Abstract: An acoustic camera comprises a first sound pick-up device, a second sound pick-up device, and a switch. The switch is respectively connected to the first sound pick-up device and the second pick-up device and used to select the first sound pick-up device or the second sound pick-up device to reconstruct the sound field of the sound source of a detected object. The first sound pick-up device has a first microphone array, and the first microphone array is a near-field uniform microphone array. The second sound pick-up device has a second microphone array, and the second microphone array is a far-field non-uniform microphone array.

Abstract: The present invention discloses a method for parameter identification and parameter optimization of microspeakers. Measurement procedures for identifying electromechanical constants of microspeaker and a GUI are developed to facilitate estimation of electroacoustic parameters of the microspeaker under test. In light of the thus identified microspeaker parameters, a parameter optimization procedure is carried out to obtain the design that attains the best acoustic performance with minimum harmonic distortion.

Abstract: A piezoelectric panel speaker and an optimal method of designing the same is disclosed. In the structure of the speaker, at least one piezoelectric plate attached at a surrounding frame supports a diaphragm inside the surrounding frame. A spacer is inserted between the piezoelectric plate and the diaphragm. The structure of the piezoelectric plates fixed at the surrounding frame improves the speaker performance within the low frequency range. The finite element method is employed to build a mathematical model to simulate the sound pressure loading of the piezoelectric panel speaker. Also, the simulated annealing method is employed to approach the optimal design parameters of the speaker structure.

Abstract: A dereverberation and noise reduction method adapted for a microphone array and an apparatus using the same are proposed. The microphone array receives a plurality of audio signals from an audio source. The dereverberation and noise reduction method includes the following steps. The received audio signals are processed by a beamforming processing, and a first audio signal is generated. Besides, the received audio signals are processed by a suppression processing, and a suppression audio vector is generated. Further, suppression audio vector is processed by an adaptive filtering processing, and a second audio signal is generated. In addition, the second audio signal is subtracted from the first audio signal to acquire an audio output signal, where parameters of the adaptive filtering processing are adjusted according to a feedback of the audio output signal.

Abstract: This invention relates to a method for virtual bass synthesis. The low frequency signal is attained by applying a low pass filter to the original. In order to reduce the operations, process of down sampling the low frequency signal, moving the low frequency signal to a series of harmonics whose frequencies are integral times as large as the frequency of low frequency signals, and then up sampling them are provided. By means of psycho-acoustic theory, the weights of harmonics are attained and applied to the harmonics. Finally the weighted harmonics are combined to produce the bass signal. As the result, the virtual bass effect which is almost the same as the low frequency of the original audio signal can be accomplished. Because the harmonic signals are high frequency ones, the virtual effect can be made in the panel speakers or ordinary low-end speakers.

Abstract: An earphone acoustic simulation system and an optimal simulation method of the same is disclosed. The earphone acoustic simulation system comprises an earphone front end simulation circuit and an earphone back end simulation circuit for simulating acoustic environment of a front cavity and a back cavity inside an earphone, and an artificial ear simulation circuit is connected respectively with the earphone front end simulation circuit and the earphone back end simulation circuit. Variation of an impedance of the artificial ear simulation circuit represents the frequency response in the earphone cavity. Besides, the optimal simulation of the earphone acoustic simulation system utilizes simulated annealing algorithm to obtain the optimal parameter of the earphone cavity, and anticipates the SPL curve related to the optimal earphone cavity through utilizing the earphone acoustic simulation system.

Abstract: A device and a method for integrating 3D sound effect processing and active noise control are proposed. A digital signal processor incorporates an artificial reverberator and a 3D spatial audio processor into an audio module. The audio signal is presented via an earphone. Next, a microphone embedded in the vicinity of the loudspeaker inside the headset is used to sense an external noise while playing, and feed it back to an active noise controller, which generates an anti-noise to eliminate the external noise. Therefore, the signal to noise ratio can be increased and the 3D sound field effect can be significantly enhanced. In addition, a head-related transfer function is more efficiently implemented on the basis of an interaural transfer function in the spatial audio processing to reduce the filter order lower and hence the computation loading.

Abstract: An intelligent classification method is proposed. The method extracts vocal features from the temporal domain, spectral domain and statistical features for measuring the vocal signal. The measured result is grouped by comparing with the trained data with single voiced source, and then different voices can be separated from the vocal signal to be classified. The vocal features are evaluated from temporal domain and spectral domain and the statistical features, and the method can improve the accuracy of the voice classification.

Abstract: An acoustic camera comprises a first sound pick-up device, a second sound pick-up device, and a switch. The switch is respectively connected to the first sound pick-up device and the second pick-up device and used to select the first sound pick-up device or the second sound pick-up device to reconstruct the sound field of the sound source of a detected object. The first sound pick-up device has a first microphone array, and the first microphone array is a near-field uniform microphone array. The second sound pick-up device has a second microphone array, and the second microphone array is a far-field non-uniform microphone array.

Abstract: A sound broadcasting mechanism and an electronic device using the same are provided. The sound broadcasting mechanism includes a sound box, a sound broadcasting element and a sound enhancing element. The sound broadcasting element is disposed at the surface of the sound box to generate a sound signal. The sound enhancing element is disposed at the surface of the sound box. The sound enhancing element includes a resonating film and a weight increasing element. The resonating film can resonate with the sound signal in a resonating frequency range to enhance the sound intensity of the sound signal in the resonating frequency range. The weight increasing element is attached to the surface of the resonating film. The weight of the weight increasing element is related to the resonating frequency range.

Abstract: A system that integrates various intelligent classification techniques and preprocessing algorithms is provided. A feature extracting unit receives audio signals and extracts audio features for identification by using various descriptors; a preprocessing unit normalized the data for data consistency; a classification unit classifying audio signals into several categories according to the audio features.

Abstract: The present invention discloses a method for parameter identification and parameter optimization of microspeakers. Measurement procedures for identifying electromechanical constants of microspeaker and a GUI are developed to facilitate estimation of electroacoustic parameters of the microspeaker under test. In light of the thus identified microspeaker parameters, a parameter optimization procedure is carried out to obtain the design that attains the best acoustic performance with minimum harmonic distortion.

Abstract: This invention relates to a method for virtual bass synthesis. The low frequency signal is attained by applying a low pass filter to the original. In order to reduce the operations, process of down sampling the low frequency signal, moving the low frequency signal to a series of harmonics whose frequencies are integral times as large as the frequency of low frequency signals, and then up sampling them are provided. By means of psycho-acoustic theory, the weights of harmonics are attained and applied to the harmonics. Finally the weighted harmonics are combined to produce the bass signal. As the result, the virtual bass effect which is almost the same as the low frequency of the original audio signal can be accomplished. Because the harmonic signals are high frequency ones, the virtual effect can be made in the panel speakers or ordinary low-end speakers.