| 341. |
- Winberg, Mathias, et al.
(författare)
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Inertial Mass Actuators, Understanding and Tuning
- 1994
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Konferensbidrag (refereegranskat)abstract
- The actuators, the error sensors and the control system are central components in active noise and vibration control systems. The actuators produce the secondary forces acting on the structure while the error sensors measure the quantity to be minimized. Inertial mass actuators are used frequently in a variety of applications, e.g. boat engine mounts and airplane fuselages. To tune the inertial mass actuator to fit a certain application or just to understand the manufactures specification it is important to have an understanding of the construction of the actuator and the equations describing it. This paper presents a mathematical model of the mechanical and electrical system for the inertial mass actuator. A mobility analogy is introduced in order to determine how the inertial mass and the spring constant of the suspension effect the resonance frequency of the actuator as well as the output force. Does the mobility of the structure, the actuator is mounted on, effect the produced output force inrelationship to the rated force of the actuator? Practical measurement results are presented in order to determine the resonance frequency and output forceof an inertial mass actuator manufactured by Metravib in France.
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| 342. |
- Yermeche, Zohra, et al.
(författare)
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A Calibrated Subband Beamforming Algorithm for Speech Enhancement
- 2002
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Konferensbidrag (refereegranskat)abstract
- This paper proposes a new calibrated adaptive frequency domain beamformer for speech enhancement. The beamformer is based on the principle of a soft constraint formed from calibration data, rather than precalculated from free-field assumptions. The benefit is that the real room acoustical properties will be taken into account. The proposed algorithm continuously estimates the spatial information for each frequency band, based on weighting of the received data. The update of the beamforming weights is done recursively where the initial precalculated correlation estimates of the speech constitute a soft constraint. The soft constraint secures the spatial-temporal passage of the desired source signal, without the need of any speech detection. The performance is evaluated in real world scenarios, both in a car-and a restaurant- environment. Interference and noise sup-pression of more than 15 dB is accomplished, while very small distortion is measured for the source signal.
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| 343. |
- Yermeche, Zohra, et al.
(författare)
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Beamforming for Moving Source Speech Enhancement
- 2005
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Ingår i: IEEE Workshop on Applications of Signal Processing to Audio and Acoustics, 2005.. - New Paltz, USA : IEEE. - 0780391543 ; , s. 25-28
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Konferensbidrag (refereegranskat)abstract
- This paper presents a new constrained subband beamforming algorithm to enhance speech signals generated by a moving source in a noisy environment. The beamformer is based on the principle of a soft constraint defined for a specified region corresponding to a known source location. The soft constraint secures the spatial-temporal passage of the desired source signal in the adaptive update of the beamforming weights and guarantees the full rank property of the matrix inverted in the update. The source of interest is modelled as a cluster of stationary point sources and source motion is accommodated by revising the point source cluster. The source modelling and its direct exploitation in the beamformer through covariance estimates are presented. An algorithm for sound source localization is used for speaker movement tracking and this information is exploited to update the spatial distribution in the source model. Evaluation in a real environment with a moving speaker shows a significant noise and hands-free interference suppression within the conventional telephone bandwidth. This is achieved with a negligible impact on speech distortion.
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| 344. |
- Yermeche, Zohra, et al.
(författare)
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Bind Subband Beamforming with Time-delay Constraints for Moving Source Speech Enhancement
- 2007
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Ingår i: IEEE Transactions on Audio, Speech, and Language Processing. - : IEEE. - 1558-7916 .- 1558-7924. ; 15:8, s. 2360-2372
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Tidskriftsartikel (refereegranskat)abstract
- A new robust microphone array method to enhance speech signals generated by a moving person in a noisy environment is presented. This blind approach is based on a two-stage scheme. First, a subband time-delay estimation method is used to localize the dominant speech source. The second stage involves speech enhancement, based on the acquired spatial information, by means of a soft-constrained subband beamformer. The novelty of the proposed method involves considering the spatial spreading of the sound source as equivalent to a time-delay spreading, thus, allowing for the estimated intersensor time-delays to be directly used in the beamforming operations. In comparison to previous approaches, this new method requires no special array geometry, knowledge of the array manifold, or acquisition of calibration data to adapt the array weights. Furthermore, such a scheme allows for the beamformer to efficiently adapt to speaker movement. The robustness of the time-delay estimation of speech signals in high noise levels is improved by making use of the non-Gaussian nature of speech trough a subband Kurtosis-weighted structure. Evaluation in a real environment with a moving speaker shows promising results, with suppression levels of up to 16 dB for background noise and interfering (speech) signals, associated to a relatively small effect of speech distortion.
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| 345. |
- Yermeche, Zohra, et al.
