• Journal of Advanced Navigation Technology
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• v.11 no.1
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• pp.87-92
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• 2007

The conventional CELP(code excited linear prediction) type vocoder has no V/UV(voiced/unvoiced) classifier. So, the unvoiced speech is processed like the voiced speech. In this paper, to reduce the bit rate, we propose a new V/UV decision algorithm minimized error rate and preprocessing computation. This V/UV classifier use the LSP(line spectrum pair) parameter which is acquired spectrum analysis process in CELP vocoders. Applying this method to the 5.3kbps ACELP(algebraic code excited linear prediction) in the G.723.1, we can get the transmission bits rate reduction of 6% approximately without degradation of speech quality.

• Proceedings of the Acoustical Society of Korea Conference
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• 1998.08a
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• pp.142-147
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• 1998

Recently, 8 kb/s CS-ACELP coder of G.729 is atandardized by ITU-T SG15 and it has been reported that the speech quality of G729 is better than or equal to that of 32kb/s ADPCM. However G.729 is the fixed rate speech coder, and it does not consider the property of voice activity in mutual conversation. If we use the voice activity, we can reduce the average bit rate in half without any degradations of the speech quality. In this paper, we propose an efficient variable rate algorithm for G.729. The variable rate algorithm consists of two main subjects, the rate determination algorithm and algorithm, we combine the energy-thresholding method, the phonetic segmentation method by integration of various feature parameters obtained through the analysis procedure, and the variable hangover period method. Through the analysis of noise features, the 1 kb/s sub rate coder is designed for coding the background noise signal. So, we design the 4 kb/s sub rate coder for the unvoiced parts. The performance of the variable rate algorithm is evaluated by the comparison of speed quality and average bit rate with G.729. Subjective quality test is also done by MOS test. Conclusively, it is verified that the proposed variable rate CS-ACELP coder produced the same speech quality as G.729, at the average bit rate of 4.4 kb/s.

• The Journal of the Acoustical Society of Korea
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• v.28 no.4
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• pp.392-400
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• 2009

In this paper, we propose an effective speech reinforcement technique base on ITU-T G.729A CS-ACELP codec under the near-end background noise environments. In general, since the intelligibility of the far-end speech for the near-end listener is significantly reduced under near-end noise environments, we require a far-end speech reinforcement approach to avoid this phenomena. In contrast to the conventional speech reinforcement algorithm, we reinforce the excitation signal of the codec's parameters received from the far-end speech signal based on the G.729A speech codec under various background noise environments. Specifically, we first estimate the excitation signal of ambient noise at the near-end through the encoder of the G.729A speech codec, reinforcing the excitation signal of the far-end speech transmitted from the far-end. we specially propose a novel approach to directly reinforce the excitation signal of far-end speech signal based on the decoder of the G.729A. The performance of the proposed algorithm is evaluated by the CCR (Comparison Category Rating) test of the method for subjective determination of transmission quality in ITU-T P.800 under various noise environments and shows better performances compared with conventional SNR Recovery methods.

• The Journal of the Acoustical Society of Korea
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• v.16 no.3E
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• pp.44-49
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• 1997

In this paper, we propose an adaptive encoding method of fixed codebook in CELP coders and implement an adaptive fixed code exited linear prediction(AF-CELP) speech coder. AF-CELP exploits the fact that the fixed codebook contribution to speech signal is also periodic like the adaptive codebook (or pitch filter) contribution. By modeling the fixed code book with the pitch lag and the gain from the adaptive codebook, AF-CELP can be implemented at low bit rates as well as low complexity. Listening tests show that a 6.4 kbit/s AF-CELP has a comparable quality to the 8 kbit/s CS-ACELP in background noise conditions.

