Abstract: Systems, methods, and computer program products are disclosed for adaptive downmixing of audio signals with improved continuity. An audio encoding system receives an input multi-channel audio signal including a primary input audio channel and L non-primary input audio channels. The system determines a set of L input gains. For each of the channels and gains, the system forms a respective scaled non-primary input audio channel. The system forms a primary output audio channel from the sum of the primary input audio channel and the scaled non-primary input audio channels. The system determines a set of L prediction gains. The system forms a prediction channel from the primary output audio channel. The system forms L non-primary output audio channels. The system forms an output multi-channel audio signal from the primary output audio channel and the L non-primary output audio channels.

Abstract: A computer implemented system for rendering captured audio soundfields to a listener comprises apparatus to deliver the audio soundfields to the listener. The delivery apparatus delivers the audio soundfields to the listener with first and second audio elements perceived by the listener as emanating from first and second virtual source locations, respectively, and with the first audio element and/or the second audio element delivered to the listener from a third virtual source location. The first virtual source location and the second virtual source location are perceived by the listener as being located to the front of the listener, and the third virtual source location is located to the rear or the side of the listener.

Abstract: The positions of a plurality of speakers at a media consumption site are determined. Audio information in an object-based format is received. Gain adjustment value for a sound content portion in the object-based format may be determined based on the position of the sound content portion and the positions of the plurality of speakers. Audio information in a ring-based channel format is received. Gain adjustment value for each ring-based channel in a set of ring-based channels may be determined based on the ring to which the ring-based channel belongs and the positions of the speakers at a media consumption site.

Abstract: Methods, systems, and media for determining sound field rotations are provided. In some embodiments, a method for determining sound field rotations involves determining an activity situation of a user. The method may involve determining a user head orientation using at least one sensor of the one or more sensors. The method may involve determining a direction of interest based on the activity situation and the user head orientation. The method may involve determining a rotation of a sound field used to present audio objects via headphones based on the direction of interest.

Abstract: The present disclosure relates to a method and system for predicting a future orientation of an orientation tracker (100). The method comprising obtaining a sequence of angular velocity samples, each angular velocity sample indicating an angular velocity at a point in time and obtaining a sequence of angular acceleration samples, each angular acceleration sample indicating an acceleration or deceleration of the angular velocity at each point in time. Wherein said method further comprises determining (S5a), for each point in time where the angular velocity is accelerating, a predicted orientation of the orientation tracker (100) based on a first order prediction of an accumulated rotation of the orientation tracker (100) and determining (S5c), for each point in time where the angular velocity is decelerating, a predicted orientation of the orientation tracker (100) based on a second order prediction of the accumulated rotation of the orientation tracker (100).

Abstract: A multi-input, multi-output audio process is implemented as a linear system for use in an audio filterbank to convert a set of frequency-domain input audio signals into a set of frequency-domain output signals. A transfer function from one input to one output is defined as a frequency dependent gain function. In some implementations, the transfer function includes a direct component that is substantially defined as a frequency dependent gain, and one or more decorrelated components that have frequency-varying group phase response. The transfer function is formed from a set of sub-band functions, with each sub-band function being formed from a set of corresponding component transfer functions including direct component and one or more decorrelated components.

Abstract: A filterbank, suitable for modifying audio signals with dynamic gains in each band, is constructed so that the perceived latency is small, while a larger group delay is applied at low frequencies to enable higher frequency resolution in the lower frequency bands. The higher group delay at low frequencies is achieved by inserting an all-pass filter into the reconstructed filter response.

Abstract: The disclosure herein generally relates to capturing, acoustic pre-processing, encoding, decoding, and rendering of directional audio of an audio scene. In particular, it relates to a device adapted to modify a directional property of a captured directional audio in response to spatial data of a microphone system capturing the directional audio. The disclosure further relates to a rendering device configured to modify a directional property of a received directional audio in response to received spatial data.

Abstract: Disclosed are methods and systems which convert a multi-microphone input signal to a multichannel output signal making use of a time-and frequency-varying matrix. For each time and frequency tile, the matrix is derived as a function of a dominant direction of arrival and a steering strength parameter. Likewise, the dominant direction and steering strength parameter are derived from characteristics of the multi-microphone signals, where those characteristics include values representative of the inter-channel amplitude and group-delay differences.

Abstract: Methods, systems, and media for determining user movement direction are provided. In some embodiments, a method involves obtaining, using a control system, user acceleration data associated with a user. The method involves determining, using the control system, a movement period associated with a movement activity of the user using the user acceleration data, wherein the movement period indicates a duration between two sequential movements by the user. The method involves determining, using the control system, a movement direction corresponding to the movement activity using the user acceleration data based on a direction of acceleration orthogonal to the movement direction in which at least a portion of the user acceleration data is anti-periodic over a period of time corresponding to the movement period.

