Abstract: A system of linear equations is used to upmix a number N of audio signals to generate a larger number M of audio signals that are psychoacoustically decorrelated with respect to one another and that can be used to improve the representation of a diffuse sound field. The linear equations are defined by a matrix that specifies a set of vectors in an M dimensional space that are substantially orthogonal to each other. Methods for deriving the system of linear equations are disclosed.

Abstract: A signal from each of an array of microphones is analyzed. For at least one subset of microphone signals, a time difference is estimated, which characterizes the relative time delays between the signals in the subset. A direction is estimated from which microphone inputs arrive from one or more acoustic sources, based at least partially on the estimated time differences. The microphone signals are filtered in relation to at least one filter transfer function, related to one or more filters. A first filter transfer function component has a value related to a first spatial orientation of the arrival direction, and a second component has a value related to a spatial orientation that is substantially orthogonal in relation to the first. A third filter function may have a fixed value. A driving signal for at least two loudspeakers is computed based on the filtering.

Abstract: A method and apparatus for limiting the absolute magnitude of an audio signal. The method may include firstly variable-gain reducing the gain of an audio signal, and then secondly variable-gain reducing the gain of the audio signal faster than the first variable-gain reduction, thereby limiting the absolute magnitude of the audio signal to a threshold. The first variable-gain reduction may include variable-gain reducing the gain of the audio signal in a first stage, and the second variable-gain reduction may include variable-gain reducing the gain of the audio signal in a second stage that reduces the gain faster than the first stage. The second variable-gain reduction may include delaying the audio signal, finding a peak among the delayed audio signal, calculating a fast gain from a found peak, and modifying the delayed audio signal with the calculated fast gain.

Abstract: Audio signals that represent a sound field with increased spatial resolution are obtained by deriving signals that represent the sound field with high-order angular terms. This is accomplished by analyzing input audio signals representing the sound field with zero-order and first-order angular terms to derive statistical characteristics of one or more angular directions of acoustic energy in the sound field. Processed signals are derived from weighted combinations of the input audio signals in which the input audio signals are weighted according to the statistical characteristics. The input audio signals and the processed signals represent the sound field as a function of angular direction with angular terms of one or more orders greater than one.

Abstract: The invention provides a method and system for forming an output impulse response function. The method includes the steps of creating an initial impulse response, and dividing the impulse response into a head portion and a tail portion. The tail portion is high pass filtered, and low frequency components of the head portion are boosted. The low frequency boosted and high pass filtered respective head and tail portions are then combined into a modified output impulse response, which can then be used to spatialize an audio signal by convolving it.

Abstract: A signal from each of an array of microphones is analyzed. For at least one subset of microphone signals, a time difference is estimated, which characterizes the relative time delays between the signals in the subset. A direction is estimated from which microphone inputs arrive from one or more acoustic sources, based at least partially on the estimated time differences. The microphone signals are filtered in relation to at least one filter transfer function, related to one or more filters. A first filter transfer function component has a value related to a first spatial orientation of the arrival direction, and a second component has a value related to a spatial orientation that is substantially orthogonal in relation to the first. A third filter function may have a fixed value. A driving signal for at least two loudspeakers is computed based on the filtering.

Abstract: A system for providing a listener with an augmented audio reality in a geographical environment said system comprising a position locating system for determining a current position and orientation of a listener in.

Abstract: Audio signals that represent a sound field with increased spatial resolution are obtained by deriving signals that represent the sound field with high-order angular terms. This is accomplished by analyzing input audio signals representing the sound field with zero-order and first-order angular terms to derive statistical characteristics of one or more angular directions of acoustic energy in the sound field. Processed signals are derived from weighted combinations of the input audio signals in which the input audio signals are weighted according to the statistical characteristics. The input audio signals and the processed signals represent the sound field as a function of angular direction with angular terms of one or more orders greater than one.

Abstract: A method to process audio signals, an apparatus accepting audio signals, a carrier medium that carried instructions for a processor to implement the method to process audio signals, and a carrier medium carrying filter data to implement a filter of audio signals. The method includes filtering a pair of audio input signals by a process that produces a pair of output signals corresponding to the results of: filtering each of the input signals with a HRTF filter pair, and adding the HRTF filtered signals. The HRTF filter pair is such that a listener listening to the pair of output signals through headphones experiences sounds from a pair of desired virtual speaker locations.

Abstract: The sample rate of a digital signal is converted by a digital simulation of an analog filter. The simulation can update the states of complex poles in the analog filter at arbitrary times using different techniques. One technique updates the states at variable rates. Other techniques update the states at a fixed rate in response to values of input or output samples that are modified to account for offsets between the times of the samples and the times the states are updated. The states may be updated by using interpolations of complex exponential functions. Values of the complex exponential functions may be obtained from a product of values obtained from multiple lookup tables.

