Patents by Inventor Kai Meng

Kai Meng has filed for patents to protect the following inventions. This listing includes patent applications that are pending as well as patents that have already been granted by the United States Patent and Trademark Office (USPTO).

  • Patent number: 11631391
    Abstract: An electronic system includes a fan module, an embedded controller, a reference microphone, an error microphone, an active noise cancellation controller, and a micro speaker module. The reference microphone outputs a wide-band noise signal associated with the operation of the fan module. The error microphone outputs an error signal by detecting the noise level of the electronic system. According to the wide-band noise signal, the error signal and the fan information provided by the embedded controller, the active noise cancellation controller provides a speaker control signal by calculating narrow-band and wide-band noises of the fan module, and drives the micro speaker module to provide a noise cancellation signal which is adjusted according to the wind pressure under the current fan speed of the fan module. The error signal may be reduced to zero by adaptively adjusting the noise cancellation signal for canceling the noises in the electronic system.
    Type: Grant
    Filed: January 16, 2022
    Date of Patent: April 18, 2023
    Assignee: ACER INCORPORATED
    Inventors: Po-Jen Tu, Ruey-Ching Shyu, Jia-Ren Chang, Kai-Meng Tzeng
  • Publication number: 20230069213
    Abstract: A conference terminal and a feedback suppression method are provided. In the method, a transmitting sound signal is divided into sub sound signals of multiple frequency bands. Different sub sound signals correspond to different frequency bands. An interfered frequency band corresponding to the howling interference is detected according to the sub sound signals of those frequency bands. The power of the sub sound signal of the interfered frequency band increases along with time. The interfered frequency band is affected by the howling interference. An interference direction is determined according to multiple input sound signals received by the microphone array and merely pass through the interfered frequency band. A sound from the interference direction leads to the howling interference. A beam pattern of the microphone array is determined according to the interference direction. The gain of the beam pattern in the interference direction is reduced.
    Type: Application
    Filed: September 14, 2021
    Publication date: March 2, 2023
    Applicant: Acer Incorporated
    Inventors: Po-Jen Tu, Jia-Ren Chang, Kai-Meng Tzeng
  • Publication number: 20230058981
    Abstract: A conference terminal and an echo cancellation method for a conference are provided. In the echo cancellation method, a synthetic speech signal is received. The synthetic speech signal includes a user speech signal of a speaking party corresponding to a first conference terminal of multiple conference terminals and an audio watermark signal corresponding to the first conference terminal. One or more delay times corresponding to the audio watermark signal are detected in a received audio signal. The received audio signal is recorded through a sound receiver of a second conference terminal of the conference terminals. An echo in the received audio signal is canceled according to the delay time.
    Type: Application
    Filed: September 14, 2021
    Publication date: February 23, 2023
    Applicant: Acer Incorporated
    Inventors: Po-Jen Tu, Jia-Ren Chang, Kai-Meng Tzeng
  • Publication number: 20230030369
    Abstract: A processing method of sound watermark and a sound watermark generating apparatus are provided. In the method, a call reception sound signal is obtained by a sound receiver. A reflection sound signal is generated according to a virtual reflection condition and the call reception sound signal. The reflection sound signal is a sound signal obtained by simulating a sound output by the sound source, then being reflected by the external object, and being further recorded by the sound receiver. The phase of the reflection sound signal is shifted according to a watermark indication code to generate a sound watermark signal. The sound watermark signal includes the reflection sound signal with the phase shift. Accordingly, at the receiver end, the sound watermark signal via the feedback path could be eliminated by echo cancellation, and the sound watermark signal would not affect the speech signal on the call transmission path.
    Type: Application
    Filed: September 16, 2021
    Publication date: February 2, 2023
    Applicant: Acer Incorporated
    Inventors: Po-Jen Tu, Jia-Ren Chang, Kai-Meng Tzeng
  • Publication number: 20230019841
    Abstract: A processing method of a sound watermark and a speech communication system are provided. Multiple sinewave signals are generated. Frequencies of the sinewave signals are different from each other, and the sinewave signals belong to a high-frequency sound signal. A watermark pattern is mapped into a time-frequency diagram, to form a watermark sound signal. Two dimensions of the watermark pattern in a two-dimensional coordinate system respectively correspond to a time axis and a frequency axis in the time-frequency diagram. Each of multiple audio frames on the time axis corresponds to the sinewave signals with different frequencies on the frequency axis. A speech signal and the watermark sound signal are synthesized in a time domain to generate a watermark-embedded signal. Accordingly, a sound watermark may be embedded in real-time.