• The Journal of the Acoustical Society of Korea
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• v.18 no.7
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• pp.33-38
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• 1999

In this paper, we combine a generalized analysis-by-synthesis (AbS) structure and an algebraic excitation scheme to propose a new 4kb/s speech codec. This codec partly uses the structure of G.729. We design a line spectrum pair (LSP) quantizer, an adaptive codebook, and an excitation codebook to fit the 4 kb/s bit rate. The codec has a 25㎳ algorithmic delay, which corresponds to a 20㎳ frame size and a 5㎳ lookahead. At the bit rates below 4kb/s, most CELP speech codecs using the AbS principle have a drawback that results a rapid degradation of speech quality. To overcome this drawback we use the generalized AbS structure which is efficient for the low bit rate speech codec. LP coefficients are converted to LSP and quantized using a predictive 2-stage VQ. A low complexity algebraic codebook which uses shifting method is used for the fixed codebook excitation, and gains of the adaptive codebook and the fixed codebook are quantized using the VQ. To evaluate the performance of the proposed codec A-B preference tests are done with the fixed rate 8kb/s QCELP. As the result of the test, the performance of the codec is similar to that of the fixed rate 8kb/s QCELP.

• The Journal of the Acoustical Society of Korea
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• v.22 no.2
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• pp.129-134
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• 2003

In this paper, we efficiently implement the AMR speech coder by reducing the complexity of algebraic codebook search. To reduce the computational complexity of the algebraic codebook search, we propose a fast algebraic codebook search method that improves conventional depth first tree search method used in AMR speech coder algorithm. The proposed method reduces the search complexity by pruning the trees which are less possible to be selected as an optimum excitation. This method needs no additional computation for selecting the trees to be pruned and reduces the computational complexity considerably compared to the original depth first tree search method with slightly degradation or speech qualify. Applying our method to the implementation or AMR speech coder with 12.2 kbps mode by using the TeakLite DSP, we reduce the search complexity about 40% compared to the conventional method.

• Journal of Broadcast Engineering
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• v.16 no.5
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• pp.693-704
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• 2011

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 contents. MPEG-D USAC standardization activities started at the 82nd MPEG meeting with a CfP and approved Study on DIS at the 96th 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, the MPEG Surround for stereo information, and window transition technology for smoothing transition between various core coder. USAC can provide consistent sound quality for both speech and music contents and can be applied to various applications such as multi-media download to mobile devices, digital radio, mobile TV and audio books.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.9C
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• pp.1337-1344
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• 2004

This paper deals with analysis and real-time Implementation of a wide band adaptive multirate speech codec (AMR-WB) using a fixed-point DSP of TI's TMS320C6201. In the AMR-WB codec, input speech is divided into two frequency bands, lower and upper bands, and processed independently. The lower band signal is encoded based on the ACELP algorithm and the upper band signal is processed using the random excitation with a linear prediction synthesis filter. The implemented AMR-WB system used 218 kbytes of program memory and 92 kbytes of data memory. And its proper operation was confirmed by comparing a decoded speech signal sample-by-sample with that of PC-based simulation. Maximum required time of 5 75 ms for processing a frame of 20 ms of speech validates real-time operation of the Implemented system.

• 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.

• Proceedings of the IEEK Conference
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• 2001.06d
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• pp.273-276
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• 2001

CELP 부호화기는 선형 예측 합성에 의한 분석 부호화의 원칙에 기본을 두고 있다. 그리고 음성 신호의 스펙트럼을 LPC 분석을 통해 부호화하는데 고정 윈도우를 사용하여 부호화한다. 그러나 음성신호는 화자의 발성속도에 따라 파형의 변화가 시간적으로 빠르게 변화하기도 하고, 반대로 유사한 파형이 일정시간 유지되기도 한다. 따라서 윈도우의 크기를 발성속도에 맞추어 분석한다면 보다 효율적인 부호화를 할 수 있다. 본 논문에서는 발성속도에 따라 전송률을 달리 적용하는 방법을 제안한다. 발성속도의 측정은 스펙트럼 변화도를 이용하여 측정하였고, 발성속도가 빠를 때는 프레임 크기를 줄여 시간적으로 빠르게 변화하는 신호에 적응적으로 분석하고 대신 파라미터 표현에 비트를 줄인다. 반대로 발성속도가 느릴 때는 프레임 크기를 키우고 파라미터 표현에 비트를 더 할당한다. 제안한 방법을 실험하기 위해 G.723.1 5.3kbps ACELP 부호화기를 이용하였다 음질의 열하 없이 평균 16.34% 전송률 감소효과를 얻을 수 있었다.