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: An acoustic signal is separated based on a difference in the distance from a sound source to a microphone. By using a filter obtained by associating a value corresponding to an estimated value of a short-distance acoustic signal which is obtained by using “a predetermined function” from a second acoustic signal derived from signals collected by “a plurality of microphones” and is emitted from a position close to “the plurality of microphones” with a value corresponding to an estimated value of a long-distance acoustic signal which is emitted from a position far from “the plurality of microphones”, a desired acoustic signal representing at least one of a sound emitted from a position close to “a specific microphone” and a sound emitted from a position far from “the specific microphone” is acquired from a first acoustic signal derived from a signal collected by “the specific microphone”.

Abstract: A frequency compressor of a vibration output apparatus generates a compressed signal by converting frequency components of a mid-band signal into low-band frequency components of a low-band signal by increasing the total number of samples by a factor of n and thus compressing the frequency of amplitude information included in mid-band frequency components of the mid-band signal to 1/n. If the level of a low-band envelope signal is lower than a predetermined threshold level, a vibration signal generator generates a vibration signal by combining the compressed signal with the low-band signal consisting of the low-band frequency components of an acoustic signal. If the level of the low-band envelope signal is higher than the predetermined threshold level, the vibration signal generator generates a vibration signal by directly using the low-band signal. A vibration output unit outputs a vibration on the basis of the generated vibration signal.

Abstract: Disclosed is a display device that includes a display panel, sound generating actuators including a first sound generating actuator in a first area of the display panel and a second sound generating actuator in a second area of the display panel, the first and second sound generating actuators configured to vibrate the display panel to generate sound, and a partition between the first sound generating actuator and the second sound generating actuator.

Abstract: A noise sensor is disposed adjacent a speaker within an ear cup of a hearing protection device. The speaker is disposed within a speaker housing and the noise sensor is disposed within a sensor housing, the sensor housing coupled to the speaker housing such that the noise sensor and speaker remain adjacent one another. The noise sensor includes at least a microphone operably coupled to a printed circuit board. The sensor housing defines an axial bore such that the noise sensor can receive acoustic signals via the axial bore. The sensor housing can be coupled to the speaker housing such that the noise sensor is sealed therebetween and receives acoustic signals via a distal end of the axial bore opposite the speaker. A calibration tool can be disposed to the axial bore via the distal end for airtight calibration of the noise sensor.

Abstract: Processes and devices for equalizing an audio system that is adapted to use a loudspeaker to transduce test audio signals into test sounds. The processes and devices can involve the use of infrared signals to convey information in one or both directions between the audio system and a portable computer device that captures test sounds, calculates audio parameters that can be used in the equalization process, and transmits these audio parameters back to the audio system for its use in equalizing audio signals that are played by the audio system.

Abstract: Systems and methods of audio processing and control for improved audibility and performance in a parametric loudspeaker system. The parametric loudspeaker system includes a parametric loudspeaker providing a capacitive load, an output stage having a plurality of switches interconnected in a bridge configuration and connected to the capacitive load of the parametric loudspeaker, and a controller operative to generate a series of pulse width modulation (PWM) pulses, and to generate a plurality of control signals synchronized with the series of PWM pulses for switchingly controlling the plurality of switches in the bridge configuration, thereby driving the capacitive load of the parametric loudspeaker.

Abstract: A method and apparatus for decompressing a Higher Order Ambisonics (HOA) signal representation is disclosed. The apparatus includes an input interface that receives an encoded directional signal and an encoded ambient signal and an audio decoder that perceptually decodes the encoded directional signal and encoded ambient signal to produce a decoded directional signal and a decoded ambient signal, respectively. The apparatus further includes an extractor for obtaining side information related to the directional signal and an inverse transformer for converting the decoded ambient signal from a spatial domain to an HOA domain representation of the ambient signal. The apparatus also includes a synthesizer for recomposing a Higher Order Ambisonics (HOA) signal from the HOA domain representation of the ambient signal and the decoded directional signal. The side information includes a direction of the directional signal selected from a set of uniformly spaced directions.

Abstract: A device for audio reproduction comprising a rest element for the head of a user, made of a deformable material; a right lateral support element and a left lateral support element located respectively at the two respective right and left of the rest element, the rest and left lateral support elements being connected to each other by the rest element, the right and left support elements defining between each other a listening space and a longitudinal direction parallel to the lateral support elements; a right loudspeaker and a left loudspeaker mounted respectively to the right and left lateral support elements have respective sound emission faces that face towards a listening space defined between the right and left support elements. The rest element provides a housing for a portion of the head configured to arrange the user with right and left ears oriented towards the right and left loudspeakers, the right and left ears creating a listening axis.

Abstract: Embodiments described herein provide for detecting presence of an object in proximity to a playback device and responsively performing one or more operations. In an example implementation, a playback device detects, via a proximity detector of the playback device, presence of an object in proximity to the playback device. In response to detecting the presence of the object, the playback device performs one or more operations.

