Abstract: A method, apparatus and computer program product are provided to permit audio signals to provide additional information to a user regarding the distance to the source of the audio signals, thereby increasing a user's situational awareness. In the context of a method, a distance and a direction from a user to an object are determined. The method also scales the distance to the object to create a modified distance within a predefined sound field region about the user. The method also causes an audio cue relating to the object to be audibly provided to the user. The audio cue is such that the object appears to be located within the predefined sound field region in the direction and at the modified distance from the user from the user.

Abstract: The present disclosure provide a volume display method and apparatus for smoothly and flexibly displaying a volume change on the premise of ensuring high time effectiveness and accuracy. The method includes: acquiring a volume value of a first audio input during a current time cycle and a volume value of a second audio input during a previous time cycle; determining volume values to be displayed according to the volume values of the first audio input and the second audio input as well as a volume unit or a threshold of the quantity of volume values to be displayed; determining, according to the quantity of the volume values and duration of the time cycle, a unit display time of volume values to be displayed; sequencing all volume values to be displayed; and successively displaying an image that corresponds to each volume value on a device screen according to the unit display time and the sequence.

Abstract: A wireless sound-emitting device includes a housing adapted to be coupled to a wall at a source of electric power, a loudspeaker positioned at a periphery of the housing, a control module outputting an electric audio signal to the at least one loudspeaker, and a wireless communications module in electrical communication with the control module. The loudspeaker emits acoustic signals in a direction parallel to the wall, when the housing is coupled to the wall, with the acoustic signals reflecting off the wall. The device may produce a sound masking noise or play a sound recorded on an internal memory. The device may include an electric plug or be adapted to replace an electric outlet faceplate. The device may have electric pass-through outlets and may be powered by the source of electric power. The device may be controlled remotely, for example via an Internet of Things (IoT) platform.

Abstract: A single knob for controlling at least an audio input signal to produce up to fourteen distinct results, plus one quiescent state, through axial translational states corresponding to: the knob being translationally biased out, pushed in, or pushed in and held; and rotational states corresponding to: rotationally biased to a first position, turned clockwise, turned clockwise and held, turned counterclockwise, and turned clockwise and held, and combinations of those translational and rotational states. The single knob controller can be linked to a Bluetooth device for controlling such device. Some applications include automotive electronics and audio systems generally.

Abstract: An audio processing system, includes: an audio collector that collects audio in a non-directivity state using audio collection elements; an operator receives an input of one or more designation directions for audio emphasis for switching from the non-directivity state to a directivity state; an emphasis processor that generates audio data in the directivity state obtained by performing an emphasis process on the audio data in the designation direction from the audio collector using audio data collected by the audio collector according to the input of the designation direction; a volume adjustor that adjusts volume of the audio data in a directivity state; and an audio output that outputs the audio in a non-directivity state collected by the audio collector or the audio in a directivity state after the volume has been adjusted by the volume adjustor.

Abstract: A power converter may include a power inductor, a plurality of switches arranged to sequentially operate in a plurality of switch configurations, an output for producing the output voltage, wherein a first switch is coupled to a first output terminal of the output and a second switch is coupled to a second output terminal of the output, and a linear amplifier coupled to the output. The controller may be configured to, in a linear amplifier mode of the power stage, enable the linear amplifier to transfer electrical energy from an input source of the power stage to the load, and in at least one mode of the power stage other than the linear amplifier mode, sequentially apply switch configurations from the plurality of switch configurations to selectively activate or deactivate each of the plurality of switches in order to transfer the electrical energy from the input source to the load.

Abstract: According to one embodiment, an active noise-reduction apparatus includes following units. The reference signal generation unit generates different reference signals based on target sound generated from a sound source. The filter processing unit generates first control signals by filtering the reference signals using first digital filters. The averaging unit generates a second control signal by averaging the first control signals. The control speaker outputs the second control signal as control sound. The error microphone detects a synthetic sound pressure of the target sound and the control sound to generate an error signal. The filter update unit updates the first digital filters so that the error signal is minimized.

Abstract: In accordance with systems and methods of the present disclosure, an adaptive noise cancellation system may include an alignment filter configured to correct misalignment of a reference microphone signal and an error microphone signal by generating a misalignment correction signal.

Abstract: In a wireless content sharing system, audio may be captured at various levels of a source device, including at an application level. Audio may also be divided into components prior to packetization and transmission, allowing different channels of audio to be sent to different target devices. Audio may be sent with timing information to coordinate playback of content. Audio may be buffered to reduce user noticeable latency.

Abstract: An example method may involve a device determining a first loudness representation for a playback device based on a first equalization setting applied to a representation of average music. The device may also determine a second loudness representation for the playback device, based on a second equalization setting applied to the representation of average music. The device may also determine a loudness adjustment factor based on the first and second loudness representations, and then causing the playback device to play back media based on the second equalization setting and the determined loudness adjustment factor.

Abstract: A mixing signal processing technique modifies digital audio recordings to simulate the linear and nonlinear effects of propagation and mixing of sounds in air. When multiple sounds or complex sounds comprised of multiple frequencies in the audible spectrum propagate in such a nonlinear medium, they transfer energy into sound at new frequencies given by the sums and differences of the original signal frequencies. The mixing signal processing technique may improve the ability of a system to reproduce the effects of a live performance using a digital audio recording.

