• The Journal of the Acoustical Society of Korea
• /
• v.15 no.2
• /
• pp.55-60
• /
• 1996

This paper describes the subjective assessment of the multi-channel audio quality according to the bitrate changing and evaluates the predistortion effect to avoid the unmasked noise after matrixing/dematrxing process in transmission and regeneration of the multi-channel audio. The simulation is processed by the perceptual coding that is MPEG-2 Audio layer II algorithm. We evaluate the quality improvement about predistortion using or not by 384, 320, 256, 128kbps. As the result of the double blind subjective assessment, 5 Grade-Impairment Scale is scored under minus one to 320kbps and so audio quality is evaluated to be perceptible, but not annoying in 3/2 channel. The effect of the predistortion is improved one level in 128kbps and especially speech test material I better improved than music test materials.

• Electronics and Telecommunications Trends
• /
• v.19 no.5 s.89
• /
• pp.114-124
• /
• 2004

유선 네트워크, 무선 이동통신 네트워크, 인터넷 등을 통합하는 유무선 통합 네트워크(BcN)에서는 VoIP기술을 사용하게 될 것이다. TTA 표준으로 2004년 7월에 제정된 VoIP 계층형 광대역 고품질 음성 코덱은 핵심계층에 G.711, G.723.1, G.729를 사용하므로 10종의 PT 를 설정하여 코덱을 협상한다. 이로 인하여 자기자신의 코덱 이외에도 G.711, G.723.1, G.729 등과 상호 호환이 되는 장점을 갖는다. 본 고는신규로 제정된 VoIP 계층형 광대역 고품질 음성 코덱을 네트워크에서 사용할 수 있도록 호 처리에 대한표준화를 추진하여야 하는데 이를 위한 표준 기술을 설명하고, 코덱과 호처리 관계 및 표준화 기술을 근거로 한 코덱 협상 처리 기술을 설명한다. 코덱 협상 처리 기술로서 PSTN/MSC 연동 코덱 협상 방안과All IP 코덱 협상 방안으로 구분하였다. All IP 코덱 협상 방안에서는 발신, 착신, MGC, 착신서버에서 호환성을 위한 호 처리 기능을 제공한다. 본 고의 호 처리 기술을 적용하면, VoIP 계층형 광대역 고품질 음성코덱은 기존 네트워크 장치 기능을 수정하지 않고 사용할 수 있다.

• Journal of the Institute of Electronics Engineers of Korea SP
• /
• v.46 no.3
• /
• pp.119-124
• /
• 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 Journal of the Acoustical Society of Korea
• /
• v.13 no.2E
• /
• pp.32-40
• /
• 1994

It is the purpose of this paper to describe the methods for establishing loudness ratings and talker echo out of transmission quality of ISDN telephone connected to fully digital network. In order to design the desirable loudness ratings and talker echo for ISDN telephone, the model system of digital speech communication for subjective tests is developed. Using this model system, opinion tests which decide the optimal CODEC input level, the range of overall loudness rating, sidetone masking rating and talker echo are performed. From the results of tests, we decided that the loudness ratings are 6 to 8dB for sending, 0 to 2dB for receiving, and 8 to 12dB for sidetone masking rating. And, the terminal coupling loss of TCLw of at least 40dB is necessary to provide echo-free telephone communications to telophone users when the overall loudness rating of ISDN telephone is normalized to 10dB.

• Journal of IKEEE
• /
• v.24 no.4
• /
• pp.1122-1128
• /
• 2020

Bandwidth Extension refers to restoring and expanding a narrow band signal(NB) that is damaged or damaged in the encoding and decoding process due to the lack of channel capacity or the characteristics of the codec installed in the mobile communication device. It means converting to a wideband signal(WB). Bandwidth extension research mainly focuses on voice signals and converts high bands into frequency domains, such as SBR (Spectral Band Replication) and IGF (Intelligent Gap Filling), and restores disappeared or damaged high bands based on complex feature extraction processes. In this paper, we propose a model that outputs an bandwidth extended signal based on an autoencoder among deep learning models, using the residual connection of one-dimensional convolutional neural networks (CNN), the bandwidth is extended by inputting a time domain signal of a certain length without complicated pre-processing. In addition, it was confirmed that the damaged high band can be restored even by training on a dataset containing various types of sound sources including music that is not limited to the speech.

• Journal of the Institute of Electronics and Information Engineers
• /
• v.50 no.7
• /
• pp.10-18
• /
• 2013

With advancing of mobile communication services and commercializing of VoLTE (Voice of LTE), it is getting to pay attention on QoS of VoLTE. This paper proposes an integrated E-model in which some factors influenced to service quality of VoIP and VoLTE based voice communication system are considered in calculating the voice quality of Wideband Codec. The model aims to calculate R value which reflects the situations of access network, network characteristics, terminals' usage and mobility. We mainly deal with the integrated E-model's structure, related algorithms and optimal parameters for VoLTE. Some experiments show that the voice quality difference between VoIP and VoiceChecker, and VoLTE and POLQA, is below 10%. With the proposed model, we can calculate the voice quality by making use of the factors directly affected to service quality and the environment of VoLTE terminal and network. As a result, we can estimate the service quality in advance, without measuring it in real wireless environment.

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

• Journal of the Institute of Electronics Engineers of Korea SD
• /
• v.38 no.7
• /
• pp.455-463
• /
• 2001

In this Paper, an approximate processing method is proposed and tested. The proposed method uses variable CSD (VCSD) coefficients which approximate filter stopband attenuation by controlling the precision of the CSD coefficient sets. A decimation filter for Audio Codec '97 specifications has been designed having processor architecture that consists of program/data memory, arithmetic unit, energy/level decision, and sinc filter blocks, and fabricated with 0.6${\mu}{\textrm}{m}$ CMOS sea-of-gate technology. For the combined two halfband FIR filters in decimation filter, the number of addition operations were reduced to 63.5%, 35.7%, and 13.9%, compared to worst-case which is not an adaptive one. Experimental results show that the total power reduction rate of the filter is varying from 3.8 % to 9.0 % with respect to worst-case. The proposed approximate processing method using variable CSD coefficients is readily applicable to various kinds of filters and suitable, especially, for the speech and audio applications, like oversampling ADCs and DACs, filter banks, voice/audio codecs, etc.