• Title/Summary/Keyword: Audio quality evaluation

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A Study of the SPR (Singing Power Ratio) on the Singing Voice in Singing Students (성악 전공 학생의 가칭 시 음성의 SPR(Singing Power Ratio)에 관한 연구)

  • Jo, Sung-Mi;Jeong, Ok-Ran;Lee, Sang-Ouk
    • Speech Sciences
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    • v.11 no.4
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    • pp.121-127
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    • 2004
  • This study attempted to provide a spectrum analysis for quantitative evaluation of singing voice quality of singing students rather than the presence or absence of the singer's formant. The regression analysis was used to analyse the relationship between ringing quality, SPR, and SPP of singing voice of college student subjects majoring in music. This study measured singing. power ratio (SPR) in 41 singing students. Digital audio recordings were made in sung vowels for acoustic analyses. Each sample was judged by 1 experienced singing teacher and 4 voice pathologists on one semantic bipolar 7-point scales (ringing-dull). The results showed that the SPR and SPP had significant correlations with ringing quality. The SPR had a significant relationship with ringing quality on singing voice in singing students. The SPR can be an important quantitative measurement for evaluating singing voice quality.

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Real-time 3D Audio Downmixing System based on Sound Rendering for the Immersive Sound of Mobile Virtual Reality Applications

  • Hong, Dukki;Kwon, Hyuck-Joo;Kim, Cheong Ghil;Park, Woo-Chan
    • KSII Transactions on Internet and Information Systems (TIIS)
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    • v.12 no.12
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    • pp.5936-5954
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    • 2018
  • Eight out of the top ten the largest technology companies in the world are involved in some way with the coming mobile VR revolution since Facebook acquired Oculus. This trend has allowed the technology related with mobile VR to achieve remarkable growth in both academic and industry. Therefore, the importance of reproducing the acoustic expression for users to experience more realistic is increasing because auditory cues can enhance the perception of the complicated surrounding environment without the visual system in VR. This paper presents a audio downmixing system for auralization based on hardware, a stage of sound rendering pipelines that can reproduce realiy-like sound but requires high computation costs. The proposed system is verified through an FPGA platform with the special focus on hardware architectural designs for low power and real-time. The results show that the proposed system on an FPGA can downmix maximum 5 sources in real-time rate (52 FPS), with 382 mW low power consumptions. Furthermore, the generated 3D sound with the proposed system was verified with satisfactory results of sound quality via the user evaluation.

An Audio Watermarking Method Using the Attribute of the Tonal Masker (토널 마스커 특성을 이용한 오디오 워터마킹)

  • 이희숙;이우선
    • The Journal of the Acoustical Society of Korea
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    • v.22 no.5
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    • pp.367-374
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    • 2003
  • In this paper, we propose an audio watermarking method using the attribute of tonal masker. First, the attribute of tonal masker as an audio watermarking attribute is analyzed. According to existing researches, it is possible to be imperceptible modulation for the energies of the frequencies that compose a tonal masker. And when the relation between the tone energy and the left or right frequency energy after various signal processing is compared with the one before the processing, very few changes are showed. We propose an audio watermarking method using these attributes of tonal masker. A watermark bit is embedded by the modulation of the difference between the two neighboring frequency energies of a tone. In the detection, the modulated the tonal masker is searched using the key wed in the embedding without original audio and the embedded watermark bit is detected. After each attack of noise insertion, band-pass filtering, re-sampling, compression, echo transform and equalization, the detection error ratios of the proposed method were average 0.11%, 1.26% for Classics and Pops. And the SDG(Subjective Diff-Grades) scale evaluation of the sound quality of the watermarked audio result in the average SDG -0.31.

Adaptive TCX Windowing Technology for Unified Structure MPEG-D USAC

  • Lee, Tae-Jin;Beack, Seung-Kwon;Kang, Kyeong-Ok;Kim, Whan-Woo
    • ETRI Journal
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    • v.34 no.3
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    • pp.474-477
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    • 2012
  • The MPEG-D unified speech and audio coding (USAC) standardization process was initiated by MPEG to develop an audio codec that is able to provide consistent quality for mixed speech and music contents. The current USAC reference model structure consists of frequency domain (FD) and linear prediction domain (LPD) core modules and is controlled using a signal classifier tool. In this letter, we propose an LPD single-mode USAC structure using an adaptive widowing-based transform-coded excitation module. We tested our system using official test items for all mono-evaluation modes. The results of the experiment show that the objective and subjective performances of the proposed single-mode USAC system are better than those of the FD/LPD dual-mode USAC system.

