• Title/Summary/Keyword: Voice Compression

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Diplophonia in Mutational Falsetto : Acoustic Characteristics and Treatment -A Case Report- (이중음성을 보인 변성발성장애 환자 음성의 음향학적 특성 및 치험례 -증 례 보 고-)

  • Lee, Jae-Yol;Lee, Sung-Eun;Lee, Sung-Eun;Choi, Hong-Shik
    • Journal of the Korean Society of Laryngology, Phoniatrics and Logopedics
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    • v.15 no.1
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    • pp.47-51
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    • 2004
  • Normally, as a result of increased laryngeal growth, the male voice drops about one octave in pitch level during adolescence. Failure of the voice to drop in pitch is consider to be a clinically significant voice disorder - 'mutational dysphonia'. The aim of this article is to evaluate the changes brought about by voice therapy, using the analysis of the EGG measure from Lx Speech Studio program(Laryngograph Ltd, UK) as well as acoustic, and aerodynamic studies in 18-year-old mutational dysphonia patient. The results from the Lx Speech Studio program demonstrated bimodal distribution of DFx(Hz), DQx(%), QxFx and diplophonic characteristic. After voice therapy combined with manual compression method, the distribution of DFx, DQx, QxFx was changed uniform with a dramatic reduction of higher pitch level. In addition, this finding suggests the EGG measure helps to choice treatment options, monitor the efficacy of therapy, and estimate the prognosis of diseases.

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Implementation of Flight Data Storage System with Compression and Security (압축 및 보안 기능이 있는 비행데이터 저장 시스템 구현)

  • Cho, Seung-Hoon;Ha, Seok-Wun;Moon, Yong-Ho
    • IEMEK Journal of Embedded Systems and Applications
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    • v.7 no.3
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    • pp.157-162
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    • 2012
  • In this paper, we propose a flight data storing system for effective data processing. Since the flight data contains critical information and their sizes are vast, encryption and compression would be needed to manage the flight data in effect. And we implemented the flight data storing system using an embedded board with DSP based on DPCM compression and AES encryption. Especially, we applied the reordering technique to advance the security function. From the simulations for two type data of voice and avionics, we found the developed system is well performed.

Design and Realization of a Novel Header Compression Scheme for Ad Hoc Networks

  • Khalid, Shahrukh;Mahboob, Athar;Azim, Choudhry Fahad;Rehman, Aqeel Ur
    • ETRI Journal
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    • v.38 no.5
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    • pp.922-933
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    • 2016
  • IP header compression schemes offer a valuable measure for bandwidth preservation. Such schemes have been practically implemented in infrastructure-based IP networks for point-to-point links. However, minimal research and practical implementation efforts have been conducted in the direction of an IP header compression strategy that can meet the peculiar requirements of multi-hop ad hoc wireless networks. In this paper, we present a practically implemented multi-hop IP header compression scheme using the Robust Header Compression (ROHC) protocol suite. The scheme runs on a novel identifier (ID) based networking architecture, known as an ID-based ad hoc network (IDHOCNET). IDHOCNET additionally solves a number of bottlenecks of pure IP-based ad hoc networks that have emerged owing to IP address auto-configuration service, distributed naming and name resolution, and the role of an IP address as an identifier at the application layer. The proposed scheme was tested on a multi-hop test bed. The results show that the implemented scheme has better gain and requires only O (1) ROHC contexts.

