The present invention relates to a method for generating a filter for an audio signal and a parameterization device for the same, and more particularly, to a method for generating a filter for an audio signal, to implement filtering of an input audio signal with a low computational complexity, and a parameterization device therefor. To this end, provided are a method for generating a filter for an audio signal, including: receiving at least one binaural room impulse response (BRIR) filter coefficients for binaural filtering of an input audio signal; converting the BRIR filter coefficients into a plurality of subband filter coefficients; obtaining average reverberation time information of a corresponding subband by using reverberation time information extracted from the subband filter coefficients; obtaining at least one coefficient for curve fitting of the obtained average reverberation time information; obtaining flag information indicating whether the length of the BRIR filter coefficients in a time domain is more than a predetermined value; obtaining filter order information for determining a truncation length of the subband filter coefficients, the filter order information being obtained by using the average reverberation time information or the at least one coefficient according to the obtained flag information and the filter order information of at least one subband being different from filter order information of another subband; and truncating the subband filter coefficient by using the obtained filter order information and a parameterization device therefor.

Examples of the disclosure describe systems and methods for presenting an audio signal to a user of a wearable head device. In an example, a received first input audio signal is processed to generate a left output audio signal and a right output audio signal presented to ears of the user. Processing the first input audio signal comprises applying a delay process to the first input audio signal to generate a left audio signal and a right audio signal; adjusting gains of the left audio signal and the right audio signal; applying head-related transfer functions (HRTFs) to the left and right audio signals to generate the left and right output audio signals. Applying the delay process to the first input audio signal comprises applying an interaural time delay (ITD) to the first input audio signal, the ITD determined based on the source location.

Embodiments are described for a system of rendering object-based audio content through a system that includes individually addressable drivers, including at least one driver that is configured to project sound waves toward one or more surfaces within a listening environment for reflection to a listening area within the listening environment; a renderer configured to receive and process audio streams and one or more metadata sets associated with each of the audio streams and specifying a playback location of a respective audio stream; and a playback system coupled to the renderer and configured to render the audio streams to a plurality of audio feeds corresponding to the array of audio drivers in accordance with the one or more metadata sets.

Methods and apparatus, including software, for collecting and managing public music performance royalties and royalty payouts are described. On the listeners side, song/audio fingerprint data is collected and transmitted to the rights owner side, where the rights owner side verifies the song/audio fingerprint data, calculates royalty payments, and in some cases, automates the royalty payments. Public music performance royalty payments are based on the song/audio fingerprint data collected by listeners/clients, as well as business logic servers.

Example systems, devices, media, and methods are described for efficiently processing an audio track of a virtual object with a head-related transfer function (HRTF). Audio tracks are processed by determining a current position (direction and optionally distance) of the virtual object with respect to the head of a user, identifying a current audio zone from predefined audio zones responsive to the determined current position where each of the audio zones has a corresponding left predefined filter and a corresponding right predefined filter, applying the left and the right predefined filters corresponding to the current audio zone to the audio track to produce a left audio signal and a right audio signal, and presenting the left audio signal with a first speaker and the right audio signal with a second speaker.

A sound signal processing device includes: a vocal remover which generates a first output signal based on first-channel and second-channel sound signals and a first coefficient indicating a vocal bandwidth to be removed; a surround sound processor which generates a second output signal by adding a surround sound effect to the first output signal; an amplifier which amplifies a signal at an amplification factor that is based on a second coefficient; a synthesizer which synthesizes the second output signal with one of the first-channel and second-channel sound signals, and synthesizes a signal that is the second output signal inverted with another one of the first-channel and second-channel sound signals; and a coefficient determination unit which sets the second coefficient such that the amplification factor, used when the vocal bandwidth to be removed is greater than a first bandwidth, is greater than the amplification factor for the first bandwidth.

An apparatus and method are disclosed for processing an audio signal. The apparatus includes an input interface, a digital filterbank having an analysis part and a synthesis part, a first phase shifter, a spectral envelope adjuster, a second phase shifter, and an output interface. The first phase shifter and the second phase shifter reduce a complexity of the digital filterbank, which includes both analysis and synthesis filters that are complex-exponential modulated versions of a prototype filter.

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.

Methods, systems, and non-transitory computer readable storage media are disclosed for rendering scene-aware audio based on acoustic properties of a user environment. For example, the disclosed system can use neural networks to analyze an audio recording to predict environment equalizations and reverberation decay times of the user environment without using a captured impulse response of the user environment. Additionally, the disclosed system can use the predicted reverberation decay times with an audio simulation of the user environment to optimize material parameters for the user environment. The disclosed system can then generate an audio sample that includes scene-aware acoustic properties based on the predicted environment equalizations, material parameters, and an environment geometry of the user environment. Furthermore, the disclosed system can augment training data for training the neural networks using frequency-dependent equalization information associated with measured and synthetic impulse responses.

The application relates to a hearing device comprising an input unit for providing first and second electric input signals representing sound signals, a beamformer filter for making frequency-dependent directional filtering of the electric input signals, the output of said beamformer filter providing a resulting beamformed output signal. The application further relates to a method of providing a directional signal. The object of the present application is to create a directional signal. The problem is solved in that the beamformer filter comprises a directional unit for providing respective first and second beamformed signals from weighted combinations of the electric input signals, an equalization unit for equalizing a phase (and possibly an amplitude) of the beamformed signals and providing first and second equalized beamformed signals, and a beamformer output unit for providing the resulting beamformed output signal from the first and second equalized beamformed signals. This has the advantage to create a directional signal where the phase of the individual components is preserved, and therefore introducing no phase distortions. The invention may e.g. be used in hearing aids, headsets, ear phones, active ear protection systems, and combinations thereof.