A DOA estimation algorithm based on equalization-cancellation theory (応用音響)
スポンサーリンク
概要
- 論文の詳細を見る
Direction of arrival (DOA) estimation plays an important role in multi-channel (binaural) speech enhancement systems and auditory humanoid robots. A number of localization methods have been presented, however, most of them require a large-size array of microphones or cannot adapt to some special conditions, e.g., humanoid robot with the effect of head-related transfer function (HRTF). In this paper, we propose a two-microphone DOA estimation algorithm, namely EC-Beam which applies equalization-cancellation (EC) model into DOA estimation through beamformer-based technique. Specifically, the EC model is integrated into beamforming to remove the signal components from a given direction and yield the energy of the remained signals from other directions. Through searching several DOA candidates in the space, the estimation of DOA is finally determined as the direction at which the energy of the remained signal gets to minimum. Interpolation method is further exploited in EC-Beam to estimate non-beamformed directions. Experimental results showed that the EC-Beam with only two microphones is able to estimate much accurately the DOA of target signal in various noise conditions and well adapt to binaural hearing systems.
- 社団法人電子情報通信学会の論文
- 2010-06-03
著者
-
Akagi Masato
School of Information Science, Japan Advanced Institute of Science and Technology
-
Chau Duc
School of Information Science, Japan Advanced Institute of Science and Technology
-
Li Junfeng
School of Information Science, Japan Advanced Institute of Science and Technology
-
Li Junfeng
School Of Information Science Japan Advanced Institute Of Science And Technology
-
Li Junfeng
School Of Aerospace Tsinghua University
-
Akagi Masato
School Of Information Sci. Japan Advanced Inst. Of Sci. And Technol. (jaist) 1-1 Asahidai Nomi Ishik
-
Chau Duc
School Of Information Science Japan Advanced Institute Of Science And Technology
-
赤木 正人
School Of Information Science Japan Advanced Institute Of Science And Technology
-
Akagi Masato
School Of Information Sci. Japan Advanced Inst. Of Sci. And Technol.
関連論文
- A DOA estimation algorithm based on equalization-cancellation theory (応用音響)
- A study on the LP-based blind model in restoring bone-conducted speech (Speech) -- (国際ワークショップ"Asian workshop on speech science and technology")
- An LP-based blind restoration method for improving intelligibility of bone-conducted speech (音声)
- Trajectory Optimization of Multi-Asteroids Exploration with Low Thrust
- A flexible spectral modification method based on temporal decomposition and Gaussian mixture model
- Trajectory Optimization of Multi-Asteroids Exploration with Low Thrust
- 加法的に付加された雑音により生じた歪を評価するための聴覚特性を考慮したスペクトル歪
- A speech dereverberation method based on the MTF concept in power envelope restoration
- An improved method based on the MTF concept for restoring the power envelope from a reverberant signal
- A DOA estimation algorithm based on equalization-cancellation theory (応用音響)
- Effects of single-channel speech enhancement algorithms on Mandarin speech intelligibility (応用音響)
- Improvement of robustness using selective sound segregation for automatic speech recognition systems in noisy environments (Speech) -- (国際ワークショップ"Asian workshop on speech science and technology")
- LP-baesd method of blind restoration to improve intelligibility of bone-conducted speech
- A Noise Reduction System in Localized and Non-Localized Noise Environments
- In Situ Resistance Measurement of Nickel-Induced Lateral Crystallization of Amorphous Silicon
- Noise reduction method based on generalized subtractive beamformer
- Fundamental frequency estimation for noisy speech based on instantaneous amplitude and frequency
- A Noise Reduction Method Based on a Generalized Subtractive Beamformer
- Comparative evaluation of modulation-transfer-function-based blind restoration of sub-band power envelopes of speech as a front-end processor for automatic speech recognition systems
- Sub-Band Temporal Envelope Restoration for ASR in Reverberation Environment (国際ワークショップ Frontiers in Speech and Hearing Research)
- A study on expressive speech and perception of semantic primitives: comparison between Taiwanese and Japanese (音声)
- A flexible temporal decomposition-based spectral modification method using asymmetric Gaussian mixture model (音声)
- A Study on Restoration of Bone-Conducted Speech with LPC-Based Model (国際ワークショップ Frontiers in Speech and Hearing Research)
- スペクトル包絡における個人性について
- A computational model of co-modulation masking release
- A method of signal extraction from noisy signal based on auditory scene analysis
- Modified Restricted Temporal Decomposition and Its Application to Low Rate Speech Coding
- Foreword to the special issue on "Applied Systems"
- Evaluations of TS-BASE for speech enhancement and binaural benefits preservation (応用音響)
- Adaptive β-order Generalized Spectral Subtraction for Speech Enhancement
- A Two-Microphone Noise Reduction Method in Highly Non-stationary Multiple-Noise-Source Environments
- A Hybrid Speech Emotion Recognition System Based on Spectral and Prosodic Features
- 会長就任にあたって : 新たな四半世紀に向けて計画から実行へ
- Effects of single-channel speech enhancement algorithms on Mandarin speech intelligibility
- 基本周波数パターンに含まれる個人性とその制御
- In Situ Resistance Measurement of Nickel-Induced Lateral Crystallization of Amorphous Silicon
- Adaptive equalization-cancellation model and its application to sound localization in noisy reverberant environments