• Title/Summary/Keyword: Digital Speech coding

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A Study on the Low Noise Delta Codec System (저잡음 델타변조방식에 관한 연구)

  • 심수보
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.9 no.3
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    • pp.120-126
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    • 1984
  • In this paper, there is presented the novel encoder circuit design method in the realization of exponential adaption process on the delta modulation coding of speech signals. The digital implementation has been adapted for the illustration of above, especially using a rate multiplier end a double integration circuit. The use of a double integration of the local decoder included in the ADM encoder in prove the undesirable characteristics which the low switching speed of the ratemultiplier couses the SQNR to decreuse, and the SQNR of the decoding signal by above realization is relatively uniformed in wide range of signal levels. The validity of the above design is verified by laboratory experiments.

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Fixed-point Implementation of LPD Decoder in MPEG-D USAC (MPEG-D USAC : LPD 복호화기의 고정 소수점 알고리즘 구현)

  • Song, Eunwoo;Song, Jeongook;Kang, Hong-Goo
    • Proceedings of the Korean Society of Broadcast Engineers Conference
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    • 2012.07a
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    • pp.254-256
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    • 2012
  • 본 논문에서는 MPEG-D 오디오 서브그룹에서 진행 중인 Unified Speech and Audio Coding (USAC) 표준의 Linear Prediction Domain (LPD) 복호화기 모듈을 고정소수점 알고리즘으로 제안한다. USAC 부호화기는 두 개의 최신 음성-오디오 부호화기가 융합된 형태로, 음성 및 오디오 신호에 대하여 우수한 성능을 갖는 부호화기이다. USAC의 표준 완료와 본격적인 서비스화에 앞서서 USAC LPD 복호화기의 구조적인 특성을 분석하고, Digital Signal Processor (DSP)구현을 위한 LPD 복호화기의 고정소수점 알고리즘을 구축하는 동시에 모듈의 복잡도를 측정하고자 한다. 또한 고정소수점 알고리즘으로 구현된 LPD 복호화기와 기존의 부동소수점 복호화기의 성능을 비교하고, LPD 복호화기의 두 가지 부호화 모드에 따른 복잡도 이슈를 다루도록 한다.

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Performance Enhancement of Underwater Acoustic Communication System Using Hydrophone Transmit Array (하이드로폰 송신 어레이를 이용한 수중 음향 통신 시스템의 성능 향상)

  • 이외형;손윤준;김기만
    • The Journal of the Acoustical Society of Korea
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    • v.21 no.7
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    • pp.606-613
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    • 2002
  • In this paper we applied a transmit beamforming technique to the underwater acoustic communication system for high rate data transmission. A prototype transmit system was designed and implemented with the general purpose DSP processor and multiple digital-to-analog converters. The performances of the implemented system were evaluated by the experiment in water tank. In order to simplify the procedure the channel coding and equalizer were omitted. And the simplest OOK (On-Off Keying) technique in digital communication methods was applied. The experimental result shows that the transmission data rate is higher about 3 times in the case of 5 hydrophone transmitting may than 1 hydrophone transmitter at bit error rate 10/sup -2/. We verified that the maximum data rate was 400 bps for speech signal transmission in water tank.

MPEG-D USAC: Unified Speech and Audio Coding Technology (MPEG-D USAC: 통합 음성 오디오 부호화 기술)

  • Lee, Tae-Jin;Kang, Kyeong-Ok;Kim, Whan-Woo
    • The Journal of the Acoustical Society of Korea
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    • v.28 no.7
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    • pp.589-598
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    • 2009
  • As mobile devices become multi-functional, and converge into a single platform, there is a strong need for a codec that is able to provide consistent quality for speech and music content MPEG-D USAC standardization activities started at the 82nd MPEG meeting with a CfP and approved WD3 at the 88th MPEG meeting. MPEG-D USAC is converged technology of AMR-WB+ and HE-AAC V2. Specifically, USAC utilizes three core codecs (AAC ACELP and TCX) for low frequency regions, SBR for high frequency regions and the MPEG Surround tool for stereo information. USAC can provide consistent sound quality for both speech and music content and can be applied to various applications such as multi-media download to mobile device Digital radio Mobile TV and audio books.

A Study on the Realization of Echo Canceller in CDMA Mobile Communication Networks (CDMA 이동통신 망에서의 반향제거기 구현에 관한 연구)

  • 유태훈;박광철;이윤희;김기두
    • Journal of the Institute of Electronics Engineers of Korea TE
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    • v.37 no.5
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    • pp.36-47
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    • 2000
  • The CDMA digital cellular systems provide better voice Quality than analog systems, however there exists inherent delays due to speech coding and transmission processing, which brings echoes returned by the BSC and PSTN interface. In this paper, we show the performance improvement of a proposed echo canceller by real time implementation, where Block Update NLMS algorithm is applied into the TMS320C54X DSP. By applying the proposed method into the practical mobile phone, we verify that various types of echoes (LE, ESE, AE) may be removed more precisely. We also cope with echo path change resulting from change of delay length after taking VAD to find echo path delay.

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