• Title/Summary/Keyword: CELP

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The Full-Duplex and Real-Time Implementation of QCELP Vocoder (QCELP 음성부호화기의 양방향 실시간 구현)

  • 장석진
    • Proceedings of the Acoustical Society of Korea Conference
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    • 1995.06a
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    • pp.239-241
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    • 1995
  • 본 논문은 CELP 계열인 QCELP의 성능 평가와 설계된 효율적인 구조, 구현된 양방향 실시간 시스템에 대해 기술한다. 공인된 음성 샘플을 이용 SNR 과 분할 SNR 의 객관적 테스트를 수행하여 QCELP의 성능을 확인하였다. 본 실에서는 QCELP 알고리듬이 하나의 DSP 칩에 이식되고, 무선 환경하에서 실시간으로 음성 부호화 과정을 수행할 수 있도록 새로운 고조를 설계하였다. 다음에 본 실에서는 양 방향 통신의 하드웨어 플랫폼을 구성하여 설계된 QCELP 구조의 타당성을 입증하였다. 본 실에서 구현된 QCELP 음성부호화기는 현재 ETRI에서 개발된 디지틀 이동 통신 시스템인 CMS -2에서 사용되어 그 성능이 입증되었다.

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Design and Implementation of a Bluetooth LAN access system for VoIP phone (Bluetooth를 이용한 VOIP Phone 의 Wireless LAN Access System 개발)

  • 김정근;김영덕;장태규
    • Proceedings of the IEEK Conference
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    • 2002.06a
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    • pp.343-346
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    • 2002
  • This paper presents a Prototype system developed for a Bluetooth interfaced VoIP system. The VoIP phone is developed based on tile implementation of a CELP coder on the TI 16bit DSP Processor A PC interfaced with Bluetooth module is used to designing a access point system. Host controller protocol stack is implemented to realize gateway between the wireless and wired line networks. A server application program for user management and call processing, which is based on TCP/IP peer to peer connection, is implemented for tile evaluation of overall interface system.

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Phisical layer of ETRI wideband CDMA with 5 MHz bandwidth (5MHz 대역폭을 갖는 ETRI Wideband CDMA 시스템의 물리계층)

  • Bang, Seung-Chan;Park, Hyeong-Rae;Han, Yeong-Nam;Lim, Myeong-Seop;Lee, Heon;Han, Gi-Cheol;Park, Hang-Gu
    • Information and Communications Magazine
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    • v.13 no.4
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    • pp.56-63
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    • 1996
  • In this paper, the phisical layer of 4.3008 Mcps wideband CDMA which is proposed as the standard of personal communication service system is introduced. It is designed to fit 5 MHz bandwidth by using $28{\times}2^n$ Hadamard code. 13 kbps CELP vocoder is chosen as the main one and 32 kbps ADPCM can be applied. In the reverse link, the burst pilot scheme is proposed and produces the increase of capacity compared with the continuous pilot method. In order to maintain the service quality when the signaling data is increased, the information data and signaling data are time-multiplexed and making use of signaling activity brings the increase of capacity. QPSK data and QPSK spreading, variable frame size, and code pair assignment for high data rate are accomplished and then information data is transmited up to 64 kbps. It is expected that the proposed techiniques here are used in the FPLMTS.

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Low Rate Speech Coding Using the Harmonic Coding Combined with CELP Coding (하모닉 코딩과 CELP방법을 이용한 저 전송률 음성 부호화 방법)

