• Title/Summary/Keyword: MPEG-2 AAC

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DSP based implementation of MPEG-2 AAC decoder (MPEG-2 AAC 디코더의 DSP 구현에 관한 연구)

  • 정종훈;김정근;이재식;장태규;장흥엽
    • Proceedings of the IEEK Conference
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    • 2001.09a
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    • pp.481-484
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    • 2001
  • 본 논문에서는 MPEG-2 AAC 디코더의 DSP구현에 관한 연구결과로서 IS0/1EC 13818-7 표준에 의거 구현된 MPEC-2 AAC 디코더의 각 세부 기능블럭들의 구성 및 동작원리에 대하여 요약 기술하고. DSP연산에 최적화된 연산구조의 연구를 바탕으로 16bit 고정소수점 연산구조를 가지는 DSP상에서 구현된 MPEG-2 AAC 디로더의 시스템의 하드웨어 및 소프트웨어 구성에 관하여 간략한 기술하였다. 구성된 디코더의 성능평가를 통하여 MPEC-2 AAC 비트스트림을 디코딩하기 위하여 필요로 하는 연산량 및 소요 메모리의 양을 측정하고, 디코더 성능의 중요 척도인 음질평가를 수행하였다. 수행방법으로서 conformance test에 의거하여 PSNR을 측정함으로써 객관적인 성능 지표의 제시와 함께, 주관적인 음질 평가도 병행하여 수행하였다.

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Architecture Design for MPEG-2 AAC Filter bank Decoder using Recursive Structure (Recursive 구조를 이용한 MPEG-2 AAC 복호화기의 필터뱅크 구현)

  • 박세기;강명수;오신범;이채욱
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.29 no.6C
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    • pp.865-873
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    • 2004
  • MPEG-2 Advanced Audio Coding(AAC) is widely used in the multi-channel audio compression standards. And it combines hi인-resolution filter bank prediction techniques, and Huffman coding algorithm to achieve the broadcast-quality audio level at very low data rates. The forward and inverse modified discrete transforms which are operated in the encoder and the decoder of the filter bank need many computations. In this paper, we propose suitable recursive structure at IMDCT processing for MPEG-2 AAC real-time decoder. We confirm the memory, the computation speed and complexity of the proposed structure.

Hardware design of the MPEG-2 AAC Decoder Module (MPEG-2 AAC 복호화기 모들의 하드웨어 설계)

  • 우광희;김수현;홍민철;차형태
    • Journal of the Institute of Convergence Signal Processing
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    • v.2 no.1
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    • pp.113-118
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    • 2001
  • In this paper, we implement modules of the MPEG-2 AAC decoder using VHDL. Tools of Huffman decoder, inverse quantizer and high-density filter bank which are necessary for the AAC decoder. We designed the high speed Huffman decoder using the method of octal tree search algorithm, and reduced computational time of filter bank using IFFT. Also, we use table of computation result for an exponential calculation of Inverse quantizer in fixed-point hardware, and reduced the size of table using linear interpolation. Modules implemented by hardware through optimization work in real time at low clock frequency are possible to reduce the system size.

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A Study On the Design of a Floating Point Unit for MPEG-2 AAC Decoder (MPEG-2 AAC 복호기를 위한 부동소수점유닛 설계에 관한 연구)

  • 구대성;김필중;김종빈
    • Journal of the Institute of Electronics Engineers of Korea TE
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    • v.39 no.4
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    • pp.355-355
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    • 2002
  • In this paper, we designed a FPU(floating point unit) that it is very important and requires of high density when digital audio is designed. Almost audio system must support the multi-channel and required for high quality. A floating point arithmetic function in MPEG-2 AAC that implemented by hardware is able to realtime decoding when DSP realization. The reason is that MPEG-2 AAC is compatible to the Audio field of MPEG-4 and afterwards. We designed a FPU by hardware to increase the speed of a floating point unit with much calculation part in the MPEG-2 AAC Decoder. A FPU is composed of a multiplier and an adder. A multiplier used the Radix-4 Booth algorithm and an adder adopted 1's complement method for speed up. A form of a floating point unit has 8bit of exponent part and 24bit of mantissa. It's compatible with the IEEE single precision format and adopted a pipeline architecture to increase the speed of a processor. All of sub blocks are based on ISO/IEC 13818-7 standard. The algorithm is tested by C language and the design does by use of VHDL(VHSIC Hardware Description Language). The maximum operation speed is 23.2MHz and the stable operation speed is 19MHz.

