Abstract: In a device or method for rendering a sound program for headphones, a location is received for placing the sound program with respect to first and second ear pieces. If the location is between the first ear piece and the second ear piece, the sound program is filtered to produce low-frequency and high-frequency portions. The high-frequency portion is panned according to the location to produce first and second high-frequency signals. The low-frequency portion and the first high-frequency signal are combined to produce a first headphone driver signal to drive the first ear piece. A second headphone driver signal is similarly produced. The sound program may be a stereo sound program. The device or method may also provide for a location that is between the first ear piece and a near-field boundary. Other aspects are also described.

Abstract: An acoustic processing device including a memory storing instructions and a processor that implements the stored instructions to execute a plurality of tasks, the tasks including: an analyzing task that analyzes an input signal; a determining task that determines an acoustic effect to be applied to the input signal, from among a first acoustic effect of virtual surround and a second acoustic effect of virtual surround different from the first acoustic effect, based on a result of the analyzing task; and an acoustic effect applying task that applies the acoustic effect determined by the determining task to the input signal.

Abstract: There is provided encoding and decoding methods for encoding and decoding of object based audio. An exemplary encoding method includes inter alia calculating M downmix signals by forming combinations of N audio objects, wherein M?N, and calculating parameters which allow reconstruction of a set of audio objects formed on basis of the N audio objects from the M downmix signals. The calculation of the M downmix signals is made according to a criterion which is independent of any loudspeaker configuration.

Abstract: An audio signal adjustment method, a storage medium, and a terminal are provided. The method may include: when it is detected that a pre-set event is triggered, controlling a microphone to acquire an audio signal, the pre-set event comprising a pre-set call event and/or a pre-set voice recording event; analyzing the sound loudness corresponding to the audio signal; dynamically adjusting, according to the analysis result and a pre-set adjustment policy, the sound loudness corresponding to the audio signal; and performing, according to the type of the pre-set event, corresponding output processing on the adjusted audio signal.

Abstract: A microphone array includes first and second microphones (Ma, Mb) placed on a first axis (f), a third microphone (Mc) placed on a plane (fg) formed by the first axis and a second axis (g) and at a position other than on the first axis, and a fourth microphone (Md) placed on a third axis (h), and at a position other than on the plane formed by the first and the second axes, and a processing circuit generates signals (Cx, Cy, Cz) having bidirectionality in first, second and third mutually perpendicular directions (x, y, z), and an omnidirectional signal (Cw), based on signals (Ba to Bd) obtained by sound collection by means of the first to fourth microphones. It is possible to generate signals having bidirectionality in mutually perpendicular directions, and an omnidirectional signal, without using special microphones, and without excessive restrictions with regard to the placement of the microphones.

Abstract: An audio decoder device for decoding a compressed input audio signal having at least one core decoder having one or more processors for generating a processor output signal based on a processor input signal, wherein a number of output channels of the processor output signal is higher than a number of input channels of the processor input signal, wherein each of the one or more processors has a decorrelator and a mixer, wherein a core decoder output signal having a plurality of channels has the processor output signal, and wherein the core decoder output signal is suitable for a reference loudspeaker setup; at least one format converter device configured to convert the core decoder output signal into an output audio signal, which is suitable for a target loudspeaker setup; and a control device configured to control at least one or more processors in such way that the decorrelator of the processor may be controlled independently from the mixer of the processor, wherein the control device is configured to contro

Abstract: A method and apparatus for reconstructing N audio channels from M audio channels is disclosed. The method includes receiving a bitstream containing an encoded audio signal representing the M audio channels and decoding the encoded audio signal to obtain a frequency domain representation of the M audio channels. The method further includes extracting a parameter from the bitstream and reconstructing at least one of the N audio channels using the parameter. The parameter represents an angle between two signals, at least one of which is included in the M audio channels.

Abstract: An exemplary behind-the-ear (“BTE”) hearing device has a side specific structure and includes a housing comprising: a proximal portion configured to be positioned towards an upper part of an ear of a recipient, the proximal portion having a curved shape configured to curve within a first reference plane such that the proximal portion is configured to follow a median shape of the ear when the BTE hearing device is worn by the recipient; and a distal portion configured to extend behind the ear when the BTE hearing device is worn by the recipient, the distal portion configured to extend within a second reference plane that intersects the first reference plane at an angle such that the distal portion is bent with respect to the proximal portion and is configured to extend toward a skull of the recipient when the BTE hearing device is worn by the recipient.

Abstract: Systems, apparatuses, and methods are described for a privacy blocking device configured to prevent receipt, by a listening device, of video and/or audio data until a trigger occurs. A blocker may be configured to prevent receipt of video and/or audio data by one or more microphones and/or one or more cameras of a listening device. The blocker may use the one or more microphones, the one or more cameras, and/or one or more second microphones and/or one or more second cameras to monitor for a trigger. The blocker may process the data. Upon detecting the trigger, the blocker may transmit data to the listening device. For example, the blocker may transmit all or a part of a spoken phrase to the listening device.

