• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2007.10a
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• pp.693-697
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• 2007

The SSB modem performs the modulation process which converts the digital voltage level to the audible frequency band signal and the demodulation process which converts reversely the audible frequency signal to the digital voltage level. The modulator and the demodulator are implemented with a single DSP chip. Because of the SSB specific character, the distortion occurs when the frequency is changed. This distortion has no effect on voice communication, but it has an significant effect on data communication. In other words, it is impossible to send data stream with adjacent 2 periods. Therefore, in case of using 2-tone FSK, it is needed to send at least 3 periods to transmit 1 bit. Therefore we implemented the modem using modified phase-delay shift keying to transmit 1 tone signal for high speed transmission. In the 1200[bps] mode, it generates 0, $187{\mu}s$ delay time at 1.3kHz symbol frequency, and in the 2400[bps] mode, 0, $70{\mu}s$, $130{\mu}s$, $200{\mu}s$ delay time at 1.5kHz symbol frequency. Finally, in the maximum 3600[bps] mode, it generates 0, $100{\mu}s$, $160{\mu}s$, $250{\mu}s$ delay time at 2.0kHz symbol frequency. The measured results of the implemented SSB modem shows a good transfer functional characteristic by spectrum analyzer, almost same bandwidth in pass band and 20dB higher SNR comparing the German PACTOR and American CLOVER and in the experimental transmitting test, we verified the transmitted data is received correctly in platform.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.42 no.12
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• pp.127-134
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• 2005

Lost packet robustness is an most important quality measure for voice over IP networks(VoIP). Recovery of the lost packet from the received information is crucial to realize this robustness. So, this paper proposes the lost packet recovery method from the received information for real-time communication for CELP coder. The proposed BS-PLC (Both Side Packet Loss Concealment) based WSOLA(Waveform Shift OverLab Add) allow the lost packet to be recovered from both the 'previous' and 'next' good packet as the LP parameter and the excitation signal are respectively recovered. The burst of packet loss is modeled by Gilbert model. The proposed scheme is applied to G.729 most used in VoIP and is evaluated through the SNR(signal to noise) and the MOS(Mean Opinion Score) test. As a simulation result, The proposed scheme provide 0.3 higher in Mean Opinion Score and 2 dB higher in terms of SNR than an error concealment procedure in the decoder of G.729 at $20\%$ average packet loss rate.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.11 no.10
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• pp.1852-1857
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• 2007

The SSB modem performs the modulation process which converts the digital voltage level to the audible frequency band signal and the demodulation process which converts reversely the audible frequency signal to the digital voltage level. The modulator and the demodulator are implemented with a single DSP chip. Because of the SSB specific character, the distortion occurs when the frequency is changed. This distortion has no effect on voice communication but it has an significant effect on data communication. In other words, it is impossible to send data stream with adjacent 2 periods. Therefore, in case of using 2-tone FSK, it is needed to send at least 3 periods to transmit 1 bit. Therefore we implemented the modem using modified phase-delay shift keying to transmit 1 tone signal for high speed transmission. In the 1200[bps] mode, it generates 0, $187{\mu}s$, delay time at 1.3kHz symbol frequency, and in the 2400[bps] mode, 0, $70{\mu}s\;130{\mu}s\;200{\mu}s$, delay time at 1.5kHz symbol frequency. Finally, in the maximum 3600[bps] mode, it generates 0, $100{\mu}s\;160{\mu}s\;250{\mu}s$ 2.0kHz symbol frequency. The measured results of the implemented SSB modem shows a good transfer functional characteristic by spectrum analyzer, almost same bandwidth in pass band and 20dB higher SNR comparing the emu FACTOR and American CLOVER and in the experimental transmitting test, we verified the transmitted data is received correctly in platform.

• Journal of the Korean Society of Laryngology, Phoniatrics and Logopedics
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• v.10 no.1
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• pp.5-11
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• 1999

Background and Objectives : Recently, many people suffering from voice change visit otolaryngologist. So, it is important to study vocal function in patients with glottic or laryngeal disease. Aerodynamic investigation is valuable information about the efficiency of the larynx in translating air pressure to acoustic signal. The purpose of this study was to investigate the aerodynamic data in patients with vocal polyp, compare this data with that of the normal Korean. Materials and Methods : In aerodynamic study, maximum phonation time, mean air flow rate, phonatory flow volume and subglottal pressure were tested by using Aerophone II voice function analyzer in 157 normal korean and 143 patients with vocal polyp, aged from 20 to 69 years randomly selected. Results and Conclusion : In vocal polyp, maximum phonation time was significantly decreased and mean air How rate was increased. Phonatory flow volume was significantly decreased and subglottic pressure was increased only in female with vocal polyp. These data will be served as basic data of evaluation after treatment and postoperative assessment of the patients with vocal polyp.

