Browsing by Author "Cheong, Kah-Meng"
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Item type:Item, A BINAURAL ALGORITHM FOR SPACE AND PITCH DETECTION(2011) ;Chou, Wen-Sheng ;Cheong, Kah-Meng ;Chi, Tai-Shih ;電機工程學系Department of Electrical and Computer EngineeringA binaural algorithm to simultaneously detect the azimuth angle and the pitch of the sound source is proposed in this paper. This algorithm is extended from the stereausis model with two-dimensional coincidence detectors in the joint Space-Pitch domain. In our simulations, sounds from different locations are produced by passing through the Head-Related-Transfer-Function (HRTF). Simulation results show that estimated azimuth angles from our proposed algorithm are more accurate than those from the stereausis model in the single sound source testing condition. Satisfactory results in streaming sound sources from the two-sound mixture by using estimated Space-Pitch information are also demonstrated in our pilot experiments.Item type:Item, A Non-uniformly Distributed Three-microphone Array for Speech Enhancement in Directional and Diffuse Noise Field(2014-01-01) ;Hsu, Chung-Chien ;Cheong, Kah-Meng ;Chi, Tai-Shih ;電機資訊學士班Undergraduate Honors Program of Electrical Engineering and Computer ScienceThis paper proposes a non-uniformly distributed three-microphone array speech enhancement system to suppress directional interferences and diffuse noise simultaneously. Each pair of microphones is designed to tackle one kind of noise. Unlike other hybrid systems, which combine different noise suppression techniques derived in different domains, the proposed system integrates two noise suppression techniques derived in the unified power spectral density domain. In terms of objective LLR and PESQ measures, the proposed hybrid system is demonstrated more effective in eliminating directional and diffuse noise under various SNR conditions than each individual suppression technique.Item type:Item, A Spectral Modulation Sensitivity Weighted Pre-emphasis Filter for Active Noise Control System(2016-01-01) ;Cheong, Kah-Meng ;Wang, Yuh-Yuan ;Chi, Tai-Shih ;電機工程學系Department of Electrical and Computer EngineeringPsychoacoustic active noise control (ANC) systems by considering human hearing thresholds in different frequency bands were developed in the past. Besides the frequency sensitivity, human hearing also shows different sensitivity to spectral and temporal modulations of the sound. In this paper, we propose a new psychoacoustic active noise control system by further considering the spectral modulation sensitivity of human hearing. In addition to the sound pressure level (SPL), the loudness level is also objectively assessed to evaluate the noise reduction performance of the proposed ANC system. Simulation results demonstrate the proposed system outperforms two compared systems under test conditions of narrowband and broadband noise in terms of the loudness level. The proposed algorithm has been validated on TI C6713 DSP platform for real-time process.Item type:Item, Acoustic Scene Change Detection by Spectro-Temporal Filtering on Spectrogram using Chirps(2016) ;Lin, Chang-Hong ;Cheong, Kah-Meng ;Huang, Mao-Chang ;Chen, Ming-YenChang, Chen-KueiIndoor acoustic scene change detection systems using periodic impulse signals were developed in the past. Comparing with impulse signals, the chirp signal is more robust against environmental noise due to its specific spectro-temporal structure, which can be easily detected using a spectro-temporal modulation filtering mechanism. In this paper, we demonstrate a system which employs the two-dimensional spectro-temporal filtering mechanism on Fourier spectrogram to measure the total energy of reverberations of the chirp signal and compares the energy difference between consecutive chirps with a predefined threshold to automatically detect the change of the acoustic scene. Simulation results show the proposed system is very effective in detecting the acoustic scene change in a real living room with great robustness against noise recorded from real-world.Item type:Item, MODULATION WIENER FILTER FOR IMPROVING SPEECH INTELLIGIBILITY(2015-01-01) ;Hsu, Chung-Chien ;Cheong, Kah-Meng ;Chien, Jen-Tzung ;Chi, Tai-Shih電機資訊學士班This paper presents a single-channel high-dimensional Wiener filter in the spectro-temporal modulation domain. Unlike other conventional noise reduction techniques, the proposed algorithm not only reduces noise but also enhances the "textures" of the speech signal. A non-iterative decision-directed noise estimation method is adopted to estimate the modulation SNR for the modulation-domain Wiener filter. The efficacy of the proposed algorithm in enhancing speech intelligibility is assessed using the short-time objective intelligibility (STOI) measure. Statistical analysis results demonstrate that our proposed algorithm can improve STOI scores in speech-shape noise (SSN) and white noise conditions, but not in babble noise condition, while the conventional Wiener filter fails to improve STOI scores in all three noise conditions.Item type:Item, MODULATION WIENER FILTER FOR IMPROVING SPEECH INTELLIGIBILITY(2015-01-01) ;Hsu, Chung-Chien ;Cheong, Kah-Meng ;Chien, Jen-Tzung ;Chi, Tai-Shih電機工程學系This paper presents a single-channel high-dimensional Wiener filter in the spectro-temporal modulation domain. Unlike other conventional noise reduction techniques, the proposed algorithm not only reduces noise but also enhances the "textures" of the speech signal. A non-iterative decision-directed noise estimation method is adopted to estimate the modulation SNR for the modulation-domain Wiener filter. The efficacy of the proposed algorithm in enhancing speech intelligibility is assessed using the short-time objective intelligibility (STOI) measure. Statistical analysis results demonstrate that our proposed algorithm can improve STOI scores in speech-shape noise (SSN) and white noise conditions, but not in babble noise condition, while the conventional Wiener filter fails to improve STOI scores in all three noise conditions.Item type:Item, Robust Voice Activity Detection Algorithm Based on Feature of Frequency Modulation of Harmonics and Its DSP Implementation(2015-10-01) ;Hsu, Chung-Chien ;Cheong, Kah-Meng ;Chi, Tai-Shih ;Tsao, Yu電機工程學系This paper proposes a voice activity detection (VAD) algorithm based on an energy related feature of the frequency modulation of harmonics. A multi-resolution spectro-temporal analysis framework, which was developed to extract texture features of the audio signal from its Fourier spectrogram, is used to extract frequency modulation features of the speech signal. The proposed algorithm labels the voice active segments of the speech signal by comparing the energy related feature of the frequency modulation of harmonics with a threshold. Then, the proposed VAD is implemented on one of Texas Instruments (TI) digital signal processor (DSP) platforms for real-time operation. Simulations conducted on the DSP platform demonstrate the proposed VAD performs significantly better than three standard VADs, ITU-T G.729B, ETSI AMR1 and AMR2, in non-stationary noise in terms of the receiver operating characteristic (ROC) curves and the recognition rates from a practical distributed speech recognition (DSR) system.Item type:Item, Spectral Modulation Sensitivity Based Perceptual Acoustic Echo Cancellation(2013-01-01) ;Chuang, Wei-Lun ;Cheong, Kah-Meng ;Hsu, Chung-Chien ;Chi, Tai-Shih電機工程學系Perceptual acoustic echo cancellers were developed by mainly considering human hearing thresholds of different acoustic frequencies in the past. In addition to the different frequency sensitivities, the human brain further analyzes sounds in terms of their spectral and temporal modulations. In this paper, we extend the perceptual normalized least mean square (P-NLMS) algorithm by adding a second pre-emphasis filter which accounts for brain's spectral modulation sensitivity. Simulation results and listening tests demonstrate our system effectively reduces the perceived residual echo during convergence due to faster convergence rates in hearing-sensitive spectral modulation bands and acoustic frequency bands.