• The Transactions of the Korea Information Processing Society
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• v.7 no.8
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• pp.2463-2472
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• 2000

In this paper, we consider pits and falls of V oIP service scheme over the air link environment. It results that V oIP over packet switching is a more attractive approach in several points. We point out the ffilIior requirements for successful V oIP service over the air. Also, we propose V oIP CP concept for efficient wireless channel utilization. Additionally, we analyze and evaluate the performance. According to the results, It shows that the long cycle VolP vocoder CODEC such as ITU-T G.723 is better than short cycle V oIP vocoder CODEC. In this case, the increase of the simultaneous user of system is almost 60% larger than conventional circuit switching.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.1C
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• pp.81-89
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• 2003

For improving the performance of noise suppression in tactical communication environments, an enhanced MELP vocoder is suggested, in which an acoustic noise suppressor is integrated into the front end of the MELP algorithm, and an FEC code into the channel side of the MELP algorithm. The acoustic noise suppressor is the modified IS-127 EVRC noise suppressor which is adapted for the MELP vocoder. As for FEC, the turbo code, which consists of rate-113 encoding and BCJR-MAP decoding algorithm, is utilized. In acoustic noise environments, the lower the SNR becomes, the more the effects of noise suppression is increased. Moreover, The suggested system has greater noise suppression effects in stationary noise than in non-stationary noise, and shows its superiority by 0.24 in MOS test to the original MELP vocoder. When the interleave size is one MELP frame, BER 10-6 is accomplished at channel bit SNR 4.2 ㏈. The iteration of decoding at 3 times is suboptimal in its complexity vs. performance. Synthetic quality is realized as more than MOS 2.5 at channel bit SNR 2 ㏈ in subjective voice quality test, when the interleave size is one MELP frame and the iteration of decoding is more than 3 times.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.33 no.9C
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• pp.691-696
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• 2008

A new vocoder of Code Excited Linear Predictive (CELP) based on Adaptive Multi Rate (AMR) 7.4kbit/s mode is proposed in this paper. The proposed vocoder achieves a better compression rate in an environment of Speaker Dependent Coding System (SDSC) and is efficiently used for systems, such as OGM(Outgoing message) and TTS(Text To Speech), which needs only one person's speech. In order to enhance the compression rate of a coder, a new Line Spectral Pairs(LSP) code-book is employed by using Centroid Neural Network (CNN) algorithm. In comparison with original(traditional) AMR 7.4 Kbit/s coder, the new coder shows 27% higher compression rate while preserving synthesized speech quality in terms of Mean Opinion Score(MOS).

• The Journal of the Acoustical Society of Korea
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• v.20 no.1
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• pp.56-61
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• 2001

This paper presents the real-time implementation of an AMR (Adaptive Multi Rate) vocoder which is included in the asynchronous International Mobile Telecommunication (IMT)-2000 mobile ASIC. The implemented AMR vocoder is a multi-rate coder with 8 modes operating at bit rates from 12.2kbps down to 4.75kbps. Not only the encoder and the decoder as basic functions of the vocoder are implemented, but VAD (Voice Activity Detection), SCR (Source Controlled Rate) operation and frame structuring blocks for the system interface are also implemented in this vocoder. The DSP for AMR vocoder implementation is a 16bit fixed-point DSP which is based on the TeakLite core and consists of memory block, serial interface block, register files for the parallel interface with CPU, and interrupt control logic. Through the implementation, we reduce the maximum operating complexity to 24MIPS by efficiently managing the memory structure. The AMR vocoder is verified throughout all the test vectors provided by 3GPP, and stable operation in the real-time testing board is also proved.

• The Journal of the Acoustical Society of Korea
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• v.41 no.1
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• pp.52-57
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• 2022

Speech consists of the sum of complex sine-waves. This study investigated the perception of electrical hearing by applying the sinusoidal model to cochlear implant simulation. Fourteen adults with normal hearing participated in this study. The sentence recognition tests were implemented using the sentence lists processed by the sinusoidal model which extracts 2, 4, 6, 8 sine-wave components and sentence lists processed by the same sinusoidal model along with cochlear implant simulation (8 channel vocoders). The results showed lower speech recognition for the sentence lists processed by the sinusoidal model and cochlear implant simulation compared to those by the sinusoidal model alone. Notably, the lower the number of sine-wave components (2), the larger the difference was. This study provides the perceptual pattern of sine-wave speech for electrical hearing by cochlear implant listeners, and basic data for development of speech processing algorithms in cochlear implants.