• Proceedings of the Acoustical Society of Korea Conference
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• spring
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• pp.57-60
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• 2004

본 연구에서는 ETSI 및 3GPP에 의해 개발된 광대역 음성부호화의 표준안인 AMR-WB 알고리즘을 분석하고, TMS320VC5510 DSK를 이용한 실시간 구현 결과를 제시하였다. AMR-WB 음성부호화기의 실시간 구현을 위해 프로그램 최적화 작업을 수행하였고, 구현된 음성 부호화기의 성능을 평가하기 위해서 프로그램 메모리와 데이터 메모리의 크기, 그리고 한 프레임당 수행 시간을 측정하였다. 구현된 시스템의 프로그램 메모리는 약65.6 kbytes, 데이터 메모리는 약 73.8kbytes 정도의 크기를 나타내었으며, 한 프레임인 20 msec를 처리하는데 소요되는 cycle 수가 평균 1,247,115 정도로 약 6.24 msec 내에 처리 할 수 있음을 보였다. 마지막으로 DSP로 구현한 AMR-WB 음성부호화기의 결과가 PC에서 시뮬레이션 한 결과와 일치함을 검증하였고 실시간으로 동작됨을 확인하였다.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.39 no.6
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• pp.696-708
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• 2002

In this paper, a novel transcoding algorithm for the Adaptive Multi Rate(AMR) and the Enhanced Variable Rate Codec(EVRC) vocoders via direct parameter transformation is proposed. In contrast to the conventional tandem transcoding algorithm, the proposed algorithm converts the parameters of one coder to the other without going through the decoding and encoding processes. The proposed algorithm consists of the parameter decoding, frame classification, mode decision, and transcoders for two frame types. The transcoders convert the parameters such as LSP, frame energy, pitch delay for the adaptive codebook, fixed codebook vector, and codebook gains. Evaluation results show that while exhibiting better computational and delay characteristics, the proposed algorithm produces equivalent speech quality to that produced by the tandem transcoding algorithm.

• Journal of Broadcast Engineering
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• v.23 no.1
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• pp.86-92
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• 2018

In this paper, we propose a method for coding history detection of digital speech signal. In digital speech communication and storage, the signal is encoded to reduce the number of bits. Therefore, when a speech signal waveform is given, we need to detect its coding history so that we can determine whether the signal is an original or an coded one, and if coded, determine the number of times of coding. In this paper, we propose a coding history detection method for 12.2kbps AMR codec in terms of original, single coding, and double coding. The proposed method extracts a speech-specific feature vector from the given speech, and models the feature vector using a deep neural network. We confirm that the proposed feature vector provides better performance in coding history detection than the feature vector computed from the general spectrogram.

• 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.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.46 no.3
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• pp.119-124
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• 2009

In this paper, AMR-WB+ audio coder is implemented, in real-time, using Intel 400MHz Xscale PXA250 with 32bit RISC processor ARM9E-J(R)core. The assembly code for ARM9E-J(R)core is developed through the serial process of C code optimization, cross compile, assembly code manual optimization and adjusting the optimized code to Embedded Visual C++ platform. C code is trimmed on Visual C++ platform. Cross compile and assembly code manual optimization are performed on CodeWarrior with ARM compiler. Through these stages the code for both ARM EVM board and PDA is implemented. The average complexities of the code are 160.75MHz on encoder and 33.05MHz on decoder. In case of static link library(SLL), the required memories are 65.21Kbyte, 32.01Kbyte and 279.81Kbyte on encoder, decoder and common sources, respectively. The implemented coder is evaluated using 16 test vectors given by 3GPP to verify the bit-exactness of the coder.

• The KIPS Transactions:PartC
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• v.16C no.5
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• pp.661-668
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• 2009

The Extended real-time Polling Service (ErtPS) is proposed tosupport QoS of VoIP service with silence suppression which generates variable size data packets in IEEE 802.16e systems. If the silence is suppressed, VoIP should support Comfort Noise Generation (CNG) which generates comfort noise for receiver's auditory sense to notify the status of connection to the user. CNG mode in silent-period generates a data with lower bit rate at long packet transmission intervals in comparison with talk-spurt. Therefore, if the ErtPS, which is designed to support service flows that generate data packets on a periodic basis, is applied to silent-period, resources of the uplink are used inefficiently. In this paper, we proposed the Improved ErtPS algorithm for efficient resource utilization of the silent-period in VoIP traffic supporting CNG. In the proposed algorithm, the base station allocates bandwidth depending on the status of voice at the appropriate interval by havingthe user inform the changes of voice status. The Improved ErtPS utilizes the Cannel Quality Information Channel (CQICH) which is an uplink subchannel for delivering quality information of channel to the base station on a periodic basis in 802.16e systems. We evaluated the performance of proposed algorithm using OPNET simulator. We validated that proposed algorithm improves the bandwidth utilization of the uplink and packet transmission latency

• 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 the Acoustical Society of Korea
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• v.30 no.1
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• pp.1-8
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• 2011

This paper presents a method for improving the performance of ISF coefficients quantizer through compensating the defect of the split structure vector quantization using the ordering property of ISF coefficients. And design the ISF coefficients quantizer for wideband speech codec using proposed method. The wideband speech codec uses split structure vector quantizer which could not use the correlation between ISF coefficients fully to reduce complexity and the size of codebook. The proposed algorithm uses the ordering property of ISF coefficients to overcome the defect. Using the ordering property, the codebook redundancy could be figured out. The codebook redundancy is replaced by the adaptive-extended codebook to improve the performance of the quantizer through using the ordering property, ISF coefficient prediction and interpolation of existing codebook. As a result, the proposed algorithm shows that the adaptive-extended codebook algorithm could get about 2 bit gains in comparison with the existing split structure ISF quantizer of AMR-WB (G.722.2) in the points of spectral distortion.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.26 no.12A
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• pp.2027-2035
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• 2001

In this paper, we present an audio stream delivery using the AMR (Adaptive Multi-Rate) coder that was adopted by ETSI and 3GPP as a standard vocoder for next generation IMT-2000 service in which includes combined sender (FEC) and receiver reconstruction technique in the Internet. By use of the media-specific FEC scheme, the possibility to recover lost packets can be much increased due to the addition of repair data to a main data stream, by which the contents of lost packets can be recovered. The AMR codec is based on the code-excited linear predictive (CELP) coding model. So we use a frame erasure concealment for CELP-based coders. The proposed scheme is evaluated with ITU-T G.729 (CS-ACELP) coder and AMR - 12.2 kbit/s through the SNR (Signal to Noise Ratio) and the MOS (Mean Opinion Score) test. The proposed scheme provides 1.1 higher in Mean Opinion Score value and 5.61 dB higher than AMR - 12.2 kbit/s in terms of SNR in 10% packet loss, and maintains the communicab1e quality speech at frame erasure rates lop to 20%.