Abstract: An apparatus for creating, utilizing a pair of oppositely opposed headphone speakers, the sensation of a sound source being spatially distant from the area between the pair of headphones, the apparatus comprising: (a) a series of audio inputs representing audio signals being projected from an idealised sound source located at a spatial location relative to the idealised listener; (b) a first mixing matrix means interconnected to the audio inputs and a series of feedback inputs for outputting a predetermined combination of the audio inputs as intermediate output signals; (c) a filter system of filtering the intermediate output signals and outputting filtered intermediate output signals and the series of feedback inputs, the filter system including separate filters for filtering the direct response and short time response and an approximation to the reverberant response, in addition to the feedback response filtering for producing the feedback inputs; and (d) a second matrix mixing means combining the filtered

Abstract: An apparatus for creating, utilizing a pair of oppositely opposed headphone speakers, the sensation of a sound source being spatially distant from the area between the pair of headphones, the apparatus comprising: (a) a series of audio inputs representing audio signals being projected from an idealised sound source located at a spatial location relative to the idealised listener; (b) a first mixing matrix means interconnected to the audio inputs and a series of feedback inputs for outputting a predetermined combination of the audio inputs as intermediate output signals; (c) a filter system of filtering the intermediate output signals and outputting filtered intermediate output signals and the series of feedback inputs, the filter system including separate filters for filtering the direct response and short time response and an approximation to the reverberant response, in addition to the feedback response filtering for producing the feedback inputs; and (d) a second matrix mixing means combining the filtered

Abstract: The sample rate of a digital signal is converted by a digital simulation of an analog filter. The simulation can update the states of complex poles in the analog filter at arbitrary times using different techniques. One technique updates the states at variable rates. Other techniques update the states at a fixed rate in response to values of input or output samples that are modified to account for offsets between the times of the samples and the times the states are updated. The states may be updated by using interpolations of complex exponential functions. Values of the complex exponential functions may be obtained from a product of values obtained from multiple lookup tables.

Abstract: The invention provides a method and system for forming an output impulse response function. The method includes the steps of creating an initial impulse response, and dividing the impulse response into a head portion and a tail portion. The tail portion is high pass filtered, and low frequency components of the head portion are boosted. The low frequency boosted and high pass filtered respective head and tail portions are then combined into a modified output impulse response, which can then be used to spatialize an audio signal by convolving it.

Abstract: A system for providing a listener with an augmented audio reality in a geographical environment said system comprising a position locating system for determining a current position and orientation of a listener in said geographical environment; an audio track creation system for creating an audio track having a predetermined spatialization component dependent on an apparent location of an apparent source associated with said audio track in said geographical environment; an audio track rendering system adapted to render an audio signal based on said audio track to a series of speakers surrounding said listener such that said listener experiences an apparent preservation of said spatialization component; and an audio track playback system interconnected to said position locating system and said audio track creation system and adapted to forward a predetermined audio track to said audio rendering system for rendering depending on said current position and orientation of said listener in said geographical environ

Abstract: In a multi viewer environment where multiple viewers simultaneously experience an audio-visual production, with the visual production occurring on a display surface, a method of increasing the perceived reality of the audio stream of the production, the method comprising the steps of: (a) locating a series of speakers along a periphery of the viewing audience; (b) panning an audio stream between the series of speakers so as to provide for the sense of an audio sound moving along the side of the viewing audience. In preferred embodiments, the output of one of the speakers is delayed relative to another speaker.

Abstract: A method of processing a series of input audio signals representing a series of virtual audio sound sources placed at predetermined positions around a listener to produce a reduced set of audio output signals for playback over speaker devices placed around a listener, the method comprising the steps of: (a) for each of the input audio signals and for each of the audio output signals: (i) convolving the input audio signals with an initial head portion of a corresponding impulse response mapping substantially the initial sound and early reflections for an impulse response of a corresponding virtual audio source to a corresponding speaker device so as to form a series of initial responses; (b) for each of the input audio signals and for each of the audio output signals: (i) forming a combined mix from the audio input signals; and (ii) forming a combined convolution tail from the tails of the corresponding impulse responses; (iii) convolving the combined mix with the combined convolution tail to form a combined t

Abstract: A method for creating a multichannel audio signal that provides the impression of spatial sound is disclosed, the method comprising providing an expected multichannel audio output signal having spatialised soundfield components; perturbing the spatial components; and utilising the perturbed spatial components to determine the multichannel audio signal. The perturbations can be substantially in accordance with the expected head movements of listeners of the audio signal and can be derived from a group of listeners to the audio signal. The expected head movements also preferable include an added substantially random movement.

Abstract: A method for the creation of acoustic impulse responses for utilization in rendering to an array of speakers comprising the steps of measuring a room response function; extracting a series of discrete time arrivals from the measured room response function so as to have a reverberant residual response function; separately rendering the extracted series and the reverberant residual response function to the array of speakers to form a discrete response and a residual response; combining the discrete response and the residual response to form an acoustic impulse response for the array of speakers.

Abstract: A method of creating a compressed audio output signal from a series of input audio signals is disclosed and claimed. In one embodiment, the method may include, for each of the input audio signals a) precomputing a transform corresponding to the desired compression format of the output audio signal. This may be followed by b) precomputing ancillary information relating to the compression of the transformed input audio. Next, the method may include c) mixing together the transformed input signals in the transform domain to produce an output transform domain signal. The method may then include d) algorithmically combining together the precomputed ancillary information to determine a suitable decompression strategy. Lastly, the method may include e) outputting compressed audio data comprising the output transform domain signal and the combined ancillary information.