• Title/Summary/Keyword: High-Fidelity Audio

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New High-Resolution Encoding System having Backward Compatibility with CDDA (상용 CDDA와 하위 호환성을 가지는 새로운 고해상도 부호화방식)

  • Moon Dong-Wook;Kim Lark-Kyo
    • The Transactions of the Korean Institute of Electrical Engineers D
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    • v.54 no.5
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    • pp.327-329
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    • 2005
  • Conventional CDDA(Compact Disc Digital Audio) system has limitation which means that bandwidth and resolution of the sign릴 are determined by the sampling frequency and quantization bit, 44.1kHz and 16 bit respectively. Though, new medium such as DVD-audio is developed for high-resolution audio recording, it has high complexity and difficulty in manufacturing process. So, CDDA is a widely used medium for high fidelity audio yet. In this paper, we design a new encoding system for high-resolution audio signal which has backward compatible with conventional CDDA. By evaluating for the encoding and decoding process. we verify the availability of our proposed system.

A Proposal for High-Resolution Encoding System with Backward Compatibility in CDDA (상용 CDDA와 하위 호환성을 가지는 고해상도 부호화방석의 제안)

  • Moon, Dong-Wook;Kim, Lark-Kyo;Nam, Moon-Hyun
    • Proceedings of the KIEE Conference
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    • 2004.11c
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    • pp.150-152
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    • 2004
  • Conventional CDDA (Compact Disc Digital Audio) system has limitations come from sampling frequency and quantization bit, 44.1kHz and 16 bit respectively. So, new medium is developed for high-resolution audio recording, like as DVD-audio etc. But CDDA is a widely used medium for high fidelity audio yet, because new medium has complexity and difficulty in manufacturing system. In this paper, we design a new encoding system for high-resolution audio signal. The system is backward compatible with conventional CDDA. By evaluation for encoding and decoding process, we describe practicability of our proposal system.

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High Frequency Enhancement of Sound Using Wavelet Transform

  • Yoon Won-Jung;Lee Kang-Kyu;Park Kyu-Sik
    • Proceedings of the IEEK Conference
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    • summer
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    • pp.233-236
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    • 2004
  • This paper proposes new method for the enhancement of nonexistent high frequency spectral contents from low sample rate audio signal. For example, Due to the protocol constraint, the audio bandwidth of MP3 is restricted to 16Khz. Although band-restricted MP3 audio provide savings of storage space and network bandwidth, it suffers a major problem of a loss in high frequency fidelity such as localization, ambient information, and bright nature of audio. This paper provides a new mathematical analysis for the adaptive estimation of the high frequency contents based on the nature of the input low sample rate audio. Proposed method can be worked globally to any kind of audio such as speech and music that are restricted by sampling rate and bandwidth.

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The Analysis of Equaizer for Improving Sound Quality of Samrtphones (스마트폰의 음질 향상을 위한 Equalizer 분석)

  • Lee, Myung-hwan;Ryu, Chang-su
    • Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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    • 2013.05a
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    • pp.190-193
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    • 2013
  • Before Smartphone releases, playing music form previous cellphone is very limited, so we need mp3 player instead. However, when Smartphone releases, a function of playing multimedia became one of the important things in Samrtphone including all functions of mp3 player. Even though it is easier to use since you can play music and multimedia in one device, it still has problem issues on the sound quality. This thesis will discuss about functions of EQ from Smartphone music players. After that, we will to achieve high fidelity sound through balance adjustment of Right Mark Audio Analyzer program.

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Digital Filter Design for the DSD Encoder with Multi-rate PCM Input (PCM 입력의 DSD 인코더를 위한 디지털 필터 설계)

  • Moon, Dong-Wook;Kim, Lark-Kyo
    • Proceedings of the KIEE Conference
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    • 2005.05a
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    • pp.170-172
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    • 2005
  • The DSD(Direct Stream Digital) encoder, which is a standard for SACD(Super Audio Compact Disc) proposed by Sony and philips, use 1 bit representation with a sampling frequency of 2.8224 MHz (64 $\times$ 44.1 kHz). For multi-rate PCM (Pulse Code Modulation) input like as 48/96/192 kHz, a external sample-rate converter is necessary to the DSD encoder. This paper has been proposed a digital filter structure composed of sample-rate converter and interpolation filter for the DSD encoder with multi-rate (48/96/192 kHz) PCM input. without a external sample-rate converter.

