Abstract: An audio rendering device includes: at least one wireless communications interface configured to receive a first plurality of audio packets from a wireless communication device, the first plurality of audio packets including frames of audio data for a first audio channel from a set of one or more audio channels; and a processing unit; wherein the audio rendering device is configured to release, at respective first buffer release times, the frames of the audio data for the first audio channel from a first buffer for rendering by the processing unit; wherein the audio rendering device is configured to receive one or more messages from an other audio rendering device; and wherein the audio rendering device is configured to synchronize the first buffer release times with second buffer release times associated with the other audio rendering device based on the received one or more messages.

Abstract: An audio system presented herein includes a transducer array, sensor array, and a controller. The controller control tactile content imparted to a user via actuation of at least one transducer in the transducer array while presenting audio content to the user. The transducer array presents the audio content with the tactile content to the user. The audio system can be part of a headset.

Abstract: A playback device includes a speaker, a controller, a first switch circuit, and a second switch circuit. The speaker has a first terminal and a second terminal. The controller is configured to output a first audio signal and a second audio signal. The controller is coupled to the first terminal of the speaker, and is configured to transmit the first audio signal to the speaker. The second switch circuit is coupled between the second terminal of the speaker and the controller, and is coupled to the first switch circuit. The second switch circuit is configured to transmit the second audio signal from the controller to the speaker when the first switch circuit is turned on.

Abstract: A multi-path audio amplification system that provides an output drive signal to electromechanical output transducers provides improved undistorted headroom, reduced path switching noise, and/or improved frequency response performance. Multiple signal amplification paths receive an audio input signal and have corresponding multiple output stages that have differing output impedances. A mode selector selects an active one of the multiple signal amplification paths is selected to supply the output drive signal. Outputs of the multiple output stages are coupled to the electromechanical transducer to provide the output drive signal and at least one of the multiple signal amplification paths includes an equalization filter for filtering the audio input signal to compensate for phase or gain differences referenced from the input to the outputs of the multiple output stages due to interaction between the differing output impedances and an impedance of the electromechanical transducer.

Abstract: A method of processing an audio signal is disclosed. According to embodiments of the method, magnitude response information of a prototype filter is determined. The magnitude response information includes a plurality of gain values, at least one of which includes a first gain corresponding to a first frequency. The magnitude response information of the prototype filter is stored. The magnitude response information of the prototype filter at the first frequency is retrieved. Gains are computed for a plurality of control frequencies based on the retrieved magnitude response information of the prototype filter at the first frequency, and the computed gains are applied to the audio signal.

Abstract: An urban air mobility (UAM) noise reduction system and method are provided, where the UAM noise reduction system includes a UAM configured to detect and to provide rotation per minute (RPM) information of a propeller and location coordinate information, and a noise canceling device configured to predict a noise canceling sound wave amplitude on the basis of the RPM information of the propeller and the location coordinate information received through the UAM and to output a noise canceling sound wave corresponding to the predicted noise canceling sound wave amplitude to the UAM.

Abstract: Noise signals are captured from one or more physical error microphones located at first locations within the vehicle. High-frequency noise signals are captured from a feedforward system sensor. A virtual microphone algorithm is utilized to estimate noise signals at a virtual location based on the noise signals, the estimation utilizing a transfer function that estimates a signal that would have been received by the one or more physical error microphones at the virtual location. The virtual microphone algorithm is utilized to estimate noise signals at the virtual location based on the high-frequency noise signal. A noise-cancelling signal is provided to cancel noise at the virtual location, the noise-cancelling signal accounting for the noise captured by both the feedforward system sensor and the one or more physical error microphones, the ANC system utilizing a working frequency for the ANC of at least 2 kHz.

Abstract: Fibers may be intertwined to form structures for electronic devices and other parts. Fibers may be intertwined using computer-controlled braiding, weaving, and knitting equipment. Binder materials may be selectively incorporated into the intertwined fibers. By controlling the properties of the intertwined fibers and the patterns of incorporated binder, structures can be formed that include antenna windows, sound-transparent and sound-blocking structures, structures that have integral rigid and flexible portions, and tubes with seamless forked portions. Fiber-based structures such as these may be used to form cables and other parts of headphones or other electronic device accessories, housings for electronic devices such as housings for portable computers, and other structures.

Abstract: A signal processer is configured to decrease total harmonic distortion plus noise of an output signal generated from an input signal. The signal processer includes a mixer, a pulse-width modulator, a power stage circuit, and a feedback circuit. The mixer is configured to mix the input signal and a feedback signal to generate a mixed signal. The pulse-width modulator is configured to module the mixed signal to generate a modulated signal and output the modulated signal from an output terminal of the pulse-width modulator. The power stage circuit is configured to amplify the modulated signal to generate the output signal and output the output signal from an output terminal of the power stage circuit. The feedback circuit is configured to generate a feedback signal selectively according to the modulated signal or the output signal.

Abstract: A driving sound generation device includes: an acquirer that acquires vehicle information; a storage that stores a plurality of sound source data including driving sound source data and masker sound source data for generating a masker sound; a first adjuster that performs a first adjustment of the plurality of sound source data in accordance with the vehicle information; a second adjuster that performs a second adjustment of at least one of the plurality of sound source data after the first adjustment when the determiner determines that a change amount is smaller than or equal to a predetermined change amount; and a synthesizer that mixes the plurality of sound source data after the second adjustment to generate a driving sound to be output into a cabin of the vehicle; and an outputter that outputs, to the cabin of the vehicle, the driving sound generated.