  • 김종학;이인성
    • The Journal of the Acoustical Society of Korea
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    • v.19 no.3
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    • pp.26-34
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    • 2000
  • In this paper, we propose a 4kbps speech coder that combines the harmonic vector excitation coding with time-separated transition coding. The harmonic vector excitation coding uses the harmonic excitation coding in the voiced frame and uses the vector excitation coding with the structure of analysis-by-synthesis in the unvoiced frame, respectively. But two mode coding method is not effective for transition frame mixed in voiced and unvoiced signal and a new method beyond using unvoiced/voiced mode coding is needed. Thus, we designed a time-separated transition coding method for transition frame in which a voiced/unvoiced decision algorithm separates unvoiced and voiced duration in a frame, and harmonic-harmonic excitation coding and vector-harmonic excitation coding method is selectively used depending on the previous frame U/V decision. In the decoder, the voiced excitation signals are generated efficiently through the inverse FFT of harmonic magnitudes and the unvoiced excitation signals are made by the inverse vector quantization. The reconstructed speech signal are synthesized by the Overlap/Add method.

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A New Vocoder based on AMR 7.4Kbit/s Mode for Speaker Dependent System (화자 의존 환경의 AMR 7.4Kbit/s모드에 기반한 보코더)

  • Min, Byung-Jae;Park, Dong-Chul
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.33 no.9C
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    • pp.691-696
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    • 2008
  • A new vocoder of Code Excited Linear Predictive (CELP) based on Adaptive Multi Rate (AMR) 7.4kbit/s mode is proposed in this paper. The proposed vocoder achieves a better compression rate in an environment of Speaker Dependent Coding System (SDSC) and is efficiently used for systems, such as OGM(Outgoing message) and TTS(Text To Speech), which needs only one person's speech. In order to enhance the compression rate of a coder, a new Line Spectral Pairs(LSP) code-book is employed by using Centroid Neural Network (CNN) algorithm. In comparison with original(traditional) AMR 7.4 Kbit/s coder, the new coder shows 27% higher compression rate while preserving synthesized speech quality in terms of Mean Opinion Score(MOS).

Signal Enhancement of a Variable Rate Vocoder with a Hybrid domain SNR Estimator

  • Park, Hyung Woo
    • KSII Transactions on Internet and Information Systems (TIIS)
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    • v.13 no.2
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    • pp.962-977
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    • 2019
  • The human voice is a convenient method of information transfer between different objects such as between men, men and machine, between machines. The development of information and communication technology, the voice has been able to transfer farther than before. The way to communicate, it is to convert the voice to another form, transmit it, and then reconvert it back to sound. In such a communication process, a vocoder is a method of converting and re-converting a voice and sound. The CELP (Code-Excited Linear Prediction) type vocoder, one of the voice codecs, is adapted as a standard codec since it provides high quality sound even though its transmission speed is relatively low. The EVRC (Enhanced Variable Rate CODEC) and QCELP (Qualcomm Code-Excited Linear Prediction), variable bit rate vocoders, are used for mobile phones in 3G environment. For the real-time implementation of a vocoder, the reduction of sound quality is a typical problem. To improve the sound quality, that is important to know the size and shape of noise. In the existing sound quality improvement method, the voice activated is detected or used, or statistical methods are used by the large mount of data. However, there is a disadvantage in that no noise can be detected, when there is a continuous signal or when a change in noise is large.This paper focused on finding a better way to decrease the reduction of sound quality in lower bit transmission environments. Based on simulation results, this study proposed a preprocessor application that estimates the SNR (Signal to Noise Ratio) using the spectral SNR estimation method. The SNR estimation method adopted the IMBE (Improved Multi-Band Excitation) instead of using the SNR, which is a continuous speech signal. Finally, this application improves the quality of the vocoder by enhancing sound quality adaptively.

Performance Comparison on Speech Codecs for Digital Watermarking Applications

  • Mamongkol, Y.;Amornraksa, T.
    • Proceedings of the IEEK Conference
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    • 2002.07a
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    • pp.466-469
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    • 2002
  • Using intelligent information contained within the speech to identify the specific hidden data in the watermarked multimedia data is considered to be an efficient method to achieve the speech digital watermarking. This paper presents the performance comparison between various types of speech codec in order to determine an appropriate one to be used in digital watermarking applications. In the experiments, the speech signal encoded by four different types of speech codec, namely CELP, GSM, SBC and G.723.1codecs is embedded into a grayscale image, and theirs performance in term of speech recognition are compared. The method for embedding the speech signal into the host data is borrowed from a watermarking method based on the zerotrees of wavelet packet coefficients. To evaluate efficiency of the speech codec used in watermarking applications, the speech signal after being extracted from the attacked watermarked image will be played back to the listeners, and then be justified whether its content is intelligible or not.