Design and Implementation of a Distributed Audio/Video Stream Service Framework based on CORBA (CORBA 기반의 분산 오디오/비디오 스트림 서비스 프레임워크의 설계 및 구현)

  • Kim, Jong-Hyeon;No, Yeong-Uk;Jeong, Gi-Dong
    • The KIPS Transactions:PartA
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    • v.9A no.2
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    • pp.207-216
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    • 2002
  • This paper present a design and implementation of a distributed audio, Video stream service framework based on CORBA for efficient processing and control of audio/video stream. We design software components which support processing, control and transmission of audio/video streams as distributed objects. For optimization of stream transmission performance, we separate the transmission path of control data and media data. Distributed objects are defined by IDL and implemented using JAVA. And device dependent facilities like media capturing, playing and communication channels are implemented using JMF (Java Media Framework) components. We show a connection establishment and control procedure of streams communication. And for evaluation, we implement a test system and experiment a system performance. Our experiments show that test system has somewhat longer connection latency time compared to TCP connection establishment, but has optimized media transmission time compared to CORBA IIOP. Also test system show acceptable service quality of media transmission.

Performance Evaluation of MCLT-based Audio Watermark in DTV System (DTV 시스템에서의 MCLT 기반 오디오 워터마크 성능 평가)

  • Jeong, Youngho;Lee, Misuk;Lee, Taejin;Kim, Huiyong
    • Proceedings of the Korean Society of Broadcast Engineers Conference
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    • 2017.06a
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    • pp.219-222
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    • 2017
  • 본 논문에서는 DTV 시스템을 대상으로 PN 시퀀스를 이용한 MCLT(Modulated Complex Lapped Transform) 기반 오디오 워터마크 알고리즘에 대한 BER 및 PEAQ(Perceptual Evaluation of Audio Quality) 성능 평가를 통해 오디오 신호 압축에 대한 워터마크의 강인성 및 워터마크 삽입에 따른 오디오 품질 열화 정도를 분석하였다. 이를 위해 오디오 신호 특성을 고려한 프로그램 장르별 시험용 방송 콘텐츠를 제작하고, Lab. Test 를 위한 DTV 송수신 시스템을 구축하였다. 오디오 인코딩 비트율 변화에 따른 성능 평가 결과, 광고 콘텐츠를 제외한 평균 BER(%)에서 192kbps 비트율이 128kpbs 비트율에 비해 0.0767 더 우수한 성능을 보였다. 오디오 워터마크 삽입에 따른 객관적 음질 평가에서는 PEAQ 점수가 약 -0.2 로 원래 오디오 신호와의 품질 차이가 매우 작은 것으로 나타났으며, 또한 DTV 시스템상의 신호 압축에 의해 발생하는 오디오 신호의 품질 저하 이외에 워터마크 삽입으로 인한 추가적인 음질 저하는 거의 발생하지 않는 것으로 분석되었다.

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Design and Implementation of the Evaluation Framework for Decentralized Multimedia Streaming Services

  • Park, Sangsoo
    • Journal of the Korea Society of Computer and Information
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    • v.25 no.9
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    • pp.91-100
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    • 2020
  • This paper presents an evaluation framework for prototyping multimedia streaming services including audio and video in a distributed and/or decentralized storage that can evaluate service quality and performance under various network conditions. The evaluation framework focuses on important indicators which measure and improve service quality by applying decentralized storage to multimedia streaming services that can mimic the scalability of the existing server-client software architecture and the issue of a single point of failure. The integrated framework not only measures performance indicators for evaluating the quality and performance of multimedia streaming on open source based multimedia content streaming services, but also adjusts network quality using network virtualization technology for comprehensive evaluations. The experimental results show that the integrated framework has low overhead in building and operating a decentralized storage with multimedia streaming services on a single host computer which validates the scalability of the developed framework.

