• The Journal of the Acoustical Society of Korea
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• v.23 no.6
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• pp.481-487
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• 2004

In this paper. the structure of bandwidth scalable wideband speech codec and its high-band codec are proposed. In the high-band codec. the signal is divided into frequency bands. and each band is quantized in DCT domain. The DCT coefficients are splitted into magnitude and sign, and each is quantized independently by a specialized method based on its characteristics. In addition. the quantized gain parameter in the low-band codec is utilized in the high-band codec for an enhanced performance. The bandwidth scalable wideband speech codec using G.729E for low-band and the proposed codec for high-band is developed, and it is confirmed that the proposed codec has better subjective performance than 24kbps G.722.1.

• The Journal of the Acoustical Society of Korea
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• v.24 no.7
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• pp.395-401
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• 2005

In this paper, the high-band codec for bandwidth scalable wideband speech codec is proposed. The wideband input speech signal is separated into low-band signal and high-band signal, and the low-band signal is encoded by the standard narrow-band speech codec and the high-band signal is encoded by the proposed codec. In the high-band codec. the signal is transformed into frequency domain by MLT on a subframe basis, and MLT coefficients are splitted into magnitude and sign for quantization. The magnitudes of MLT coefficients are arranged into several time-frequency bands and each band is quantized in 2D-DCT domain, where the low-band information is utilized for better performance. The sign of MLT coefficient is quantized based on a priority selection process with the weighting measurement. The objective and subjective performance of wideband speech codec including the proposed high-band codec is measured, and it is confirmed that the proposed codec has better performance than 32kbps G.722.1.

• The Journal of the Acoustical Society of Korea
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• v.22 no.4
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• pp.308-314
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• 2003

Wideband speech communications with 7㎑ bandwidth can provide high-quality speech services that are almost impossible with current narrow-band speech communications with 3.4 ㎑ bandwidth, and AMR wideband codec was recently developed for these services. The performance of AMR wideband codec is excellent due to its wideband information and partially to ACELP structure, but it requires high computational complexity especially in codebook search. In this paper, to solve this problem, an efficient codebook search method for AMR wideband codec is proposed. The proposed method first determines the coarse initial codevector, then improves the performance of codevector by replacing a poor pulse in codevector with better one iteratively. Simulations show that AMR wideband codec with proposed codebook search method has higher performance with much less computational cost than conventional AMR wideband codec.

• ETRI Journal
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• v.27 no.2
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• pp.231-234
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• 2005

This letter proposes a new embedded speech coding structure based on the Adaptive Multi-Rate Wideband (AMR-WB) standard codec. The proposed coding scheme consists of three different bitrates where the two lower bitrates are embedded into the highest one. The embedded bitstream was achieved by modifying the algebraic codebook search procedure adopted for the AMR-WB codec. The proposed method provides the advantage of scalability due to the embedded bitstream, while it inevitably requires some additional computational complexity for obtaining two different code vectors of the higher bitrate modes. Compared to the AMR-WB codec, the embedded coder shows improved speech qualities for two higher bitrate modes with a slightly increased bitrate caused by the decreased coding efficiency of the algebraic codebook.

• The Journal of the Acoustical Society of Korea
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• v.21 no.1
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• pp.5-10
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• 2002

This paper proposes new 16 kbps wideband speech codec with bandwidth of 7 kHz. The proposed codec decomposes the input speech signal into low-band and high-band signals using QMF (Quadrature Mirror Filter), then AMR (Adaptive Multi Rate) speech codec processes the low-band signal and new transform-domain codec based on G.722.1 wideband cosec compresses the high-band signal. The proposed codec allocates different number of bits to each band in an adaptive way according to the property of input signal, which provides better performance than the codec with the fixed bit allocation scheme. In addition, the proposed cosec processes high-band signal using wavelet transform for better performance. The performance of proposed codec is measured in a subjective method. and the simulations with various speech data show that the proposed coders has better performance than G.722 48 kbps SB-ADPCM.

• Broadcasting and Media Magazine
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• v.19 no.1
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• pp.48-55
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• 2014

One of the recent noticeable evolutions in mobile communication systems is that wideband-codec is deployed rapidly in VoLTE (Voice over Long Term Evolution) service or HD voice. This paper is concerned with next generation HD voice or VoLTE service that is coined to describe high quality communication with super-wideband voice codec. 3GPP EVS (Enhanced Voice Service) Codec is being standardized to develop the super-wideband voice codec. This paper deals with the codec design constraints, performance requirements, the status of standardization, and finally perspective on VoLTE service in future.

• ETRI Journal
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• v.30 no.5
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• pp.738-740
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• 2008

In this paper, block constrained trellis coded vector quantization (BC-TCVQ) is presented for quantizing the line spectrum frequency parameters of the wideband speech codec. Both a predictive structure and a safety-net concept are combined into BC-TCVQ to develop the predictive BC-TCVQ. The performance of this quantization is compared with that of the linear predictive coding vector quantizer used in the AMRWB codec, demonstrating reductions in spectral distortion.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.33 no.8C
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• pp.575-582
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• 2008

In this paper we described the process and the results of real-time implementation of G.729.1 wideband speech codec which is standardized in SG15 of ITU-T. To apply the codec on ARM926EJ-S(R) processor core. we transformed some parts of the codec C program including basic operations and arithmetic functions into assembly language to operate the codec in real-time. G.729.1 is the standard wideband speech codec of ITU-T having variable bit rates of $8{\sim}32kbps$ and inputs quantized 16 bits PCM signal per sample at the rate of 8kHz or 16kHz sampling. This codec is interoperable with the G.729 and G.729A and the bandwidth extended wideband($50{\sim}7,000Hz$) version of existing narrowband($300{\sim}3,400Hz$) codec to enhance voice quality. The implemented G.729.1 wideband speech codec has the complexity of 31.2 MCPS for encoder and 22.8 MCPS for decoder and the execution time of the codec takes 11.5ms total on the target with 6.75ms and 4.76ms respectively. Also this codec was tested bit by bit exactly against all set of test vectors provided by ITU-T and passed all the test vectors. Besides the codec operated well on the Internet phone in real-time.

• 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.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.33 no.10B
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• pp.939-945
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• 2008

Tn this paper we described the process and the results of an implementation of Internet terminal using G.729.1 wideband speech codec for next generation network. For this purpose firstly we chose a high performance RISC application processor having DSP features for speech codec processing and enhanced Multimedia Accelerator(eMMA) function for video codec. In the implementation of this terminal, we used G.729.1 codec recently standardized in ITU-T which is a new scalable speech and audio codec that extends 0.729 speech coding standard. To adopt G.729.1 codec to this terminal we transformed most of the fixed point C codes which require more complexity into assembly codes so as to minimize processing time in the processor. As a result of this work we reduced the execution time of the original C codes about 80% and operated in real time on the terminal. For video we used H.263/MPEG-4 codec which is supported by the eMMA with hardware in the processor. In the SIP call processing test connected to real network we obtained under looms end-to-end delay and 3.8 MOS value measured with PESQ instrument. Besides this terminal operated well with commercial terminals.