Abstract: A wireless earpiece includes a wireless earpiece housing, a processor disposed within the wireless earpiece housing, at least one microphone operatively connected to the processor, and at least one speaker operatively connected to the processor. The processor is configured to receive audio from the at least one microphone, perform processing of the audio to provide processed audio, and output the processed audio to the at least one speaker. The processing of the audio involves identifying body generated sounds generated by a body of a user of the wireless earpiece and removing the body generated sounds.

Abstract: Distributed audio mixing may include transmitting a set of parameters from a local location to one or more remote locations at least multiple miles away from the local location for, at each of the one or more remote locations, one or more remote audio sources to be processed according to the parameters to produce respective one or more remote audio mixes; processing one or more local audio sources according to the parameters to produce a local audio mix; receiving the one or more remote audio mixes; and locally summing the one or more remote audio mixes to the local audio mix to obtain a final audio mix.

Abstract: A sound signal processing method includes: obtaining a plurality of sound signals respectively collected by a plurality of microphones arranged in a space; adjusting respective levels of the plurality of sound signals in accordance with respective positions of the plurality of microphones; mixing the plurality of sound signals having the adjusted respective levels to thereby obtain a mixed signal; and generating a reflected sound by using the obtained mixed signal.

Abstract: Provided are an audio circuit and a mobile terminal having the audio circuit. The audio circuit includes an audio codec chip circuit, a radio frequency circuit and a main circuit board reference ground; the sub-circuit board is provided with a microphone circuit, an antenna circuit, a sub-circuit board reference ground, a decoupling capacitor and a first anti-static device; where the audio codec chip circuit is configured to provide a bias voltage for the sub-circuit board through a transmission line in the main circuit board; the decoupling capacitor and the first anti-static device are connected in parallel and then are connected to the transmission line, and the transmission line is connected to the main circuit board reference ground through the decoupling capacitor and the first anti-static device which are connected in parallel.

Abstract: A vehicle system includes a controller that is configured to (a) store a plurality of passenger profiles regarding passengers of a vehicle where each of the plurality of passenger profiles include pre-stored characteristics of a respective passenger associated therewith, (b) acquire passenger identifying data regarding a present passenger in the vehicle, (c) identify a respective passenger profile associated with the present passenger from the plurality of passenger profiles based on the passenger identifying data, and (d) control a speaker positioned within the vehicle based at least in part on the respective passenger profile to emit noise-canceling sound waves to generate a noise suppression zone to suppress sound waves perceived by the present passenger.

Abstract: A sound signal processing device includes a plurality of sound signal processing units and a plurality of buffers. Each of the plurality of sound signal processing units belongs to one of a plurality of groups and executes a predetermined process by using a supplied sound signal. The plurality of buffers are provided with mutually different buffer sizes where each of the plurality of buffers is associated with one of the groups and supplies an accumulated sound signal to a sound signal processing unit that belongs to an associated group.

Abstract: An audio playback device comprises a microphone, a speaker, and a processor. The processor is arranged to output by the speaker first audio content and receive by the microphone an indication of the first audio content. A first acoustic response of a room in which the audio playback device is located is determined based on the received indication of first audio content. A mapping is applied to the first acoustic response to determine a second acoustic response. The second acoustic response is indicative of an approximated acoustic response of the room at a spatial location different from a spatial location of the microphone. The second audio content output by the speaker is adjusted based on the second response.

Abstract: A loudspeaker described herein have a radiation pattern which is constant over a wide frequency range without requiring any attenuation. The system includes plurality of drivers not uniformly arranged so that the relative velocity of the speaker follows the Legendre shading function. By making the driver density proportional to the Legendre function SSCBT allows each driver to play at max volume. The purpose behind Space Shaded Constant Beamwidth Transducer (SSCBT) is to replace attenuation with incremental spacing between the drivers. Alternatively, the angles each driver is placed by doubling the distance each driver is placed to accomplish the region which is 3 db lower. When the distance is doubled, the angle between each driver increases the further from 0 it is and is consistent with the Legendre function on its surface.

Abstract: The various implementations described herein include methods, devices, and systems for automatic audio equalization. In one aspect, a method is performed at an electronic device that includes speakers, microphones, processors and memory. The electronic device outputs audio user content from the speakers and automatically equalizes subsequent audio output of the device without user input. The automatic equalization includes: (1) obtaining audio content signals, including receiving outputted audio content at each microphone; (2) determining from the audio content signals phase differences between microphones; (3) obtaining a feature vector based on the phase differences; (4) obtaining a frequency correction from a correction database based on the obtained feature vector; and (5) applying the obtained frequency correction to the subsequent audio output.

Abstract: This disclosure relates to a signal generating circuit and an audio processing device. The circuit includes a switch module, a voltage producing module, and a signal generating module; the switch module is connected to the voltage producing module, including at least one control switch, and is used for receiving a frequency division signal. Based on the frequency division signal, the at least one control switch is turned on or turned off; the voltage producing module is separately connected to the switch module and the signal generating module and used for producing a first voltage and a second voltage. The at least one control switch controls the first voltage and the second voltage to change. The signal generating module is connected to the voltage producing module and used for generating a carrier signal with the same frequency as the frequency division signal according to the received first and second voltages.

Abstract: A method is given for a broadband constant beam width acoustic array using shading function parameters for a three dimensional axially symmetric beam. Coefficients are calculated for an estimated shading function fitting the parameters that is a summation of Legendre polynomial orders. The number of orders is user specified. Null bearing locations can be determined from the parameters or from the shading function. A constant beam width shading function can be created from these parameters and used as amplifications and attenuations for a conical spherical array of transducers. The array can be truncated at the null bearing locations. The estimated shading function can be further refined by provided additional Legendre polynomial orders.