Channel Expansion Technology in MPEG Audio (MPEG 오디오의 채널 확장 기술)

  • Pang, Hee-Suk
    • Journal of Broadcast Engineering
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    • v.16 no.5
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    • pp.714-721
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    • 2011
  • MPEG audio uses the masking effect, high frequency component synthesis based on spectral band replication, and channel expansion based on parametric stereo for efficient compression of audio signals. In this paper, we present an overview of the state-of-the-art channel expansion technology in MPEG audio. We also present technical overviews and application examples to broadcasting services for HE-AAC v.2, MPEG Surround, spatial audio object coding (SAOC), and unified speech and audio coding (USAC) which are MPEG audio codecs based on the channel expansion technology.

Internet Audio Broadcasting Technology Using MPEG-2 AAC Streaming (MPEG-2 AAC 스트리밍을 이용한 인터넷 오디오 방송기술)

  • 이태진;홍진우
    • The Journal of the Acoustical Society of Korea
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    • v.21 no.2
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    • pp.93-101
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    • 2002
  • This paper presents the Internet audio broadcasting technology based on the streaming technology. In this paper, we choose the MPEG-2 AAC for multimedia data, and for the streaming of this data we use RTP/RTCP protocol. We use RTSP protocol for the control of streaming data and TCP/IP for the exchange of information between server and client. By using all of these protocols and MPEBG-2 AAC, we explain the implementation method for the unicast/multicast streaming server/client system. Our system was tested by ETRI intranet, which is connected by 2000 researchers. Experimental result show that our system can be process the packet loss and jitter by retransmission and variable length buffer. Multicast streaming server can be used for the audio broadcasting service inside the company, unicast streaming server can be used for the AOD (Audio On Demand) service.

Multi-symbol Accessing Huffman Decoding Method for MPEG-2 AAC

  • Lee, Eun-Seo;Lee, Kyoung-Cheol;Son, Kyou-Jung;Moon, Seong-Pil;Chang, Tae-Gyu
    • Journal of Electrical Engineering and Technology
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    • v.9 no.4
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    • pp.1411-1417
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    • 2014
  • An MPEG-2 AAC Huffman decoding method based on the fixed length compacted codeword tables, where each codeword can contain multiple number of Huffman codes, was proposed. The proposed method enhances the searching efficiency by finding multiple symbols in a single search, i.e., a direct memory reading of the compacted codeword table. The memory usage is significantly saved by separately handling the Huffman codes that exceed the length of the compacted codewords. The trade-off relation between the computational complexity and the amount of memory usage was analytically derived to find the proper codeword length of the compacted codewords for the design of MPEG-2 AAC decoder. To validate the proposed algorithm, its performance was experimentally evaluated with an implemented MPEG-2 AAC decoder. The results showed that the computational complexity of the proposed method is reduced to 54% of that of the most up-to-date method.

The Audio Watermarking method Using the MPEG-2 AAC Psychoacoustic Model (MPEG-2 AAC 심리음향 모델을 이용한 오디오 워터마킹 기법)

  • 성종수;강상구;신재호
    • Proceedings of the IEEK Conference
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    • 1999.06a
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    • pp.716-719
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    • 1999
  • In this Paper, we Present a method for embedding digital watermarks into digital audio signals. The watermarking must be imperceptible and should be robust to attacks, such as filtering and compression etc. In our method, we adaptively embedded the watermarks changing the scale factor using the spread spectrum and MPEG-2 AAC psychoacoustic model.

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Design on MPEC2 AAC Decoder

  • NOH, Jin Soo;Kang, Dongshik;RHEE, Kang Hyeon
    • Proceedings of the IEEK Conference
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    • 2002.07c
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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-D USAC: Unified Speech and Audio Coding Technology (MPEG-D USAC: 통합 음성 오디오 부호화 기술)

  • Lee, Tae-Jin;Kang, Kyeong-Ok;Kim, Whan-Woo
    • The Journal of the Acoustical Society of Korea
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    • v.28 no.7
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    • pp.589-598
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    • 2009
  • As mobile devices become multi-functional, and converge into a single platform, there is a strong need for a codec that is able to provide consistent quality for speech and music content MPEG-D USAC standardization activities started at the 82nd MPEG meeting with a CfP and approved WD3 at the 88th MPEG meeting. MPEG-D USAC is converged technology of AMR-WB+ and HE-AAC V2. Specifically, USAC utilizes three core codecs (AAC ACELP and TCX) for low frequency regions, SBR for high frequency regions and the MPEG Surround tool for stereo information. USAC can provide consistent sound quality for both speech and music content and can be applied to various applications such as multi-media download to mobile device Digital radio Mobile TV and audio books.