    Type: Application
    Filed: August 16, 2021
    Publication date: January 19, 2023
    Applicant: Acer Incorporated
    Inventors: Po-Jen Tu, Jia-Ren Chang, Kai-Meng Tzeng
  • Publication number: 20230001861
    Abstract: A mounting kit for installing a soundbar in a vehicle includes a tube having an elongated body and at least one chamber having a window. The mounting kit includes at least one bolt having a head and a shank, the head arranged to be coupled to the soundbar and the shank arranged to be received in the at least one chamber and accessible through the window to secure the soundbar to the tube. A first bracket is arranged to be secured to a cross member and first pillar of the vehicle, and a second bracket is arranged to be secured to the cross member and a second pillar of the vehicle, each of the first and second brackets having a cradle arranged to receive an end portion of the tube for installing the tube and attached soundbar in the vehicle.
    Type: Application
    Filed: July 1, 2021
    Publication date: January 5, 2023
    Inventors: Jeffery FAY, Kai MENG, Linghua JIANG
  • Patent number: 11540051
    Abstract: A two-channel balance method and an electronic device using the same are provided. The two-channel balance method includes the following steps. A gain-frequency information of a two-channel signal is adjusted. A sampling delay information of the two-channel signal is calculated according to a distance information among a sound receiving unit, a left speaker unit and a right speaker unit. A forward test audio file or a surround test audio file is generated according to the sampling delay information. A phase offset information is estimated according to at least the forward test audio file or the surround test audio file. A phase offset direction information is determined. A phase information of the two-channel signal is adjusted according to the phase offset information and the phase offset direction information.
    Type: Grant
    Filed: April 22, 2021
    Date of Patent: December 27, 2022
    Assignee: ACER INCORPORATED
    Inventors: Po-Jen Tu, Jia-Ren Chang, Kai-Meng Tzeng
  • Publication number: 20220406317
    Abstract: A conference terminal and an embedding method of audio watermarks are provided. In the method, a first speech signal and a first audio watermark signal are received respectively. The first speech signal relates to a speaker corresponding to another conference terminal, and the first audio watermark signal corresponds to the another conference terminal. The first speech signal is assigned to a host path to output a second speech signal. The first audio watermark signal is assigned to an offload path to output a second audio watermark signal. The host path provides more digital signal processing (DSP) effects than the offload path. The second speech signal and the second audio watermark signal are synthesized to output a synthesized audio signal. The synthesized audio signal is adapted for audio playback. A completed audio watermark signal is outputted accordingly.
    Type: Application
    Filed: August 16, 2021
    Publication date: December 22, 2022
    Applicant: Acer Incorporated
    Inventors: Po-Jen Tu, Jia-Ren Chang, Kai-Meng Tzeng
  • Publication number: 20220402118
    Abstract: A nanoscale positioning apparatus with a large stroke and multiple degrees of freedom and a control method thereof are provided. The nanoscale positioning apparatus includes a base, a plurality of parallel branch chain mechanisms and a working table. Each of the parallel branch chain mechanisms includes an electric cylinder, a micro-motion drive mechanism, a laser interferometer, a grating measuring device, a self-locking upper hinge and a self-locking lower hinge. The top of the base is connected to one end of the electric cylinder through the self-locking lower hinge. The other end of the electric cylinder is connected to one end of the micro-motion drive mechanism. The other end of the micro-motion drive mechanism is connected to the bottom of the working table through the self-locking upper hinge. The positioning apparatus has multiple degrees of freedom, and realizes multi-degree-of-freedom arbitrary position adjustment of the working table through parallel branch chain mechanisms.
    Type: Application
    Filed: May 22, 2020
    Publication date: December 22, 2022
    Applicant: WUXI FRIEDRICH MEASUREMENT AND CONTROL INSTRUMENTS CO., LTD
    Inventors: Fengwei ZHOU, Xiaoming QIAN, Kai MENG
  • Publication number: 20220343889
    Abstract: A method and an apparatus for audio signal processing selection are provided. In the method, multiple audio signal processing operations are performed on a synthesized audio signal to generate multiple processed audio signals, the audio signal processing operations are evaluated according to the comparison results between the processed audio signals and the primary signal, and the audio signal processing operation corresponding to a designated application and the designated audio output mode is selected according to the evaluation result of the audio signal processing operations. The synthesized audio signal is generated by adding a secondary signal into a primary signal. The signal processing is related to remove the secondary signal from the synthesized audio signal. Those processed audio signals are used by the designated application at the designated audio output mode. The comparison result is related to signal similarity. The evaluation result is related to the highest signal similarity.
