• The Journal of Korean Institute of Communications and Information Sciences
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• v.38C no.3
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• pp.239-246
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• 2013

The IEEE 802.15.4 offers standard about PHY and MAC layer and features low power, low bandwidth, and low speed data communication. Because of this reason, IEEE 802.15.4 is only within a limited range such as sensor detection and home network; nevertheless, the research about transmission multimedia data like voice packet through wireless sensor networks is conducted widely. In this paper, we proposed the group communication system based on the sensor network. TDMA/TDD MAC based on the IEEE 802.15.4 PHY for voice communication on the sensor network is designed by improvement existing peer-to-peer voice communication on the sensor network and hardware is implemented for group communication. To measure the quality of designed system, mean opinion score (MOS) is obtained from the experiment and verified by using sine wave method. As a result of an experiment, we expect that a many cases of application solution can be developed using presented system.

• The Journal of the Acoustical Society of Korea
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• v.24 no.1
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• pp.43-51
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• 2005

In this paper, we propose an algorithm that improves a tone quality of a noisy audio signal in order to enhance a Performance of perceptual filter using noise energy control. Most of the algorithms which were proposed by the other researchers usually applied a filter using the noise energy acquired from a silent range. In this case. the improvement rate of tone quality decreases if the noise energy is changed by the magnitude or environment variation in a signal frame. But the Proposed method Provides the means to find a food estimated noise through energy control of the estimated noise which is obtained from a silent range. Also we can get the enhancement of tone qualify in low frequency band unlike other methods. To show the performance of the Proposed algorithm, various input signals which had a different signal-to-noise ratio (SNR) such as 5dB, l0dB, 15dB and 20dB were used to test the proposed algorithm. With the proposed algorithm, we could confirm the enhancement of tone quality in terms of segmental SNR (SSNR). noise-to-mask ration (NMR) and mean opinion score (MOS) test.

• 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).

• Journal of the Korean Institute of Intelligent Systems
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• v.24 no.1
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• pp.28-32
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• 2014

For a signal such as speech showing piece-wise linear shape in a very short time period, a nonuniform sampling method based on the inflection point detection (IPD) is proposed to reduce data rate. The method exploits the geometrical characteristics of signal further than the existing local maxima/minima detection (MMD) based sampling method. As results, the reconstructed signal by the interpolation of the IPD based sampled data resembles the original speech more. Computer simulation shows that the proposed IPD based method produces about 9~23 dB improvement over the existing MMD method. To show the usefulness of the IPD technique, it is applied to speech coding, and compared to the continuously variable slope delta modulation (CVSD). The nonuniformly sampled data is binary coded with one bit flag set "1". Noninflection samples are not sent, but only flag bits set 0 are sent. The method shows 0.3 ~ 9 dB SNR and 0.5 ~ 1.3 mean opinion score (MOS) improvements over the CVSD.

• The KIPS Transactions:PartC
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• v.19C no.2
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• pp.119-126
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• 2012

This paper covers the service quality of packet-based voice service that is provided over wireless broadband (WiBro) and high speed downlink packet access (HSDPA) systems. Using a measurement software that has been developed in the course of preparing a advanced service quality management scheme for the packet-based voice service over wireless networks, a huge scale of experiment is conducted to measure the real quality of the voice service. Based on our analysis of the measurement results, the service quality of the voice service is supposed to be quite good over both wireless systems. In addition, another experiment to investigate the effect of degradation of wireless transmission conditions on the service quality of the voice service shows the values of wireless service metrics in which mean opinion score (MOS) starts to decrease.

