• Title/Summary/Keyword: speech codec

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Transmission of Channel Error Information over Voice Packet (음성 패킷을 이용한 채널의 에러 정보 전달)

  • 박호종;차성호
    • The Journal of the Acoustical Society of Korea
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    • v.21 no.4
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    • pp.394-400
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    • 2002
  • In digital speech communications, the quality of service can be increased by speech coding scheme that is adaptive to the error rate of voice packet transmission. However, current communication protocol in cellular and internet communications does not provide the function that transmits the channel error information. To solute this problem, in this paper, new method for real-time transmission of channel error information is proposed, where channel error information is embedded in voice packet. The proposed method utilizes the pulse positions of codevector in ACELP speech codec, which results in little degradation in speech quality and low false alarm rate. The simulations with various speech data show that the proposed method meets the requirement in speech quality, detection rate, and false alarm rate.

Implementation of G.723.1 speech codec on OAK DSP Core based CSD17C00 (OAK DSP Core 기반 CSD17C00에서의 G. 723.1 Speech Codec 의 구현)

  • 성유나
    • Proceedings of the Acoustical Society of Korea Conference
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    • 1998.06c
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    • pp.151-154
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    • 1998
  • 이중 전송율(5.3 과 6.3kbit/s)을 제공하는 G.723.1 음성 코더는 공중망을 통한 H.324 POTS 영상 회의 규격의 음성 코더로 채택된 것으로, MPMLQ, ACELP 알고리즘에 근거한다. 본 논문에서는 Annex A를 포함한 G.723.1 음성 코더 알고리즘을 C&S Technology에서 개발한 음성 신호 처리를 위한 범용 DSP인 CSD17C00 칩을 이용하여 실시간 응용이 가능하도록 구현하였다. G.723.1 에 대한 양방향 평가가 Codec loopback을 통해 수행되었으며, ITU에서 제공한 테스트 절차에 따라 평가되었다. 또한, 본 논문에서 구현된 G.723.1 음성 코더는 27MIPS의 계산 속도를 갖으며, 프로그램 ROM의 크기는 8.85K Words이고, 10K 데이터 ROM과 4K 데이터 RAM을 필요로 하고 있다. 경쟁 제품과의 MOS 측정 음질 평가를 실시한 결과, CSD17C00에서의 음질 성능이 더 우수함을 입증 함으로써, 본 논문에서 보여준 CSD17C00을 기반으로 구현된 G.723.1 알고리즘의 실시간 구현기술의 타당성을 검증하게 되었다.

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Implementation of 2.4 kbps STC Speech Codec on the TMS320C6201 (TMS320C6201을 이용한 2.4 kbps STC 음성 부호화기의 실시간 구현)

  • 유승형;이승원;배건성
    • Proceedings of the IEEK Conference
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    • 2002.06d
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    • pp.167-170
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    • 2002
  • In this paper, we implement a 2.4 kbps STC speech codec using the TMS320C6201 DSP The main job for this work is twofold: one is to convert floating-point operation in the codec into fixed-point operation while maintaining the high resolution, and the other is to optimize the program to make it run in real time with memory size as small as possible. The implemented decoder uses 54.8 kbyte of program memory, 29.7 kbyte of data ROM and 55.2 kbyte of data RAM, respectively. It also uses about 45% of maximum computation capacity of TMS320C6201.

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Non-Intrusive Speech Quality Estimation of G.729 Codec using a Packet Loss Effect Model (G.729 코덱의 패킷 손실 영향 모델을 이용한 비 침입적 음질 예측 기법)

  • Lee, Min-Ki;Kang, Hong-Goo
    • The Journal of the Acoustical Society of Korea
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    • v.32 no.2
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    • pp.157-166
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    • 2013
  • This paper proposes a non-intrusive speech quality estimation method considering the effects of packet loss to perceptual quality. Packet loss is a major reason of quality degradation in a packet based speech communications network, whose effects are different according to the input speech characteristics or the performance of the embedded packet loss concealment (PLC) algorithm. For the quality estimation system that involves packet loss effects, we first observe the packet loss of G.729 codec which is one of narrowband codec in VoIP system. In order to quantify the lost packet affects, we design a classification algorithm only using speech parameters of G.729 decoder. Then, the degradation values of each class are iteratively selected that maximizes the correlation with the degradation PESQ-LQ scores, and total quality degradation is modeled by the weighted sum. From analyzing the correlation measures, we obtained correlation values of 0.8950 for the intrusive model and 0.8911 for the non-intrusive method.

Design of a Bitrate Scalable Speech Codec Based on G.723.1 (G.723.1 기반 비트율 scalable 음성 코덱 개발)

  • Kang Sangwon;Lee Kangeun;Park Dongwon;Lee Joonseok
    • The Journal of the Acoustical Society of Korea
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    • v.24 no.6
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    • pp.358-364
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    • 2005
  • In this Paper. we present a bitrate scalable speech codec which uses an ITU-T G.723.1 as the baseline coder and encodes the synthesis error signal in an enhancement coder. ITU-T P.862 (PESQ) is used to evaluate the Performance of the bitrate scalable coder. Experiments show that 6.7kbps enhancement layer based on G.723.1 5.3kbps produces the increase of 0.39 in MOS and 5.7kbps enhancement layer based on G.723.1 6.3kbps Produces the increase of 0.267 in MOS.

