• Proceedings of the Korea Information Processing Society Conference
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• 2009.11a
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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.

• Journal of the Korean Institute of Telematics and Electronics
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• v.24 no.4
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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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• v.22 no.1
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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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• v.40 no.1
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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.

• The Journal of the Acoustical Society of Korea
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• v.24 no.7
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• pp.387-394
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• 2005

In this paper, we propose a new transmission technique of audio signals for the serial connection of the speakers of multiple-channel audio systems. Analog audio signals from a multi-channel audio system are converted into digital signals with signal processing steps and transferred to each speaker through a serial line. The signal processing steps contain data compression and packet generation in association with audio signal characteristics. Each speaker gets its corresponding digital audio signals from the transmitted packets and converts the signals into analog audio signals to make sounds with the speaker All the proposed functions in this paper are modeled in VHDL. implemented with FPGA chips, and tested for actual multi-channel audio systems.

• Journal of Information Processing Systems
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• v.2 no.1
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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.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.41 no.6
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• pp.271-278
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• 2004

In this paper, we propose a content-based audio genre classification algorithm that automatically classifies the query audio into five genres such as Classic, Hiphop, Jazz, Rock, Speech using digital sign processing approach. From the 20 seconds query audio file, the audio signal is segmented into 23ms frame with non-overlapped hamming window and 54 dimensional feature vectors, including Spectral Centroid, Rolloff, Flux, LPC, MFCC, is extracted from each query audio. For the classification algorithm, k-NN, Gaussian, GMM classifier is used. In order to choose optimum features from the 54 dimension feature vectors, SFS(Sequential Forward Selection) method is applied to draw 10 dimension optimum features and these are used for the genre classification algorithm. From the experimental result, we can verify the superior performance of the proposed method that provides near 90% success rate for the genre classification which means 10%∼20% improvements over the previous methods. For the case of actual user system environment, feature vector is extracted from the random interval of the query audio and it shows overall 80% success rate except extreme cases of beginning and ending portion of the query audio file.

• IEIE Transactions on Smart Processing and Computing
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• v.2 no.1
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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.

• Journal of Institute of Control, Robotics and Systems
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• v.13 no.4
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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.

• Proceedings of the Korean Society of Computer Information Conference
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• 2014.01a
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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.