Abstract: A method of using a dual-port measurement system to measure acoustic impedance is used to measure an acoustic impedance Z of a tested object. The tested object includes an input end and an output end opposite to the input end. The dual-port measurement system comprises a first impedance tube and a second impedance tube. The first impedance tube includes a first inlet where a plane wave of a sound source is input, and a first outlet connected with the input end. The second impedance tube includes a second inlet connected with the output end, and a second outlet where the plane wave is output. The method uses the dual-port measurement system and a two-boundary method to obtain the acoustic impedances Z, whereby the dual-port measurement system is conveniently applied to various fields, such as the design of earphones, muffler tubes, sound absorption materials, and artificial ears.

Abstract: A method for recording and reconstructing a three-dimensional (3D) sound field, wherein a microphone array is established in a 3D sound field to track and locate sound sources in the 3D sound field and retrieve corresponding sound source signals. A plurality of control points is established inside an area where the 3D sound field is to be reconstructed. The control points are used to establish relational expressions of the sound source signals, the 3D sound field, a reconstructed sound field, and reconstructed sound source signals. The reconstructed sound source signals are obtained via solving the relational expressions and input into a speaker array arranged outside the area to establish the reconstructed sound field in the area. The present invention truly records the 3D sound field without using any extra transformation process and replays the reconstructed sound field with a larger sweet spot in higher fidelity.

Abstract: A method using an may microphone to cancel echo applies to a sound receiving system and comprises steps: an array microphone receiving a sound source and outputting a plurality of analog acoustic signals formed from the sound source; an A/D converter converting the analog acoustic signals into a plurality of digital acoustic signals; a digital signal processor respectively using an adaptive beamforming process and a blocking matrix filtering process to convert the digital acoustic signals into a primary acoustic signal and at least one noise signal; and the digital signal processor using a multiple-input cancelling process to subtract the noise signal from the primary acoustic signal to obtain an acoustic signal where the echo has been cancelled. Thereby, the present invention can eliminate the systematic errors of the array microphone of the sound receiving system and improves the robustness of the acoustic signal.

Abstract: A method of using a dual-port measurement system to measure acoustic impedance is used to measure an acoustic impedance Z of a tested object. The tested object includes an input end and an output end opposite to the input end. The dual-port measurement system comprises a first impedance tube and a second impedance tube. The first impedance tube includes a first inlet where a plane wave of a sound source is input, and a first outlet connected with the input end. The second impedance tube includes a second inlet connected with the output end, and a second outlet where the plane wave is output. The method uses the dual-port measurement system and a two-boundary method to obtain the acoustic impedances Z, whereby the dual-port measurement system is conveniently applied to various fields, such as the design of earphones, muffler tubes, sound absorption materials, and artificial ears.

Abstract: A method for recording and reconstructing a three-dimensional (3D) sound field, wherein a microphone array is established in a 3D sound field to track and locate sound sources in the 3D sound field and retrieve corresponding sound source signals. A plurality of control points is established inside an area where the 3D sound field is to be reconstructed. The control points are used to establish relational expressions of the sound source signals, the 3D sound field, a reconstructed sound field, and reconstructed sound source signals. The reconstructed sound source signals are obtained via solving the relational expressions and input into a speaker array arranged outside the area to establish the reconstructed sound field in the area. The present invention truly records the 3D sound field without using any extra transformation process and replays the reconstructed sound field with a larger sweet spot in higher fidelity.

Abstract: A method for visualizing sound source energy distribution in an echoic environment comprises steps: arranging in an echoic environment a plurality of arrayed sound pickup units, wherein each sound pickup unit includes at least two microphones separated by a directive distance enabling the sound pickup unit to have a primary pickup direction; disposing the sound pickup units with the primary pickup directions thereof pointing toward a sound source in the echoic environment, and measuring the sound source by the sound pickup units to obtain a sound source-related parameter; substituting the directive distance and the parameter into an algorithm to make the parameter have directivity; and then substituting the parameter into an ESM algorithm to establish a sound source energy distribution profile. Thereby, the method can measure a sound source in a specified direction in an echoic environment and establish a visualized sound source energy distribution profile.

