Abstract: On the basis of a bitstream (P), an n-channel audio signal (X) is reconstructed by deriving an m-channel core signal (Y) and multichannel coding parameters (?) from the bitstream, where 1?m<n. Also derived from the bitstream are pre-processing dynamic range control, DRC, parameters (DRC2) quantifying an encoder-side dynamic range limiting of the core signal. The n-channel audio signal is obtained by parametric synthesis in accordance with the multichannel coding parameters and while cancelling any encoder-side dynamic range limiting based on the pre-processing DRC parameters. In particular embodiments, the reconstruction further includes use of compensated post-processing DRC parameters quantifying a potential decoder-side dynamic range compression. Cancellation of an encoder-side range limitation and range compression are preferably performed by different decoder-side components. Cancellation and compression may be coordinated by a DRC pre-processor.

Abstract: A method and apparatus may be used to perform personalized audio virtualization. The apparatus may include a speaker, a headphone (over-the-ear, on-ear, or in-ear), a microphone, a computer, a mobile device, a home theater receiver, a television, a Blu-ray (BD) player, a compact disc (CD) player, a digital media player, or the like. The apparatus may be configured to receive an audio signal, scale the audio signal, and perform a convolution and reverberation on the scaled audio signal to produce a convolved audio signal. The apparatus may be configured to filter the convolved audio signal and process the filtered audio signal for output.

Abstract: Audio content coded for a reference speaker configuration is downmixed to downmix audio content coded for a specific speaker configuration. One or more gain adjustments are performed on individual portions of the downmix audio content coded for the specific speaker configuration. Loudness measurements are then performed on the individual portions of the downmix audio content. An audio signal that comprises the audio content coded for the reference speaker configuration and downmix loudness metadata is generated. The downmix loudness metadata is created based at least in part on the loudness measurements on the individual portions of the downmix audio content.

Abstract: An audio system for a vehicle has at least one source of audio signals. A respective directional loudspeaker array is mounted at each seat position and coupled to the at least one source. The at least one source includes a microphone that detects speech from an occupant of the first seat position. Processing circuitry receives signals from the microphone that correspond to the detect speech and drives each second respective loudspeaker array at the other seat positions to radiate acoustic energy corresponding to the detected speech.

Abstract: A head mounted display (HMD) includes a memory that stores head-related transfer functions (HRTFs), a digital signal processor (DSP) that processes sound into binaural sound with a pair of the HRTFs, and head tracking that tracks head movements of the user. Speakers play the binaural sound to the user while the user wears the HMD, and the HMD corrects an error where the user hears the binaural sound.

Abstract: Systems and methods disclosed herein include, determining that a requirement exists to output a system response. In response to determining that a requirement exists to output a system response, (i) setting a system response volume for a first speaker associated with the first networked device and (ii) outputting a system response at the set system response volume via the first speaker associated with the first networked device.

Abstract: The present application provides method and apparatus for a binaural hearing assistance system using a monaural audio signal input. The system, in various examples, provides adjustable delay/phase adjustment and sound level adjustment. Different embodiments are provided for receiving the monaural signal and distributing it to a plurality of hearing assistance devices. Different relaying modes are provided. Special functions are supported, such as telecoil functions. The system also has examples that account for a head-related transfer function in providing advanced sound processing for the wearer. Other examples are provided that are described in the detailed description.

Abstract: Provided are systems and methods for utilizing digital microphones in low power keyword detection and noise suppression. An example method includes receiving a first acoustic signal representing at least one sound captured by a digital microphone. The first acoustic signal includes buffered data transmitted with a first clock frequency. The digital microphone may provide voice activity detection. The example method also includes receiving at least one second acoustic signal representing the at least one sound captured by a second microphone, the at least one second acoustic signal including real-time data. The first and second acoustic signals are provided to an audio processing system which may include noise suppression and keyword detection. The buffered portion may be sent with a higher, second clock frequency to eliminate a delay of the first acoustic signal from the second acoustic signal. Providing the signals may also include delaying the second acoustic signal.

Abstract: A virtual reality (VR) system simulates sounds that a user of the VR system perceives to have originated from sources at desired virtual locations of the VR system. The simulated sounds are generated based on personalized head-related transfer functions (HRTF) of the user that are constructed by applying machine-learned models to a set of anatomical features identified for the user. The set of anatomical features may be identified from images of the user captured by a camera. In one instance, the HRTF is represented as a reduced set of parameters that allow the machine-learned models to capture the variability in HRTF across individual users while being trained in a computationally-efficient manner.

