Browsing by Author "Yang, Cheng-Yen"

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A Systematic ANSI S1.11 Filter Bank Specification Relaxation and Its Efficient Multirate Architecture for Hearing-Aid Systems
(2016-08)
Yang, Cheng-Yen
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Liu, Chih-Wei
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Jou, Shyh-Jye
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電子工程學系及電子研究所
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Department of Electronics Engineering and Institute of Electronics
Recently, emerging mobile computing requires the high integration of hearing aids into a single system-on-chip. Modern hearing aid systems include a frequency decomposer, noise reduction, feedback cancellation, auditory compensation, and intelligent adaptation. The majority of existing works concentrated on improving the performance and efficiency on a single-signal processing block or one-sided effect. These works lacked comprehensive discussions on system-wide aspects regarding the overall impacts. To design an optimal hearing aid system, frequency decomposers, or the filter banks that dominate in hearing aid systems, are the first priority. We propose a systematic relaxation of the ANSI S1.11 specification and its design procedure for filter banks. The proposed design procedure overcomes the drawbacks of previous works and changes the five performance indices of the filter bank: delay, complexity, sub-band rate reduction, ripples of synthesized output, and prescription matching errors. These performance indices help system or algorithm designers in selecting a beneficial system-relaxed filter bank to achieve optimal hearing aids. The proposed multirate filter bank using the resampling method provides an efficient, low complexity, and delay-constrained computing architecture. Finally, seven design cases are used to demonstrate the proposed method, and comprehensive discussions of the five performance indices are presented.
AN EFFICIENT 18-BAND QUASI-ANSI 1/3-OCTAVE FILTER BANK USING RE-SAMPLING METHOD FOR DIGITAL HEARING AIDS
(2014-01-01)
Yang, Cheng-Yen
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Liu, Chih-Wei
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Jou, Shyh-Jye
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電子工程學系及電子研究所
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Department of Electronics Engineering and Institute of Electronics
This paper presents the multirate and re-sampling techniques to realize a low-delay, 18-band quasi-ANSI filter bank for digital hearing aids, which not only achieves a rather low computation complexity without a significant increase in the latency, but reduces greatly the total computation complexity for sub-band signal processing followed by the filter bank, such as noise reduction as well as wide dynamic range compression (WDRC). Researches done in the literature all focused on how to reduce the computation complexity of the filter bank. In particular, with the efficient multirate and interpolated FIR (IFIR) approaches for a 10-ms, 18-band quasi-ANSI filter bank, approximately 93% of the multiplications are saved, compared that with a straightforward parallel FIR filters architecture. However, they did not consider the computation complexity of the sub-band signal processing. In this paper, we first investigate realizing the FIR filter bank efficiently by using the multirate re-sampling techniques. To reduce the complexity, the optimized re-sampling factor for each filter band is explored carefully. Then, with the resampling technique, an efficient multirate quasi-ANSI FIR filter bank architecture is proposed. Compare to the state-of-the-art quasi-ANSI filter bank, approximately 17.7% of multiplicative complexity is reduced further and, up to 25% of the total computation complexity for sub-band signal processing followed by the filter bank is saved, but with only a slight increase in latency, i.e. 13.6 ms.
Binaural-Cue-Based Noise Reduction Using Multirate Quasi-ANSI Filter Bank for Hearing Aids
(2016)
Tang, Huei-Shiuan
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Yang, Cheng-Yen
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Liu, Chih Wei
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Chien, Chia-Cheng
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電子工程學系及電子研究所
This paper presents a low-power and real-time noise reduction (NR) algorithm using 18-band 1/3-octave quasi-ANSI filter bank for binaural hearing aids. With aids of binaural cues and minima controlled recursive average (MCRA) approach, the proposed NR algorithm consists of a directional mask and an estimated mask. The directional mask exploits the interaural time difference (ITD) together with the interaural level difference (ILD) of two input sources to attenuate the lateral noise, while the estimated mask is used to further reduce the background noise and the coherence noise. Investigated by the circumstances with the -3 dB, 0 dB, and 3 dB, respectively, cocktail-party effect, the simulation results show that the proposed binaural NR algorithm gains an average of approximately 5.07 dB SNR improvement. Moreover, the speech intelligibility index (SII performance is superior, comparing that with the state-of-the-art NR algorithms for hearing aids. The proposed binaural NR algorithm has been implemented in TSMC 90 nm CMOS high-VT technology. The chip design is operated by 288 KHz and consumes approximately 121.6 mu W (@1 V). The chip design can be operated by 0.6 V for real-time processing 24 kHz audio. The simulated power consumption is approximately 64.52 W for binaural hearing aids.
Method for image processing
(2010-06-24)
Wu, Bing-Fei
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Chen, Chao-Jung
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Kao, Chih-Chung
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Chung, Meng-Liang
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Chiu, Chung-Cheng
The invention relates to a method for image processing. First, establish the initial image background information. And retrieve the instant image information. Then calculate the initial image background information and color intensity information of the instant image. Furthermore, adjust the instant image information. Then calculate the moving-object information. Finally, track the moving-object information. It can improve the accuracy rate of detection without the influence of erected height.
Method for tracking and processing image
(2010-06-24)
Wu, Bing-Fei
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Chen, Chao-Jung
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Kao, Chih-Chung
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Chung, Meng-Liang
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Chiu, Chung-Cheng
The invention relates to a method for image processing, which can be used to calibrate the background quickly. When the external environment is changed due to the switch of light, the color of background is calibrated quickly, and the background can be updated together. The method not only is used to update the background, but also can be used to eliminate the convergence of background again.
SPATIAL-CUE-BASED MULTI-BAND BINAURAL NOISE REDUCTION FOR HEARING AIDS
(2013)
Yang, Cheng-Yen
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Chou, Wen-Sheng
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Chang, Kuo-Chiang
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Liu, Chih-Wei
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Chi, Tai-Shih
A noise reduction algorithm for binaural hearing aids is proposed in this paper. The proposed algorithm suppresses the directional interferences and diffuse noise using the interaural time difference (ITD), interaural level difference (ILD) and interaural coherence cues via the time-frequency (T-F) masking process. Furthermore, the proposed algorithm is evaluated under different specifications of ANSI-like filter banks for comprehending dominant factors in boosting the de-noise performance. Experimental results suggest that a higher resolution in low frequencies (approximately less than 1000 Hz) and a higher overlapping ratio of the filters are beneficial to de-noise performance in terms of sound quality.