Abstract: A filterbank, suitable for modifying audio signals with dynamic gains in each band, is constructed so that the perceived latency is small, while a larger group delay is applied at low frequencies to enable higher frequency resolution in the lower frequency bands. The higher group delay at low frequencies is achieved by inserting an all-pass filter into the reconstructed filter response.

Abstract: A computer implemented system for rendering captured audio soundfields to a listener comprises apparatus to deliver the audio soundfields to the listener. The delivery apparatus delivers the audio soundfields to the listener with first and second audio elements perceived by the listener as emanating from first and second virtual source locations, respectively, and with the first audio element and/or the second audio element delivered to the listener from a third virtual source location. The first virtual source location and the second virtual source location are perceived by the listener as being located to the front of the listener, and the third virtual source location is located to the rear or the side of the listener.

Abstract: The disclosure herein generally relates to capturing, acoustic pre-processing, encoding, decoding, and rendering of directional audio of an audio scene. In particular, it relates to a device adapted to modify a directional property of a captured directional audio in response to spatial data of a microphone system capturing the directional audio. The disclosure further relates to a rendering device configured to modify a directional property of a received directional audio in response to received spatial data.

Abstract: The positions of a plurality of speakers at a media consumption site are determined. Audio information in an object-based format is received. Gain adjustment value for a sound content portion in the object-based format may be determined based on the position of the sound content portion and the positions of the plurality of speakers. Audio information in a ring-based channel format is received. Gain adjustment value for each ring-based channel in a set of ring-based channels may be determined based on the ring to which the ring-based channel belongs and the positions of the speakers at a media consumption site.

Abstract: The disclosed embodiments enable converting audio signals captured in various formats by various capture devices into a limited number of formats that can be processed by an audio codec (e.g., an Immersive Voice and Audio Services (IVAS) codec). In an embodiment, a simplification unit of the audio device receives an audio signal captured by one or more audio capture devices coupled to the audio device. The simplification unit determines whether the audio signal is in a format that is supported/not supported by an encoding unit of the audio device. Based on the determining, the simplification unit, converts the audio signal into a format that is supported by the encoding unit. In an embodiment, if the simplification unit determines that the audio signal is in a spatial format, the simplification unit can convert the audio signal into a spatial “mezzanine” format supported by the encoding.

Abstract: Disclosed are methods and systems which convert a multi-microphone input signal to a multichannel output signal making use of a time- and frequency-varying matrix. For each time and frequency tile, the matrix is derived as a function of a dominant direction of arrival and a steering strength parameter. Likewise, the dominant direction and steering strength parameter are derived from characteristics of the multi-microphone signals, where those characteristics include values representative of the inter-channel amplitude and group-delay differences.

Abstract: Methods and systems for improving signal processing by smoothing the covariance matrix of a multi-channel signal by setting a forgetting factor based on the bins of a band. A method and system for resetting the smoothing based on transient detection is also disclosed. A method and system for resampling for the smoothing during a banding transition is also disclosed.

Abstract: The disclosed embodiments enable converting audio signals captured in various formats by various capture devices into a limited number of formats that can be processed by an audio codec (e.g., an Immersive Voice and Audio Services (IVAS) codec). In an embodiment, a simplification unit of the audio device receives an audio signal captured by one or more audio capture devices coupled to the audio device. The simplification unit determines whether the audio signal is in a format that is supported/not supported by an encoding unit of the audio device. Based on the determining, the simplification unit, converts the audio signal into a format that is supported by the encoding unit. In an embodiment, if the simplification unit determines that the audio signal is in a spatial format, the simplification unit can convert the audio signal into a spatial “mezzanine” format supported by the encoding.

Abstract: A method for performing calibration of magnetometers is provided. In some embodiments, the method involves obtaining a sequence of gyroscope measurements from one or more gyroscopes and a sequence of magnetometer measurements from one or more magnetometers. In some embodiments, the method involves determining a sequence of angular velocity estimates based on the sequence of gyroscope measurements. In some embodiments, the method involves determining a first estimate of a derivative of an external magnetic field based on the sequence of magnetometer measurements. In some embodiments, the method involves determining a second estimate of the derivative of the external magnetic field based on the sequence of angular velocity estimates. In some embodiments, the method involves identifying magnetometer calibration constants based on a difference between the first estimate of the derivative and the second estimate of the derivative.

Abstract: Methods and systems for improving signal processing by smoothing the covariance matrix of a multi-channel signal by setting a forgetting factor based on the bins of a band. A method and system for resetting the smoothing based on transient detection is also disclosed. A method and system for resampling for the smoothing during a banding transition is also disclosed.