Abstract: A system and method that incorporates the subject disclosure may include, for example, receiving a multichannel audio stream; forming a front channel audio stream of the multichannel audio stream, including combining a first subset of audio channels of the multichannel audio stream to form the front channel audio stream; forming a surround channel audio stream of the multichannel audio stream including combining a second subset of audio channels of the multichannel audio stream to form the surround channel audio stream; providing the front channel audio stream to a primary set of speakers positioned in front of a listener and providing the surround channel audio stream to a supplemental speaker positioned behind the listener; and synchronizing the front channel audio stream and the surround channel audio stream. Additional embodiments are disclosed.

Abstract: An apparatus comprises a receiver (201) receiving a multi-channel audio signal representing audio for a scene. An extractor (203) extracts at least one directional audio component by applying a spatial filtering to the multi-channel signal where the spatial filtering is dependent on the multi-channel audio signal. A feature processor (205) determines a set of features for the first directional audio component and a categorizer (207) determines a first audio source category out of a plurality of audio source categories for the directional audio signal in response to the set of features. An assigner (209) assigns a first audio source property to the first directional audio component from a set of audio source properties for the first audio source category. The apparatus may provide very advantageous categorization and characterization of individual audio sources/components present in a multi-channel signal. This may be advantageous e.g. for visualization of audio events.

Abstract: The invention relates to a method for processing a signal, in particular a digital audio signal, suitable for being implemented by a digital signal processor (DSP) having libraries for calculating Fourier transforms from the complex number space to the complex number space, for digitally processing P input signals, P being an integer at least equal to 2, more particularly for filtering said P input signals by the convolution of sampled fast Fourier transforms (FFT), thus obtaining Q output signals, Q being an integer at least equal to 2.

Abstract: The invention relates to a method for processing a signal, in particular a digital audio signal, suitable for being implemented by a digital signal processor (DSP) having libraries for calculating Fourier transforms from the complex number space to the complex number space, for digitally processing P input signals, P being an integer at least equal to 2, more particularly for filtering said P input signals by the convolution of sampled fast Fourier transforms (FFT), thus obtaining Q output signals, Q being an integer at least equal to 2.