• The Journal of the Acoustical Society of Korea
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• v.19 no.8
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• pp.35-46
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• 2000

In this paper, we propose a subband acoustic echo canceller using the weighted Overlap-add adaptive filter bank to prevent the decrease of convergence speed in full-band US processing, and make it possible to realize the adaptive filter in block-parallel processing, this paper introduces the weighted overlap-add technique for subband echo canceller. Moreover, we propose a new double-talk detector which employs dedicated filter in addition to the energy comparison method simultaneously. The computer simulation results show that the performance of the proposed subband adaptive echo canceller double-talk detection

• Proceedings of the IEEK Conference
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• 2000.09a
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• pp.541-544
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• 2000

본 논문은 부밴드별 반향제거 필터 뱅크를 구현하기 위한 가중 Overlap-add 적응필터를 이용한 단일 채널 음향 반향 제거기를 제안한다. 기준 입력 신호의 고유치 분포율에 의존하여 수렴 특성이 결정되는 NLSM알고리즘을 사용하여 전대역 처리 과정에서 발생하는 수렴성능의 저하를 방지하고, 효율적인 블록별 병렬 처리가 가능한 부밴드 처리기법인 가중 Overlap-add 방식을 적용한 적응 반향제거기의 성능을 고찰한다 또한 본 논문에서는 동시 통화 검출을 위한 전용 필터와 에너지 비교 방법을 동시에 사용하는 새로운 형태의 동시통화 검출 기법을 제안한다.

• 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.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.32 no.12C
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• pp.1149-1155
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• 2007

In this paper, we propose a new overlap and add speech synthesis method which demonstrates improved continuity performance. The proposed method uses a weighted phase error function and minimizes the wave discontinuity of the synthesis signal, rather than the phase discontinuity, to estimate the mid-point phase. Experimental results show that the proposed method improves the continuity between the synthesized signals relative to the existing method.

• Proceedings of the IEEK Conference
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• 2007.07a
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• pp.409-410
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• 2007

In this paper, we propose a new method for overlap and add synthesis using phase shaping factor in a sinusoidal synthesis method of speech signal, which improves continuity and SNR(Signal Noise Ratio) efficiency of synthesized speech.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.34 no.3C
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• pp.251-256
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• 2009

In this paper, we propose a new method with the improved continuity performance of overlap and add speech signal synthesis method using damping harmonic amplitude parameter. The existing method uses the average value of past and current parameters for the sinusoidal amplitude used as the weight of phase error function. But, the proposed method extracts the more accurate sinusoidal amplitude by using a correlation between the original signals and the synthesized signals for the sinusodal amplitude used as the weights. To verify the performance of the proposed method, we observed the average differential error value between the synthesized signals.

• Journal of the Korean Institute of Intelligent Systems
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• v.14 no.6
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• pp.759-763
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• 2004

In this paper, real-time voice change method using pitch change technique is proposed to change one's voice to the other voice. For this purpose, sampling rate change method using DFT (Discrete Fourier Transform) method and time scale modification method using SOLA (Synchronized Overlap and Add) method is combined to change pitch. In order to evaluate the performance of the proposed method, voice transformation experiments were conducted. Experimental results showed that original speech signal is changed to the other speech signal in which original speaker's identity is difficult to find. The system is implemented using TI TMS320C6711DSK board to verify the system runs in real time.

• The Journal of Korea Institute of Information, Electronics, and Communication Technology
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• v.16 no.2
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• pp.69-76
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• 2023

A high-quality audio/voice codec using the MDCT/IMDCT process can perfectly restore the current frame through an overlap-add process with the previous frame. In the overlap-add process, an algorithm delay equal to the frame length occurs. In this paper, we propose a MDCT/IMDCT process that reduces algorithm delay by using a variable phase shift in MDCT/IMDCT process. In this paper, a low-delay audio/speech codec was proposed by applying the low delay MDCT/IMDCT algorithm to the ITU-T standard codec G.729.1 codec. The algorithm delay in the MDCT/IMDCT process can be reduced from 20 ms to 1.25 ms. The performance of the decoded output signal of the audio/speech codec to which low-delay MDCT/IMDCT is applied is evaluated through the PESQ test, which is an objective quality test method. Despite of the reduction in transmission delay, it was confirmed that there is no difference in sound quality from the conventional method.

• Proceedings of the IEEK Conference
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• 1999.11a
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• pp.1127-1130
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• 1999

In this paper, using one method of Additive Synthesis, Analysis-by-synthesis/Overlap-Add (ABS/OLA) method, analysis and synthesis of musical tones is processed. But peak detection of frequency domain is processed by proposed method considering the view of acoustics. It is that that harmonics frequency is times of main frequency. Using this fact, peak detection of frequency domain is useful for detection of tonal component identified musical note. It is possible to realize high-quality lour bit rate audio.

• The Journal of the Acoustical Society of Korea
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• v.33 no.4
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• pp.262-267
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• 2014

Recently, long pulses are transmitted for target detection in active sonar application. Matched filtering implemented by simple convolution algorithm, requires massive computational power for long replica. The computational loads are reduced significantly by implementing the convolution in the frequency domain with overlap add method, but the performance degrades for specified input/output system delay which constrains the size of FFT function. For performance improvement, the replica could be partitioned into uniform blocks (FDL) by re-using IFFT operations, or variable blocks of increasing length (MC) by using the largest possible blocks to calculate the convolution. In this paper, by combining the strong points of the two methods, we propose a new filter partition structure that allows for further optimization of the previous two methods.