Abstract: A miniature electronic shotgun microphone, which is used to receive a sound source from a specified direction, comprises a pick-up member, an A/D (Analog/Digital) conversion unit, and a digital signal processor. The pick-up member includes a first pick-up unit, a second pick-up unit separated from the first pick-up unit by a first distance, and a third pick-up unit separated from the second pick-up unit by a second distance; the first distance is greater than the second distance. The first pick-up unit, the second pick-up unit and the third pick-up unit respectively receive the sound source and output an analog signal. The A/D conversion unit and the digital signal processor process the analog signals, and convert them into a directional digital acoustic signal. Thus, the directional digital acoustic signal has a maximum pick-up frequency. Thereby is decreased grating lobes and spatial aliasing.

Abstract: A method of using microphones to measure a particle velocity comprises steps: arranging a sound source, a first microphone and a second microphone in a space, wherein the first microphone is arranged between the sound source and second microphone, and wherein the first microphone is located at a first position and the second microphone is located at a second position more far away from the sound source than the first position; using the first and second microphones to measure the sound source and obtain first and second acoustic pressures respectively; using the first and second positions and an equivalent source method to establish a free-space Green's function, and using the first and second acoustic pressures and the equivalent source method to establish an acoustic pressure function; and using the free-space Green's function and acoustic pressure function to predict state space of sound amplitude and then obtain a particle velocity.

Abstract: A method using an may microphone to cancel echo applies to a sound receiving system and comprises steps: an array microphone receiving a sound source and outputting a plurality of analog acoustic signals formed from the sound source; an A/D converter converting the analog acoustic signals into a plurality of digital acoustic signals; a digital signal processor respectively using an adaptive beamforming process and a blocking matrix filtering process to convert the digital acoustic signals into a primary acoustic signal and at least one noise signal; and the digital signal processor using a multiple-input cancelling process to subtract the noise signal from the primary acoustic signal to obtain an acoustic signal where the echo has been cancelled. Thereby, the present invention can eliminate the systematic errors of the array microphone of the sound receiving system and improves the robustness of the acoustic signal.

Abstract: The present invention discloses a microphone array structure able to reduce noise and improve speech quality and a method thereof. The method of the present invention comprises steps: using at least two microphone to receive at least two microphone signals each containing a noise signal and a speech signal; using FFT modules to transform the microphone signals into frequency-domain signals; calculating an included angle between a speech signal and a noise signal of the microphone signal, and selecting a phase difference estimation algorithm, a noise reduction algorithm or both to reduce noise according to the included angle; if the phase difference estimation algorithm is used, calculating phase difference of the microphone signals to obtain a time-space domain mask signal; and multiplying the mask signal and the average of the microphone signals to obtain the speech signals of the microphone signals. Thereby is eliminated noise and improve speech quality.

Abstract: The present invention provides a time-reversal-based impact-localization and haptic-feedback method for touch panels. Firstly, a plate model for an elastic plate and the impulse responses thereof are constructed according a plate theory. Next, a mathematical model is established according to the impulse responses and a time-reversal approach. When an impact force hits the elastic plate, the touched point on the elastic plate generates a touch signal, and the touch signal is received by at least a sensor at the corner of the elastic plate. The touch signal is converted into a time-reversal signal according to the mathematical model by a simulator. The time-reversal signal is reversed to the touched point for simulating reversal vibration waves of the time-reversal signal on the elastic plate, and locating the touched point.