    Type: Application
    Filed: October 4, 2021
    Publication date: October 27, 2022
    Applicant: Acer Incorporated
    Inventors: Po-Jen Tu, Jia-Ren Chang, Kai-Meng Tzeng, Ming-Chun Fang
  • Publication number: 20220208171
    Abstract: A method and an apparatus for audio signal processing evaluation are provided. The audio signal processing is performed on a synthesized audio signal to generate a processed audio signal. The synthesized audio signal is generated by adding a secondary signal into a master signal. The master signal is merely a speech signal. The signal processing is related to removing the secondary signal from the synthesized audio signal. The sound characteristics of the processed audio signal and the master signal are obtained, respectively. The sound characteristics include text content, and the text content is generated by performing speech-to-text on the processed audio signal and the master signal. The audio signal processing is evaluated according to the compared result between the sound characteristics of the processed audio signal and the master signal. The compared result includes the correctness of the text content of the processed audio signal relative to the master signal.
    Type: Application
    Filed: February 3, 2021
    Publication date: June 30, 2022
    Applicant: Acer Incorporated
    Inventors: Po-Jen Tu, Jia-Ren Chang, Kai-Meng Tzeng
  • Publication number: 20220141341
    Abstract: A conference terminal and a multi-device coordinating method for the conference are provided. In the method, multiple conference terminals are allocated to a plurality of areas according to location relationship. Each area includes one or more conference terminals that are close in the location relationship. An input sound signal is obtained from picking up a sound by the conference terminal in each area. The input sound signal of the one or more conference terminals in a first area among the areas is allocated to the one or more conference terminals in a second area among the areas to be played. The input sound signal obtained from picking up the sound by the conference terminal in each area is not played by any conference terminal in the same area.
    Type: Application
    Filed: January 7, 2021
    Publication date: May 5, 2022
    Applicant: Acer Incorporated
    Inventors: Po-Jen Tu, Jia-Ren Chang, Kai-Meng Tzeng, Chao-Kuang Yang
  • Patent number: 11259118
    Abstract: A Finite Impulse Response (FIR) filter is configured to minimize delay and maximize passband power by adjusting the filter coefficients applied to the sampled values. The FIR filter obtains an input signal and samples the input signal to generate a set of sampled input values. The FIR filter generates a set of filter coefficients, with each filter coefficient based on a corresponding sampled input value in the set of sample input values. The FIR filter selects a subset of sampled input values that have been most recently sampled from the input signal, and selects a subset of filter coefficients corresponding to sampled input values that are not the most recently sampled. The subset of sampled input values is combined with the subset of filter coefficients to generate an output value for the FIR filter.
    Type: Grant
    Filed: October 1, 2019
    Date of Patent: February 22, 2022
    Assignee: ACER INCORPORATED
    Inventors: Po-Jen Tu, Jia-Ren Chang, Kai-Meng Tzeng
  • Patent number: 11252505
    Abstract: A method for regulating a sound source of a designated object and an audio processing device using same are provided. The method includes the following steps. An original two-channel signal is obtained. An included angle of the designated object with respect to an ear of a user is detected. A first beam and a second beam are respectively formed in a clockwise direction and a counterclockwise direction according to the included angle to obtain a bidirectional sound signal. A sound rotation process is performed, so that the ear is directed toward sound source of the designated object, and a rotated two-channel sound signal is obtained. A unidirectional sound signal towards the sound source is obtained. A sound signal characteristic of the designated object is obtained according to the bidirectional sound signal and the unidirectional sound signal and then is regulated to synthesize a regulated two-channel signal.
    Type: Grant
    Filed: January 19, 2021
    Date of Patent: February 15, 2022
    Assignee: ACER INCORPORATED
    Inventors: Po-Jen Tu, Jia-Ren Chang, Kai-Meng Tzeng, Chao-Kuang Yang
  • Publication number: 20210337311
    Abstract: A two-channel balance method and an electronic device using the same are provided. The two-channel balance method includes the following steps. A gain-frequency information of a two-channel signal is adjusted. A sampling delay information of the two-channel signal is calculated according to a distance information among a sound receiving unit, a left speaker unit and a right speaker unit. A forward test audio file or a surround test audio file is generated according to the sampling delay information. A phase offset information is estimated according to at least the forward test audio file or the surround test audio file. A phase offset direction information is determined. A phase information of the two-channel signal is adjusted according to the phase offset information and the phase offset direction information.
