• The Journal of the Acoustical Society of Korea
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• v.39 no.6
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• pp.600-605
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• 2020

In this paper, we propose a sound event detection method using a multi-channel multi-scale neural networks for sound sensing home monitoring for the hearing impaired. In the proposed system, two channels with high signal quality are selected from several wireless microphone sensors in home. The three features (time difference of arrival, pitch range, and outputs obtained by applying multi-scale convolutional neural network to log mel spectrogram) extracted from the sensor signals are applied to a classifier based on a bidirectional gated recurrent neural network to further improve the performance of sound event detection. The detected sound event result is converted into text along with the sensor position of the selected channel and provided to the hearing impaired. The experimental results show that the sound event detection method of the proposed system is superior to the existing method and can effectively deliver sound information to the hearing impaired.

• Journal of the Institute of Electronics Engineers of Korea SC
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• v.43 no.2 s.308
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• pp.28-32
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• 2006

In this paper, NTSC TV Transmitter Modulo will be designed and produced which make possible playing the motion picture not only on TV but also on portable TV. NTSC TV Transmitter Module modulates signals that received Video and Audio signals from a mobile on NTSC TV CH4 mechanism. so it has an advantage of convenience that watching the motion picture of mobile on TV without any other cable through transmitting signals by wireless. But it has some demerits of long size antenna and noise sensitiveness. In the future, if some problems like a size of antenna distortion of signal and noise can be solved through continuous researching about Radio Frequency part, it is possible to play mobile motion pictures on the more media like a camcorder, DVD player and so on.

• Journal of the Institute of Electronics Engineers of Korea SC
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• v.43 no.3 s.309
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• pp.51-59
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• 2006

In this paper, we propose an excellent ultrasonic transceiver system based on a QPSK modulation technique for underwater communication. The transmitter sends a still image at the level of 187dB re $1{\mu}Pa/V@1m$ through a power amplifier by driving an ultrasonic sensor. The receiver performs digital conversion at the 100kHz sampling frequency, demodulation and decoding process for the image sent from the transmitter through the underwater communication. We have shown that the processed image at the receiver is almost the same as the orignal one. The maximum detection distance of the system proposed in this paper is approximately 1.17km. To cope with the difficulties of transmission loss, this paper proposes, implements and analyzes important parameters of sensors and circuits used in the system. Most of the underwater communication has focused on the transmission of audio signal, but this paper suggests an efficient underwater communication system for still image transmission.

• Journal of KIISE:Information Networking
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• v.28 no.4
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• pp.540-549
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• 2001

Internet telephony service is very cheap and very easy to introduce the value-added service than the POTS, but is difficult to the QoS of telephone service. The existing Internet typically offers 'best effort' services only, which do not make any commitment about delay, packet loss and jitter. This paper compensates the low quality of the speech for packet loss or delay using SOLA algorithm in Internet phone. SOLA algorithm is a popular technique for Time Scale Modification of speech and audio signal. In the proposed algorithm, the receiver expands the received packet under resonable threshold, and hence compensates the QoS of speech. From the simulation, this algorithm can conceals packet loss considerably, and is also improved the quality of the speech.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.8 no.4
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• pp.53-58
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• 2008

This paper designs sigma-delta D/A Converter with a high resolution and low power consumption. It reorganizes the input data along LJ, RJ, I2S mode and bit mode to the output data of A/D converter. The D/A converter decodes the original analog signal through HBF, Hold and 5th CIFB(Cascaded Integrators with distributed Feedback as well as distributed input coupling) sigma-delta modulation blocks. It uses repeatedly the addition operation in instead of the multiply operation for the chip area and the performance. Also, the half band filters of similar architecture composed the one block and it used the sample-hold block instead of the sinc filter. We supposed simple D/A Converter decreased in area. The filters of the block analyzed using the matlab tool. The top block designed using the top-down method by verilog language. The designed block is fabricated using Samsung 0.35um CMOS standard cell library. The chip area is 1500*1500um.