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The Research of Reducing the Fixed Codebook Search Time of G.723.1 MP-MLQ (잡음 환경에서의 전송율 감소를 위한 G.723.1 VAD 성능개선에 관한 연구)

  • 김정진;박영호;배명진
    • Proceedings of the IEEK Conference
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    • 2000.06d
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    • pp.98-101
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    • 2000
  • On CELP type Vocoders G.723.1 6.3kbps/5.3kbps Dual Rate Speech Codec, which is developed for Internet Phone and videoconferencing, uses VAD(Voice Activity Detection)/CNG (Comfort Noise Generator) in order to reduce the bit rate in a silence period. In order to reduce the bit rate effectively in this paper, we first set the boundary condition of the energy threshold to prevent the consumption of unnecessary processing time, and use three decision rules to detect an active frame by energy, pitch gain and LSP distance. To evaluate the performance of the proposed algorithm we use silence-inserted speech data with 0, 5, 10, 20dB of SNR. As a result when SNR is over 5dB, the bit rate is reduced up to about 40% without speech degradation and the processing time is additionally decreased.

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Adaptive Multi-Rate(AMR) Speech Coding Algorithm (Adaptive Multi-Rate(AMR) 음성부호화 알고리즘)

  • 서정욱;배건성
    • Proceedings of the IEEK Conference
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    • 2000.06d
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    • pp.92-97
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    • 2000
  • An AMR(Adaptive Multi-Rate) speech coding algorithm has been adopted as a standard speech codec for IMT-2000. It is based on the algebraic CELP, and consists of eight speech coding modes having the bit rate from 4.75 kbit/s to 12.2 kbit/s. It also contains the VAD(Voice Activity Detector), SCR (Source Controlled Rate) operation, and error concealment scheme for robustness in a radio channel. The bit rate of AMR is changed on a frame basis depending on the channel condition. In this paper, we introduced AMR speech coding algorithm and performed the real-time implementation using TMS320C6201, i.e., a Texas Instrument's fixed-point DSP. With the ANSI C source code released from ETSI and 3GPP, we convert and optimize the program to make it run in real time using the C compiler and assembly language. It is verified that the decoded result of the implemented speech codec on the DSP is identical with the PC simulation result using ANSI C code for test sequences. Also, actual sound input/output test using microphone and speaker demonstrates its proper real-time operation without distortions or delays.

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On a Study of Measurement Method of Utterance Velocity for the Reduction of Transmission Rate in CELP Vocoder. (LSP 파라미터를 이용한 발성측정법)

  • 장경아;배명진
    • Proceedings of the IEEK Conference
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    • 2000.11d
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    • pp.199-202
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    • 2000
  • Speaking Rate has variety depends on the situation and habit of speakers. It has been many studied about speaking rate In speaker recognition. The study of speaking rate in speech recognition is one of considerable matter when It is recognized the speakers and it is measured by many speech data base and complicate estimation for accuracy. In this paper, conventional vocoder process the speech signal when encoding and transmitting without regard to speaking rate so in order to apply the speaking rate for vocoder It should be considered the simpler algorithm and less computation amount than the conventional method of speaking rate used In speech recognition. We proposed the speaking rate algorithm which is used the simple parameter with Line Spectrum Pair (LSP). The proposed peaking rate method is measured by the information of LSP in speech. We measured the variety rate of phenomenon about utterances which have different velocity, respectively. As a result, It has distinct variation rate of phenomenon between utterances uttered fast and slow and the rate is 42.8% higher in case of uttered fast than in case of uttered slow.

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