A Study on Measurement Method of Audio Playback Time for Standardization of Wireless Earphone Quality (무선이어폰 품질 표준화를 위한 오디오 재생 시간 측정법에 관한 연구)

  • HAN, Munhwan;Jeong, Inho
    • The Journal of the Institute of Internet, Broadcasting and Communication
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    • v.22 no.1
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    • pp.141-151
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    • 2022
  • Wireless earphones are products that are consumed together with smart devices (mobile phones, etc), and there is no twisting and convenience compared to general earphones. However, due to the lack of information on the quality of wireless earphones, consumers tend to purchase products based on brand awareness, and manufacturers deliver information to consumers based on different standards for each product due to the lack of standards for measurement methods for quality evaluation. In particular, the playback time of wireless earphones is a factor that can directly affect consumers' purchases, so it is necessary to prepare a standardized test method to properly measure it. This paper introduces the current status of wireless earphones and related standard trends, and proposes a method for measuring the audio playback time of wireless earphones developed through this. In addition, this measurement method will be proposed as an international standard (IEC) after being established as the national standard, the Korean Industrial Standard (KS).

Performance Evaluation of Six Jitter Control Algorithms for Improving Audio Quality (오디오 품질을 개선하기 위한 6개의 Jitter Control 알고리즘의 성능 분석)

  • 나승구;유홍준;안종석;이태진
    • Proceedings of the Korean Society of Broadcast Engineers Conference
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    • 2000.11b
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    • pp.29-35
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    • 2000
  • 음성 데이터의 패킷 지터(jitter)가 심할수록 오디오 플레이어가 오디오 데이터를 자연스럽게 재생하지 못하기 때문에 사용자는 원래의 음성을 거의 알아들을 수 없게 된다. 이 문제점을 해결하기 위하여 오디오 수신자는 전송 받은 오디오 데이터를 바로 재생하지 않고 재생시간을 지연시키는 방법을 사용한다. 본 연구자의 조사에 의하면 이러한 재생시간을 지연하는 대표적인 지터 컨트롤 알고리즘으로 6가지 방식이 제안되고 있다. 그 중 세 가지는 NeVot, Vat, Open H.323 프로그램 등에 구현되어 실제로 사용되고 있다 본 논문에서는 이들 6가지의 모델의 지터 컨트롤 알고리즘의 특성을 알아보고 어느 알고리즘이 효율적인지 알아보기 위해 현재 인터넷의 성능을 파악하고 이를 기초로 제안된 6가지 알고리즘 중 어느 것이 가장 효율적인가를 파악하여 오디오의 음질을 개선하기 위한 방법을 제시하고자 한다.

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Analysis and Evaluation of PEAQ : Objective Method for Perceived Audio Quality Measurement (객관적 음질 평가를 위한 PEAQ의 성능 평가 및 분석)

  • Park Se-Hyoung;Ryu Seung-Wan;Park Jeong-Yeol;Shin Jae-Ho
    • 한국정보통신설비학회:학술대회논문집
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    • 2003.08a
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    • pp.234-239
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    • 2003
  • 디지털방송, DAB 등과 같은 디지털 오디오 방송 서비스를 위한 디지털 시스템을 설계하기 위해서는 오디오 음질을 평가하기 위한 방법이 필수적이다. 기존의 방식은 인간의 귀를 이용한 주관적 방식을 이용함으로서 많은 시간과 비용을 들이게 되며, 음질평가를 하는 사람의 주관적 의견에 많이 좌우하게 된다. 그러나 최근 ITV-R에서는 오디오 음질의 객관적 평가를 위한 BS.1387(PEAQ)를 제안함으로 많은 시간과 비용을 절감하고 신뢰할 수 있는 결과를 얻게 되었다. PEAQ는 인간의 귀에서의 신호의 처리과정과 인식과정을 심리음향모델과 인식모델로 분리하여 구성함으로써 주관적 평가의 SDG(Subjective Difference Grade)에 대응하는 ODG(Objective Difference Grade)를 구하게 된다. 본 논문에서는 이러한 PEAQ의 심리음향 모델과 인식 모델을 원리와 과정을 평가 분석하였다.

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