• Journal of Korean Institute of Industrial Engineers
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• v.31 no.4
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• pp.349-357
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• 2005

The Adaptive Multi-Rate (AMR) speech codec is the mandatory for voice service in WCDMA systems. The AMR codec can be used efficiently to provide a balanced trade-off between the capacity and quality of voice by adjusting various service rates. In this paper, three ways of AMR mode allocation schemes on the downlink in WCDMA system are evaluated. To evaluate users satisfaction efficiently, new system performance measure and analytic models are proposed. The proposed analytic models can be applied to obtain optimal mode allocation ways while considering the system capacity and quality of voice. In numerical examples, the ways of finding optimal parameters are illustrated for the given traffic loads and the performances of three mode allocation schemes are compared.

• 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.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2018.10a
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• pp.490-492
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• 2018

This paper presents an analysis of the AMR speech data as a basic work to study the technique of inputting and transmitting AMR voice data which is widely used in the public cell phone. AMR consists of HEADER and Speech Data, and it is transmitted in bit format and has 8 bit-rate modes in total. HEADER contains mode information of Speech Data, and the length of Speech Data differs depending on the mode. We chose the best mode which is best to input into QR code and analyzed that mode. It is a goal to show a higher compression ratio for voice data by the analysis and experiments. This analysis shows improvement in that it can transmit voice data more effectively.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.38 no.3
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• pp.304-315
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• 2001

In this paper, we peformed the real-time implementation of AMR(Adaptive Multi-Rate) speech coding algorithm which is adopted for IMT-2000 service using TMS320C6201, i.e., a Texas Instrument´s fixed-point DSP. With the ANSI C source code released from ETSI, optimization is performed to make it run in real-time with memory as small as possible using the C compiler and assembly language. Implemented AMR speech codec has the size of 32.06 kWords program memory, 9.75 kWords data RAM memory, and 19.89 kWords data ROM memory. And, The time required for processing one frame of 20 ms length speech data is about 4.38 ms, and it is short enough for real-time operation. 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.

• 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.

• The Journal of the Acoustical Society of Korea
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• v.20 no.6
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• pp.34-39
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• 2001

An adaptive multi-rate (AMR) speech coder was adopted as a standard of W-CDMA by 3GPP and ETSI. The AMR coder is based on the CELP algorithm operating at rates ranging from 12.2 kbps down to 4.75 kbps, and it is a source controlled codec according to the channel error conditions and the traffic loading. In this paper, we implement the DSP S/W of the AMR coder using OakDSPCore. The implementation is based on the CSD17C00A chip developed by C&S Technology, and it is tested using test vectors, for the AMR speech codec, provided by ETSI for the bit exact implementation. The DSP B/W requires 20.6 MIPS for the encoder and 2.7 MIPS for the decoder. Memories required by the Am coder were 21.97 kwords, 6.64 kwords and 15.1 kwords for code, data sections and data ROM, respectively. Also, actual sound input/output test using microphone and speaker demonstrates its proper real-time operation without distortions or delays.

• Proceedings of the Acoustical Society of Korea Conference
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• autumn
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• pp.93-96
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• 2001

ACELP 구조의 음성 압축기는 우수한 음질을 제공하지만 최적의 코드 벡터를 구하기 위한 계산량이 상당히 많은 단점이 있다. 이를 해결하기 위해서 본 논문에서는 AMR 음성 압축기의 코드북을 매우 효율적으로 검색하는 새로운 방법을 제안한다. 제안하는 코드북 검색 방법은 완전 순차적인 검색 방법을 사용하여 대략적인 코드 벡터를 구하고, 코드 벡터의 각 펄스들의 중요도를 계산하여 중요도가 낮은 펄스를 새로운 펄스로 교환하는 펄스 교환 과정을 수행하여 코드 벡터의 성능을 향상시키는 방법을 사용한다. 또한, AMR 음성 압축기의 구조에 맞도록 트랙별로 이동하면서 순차적으로 코드북을 검색하여 다수의 대략적인 코드벡터를 찾은 후, 각 코드 벡터에 대하여 펄스 교환 과정을 수행하여 최적의 코드 벡터를 구한다. 제안한 코드북 검색 방법을 AMR 음성 압축기의 모든 모드에 적용하여 코드북 검색을 위한 계산량과 성능을 측정하였으며, 모든 모드에 대하여 매우 적은 계산량으로 동등한 성능을 가지는 것을 확인하였다

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.40 no.11
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• pp.63-71
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• 2003

This paper suggests the packet loss recovery method to communicate in real time in the Internet. To reduce the effects of packet loss, Forward Error Correction (FEC) that adds redundant information to voice packets can be used. Adaptive Multi Rate Wideband(AMR-WB) codec which is recently selected by the Third Generation Partnership Project(3GPP) for GSM and the third generation mobile communication WCDMA system and has also been standardized in ITU-T for providing wideband speech services is used. The major cause for speech qualitly degradation in IP-networks is packet loss. So, We recovered single lossy packet by using FEC method and concealed continued errors. The proposed scheme if evaluated in the Gilbert Internet channel model. The high quality of audio maintained up to 30% packet loss.

• The Journal of the Acoustical Society of Korea
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• v.30 no.5
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• pp.265-272
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• 2011

In most of the ISF (or LSF) based real time speech codec, SVQ (split vector quantization) method is used to decrease the quantizer complexity and memory size of codebook. However, it produces drawback that the level of correlation between code vectors can not be used during vector splits. This paper presents a new method of adaptive ISF vector quantization, which compensates the drawbacks of SVQ structured quantizer for wideband speech codec. In each different frame, the proposed method makes use of the correlation between splitted vectors by adaptively changing codebook distribution according to ordering property of ISF. The algorithm is evaluated in AMR-WB, and shows about 1.5 bit per frame improvement.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.38 no.3
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• pp.68-68
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• 2001

본 논문에서는 3GPP와 ETSI에서 IMT-2000의 음성부호화 방식 표준안으로 채택한 AMR 음성부호화 알고리즘을 분석하고 C 컴파일러와 어셈블리 언어를 이용한 최적화 과정을 거친 후, 고정 소수점 DSP 칩인 TMS320C6201을 이용하여 실시간 구현하였다. 구현된 codec의 프로그램 메모리는 약 31.06 kWords, 데이터 RAM 메모리는 약 9.75 kWords, 그리고 데이터 ROM 메모리는 약 19.89 kWords 정도를 가지며, 한 프레임(20 ms)을 처리하는데 약 4.38 ms가 소요되어 TMS320C6201 DSP 칩의 전체 가용한 clock의 21.94%만 사용하여도 충분히 실시간으로 동작 가능함을 확인하였다. 또한, DSP 보드상에서 구현한 결과가 ETSI에서 공개한 ANSI C 소스 프로그램의 수행 결과와 일치함을 검증하였고, 구현된 AMR 음성부호화기를 sound I/O 모듈과 결합하여 실험한 결과, 어떠한 음질의 왜곡이나 지연 없이 실시간으로 충분히 동작함을 확인하였다. 마지막으로, Host I/O와 LAN 케이블을 이용하여 AMR 음성부호화 알고리즘을 통한 쌍방간 실시간 통신을 full-duplex 모드로 확인하였다.