• The Transactions of The Korean Institute of Electrical Engineers
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• v.58 no.4
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• pp.857-866
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• 2009

In this paper, a noise cancellation-method using microphone array for digital hearing aids is proposed. The microphone array is located around the ear of a dummy. Speech sound is generated from the forward speaker positioned in the front of the dummy and noise sound is generated from the backward speaker. The speech and noise are mixed in the air space and entered into the microphones. VAD(voice activity detector) and ANC(adaptive noise cancellation) methods were used to eliminate noise in the sound of the microphones. 10 two-syllable words and 4 sentences were used for speech signals. Babble and car interior noise were used for noise signals. The performance of the proposed algorithm was evaluated by SNR(signal-to-noise ratio) and PESQ-MOS(perceptual evaluation of speech quality-mean opinion score). In babble noise condition, SNR was improved as much as $7.963{\pm}1.3620dB\;and\;3.968{\pm}0.6659dB$ for words and sentences respectively. In the case of car interior noise, SNR was improved as $10.512{\pm}2.0665dB\;and\;6.000{\pm}1.7642dB$ for words and sentences respectively. PESQ-MOS of the babble noise was improved as much as $0.1722{\pm}0.0861$ score for words and $0.083{\pm}0.0417$ score for sentences. And PESQ-MOS of the car interior noise was improved as $0.2661{\pm}0.0335$ score and $0.040{\pm}0.0201$ score for words and sentences respectively. It is verified that the proposed algorithm has a good performance in noise cancellation of microphone array for digital hearing aids.

• Proceedings of the IEEK Conference
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• 2001.09a
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• pp.939-942
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• 2001

본 논문에서는 ITU-T G.723.1, G.729 부호화기와 MLT(Modulated Lapped Transform) 방법을 이용한 광대역 음성 부호화방법을 제안한다. 제안된 광대역 음성부호화 방법은 16 kHz로 샘플링된 입력신호를 QMF(Quadrature Mirror Filter)사용하여 저대역과 고대역으로 나누며, 각 대역은 8 kHz의 샘플링을 갖는 협대역 음성 신호로 변환된다. 고대역은 MLT변환 후 벡터 양자화하며 또한 MLT를 사용한 ATC(Adaptive Transform Coding)방법을 적용하여 표현하며 저대역은 G.723.1과 G.729 부호화기를 사용한다. 설계된 광대역 음성부호화기의 성능을 평가하기 위하여 MOS (Mean Opinion score)실험을 수행하였다. MOS 실험을 통해 16 kbps G.729-MLT VQ방식이 G.722 56kbps 와 비슷한 음질을 나타내었다.

• MALSORI
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• v.68
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• pp.49-63
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• 2008

This paper describes the development and assessment of a hidden Markov model (HMM) based Tagalog speech synthesis system, where Tagalog is the most widely spoken indigenous language of the Philippines. Several aspects of the design process are discussed here. In order to build the synthesizer a speech database is recorded and phonetically segmented. The constructed speech corpus contains approximately 89 minutes of Tagalog speech organized in 596 spoken utterances. Furthermore, contextual information is determined. The quality of the synthesized speech is assessed by subjective tests employing 25 native Tagalog speakers as respondents. Experimental results show that the new system is able to obtain a 3.29 MOS which indicates that the developed system is able to produce highly intelligible neutral Tagalog speech with stable quality even when a small amount of speech data is used for HMM training.

• The Journal of the Acoustical Society of Korea
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• v.18 no.1E
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• pp.32-36
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• 1999

In the area of the speech synthesis techniques, the waveform coding methods maintain the intelligibility and naturalness of synthetic speech. In order to apply the waveform coding techniques to synthesis by rule, we must be able to alter the pitches of synthetic speech. In this paper, we propose a new pitch altering method that compensates phase distortion of the cepstral pitch alteration method with time scaling method in the time domain. This method can remove some spectrum distortion which is occurred in conjunction point between the waveforms. For performance test the spectrum distortion rate was used as objective criterion and the MOS(Mean Opinion Score) was used as subjective criterion. As a result, the spectrum distortion and MOS are obtained by 0.66% and 3.9, respectively.

• Journal of information and communication convergence engineering
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• v.7 no.3
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• pp.309-313
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• 2009

In this paper, end-to-end VoIP(Voice over Internet Protocol) performance is evaluated by simulation with NS-2 simulation tool. There are many results studied and published for VoIP performance over TCP/IP networks. But, almost all of them were focused on wired or wireless Internet environments. About MANET (Mobile Ad Hoc Network), VoIP is currently studying several points of research. In this paper, analysis of VoIP performance is done with focusing on the mobility of MANETs. MOS(Mean Opinion Score), network delay, packet loss rates are considered as end-to-end QoS performance parameters.