Abstract: A cinema system comprises video equipment configured to generate video images from video data for presentation on a screen; a headphone system including left and right ear cups, each including at least one driver configured to drive highs and/or mids based on headphone sound signals and not configured to drive lows; a first DAC configured to convert audio data based on the headphone sound data to the headphone sound signals; and a control system configured to generate the audio data from at least headphone sound data; one or more low-frequency speakers configured to drive lows based on low-frequency speaker signals; a second DAC configured to generate the low-frequency speaker signals from low-frequency speaker data; and a server system configured to assist in providing the video data to the video equipment, the headphone sound data to the one or more headphone systems, and the low-frequency speaker data to the second DAC.

Abstract: A software-based conferencing platform is provided. The platform comprises a plurality of audio sources providing input audio signals, the audio sources including a virtual audio device driver configured to receive far-end input audio signals from a conferencing software module, and a network audio library configured to receive near-end input audio signals from one or more near-end audio devices. The platform further comprises a digital signal processing component configured to receive the input audio signals from the audio sources and generate audio output signals based the received signals, the digital signal processing component comprising an acoustic echo cancellation module configured to apply acoustic echo cancellation techniques to one or more of the near-end input audio signals.

Abstract: A crosstalk cancellation filter set configured for use in delivering binaural signals to human ears is provided. The crosstalk cancellation filter set includes a pressure matching system configured to perform spatial filtering or sound field control and an obstructed field model in communication with the pressure matching system. The crosstalk cancellation filter set is configured to take acoustic advantage of scattering effects and occlusional effects caused by violations to a free-field assumption, thereby delivering improved crosstalk cancellation acoustic displays to a listener without the use of headphones.

Abstract: In some embodiments, a method for processing an audio signal in an audio processing apparatus is disclosed. The method includes receiving an audio signal and a parameter, the parameter indicating a location of an auditory event boundary. An audio portion between consecutive auditory event boundaries constitutes an auditory event. The method further includes applying a modification to the audio signal based in part on an occurrence of the auditory event. The parameter may be generated by monitoring a characteristic of the audio signal and identifying a change in the characteristic.

Abstract: An apparatus including means configured to: obtain at least one audio stream, wherein the at least one audio stream includes one or more transport audio signals, wherein the one or more transport audio signals is a defined type of transport audio signal; and convert the one or more transport audio signals to at least one or more further transport audio signals, the one or more further transport audio signals being a further defined type of transport audio signal.

Abstract: Various techniques for driving phased array systems are described, specifically intended for acoustic phased arrays with applications to mid-air haptics, parametric audio, acoustic levitation and acoustic imaging, including a system: 1) that is capable of mitigating the effect of the changes in the air to provide a consistent haptic experience; 2) that produces trap points in air; 3) that defines phased-array optimization in terms of vectors for the production of more consistent haptic effects; 4) that defines one or more control points or regions in space via a controlled acoustic field; 5) that uses a reduced representation method for the construction of acoustic basis functions; 6) that performs efficient evaluation of complex-valued functions for a large quantity of throughput; 7) that generates a Krylov sub-space of a matrix; and 8) that maximizes an objective described by different control points and/or regions to those used to create the acoustic basis functions.

Abstract: A control device includes: a difference calculator that calculates a maximum gain difference between frequency characteristics of an audio signal and desired frequency characteristics in a predetermined frequency band; a center frequency calculator that calculates a center frequency between a first frequency at which the gain difference is 1/?2 times the maximum gain difference and which is closest to a peak frequency corresponding to the maximum gain difference among one or more frequencies lower than the peak frequency and a second frequency at which the gain difference is 1/?2 times the maximum gain difference and which is closest to the peak frequency among one or more frequencies higher than the peak frequency; a determiner that determines parameters of an equalizer based on the maximum gain difference and the center frequency; and an outputter that outputs the parameters to the equalizer.

Abstract: A full-range speaker is driven by an original sound signal output from a sound source. A vibration detection unit detects vibration of the full-range speaker and outputs a playback signal. A subtractor outputs an error signal indicating the difference between the original sound signal and the play back signal. The error signal is amplified by the amplifier to drive the woofer which increases the sound pressure level in the low frequency range by outputting a sound in the same phase as the sound of the full range speaker if the sound pressure level of the full range speaker is insufficient, or in the opposite phase as the sound of the full range speaker if the sound pressure level is excessive. If the sound pressure level is excessive, the sound pressure level is reduced by outputting a sound in phase with the sound of the full range speaker.