Effects of Chest Compression Quality between Rescuer's Simplified Verbal-Order Method and Continued Verbal-Order Method during Cardiopulmonary Resuscitation (심폐소생술 시 구조자의 간소화된 구령방법과 연속된 구령방법 간의 가슴압박 질 효과)

  • Baek, Hong-Seok;Park, Sang-Sub
    • The Journal of the Korea Contents Association
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    • v.13 no.4
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    • pp.320-330
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    • 2013
  • The purpose of this study is to increase efficiency of CPR through comparing the chest-compression quality between rescuer's simplified verbal order method and the continued verbal order method by utilizing voice meter during CPR. Subjects were 89 people(45 people for the experimental group, 44 people for the control group) who completed the 15-week CPR curriculum as undergraduates for the department of Emergency Medical Technology in C Province and were carried out by being randomly extracted. The group division was set for the experimental group as the group with the simplified verbal order and for the control group as the group with the continued verbal order. The period of measurement was progressed primarily(November 10, November 28, 2011) and secondarily(September 3-September 4, 2012). An analysis was used SPSS WIN 12.0 program. As a result of research, as for the implementation of appropriate chest compression(time, %), the quality was higher(p<.05) in the experimental group(102.86 times, 67.79%) than the control group(85.31 times, 55.84%). As a result of research, the chest compression(time, %) in the experimental group(102.86 times, 67.79%) had the higher effect of chest compression quality(p<.05) than the control group(85.31 times, 55.84%). On the other hand, the operation of weak chest compression(time) was higher in control group(61.13 times) than experimental group(35.54 times). The proper chest compression was shown(p<.05) in men of the experimental group as for gender and in over 60kg of the experimental group as for weight.

A study on the Implementation Extended Concept of GTS in IEEE 802.15.4 (IEEE 802.15.4에서 GTS의 확장개념에 관한 연구)

  • Jeon, Dong-Keun
    • The Journal of the Korea institute of electronic communication sciences
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    • v.10 no.3
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    • pp.319-325
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    • 2015
  • Remarkable advances in wireless communication technology have enabled communications among people who are far away from each other. In recent, the needs of local area voice communication using a wireless system based on low-cost and simple hardware are rapidly rising. However, since these applications require that the multi-users communicate on the same wireless channel in a small area, the existing voice technologies are not suitable for directly applying to these applications. Therefore, in this paper I propose a novel idea enabling multi-user voice communication. In particular, as a short range wireless solution, I employ the IEEE 802.15.4 based on low power and low cost. However, since originally the standard is not developed for voice communication, we extend the original scheme to be suitable for the voice communication by utilizing the extended concept of GTS. The capacity and validity of the proposed scheme are evaluated through quantitative analysis in various voice compression rates.

A Study On Intelligent Robot Control Based On Voice Recognition For Smart FA (스마트 FA를 위한 음성인식 지능로봇제어에 관한 연구)

  • Sim, H.S.;Kim, M.S.;Choi, M.H.;Bae, H.Y.;Kim, H.J.;Kim, D.B.;Han, S.H.
    • Journal of the Korean Society of Industry Convergence
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    • v.21 no.2
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    • pp.87-93
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    • 2018
  • This Study Propose A New Approach To Impliment A Intelligent Robot Control Based on Voice Recognition For Smart Factory Automation Since human usually communicate each other by voices, it is very convenient if voice is used to command humanoid robots or the other type robot system. A lot of researches has been performed about voice recognition systems for this purpose. Hidden Markov Model is a robust statistical methodology for efficient voice recognition in noise environments. It has being tested in a wide range of applications. A prediction approach traditionally applied for the text compression and coding, Prediction by Partial Matching which is a finite-context statistical modeling technique and can predict the next characters based on the context, has shown a great potential in developing novel solutions to several language modeling problems in speech recognition. It was illustrated the reliability of voice recognition by experiments for humanoid robot with 26 joints as the purpose of application to the manufacturing process.