Abstract: One or more computer processors detect a contextual need for sound generation on a rollable display device, wherein the rollable display device comprises an array of micro-speakers mounted on one or more microfluidics panels and a plurality of embedded piezoelectric strips. The one or more computer processors identify one or more environmental parameters associated with an environment surrounding the rollable display device. The one or more computer processors determine a rolling profile associated with the rollable based on the detected contextual need and the one or more identified environmental parameters. The one or more computer processors roll the rollable display device based on the determined rolling profile utilizing the plurality of piezoelectric strips. The one or more computer processors adjust an output direction of each micro-speaker in the array of micro-speakers utilizing the respective microfluidics panel.

Abstract: Embodiments are provided for intelligently activating an amplifier in a playback device based on proximity detection. The playback device may be in a quasi-idle state when the playback device is not rendering media content. The quasi-idle state of the playback device may involve an amplifier in the playback device being inactive, while some other components or modules of the playback devices remain active. The playback device may include a proximity sensor configured to detect movement relative to the playback device. If movement is detected indicating that a user input to cause the playback device to render media content is anticipated, the amplifier may be pre-emptively activated such that the playback device enters an active state from the quasi-idle state. In some cases, the playback device may send a message to one or more other playback devices to cause the other playback devices to enter an active state.

Abstract: An integrated audio-visual system is disclosed for delivering an event to one or more spectators. The integrated audio-visual system includes one or more loudspeakers that are positioned behind one or more visual displays to be effectively behind the one or more visual displays to be shielded from the field of view of the one or more spectators. The one or more visual displays are specially designed and manufactured to allow sound associated with the event to propagate from the one or more loudspeakers with minimal acoustical distortion and/or minimum acoustical vibration while presenting a visual representation of the event to the one or more spectators. Moreover, the one or more loudspeakers and the one or more visual displays are situated to be a predetermined displacement from each other to further minimize acoustical distortion.

Abstract: The present invention provides a signal processing device that allows a user, who adjusts a signal level to match a reference level, to adjust the signal level while checking a degree of difference between the signal level and the reference level. A signal processing device according to the present invention comprises: an input part 20 to which an input signal is input; a storage 40 that stores therein one or more setting values of a parameter to be used for amplifying the input signal and a reference value of the setting value; an amplification part 30 that generates an output signal by amplifying a level of the input signal based on the setting value; an output part 70 that outputs the output signal; an adjustment part 50 that adjusts the setting value; and a display 60 that displays a display result determined based on the setting value adjusted by the adjustment part and the reference value.

Abstract: A remote speaker microphone (RSM) unit enables a user to use either a headset or an in-ear speaker with the RSM unit. The RSM unit includes a first RSM connection for communicating and connectable with the first connection of the headset; and a second RSM connection for communicating and connectable with the second connection of the in-ear speaker. Connecting either the headset or the in-ear speaker to the RSM unit disables the speaker and microphone of the RSM unit. At least one of the first and second connections are wireless, e.g., Bluetooth or NFC, and at least one of the first and second RSM connections are wireless connections communicable with the first and second connections, respectively. In addition or the alternative, at least one of said first connection and said second connection are electrical plugs and at least one of said first and second RSM connections are corresponding electrical jacks.

Abstract: A hearing protection system, a hearing protection device and related method, the hearing protection device comprising an earpiece device comprising a first connector, and first and second earpieces, the first earpiece comprising a first primary microphone and a first receiver wired to the first connector and the second earpiece comprising a second primary microphone and a second receiver wired to the first connector; and a processing device comprising a processor, an earpiece interface, and a wireless communication interface, wherein the processor is configured to receive a first microphone input signal from the first primary microphone; process the first microphone input signal for provision of a first primary output signal for the first receiver based on the first microphone input signal; receive a communication request from a communication device via the communication interface; and in response to receipt of the communication request, communicate with the communication device via the communication interfac

Abstract: In some examples, an audio sending device receives a stream of application audio data, encodes the stream of application audio data, and in response to detecting an end of the stream of application audio data, provides pre-encoded filler audio data from a buffer in the audio sending device as an encoded stream of filler audio data. The audio sending device transmits the encoded stream of application audio data and the encoded stream of filler audio data in an encoded output data stream over a transport to an audio receiving device.

Abstract: A device is configured to (i) receive media content from a media source, (ii) generate a first series of frames including first portions of the media content and first playback timing information, (iii) generate a second series of frames including second portions of the media content and second playback timing information, (iv) transmit the first series of frames to a first playback device in a synchrony group for playback of the first portions of the media content in accordance with the first playback timing information, and (v) transmit the second series of frames to a second playback device in the synchrony group for playback of the second portions of the media content in accordance with the second playback timing information such that the media content is played back in synchrony by the synchrony group.

Abstract: A system for capturing sound comprising a plurality of discrete microphones (112, 14, 116, 118) and a processing system (408). The plurality of discrete microphones are arranged in a circular array. The processing system (408) arranged to perform a first signal processing algorithm on sound originating from one or more of a first set of directions relative to the array to isolate a first sound source. The processing system (408) is further arranged to perform a second signal processing algorithm on sound originating from one or more of a second set of directions relative to the array to isolate a second sound source therein. A method for receiving sound at a plurality of discrete microphones (112, 114, 116, 118) arranged in a circular array is also described.