Abstract: A multi-purpose Bluetooth interface and controller includes a microcontrol unit and at least one first microphone, the first microphone being electrically connected to the MCU and configured to collect sound signals and convert the sound signals into electrical signals that are received by the MCU; wherein the Bluetooth interface and controller further includes at least one connection port which is electrically connected to the MCU and is configured to connect an external device that includes a wired microphone, the wired microphone also being configured to collect sound signals that are received by the MCU; and the MCU encodes electrical sound signals received from the first microphone and the external wired microphone, and transmits the encoded sound signals in uplink to a mobile terminal via a Bluetooth wireless connection established between the MCU and the mobile terminal for subsequent processing by the mobile terminal.

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.

Abstract: Various embodiments of the present invention comprise: a circuit board comprising a first grounding portion and a second grounding portion; a wireless communication circuit arranged in a first area of the circuit board, the wireless communication circuit comprising a first grounding terminal electrically connected to the first grounding portion; an audio codec circuit arranged in a second area of the circuit board, the audio codec circuit comprising a signal input terminal and a second grounding terminal electrically connected to the second grounding portion; and a connector electrically connected to the input terminal, the connector comprising a first pin that can be electrically connected to a signal terminal of a microphone included in an external electronic device and a second pin that can be electrically connected to a grounding terminal of the microphone, the second pin being electrically connectable to the second grounding portion.

Abstract: A mobile computing device includes a digital signal processor, a first speaker having a first sensitivity and a second speaker having a second sensitivity, where the second sensitivity is greater than the first sensitivity.

Abstract: An electronic device includes a first microphone, a second microphone, a digital signal processor, and a detector. The first microphone samples a sound from a sound source at a first location to generate a first signal. The second microphone samples the sound from the sound source at a second location spaced from the first location to generate a second signal. The digital signal processor compares the first signal and the second signal to obtain external environment information associated with a location of the sound source. The detector determines a call mode, based on the external environment information.

Abstract: An active noise control unit generates a noise canceling sound of a single frequency that cancels engine sound heard by an occupant and outputs the noise canceling sound to a speaker through an amplifier. A second harmonic detector and a third harmonic detector detect a magnitude of a harmonic component of the noise canceling sound included in sound collected by a microphone. A controller controls a frequency of the noise canceling sound generated by the active noise control unit to a fundamental frequency of the engine sound based on a rotation frequency of the engine, and in addition, causes an output level control unit to control a level of the noise canceling sound to be output to the amplifier so that the level of the noise canceling sound becomes higher when the magnitude of the harmonic component exceeds a predetermined threshold value.

Abstract: An active vibratory noise reduction system includes: a canceling vibratory sound generator; an error signal detector for detecting a canceling error between the canceling vibratory sound and a vibratory noise as an error signal; and an active vibratory noise controller for generating, based on the error signal, a control signal for controlling the canceling vibratory sound generator. The active vibratory noise controller is provided with a stability improving unit including: a correction value generation unit for generating an error signal correction value by multiplying a reaching control sound estimation value by a stabilization coefficient; an error signal correction unit for correcting the error signal by using the error signal correction value to generate a corrected error signal, and a stabilization coefficient updating unit for sequentially updating the stabilization coefficient based on the corrected error signal and the reaching control sound estimation value by using an adaptive algorithm.

Abstract: Methods and systems are provided for providing one or more sound component to a viewer within a media presentation space by a robot having one or more speakers. A method includes an operation for establishing a connection to a media device that is presenting media content to the viewer, and an operation for capturing input from a plurality of sensors for obtaining a map of the media presentation space. The method additionally provides for identifying a position of the viewer and a speaker configuration in the media presentation space. The method allows the robot to move to a location based on the map of the media presentation space, the position of the viewer and the speaker configuration. Moreover, the method provides an operation for receiving one or more sound components for presentation to the user.

Abstract: An apparatus including an interface and a processor. The interface may be configured to receive video frames corresponding to an interior of a vehicle. The processor may be configured to perform video operations on the video frames to detect objects in the video frames, detect one or more passengers based on the objects detected in the video frames, determine a location of each of the passengers detected, determine which of the passengers may be an audio source and generate a control signal in response to the audio source. The control signal may be configured to adjust an input of a microphone towards the audio source. The video operations may be used to determine the location and the audio source.

Abstract: Determination of material acoustic parameters for a headset is presented herein. A value of a material acoustic parameter is initialized. A simulation is performed using the value of the material acoustic parameter and a model. The model includes a three-dimensional representation of a local area occupied by the headset. During the simulation, the value of the material acoustic parameter is dynamically modified until a reverberation time calculated based on the modified value of the material acoustic parameter falls within a threshold value of a target reverberation time. The model is updated with the modified value of the material acoustic parameter. The model is used to determine one or more acoustic parameters. Audio content is rendered based on the one or more acoustic parameters so that the audio content appears originating from an object in the local area.

Abstract: A sound processing apparatus includes n number of microphones that are disposed correspondingly to n number of persons and that mainly collect sound signals uttered by respective relevant persons, a filter that suppresses crosstalk components included in a talker sound signal collected by a microphone corresponding to at least one talker using the sound signals collected by the n number of microphones, a parameter updater that updates a parameter of the filter for suppressing the crosstalk components and stores an update result in the memory in a case where a predetermined condition including time at which at least one talker talks is satisfied, and a sound output controller that outputs the sound signals, acquired by subtracting the crosstalk components by the filter from the talker sound signals based on the update result, from a speaker.