Abstract: A system is disclosed for using an audio time and level difference renderer (TLDR) to generate spatialized audio content for multiple channels from an audio signal received at a single channel. The system selects an audio TLDR from a set of audio TLDRs based on received input parameters. The system configures the selected audio TLDR based on received input parameters using a filter parameter model to generate a configured audio TLDR that comprises a set of configured binaural dynamic filters, and a configured delay between the multiple channel. The system applies the configured audio TLDR to an audio signal received at the single channel to generate spatialized multiple channel audio content for each channel of the multiple audio channel and presents the generated spatialized audio content at multiple channels to a user via a headset.

Abstract: A communication device for hearing protection devices provides communication capabilities to users adorning hearing protection devices, which would ordinarily hinder or block communication. The communication device removably attaches to a cup of a hearing protection device and connects to the speakers thereof. One or more wireless transceivers provide wireless transmission of vocal communication and other sound between users of hearing protection devices enhanced with the communication device. Users remain protected by their hearing protection devices and while communicating freely in high decibel environments.

Abstract: This invention corresponds to a device for binaural capture of sound comprising an upper module, a first external ear and a second external ear incorporated into the upper module in a removable way; two transducers, a transducer is arranged in the first external ear and the other transducer is arranged in the second external ear; and, a clamping-coupling located in the lower part of the upper module.

Abstract: A headphone for spatial audio rendering includes a first database having an impulse response pair corresponding to a reference speaker location. A head sensor provides head orientation information to a second database having rotation filters, the filters corresponding to different azimuth and elevation positions relative to the reference speaker location. A digital signal processor combines the rotation filters with the impulse response pair to generate an output binaural audio signal to transducers of the headphone. Efficiencies in creating impulse response or HRTF databases are achieved by sampling the impulse response less frequently than in conventional methods. This sampling at coarser intervals reduces the number of data measurements required to generate a spherical grid and reduces the time involved in capturing the impulse responses. Impulse responses for data points falling between the sampled data points are generated by interpolating in the frequency domain.

Abstract: This patent includes a method to generate a six dimensional audio dataset. When listening to the six dimensional audio dataset with an advanced headset disclosed in this patent, a user will enable a user to have precision sound localization and maximize the listening experience. For example, when the audio file is registered to a living room, a user would be able to sit in the “middle of the band” and if desired, reposition the instruments to different locations within the living room to change the audio experience.

Abstract: The present application relates to a system and method for receiving, via a microphone array, an incoming acoustic signal including a desired signal and an undesired signal, performing a first minimum variance distortionless response beamforming operation on the incoming acoustic signal to isolate the desired signal, performing a second minimum variance distortionless response beamforming operation, by the signal processor, on the incoming acoust signal to isolate the undesired signal, filtering the desired signal to generate a filtered desired signal, combining an inverse of the filtered desired signal with the undesired signal to generate a combined undesired signal, filtering the combined undesired signal to generate a filtered combined undesired signal, combining an inverse of the combined undesired signal and the desired signal to generate an output signal, and generating a data signal in response to the output signal for transmission via a wireless network.

Abstract: The present disclosure relates generally to a speech aid device comprising a mount configured to couple to a head of a user, at least one sonic signal transmitter coupled to the mount, and a power supply electrically coupled to the at least one sonic signal transmitter. The at least one sonic signal transmitter is configured to transmit a sonic signal into at least one of a nasal cavity or a mouth cavity of the user when the mount is coupled to the head of the user.

Abstract: A method for processing a sound signal including synchronously acquiring an input sound signal Sinput by means of at least two omnidirectional microphones, encoding the input sound signal Sentréeinput in a sound data D format of the ambisonics type of order R, R being a natural number greater than or equal to one, the encoding step including a directivity optimisation sub-step carried out by means of filters of the Finite Impulse Response filter type. Each of the signals acquired by the microphones is filtered during the directivity optimisation sub-step by a FIR filter, then subtracted from an unfiltered version of each of the other signals in order to obtain N enhanced signals. The present invention also relates to a system for processing the sound signal.

Abstract: In an audio encoder, for audio content received in a source audio format, default gains are generated based on a default dynamic range compression (DRC) curve, and non-default gains are generated for a non-default gain profile. Based on the default gains and non-default gains, differential gains are generated. An audio signal comprising the audio content, the default DRC curve, and differential gains is generated. In an audio decoder, the default DRC curve and the differential gains are identified from the audio signal. Default gains are re-generated based on the default DRC curve. Based on the combination of the re-generated default gains and the differential gains, operations are performed on the audio content extracted from the audio signal.

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. Also frequency and or phase response can be measured with the presence of a de-sensing signal and or another signal that induce near slew rate limiting or near overload condition of the device under test.

Abstract: A system comprises an ear-worn electronic device configured to be worn by a wearer. The ear-worn electronic device comprises a processor and memory coupled to the processor. The memory is configured to store an annoying sound dictionary representative of a plurality of annoying sounds pre-identified by the wearer. A microphone is coupled to the processor and configured to monitor an acoustic environment of the wearer. A speaker or a receiver is coupled to the processor. The processor is configured to identify different background noises present in the acoustic environment, determine which of the background noises correspond to one or more of the plurality of annoying sounds, and attenuate the one or more annoying sounds in an output signal provided to the speaker or receiver.

