Abstract: An audio signal processing device includes a volume setting unit that changes a volume of an audio signal according to a control for changing the volume, and a level control unit that controls a level of a predetermined band of the audio signal according to the control.

Abstract: An electronic device includes a first housing component including a first surface facing a first side and a second surface facing a second side, the first housing component including a first through-hole connecting the first surface and the second surface, a waterproof film located to cover the first through-hole, an audio component located on the first side of the waterproof film, and a second housing component attached removably to the second side of the first housing component. The first through-hole has such a shape that the waterproof film is directly visible when the first side is seen from the second side without the second housing component attached to the first housing component. The first housing component and the second housing component are combined together to form a passage leading from an end of the first through-hole on the second surface side to a first opening.

Abstract: A reflector and an electronic system produce a diffuse way by creating time delays in accordance with a number sequence. An acoustical passive reflector incorporates a series of wells in its surface to transform an acoustical wave into a series of acoustical waves having a time difference based on a number sequence. The electronic signal conversion system converts a signal into a series of signals having a time difference based on a number sequence. This can be used in an audio speaker system having N×N array of speakers where N is an odd prime number, arranged to be driven by the electronic signal conversion system in which the signal is converted into a series of signals centered on the signal with at least one signal being timed to precede the signal and at least one signal to follow the signal and the signal being arranged to be sent to the central speaker in the N×N array.

Abstract: A speaker system can include a speaker core and a removable jacket. The speaker core can have a housing and a speaker positioned within the housing. The removable jacket can cover the speaker on an outside surface of the housing, the jacket extending along at least three sides of the housing. Among other features, the speaker core can be used with one of among a number of different jackets to form different speaker systems.

Abstract: Configuring an adaptive microphone array to gather signals from a main lobe of the array, and configuring the array to reduce side interference gathered from sources that are not situated within the main lobe. A memory stores test signals gathered by the array in an anechoic chamber, a free-field environment, or other echo, interference and noise-minimized acoustic environment. Signals gathered in real time are processed to provide a preliminary output and preliminary weights. The test signals are retrieved from memory. The preliminary weights are applied to the test signals to provide equalizer weights. The equalizer weights and the preliminary output generate an equalized output that minimizes degradations resulting from side interference reduction.

Abstract: A microphone or instrument sound pickup for various musical instruments and singers is coupled to an amplifier, the output of which is coupled to a speaker system. A microphone is positioned within the output range of the speaker system and is coupled to one input of a comparator circuit. The remaining input of the comparator circuit is coupled to the microphones or pickups. An additive device receives the output signal of the comparator and the originating signal from the microphones or pickups to form a correcting audio signal which is applied to the amplifier. In an alternate embodiment, an adaptive preamplifier is interposed between the originating microphones and audio pickups and the amplifier. The adaptive preamplifier receives the output of the comparator and imposes alteration upon the signal processed through the preamplifier for application to the amplifier.

Abstract: Devices, systems, and methods provide direction finding of an acoustic signal source with respect to a voice-controlled device. The direction can be found without using elevation data, instead determining the horizontal location based on power values of the received signal. A large number of candidate vectors having values for azimuth, elevation, and power may be generated by a steered response power algorithm. The large number of vectors is reduced to a small number of reference azimuths spanning an azimuth range by associating the vectors with the closest reference azimuth and then calculating an average and/or maximum power of the associated vectors at each reference azimuth. The reference azimuth with the highest average (or maximum) power may be set as the direction of the signal source. Alternatively, each reference azimuth having an average (or maximum) power exceeding a threshold may be considered a direction of one of multiple sources.

Abstract: In one embodiment, a device includes at least one transducer configured to vibrate within a first frequency range to generate a signal audible by a wearer of the device. The device includes the first material, which has an acoustic impedance at one or more frequencies within the first frequency range that is substantially similar to an acoustic impedance of the wearer's skin at the one or more frequencies. The device includes a second material that has an acoustic impedance at one or more frequencies within the first frequency range that substantially differs from an acoustic impedance of an environment of the device.

Abstract: The present invention provides for an apparatus, system, and method for generating a head related audio transfer function in real time. Specifically, the present invention utilizes unique structural components including a tragus structure and an antihelix structure in connection with a microphone in order to communicate the location of a sound in three dimensional space to a user.

Abstract: An adaptive filter of an adaptive blocking matrix in an adaptive beam former or null former may be modified to track and maintain noise correlation between an input and a reference noise signal to the adaptive noise canceller module. That is, a noise correlation factor may be determined, and that noise correlation factor may be used in an inter-sensor signal model applied when generating the blocking matrix output signal. The output signal may then be further processed within the adaptive beamformer to generate a less-noisy representation of the speech signal received at the microphones. The inter-sensor signal model may be estimated using a gradient decent total least squares (GrTLS) algorithm. Further, spatial pre-whitening may be applied in the adaptive blocking matrix to further improve noise reduction.

Abstract: According to an embodiment, a speaker system includes filters, speakers, and a sound collection unit. Each of the filters filters a first signal to generate a second signal. The speakers are arranged so that transfer characteristics from the speakers to an evaluation point are different from each other. Each of the speakers converts the second signal generated by a corresponding one of the filters into a sound wave. The sound collection unit combines sound waves output from the speakers to generate a combined sound wave. The filters are formed so that a transfer characteristic from the first signal to an output signal matches a target transfer characteristic, the output signal indicating a sound pressure of the combined sound wave at the evaluation point.

