 신호 전송 기법을 제시한다. 다채널 오디오 본체로부터의 아날로그 신호는 디지털 신호로 변환되고 신호 처리 과정을 거쳐서 직렬로 연결된 각 스피커에 전달된다. 여기서 신호 처리 과정은 오디오 신호의 특성을 고려한 데이터 압축과 전송을 위한 패킷 생성을 포함한다. 각 스피커는 전달된 패킷으로부터 해당하는 디지털 신호만을 검출하여 아날로그 신호로 다시 변환하여 음향을 재생한다. 제시된 모든 기능은 VHDL을 사용하여 모델링되었으며 FPGA 칩으로 구현하였고 실제 다채널 오디오 시스템에서 테스트하였다.

• Journal of Information Processing Systems
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• 제2권1호
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• pp.13-17
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• 2006

In this paper, we propose an audio extraction method to decrease the influence of the original signal by modifying the watermarking detection system proposed by P. Bassia et al. In the extraction of the watermark, we employ a simple mean filter to remove the influence of the original signal as a preprocessing of extraction and the repetitive insertion of the watermark. As the result of the experiment, for which we used about 20 kinds of actual audio data, we obtain a watermark detection rate of about 95% and a good performance even after the various signal processing attacks.

• 대한전자공학회논문지SP
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• 제41권6호
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• pp.271-278
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• 2004

본 논문에서는 디지털 신호처리를 이용하여 Classic, Hiphop, Jazz, Rock, Speech 등 5개의 오디오 장르를 자동적으로 분류하는 내용기반 오디오 장르 분류기를 제안하였다. 20초 분량의 질의 오디오로부터 23ms 크기의 Hamming window를 이동시켜 가며 Spectral Centroid, Rolloff, Flux 등 STFT 기반의 특징 계수들과 MFCC, LPC 등의 계수들을 구하여 총 54차에 해당하는 특징 벡터 열을 추출하였으며 분류 알고리즘으로는 k-NN, Gaussian, GMM 분류기를 사용하였다. 최적의 특징 벡터를 선별하는 알고리즘으로 총 54차의 특징벡터 중 가장 성능이 좋은 특징 계수들을 찾아 순차적으로 재배치하는 SFS(Sequential Forward Selection)방법을 사용하였고, 이를 이용하여 최적화 된 10차의 특징 벡터만을 선정해서 오디오 장르 분류에 사용하였다. SFS를 적용한 실험 결과 약 90% 가까운 분류 성공률을 보이고 있어 기존 연구에 비하여 약 10%∼20% 정도의 성능 향상을 꾀 할 수 있었다. 한편 실제 사용자들이 오디오 자동 장르 분류 시스템을 사용할 때 일어날 수 있는 상황을 가정하여 임의 구간에서 질의 데이터를 추출하여 실험을 수행하였으며 실험 결과 오디오 파일의 맨 앞과 맨 뒤 등 worst-case 질의를 제외하고는 약 80%대의 분류 성공률을 얻을 수 있었다.

• IEIE Transactions on Smart Processing and Computing
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• 제2권1호
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• pp.11-19
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• 2013

A video-zoom driven audio-zoom algorithm is proposed to provide audio zooming effects according to the degree of video-zoom. The proposed algorithm is designed based on a super-directive beamformer operating with a 4-channel microphone array in conjunction with a soft masking process that uses the phase differences between microphones. The audio-zoom processed signal is obtained by multiplying the audio gain derived from the video-zoom level by the masked signal. The proposed algorithm is then implemented on a portable digital imaging device with a clock speed of 600 MHz after different levels of optimization, such as algorithmic level, C-code and memory optimization. As a result, the processing time of the proposed audio-zoom algorithm occupies 14.6% or less of the clock speed of the device. The performance evaluation conducted in a semi-anechoic chamber shows that the signals from the front direction can be amplified by approximately 10 dB compared to the other directions.

• 제어로봇시스템학회논문지
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• 제13권4호
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• pp.385-391
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• 2007

A class D digital audio amplifier with small size, low cost, and high quality is positively necessary in the multimedia era. Since the digital audio amplifier is based on the PWM signal processing, it is improper to analyze the principle of signal generation using linear system theories. In this paper, a class D digital audio amplifier based ADSM (Advanced Delta-Sigma Modulation) is considered. We first model the digital audio amplifier and then explain the operation principle using variable structure control algorithm. Moreover, the ripple signal generated by the hysteresis in the comparator has a significant effect on the system performance. Thus, we present a method to find the magnitude and the frequency of the ripple signal using describing function. Finally, simulations and experiments are provided to show the validity of the proposed methods.

• 한국컴퓨터정보학회:학술대회논문집
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• 한국컴퓨터정보학회 2014년도 제49차 동계학술대회논문집 22권1호
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• pp.89-92
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• 2014

This paper presents a secure and blind adaptive audio watermarking algorithm based on Empirical Mode Decomposition (EMD). The audio signal is divided into frames and each one is decomposed adaptively, by EMD, into several Intrinsic Mode Functions (IMFs). The watermark and the synchronization codes are then embedded into the extrema of the last IMF. The experimental results show that the proposed method has good imperceptibility and robustness against signal processing attacks.