• Title/Summary/Keyword: MOS test

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The in Fluence of Stage Make-Up to Psychological Condition of Performers and Performance (무대분장이 공연자의 심리상태 교 공연수행에 미치는 영향)

  • Ryu, Se-Ja;Park, Meegn-Ee
    • Journal of the Korean Society of Costume
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    • v.55 no.7 s.98
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    • pp.51-61
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    • 2005
  • The purpose of this study is to identify function of make-up as an Important variable in terms of its influence to psychology of performers and outcome of their performance. From June to November 2004 performers in 4 types of fields, i.e., opera, musical, drama and dancing being performed in Seoul and Cheongju were chosen as subjects and totally 450 questionnaires were prepared and distributed and among them 416 were used as data for final analysis. For data analysis frequency analysis, factor analysis, T-test, one-way ANOVA, path analysis, chi-square test were conducted by means of SPS 12.0 and MOS 4.0 statistical programs and as ex post facto checking Duncan's multiful range test was conducted. Make-up is an important element in acting and it has great influence on level of psychological satisfaction of an individual. It was disclosed that psychological factor of concentration and lethargy have direct bearing on acting performance. In order to maximize actor or actresses performance skill perfect make-up is essential and role of make-up specialist can become a critical factor for inducing success in performance. Role and duty of make-up artists in terms of scope of their responsibility should be extended so that they may give their full support to the performers to be most successful in their performance.

Intonatin Conversion using the Other Speaker's Excitation Signal (他話者의 勵起信號를 이용한 抑揚變換)

  • Lee, Ki-Young;Choi, Chang-Seok;Choi, Kap-Seok;Lee, Hyun-Soo
    • The Journal of the Acoustical Society of Korea
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    • v.14 no.4
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    • pp.21-28
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    • 1995
  • In this paper an intonation conversion method is presented which provides the basic study on converting the original speech into the artificially intoned one. This method employs the other speaker's excitation signals as intonation information and the original vocal tract spectra, which are warped with the other speaker's ones by using DTW. as vocal features, and intonation converted speech signals are synthesized through short-time inverse Fourier transform(STIFT) of their product. To evaluate the intonation converted speech by this method, we collect Korean single vowels and sentences spoken by 30 males and compare fundamental frequency contours spectrograms, distortion measures and MOS test between the original speech and the converted one. The result shows that this method can convert and speech into the intoned one of the other speaker's.

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Reconstruction of Transmitted Frames for Visual Quality Assessment of Streaming Video (스트리밍 비디오 화질 평가를 위한 수신 영상 복원)

  • Park, Su-Kyung;Sim, Dong-Gyu
    • Journal of the Institute of Electronics Engineers of Korea SP
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    • v.46 no.1
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    • pp.32-40
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    • 2009
  • In this paper, we proposed an reconstruction algorithm of transmitted frames from displayed image on video terminal. For image quality assessment of the video streaming in the wireless network, we need information of the image that is transmitted to the end-user's device. Generally, subjective methods are widely used to evaluate the image quality by human beings because it is difficult to extract the transmitted image from the end-user's device. This paper presents an image reconstruction algerian based on the displayed image in video terminal for the extraction of the transmitted image. In the proposed method, we acquired the displayed image on video terminal using the camera. Camera-acquired images exhibit geometric and color distortions caused by characteristics of cameras and display devices. Therefore we correct the geometric distortion by exploiting the homography and color distortion by pre-computed look-up table. The experimental results show that the proposed measurement system yields promising estimation performance in terms of PSNR of $27{\sim}28dB$. We also carried out performance evaluation of the proposed method in terms of EPSNR and the quality of the estimated images by the proposed algerian was in fairly good range of MOS test scale.

Design of the 5th-order Elliptic Low Pass Filter for Audio Frequency using CMOS Switched Capacitor (CMOS 스위치드 캐패시터 방식의 가청주파수대 5차 타원 저역 통과 여파기의 설계 및 구현)

  • Song, Han-Jung;Kwack, Kae-Dal
    • Journal of the Korean Institute of Telematics and Electronics C
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    • v.36C no.1
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    • pp.49-58
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    • 1999
  • This paper describes an integrated low pass filter fabricated by using $0.8{\mu}m$ single poly CMOS ASIC technology. The filter has been designed for a 5th-order elliptic switched capacitor filter with cutoff frequency of 5khz, 0.1dB passband ripple. The filter consists of MOS swiches poly capacitors and five CMOS op-amps. For the realization of the SC filter, continuous time transfer function H(s) is obtained from LC passive type, and transfered as discrete time transfer H(z) through bilinear-z transform. Another filter has been designed by capacitor scaling for reduced chip area, considering dynamic range of the op-amp. The test results of two fabricated filters are cutoff frequency of 4.96~4.98khz, 35~38dB gain attenuation and 0.72~0.81dB passband ripple with the ${\pm}2.5V$power supply clock of 50KHz.