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: In one example, a zone player may include a network interface, at least one processor, a non-transitory computer-readable medium, and program instructions stored on the non-transitory computer-readable medium that may be executable by the at least one processor such that the zone player is configured to receive an instruction to change a volume setting of the zone player to a requested volume level; in response to receiving the instruction to change the volume setting of the zone player to the requested volume level, adjust the volume setting of the zone player to an adapted volume level that is lower than the requested volume level; and after adjusting the volume setting of the zone player to the adapted volume level, send an indication that the volume setting of the zone player has been adjusted to the requested volume level rather than the adapted volume level.

Abstract: Techniques for improving acoustic echo cancellation are described. Energy levels of audio data received from a microphone and representing near-end audio and reference audio data representing far-end audio are determined. If near-end audio is detected but far-end audio is not detected, a controller turns of or bypasses an acoustic echo cancellation system until far-end audio is again detected, thereby decreasing or eliminating distortion of the near-end audio by the acoustic echo cancellation system.

Abstract: Various implementations include systems for processing audio signals. In particular implementations, a system for processing audio signals includes: a wearable audio device having a transducer and a communication system; and an accessory device configured to wirelessly communicate with the wearable audio device, the accessory device having a processor configured to process a source audio signal according to a method that includes: source separating the source audio signal; and providing a source separated audio signal to the wearable audio device for transduction.

Abstract: An audio filter and corresponding method with through-zero linearly variable resonant frequency are described. An example method includes receiving an input signal; receiving one or more control input signals; and providing a resonant filter for electronic music for processing the input signal. The resonant filter can include an exponential times linear multiplication to produce the resonant frequency control responsive to control voltages exponentially for musical octave/semitone control, and linearly for timbre modulation, the waveform being invariant over corresponding exponential changes in pitch of both input signal and filter resonance. The linear modulation can include inverting the sign of a bandpass feedback based on the sign of the resonant frequency variable so as to maintain stability. In the example method, to prevent a glitch on the output when frequency changes sign, a separate highpass output signal may be generated.

Abstract: Provided is an information processor including a signal processing section that acquires a first signal to be detected by an acoustic input section of a first unit including the acoustic input section disposed within a predetermined distance from one ear hole of a user in a state of being worn by the user, acquires a second signal, which indicates a noise generated from a noise source, to be acquired by a second unit, and generates a noise cancellation signal directed to the noise on the basis of the first signal and the second signal.

Abstract: A novel phantom-powered inline preamplifier is configured to provide selective processing of a sound source signal by intelligently determining the need for a low or high impedance matching and transformer coupling, based on high fidelity requirements of a particular sound source signal. For example, the phantom-powered inline preamplifier can intelligently detect a microphone-originating sound source signal and automatically route the microphone-originating sound source signal to a low impedance matching circuit pathway that leads to a phantom-powered output terminal for an optimal hi-fidelity processing specific to the microphone-originating sound source signal.

Abstract: Examples described herein involve validating motion of a microphone during calibration of a playback device. An example implementation involves a mobile device detecting, via one or more microphones, audio signals emitted from one or more playback devices as part of a calibration process. After the one or more playback devices emit the audio signals, the mobile device determines whether the detected audio signals indicate that sufficient horizontal translation of the mobile device occurred during the calibration process. When the detected audio signals indicate that insufficient horizontal translation occurred, the mobile device displays a prompt to move the mobile device more while the one or more playback devices emit one or more additional audio signals as part of the calibration process. When the detected audio signals indicate that sufficient horizontal translation occurred, the mobile device calibrates the one or more playback devices with a calibration based on the detected audio signals.

Abstract: A signal processing system includes microphone units connected in series and a host device connected to one of the microphone units. Each of the microphone units has a microphone, a temporary storage memory, and a processing section for processing the sound picked up by the microphone. The host device has a non-volatile memory in which a sound signal processing program for the microphone units is stored. The host device transmits the sound signal processing program read from the non-volatile memory to each of the microphone units. Each of the microphone units temporarily stores the sound signal processing program in the temporary storage memory. The processing section performs a process corresponding to the sound signal processing program temporarily stored in the temporary storage memory and transmits the processed sound to the host device.

Abstract: A method for compensating for acoustic influence of a listening room on an acoustic output from an audio system including at least a left and a right loudspeaker, the method comprising determining a left frequency response and a right frequency response, designing a left compensation filter FL, and a right compensation filter FR, and during playback applying the left and right filters to left and right channel inputs. According to the invention, a target response in the listening position is simulated, and the left and right compensation filters are designed to filter transfer functions based on the simulated target function multiplied by an inverse of the left/right frequency responses. By relying on a simulated target instead of relying on an empirical approach, the general impact of a room can be more accurately captured by the target functions.