Playout Scheduling Method Based on Adaptive Jitter Estimation for Enhancing VoIP Speech Quality (VoIP 음질향상을 위한 적응적 지터추정 기반의 플레이아웃 스케줄링 방법)

  • Ryu, Sang-Hyeon;Kim, Hyoung-Gook
    • The Journal of the Acoustical Society of Korea
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    • v.33 no.2
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    • pp.133-138
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    • 2014
  • Packet arrival-delay variation, so-called 'jitter' is one of the main factors that degrade the quality of voice in mobile devices at the Voice over Internet Protocol (VoIP). To resolve this issue, a playout scheduling based on adaptive jitter estimation for enhancing VoIP speech quality is proposed. The proposed algorithm copes with the effect of transmission jitter by expanding or compressing each packet according to the predicted network delay and variations. Additionally, the active network jitter estimation incorporates rapid detection of delay spikes and reacts to changes in network conditions. The experimental results have shown that the proposed algorithm delivers high voice quality in unstable network environment.

Adaptive Video Watermarking Using Half-cell Motion Vector (반화소 움직임 벡터를 이용한 적응적 비디오워터마킹)

  • Shinn Brian-B.;Kim Min-Yeong;D Khongorzul;Lee In-Sung
    • Journal of the Korea Institute of Information and Communication Engineering
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    • v.10 no.7
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    • pp.1214-1221
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    • 2006
  • Header compression scheme is suggested as a solution to reduce the inefficient overhead of general packet stream data. Especially, it is shown that there are more overhead rate for real-time media stream links such as voice because of its short payload size, and it is possible to get higher bandwidth efficiency using the header compression scheme. There are two kinds of error recovery in header compression such as Periodic Header Refresh(PHR) and Header Request(HR) schemes. In this paper, we analyze the performance of these two compression recovery schemes, and some results such as the overhead rate, bandwidth gain and bandwidth efficiency(BE) are presented.

On a Speech Coding Algorithm for Low Cost Implementation of Voice Telegram System (보이스 전보 시스템 구현을 위한 저가형 음성파형 부호화 알고리즘)

  • 나덕수;민소연;배명진
    • The Journal of the Acoustical Society of Korea
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    • v.19 no.2
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    • pp.101-105
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    • 2000
  • A telegram has been used to transmit the emergency news or celebration message. So, it has been very important media in our life. Although the telegram processing is more and more convenient, on the other hand, the telegram service contains only text message. The voice telegram is that delivering user's voice with text message. So, the voice telegram can be delivered sender's emotions and feelings. However, since voice information contains lots of data, large memory size and high cost processor are needed to deliver itself. In this paper, we proposed a new speech waveform coding method that has low complexity and low cost implementation for the voice telegram system. First, we fixed one basic speech waveform per pitch period and measured the waveform similarity between basic and neighbor speech waveform. Second, if the similarity satisfied threshold values, we compress the neighbor speech waveform with pitch and magnitude value per pitch period and if not, we save speech waveform. When the compression is about 45%, we obtained about 4 point in MOS.

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Two-Channel Multiwavelet Transform and Pre/Post-Filtering for Image Compression (영상 데이터 압축을 위한 2-채널 멀티웨이브렛 변환과 전후처리 필터의 적용)

  • Heo, Ung;Choi, Jae-Ho
    • Journal of the Korea Computer Industry Society
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    • v.5 no.5
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    • pp.737-746
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    • 2004
  • Two-channel multiwavelet system is investigated for image compression application in this paper. Generally, multiwavelets are known for their superb capability of compressing non-stationary signals like voice. However, multivavelet system have a critical problem in processing and compressing image data due to mesh-grid visual artifacts. In our two-channel multiwavelet system we have investigated incorporation of pre and post filtering to the multiwavelet transform and compression system for alleviating those ingerent visual artifacts due to multiwavelet effect. In addition, to quantify the image data compression performance of proposed multiwavelet system, computer simulations have been performed using various image data. For bit allocation and quantization, the Lagrange multiplier technique considering data rate vs. distortion rate along with a nonlinear companding method are applied equallly to all systems considered, here. The simulation results have yielded 1 ~ 2 dB compression enhancement over the scalar savelet systems. If the more advanced compression methods like SPIHT and run-length channel coding were adopted for the proposed multiwavelet system, a much higher compression gain could be obtained.

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