Abstract: An audio accessory and an audio output method are provided. The audio accessory includes a connector configured to be connected to an external device, an audio output unit configured to output a sound, and a switching unit configured to select a signal path according to whether the sound output through the audio output unit is in a balanced mode or an unbalanced mode.

Abstract: The present invention relates to a method of noise reduction including the steps of filtering reference signals and representing noise by an adaptive filter comprising adaptive filter coefficients to obtain actuator driving signals, outputting the actuator driving signals by loudspeakers to obtain loudspeaker signals. The method further includes detecting the loudspeaker signals by microphones and filtering the reference signals by estimated transfer functions representing the transfer of the loudspeaker signals output by the loudspeakers to the microphones to obtain filtered reference signals. The method further includes updating the filter coefficients of the adaptive filter based on the filtered reference signals and based on the previously updated filter coefficients of the adaptive filter multiplied by leakage factors.

Abstract: An electronic device including a voice obtaining microphone configured to receive a voice signal including a noise at a first level; a noise obtaining microphone configured to receive a voice signal including a noise at a second level higher than the first level; and a controller configured to estimate the noise at the second level from the voice signal received by the noise obtaining microphone, remove the noise at the first level from the voice signal received by the voice obtaining microphone based on the estimated noise, and perform an operation corresponding to the voice signal having the noise at the first level eliminated therefrom.

Abstract: Technologies pertaining to calibration of filters of an audio system are described herein. A mobile computing device is configured to compute values for respective filters, such as equalizer filters, and transmit the values to a receiver device in the audio system. The receiver device causes audio to be emitted from a speaker based upon the values for the filters.

Abstract: Sound output devices, output systems and methods are provided. The sound output device includes a speaker; a communicator configured to perform communication with an external speaker and a content source that provides sound content to the external speaker; and a processor configured to recognize the external speaker in response to the device entering into a predetermined range from the external speaker, to perform pairing with the external speaker, to receive information about the content source from the external speaker, to receive the sound content from the content source based on the information about the content source, and to output the received sound content through the speaker.

Abstract: The present disclosure provides a therapeutic method applicable to learning, attention, cognitive and developmental disorders. This method combines auditory, visual and vestibular stimuli into a systematic method which can be customized to address individual learning, attention and processing disorders. This method may be wirelessly implemented in a group setting. A device for providing vestibular and somatosensory stimulation through bone conduction of sound waves to skeletal bones is provided in the form of a pad that is applied to or wrapped around the individual in treatment, which is comfortable in use and may contain particulate filler material capable of providing bone conduction stimulation adjacent to the location of a bone conduction transducer and expand the sound conduction site over a larger area.

Abstract: A system may include control circuitry for detecting a plosive event associated with a microphone transducer and in response to the plosive event, causing restoration of acoustic sense operation of the microphone transducer and a processing circuit associated with the microphone transducer. A system for configuring a filter having at least two frequency response configurations to achieve an effective frequency response configuration intermediate to the at least two frequency response configurations may include control circuitry for rapidly switching between the at least two frequency response configurations such that a weighted average frequency response of the filter corresponds to the effective frequency response configuration.

Abstract: Sound output devices, output systems and methods are provided. The sound output device includes a speaker; a communicator configured to perform communication with an external speaker and a content source that provides sound content to the external speaker; and a processor configured to recognize the external speaker in response to the device entering into a predetermined range from the external speaker, to perform pairing with the external speaker, to receive information about the content source from the external speaker, to receive the sound content from the content source based on the information about the content source, and to output the received sound content through the speaker.

Abstract: A D/A converter converts digital audio data DIN into analog differential audio signals VP and VN. A differential to single-ended conversion circuit converts the differential audio signals VP and VN into a single-ended audio signal VSE. A volume circuit receives the single-ended audio signal VSE, and amplifies the single-ended audio signal VSE with a gain that corresponds to a volume value. A reference voltage source generates a reference voltage VREF commonly referred by the differential to single-ended conversion circuit and the volume circuit. In a calibration operation, a calibration circuit controls the D/A converter so as to shift at least one from among the differential audio signals VP and VN such that the difference between the output voltage VSE of the differential to single-ended conversion circuit and the reference voltage VREF approaches zero.

Abstract: This application describes methods and apparatus for mitigating the effects of crosstalk in multichannel audio. An audio driver circuit (200) for driving first and second audio loads (103) having a common return path (RC), has first and second signal paths (Left and Right). A crosstalk compensation block (205) is configured to add a first compensation signal to the first signal path and add a second compensation signal to the second signal path. The first compensation signal is generated based on the second audio signal and a first compensation function and the second compensation signal is generated based on the first audio signal and a second compensation function. Each of the first and second compensation functions is based on a predetermined impedance value for at least part of the common return path (RH1) and is also based on a determined DC impedance value (ZL, ZR) for one of the first and second audio loads which is modified by a band correction factor (?).

Abstract: A class-D amplifier includes an error amplification circuit, a duty signal generator, a level selection circuit, a driver and control block and an output stage. The class-D amplifier divides peal levels of an error signal into multi-level and changes a scheme for modulating the error signal when the error signal crosses each level boundary of the multi-level thereby to have an effect such as the error signal is folded. Therefore, the class-D amplifier drives output nodes with multi-level and thus the class-D amplifier may increasing efficiency while reducing EMI.

