Examples of the disclosure describe systems and methods for presenting an audio signal to a user of a wearable head device. According to an example method, a first input audio signal is received, the first input audio signal corresponding to a source location in a virtual environment presented to the user via the wearable head device. The first input audio signal is processed to generate a left output audio signal and a right output audio signal. The left output audio signal is presented to the left ear of the user via a left speaker associated with the wearable head device. The right output audio signal is presented to the right ear of the user via a right speaker associated with the wearable head device. 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 a gain of the left audio signal; adjusting a gain of the right audio signal; applying a first head-related transfer function (HRTF) to the left audio signal to generate the left output audio signal; and applying a second HRTF to the right audio signal to generate the right output audio signal. 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.

Systems and methods are disclosed for providing voice and noise activity detection with audio automixers that can reject errant non-voice or non-human noises while maximizing signal-to-noise ratio and minimizing audio latency.

Examples disclosed herein include a speaker. The speaker may include a group of microphones and a processor. The processor may determine a first speaker-channel identifier for a multi-speaker system at least partially responsive to a first tone captured at the group of microphones. The processor may also determine a position of a source of the captured first tone relative to the speaker at least partially responsive to position information derived from the captured first tone. The processor may also determine a second speaker-channel identifier at least partially responsive to the first speaker-channel identifier and the position of the source of the captured first tone. The processor may also determine speaker settings at least partially responsive to the second speaker-channel identifier. Related devices, systems and methods are also disclosed.

The present embodiments generally relate to enabling participants in an online gathering with networked audio to use a cancelling auralizer at their respective locations to create a common acoustic space or set of acoustic spaces shared among subgroups of participants. For example, there are a set of network connected nodes, and the nodes can contain speakers and microphones, as well as participants and node mixing-processing blocks. The node mixing-processing blocks generate and manipulate signals for playback over the node loudspeakers and for distribution to and from the network. This processing can include cancellation of loudspeaker signals from the microphone signals and auralization of signals according to control parameters that are developed locally and from the network. A network block can contain network routing and processing functions, including auralization, synthesis, and cancellation of audio signals, synthesis and processing of control parameters, and audio signal and control parameter routing.

A system for producing an encoded digital audio recording has an audio encoder that encodes a digital audio recording having a number of audio channels or audio objects. An equalization (EQ) value generator produces a sequence of EQ values which define EQ filtering that is to be applied when decoding the encoded digital audio recording, wherein the EQ filtering is to be applied to a group of one or more of the audio channels or audio objects of the recording independent of any downmix. A bitstream multiplexer combines the encoded digital audio recording with the sequence of EQ values, the latter as metadata associated with the encoded digital audio recording. Other embodiments are also described including a system for decoding the encoded audio recording.

A dual-zone automotive multimedia system may include a first infotainment device associated with a front zone of a vehicle, at least one second infotainment device associated with a rear zone of a vehicle, wherein the at least one second infotainment device includes a directional loudspeaker arranged facing the rear zone of the vehicle, and a processor programmed to transmit audio signals to the first and second infotainment devices to create sound at each of the front and rear zones, wherein the audio signal transmitted to the directional loudspeaker relates to playback at the rear zone.

Individual loudspeaker dynamics processing configuration data, for each of a plurality of loudspeakers of a listening environment, may be obtained. Listening environment dynamics processing configuration data may be determined, based on the individual loudspeaker dynamics processing configuration data. Dynamics processing may be performed on received audio data based on the listening environment dynamics processing configuration data, to generate processed audio data. The processed audio data may be rendered for reproduction via a set of loudspeakers that includes at least some of the plurality of loudspeakers, to produce rendered audio signals. The rendered audio signals may be provided to, and reproduced by, the set of loudspeakers.

The following coding scenario is addressed: A number of audio source signals need to be transmitted or stored for the purpose of mixing wave field synthesis, multi-channel surround, or stereo signals after decoding the source signals. The proposed technique offers significant coding gain when jointly coding the source signals, compared to separately coding them, even when no redundancy is present between the source signals. This is possible by considering statistical properties of the source signals, the properties of mixing techniques, and spatial hearing. The sum of the source signals is transmitted plus the statistical properties of the source signals, which mostly determine the perceptually important spatial cues of the final mixed audio channels. Source signals are recovered at the receiver such that their statistical properties approximate the corresponding properties of the original source signals. Subjective evaluations indicate that high audio quality is achieved by the proposed scheme.

An apparatus comprising means configured to: obtain at least one impulse response; obtain at least one reflection filter based on the obtained at least one impulse response, wherein the at least one reflection filter is configured to determine at least one early reflection from an acoustic surface which is not overlapped in time by any other reflection, wherein a duration of the at least one early reflection is shorter than a duration of the obtained at least one impulse response. In addition, an apparatus comprising means configured to: obtain at least one impulse response, wherein the at least one impulse response is configured with a perceivable timbre during rendering; create a timbral modification filter; obtain at least one audio signal; and render at least one output audio signal based n the at least one audio signal, wherein the at least one output signal is based on an application of the timbral modification filter.

The invention relates to a method for improving the sound quality of a sound reproduction or recording in a room, the method comprising the steps of measuring an impulse response that comprises the linear response of the room; performing a time domain analysis to determine the resonances of the room and for a chosen group of room resonances determining a corresponding group of filters that, when inserted in a sound reproduction or recording chain in said room will counteract the unwanted effect of said chosen group of room resonances on the sound quality of sound reproduction or recording made in the room. The invention further relates to a device designed to implement the method according to the invention and to the use of a measure of amplitude decay as a function of frequency of a measured impulse response of a sound reproduction or recording system in a room to determine one or more resonance frequencies, the total or partial compensation of which will improve the sound quality of sound reproductions or recordings made in the room.