• Title/Summary/Keyword: Voice Compression

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A Low Power Multi-Function Digital Audio SoC

  • Lim, Chae-Duck;Lee, Kyo-Sik
    • Proceedings of the IEEK Conference
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    • 2004.06b
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    • pp.399-402
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    • 2004
  • This paper presents a system-on-chip prototype implementing a full integration for a portable digital audio system. The chip is composed of a audio processor block to implements audio decoding and voice compression or decompression software, a system control block including 8-bit MCU core and Memory Management Unit (MMU) a low power 16-bit ${\Sigma}{\Delta}$ CODEC, two DC-to-BC converter, and a flash memory controller. In order to support other audio algorithms except Mask ROM type's fixed codes, a novel 16-bit fixed-point DSP core with the program-download architecture is proposed. Funker, an efficient power management technique such as task-based clock management is implemented to reduce power consumption for portable application. The proposed chip has been fabricated with a 4 metal 0.25um CMOS technology and the chip area is about 7.1 mm ${\times}$ 7.1mm with 100mW power dissipation at 2.5V power supply.

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Transmission of Image Information in Wireless System (무선 환경에서의 영상 정보의 전송)

  • Jeonh, Sang-Hoon;Lim, Joon-Hong
    • Proceedings of the KIEE Conference
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    • 2004.11c
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    • pp.268-270
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    • 2004
  • There are various researches on MPEG techniques. MPEG technique is used to digital TV(DTV) and internet image communication. Connection method between server and client is usually using wire. Applications may be expanded, if wireless technology is used between server and client system. In this paper, Bluetooth is used for connection method between server and client. Bluetooth offers fast and reliable transmissions of both voice and data over the globally available 2.4GHz ISM (Industrial, Scientific and Medical) band. One of the major application purposes of Bluetooth is the cable replacement for mobile and peripheral devices. Bluetooth has the advantage of small size, low power and low cost. It has the disadvantage of limited bandwidth and limited range. In order to transfer effectively image Information between server and client using Bluetooth, we apply MPEG-2 and MPEG-4 image compression techniques and the results are compared with each other.

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Thinning algorithm of hand-printed korean character using wavelet transform (웨이브렛 변환을 이용한 필기체 한글 문자의 세선화 알고리즘)

  • 길문호;유기형;박정호;최재호;곽훈성
    • Proceedings of the IEEK Conference
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    • 1998.06a
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    • pp.745-748
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    • 1998
  • Recently, image and voice processing part is using wavelet transform. We propose thining algorithm using wavelet tranform. Wavelet transform consists of low frequency and high frequency in the spatial and frequency domain. After the wavelet decomposition, more than 90 percents of energy are contained in lowest frequency band. Therefor, for images with large difference of gray value between foreground and background like character images, we can more accurately in the lowest frequency band. Lowest frequency band has wavelet transform significant coefficient(WTS) that is required for the thinning algorithm we proposed Paper [3][5][7][8] can not separate consonants and vowels of korean characters. Becuase korean characters have structural feature. This paper can separate consonants and vowels. Simulation executed low frequency image and data compression can reduce 1/4$^{n}$ with level n. we can redcue time complexity 3/8.

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A Study of Speech Coding for the Transmission on Network by the Wavelet Packets (Wavelet Packet을 이용한 Network 상의 음성 코드에 관한 연구)

  • Baek, Han-Wook;Chung, Chin-Hyun
    • Proceedings of the KIEE Conference
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    • 2000.07d
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    • pp.3028-3030
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    • 2000
  • In general. a speech coding is dedicated to the compression performance or the speech quality. But. the speech coding in this paper is focused on the performance of flexible transmission to the, network speed. For this. the subbanding coding is needed. which is used the wavelet packet concept in the signal analysis. The extraction of each frequency-band is difficult to general signal analysis methods, after coding each band, the reconstruction of these is also a difficult problem. But. with the wavelet packet concept(perfect reconstruction) and its fast computation algorithm. the extraction of each band and the reconstruction are more natural. Also, this paper describes a direct solution of the voice transmission on network and implement this algorithm at the TCP/IP network environment of PC.

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Thyroplasty for the Restoration of a Normal Voice (음성개선을 위한 갑상연골성형술)

  • 김기령;김광문;정명현;이원상;정승규
    • Proceedings of the KOR-BRONCHOESO Conference
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    • 1982.05a
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    • pp.10.1-10
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    • 1982
  • The use of phonosurgery in the recent development of laryngomicrosurgery has enabled the restoration of a normal voice in respect to functional laryngeal surgery which in Korea in the past limited to simple removal of benign laryngeal tumor such as laryngeal polyp or nodules and cordal injection of $Teflon^{{\circledR}}$ for the treatment of recurrent nerve paralysis under the vision of suspension laryngoscopy. Performance of phonosurgery for the treatment of cord paralysis, mutational dysphonia, vocal cord atrophy, hyperkinetic dysphonia and sulcus vocalis is a happy event in the view point of development of phonosurgery in Korea. In this aspect thyroplasty to change the position and physical characteristics of the cord outside the glottis instead of the direct handling of the vocal cord through direct endoscopy is popular. Among the 4 types of thyroplasty, classified by Insshiki(1974), type I thyroplasty(1ateral compression of vocal cord) and type IV thyroplasty(lengthening of vocal cord) were effective in the treatment of unilateral vocal cord paralysis. Advantages of this operation are the fine adjustment of the degree of lateral compression under local anesthesia according to the phonation of the patient during operation and avoidance of dyspnea and intralaryngeal hemorrhage due to the manipulation outside the internal perichondrium of the thyroid cartilage. We did 7 cases of thyroplasty for the treatment of unilateral vocal cord paralysis in the 7 months from September 1981 to March 1982. Before the operation aerodynamic study, psychoacoustical evaluation, stroboscopy and sound spectrographic analysis were done. Two months after the operation the above procedures were performed again. Results of preoperative and postoperative examination were compared and the following results were obtained. 1) In the aerodynamic study, maximum phonation time increased to 158% of the preoperative value and the phonation quotient and the mean flow rate decreased to 58% and 54% of preoperative values. 2) The degree of hoarseness improved in the psychoacoustical evaluation and the glottic chink during phonation was decreased in the stroboscopic examiantion. 3) In the sound spectrographic analysis, periodicity was much restored and noise distribution decreased especially in the high frequency area.

