• 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.

• ETRI Journal
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• v.31 no.3
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• pp.336-338
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• 2009

This letter proposes a complexity reduction method to speed up the noiseless decoding of a bit-sliced arithmetic coding (BSAC) decoder. This scheme fully utilizes the group of consecutive arithmetic-coded symbols known as the decoding band and the significance tree structure sorted in order of significance at every decoding band. With the same audio quality, the proposed method reduces the number of calculations that are performed during the noiseless decoding in BSAC to about 22% of the amount of calculations with the conventional full-search method.

• Proceedings of the IEEK Conference
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• 2003.07e
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• pp.2471-2474
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• 2003

The modified discrete cosine transform (MDCT) and its inverse transform (IMDCT) are employed in subband/transform coding schemes as the analysis/synthesis filter bank based on time domain aliasing cancellation (TDAC). And they are the most computational intensive operations in layer III of the MPEG audio coding standard. In this paper, we propose a new efficient algorithm for the MDCT/IMDCT computation. It is based on the MDCT/IMDCT computation algorithm using the discrete cosine transforms (DCTs), and it employs two discrete cosine transform of type II(DCT-II) to compute the MDCT/IMDCT. In addition to, it takes advantage of ability in calculating the MDCT/IMDCT computation, where the length of a data block is divisible by 4. The proposed algorithm in this paper requires less calculation complexity than the existing methods. Also, it can be implemented by the parallel structure,, and its structure is particularly suitable for VLSI realization.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.41 no.7
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• pp.823-832
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• 2016

This paper presents a generation method of test streams for verifying conformance of an UHD broadcasting receiver including decoders for video and audio as well as parsers for PSIP and closed caption data. The proposed test streams for video/audio signals can evaluate conformance of HEVC, AC-3 and DTS-HD standards. Especially, test streams for HEVC video compression standard can be used for testing syntax compliance and error resilience for a HEVC decoder. Moreover, the proposed test streams for system/program and closed caption can be applied for verifying parsers for PSIP and CEA-708 standards.

• Journal of the Institute of Electronics and Information Engineers
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• v.50 no.2
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• pp.185-192
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• 2013

In this paper, proposed to improve the performance of speech and mixed content signal classification using MFCC based on GMM probability model used for the MPEG USAC(Unified Speech and Audio Coding) standard. For effective pattern recognition, the Gaussian mixture model (GMM) probability model is used. For the optimal GMM parameter extraction, we use the expectation maximization (EM) algorithm. The proposed classification algorithm is divided into two significant parts. The first one extracts the optimal parameters for the GMM. The second distinguishes between speech and mixed content signals using MFCC feature parameters. The performance of the proposed classification algorithm shows better results compared to the conventionally implemented USAC scheme.

• Broadcasting and Media Magazine
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• v.16 no.4
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• pp.59-71
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• 2011

오디오와 음성 압축 기술적 근간은 서로 다르지만, 최근의 모바일 멀티미디어 기기 시장의 컨버전스 현상에 따라 압축하고자 하는 신호가 혼용되고 있으며, 비슷한 목표 전송률과 음질로 수렴하고 있다. 현재는 동일 기기에서 서로 다른 압축 기술을 적용하고 있으나, 음성과 음악이 동시에 서비스 되는 멀티미디어 기기에서는 단일 압축 방식으로 처리하고자 하는 이슈가 부각되고 있다. 특히, 스마트 폰 및 음악 콘텐츠 포탈 서비스의 대중화를 고려할 때, 음성 및 음악 신호 모두를 효율적으로 압축하는 음악 및 음성 신호의 융합 압축 기술이 더욱 필요해 보인다. 본 고에서는 MPEG 오디오 그룹에서 가장 최근 진행한 Unified Speech and Audio Coding(USAC)의 탄생 배경 및 표준화 현황을 소개한다. USAC는 64kbps 이하에서 기술적으로 최고 성능을 지닌 AMR-WB+ 및 HE-AAC v2보다도 우월한 음질을 보이며, 높은 비트율에서도 동등한 음질을 보장한다. 이런 우수한 음질에 기여한 USAC의 스위칭 구조와 더불어 기술적으로 향상된 주요 모듈인 파라미터 기반 스테레오 및 고주파 압축, 그리고 엔트로피 코딩 방식에 대해서 살펴 본다. 향후, 다양한 오디오 신호를 효율적으로 압축하는 USAC는 디지털 라디오, 모바일 TV, 그리고 오디오 북과 같은 사용자 시나리오에서 사용될 확률이 높아 보인다. 또한, USAC는 배경 잡음이나 배경 음악이 있는 경우에도 성능이 우수하기 때문에 YouTube 및 podcast 등과 같이 사용자가 콘텐츠를 생성할 때도 유용하게 사용 될 수 있다.

• International Journal of Fuzzy Logic and Intelligent Systems
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• v.4 no.1
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• pp.34-39
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• 2004

Huffman coding which has been used in many data compression algorithms is a popular data compression technique used to reduce statistical redundancy of a signal. It has been proposed that the Huffman algorithm can decode efficiently using characteristics of the Huffman tables and patterns of the Huffman codeword. We propose a new Huffman decoding algorithm which used a multi way tree search and present an efficient hardware implementation method. This algorithm has a small logic area and memory space and is optimized for high speed decoding. The proposed Huffman decoding algorithm can be applied for many multimedia systems such as MPEG audio decoder.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2015.07a
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• pp.3-4
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• 2015

본 논문에서는 공간 오디오 부호화 기법인 MPEG 서라운드에서 공간 파라미터 전송 시 위상 파라미터를 생략하는 기법에 대해 다룬다. 기존 방법에서는 한 프레임이 모두 적은 위상차를 가지는 경우에도 정상적으로 처리하여 전송한다. 이러한 경우 위상차 파라미터를 생략하여 비트 효율을 향상시킬 수 있다. 스테레오 복원 과정에서 발생하는 채널 간 시간차에 기반해 설계된 양자화기를 생략 기법에 적용하면 기존에 비해 평균적으로 40 ~ 50% 정도의 위상 파라미터 절감 효과를 얻을 수 있다.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.10C
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• pp.1451-1459
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• 2004

In this paper, a low-power 32-point IMDCT structure is proposed for MP3. Through re-odering of IMDCT matrices, we propose the systolic structure operating with 16, 8, 4, 2, and 1 cycle, respectively. To reduce power consumption, multiplication of each sub blocks are implemented by add and shift operation with CSD(Canrmic sigled digit) form coefficients. To reduce, furthermore, the number of adders, we utilize the common sub-expression sharing techniques. With these techniques, the relative power consumption of the proposed structure is reduced by 58.4% comparison to the conventional structure using only 2's complement form coefficient. Validity of the proposed structure is proved through Verilog-HDL coding.

• Journal of the Institute of Electronics and Information Engineers
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• v.51 no.6
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• pp.221-227
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• 2014

In this paper, we propose a synthesis method that can effectively suppress the ambience which affects tonal components in the PS decoder. Ambience component was obtained by using decorrelation filter and the weighting of the ambience in the decoder was determined through IC parameter. However, since the parameters are extracted in the sub-band domain, a low IC value could be analyzed even if the tonal component is dominant. The quality of the output signal may be degraded. To prevent this problem, the tonality was measured in the downmixed signal and the weighting of the ambience components were adjusted appropriately according to the measured tonality index. The performance of the proposed method was evaluated by simulations. Furthermore, the subjective test was performed and the results confirmed that the proposed method offers improved quality.