• 제목/요약/키워드: Audio Compression

검색결과 135건 처리시간 0.024초

A Robust Audio Fingerprinting Method Based on Segmentation Boundaries

  • Seo, Jin-Soo
    • 한국음향학회지
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    • 제31권4호
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    • pp.260-265
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    • 2012
  • A robust audio fingerprinting method is presented based on segmentation boundaries. In order to obtain robustness against linear speed changes, fingerprint extraction and matching are synchronized with the segmentation boundaries. Experimental results show that the proposed method is also robust against other common audio processing steps including low bit-rate compression, equalization, and time-scale modification.

저작권자의 음성 삽입을 위한 영상 워터마킹 방법 (An Image Watermarking Method for Embedding Copyrighter's Audio Signal)

  • 최재승;김정화;고성식
    • 한국음향학회지
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    • 제24권4호
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    • pp.202-209
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    • 2005
  • 디지털 미디어와 통신 네트워크의 급속한 발전으로 지적소유권 보호 기술이 절실히 요구된다. 본 논문에서 영상컨텐츠에 소유자의 음성신호를 삽입할 수 있는 새로운 영상 워터마킹 방법을 제안한다. 제안한 방법은 삽입할 워터마크로 음성신호를 이용하기 때문에 청각적으로 소유권을 주장하는데 매우 유리하다 그리고 LBX (Linear Bit expansion)에 의해 인터리빙하여 음성 워터마크를 삽입하기 때문에 공격에 의해 변형되거나 특히 제거된 음성신호를 복구할 수 있는 이점을 가진다. 본 방법은 다음세가지 기본 절차를 포함한다. 첫째, 아날그 소유자음성 신호를 PCM에 의해 부호화하고 새로운 디지털 워터마크를 생성한다. 둘째, 제안한 LBX에 의해 음성 워터마크를 인터리빙한다. 마지막으로 영상 컨텐츠를 이산 Haar 웨이브렛변환 (DHWT) 하고 저주파 영역을 마킹공간으로 하여 인터리빙 된 음성워터마크를 삽입한다. 실험 결과 본 연구에서 제안한 소유자 음성신호의 워터마크 삽입방법은 기존 워터마크 정보보다 눈에 보이지 않게 많은 정보량을 삽입할 수 있고 표준영상압축방식인 JPEG 손실압축과 특히 영상의 일부가 제거되는 영상절단과 영상회전 공격에 대해 강인성을 강건히 할 수 있었다.

Design on MPEC2 AAC Decoder

  • NOH, Jin Soo;Kang, Dongshik;RHEE, Kang Hyeon
    • 대한전자공학회:학술대회논문집
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    • 대한전자공학회 2002년도 ITC-CSCC -3
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    • pp.1567-1570
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    • 2002
  • This paper deals with FPGA(Field Programmable Gate Array) implementation of the AAC(Advanced Audio Coding) decoder. On modern computer culture, according to the high quality data is required in multimedia systems area such as CD, DAT(Digital Audio Tape) and modem. So, the technology of data compression far data transmission is necessity now. MPEG(Moving Picture Experts Group) would be a standard of those technology. MPEG-2 AAC is the availableness and ITU-R advanced coding scheme far high quality audio coding. This MPEG-2 AAC audio standard allows ITU-R 'indistinguishable' quality according to at data rates of 320 Kbit/sec for five full-bandwidth channel audio signals. The compression ratio is around a factor of 1.4 better compared to MPEG Layer-III, it gets the same quality at 70% of the titrate. In this paper, for a real time processing MPEG2 AAC decoding, it is implemented on FPGA chip. The architecture designed is composed of general DSP(Digital Signal Processor). And the Processor designed is coded using VHDL language. The verification is operated with the simulator of C language programmed and ECAD tool.

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MPEG Audio을 위 한 MDCT/IMDCT의 설계에 관한 연구 (A Study on the Design of MDCT/IMDCT for MPEG Audio)

  • 김정태;방기천;이강현
    • 대한전자공학회:학술대회논문집
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    • 대한전자공학회 1999년도 하계종합학술대회 논문집
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    • pp.530-533
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    • 1999
  • During the last decade, high quality digital audio has essentially replaced analog audio. During this period, digital audio have applied many application areas of the info-industry. These applications have created a demand for high quality digital audio. In audio compression, the methods using human auditory nervous properties are used and introduced from psychoacoustical model utilized perceptual audio coding unable to code above the limitation of human perception. The discussion concentrates on architectures and applications of those techniques which utilize psychoacoustical models to exploit efficiently masking characteristics of the human receiver. In this paper, the designed MDCT/IMBCT as a standard of current MPEG is implemented onto FPGA.

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음성신호의 압축원리를 이용한 사운드 마스킹 효과로 음향 환경 최적화 (Optimize the Acoustic Environment Using a Sound Masking Effects of the Audio Signal Compression Principle)

  • 안숙향
    • 한국전기전자재료학회논문지
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    • 제28권11호
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    • pp.748-751
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    • 2015
  • Sound Masking System technology as by sound the same on all bands and artificially generates a constant sound shield People want to hear or recognize the people with the noise generated from the interior of the way. Prevent hearing or prevent recognition by using the technology to control the audible frequency band Continue to emit constant and uniform shielding sound audible frequency band Even the security content of speech (20 Hz~20 KHz). That interception laser eavesdropping, internal solicitations, during recording Or delay the decoding was a result of the effect of interference calculated Experience noise disturbance index is applied around the Stress Index is the average index is 10.16 was a luxury for the average index is then applied to the index 3.07 Noise is significantly lower stress level has improved noise conditions.