• Journal of Digital Contents Society
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• v.17 no.2
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• pp.71-79
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• 2016

This study is intended to compare the performances of sound source localization methods used for stable automobile control by improving voice recognition rate in automobile environment and suggest how to improve their performances. Generally, sound source location estimation methods employ the TDOA algorithm, and there are two ways for it; one is to use a cross correlation function in the time domain, and the other is GCC-PHAT calculated in the frequency domain. Among these ways, GCC-PHAT is known to have stronger characteristics against echo and noise than the cross correlation function. This study compared the performances of the two methods above in automobile environment full of echo and vibration noise and suggested the use of a median filter additionally. We found that median filter helps both estimation methods have good performances and variance values to be decreased. According to the experimental results, there is almost no difference in the two methods' performances in the experiment using voice; however, using the signal of a song, GCC-PHAT is 10% more excellent than the cross correlation function in terms of the recognition rate. Also, when the median filter was added, the cross correlation function's recognition rate could be improved up to 11%. And in regarding to variance values, both methods showed stable performances.

• The Transactions of the Korea Information Processing Society
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• v.7 no.10
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• pp.3210-3218
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• 2000

In a IS-95 cellular systm, blocking will occur when the reverse link user interference power reaches a predepermmed level which is set to maintam acceptable signal quality. In this paper, it is assumed that a mobile rdio channel is a shadowing channel and Erlang capacity is calculated for the reverse limk of an imperfect power controlled IS-95 cellular system. the blocking probability is derived using lognornal pproximation and the results according to guassian and lognormal approximation method are compared and analyzed respcctively. Assuming that blocking probability is 1% at the data rate of $R_b$=9.6kbps and $R_b$=14.4kbps, it is shown that Erlang capacity using Iognormal approximation is 13.68 Erlang and 7.08 Erlang and then the approximation erroris occurred about 24.4% and 40.4% inthe garssian approximation, respectively. It is also observed that if the power control becomes periect, the Erlang capacity is increased more 6.99 and 4.21 Erlang than that of the imperfect power control that the power contrl error is 2.5dB, and if voice activity is considered as 10%, the Erlang capacity is increased more 8.21 and 1.25 Erlang than that using non voice activity, respectively.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.10 no.1
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• pp.29-35
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• 2006

The third generation mobile communications system requiring the reliable multimedia data transmission has provided with the reliable voice, data and video services over the variable propagation environment. However the broadband wireless multiple access technologies cause Inter Symbol Interference(ISI) or Multiple Access Interference(MAI) to degrade the performance of CDMA(Code Division Multiple Access) system. Constant Modulus Algorithm which is frequently used as the adaptive blind equalizers to remove the interfering signal has ill-convergence phenomenon without proper initialization. In this paper, new blind equalization method based on conventional CMA is proposed to improve the channel efficiency, and through computer simulation this is tested over the time varying fading environment of mobile communication system. consequently, new blind equalization method into concatenated Kalman filter with CMA is verified better than conventional CMA through adopting minimum mean square errors and eye-pattern obtained from algorithm are compared.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.49 no.3
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• pp.98-105
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• 2012

This paper describes a new method for a speech playback speed control with high quality. The proposed method provides an adaptive threshold filtering solution for detecting active regions of a speech signal that are followed by playback speed. For a given playback speed, threshold value is adaptively determined with the statistics(:mean and standard deviation) of each frame in speech, and is used to select only active blocks within the current frame. To minimize quality degradation(i.e., pitch degradation) caused due to high-speed playback, the threshold filtering priorly eliminates relatively low-activity blocks including voice and unvoice. Simulation results show that the proposed scheme provides a playback speed control solution with higher quality than SOLA(Synchonized OverLap Add) method using the pitch extraction of speech.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.40 no.11
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• pp.63-71
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• 2003

This paper suggests the packet loss recovery method to communicate in real time in the Internet. To reduce the effects of packet loss, Forward Error Correction (FEC) that adds redundant information to voice packets can be used. Adaptive Multi Rate Wideband(AMR-WB) codec which is recently selected by the Third Generation Partnership Project(3GPP) for GSM and the third generation mobile communication WCDMA system and has also been standardized in ITU-T for providing wideband speech services is used. The major cause for speech qualitly degradation in IP-networks is packet loss. So, We recovered single lossy packet by using FEC method and concealed continued errors. The proposed scheme if evaluated in the Gilbert Internet channel model. The high quality of audio maintained up to 30% packet loss.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.16 no.6 s.97
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• pp.603-614
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• 2005

In this paper, digital ultra-narrowband Walky-Talky using direct conversion method for CQPSK modulation scheme is implemented with satisfying the requirements of APCO P25. RF transceiver design and implementation scheme that minimize the influence of DC-offset and AC-coupling at ultra-narrowband is proposed. This scheme also minimizes the influence of nonlinear characteristic at power amplifier fir CQPSK modulation method. Test results of full system including DSP module and direct conversion RF transceiver show that FCC emission mask at 36.8 dBm PEP meets the standard requirements. The characteristic of receiver AGC by PWM control signal is linear at 40 dB dynamic range and voice communication at input power level of -116 dBm is successful. Also it is verified that the performance of BER versus frequency offset and versus SNR meets the standard requirements.