• International journal of advanced smart convergence
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• v.11 no.1
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• pp.55-63
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• 2022

Although research on the leader's or speaker's voice has been continuously conducted, existing research has a single point of view. Sound analysis of voice characteristics has been studied from engineering perspectives, and leadership trait theory has been studied from a business perspective. Convergence studies on leader voice and member cognition are being attempted today. Convergence research on voice has a positive effect on refinement of voice analysis, diversification of voice use, and establishment of voice utilization strategy. This study explains the current flow of research on convergence between speaker's voice and listener's perception, and suggests a direction for the future development of voice fusion research. Furthermore, in connection with AI in the 4th industrial age, new attempts for voice research are sought. First, advances in AI focus on strategically generating the voices needed for individual situations. Second, the voice corrected in real time will support the leader and speaker to utilize the desired voice type. Third, voices through AI based on big data will affect the cognition, attitude and behavior of individual listeners who members, customers, and students in more diverse situations. The purpose and significance of this study is to suggest the way to research the leader's voice recognized by members, and to suggest a method that can be applied in various situations.

• Proceedings of the KIEE Conference
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• 2007.04a
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• pp.275-277
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• 2007

Recent activation of DMB has enabled various high quality video, audio and data services. And there are various user facilities functions using digital data transmission. One of the various user facilities functions is EPG(Electronic Program Guide). EPG supports schedule of programs on screen for. audiences. EPG is composed to time, title, channel, genre etc. Users can select a program what they want to browsing. Currently EPG services are displaying program schedule on screen visually and make users to input ke:ywords with keypads, remote control devices or touch screen etc. However, this approach could cause a serious restriction to some users like to drivers or visually handicapped persons. A standard for a voice EPG to T-DMB is proposed. This method must be transferred VoiceXML based EPG files from the transmitter to receivers. This approach has a problem to process a standardization because the transmitter and receivers should be modified. We proposed and implemented a voice EPG platform that generates the voice EPG files from T-DMB SI without transferring voice EPG file from the transmitter.

• Journal of Institute of Control, Robotics and Systems
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• v.22 no.10
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• pp.797-802
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• 2016

This paper proposes a noble voice recognition method based on an adaptive MFCC and deep learning for embedded systems. To enhance the recognition ratio of the proposed voice recognizer, ambient noise mixed into the voice signal has to be eliminated. However, noise filtering processes, which may damage voice data, diminishes the recognition ratio. In this paper, a filter has been designed for the frequency range within a voice signal, and imposed weights are used to reduce data deterioration. In addition, a deep learning algorithm, which does not require a database in the recognition algorithm, has been adapted for embedded systems, which inherently require small amounts of memory. The experimental results suggest that the proposed deep learning algorithm and HMM voice recognizer, utilizing the proposed adaptive MFCC algorithm, perform better than conventional MFCC algorithms in its recognition ratio within a noisy environment.

• Proceedings of the IEEK Conference
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• 2000.11a
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• pp.189-192
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• 2000

This paper proposes a fuzzy-based random access controller with a superimposed frame structure (F$^2$RAC) fur voice/data-integrated wireless networks. F$^2$RAC adopts mini-slot technique for reducing contention cost, and these mini-slots of which number may dynamically vary from one frame to the next as a function of the traffic load are further partitioned into two regions for access requests coming from voice and data traffic with their respective QoS requirements. And F$^2$RAC is designed to properly determine the access regions and permission probabilities for enhancing the data packet delay while ensuring the voice packet dropping probability constraint. It mainly consists of the estimator with Pseudo-Bayesian algorithm and fuzzy logic controller with Sugeno-type of fuzzy rules. Simulation results prove that F$^2$RAC can guarantee QoS requirement of voice and provide the highest throughput efficiency and the smallest data packet delay amongst the different alternatives including PRMA［1］, IPRMA［2］, and SIR［3］.

• The KIPS Transactions:PartD
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• v.15D no.2
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• pp.257-262
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• 2008

This paper describes the implementations of the application service based on embedded Linux; Personal Audio Recorder (PAR) which uses WiBro network for data communications and CDMA network for voice communications. At PAR, when PAR client starts voice recording on a dual-mode terminal, the CDMA voice data of caller and callee is transmitted to storage server located in the Internet through WiBro network. Then, PAR server stores voice data on storage server according to the call number and call time. In case of shortage of storage space on terminal, PAR makes user to store voice data. And, PAR can search a catalog of stored data on server and play the specific content.