MPEG-2 오디오를 위한 MDCT 설계에 관한 연구 (A Study on the MDCT Design for MPEG-2 Audio)

  • 김정태;구대성;이강현
    • 대한전자공학회:학술대회논문집
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    • 대한전자공학회 2000년도 추계종합학술대회 논문집(3)
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    • pp.97-100
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    • 2000
  • The most important technology is the compression methods in the multimedia society. Audio files are rapidly propagated through internet. MP-3(MPEG-1 Layer3) is offered to CD tone quality in 128kbps, but 64kbps below tone-quality is abruptly down. On the other hand, MPEG-II AAC (Advanced Audio Coding) is not compatible with MPEG-I, but AAC has a high compression ratio 1.4 times better than MP-3 and it has max. 7.1 channel and 96KHz sampling rate. In this paper, we designed the optimized MDCT (Modified Discrete Cosine Transform) that could decrease the capacity of enormous computation and could increase the processing speed in the MPEG-2 AAC encoder.

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오디오 워터마크를 이용한 실시간 방송동기화시스템의 구현 (The Implemetation of Real-time Broadcast Synchronizing System Using Audio Watermark)

  • 신동환;김종원
    • 대한전기학회논문지:시스템및제어부문D
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    • 제54권12호
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    • pp.716-722
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    • 2005
  • In this paper, we propose the audio watermarking algorithm based on the critical band of HAS(human auditory system) without audibly affecting the quality of the watermarked audio and implement the detecting algorithm on the BSS(broadcast synchronizing system) for testing the proposed algorithm. According to the audio quality test, the SNR(signal to noise ratio) of the watermarked audio objectively is 66dB above. In the robustness test, the proposed algorithm can detect the watermark more than $90\%$ from various compression(MP3, AAC), A/D and D/A conversions, sampling rate conversions and especially asynchronizing attacks. The BSS automatically switches the programs between the key station and the local station in broadcasting system. The result of reliability test of implemented system by using the real broadcasting audio has no false positive error during 30 days. Because of detecting once processing per 0.5 second, we can judge that the false positive error does not occur.

MPEG Surround Extension Technique for MPEG-H 3D Audio

  • Beack, Seungkwon;Sung, Jongmo;Seo, Jeongil;Lee, Taejin
    • ETRI Journal
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    • 제38권5호
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    • pp.829-837
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    • 2016
  • In this paper, we introduce extension tools for MPEG Surround, which were recently adopted as MPEG-H 3D Audio tools by the ISO/MPEG standardization group. MPEG-H 3D Audio is a next-generation technology for representing spatial audio in an immersive manner. However, considerably large numbers of input signals can degrade the compression performance during a low bitrate operation. The proposed extension of MPEG Surround was basically designed based on the original MPEG Surround technology, where the limitations of MPEG Surround were revised by adopting a new coding structure. The proposed MPEG-H 3D Audio technologies will play a pivotal role in dramatically improving the sound quality during a lower bitrate operation.

Design and Development of T-DMB Multichannel Audio Service System Based on Spatial Audio Coding

  • Lee, Yong-Ju;Seo, Jeong-Il;Beack, Seung-Kwon;Jang, Dae-Young;Kang, Kyeong-Ok;Kim, Jin-Woong;Hong, Jin-Woo
    • ETRI Journal
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    • 제31권4호
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    • pp.365-375
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    • 2009
  • In this paper, a terrestrial digital multimedia broadcasting (T-DMB) multichannel audio broadcasting system based on spatial audio coding is presented. The proposed system provides realistic multichannel audio service via T-DMB with a small increase of data rate as well as backward compatibility with the conventional stereo-based T-DMB player. To reduce the data rate for additional multichannel audio signals, we compress the multichannel audio signals using the sound source location cue coding algorithm, which is an efficient parametric multichannel audio compression technique. For compatibility, we use the dependent property of an elementary stream descriptor, and this property should be ignored in a conventional T-DMB player. To verify the feasibility of the proposed system, we implement the T-DMB multichannel audio encoder and a prototype player. We perform a compatibility test using the T-DMB multichannel audio encoder and conventional T-DMB players. The test demonstrates that the proposed system is compatible with a conventional T-DMB player and that it can provide a promisingly rich audio service.

Recursive 구조를 이용한 MPEG-2 BC/AAC 오디오 공용 합성 필터 (A Common Synthesis Filter for MPEG-2 BC/AAC Audio Using Recursive Structure)

  • 강명수;박세기;오신범;이채욱
    • 한국통신학회논문지
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    • 제29권6C호
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    • pp.874-882
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    • 2004
  • MPEG 오디오 표준은 고품질 오디오를 제공하기 위해 인간 청각계의 지각현상을 이용한 오디오 압축 알고리듬이다. 다수의 표준안 중 MPEG-2 BC와 MPEG-2 AAC에 대한 공용 복호화 시스템은 아직 발표되어 있지 않다. 본 논문에서는 BC/AAC의 공용 시스템을 구축하여 복호화 과정에서 공통된 구조로 연산을 수행하고 면적감소, 비용절감 등을 목표로 한다. 본 논문에서는 공통된 연산구조를 가지기 위해서 recursive 구조를 이용하여 복호화 과정중 합성 필터링 과정을 공통된 구조로 연산을 수행할 수 있는 공용 합성 필터 구조를 제안한다. 제안한 구조는 연산시간의 개선을 위하여 고속 알고리듬을 적용하였으며 기존의 MPEG-2 AAC 함성 필터링에 대한 recursive 알고리듬을 2단 recursive 구조로 변경, 적용하였고 그 결과 같은 구조로 MPEG-2 BC의 서브밴드 합성필터링도 연산 가능함을 보였다.