• 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.

• Journal of the Institute of Electronics and Information Engineers
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• v.52 no.3
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• pp.162-170
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• 2015

In the bandpass sampling (BPS), the sampling frequency is lower than the frequency of the signal to be sampled. In this method, the baseband spectrum can be directly obtained by the sampling operation. This makes the frequency down converter unnecessary as well as the receiver's circuit simpler. In the second-order BPS system, two sampling devices are used. When aliasing occurs due to the sampling operation, the aliased component can be cancelled by combining the two sampled signals. In this paper, it is presented a design method of the second-order BPS system when multiple interferences are simultaneously aliased to the signal component. The optimum phase of the interpolant filter is searched for maximizing the signal-to-interference ratio, and a practical formula for the suboptimal phase is derived in terms of the power spectrum profile of the BPS input. A computer simulation has been performed for the proposed second-order BPS system, and it has been shown that the signal-to-interference ratio can be increased by considering multiple aliasing.

• The Journal of the Acoustical Society of Korea
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• v.19 no.2
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• pp.13-18
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• 2000

This paper presents the result of a technique to enhance TDAC in the AC-3 algorithm. To reduce block boundary noise without decreasing the performance of transform coding, We propose new special windows which improve the defect of the AC-3 algorithm that could not properly cancel aliasing in the transient period. In addition, a fast MDCT calculation algorithm based on a fast Fourier transform, is adopted.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.25 no.6B
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• pp.1106-1113
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• 2000

In this, paper, a DFT based polyphase filter bank for the fractional ratio oversampling is proposed. Proper fractional oversampling ratio gives lower aliasing than the critical sampling and, at the same time, lower computational load than the integer ratio oversampling. In addition, filter bank design becomes easier by the reduced aliasing effect of fractional ratio oversampling. Proposed fractional ratio oaversampling polyphase structure is applied to a subband adaptive filter for acoustic echo cancellation where long adaptive filter are ofter required. Echo cancellation results show that fractional ratio oversampling gives comparable performance to the integer ratio oversampling with less computational load.

• ETRI Journal
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• v.35 no.5
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• pp.935-938
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• 2013

This letter presents an algorithm for selecting a low delay for the modified discrete cosine transform (MDCT) and inverse MDCT (IMDCT). The implementation of conventional MDCT and IMDCT requires a 50% overlap-add (OLA) for a perfect reconstruction. In the OLA process, an algorithmic delay in the frame length is employed. A reduced overlap window and MDCT/IMDCT phase shifting is used to reduce the algorithmic delay. The performance of the proposed algorithm is evaluated by applying the low-delay MDCT to the G.729.1 speech codec.

• Proceedings of the IEEK Conference
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• 2000.06d
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• pp.178-181
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• 2000

In this paper, we present the implementation of filterbank for MPEG-2 Advanced Audio Coding (AAC) decoder with VHDL. The filterbank of AAC employs a technique called time-domain aliasing cancellation (TDAC). In order to make the algorithm more efficiently, we decompose and reorganize the filterbank algorithm lot the high speed decoding process and lower computational cost. And we make this filterbank algorithm to be used with other modules of AAC decoder in parallel processing.

• 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

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.41 no.6
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• pp.261-269
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• 2004

In this paper, an efficient algorithm for computing in the MPEG/audio Layer Ⅲ (MP3) encoder is proposed. The proposed algerian performs a full-band 1024-point FFT by computing 32-point FFT's of 32 subband outputs. To reduce the aliasing caused by the analysis filter bank, an aliasing cancellation butterfly is developed. A major benefit of the proposed algorithm is the computational saving. By using the proposed algorithm, it is possible to save 40~50% of computations for FFT, which results in about 20% reduction of the PAM-2 complexity.

• 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.26 no.8B
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• pp.1057-1069
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• 2001

오디오 부호화 시스템에 널리 쓰이는 시간 영역 엘리어싱 제거(TDAC) 필터 뱅크의 성능 개선을 위하여, 다중 중첩-합 구조를 바탕으로 한 개선된 TDAC 필터 뱅크를 제안하였다. 제안된 구조는 필터 뱅크의 분해 부분과 합성 부분 사이에서 발생하는 양자화 잡음의 효과를 줄이도록 제안되었다. 모의 실험을 통해 같은 양자화 비트 수를 사용하는 경우에 제안한 시스템이 SNR 측면에서 보다 나은 성능을 나타냄을 보였으며, 망 전송 데이터 양을 같게 한 경우에도 제안한 시스템이 더 적은 데이터 양의 블록 단위를 가질 수 있으므로 데이터 망의 혼잡 제어에 있어 보다 유리할 수 있음을 보였다.