Implementation of Speaker Independent Speech Recognition System Using Independent Component Analysis based on DSP (독립성분분석을 이용한 DSP 기반의 화자 독립 음성 인식 시스템의 구현)

  • 김창근;박진영;박정원;이광석;허강인
    • Journal of the Korea Institute of Information and Communication Engineering
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    • v.8 no.2
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    • pp.359-364
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    • 2004
  • In this paper, we implemented real-time speaker undependent speech recognizer that is robust in noise environment using DSP(Digital Signal Processor). Implemented system is composed of TMS320C32 that is floating-point DSP of Texas Instrument Inc. and CODEC for real-time speech input. Speech feature parameter of the speech recognizer used robust feature parameter in noise environment that is transformed feature space of MFCC(met frequency cepstral coefficient) using ICA(Independent Component Analysis) on behalf of MFCC. In recognition result in noise environment, we hew that recognition performance of ICA feature parameter is superior than that of MFCC.

A LSF Quantizer for the Wideband Speech Using the Predictive VQ-Pyramid VQ (예측 VQ-Pyramid VQ를 이용한 광대역 음성용 LSF 양자학기 설계)

  • 이강은;이인성;강상원
    • The Journal of the Acoustical Society of Korea
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    • v.23 no.4
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    • pp.333-339
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    • 2004
  • This Paper proposes the vector quantizer-pyramid vector quantizer(VQ-PVQ) structure. Also both predictive structure and safety-net concept are combined into the VQ-PVQ to quantize the IPC parameter of wideband speech codec. The Performance is compared to the LPC vector quantizer used in the AMR-WB(ITU-T G.722.2). demonstrating reduction in both spectral distortion and encoding memory.

Design of the LSF Parameter Quantizer for the Wideband Speech Codec (광대역 음성 부호화기용 선 스펙트럼 주파수 계수 양자화기 설계)

  • 지상현;강상원;윤병식
    • The Journal of the Acoustical Society of Korea
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    • v.20 no.4
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    • pp.29-34
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    • 2001
  • In this paper, we designed an LSF coefficient quantizer of the wideband speech codec that can produce high quality speech service. For the efficient LSF coefficient quantizer, the interframe correlation was used. Also we separately quantized the LSF coefficients with high and low interframe correlation. Predictive pyramid vector quantizer (PVQ) was used for quantizing the LSF coefficients with high interframe correlation, and PVQ was used for quantizing the LSF coefficients with low interframe correlation. Experiments show that the proposed UF quantizer can quantize LSF information in 40 bits/frame, with an average spectral distortion (SD) of 1 dB and less than 3.87% frames having SD greater than 2 dB.

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Recognition of Continuous speech via 64kbit/s(7 kHz) Codec (64kbit/s(7 kHz) Codec을 경유한 연속음성의 인식)

  • 정현열
    • Proceedings of the Acoustical Society of Korea Conference
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    • 1993.06a
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    • pp.125-127
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    • 1993
  • 오디오 혹은 비디오화의, 방송 고품질전화 등의 음성신호의 전송을 위해 마련된 CCITT Recommendation G.722에 의거 Codec을 구성하고 이를 통과한 연속음성을 CMU의 불특정 화자 연속음성인식 시스템인 SPHINX에 입력하여 인식률을 조사 한 후 CODING전의 인식결과와 비교하였다. 이때 CODEC은 크게 네 부분(Trans Quarature Mirror Filter, Encoder, Decoder, Receive QMF)으로 구성하고 입력음성 데이터는 150화자에 의한 1018문장을 훈련용으로, 140문장을 테스트용으로 하였을 때의 단어 인식률을 인식률로 하였다. 또 이때 특징벡터로는 12차 Melcepstrum 계수를 사용하였다. 인식결과 코딩전(close talk Mic를 이용하여 직접입력)의 단어 인식률이 86.7%인데 비해 코딩후의 인식률은 85.6%로 나타나 약 1%의 인식률 저하를 가져와 코딩으로 인한 Error에 비해 비교적 양호한 결과를 얻을 수 있었다. 인식률 저하의 원인으로서는 코딩시의 BER(Bit Error Rate)에 의한 것으로 생각된다.

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Real-Time Implementation of a SBC Codec Using a NEC 7720 DSP (NEC 7720 DSP를 이용한 SBC codec의 실시간 구현)

  • Oh, Soo Hwan;Lee, Sang Uk
    • Journal of the Korean Institute of Telematics and Electronics
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    • v.23 no.4
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    • pp.429-438
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    • 1986
  • In this paper we have designed and implemented a real-time, full-duplex SBC (sub-band coding) codec at 16kbps using a high speed digital signal processor, NEC 7720. The SBC codec employs a QMF(quadrature mirror filter) filter bank based on the tree structures of two-band analysis-synthesis pairs to partition speech signal into 4 octabe bands. Computer simulation has been done to investigate the effect of fixed-point computation of the NEC 7720. Three different performance measures, the conventional signal-to-noise ratio, the informal listening test, and an LPC(linear predictive coding)distance measure, have been used in this simulation. The necessary parameters have been optimized through the simulation. The developed hardware and software have been tested in real-time operation using a hardware emulator.

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