Abstract: A microphone device including a first sound receiving module, a second sound receiving module and a sound collecting trough is provided. The first sound receiving module receives a sound signal to output a first electronic signal. The second sound receiving module receives the sound signal to output a second electronic signal. The first sound receiving module is coupled to the second sound receiving module, and the phase of the first electronic signal and the phase of the second electronic signal are inverse to each other. A distance between the first sound receiving module and the sound collecting trough is smaller than a distance between the second sound receiving module and the sound collecting trough, and another sound signal is transferred to the first sound receiving module through the sound collecting trough.

Abstract: Digital filters for processing an input audio signal for extending the audio capabilities and reducing distortion of a loudspeaker. Such extension may include a general sound pressure level (SPL) extension or a more targeted low frequency (LF) extension. This may be accomplished by modifying an original frequency response of the loudspeaker to achieve a particular extension, and applying a digital linear filter based on the modified frequency response to the audio signal. A nonlinear digital filter may also be applied to the audio signal reduce nonlinear distortions of the loudspeaker. The nonlinear digital filter may be based on an inverse of an electro-mechanical model of the loudspeaker. In this manner, a loudspeaker may be driven close to its maximum theoretical power capability without increasing distortion in the loudspeaker output.

Abstract: An earphone apparatus includes at least one sound broadcasting device. Each of the sound broadcasting device includes a main shell, a plurality of first electrodes, a speaker device and an expanding speaker device. The main shell has a first connector. The speaker device is disposed in the main shell, and receives a sound signal through a signal transmitting wire coupled to the first electrodes. The expanding speaker device has a second connector, and the expanding speaker device is detachably connected to the first connector through the second connector. When the second connector is connected to the first connector, the expanding speaker device is electronically connected to the first electrodes, and receives the sound signal transmitted by the signal transmitting wire.

Abstract: Systems, methods, and apparatus are provided for recording high output power levels of sound at low sound pressure levels. For example, an apparatus comprises an enclosure, a speaker disposed within the enclosure, a microphone disposed within the enclosure, and an evacuation port. The evacuation port is configured to connect to a system that reduces a pressure level within the enclosure to a level that is less than an ambient air pressure level outside the enclosure. The enclosure is sealed or otherwise configured to provide a sealed enclosure, to maintain the reduced air pressure within the enclosure. The speaker can be driven by an amplifier at high output power levels to generate a distorted sound of an amplified electric musical instrument for recording purposes, while the reduced pressure level within the enclosure serves to attenuate the sound pressure level and perceived loudness which emanates from the speaker.

Abstract: Example embodiments disclosed herein relate to audio content rendering. A method of rendering audio content is disclosed, which includes determining a priority level for an audio object in the audio content, selecting a rendering mode from a plurality of rendering modes for the audio object based on the determined priority level and rendering the audio object in accordance with the selected rendering mode, the rendering mode indicating an accuracy of the rendered audio object. Corresponding system and computer program product are also disclosed.

Abstract: Disclosed herein are systems and methods for receiving a voice command and determining an appropriate action for the media playback system to execute based on user identification. The systems and methods receive a voice command for a media playback system, and determines whether the voice command was received from a registered user of the media playback system. In response to determining that the voice command was received from a registered user, the systems and methods configure an instruction for the media playback system based on content from the voice command and information in a user profile for the registered user.

Abstract: Methods, computing devices, and systems that apply equalization settings for particular audio devices based on a content audio profile. A computing device may detect at least one audio device connected to the computing device and request a device sound profile of the at least one audio device from a database remote from the computing device. Computing device may receive the device sound profile of the at least one audio device from the remote database. The computing device may receive content comprising an audio component and a video component and a content audio profile associated with the audio component. The computing device can determine whether to adjust equalizer settings for sound reproduction of the audio component based on the audio profile and the device sound profile of the at least one audio device.

Abstract: Systems and methods are disclosed, which include or present “soundscapes” in or for online mapping utilities. To obtain the data for such soundscapes, along with cameras for visual images, a microphone array can be mounted on a vehicle to record sounds along the streets travelled for linking to an online map. A speech recognition algorithm is used to identify private conversations and remove them from the recording. The systems and methods for accomplishing this task include use of an array of microphones mounted in a special pattern with special materials on top of the vehicle to record sounds as the vehicle travels through space and time. A set of signal processing algorithms is also used to process the microphone signals autonomously in real-time, allowing the operator to immediately review them for quality assurance.

Abstract: The behavior of a NOS DAC and an analog filter may be emulated by electronic components of an integrated circuit (IC) by upsampling data and applying a digital filter to the upsampled data. For example, the IC may include a zero-order-hold circuit that upsamples data from a first input sample rate to a second, higher input rate. The upsampled data may be passed to an Asynchronous Sample Rate Converter (ASRC) that performs further upsampling (e.g., from 8*Fs-64*Fs). The upsampled data may be passed to a digital low pass filter. The digital low pass filter may emulate, for example, a response of a fifth order Butterworth analog filter to mimic the effect of analog processing. The IC may integrate the upsampling circuit, the low pass digital filter, a digital-to-analog converter (DAC) and an amplifier to provide an audio solution for playing high-fidelity music in a mobile device.