Abstract: An in-car audio system includes a speaker unit. A frame of the speaker unit is fixed to a support plate serving as part of the structural member of a vehicle via a partition wall member. The support plate has an opening formed therein. A duct is connected to the opening, and the opening communicates with the outside of the vehicle. A space surrounded by the speaker unit, the support plate, and a partition wall member serves as a first acoustic resonance space, and a communication channel located inside the duct serves as a second acoustic resonance space.

Abstract: A unidirectional condenser microphone includes a diaphragm, a fixed electrode disposed opposite a back face of the diaphragm and an electrode extraction part for the fixed electrode disposed at a backside of the fixed electrode and having a through hole adapted to capture a sound wave from a rear acoustic terminal into a backside of the diaphragm. The through hole has a horn-shaped opening formed in continuation of the through hole at the rear acoustic terminal side thereof such that an inner diameter of the horn-shaped opening is increased toward the rear acoustic terminal side. The electrode extraction part has a plurality of the through holes formed therein at regular intervals along a concentric circle around an axial center of the electrode extraction part, and all the through holes along the concentric circle have the horn-shaped openings formed therein, respectively.

Abstract: The present invention provides a method and an apparatus for exchanging control functions of volume control units, and a terminal, which relate to the field of communications technologies and can make it more convenient for a user to adjust volume. The method includes: obtaining current placement status information of the terminal, where the placement status information is used to indicate a current placement status of the terminal, and at least two volume control units are disposed in the terminal; and exchanging volume control functions of the at least two volume control units when the current placement status of the terminal deviates from a normal placement status of the terminal, where the at least two volume control units include a volume control unit whose volume control function is to control volume to increase and a volume control unit whose volume control function is to control volume to decrease.

Abstract: The present invention discloses an audio codec with a jack detection function and an audio jack detection method. An embodiment of the audio codec comprises: a first-contact signal input end operable to electrically connect with a sleeve contact of a plug and receive a first-contact signal indicating the presence or absence of the plug; a second-contact signal input end operable to electrically connect with a ring contact or the sleeve contact of the plug and receive a second-contact signal indicating the presence or absence of the plug; a plug detecting circuit coupled to the first-contact or second-contact signal input end and operable to detect whether a voltage level of the first-contact or second-contact signal has changed to thereby generate a plug detection signal; and a control circuit coupled to the plug detecting circuit and operable to determine the presence or absence of the plug according to the plug detection signal.

Abstract: A wearable sound box apparatus is disclosed. In one embodiment, the disclosed sound box apparatus may include a wearable element having both ends thereof extended to form two sound box accommodating areas. The wearable element may be structurally hollow, allowing for a printed circuit board (PCB), a battery, and a reflection tube to be placed therein. A sound unit placed within each of the sound box accommodating areas may be electrically connected to the PCB through a signal line. When the sound unit operates, sound waves from the sound unit are reflected within the wearable element and corresponding inverted sound waves are transmitted out of the wearable element by the reflection tube.

Abstract: A hearing aid system is enabled for binaural microphone adjustment by way of the wearer's own voice. The hearing aid system has a first and a second hearing aid having a first and a second acousto-electric converter. The converters have a signal link with a signal processing unit which determines and compensates for signal deviations caused by the first and second acousto-electric converters on the basis of signals from the wearer's own voice. The wearer's own voice is recognized by a special facility.

Abstract: An audio device adapted for altering the structure of water is provided. The device includes a rigid, enclosed cabinet with speakers mounted to the cabinet. The speakers are positioned on opposing walls, facing each other, within the cavity of the cabinet and a watertight container is placed between the speakers. The exterior surface of the container rests directly against with the cone or the voice coil of the speakers, which are connected to an external sound system. Because the container is flush against the surface of the speakers, vibrations from the sound waves generated by the cone or the voice coil are transferred directly through the water. The sound waves alter the structure of the water from a liquid form to a liquid crystalline form, such that the liquid crystalline takes coherent geometric shapes.

Abstract: A hearing aid has an acoustoelectric transducer, a pickup device for picking up an electrical or electromagnetic signal and converting it into an electrical signal. The hearing aid further has a signal processing device, an electromechanical transducer for outputting an acoustic signal, and a signal estimation device. The signal estimation device determines a correlation between signals of the acoustoelectric transducer and of the pickup device. The signal processing device adjusts a mixing ratio of the signals in the output signal depending on the correlation.

Abstract: Technologies pertaining to provision of customized audio to each listener in a plurality of listeners are described herein. A sensor outputs data that is indicative of locations of multiple listeners in an environment. The data is processed to determine locations and orientations of the respective heads of the multiple listener in the environment. Based on the locations and orientations of heads of the listeners in the environment, for each listener, respective customized audio signals are generated. The customized audio signals are transmitted to respective beamforming transducers. The beamforming transducers directionally output customized beams for the first listener and the second listener based upon the customized audio signals and locations of the heads of the listeners.