• The Journal of Korean Institute of Communications and Information Sciences
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• v.11 no.3
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• pp.172-178
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• 1986

The recent interest in the design of filters is motivatied by the fact that such filter can be fully integrated using standard metal-oxide-semiconductor processing technology. This is due to replacing all the resistors in the active RC filter network by the switched capacitors. The voltage gain of a SC filter depends only on the rations of capacitance and these ratios can be obtained and maintained to high accuracy. Therefore, it is known that a switched capacitor is much better than a resistor in temperature and linearity characteristics. This paper proposed a programmable SC filter and proved the fact that ${omega}_0$ Q and G of this circuit can be controlled by digital signal. Experiments show that SC filter remains the low sensitivities but it can't avoid little influence of parasitic capacitance. As the transfer characteristic of the SC filter is varied with sampling frequency and resistor array, SC filtering technigue can be applied for digital processing, speech analysis and synthesis and so on.

• Journal of Digital Convergence
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• v.13 no.12
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• pp.187-192
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• 2015

If the software is developed to analyze the speech disorder, the application of various converged areas will be very high. This paper implements the user-friendly program based on CART(Classification and regression trees) analysis to distinguish between normal and pathological voices utilizing combination of the acoustical and HOS(Higher-order statistics) parameters. It means convergence between medical information and signal processing. Then the acoustical parameters are Jitter(%) and Shimmer(%). The proposed HOS parameters are means and variances of skewness(MOS and VOS) and kurtosis(MOK and VOK). Database consist of 53 normal and 173 pathological voices distributed by Kay Elemetrics. When the acoustical and proposed parameters together are used to generate the decision tree, the average accuracy is 83.11%. Finally, we developed a program with more user-friendly interface and frameworks.

• Journal of Biomedical Engineering Research
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• v.19 no.1
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• pp.59-68
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• 1998

Digital hearing aids offer many advantages over conventional analog hearing aids. With the advent of high speed digital signal processing chips, new digital techniques have been introduced to digital hearing aids. In addition, the evaluation of new ideas in hearing aids is necessarily accompanied by intensive subject-based clinical tests which requires much time and cost. In this paper, we present an objective method to evaluate and predict the performance of hearing aid systems without the help of such subject-based tests. In the hearing impairment simulation(HIS) algorithm, a sensorineural hearing impairment medel is established from auditory test data of the impaired subject being simulated. Also, the nonlinear behavior of the loudness recruitment is defined using hearing loss functions generated from the measurements. To transform the natural input sound into the impaired one, a frequency sampling filter is designed. The filter is continuously refreshed with the level-dependent frequency response function provided by the impairment model. To assess the performance, the HIS algorithm was implemented in real-time using a floating-point DSP. Signals processed with the real-time system were presented to normal subjects and their auditory data modified by the system was measured. The sensorineural hearing impairment was simulated and tested. The threshold of hearing and the speech discrimination tests exhibited the efficiency of the system in its use for the hearing impairment simulation. Using the HIS system we evaluated three typical hearing aid algorithms.

• Journal of Digital Contents Society
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• v.16 no.2
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• pp.291-298
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• 2015

In a fast-paced information technology environment, consumption of video content is changing from one-way television viewing to VOD (Video on Demand) playing anywhere, anytime, on any device. This video-watching trend gives additional importance to videos with fine-speed-control, in addition to the strength of the digital video signal. Currently, many video players provide a fine-speed-control function which can speed up the video to skip a boring part, or slow it down to focus on an exciting scene. The audio information is just as important as the visual information for understanding the content of the speed-controlled video. Thus, a number of algorithms for fine-speed-control video-playing technologies have been proposed to solve the pitch distortion in the audio-processing area. In this study, well-known techniques for prosodic modification of speech signals, WSOLA (Waveform-Similarity-Based Overlap-Add), have been applied to analyze users' needs for fine-speed-control video playing. By surveying the users' preferred speeds on categorized video content and analyzing the results, this paper proposes that various fine-speed adjustments are needed to accommodate users' preferred video consumption.

• Journal of the Korean Institute of Telematics and Electronics
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• v.22 no.5
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• pp.29-38
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• 1985

In this paper we have designed and implemented in real time a simple, efficient and flexible AOPCM cosec using a high speed digital processor, NEC 7720. For ADPCM system, we have used an instantaneous adaptive quantizer and a first-order fixed predictor. The software for NEC 7720 has been developed and it was found that the NEC 7720 was capable of performing the entire ADPCAt algorithm for 4 channels in real time as optimizing the program. Computer simulation has born made to investigate a computational accuracr of NEC 7720 and to de-termine necessary parameters for a ADPCM codec. Real telephone speech, RC-shaped Gaussian noise and 1004 Hz tone signal were used for simulation. In simulation, the parameters werc optimized from the computed SNR and the informal listening test. The developed software was tested in real time operation using a hardware emulator for NEC 7720. It took a maximum 23.25$\mu$s to encode one sample and 113.5$\mu$s, including all the necessary 1/0 operations, to encode 4 channels. In the case of decoding process, it took 24.75$\mu$s to decode one sample and 119.5$\mu$s to decode 4 channels.

• The Journal of the Acoustical Society of Korea
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• v.20 no.1
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• pp.56-61
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• 2001

This paper presents the real-time implementation of an AMR (Adaptive Multi Rate) vocoder which is included in the asynchronous International Mobile Telecommunication (IMT)-2000 mobile ASIC. The implemented AMR vocoder is a multi-rate coder with 8 modes operating at bit rates from 12.2kbps down to 4.75kbps. Not only the encoder and the decoder as basic functions of the vocoder are implemented, but VAD (Voice Activity Detection), SCR (Source Controlled Rate) operation and frame structuring blocks for the system interface are also implemented in this vocoder. The DSP for AMR vocoder implementation is a 16bit fixed-point DSP which is based on the TeakLite core and consists of memory block, serial interface block, register files for the parallel interface with CPU, and interrupt control logic. Through the implementation, we reduce the maximum operating complexity to 24MIPS by efficiently managing the memory structure. The AMR vocoder is verified throughout all the test vectors provided by 3GPP, and stable operation in the real-time testing board is also proved.