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Cervico-Mediastinal Lipoma with Horner's Syndrome -A case report- (호너 증후군이 유발된 경부-종격동 지방종 -1예 보고-)

  • 김응수
    • Journal of Chest Surgery
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    • v.36 no.6
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    • pp.448-450
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    • 2003
  • Lipoma is a circumscribed mesenchymal tumor originating from the adipose tissue. The lesion is usually small and asymptomatic. The most common location is in the neck region, however, lipoma can be found in the mediastinum in rare occasions. Although lipoma reach to the large proportions in the mediastinum, it rarely compresses the neurovascular structure. We present a case of a 58-year-old man, in which a hourglass-type cervicomediastinal lipoma produced Horner's syndrome with voice change. The man presented a swelling at the right side of his neck, ptosis and anhidrosis on the right side of his face, and right chest discomfort. After the removal of the mass, all the symptoms which had been provoked by compression, as well as Horner's syndrome and hoarseness, nearly disappeared.

Control of mobile robot system with wireless transmission of image information.

  • Jeong, Sang-Hoon;Kwak, Jae-Hyuk;Lim, Joon-Hong
    • 제어로봇시스템학회:학술대회논문집
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    • 2004.08a
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    • pp.908-911
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    • 2004
  • There are various researches on mobile robot systems. Connection method between server and client of mobile robot system is one of them. In the case of mobile robot system, when connection method between server and client is wireless than wire, applications may be expanded. Also in remote monitoring environment using mobile robot system, we are interested in an effective transmission of the image information between server and client. In this paper, Bluetooth is used for connection method between server and client. One of the major applications of Bluetooth is the cable replacement for mobile and peripheral devices. Using Bluetooth, we propose the control method of mobile robot system. Bluetooth offers fast and reliable transmissions of both voice and data over the globally available 2.4GHz ISM (Industrial, Scientific and Medical) band. It has the advantage of small size, low power and low cost. It has the disadvantage of limited range and limited bandwidth. Also in order to transfer effectively image information between remote site(server) and mobile robot system(client) using Bluetooth, we applied to MPEG-2 and MPEG-4 image compression techniques and the results are compared with each other.

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Adaptive Buffer Management Method for Quality of Service of Internet Telephony (인터넷폰의 QoS를 위한 적응적인 버퍼관리 방식)

  • 류태욱;이정훈;강성호;엄기환
    • Journal of the Korea Institute of Information and Communication Engineering
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    • v.6 no.3
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    • pp.386-392
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    • 2002
  • Internet telephony is an application that transmits voice data for conversation. Therefore it must provide high sound quality. However while audio packets are transferred through the network, they are affected by delay variations and jitters, which could result in poor sound quality of the receiving end does not have an appropriate jitter buffer to overcome network factors. This thesis introduces a buffer management algorithm that could be used to provide better sound quality for Internet phone terminals. This algorithm actively responds to both the compression algorithms that are used by the terminals, as well as to the received data to provide an improvement in sound quality. In order to verify the effectiveness of the proposed algorithm, we experimented in variance network settings. The results show that the proposed algorithm improves on the performance of the conventional buffer management algorithm.

Multimedia Data Security of Video Conferencing System (영상회의 시스템에서의 멀티미디어 데이터 보안)

  • 이원호;한군희
    • Proceedings of the Korea Contents Association Conference
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    • 2003.05a
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    • pp.231-236
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    • 2003
  • Video conferencing system it is various at internet and uses the reading is become accomplished. Research of like this portion synchronization of audio, the video compression technique and multimedia data, supports the video conference the research of the Mbone of the If multicast for being active, being become accomplished the multimedia service which is various an video from internet, the line speed of communication becomes high-speed anger and to follow leads is become accomplished. The video conference from opening elder brother dispersion internet network environment the problem against the image which is an image conference data and a voice security is serious and it raises its head. To sleep it presents the security method which from the video conference it follows in quality of multimedia data from the dissertation which it sees and it does.

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Adaptive Buffer Management Method for QoS of Internet Telephony (인터넷폰의 QoS를 위한 적응적인 버퍼관리 방식)

  • 류태욱;이현관;이용구;김주웅;엄기환
    • Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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    • 2002.05a
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    • pp.384-387
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    • 2002
  • Internet telephony is an application that transmits voice data for conversation. Therefore it must provide high sound quality. However while audio packets are transferred through the network, they are affected by delay variations and jitters, which could result in poor sound quality if the receiving end does not have an appropriate jitter buffer to overcome network factors. This thesis introduces a buffer management algorithm that could be used to provide better sound quality for Internet phone terminals. This algorithm actively responds to both the compression algorithms that are used by the terminals, as well as to the received data to provide an improvement in sound quality. In order to confirm the validity of the suggested algorithm, comparisons of the performance have been made between the existing buffer management algorithms and this new algorithm in various network settings.

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