Abstract: Compensation accuracy is increased with respect to clip compensation in a case where signals from a plurality of microphones are subjected to an echo cancellation process. A signal processing device according to an embodiment of the present technology includes an echo cancellation unit that performs an echo cancellation process of canceling an output signal component from a speaker on signals from a plurality of microphones, a clip detection unit that performs a clip detection for signals from the plurality of microphones, and a clip compensation unit that compensates for a signal after the echo cancellation process of clipped one of the microphones on the basis of a signal of non-clipped one of the microphones.

Abstract: A binaural room impulse response (BRIR) can be generated based on a position of a listener's head, and a plurality of head related impulse responses (HRIRs). Each of the plurality of HRIRs are selected for a respective one of a plurality of acoustic reflections which, when taken together, approximate reverberation of a room. Each of the acoustic reflections have a direction and a delay. The BRIR filter is applied to source audio to generate binaural audio output.

Abstract: In one embodiment, a system for providing three-dimensional (3D) immersive sound is provided. The system includes a loudspeaker and at least one controller. The loudspeaker transmits an audio output signal in a listening environment. The at least one controller is programmed to store a plurality of directional bands with each directional band being defined by a narrowband frequency interval and to store at least psychoacoustic scale including a sub-band for each directional band. The at least one controller is further programmed to determine an energy for the sub-band and generate a loudspeaker driving signal based at least on the energy for the sub-band to drive the loudspeaker to transmit the audio output signal.

Abstract: Methods and systems are described for a media guidance application that tailors media content recommendations for downloadable content to the current interests of the users. For example, the media guidance application may determine the current interests of a user based on the media content recently consumed by the friends of the user, and automatically download that media content. In such cases, the media guidance application may monitor social networks of the user for indications of media content that may have been recently consumed by the friends of a user as such media content is likely to be of current interest to the user. In response to detecting media content that was recently consumed by the friend of the user, the media guidance application may automatically download the media content to a device of the user.

Abstract: A sound system configured to output sound signals to a user located in a defined environment, comprising a plurality of transducers configured to output the sound signals, a processing unit configured to individually control at least an electrical delay, a frequency characteristic and a sound level with which the sound signals transmitted to the plurality of transducers are output by each of the plurality of transducers, and a seat for a user to which the sound signals are output, the seat comprising a seating surface and a backrest configured support a back of the user sitting on the seating surface. The backrest comprises an upper surface defining an upper end surface of the backrest. The defined environment influences the sound signals as perceived by the user sitting on the seat.

Abstract: An apparatus for providing a contrast mode and a front optimized mode for audio in a vehicle is provided. An audio controller is programmed to transmit first audio content in a first zone seating area and to transmit second audio content in a second zone seating area. The audio controller receives a first indication to transmit the first audio content in the first zone seating area and the second audio content in the second zone seating area in the contrast mode to provide an equal listening experience. The audio controller receives a second indication to transmit the first audio content in the first zone seating area and the second audio content in the second zone seating area in the front optimized mode to increase a quality of sound in the first zone seating area and to decrease a quality of sound in the second zone seating area.

Abstract: Functionalities of new data types are enabled for data stored using native data types (“recognized data types”) supported by DBMSs. Values having a latent data type may be stored using a recognized data type of a DBMS. That is, a value stored in a database table may be recognized by a DBMS as having a recognized data type (e.g., character string) but the value may also include information that indicates the value conforms to a particular latent data type. DBMSs compile database statements. The database statement can contain multiple expressions that may specify a latent data type or that may have a recognized data type but that also imply latent data types. By determining that various expressions have latent data types, operations that rely on the expressions as having a latent data type may be validated in a query and executed accordingly.

Abstract: A method for conducting a cross frequency simultaneous masking (xF SM) test begins with generating a signal probe and a masker probe. The center frequencies of the signal and masker probes are separated by a fixed frequency ratio. An xF SM curve is generated by sweeping the signal and masker probes across a given frequency range, while maintaining the fixed frequency ratio between the two. While sweeping, the masker probe is maintained at a pre-determined masker amplitude or a series of pre-determined masker amplitudes. The amplitude of the signal probe is adjusted in response to a series of user inputs, which are then interpolated to generate the xF SM curve. Additionally, while sweeping, the signal probe can be maintained at one or more pre-determine amplitudes and the amplitude of the masker probe adjusted in response to user inputs, which are then interpolated to generate the xF SM curve.

Abstract: Methods, systems, computer-readable media, and apparatuses for audio signal processing are presented. Some configurations include determining that first audio activity in at least one microphone signal is voice activity; determining whether the voice activity is voice activity of a participant in an application session active on a device; based at least on a result of the determining whether the voice activity is voice activity of a participant in the application session, generating an antinoise signal to cancel the first audio activity; and by a loudspeaker, producing an acoustic signal that is based on the antinoise signal. Applications relating to shared virtual spaces are described.