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Implementation of 1.9GHz RF Frequency Synthesizer for USN Sensor Nodes (USN 센서노드용 1.9GHz RF 주파수합성기의 구현)

  • Kang, Ho-Yong;Kim, Nae-Soo;Chai, Sang-Hoon
    • Journal of the Institute of Electronics Engineers of Korea SD
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    • v.46 no.5
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    • pp.49-54
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    • 2009
  • This paper describes implementation of the 1.9GHz RF frequency synthesizer with $0.18{\mu}m$ silicon CMOS technology being used as an application of the USN sensor node transceiver modules. To get good performance of speed and noise, design of the each module like VCO, prescaler, 1/N divider, fractional divider with ${\Sigma }-{\Delta}$ modulator, and common circuits of the PLL has been optimized. Especially to get good performance of speed, power consumption, and wide tuning range, N-P MOS core structure has been used in design of the VCO. The chip area including pads for testing is $1.2{\times}0.7mm^2$, and the chip area only core for IP in SoC is $1.1{\times}0.4mm^2$. The test results show that there is no special spurs except -63.06dB of the 6MHz reference spurs in the PLL circuitry. There is good phase noise performance like -116.17dBc/Hz in 1MHz offset frequency.

Real-time Implementation of MPEG-4 HVXC Encoder and Decoder on Floating Point DSP (부동 소수점 DSP를 이용한 MPEG-4 HVXC 인코더 및 디코더의 실시간 구현)

  • Kang, Kyeong-ok;Na, Hoon;Hong, Jin-Woo;Jeong, Dae-Gwon
    • The Journal of the Acoustical Society of Korea
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    • v.19 no.4
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    • pp.37-44
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    • 2000
  • In this paper, we described the real-time implementation effort of MPEG-4 audio HVXC (Harmonic Vector eXcitation Coding) algorithm for very low bitrates, which has target applications from mobile communications to Internet telephony, on current high performance floating point TMS320C6701 DSP. We adopted a hardware structure for real-time operation. In order for software optimization, we used C- and assembly-language level optimizations for time-critical functional codes. Utilizing the internal program memory of the DSP as the program cache, the internal data memory overlap technique and DMA functionality, we could get a goal of realtime operation of HVXC codec both at 2 kbit/s and at 4 kbit/s. For an encoder at 2 kbit/s, the optimization ratio to original code is about 96 %. Finally, we got the subjective quality of MOS 2.45 at 2 kbit/s from an informal quality test.

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Voice Conversion using Generative Adversarial Nets conditioned by Phonetic Posterior Grams (Phonetic Posterior Grams에 의해 조건화된 적대적 생성 신경망을 사용한 음성 변환 시스템)

  • Lim, Jin-su;Kang, Cheon-seong;Kim, Dong-Ha;Kim, Kyung-sup
    • Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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    • 2018.10a
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    • pp.369-372
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    • 2018
  • This paper suggests non-parallel-voice-conversion network conversing voice between unmapped voice pair as source voice and target voice. Conventional voice conversion researches used learning methods that minimize spectrogram's distance error. Not only these researches have some problem that is lost spectrogram resolution by methods averaging pixels. But also have used parallel data that is hard to collect. This research uses PPGs that is input voice's phonetic data and a GAN learning method to generate more clear voices. To evaluate the suggested method, we conduct MOS test with GMM based Model. We found that the performance is improved compared to the conventional methods.

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DTMOS Schmitt Trigger Logic Performance Validation Using Standard CMOS Process for EM Immunity Enhancement (범용 CMOS 공정을 사용한 DTMOS 슈미트 트리거 로직의 구현을 통한 EM Immunity 향상 검증)

  • Park, SangHyeok;Kim, SoYoung
    • The Journal of Korean Institute of Electromagnetic Engineering and Science
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    • v.27 no.10
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    • pp.917-925
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    • 2016
  • Schmitt Trigger logic is a gate level design method to have hysteresis characteristics to improve noise immunity in digital circuits. Dynamic Threshold voltage MOS(DTMOS) Schmitt trigger circuits can improve noise immunity without adding additional transistors but by controlling substrate bias. The performance of DTMOS Schmitt trigger logic has not been verified yet in standard CMOS process through measurement. In this paper, DTMOS Schmitt trigger logic was implemented and verified using Magna $0.18{\mu}m$ MPW process. DTMOS Schmitt trigger buffer, inverter, NAND, NOR and simple digital logic circuits were made for our verification. Hysteresis characteristics, power consumption, and delay were measured and compared with common CMOS logic gates. EM Immunity enhancement was verified through Direct Power Injection(DPI) noise immunity test method. DTMOS Schmitt trigger logics fabricated using CMOS process showed a significantly improved EM Immunity in 10 M~1 GHz frequency range.

Audio Stream Delivery Using AMR(Adaptive Multi-Rate) Coder with Forward Error Correction in the Internet (인터넷 환경에서 FEC 기능이 추가된 AMR음성 부호화기를 이용한 오디오 스트림 전송)

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

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BS-PLC(Both Side-Packet Loss Concealment) for CELP Coder (CELP 부호화기를 위한 양방향 패킷 손실 은닉 알고리즘)

  • Lee In-Sung;Hwang Jeong-Joon;Jeong Gyu-Hyeok
    • 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.