• Journal of the Korean Society of Laryngology, Phoniatrics and Logopedics
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• v.32 no.2
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• pp.87-93
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• 2021

There are a number of voice analysis programs around the world. Domestic voice analysis is performed by relying heavily on specific commercial program. We intend to develop coding for voice analysis using Praat and apply it to clinical practice. This study consisted of Experiment 1 and Experiment 2. Experiment 1 was the development of automated voice analysis coding based on Praat. The coding was largely divided into a recording, an analysis, and a storage section. Experiment 2 was applied to the voice analysis of 2 male patients pre- and post-operation with this coding. The analysis parameters of this coding provided 26 parameters for vowel /a/, nine parameters for sentence analysis, and a total of 4 parameters for voice range profile analysis. In two male patients, the pitch and the intensity increased, the voice quality improved, and the sentence length decreased after surgery. The coding was well made, so the output was good in real time. The code is automated as much as possible to block manual errors and increases convenience and efficiency by generating the result sheet in real time.

• Journal of Broadcast Engineering
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• v.13 no.5
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• pp.612-621
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• 2008

Due to the various constraints such as feasible size of data payload and low transmission power, no technical specifications on the voice communication are included in the Zigbee standard. In this paper, a voice coding technique for application to the IEEE 802.15.4 standard, which is the basis of Zigbee communication, is presented. Here, both high compression and good waveform recovery are essential. To meet those requirements, a multi-stage discrete wavelet transform (DWT) block and a binary coding block consisting of two different pulse-code modulations are exploited. Theoretical analysis and simulation results in an indoor wireless channel show that the voice coder with 2-stage DWT is most appropriate from the viewpoint of compression and waveform recovery. When the line-of-sight component is dominant, the voice coding scheme has good recovery capability even in the moderate signal-to-noise power ratios. Hence, it is considered that the presented scheme will be a technical reference for the future recommendation of voice communication exploiting Zigbee.

• The Journal of the Acoustical Society of Korea
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• v.18 no.6
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• pp.71-76
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• 1999

In digital cellular communication systems, tandem coding occurs in communications between mobile phones with different speech codecs, resulting in poor voice quality, high computational load, and long transmission delay. In this paper, voice packet conversion technique is proposed to solve the tandem coding problems, and packet conversion algorithm from 13kbps QCELP to 8kbps QCELP is developed. Simulations using various speech data show that the proposed packet conversion method produces voice quality which is equivalent to that by the conventional tandem coding method with shorter transmission delay using about 33% computational load.

• Journal of the Korean Society of Surveying, Geodesy, Photogrammetry and Cartography
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• v.18 no.3
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• pp.279-285
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• 2000

Men's desire for the human interface, due to the development of voice processing technology of computer, and the development of intelligent MMI (Man-Machine Interface) computer technology enabled us to operate computers with our voice without using keyboards or other input systems. Especially, by obtaining field data and layout from the complicated surveying environment and applying the voice recognition technology to the actual surveying work, we can save a lot of working hours and costs. According to the result of this study, the real time Geo-Coding and graphic data-coding were possible with only 25 words by connecting the software engine which recognizes 50,000 different words and the voice recognition technology based on the super IC which recognizes 60 different words with the Total-station and the RTK-GPS.

• Journal of Internet Computing and Services
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• v.19 no.6
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• pp.1-7
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• 2018