(författare)
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Blind Subband Beamforming for Speech Enhancement of Multiple Speakers
- 2006
-
Konferensbidrag (refereegranskat)abstract
- A new blind microphone array method to enhance speech signals generated by multiple sources in a noisy environment is proposed. This approach is based on a two-stage scheme. A subband time-delay estimation algorithm is first used to localize the dominant speech sources. The speech enhancement is performed in a second stage, based on the acquired spatial information, by means of a spatially constrained subband beamformer. The robustness of this structure is ensured by the spatial constraint constructed to include the discrepancies in the acoustical environment model as well as errors in the time-delay estimation. Such scheme also allows for an efficient adaptation of the beamformer to speakers movement. The proposed subband approach for time-delay estimation exploits the sparseness of speech signals in the time-frequency domain to localize multiple speakers simultaneously. It also provides means to select the number of target sources. Evaluation in a real environment shows promising results.
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| 346. |
- Yermeche, Zohra, et al.
(författare)
-
Moving Source Speech Enhancement Using Time-Delay Estimation
- 2005
-
Konferensbidrag (refereegranskat)abstract
- This paper presents a new constrained subband beamforming algorithm to enhance speech signals generated by a moving source in a noisy environment. The beamformer is based on the principle of a soft constraint calculated from an estimated source position. The soft constraint secures the spatial-temporal passage of the desired source signal in the adaptive update of the beamforming weights and guaranties the full rank property of the covariance matrix inverted in the update. This approach allows for an efficient adaptation of the beamformer to speaker movement by using a tracking algorithm for sound source time-delay estimation. The proposed method has the benefit of taking into consideration the discrepancies in the acoustical environment model as well as errors in the time-delay estimation. Evaluation in a real environment with a moving speaker in a hands-free situation shows up to 10~dB noise suppression and 20~dB interference suppression within the conventional telephone bandwidth. This is achieved with a negligible impact on target signal distortion.
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| 347. |
- Yermeche, Zohra, et al.
(författare)
-
Real-Time DSP Implementation of a Subband Beamforming Algorithm for Dual Microphone Speech Enhancement
- 2007
-
Konferensbidrag (refereegranskat)abstract
- A real-time Digital Signal Processor (DSP) based implementation of a subband beamforming algorithm and its evaluation for dual microphone speech enhancement is presented. The algorithm, a calibrated constrained beamformer, is described theoretically and a real-time structure is proposed, including an efficient approach for multichannel data transformation. Measurements show that the battery driven DSP implementation supports 20 h operation-time, with an improved Signal-to-Noise Ratio (SNR) of up to 14~dB in high-noise factory environment. Further, less than half the provided computational performance of the DSP is used by the proposed method, hence, processing of additional tasks may be included.
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| 348. |
- Yermeche, Zohra, et al.
(författare)
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Spatial Filter Bank Design for Speech Enhancement Beamforming Applications
- 2004
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Ingår i: Sensor Array and Multichannel Signal Processing Workshop Proceedings, 2004. - Barcelona : IEEE. - 0780385454 ; , s. 557-560
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Konferensbidrag (refereegranskat)abstract
- In this paper, a new spatial filter bank design method for speech enhancement beamforming applications is presented. The aim of this design is to construct a set of different filter banks that would include the constraint of signal passage at one position (and closing in other positions corresponding to known disturbing sources). By performing the directional opening towards the desired location in the fixed filter bank structure, the beamformer is left with the task of tracking and suppressing the continuously emerging noise sources. This algorithm has been implemented in MATLAB and tested on real speech recordings conducted in a car hands-free communication situation. Results show that a reduction of the total complexity can be achieved while maintaining the noise suppression performance and reducing the speech distortion.
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| 349. |
- Yermeche, Zohra, et al.
(författare)
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Speech enhancement of multiple moving sources based on subband clustering time-delay estimation
- 2006
-
Konferensbidrag (refereegranskat)abstract
- A new robust blind microphone array method to enhance speech signals generated by multiple moving sources in a noisy environment is presented. This approach is based on a two-stage scheme. A subband clustering time-delay estimation algorithm is first used to localize the dominant speech sources. The speech enhancement is performed in a second stage, based on the acquired spatial information, by means of a soft-constrained subband beamformer. The robustness of this structure is ensured by the spatial constraint constructed to include the discrepancies in the acoustical environment model as well as errors in the time-delay estimation. Such scheme also allows for an efficient adaptation of the beamformer to speakers movement. The proposed subband clustering approach for time-delay estimation exploits the sparseness of speech signals in the time-frequency domain to localize multiple speakers simultaneously. It also provides means to select the number of target sources. Evaluation in a real environment with moving speakers shows promising results.
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| 350. |
- Yiu, KFC, et al.
(författare)
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A fast algorithm for the optimal design of high accuracy windows in signal processing
- 2013
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Ingår i: Optimization Methods and Software. - : Taylor & Francis. - 1055-6788 .- 1029-4937. ; 28:4, s. 900-916
-
Tidskriftsartikel (refereegranskat)abstract
- This paper presents a new optimal window design method with a general window design specification for the passband and stopband. The design problem is formulated as a semi-infinite linear programming problem. With suitable discretizations, an exchange algorithm is employed. The convergence of the proposed algorithm is established. In the formulation, since the stopband is minimized, the method can be employed for the design of very highly optimized windows.
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