    Type: Application
    Filed: April 22, 2021
    Publication date: October 28, 2021
    Applicant: Acer Incorporated
    Inventors: Po-Jen TU, Jia-Ren CHANG, Kai-Meng TZENG
  • Patent number: 11158301
    Abstract: A method for eliminating a specific object voice and an ear-wearing audio device using the same are provided. The ear-wearing audio device includes a plurality of voice receiving units, a voice direction tracking unit, a direction enhancement unit, a window cutting unit, a voiceprint recognition unit, a voice cancellation unit and two speakers. The voice receiving units are arranged in an array to obtain a sound signal. The voice direction tracking unit is configured to track a plurality of sound sources to obtain a plurality of sound source directions. The voiceprint recognition unit determines whether the sound signal contains a specific object voice in each of the sound source directions. If the sound signal contains the specific object voice in one of the sound source directions, the voice cancellation adjusts a field pattern using a beamforming technique to eliminate the specific object voice.
    Type: Grant
    Filed: November 20, 2020
    Date of Patent: October 26, 2021
    Assignee: ACER INCORPORATED
    Inventors: Po-Jen Tu, Jia-Ren Chang, Kai-Meng Tzeng, Kuei-Ting Tai, Chih-Ta Lin
  • Patent number: 11153703
    Abstract: A specific sound source automatic adjusting method and an electronic device using the same are provided. The electronic device includes a first audio recognition unit, a first multi-sound source determination unit, a directivity analysis unit, a directional separation unit, a second audio recognition unit, a second multi-sound source determination unit and an audio adjustment unit. If the number of sound sources of the original sound signal is larger than or equal to 2, the directivity analysis unit performs a directionality analysis procedure on the original sound signal. The directional separation unit separates out at least one specific directional sub-signal from the original sound signal according to the result of directional analysis procedure. If the number of sound sources of the specific directional sub-signal is equal to 1, the audio adjustment unit performs a sound source adjustment procedure.
    Type: Grant
    Filed: August 31, 2020
    Date of Patent: October 19, 2021
    Assignee: ACER INCORPORATED
    Inventors: Po-Jen Tu, Jia-Ren Chang, Kai-Meng Tzeng
  • Patent number: 11109175
    Abstract: A sound outputting device, a processing device and a sound controlling method thereof are provided. The sound controlling method includes the following steps. An original left sound signal and an original right sound signal are received. The original left sound signal and the original right sound signal are transformed to be a virtual left sound signal and a virtual right sound signal of a virtual sound source. A rotation degree of a user is detected. The virtual left sound signal and the virtual right sound signal are transformed to be an updated left sound signal and an updated right sound signal.
    Type: Grant
    Filed: July 9, 2019
    Date of Patent: August 31, 2021
    Assignee: ACER INCORPORATED
    Inventors: Po-Jen Tu, Jia-Ren Chang, Kai-Meng Tzeng
  • Publication number: 20210250682
    Abstract: A method for regulating a sound source of a designated object and an audio processing device using same are provided. The method includes the following steps. An original two-channel signal is obtained. An included angle of the designated object with respect to an ear of a user is detected. A first beam and a second beam are respectively formed in a clockwise direction and a counterclockwise direction according to the included angle to obtain a bidirectional sound signal. A sound rotation process is performed, so that the ear is directed toward sound source of the designated object, and a rotated two-channel sound signal is obtained. A unidirectional sound signal towards the sound source is obtained. A sound signal characteristic of the designated object is obtained according to the bidirectional sound signal and the unidirectional sound signal and then is regulated to synthesize a regulated two-channel signal.
    Type: Application
    Filed: January 19, 2021
    Publication date: August 12, 2021
    Applicant: Acer Incorporated
    Inventors: Po-Jen TU, Jia-Ren CHANG, Kai-Meng TZENG, Chao-Kuang YANG
  • Publication number: 20210248992
    Abstract: A method for eliminating a specific object voice and an ear-wearing audio device using the same are provided. The ear-wearing audio device includes a plurality of voice receiving units, a voice direction tracking unit, a direction enhancement unit, a window cutting unit, a voiceprint recognition unit, a voice cancellation unit and two speakers. The voice receiving units are arranged in an array to obtain a sound signal. The voice direction tracking unit is configured to track a plurality of sound sources to obtain a plurality of sound source directions. The voiceprint recognition unit determines whether the sound signal contains a specific object voice in each of the sound source directions. If the sound signal contains the specific object voice in one of the sound source directions, the voice cancellation adjusts a field pattern using a beamforming technique to eliminate the specific object voice.
    Type: Application
    Filed: November 20, 2020
    Publication date: August 12, 2021
    Applicant: Acer Incorporated
    Inventors: Po-Jen TU, Jia-Ren CHANG, Kai-Meng TZENG, Kuei-Ting TAI, Chih-Ta LIN