This paper present an 8kbps ACLMS-MPC(Amplitude Compensation and Least Mean Square - Multi Pulse Coding) coding method integrated with ACFBD-MPC(Amplitude Compensation Frequency Band Division - Multi Pulse Coding) and LMS-MPC(Least Mean Square - Multi Pulse Coding) used V/UV/S(Voiced / Unvoiced / Silence) switching, compensation in a multi-pulses each pitch interval and Unvoiced approximate-synthesis by using specific frequency in order to reduce distortion of synthesis waveform. In integrating several methods, it is important to adjust the bit rate of voiced and unvoiced sound source to 8kbps while reducing the distortion of the speech waveform. In adjusting the bit rate of voiced and unvoiced sound source to 8 kbps, the speech waveform can be synthesized efficiently by restoring the individual pitch intervals using multi pulse in the representative interval. I was implemented that the ACLMS-MPC method and evaluate the SNR of APC-LMS in coding condition in 8kbps. As a result, SNR of ACLMS-MPC was 15.0dB for female voice and 14.3dB for male voice respectively. Therefore, I found that ACLMS-MPC was improved by 0.3dB~1.8dB for male voice and 0.3dB~1.6dB for female voice compared to existing MPC, ACFBD-MPC and LMS-MPC. These methods are expected to be applied to a method of speech coding using sound source in a low bit rate such as a cellular phone or internet phone. In the future, I will study the evaluation of the sound quality of 6.9kbps speech coding method that simultaneously compensation the amplitude and position of multi-pulse source.

• Phonetics and Speech Sciences
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• v.3 no.2
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• pp.89-96
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• 2011

In this paper, details of the design and implementation of a voice source simulator using Simulink and Matlab are discussed. This simulator is an implementation by model-based design concept. Voice sources can be analyzed and manipulated through various factors by choosing options from GUI input and selecting pre-defined blocks or user created ones. This kind of simulation tool can simplify the procedure of analyzing speech signals for various purposes such as voice quality analysis, pathological voice analysis, and speech coding. Also, basic analysis functions are supported to compare the original signal and the manipulated ones.

• Journal of Digital Convergence
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• v.13 no.7
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• pp.177-182
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• 2015

In a MPC(Multi-Pulse Coding) using excitation source of voiced and unvoiced, it would be a distortion of voice waveform. This is caused by normalization of synthesis speech waveform of voiced in the process of restoration. To solve this problem, this paper present APC-MPC of amplitude-position compensation in a multi-pulses each pitch interval in order to reduce distortion of synthesis waveform. Also, I was implemented that the APC-MPC in coding system. And I evaluate the SNRseg of APC-MPC in 8kbps coding condition of convergence system. As a result, SNRseg of APC-MPC was 13.9dB for female voice and 14.3dB for male voice respectively. And so, I expect to be able to this method for cellular phone and smart phone using excitation source of low bit rate.

• The Journal of the Acoustical Society of Korea
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• v.18 no.3E
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• pp.26-29
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• 1999

In this paper, we propose a new voice coding using only the features of the face image such as mouth height(H), width(W), rate(R=W/H), area(S), and ellipse's feature(P). It provides high security and is not affected by acoustic noise because we use only the features of face image for speech. In the proposed algorithm, the mean recognition rate for the vowels approximately rises between 70% and 96% after many tests.

• Proceedings of the IEEK Conference
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• 1998.10a
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• pp.1247-1250
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• 1998

In speech synthesis, LF model is widely used for excitation signal for voice source coding system. But LF model does not represent the harmonic frequencies of excitation signal. We propose an effective method which use sinusoidal functions for representing the harmonics of voice source signal. The proposed method could achieve more exact voice source waveform and better synthesized speech quality than LF model.

• Journal of the Korea Society for Simulation
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• v.10 no.2
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• pp.15-24
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• 2001

In this paper we study the delay patterns of voice traffic for end-to-end users Caused by serving the whole bunch of applications traffic at the same time on the Voice over Internet Protocol (VoIP) network. Given the current situation that voice traffic is served along with other application services on the VoIP network, it is quite necessary to figure out how and by what the voice traffic requiring high QoS is delayed. We compare the delay performance of voice traffic on the VoIP network under FIFO with the one under Weighted Fair Queuing(WFQ), and discover the differences of the delay performance resulting from the use of different voice codec algorithms. The results of our study show that using the voice codec algorithm with a higher coding rate nd the queuing algorithm of WEQ can provide users with high-quality voice traffic.