Abstract: A method for audio rendering by an apparatus comprising at least one audio rendering device, the method comprising: extracting (S10) a plurality of frequency band components from an input audio signal; determining (S15) from the device frequency response and the plurality of extracted frequency band components at least one indicator representative of masking frequencies energy, masking frequencies corresponding to frequency bands that are above a frequency threshold; determining at least one correction factor (S16) from the indicator representative of masking frequencies energy; e) for each frequency band, determining a second acoustic level threshold (S20) by modifying (S17) with the correction factor a predetermined first acoustic level threshold associated with said frequency band, and determining a reduction gain from a comparison (S30) between an acoustic level of the extracted frequency band and the second acoustic level threshold associated to said extracted frequency band, and applying (S40) the reduc

Abstract: An audio engine is provided for acoustically rendering a three-dimensional virtual environment. The audio engine uses geometric volumes to represent sound sources and any sound occluders. A volumetric response is generated based on sound projected from a volumetric sound source to a listener, taking into consideration any volumetric occluders in-between. The audio engine also provides for modification of a level of detail of sound over time based on distance between a listener and a sound source. Other aspects are also described and claimed.

Abstract: A single-ended to differential converter includes a converter input, a first converter output, a second converter output, and an internal node, wherein the first converter output and the second converter output comprise a differential output; a non-inverting amplifier having an input coupled to the converter input, and an output coupled to the first converter output; an inverting amplifier having an input coupled to the first converter output, and an output coupled to the second converter output; a charge pump having a charge pump output capacitor coupled between the second converter output and the internal node; and a feedback capacitor coupled between the first converter output and the internal node.

Abstract: A method or apparatus switches between binaural sound and stereo sound in response to head movements of a listener. An electronic device provides the listener with binaural sound at a sound localization point (SLP) in a field-of-view of the listener. Binaural sound at the SLP switches to one of mono or stereo sound when head movements of the listener cause the SLP to move outside the field-of-view.

Abstract: Disclosed are a TWS earphone, a method and apparatus for reducing earphone energy consumption, and a computer-readable storage medium. The method for reducing earphone energy consumption comprises: determining whether a target earphone is in an off-ear state and is not placed in a charging case; if so, acquiring a timekeeping duration since the target earphone is taken off of an ear; determining whether the timekeeping duration exceeds a first preset threshold value; and if the timekeeping duration exceeds the first preset threshold value, performing energy consumption reduction processing on the earphone.

Abstract: A method and corresponding device for processing virtual-reality environment data, wherein at least one part of the body of a user of the virtual-reality environment is detected. The part of the body of the user is capable of interacting with at least one object of the virtual-reality environment. At least one sound signal is generated according to the detection of the at least one part of the body of the user of the virtual-reality environment.

Abstract: Sound source enhancement technology is provided that is capable of improving sound source enhancement capabilities in accordance with settings of usage and applications. A PSD optimization device includes a PSD updating unit that takes a target sound PSD input value {circumflex over (?)}?S(?, ?), an interference noise PSD input value {circumflex over (?)}?IN(?, ?), and a background noise PSD input value {circumflex over (?)}?BN(?, ?) as input, and generates a target sound PSD output value ?S(?, ?), an interference noise PSD output value {circumflex over (?)}?IN(?, ?), and a background noise PSD output value {circumflex over (?)}?BN(?, ?), by solving an optimization problem for a cost function relating to a variable uS representing a target sound PSD, a variable uIN representing an interference noise PSD, and a variable uBN representing a background noise PSD.

Abstract: A method of equalising an audio signal derived from a microphone, the method comprising: receiving the audio signal; applying an order-statistic filter to the audio signal in the frequency domain to generate a statistically filtered audio signal; equalising the received audio signal based on the statistically filtered audio signal to generate an equalised audio signal.

Abstract: An electronic device and a method for controlling same are provided. The present electronic device comprises: a communication unit; and a processor configured to receive multiple audio signals via the communication unit, the multiple audio signals being acquired by multiple external electronic devices which have microphones, respectively, and which are positioned at different places, via microphones thereof, the processor being configured to determine at least one audio signal including a user voice uttered by a user among the multiple audio signals and to perform voice recognition regarding an audio signal acquired from the determined audio signals on the basis of the intensity of the determined audio signals.

Abstract: A method for processing an input audio signal includes using a first analytical filter bank to divide the input audio signal in a first frequency splitting process into a plurality of first frequency bands. The first frequency bands of a first subgroup are divided in a further frequency splitting process by a further analytical filter bank into a plurality of frequency subbands. The divided input audio signal is frequency-selectively processed or amplified. The divided and processed input audio signal is then combined again into an output audio signal. A prediction is applied to the first frequency bands of the first subgroup and/or the frequency subbands derived therefrom, to compensate for latency differences between the first frequency bands and the frequency subbands as a result of the or each further frequency splitting process. A device or hearing aid for carrying out the method is also provided.

Abstract: This application relates to a method of extracting an inter channel feature from a multi-channel multi-sound source mixed audio signal performed at a computing device.

Abstract: An example system is configured to: (i) receive data indicating a command to play back audio content, where the data indicating the command to play back audio content is received via a voice input that is detected by a first zone player, (ii) based on the data indicating the command, determine a playback condition comprising a particular zone player, (iii) identify a user account associated with the first zone player, (iv) based at least on (a) the command and (b) the identified user account, determine audio content for playback, and (v) play back the determined audio content via the particular zone player.