• Proceedings of the Acoustical Society of Korea Conference
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• autumn
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• pp.13-16
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• 2001

본 논문에서는 3GPP 규격에 따른 IMT-2000 시스템용 UE AMR 코덱의 소프트웨어 및 하드웨어 개발에 관하여 논한다 UE AMR 코덱은 ASIC 개발을 고려하여 Teak-Lite DSP 칩을 사용하여 개발하였다 AMR 코덱을 구현하기 위한 효율적인 소프트웨어 개발 기법을 설명하고 하드웨어 디자인도 논한다 개발된 UE AMR 코덱에는 음성 데이터 입출력 기능은 물론 리부 호스트 프로세서와의 통신 기능도 포함된다. Teak-Lite EVM보드를 사용하여 실시간으로 동작하는 AMR 코덱 소프트웨어를 개발하였다. 또한 동시에 UE AMR 코덱용 하드웨어도 개발하였다. ETRI에서 개발 및 시험 중인 IMT-2000 시스템 상에서 개발한 UE AMR 코덱의 동작 및 기능을 검증하였다.

• Proceedings of the Korean Operations and Management Science Society Conference
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• 2005.05a
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• pp.230-237
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• 2005

WCDMA방식에서 채택한 AMR(Adaptive Multirate) 음성 코덱은 4.75kbps에서 12.2kbps까지 8개의 가변 출력률을 가진다. 기지국제어시스템(Radio Network Controller)은 무선망 상황에 따라 AMR 출력 모드를 가변적으로 조정해 줌으로써 높은 사용자 QoS와 효율적인 시스템 성능을 얻을 수 있다. 본 연구에서는 순방향 WCDMA 채널에서 세 가지의 AMR 출력 모드 할당 방식을 제안하고, 음성 사용자가 경험하는 QoS 만족도를 시스템 성능 척도로하여 제안된 방식들을비교 할 수 있는 분석적 방법을 제시한다. 실험 예제를 통해서 시스템 부하에 따른 시스템 성능 척도의 변화를 도시함으로써 최적의 AMR 모드 할당 방식을 결정하는 기준을 제시한다.

• The Journal of the Acoustical Society of Korea
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• v.30 no.4
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• pp.190-196
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• 2011

Bandwidth extension is a technique to improve speech quality and intelligibility, extending from 300-3400 Hz narrowband speech to 50-7000 Hz wideband speech. This paper designs an artificial bandwidth extension (ABE) module embedded in the AMR (adaptive multi-rate) decoder, reducing LPC/LSP analysis and algorithm delay of the ABE module. We also introduce a fast search codebook mapping method for ABE, and design a low power BWE technique based on the AMR decoder. The proposed ABE method reduces the computational complexity and the algorithm delay, respectively, by 28 % and 20 msec, compared to the traditional DTE (decode then extend) method. We also introduce a weighted classified codebook mapping method for constructing the spectral envelope of the wideband speech signal.

• Proceedings of the IEEK Conference
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• 2000.09a
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• pp.811-814
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• 2000

본 논문에서는 AMR 음성 부호화 알고리즘을 분석하고 C프로그램 최적화 과정을 거친후 OakDSPCore?를 기반으로 설계된 C&S Technology사의 CSD17C00A칩을 이용하여 전과정을 어셈블리어로 실시간 구현 하였다. 구현된 코덱은 최대의 계산량을 요구하는 6.7kbps 모드일때, 인코더부분이 최대 20.6MIPS 이며 디코더부분은 약 2.7MIPS 의 복잡도를 나타낸다. 사용된 프로그램 메모리는 약 21.97kwords, 데이터 RAM 메모리는 약 6.64kwords를 가지며 데이터 ROM 메모리는 약 15.1kwords 이다. 구현된 코덱은 최대 약23.29MIPS의 복잡도를 가지고 있으므로 40MIPS의 처리용량을 가지는 CSD17C00A 를 이용한 보드상에서 실시간 동작이 가능함을 확인하였다. 구현된 프로그램은 3GPP에서 제공하는 21개의 test 벡터들을 통하여 bit-exact 함을 확인하였다.

• 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.

• 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 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

• The Journal of the Acoustical Society of Korea
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• v.28 no.3
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• pp.290-297
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• 2009

This paper presents an improved adaptive multi-rate wideband (AMR-WB) algorithm for the efficient Text-To-Speech (TTS) database compression. The proposed algorithm includes unnecessary common bit-stream (CBS) removal and parameter delta coding combined with speaker-dependent huffman coding to reduce the required bit-rate without any quality degradation. We also propose lossy coding schemes to produce the maximum bit-rate reduction with negligible quality degradation. The proposed lossless algorithm including CBS removal can reduce bit-rate by 12.40% without quality degradation compared with the 12.65 kbps AMR-WB mode. The proposed lossy algorithm can reduce bit-rate by 20.00% with 0.12 PESQ degradation.

• The Journal of the Acoustical Society of Korea
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• v.22 no.4
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• pp.293-298
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• 2003

In this paper. we propose a fast search algorithm for nearest neighbor vector quantization (NNVQ). The proposed algorithm rejects those codewords which can not be the nearest codeword and reduces the search range of codebook. Hence it reduces computational time and complexity in encoding process, while it provides the same SD performance as the conventional full search algorithm. We apply the proposed algorithm to the adaptive multi-rate (AMR) speech coder and a general vector quantizer designed by LBG. algorithm. Simulation results show effectiveness of the proposed algorithm.

• Journal of the Korea Institute of Information Security & Cryptology
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• v.22 no.1
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• pp.57-65
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• 2012

People leave voicemails or record phone conversations in their daily cell phone use. Sometimes important voice data is deleted by the user accidently, or purposely to cover up criminal activity. In these cases, deleted voice data must be able to be recovered for forensics, since the voice data can be used as evidence in a criminal case. Because cell phones store data that is easily fragmented in flash memory, voice data recovery is very difficult. However, if there are identifiable patterns for the deleted voice data, we can recover a significant amount of it by researching images of it. There are several types of voice data, such as QCP, AMR, MP4, etc.. This study researches the data recovery solutions for EVRC codec and AMR codec in QCP file, Qualcumm's voice data format in cell phone.