Abstract: A conversation assistance system with a bi-lateral array of microphones arranged externally of a space that does not include any array microphones, where the space has a left side, a right side, a front and a back, the array comprising a left side sub-array of multiple microphones and a right side sub-array of multiple microphones, where each microphone has a microphone output signal, and a processor that creates from the microphone output signals a left-ear audio signal and a right-ear audio signal. The left-ear audio signal is created based on the microphone output signals from one or more of the microphones of the left-side sub-array and one or more of the microphones of the right-side sub-array and the right-ear audio signal is created based on the microphone output signals from one or more of the microphones of the left-side sub-array and one or more of the microphones of the right-side sub-array.

Abstract: In some embodiments, a method for automatic detection of polarity of speakers, e.g., speakers installed in cinema environments. In some embodiments, the method determines relative polarities of a set of speakers (e.g., loudspeakers and/or drivers of a multi-driver loudspeaker) using a set of microphones, including by measuring impulse responses, including an impulse response for each speaker-microphone pair; clustering the speakers into a set of groups, each group including at least two of the speakers which are similar to each other in at least one respect; and for each group, determining and analyzing cross-correlations of pairs of impulse responses (e.g., pairs of processed versions of impulse responses) of speakers in the group to determine relative polarities of the speakers. Other aspects include systems configured (e.g., programmed) to perform any embodiment of the inventive method, and computer readable media (e.g., discs) which store code for implementing any embodiment of the inventive method.

Abstract: A Bluetooth communication bracelet includes a main body and a Bluetooth earphone. The main body includes a wearing member, a charging plate, an USB interface, and an audio jack. The wearing member defines a receiving groove, a positioning portion and a charging electrode. The receiving groove is located on the outer peripheral of the wearing member, the positioning portion. The charging electrode is located on sidewalls of the receiving groove. The charging plate is located in the wearing member. The USB interface is electrically coupled to the charging plate for charging the charging plate. The Bluetooth earphone, which is received in the receiving groove, includes a carrier, an earplug, and a microphone. The carrier sets a matching portion and a contact electrode. The matching portion matches the positioning portion to make the Bluetooth earphone releasably mounted to the wearing member. The contact electrode is in contact with the charging plate.

Abstract: Disclosed are an apparatus and method of processing an audio signal to optimize audio for a room environment. One example method of operation may include recording the audio signal generated within a particular room environment and processing the audio signal to create an original frequency response based on the audio signal. The method may also include creating at least two iterative filters based on at least two separate frequency ranges of the original frequency response, calculating an error difference between the frequency response modified by the at least two iterative filters and the original frequency response, and applying the error difference to the audio signal.

Abstract: A microphone package is integrated with a built-in speaker driver. A microphone application-specific integrated circuit (ASIC) and the speaker driver can be directly coupled to an external application processor, eliminating a need for a codec and thus, reducing the size, cost, and/or complexity of a device. In one aspect, the speaker driver and the microphone ASIC are implemented as separate dice mounted on the package substrate. In another aspect, the speaker driver and the microphone ASIC are implemented as stacked die on the package substrate. In yet another aspect, the speaker driver and the microphone ASIC are implemented as a single die on the package substrate.

Abstract: Disclosed herein are systems, methods, and computer-readable storage device for analyzing a first audiofile, to yield a first analysis, wherein the first analysis identifies a first segment of a first plurality of segments within the first audiofile, the first segment being one of a most interesting segment, a most important segment, a most relevant segment, and a most representative segment. A same analysis can be performed on a second audiofile, to yield a second analysis, wherein the second analysis identifies a second segment of a second plurality of segments within the second audiofile, the second segment comprising one of a most interesting segment, a most important segment, a most relevant segment, and a most representative segment. The first segment is presented as a representative segment of the first audiofile and the second segment is presented as a representative of the second audiofile within a three-dimensional audio space.

Abstract: An example implementation may involve a playback device assigning, to the playback device, a zone property that is assigned to a device base (e.g., a charging base) that the playback device is placed upon. For instance, a playback device may identify a particular base onto which the playback device is placed. The playback device may determine that the particular base is assigned a zone property that is associated with a first zone of a media playback system. The playback device may assign, to the playback device, the zone property that is assigned to the particular base. By acquiring the zone property, the playback device may join the first zone of the media playback system. Other example implementations may involve assigning a zone property to a device base, or modify a zone property of a device base, among other examples.

Abstract: According to the embodiment of the present invention, the stereo echo suppressing device that suppresses an acoustic echo signal based on a near-end input signal and far-end signals of two channels as a stereo signal includes: first and second far-end signal amplitude spectrum calculating units for calculating amplitude spectrums of a first far-end signal and a second far-end signal; a near-end input signal amplitude spectrum calculating unit for calculating an amplitude spectrum of the near-end input signal; first and second estimation echo signal estimating units for estimating amplitude spectrums of a first estimation echo signal and a second estimation echo signal; first and second echo suppressing units for suppressing the acoustic echo signal; first and second echo path characteristic updating unit for calculating and updating echo path characteristics of the first far-end signal and the second far-end signal.

Abstract: A reverberation suppression apparatus includes: an instantaneous value calculation unit that calculates an instantaneous value/instantaneous values in an envelope of values correlating with the absolute value or the square of an input signal; a reverberation estimation unit that calculates an exponential moving average of the instantaneous value(s) as an estimated reverberation component; a gain derivation unit that derives a gain corresponding to the input signal according to the estimated reverberation component and the instantaneous value(s) when the each instantaneous value is larger than the estimated reverberation component, and derives a lower limit of the gain as the gain corresponding to the input signal when the each instantaneous value is smaller than the estimated reverberation component; a smoothing unit that performs a smoothing process on the gain; and a gain processing unit that applies the gain to amplitude adjustment of the input signal thereafter.