Abstract: An audio processing apparatus includes a first microphone configured to obtain ambient sound, a second microphone configured to obtain noise from a noise source, a first conversion unit configured to perform a Fourier transform on an audio signal from the first microphone to generate a first audio signal, a second conversion unit configured to perform a Fourier transform on an audio signal from the second microphone to generate a second audio signal, a generation unit configured to generate noise data by using the second audio signal and a parameter related to the noise from the noise source, a subtraction unit configured to subtract the noise data from the first audio signal, and a third conversion unit configured to perform an inverse Fourier transform on an audio signal from the subtraction unit.

Abstract: A method that generates binaural headphone playback signals given multiple audio source signals with an associated metadata and binaural room impulse response (BRIR) database, wherein the audio source signals are channel-based, object-based, or a mixture of both channel-based and object-based signals. The method includes parameterizing BRIR to be used for rendering, dividing each audio source signal to be rendered into a number of blocks and frames, and averaging the parameterized BRIR sequences. The method also includes downmixing the divided audio source signals using the diffuse blocks of BRIRs, and performing late reverberation processing on the downmixed version of the previous blocks of the audio source signals.

Abstract: At least a system or a method is provided for remote delivery or collection of a device such as an audio collection device. For example, a device comprising an aperture collection and retrieval pin is provided. An apparatus is provided having an aperture receiver, an aperture drive gear and a drive motor. The drive motor is configured to drive the aperture drive gear to open or close the aperture receiver of the apparatus for retrieving or releasing the device comprising the aperture collection pin.

Abstract: Example embodiments provide a process that includes one or more of receiving an audio signal at a feedback compressor circuit, receiving an auxiliary attenuation signal from an auxiliary attenuation source, determining a threshold power level based on a value of the auxiliary attenuation signal, determining an output power level of the audio signal exceeds the threshold power level, combining the audio signal with the auxiliary attenuation signal from the auxiliary attenuation source and a compressed attenuation signal from the feedback compressor circuit to create a combination signal, and generating an audio output signal of the feedback compressor circuit based on the combination signal.

Abstract: A recurrent neural network is employed in a loudspeaker system to compensate the distortion of the system based upon a source signal (content) and the sensing output of a sensing circuit (context). A frequency domain transform is selected to provide mapping between the source signal and a recorded signal; and enable reconstruction of desirable playback. Various sensing-related features and source-related features are derived to serve as the auxiliary information. A desirable content is therefore generated based upon the original content and the context.

Abstract: An audio cancellation system includes a voice enabled computing system that is connected to an audio output device using a wired or wireless communication network. The voice enabled computing device can provide media content to a user and receive a voice command from the user. The connection between the voice enabled computing system and the audio output device introduces a time delay between the media content being generated at the voice enabled computing device and the media content being reproduced at the audio output device. The system operates to determine a calibration value adapted for the voice enabled computing system and the audio output device. The system uses the calibration value to filter the user's voice command from a recording of ambient sound including the media content, without requiring significant use of memory and computing resources.

Abstract: Embodiments herein enable fast and easy interconnectivity among multimedia accessories including mobile devices and other devices. There is only limited space on mobile devices yet there are numerous input connectors. The standard TRRS audio jack is one such input that has and remains common, primarily because it is the accepted standard for audio input; namely, headphones and earpieces for listening purposes. Embodiments herein describe an intelligent switch to that audio jack that permits for additional backward and forward compatibility. It transparently allows a user to insert analog or digital audio devices, such as earphones, without the need to manually reconfigure device settings. The device herein automatically converts between input connector types using the same input convention present on their existing mobile devices. Other embodiments are disclosed.

Abstract: In multichannel audio coding, an improved coding efficiency is achieved by the following measure: the noise filling of zero-quantized scale factor bands is performed using noise filling sources other than artificially generated noise or spectral replica. In particular, the coding efficiency in multichannel audio coding may be rendered more efficient by performing the noise filling based on noise generated using spectral lines from a previous frame of, or a different channel of the current frame of, the multichannel audio signal.

Abstract: A system for rotating sound or selective listening to sound provides for the ability of the apparent direction of sound sources in a listening environment to remain in consistent orientations in space despite rotations of the microphones used to capture the sound and despite rotations of the head of the listener, even when wearing headphones. Modules are provided in the system to distinguish the sound sources and their apparent directions, as well as to optionally rotate the sound sources in response to detected rotations of the listener's head and/or detected rotations of the microphones.

Abstract: Methods for calibrating active noise-cancelling headphones, including placing the active noise-cancelling headphones on a measuring device; exciting the active noise-cancelling filter; measuring one or more relevant transmission pathways selected from x(n), m(n), and p(n) for feedforward and/or h(n) for feedback; defining at least one goal function for feedforward or feedback; calculating a complementary function for the defined goal function for at least one branch of the active noise-cancelling filter; calculating an impulse response of the complementary function from the measurements of the relevant transmission pathways; approximating operating parameters for the active noise-cancelling filter using the Prony method; and implementing the approximated operating parameters in the active noise-cancelling filter on the signal processor in order to create an approximated complementary active noise-cancelling filter, thereby calibrating the active noise-cancelling headphones.