In one example in accordance with the present disclosure, a system is described. The system includes a decompose device to decompose a multi-channel audio stream into at least a first portion and a second portion. A synthesis device of the system independently synthesizes harmonics in each of the first portion and the second portion using different harmonic models. An audio generator of the system combines synthesized harmonics from the first portion and the second portion with the multi-channel audio stream to generate a synthesized audio output.

Systems and methods for rendering audio signals are disclosed. In some embodiments, a method may receive an input signal including a first portion and the second portion. A first processing stage comprising a first filter is applied to the first portion to generate a first filtered signal. A second processing stage comprising a second filter is applied to the first portion to generate a second filtered signal. A third processing stage comprising a third filter is applied to the second portion to generate a third filtered signal. A fourth processing stage comprising a fourth filter is applied to the second portion to generate a fourth filtered signal. A first output signal is determined based on a sum of the first filtered signal and the third filtered signal. A second output signal is determined based on a sum of the second filtered signal and the fourth filtered signal. The first output signal is presented to a first ear of a user of a virtual environment, and the second output signal is presented to the second ear of the user. The first portion of the input signal corresponds to a first location in the virtual environment, and the second portion of the input signal corresponds to a second location in the virtual environment.

Environmental sound is recorded using one or more microphones. A source of the recorded environmental sound is classified. The recorded environmental sound is weighted based on the classification of the source and the source media sound using a weighting mode to determine whether to mix the recorded environmental. The recorded environmental sound is mixed with source media sound to produce a mixed sound based on the determination. The mixed sound is played over one or more speakers.

Determination of a composite acoustic parameter value for a headset is presented herein. A directionally enhanced audio signal is generated based on audio signals from an acoustic sensor array and a spatial signal enhancement filter that is directed for enhancement of a sound source. A SNR improvement value is determined based on a SNR value of the directionally enhanced audio signal and a SNR value of an audio signal from an acoustic sensor of the acoustic sensor array. The SNR improvement value is input into a model that maps SNR improvement values to spatial acoustic parameters to determine a spatial acoustic parameter. A temporal acoustic parameter is determined based on the audio signals. The composite acoustic parameter value is determined based on the spatial acoustic parameter and a temporal acoustic parameter value. Audio content presented to a user is adjusted based in part on the composite acoustic parameter value.

The present invention provides a method for processing audio data, comprising the steps of providing input audio data containing a mixture of audio data including first audio data of a first musical timbre and second audio data of a second musical timbre different from said first musical timbre, decomposing the input audio data to provide decomposed data representative of the first audio data, transforming the decomposed data to obtain third audio data.

A method for spatial audio signal processing including: obtaining, from a first capture device, at least one first audio signal and at least one first direction parameter for at least one frequency band; obtaining, from a second capture device, at least one second audio signal and at least one second direction parameter for the at least one frequency band; obtaining a first position associated with the first capture device; obtaining a second position associated with the second capture device; determining a distance parameter for the at least one frequency band in relation to the first position based, at least partially, on the at least one first direction parameter and the at least one second direction parameter; and enabling an output and/or store of the at least one first audio signal, the at least one first direction parameter and the distance parameter.

An audio signal processing method and apparatus for adaptively adjusting a decorrelator. The method comprises obtaining a control parameter and calculating mean and variation of the control parameter. Ratio of the variation and mean of the control parameter is calculated, and a decorrelation parameter is calculated based on the said ratio. The decorrelation parameter is then provided to a decorrelator.

The present disclosure relates to an audio device (900) for providing an improved three-dimensional sound experience by means of the generated soundfield. To achieve this, the audio device (900) comprises a housing (901), which has an elliptical torus shape and a plurality of loudspeakers (903a-903h), and a processing circuitry (1310). The processing circuitry is configured to process a plurality of input signals (L, R, UL, UR) in a manner, which enables the plurality of loudspeakers (903a-903h) to form at least a first (DH1, DH3) and second (DH2) horizontal dipoles for crosstalk cancellation within at least two different frequency ranges (HF, MF), and to form at least a first vertical dipole (DV1, DV3) for sound elevation (1204a, 1204b) of the soundfield. Hereby, the desired frequency ranges (HF, MF) may be adjusted using an appropriated distance of the plurality of loudspeakers (903a-903h).

Methods for rendering audio for playback by two or more speakers are disclosed. The audio includes one or more audio signals, each with an associated intended perceived spatial position. Relative activation of the speakers may be a cost function of a model of perceived spatial position of the audio signals when played back over the speakers, a measure of proximity of the intended perceived spatial position of the audio signals to positions of the speakers, and one or more additional dynamically configurable functions. The dynamically configurable functions may be based on at least one or more properties of the audio signals, one or more properties of the set of speakers and/or one or more external inputs.

The present invention relates to a method and an apparatus for processing a signal, which are used for effectively reproducing a multimedia signal, and more particularly, to a method and an apparatus for processing a signal, which are used for implementing filtering for multimedia signal having a plurality of subbands with a low calculation amount. To this end, provided are a method for processing a multimedia signal including: receiving a multimedia signal having a plurality of subbands; receiving at least one proto-type filter coefficients for filtering each subband signal of the multimedia signal; converting the proto-type filter coefficients into a plurality of subband filter coefficients; truncating each subband filter coefficients based on filter order information obtained by at least partially using characteristic information extracted from the corresponding subband filter coefficients, the length of at least one truncated subband filter coefficients being different from the length of truncated subband filter coefficients of another subband; and filtering the multimedia signal by using the truncated subband filter coefficients corresponding to each subband signal and an apparatus for processing a multimedia signal using the same.