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A Study on the Assessment Factors for Car Audio Sound Quaility (자동차용 오디오의 실내음질 평가인자에 관한 연구)

  • 오양기;한명호
    • Journal of KSNVE
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    • v.9 no.5
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    • pp.991-997
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    • 1999
  • It is getting higher the demand on a good sound in a car. Like some other sensory qualities, achieving a good sound is still treated as a black-box followed by the individual taste of tuning engineers. This study aims to define the attributes of a good car audio sound and the assessment factors affecting the sound quality. With a variety of surveys on professional listening groups and car audio installers, 26 most frequently used vocabularies are selected as typical presentations of car audio sound quality. After repetitions of listening tests with deliberately selected test CD, pre-recorded 15 different digital car-recordings, headphones replay system, and three different listening groups, it is proved that there are at least 5 different assessment factors which significantly affect the car audio sound quality. They are high fidelity, spectral balance, tonal accuracy, sound stage and acoustic fault.

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An Improved Digital Filter Design for the DSD Encoder with Multi-rate PCM Input (다중 표본화율의 PCM 입력을 위한 개선된 DSD 인코더용 디지털 필털 설계)

  • Moon, Dong-Wook;Kim, Lark-Kyo
    • Proceedings of the KIEE Conference
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    • 2005.10b
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    • pp.358-360
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    • 2005
  • The DSD(Direct Stream Digital) encoder, which is a standard for SACD(Super Audio Compact Disc) proposed by Sony and philips, uses 1 bit representation with a sampling frequency of 2.8224MHz (64${\times}$44.1kHz). For multi-rate PCM (Pulse Code Modulation) input such as 8${\sim}$192kHz, a external sample-rate converter is necessary to the DSD encoder. This paper has been proposed a digital mter structure composed of sample-rate converter and interpolaton filter for the DSD encoder with multi-rate (8${\sim}$192kHz) PCM input, without a external sample-rate converter.

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An Implementation of an ARM Platform based MP3 Sound Enhancement System (ARM 플랫폼 기반의 MP3 오디오 음질 향상 시스템 구현)

  • Oh, Sang-Hun;Park, Kyu-Sik
    • Journal of the Institute of Electronics Engineers of Korea SP
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    • v.44 no.1
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    • pp.70-75
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    • 2007
  • In order to mitigate the problems in storage space and network bandwidth for the full CD quality audio with 44.1 kHz sampling rate, current existing digital audio is always restricted by sampling rate and bandwidth. This kind of restriction normally can be resolved by using low bit rate audio codec such as MP3, OGG, and AAC. However it suffers a major problem such as a loss of high frequency fidelity. This high frequency loss will reproduce only the band-limited low-frequency part of audio in the standard CD-quality audio. In general, the high frequency contents of audio have lots of information such as localization and ambient information, and bright nature of audio. The purpose of this paper is to implement on ARM platform system that can effectively estimate and compensate the missing high frequency contents of MP3 audio. From the experimental results with spectrum analysis and listening test, we confirm the superiority of the proposed algorithms for MP3 audio quality enhancement.

An Implementation of Sound Enhanced MPEG-1 Audio Decoder on Embedded OS Platform (음질향상 알고리즘을 내장한 MPEG-1 오디오 디코더의 Embedded OS 플랫폼에의 구현)

  • Hong, Sung-Min;Park, Kyu-Sik
    • Journal of Korea Multimedia Society
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    • v.10 no.8
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    • pp.958-966
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    • 2007
  • In this paper, we implement a sound-enhanced MPEG-1 audio decoder on embedded OS Platform. Low bit rate lossy audio codecs such as MP3, OGG, and AAC for mitigating the problems in storage space and network bandwidth suffer a major common problem such as a loss of high frequency fidelity of audio signal. This high frequency loss will reproduce only a band-limited low-frequency part of audio in the standard CD-quality audio. In order to overcome this problem, we embedded a sound enhancement algorithm into the MPEG-1 audio decoder and then the algorithms optimized according to the characteristic of the MPEG-1 audio layer I, II, III were implemented on an embedded OS platform. From the experimental results with spectrum analysis and listening test, we confirm the superiority of the proposed system compared to the standard MPEG-1 audio decoder.

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Neural perceptron-based Training and Classification of Acoustic Signal

  • Kim, Yoon-Ho
    • Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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    • v.9 no.1
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    • pp.1133-1136
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    • 2005
  • The MPEG/audio standard results from three years of co-work by an international committe of high-fidelity audio compression experts in the Moving Picture Experts Group (MPEG/audio). The MPEG standard is rigid only where necessary to ensure interoperability. In this paper, a new approach of training and classification of acoustic signal is addressed. This is some what a fields of application aspects rather than technonical problems such as MPEG/codec, MIDI. In preprocessing, acoustic signal is transformmed using DWT so as to extract a feature parameters of sound such as loudness, pitch, bandwidth and harmonicity. these accoustic parameters are exploited to the input vector of neural perceptron. Experimental results showed that proposed approach can be used for tunning the dissonance chord.

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