Abstract: A method of using microphones to measure a particle velocity comprises steps: arranging a sound source, a first microphone and a second microphone in a space, wherein the first microphone is arranged between the sound source and second microphone, and wherein the first microphone is located at a first position and the second microphone is located at a second position more far away from the sound source than the first position; using the first and second microphones to measure the sound source and obtain first and second acoustic pressures respectively; using the first and second positions and an equivalent source method to establish a free-space Green's function, and using the first and second acoustic pressures and the equivalent source method to establish an acoustic pressure function; and using the free-space Green's function and acoustic pressure function to predict state space of sound amplitude and then obtain a particle velocity.

Abstract: A miniature electronic shotgun microphone, which is used to receive a sound source from a specified direction, comprises a pick-up member, an A/D (Analog/Digital) conversion unit, and a digital signal processor. The pick-up member includes a first pick-up unit, a second pick-up unit separated from the first pick-up unit by a first distance, and a third pick-up unit separated from the second pick-up unit by a second distance; the first distance is greater than the second distance. The first pick-up unit, the second pick-up unit and the third pick-up unit respectively receive the sound source and output an analog signal. The A/D conversion unit and the digital signal processor process the analog signals, and convert them into a directional digital acoustic signal. Thus, the directional digital acoustic signal has a maximum pick-up frequency. Thereby is decreased grating lobes and spatial aliasing.

Abstract: A method for visualizing sound source energy distribution in an echoic environment comprises steps: arranging in an echoic environment a plurality of arrayed sound pickup units, wherein each sound pickup unit includes at least two microphones separated by a directive distance enabling the sound pickup unit to have a primary pickup direction; disposing the sound pickup units with the primary pickup directions thereof pointing toward a sound source in the echoic environment, and measuring the sound source by the sound pickup units to obtain a sound source-related parameter; substituting the directive distance and the parameter into an algorithm to make the parameter have directivity; and then substituting the parameter into an ESM algorithm to establish a sound source energy distribution profile. Thereby, the method can measure a sound source in a specified direction in an echoic environment and establish a visualized sound source energy distribution profile.

Abstract: A method for diagnosis of diseases adopted on an electronic stethoscope which includes at least two sound receiving portions, a noise control portion, a processing portion, a data portion and an output portion. The method includes: first, the sound receiving portions receive sound signals issued from a patient's lungs included external noises; next, the sound signals are sent to the noise control portion which eliminates the external noise, and the processing portion to be overlapped and intensified; then characteristic values are retrieved from the sound signals to be compared with disease sound signal data in the data portion; finally the output portion outputs a diseases judgment result. Thus the electronic stethoscope can perform automatic interpretation of diseases to reduce human erroneous diagnostic judgment. Users also can get preliminary understanding of their body conditions when doctors are absent.

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: 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 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: The present invention discloses a microphone array structure able to reduce noise and improve speech quality and a method thereof. The method of the present invention comprises steps: using at least two microphone to receive at least two microphone signals each containing a noise signal and a speech signal; using FFT modules to transform the microphone signals into frequency-domain signals; calculating an included angle between a speech signal and a noise signal of the microphone signal, and selecting a phase difference estimation algorithm, a noise reduction algorithm or both to reduce noise according to the included angle; if the phase difference estimation algorithm is used, calculating phase difference of the microphone signals to obtain a time-space domain mask signal; and multiplying the mask signal and the average of the microphone signals to obtain the speech signals of the microphone signals. Thereby is eliminated noise and improve speech quality.

Abstract: The present invention provides a time-reversal-based impact-localization and haptic-feedback method for touch panels. Firstly, a plate model for an elastic plate and the impulse responses thereof are constructed according a plate theory. Next, a mathematical model is established according to the impulse responses and a time-reversal approach. When an impact force hits the elastic plate, the touched point on the elastic plate generates a touch signal, and the touch signal is received by at least a sensor at the corner of the elastic plate. The touch signal is converted into a time-reversal signal according to the mathematical model by a simulator. The time-reversal signal is reversed to the touched point for simulating reversal vibration waves of the time-reversal signal on the elastic plate, and locating the touched point.