• The KIPS Transactions:PartB
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• v.8B no.5
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• pp.579-586
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• 2001

Digital watermarking is to embed imperceptible mark into image, video, audio, and text data to prevent the illegal copy of multimedia data. arbitrary modification, and also illegal sales of the copies without agreement of copyright ownership. In this paper, we study for the embedding and extraction of watermark key using wavelet in the luminance signal in order to implement the system to protect the copyright for image MPEG. First, the original image is analyzed into frequency domain by discrete wavelet transform. The RSA(Rivest, Shamir, Aldeman) public key of the coded target is RUN parameter of VLD(variable length coding). Because the high relationship among the adjacent RUN parameters effect the whole image, it prevents non-authorizer not to possess private key from behaving illegally. The Results show that the proposed method provides better moving picture and the distortion more key of insert than direct coded method on low-frequency domain based DCT.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.37 no.8A
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• pp.712-719
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• 2012

This paper presents Single-bit Third order Delta-Sigma Modulator, which can be applied to the Low speed High resolution ADC in Audio signal Process System. Whereas the Operational Amplifier in modulator consumed static power dissipation in operating, this modulator used Switching on/off techniques, which makes the Power dissipation of the modulator reduced. Also proposed modulator minimizes frequency characteristic variation by optimizing switch position. And this modulator chooses Single-bit type to guarantee stability. The designed ADC went through 0.35um CMOS n-well 1-poly 4-metal process to be a final product, and the final product has shown 17.1mW of power dissipation with 3.3V of Supply Voltage, 6.4MHz of conversion rate. And 84.3dB SNDR and 13.5bit ENOB with 20KHz of input frequency.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.34 no.8C
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• pp.822-829
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• 2009

This paper proposes a new method of improving the quality of High Efficiency Advanced Audio Coding (HE-AAC). HE-AAC encodes input source by allocating bits for each scalefactor bands appropriately according to human ear's psychoacoustic property. As a result, insufficient bits are assigned to the bands which have relatively low energy. This imbalance between different energy bands can cause decreasing of sound quality like musical noise. In the proposed system, a Linear Prediction (LP) module is combined with HE-AAC as a pre-processor to improve sound quality by even bits distribution. To apply accurate human being's psychoacoustic property, the psychoacoustic model uses Fast Fourier Transform (FFT) spectrum of original input signal to make masking threshold. In its implementation, masking threshold of psychoacoustic model is normalized using the LP spectral envelope in prior to quantization of the LP residual. Experimental result shows that, the proposed algorithm allocates bits appropriately for insufficient bits condition and improves the performance of HE-AAC.

• Journal of KIISE
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• v.45 no.3
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• pp.311-320
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• 2018

Recently, the use of digital audio and video as proof in criminal and all kinds of litigation is increasing, and scientific investigation using digital forensic technique is developing. With the development of computing and file editing technologies, anyone can simply manipulate video files, and the number of cases of manipulating digital data is increasing. As a result, the integrity of the evidence and the reliability of the evidence Is required. In this paper, we propose a technique for extracting the Electrical Network Frequency (ENF) through a grid of power grids according to the geographical environment for power supply, and then performing signal processing for peak detection using QIFFT. Through the detection algorithm using the standard deviation, it was confirmed that the video file was falsified with 73% accuracy and the forgery point was found.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.23 no.6
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• pp.1409-1418
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• 1998

The recent audio CODEC(Coding/Decoding) algorithms are complex of several coding techniques, and can be divided into DSP tasks, controller tasks and mixed tasks. The traditional DSP processor has been designed for fast processing of DSP tasks only, but not for controller and mixed tasks. This paper presents a new architecture that achieves high throughput on both controller and mixed tasks of such algorithms while maintaining high performance for DSP tasks. The proposed processor, YSP-3, operates four algorithms while maintaining high performance for DSP tasks. The proposed processor, YSP-3, operates functional units (Multiplier, two ALUs, Load/Store Unit) in parallel via 4-issue super-scalar instruction structure. The performance evaluation of YSP-3 has been done through the implementation of the several DSP algorithms and the part of the AC-3 decoding algorithms.