Abstract: A loudspeaker includes a horn including a first end panel, a second end panel, a first side panel, and a second side panel. Edges of at least the first and second side panels define a diffraction slot opening. The first and second side panels are each fabricated from a sheet of flexible material held in a stressed, curved shape by at least a rigid support member. The panels are designed by an automated process based on a number of electro-acoustic transducers to be used in a loudspeaker, horizontal and vertical coverage angles for the loudspeaker, and a wall length for a horn of the loudspeaker.

Abstract: A Closed Loop Headphone Apparatus with 1, 2, 3, 4, 5, 6, 7 or more built-in speakers and 2 Ear Openings, which go around the head of the user and imparting real surround sound emitted from each ear opening.

Abstract: A noise signal is estimated based on a captured audio signal captured from a sound capture unit. It is determined whether the estimated noise signal thus estimated is in a noiseless state. If it is determined that the estimated noise signal is in the noiseless state, the captured audio signal is analyzed as a target sound signal, and a characteristic obtained by the analysis is learned and modeled, thereby generating a target sound model.

Abstract: An audio control arrangement for a motor vehicle includes a camera capturing images of a passenger of the motor vehicle. A processor is communicatively coupled to the camera and determines, based on the captured images, whether the passenger is at least partially asleep. An audio system is communicatively coupled to the processor and reduces a volume of an audio signal in response to the processor determining that the passenger is at least partially asleep. A loudspeaker is communicatively coupled to the audio system and emits sounds based on the audio signal.

Abstract: When a channel signal, such as a 22.2 channel signal, is rendered into a 5.1 channel signal, a three-dimensional (3D) audio may be reproduced by using a two-dimensional (2D) output channel, however, when an elevation angle of an input channel is different from a standard elevation angle, if elevation rendering parameters according to the standard elevation angle are used, distortion may occur in a sound image.

Abstract: Methods, apparatus, systems and articles of manufacture to implement selective suppression of audio emitted from an audio source are disclosed. Example apparatus disclosed herein include a speaker, an audio output driver in communication with the speaker, memory including computer readable instructions, and a processor. In some disclosed example apparatus, the processor is to execute the instructions to perform operations including obtaining reference audio data corresponding to an audio signal to be output subsequently by the audio source and including identification data identifying the audio source, and in response to an input selection corresponding to the identification data, generating a suppression signal to provide to the audio output driver, the suppression signal based on the reference audio data and a time delay between a first time at which the reference audio data was received and a second time at which the audio signal is to be output by the audio source.

Abstract: A control apparatus comprises a first obtaining unit configured to obtain a result of sound collection performed by a sound collection apparatus that collects a measurement signal as a sound, the measurement signal being output from an output apparatus; a detection unit configured to detect a position of the output apparatus or the sound collection apparatus relative to a measurement-subject person; and a second obtaining unit configured to obtain a head-related transfer function of the measurement-subject person based on the result of sound collection obtained by the first obtaining unit and the position detected by the detection unit.

Abstract: A system is provided to analyze cross-modulation distortion in audio devices, which may include testing with audio frequencies. One or more distortion signals from the audio device may be measured for an amplitude, phase, and or frequency modulation effect. In another embodiment a musical signal may be used as a test signal. Providing additional test signals to the audio device can induce a time varying cross-modulation distortion signal from an output of the audio device. Also utilizing at least one additional filter, filter bank, demodulator and or frequency converter and or frequency multiplier provides extra examination of distortion.

Abstract: Methods of operating an audio device are provided. A method includes measuring sound pressure levels (SPLECM) for acoustic energy received by an ear canal microphone (ECM) during a time increment ?t; and calculating a SPL_Dose?t during the time increment ?t using SPLECM.

Abstract: An input signal includes a channel-based audio signal and an object-based audio signal, and an audio encoding device includes an audio scene analysis unit configured to determine an audio scene from the input signal and detect audio scene information; a channel-based encoder that encodes the channel-based audio signal output from the audio scene analysis unit; an object-based encoder that encodes the object-based audio signal output from the audio scene analysis unit; and an audio scene encoding unit configured to encode the audio scene information.

Abstract: Media device systems and methods synchronize video content with audio content presented by a plurality of wireless audio headsets. In an exemplary embodiment, a first time delay corresponds to a first duration of time between communication of the audio content from the media device and presentation of the audio content by a first wireless audio headset. A second time delay corresponds to a second duration of time between communication of the audio content from the media device and presentation of the audio content by a second wireless audio headset, wherein the first time delay is greater than the second time delay. Video content communicated to a display is delayed by the first time delay. Audio content communicated to the second wireless audio headset is delayed by a time delay difference between the first time delay and the second time delay.