• The Journal of the Acoustical Society of Korea
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• v.32 no.6
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• pp.525-531
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• 2013

In this paper, we propose to apply low delay algorithm using power complementary window to Integer Modified Discrete Cosine Transform (IntMDCT). Conventional transform, the Modified Discrete Cosine Transform (MDCT) usually produces floating point values for integer input values. This causes the expansion of the data. To refine on this, IntMDCT that produces integer values even for integer input values have emerged. However, IntMDCT has a problem of the algorithm delay, such as MDCT. Delay has became a key issue in environments for the purpose of real-time communications. In order to reduce the delay, the proposed algorithm was applied and the results of the performance evaluation show that delay of IntMDCT has reduced by halfexisting delay.

• The Journal of the Acoustical Society of Korea
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• v.37 no.2
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• pp.110-117
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• 2018

In this paper, we propose a low delay window switching MDCT (Modified Discrete Cosine Transform) method for speech/audio coder. The window switching algorithm is used to reduce the degradation of sound quality in non-stationary trasient duration and to reduce the algorithm delay by using the low delay TDAC (Time Domain Aliasing Cancellation). While the conventional window switching algorithms uses overlap-add with different lengths, the proposed method uses the fixed overlap add length. It results the reduction of algorithm delay by half and 1 bit reduction in frame indication information by using 2 window types. We apply the proposed algorithm to G.729.1 based on MDCT in order to evaluate the performance. The propose method shows the reduction of algorithm delay by half while speech quality of the proposed method maintains same as the conventional 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.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.40 no.6
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• pp.106-113
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• 2003

The modified discrete cosine transform (MDCT) and inverse MDCT (IMDCT) are employed in subband/transform coding schemes as the analysis/synthesis filter bank based on time domain aliasing cancellation (TDAC). And the MDCT and IMDCT 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 in various audio coding systems. 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, 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 method does. Also, it can be implemented by the parallel structure, therefore its structure is particularly suitable for VLSI realization

• Proceedings of the IEEK Conference
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• 1999.06a
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• pp.538-541
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• 1999

MPEG-2 AAC(Advanced Audio Coding) is the most advanced coding scheme available for high quality audio coding. This MPEG-2 AAC audio Standard allows for ITU-R ‘indistinguishable’ quality according to at data rates of 320 kb/s for five full-bandwidth channel audio signals. The compression ratio is around a factor of 1.4 better compared to MPEG Layer 3, you get the same quality at 70% of the bitrate. This paper suggest optimization method for MDCT/IMDCT (Modified Discrete Cosine Transform/Inverse Modified Discrete Cosine Transform) in Encoder and Decoder for AAC.

• Proceedings of the IEEK Conference
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• 1999.06a
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• pp.329-332
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• 1999

Currently, the most important technology is compression methods in the multimedia society. In audio compression, the method using human auditory nervous property is used. This method using psychoacoustical model is applied to perceptual audio coding, because human's audibility is limited. MPEG-II AAC(Advanced Audio Coding) is the most advanced coding scheme that is of benefit to high quality audio coding. The compression ratio is 1.4 times compared with MPEG-I layer-III. In this paper, the vector processor for MDCT/IMDCT(Modified Discrete Cosine Transform /Inverse Modified Discrete Cosine Transform) of MPEG-II AAC is designed.

• Proceedings of the Korea Multimedia Society Conference
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• 2003.05b
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• pp.69-73
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• 2003

음향압축에 있어서 인간의 청각신경의 특성을 이용하는 방식이 사용되고 있다. 이러한 방법은 심리음향모델(psychoacustical model)에서 도입되었다. 음향압축에서는 이러한 심리음향모델을 사용하여 인간이 지각할 수 없는 한도 내에서 부호화하지 않는 지각음향부호화(perceptual audio coding)사용한다. 지각음향부호화는 분석필터와 합성필터로 각각 부호화 복호화하는데 이것은 필터뱅크(filter bank)로 구현된 서브밴드코더(subband coder) 이다. 본 논문에서는 분석필터와 합성필터에 사용되는 MDCT(Modified Discrete Cosine Transform)와 IMDCT(Inverse Modified Discrete Cosine Transform)를 FPGA에 구현하였다.

• The Journal of the Acoustical Society of Korea
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• v.25 no.2E
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• pp.74-78
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• 2006

This paper proposes an effective method for classifying emotions of the music from its acoustical signals. Two feature sets, timbre and tempo, are directly extracted from the modified discrete cosine transform coefficients (MDCT), which are the output of partial MP3 (MPEG 1 Layer 3) decoder. Our tempo feature extraction method is based on the long-term modulation spectrum analysis. In order to effectively combine these two feature sets with different time resolution in an integrated system, a classifier with two layers based on AdaBoost algorithm is used. In the first layer the MDCT-driven timbre features are employed. By adding the MDCT-driven tempo feature in the second layer, the classification precision is improved dramatically.

• The Journal of Korea Institute of Information, Electronics, and Communication Technology
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• v.2 no.2
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• pp.37-43
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• 2009

In this study, multi-channel digital hearing aid algorithm for low power system is proposed. First, MDCT(Modified Discrete Cosine Transform) method converts time domain of input speech signal into frequency domain of it. Output signal from MDCT makes a group about each channel, and then each channel signal adjusts a gain using LCF(Loudness Compensation Function) table depending on hearing loss of an auditory person. Finally, compensation signal is composed by TDAC and IMDCT. Its all of process make progress 16-bit fixed-point operation. We use fast-MDCT instead of MDCT for reducing system complexity and previously computed tables instead of log computation for estimating a gain. This algorithm evaluate through computer simulation.

• ETRI Journal
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• v.35 no.3
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• pp.542-545
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• 2013

State-of-the-art voice codecs have been developed to extend the input bandwidth to enhance quality while maintaining interoperability with a legacy codec. Most of them employ a modified discrete cosine transform (MDCT) for coding their extended band. We propose a source filter model-based coding algorithm of MDCT spectral coefficients, apply it to the ITU-T G.711.1 super wideband (SWB) extension codec, and subjectively test it to validate the model. A subjective test shows a better quality over the standardized SWB codec.