• Journal of Korean Society of Industrial and Systems Engineering
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• v.29 no.4
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• pp.100-106
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• 2006

We analyze a voice/data integrated traffic model of the cellular mobile communication system with hybrid guard channels for voice and handoff calls. In a multi-service integrated wireless environment, quality of service guarantee is crucial for smooth transportation of real time information. Real time voice traffic requires a guaranteed upper bounded on both delay and packet error rate, whereas data traffic does not. Voice traffic has high transmission priority over data packets. Thus one of the important problems is the design of admission control schemes which can efficiently accommodate the differential quality of service requirements. In this paper, a hybrid guard channel scheme is considered in which arriving calls are assigned channels as long as the number of busy channels in the cell is below a predetermined first threshold. When the number of busy channels reaches the first threshold, new originating data calls are queued in the infinite data buffer. Then reaches second threshold, only handoff calls are assigned the remaining channels and new originating voice calls are blocked. We evaluate the system by a two-dimensional Markov chain approach and generating function method and obtain performance measures included blocking probability and forced termination probability.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.40 no.9
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• pp.373-383
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• 2003

In this paper we discuss the effect of the delay limit and the packet size related to the quality of service on a VoIP system using the Internet. We also provide a guideline to determining the optimal packet size of the voice data for a given delay limit. Empirical studies are done with two personal computers connected through the packet switched public IP network. The sender encodes the voice signal from the microphone to get PCM and ADPCM data and sends the data to the receiver using UDP packets. The receiver plays the reconstructed voice from the stream with lost and delayed packets. The quality of the reconstructed voice is evaluated offline by the MNB (Measuring Normal Block) method using the data acquired from the both sides. The result shows that under the delay limit of 100ms for 40Kbps, 32Kbps and l6Kbps of ADPCM data, the minimum packet size should be 300bytes, 400bytes and 600bytes respectively and the maximum packet size should be l200bytes commonly for the best quality of voice.

• Speech Sciences
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• v.10 no.4
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• pp.201-219
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• 2003

Background: The perceptual assessment is generally performed by the voice specialist. The objective evaluation is performed in a voice laboratory. Research in voice laboratories has generated a variety of different objective tests and parameters. The perceptual evaluation is one of the most controversial topics in voice research. Review of literature reveals a wide variety of rating scales and reliability data fluctuating from study to study. Unfortunately, there is no widely accepted valid method for classifying voice disorders and assessing outcome after voice treatment. Objectives: The goals of this research were to identify important objective acoustic parameters of vocal quality, and to establish an objective and quantitative correlate of the perceived vocal quality. Materials and Methods : We evaluated the voice analyzed data from 122 dysphonic patients and 20 normal volunteers. A computerized speech lab. 4300B(CSL) was used to carry out the analysis of each voice sample. Results: Three dysphonia severity indices(DSI) were created using discriminant analysis. DSI is based on the weighted combination of the following selected set of acoustic parameters: absolute jitter(Jita in us), smoothed pitch period perturbation (sPPQ in %), amplitude perturbation quotient(APQ in %), soft phonation index(SPI), average fundamental frequency(Fo in Hz), lowest fundamental frequency(Flo in Hz), and smoothed amplitude perturbation quotient(sAPQ in %). The DSI, being the discriminating rule calculated by the logistic regression, consists of three equation based on statistically significant acoustic parameters. Three DSI were created to reflects best the degree of hoarseness as expressed by G from the GRBAS scale. The more positive this DSI is for a patient, the worse the vocal quality. The more it is negative, the better it is. The effect of sex is included implicitly in the DSI-1 and DSI-2, so that a separate DSI-1 and DSI-2 for males and females need not be used. The DSI is objective because no perceptual input is required for its calculation. Conculsion : This research demonstrates that the voice function values calculated from three different multivariate objective dysphonia severity indices are significantly associated with subjective voice assessments. These multivariate objective dysphonia severity indices may be appropriate for use in clinical trials and outcomes research on treatment effectiveness for voice disorders.

• Journal of information and communication convergence engineering
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• v.5 no.2
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• pp.88-92
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• 2007

In this paper, an SS-RRA protocol that is based on Code Division Multiple Access is proposed and analyzed under the integrated voice and data traffic load. The backward logical channels consist of slotted time division frames with multiple spreading codes per slot. The protocol uses a reservation mechanism for the voice traffic, and a random access scheme for the data traffic. A discrete-time, discrete-state Markov chain is used to evaluate the performance. The numerical results show that the performance can be significantly improved by a few distinct spreading codes.

• Proceedings of the Korea Information Processing Society Conference
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• 2000.04a
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• pp.692-697
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• 2000

멀티미디어 개인 휴대 통신(MPCS)에서 다중 클래스호에 대한 효율적인 채널할당은 매우 중요하다고 할 수 있다. 본 논문에서는 Image/Voice/Data 호에 대하여 가변적 보호 채널을 할당하는 새로운 방식을 제안하였다. 이러한 방식은 3차원 상태 천이도로 모델링 되며 보호 채널의 크기를 가변적으로 조절함으로써 보다 융통성있는 서비스를 제공하게 되며, 또한 수학적 분석과 시뮬레이션을 통해 비교분석을 수행하였다.