A method for designing a line array loudspeaker arrangement comprises providing a loudspeaker arrangement based on design start parameters and including at least a vertical front array and measuring the frequency responses of the loudspeaker arrangement with bypassed or omitted electronic filters at predefined horizontal angle increments. The method further comprises computing combined beam forming and crossover filter frequency responses for the vertical front array based on the measured frequency responses of the loudspeaker arrangement and first target frequency responses at various frequency points and various positions; and computing combined equalizing and crossover filter frequency responses for the vertical front array based on second target frequency responses, the combined equalizing and crossover filter frequency responses being configured to obtain acoustic linear phase responses of the loudspeaker arrangement.

An earphone including a speaker that outputs sounds to a uniaxial direction; a sound guide tube offset from a center of the speaker to and disposed on one side of a direction orthogonal to the uniaxial direction, the sound guide tube extending to a direction parallel to the uniaxial direction; and a housing accommodating the speaker and supporting a base end of the sound guide tube, wherein the housing is broadened from the base end to other side opposite to the one side, toward a location where the speaker is accommodated, and has an inclined surface that inclines at the broadened location, and a separation distance between a central axis of the speaker and a central axis of the sound guide tube is smaller than an internal radius of the sound guide tube.

An earphone including a speaker that outputs sounds to a uniaxial direction; a sound guide tube offset from a center of the speaker to and disposed on one side of a direction orthogonal to the uniaxial direction, the sound guide tube extending to a direction parallel to the uniaxial direction; and a housing accommodating the speaker and supporting a base end of the sound guide tube, wherein the housing is broadened from the base end to other side opposite to the one side, toward a location where the speaker is accommodated, and has an inclined surface that inclines at the broadened location, and a separation distance between a central axis of the speaker and a central axis of the sound guide tube is smaller than an internal radius of the sound guide tube.

A computing device with a microphone system is disclosed. The computing device includes a microphone system with an environment microphone and a noise microphone. The environment microphone picks up an environment microphone signal which includes (1) a desired signal component based on desired sound and (2) a noise component based on noise from a noise source. The noise microphone picks up a noise microphone signal based on the noise, and is configured such that contributions to the noise microphone signal from the desired sound, if present, are attenuated relative to the environment microphone. A controller receives and processes time samples from the noise microphone signal to yield a noise estimation of the noise component. The estimation is subtracted from the environment microphone signal to yield and end-user output.

A method for operating a bone conduction communication system can include establishing a communicable connection, operating a transducer in an input mode wherein the bone conduction transducers are configured to detect a vibration associated with a bone of the user; transmitting an audio signal over the communicable connection; and operating the transducers responsive to the audio signal.

A method for operating a bone conduction communication system can include establishing a communicable connection, operating a transducer in an input mode wherein the bone conduction transducers are configured to detect a vibration associated with a bone of the user; transmitting an audio signal over the communicable connection; and operating the transducers responsive to the audio signal.

The present disclosure provides an acoustic output apparatus including one or more status sensors, at least one low-frequency acoustic driver, at least one high-frequency acoustic driver, at least two first sound guiding holes, and at least two second sound guiding holes. The status sensors may detect status information of a user. The low-frequency acoustic driver may generate at least one first sound, a frequency of which is within a first frequency range. The high-frequency acoustic driver may generate at least one second sound, a frequency of which is within a second frequency range including at least one frequency exceeding the first frequency range. The first and second sound guiding holes may output the first and second spatial sound, respectively. The first and second sound may be generated based on the status information, and may simulate a target sound coming from at least one virtual direction with respect to the user.

A microphone may comprise a microphone element for detecting sound, and a digital signal processor configured to process a first audio signal that is based on the sound in accordance with a selected one of a plurality of digital signal processing (DSP) modes. Each of the DSP modes may be for processing the first audio signal in a different way. For example, the DSP modes may account for distance of the person speaking (e.g., near versus far) and/or desired tone (e.g., darker, neutral, or bright tone). At least some of the modes may have, for example, an automatic level control setting to provide a more consistent volume as the user changes their distance from the microphone or changes their speaking level, and that may be associated with particular default (and/or adjustable) values of the parameters attack, hold, decay, maximum gain, and/or target gain, each depending upon which DSP is being applied.

A microphone may comprise a microphone element for detecting sound, and a digital signal processor configured to process a first audio signal that is based on the sound in accordance with a selected one of a plurality of digital signal processing (DSP) modes. Each of the DSP modes may be for processing the first audio signal in a different way. For example, the DSP modes may account for distance of the person speaking (e.g., near versus far) and/or desired tone (e.g., darker, neutral, or bright tone). At least some of the modes may have, for example, an automatic level control setting to provide a more consistent volume as the user changes their distance from the microphone or changes their speaking level, and that may be associated with particular default (and/or adjustable) values of the parameters attack, hold, decay, maximum gain, and/or target gain, each depending upon which DSP is being applied.

The present application relates to the technical field of loudspeakers. Disclosed are a loudspeaker device, a method, apparatus and device for adjusting the sound effect thereof, and a medium. The method is used for increasing the diversity of the sound effect of the loudspeaker device, and comprises: determining the spatial distribution state of multiple loudspeaker units in the loudspeaker device; determining a sound effect mode corresponding to the spatial distribution state; and adjusting the sound effect of the loudspeaker device according to the sound effect mode. According to the present application, the sound effect mode is accordingly adjusted according to the spatial distribution state of multiple loudspeaker units in the loudspeaker device. In this way, when the spatial distribution state changes, the sound effect mode of the loudspeaker device also changes, thereby overcoming the defect of the single sound effect of the loudspeaker device, i.e., according to the present application, the diversity of the sound effect of the loudspeaker device can be increased efficiently.