Abstract: The present embodiments relate to audio effect processing, and more particularly to a method for creating reverberation impulse responses from prerecorded or live source materials forms the basis of a family of reverberation effects. In one embodiment, segments of audio are selected and processed to form an evolving sequence of reverberation impulse responses that are applied to the original source material—that is, an audio stream reverberating itself. In another embodiment, impulse responses derived from one audio track are applied to another audio track. In a further embodiment, reverberation impulse responses are formed by summing randomly selected segments of the source audio, and imposing reverberation characteristics, including reverberation time, wet equalization, wet-dry mix, and predelay. By controlling the number and timing of the selected source audio segments, the method produces a collection of impulse responses that represent a trajectory through the source material.

Abstract: The implementation of modal processors, which involve the parallel combination resonant filters, may be costly for applications such as artificial reverberation that can require thousands of modes. In one embodiment, the input signal is decomposed into a plurality of subbands, the outputs of which are downsampled. In each downsampled band, resonant filters are applied at the downsampled sampling rate, and their output is upsampled and filtered to form the band output. In these and other embodiments, a feature of responses of the mode filters have been optimized to minimize an aspect of a residual error after a point in time.

Abstract: Preparing a sum of complex exponentials in the form of a modal filter to approximate a given impulse response is considered, and a non-iterative solution is described. In one embodiment, the mode count, and the mode frequencies, dampings, and amplitudes are estimated using the generalized eigenvalues of Hankel matrices of samples of the given impulse response.

Abstract: Preparing a sum of complex exponentials in the form of a modal filter to approximate a given impulse response is considered, and a non-iterative solution is described. In one embodiment, the mode count, and the mode frequencies, dampings, and amplitudes are estimated using the generalized eigenvalues of Hankel matrices of samples of the given impulse response.

Abstract: The implementation of modal processors, which involve the parallel combination resonant filters, may be costly for applications such as artificial reverberation that can require thousands of modes. In one embodiment, the input signal is decomposed into a plurality of subbands, the outputs of which are downsampled. In each downsampled band, resonant filters are applied at the downsampled sampling rate, and their output is upsampled and filtered to form the band output. In these and other embodiments, a feature of responses of the mode filters have been optimized to minimize an aspect of a residual error after a point in time.

Abstract: The implementation of modal processors, which involve the parallel combination resonant filters, may be costly for applications such as artificial reverberation that can require thousands of modes. In one embodiment, the input signal is decomposed into a plurality of subbands, the outputs of which are downsampled. In each downsampled band, resonant filters are applied at the downsampled sampling rate, and their output is upsampled and filtered to form the band output.

Abstract: The implementation of modal processors, which involve the parallel combination resonant filters, may be costly for applications such as artificial reverberation that can require thousands of modes. In one embodiment, the input signal is decomposed into a plurality of subbands, the outputs of which are downsampled. In each downsampled band, resonant filters are applied at the downsampled sampling rate, and their output is upsampled and filtered to form the band output.

Abstract: Spatialization of soundfields is accomplished by filtering audio signals using filters having unvarying frequency response characteristics and amplifying signals using amplifier gains adapted in response to signals representing sound source location and/or listener position. The filters are derived using a singular value decomposition process which finds the best set of component impulse responses to approximate a given target set of impulse responses corresponding to head related transfer functions. Efficient implementations for rendering reflection effects, air absorption losses and other ambient effects, and for spatializing multiple sound sources and/or generating multiple output signals are disclosed.

Abstract: A method and apparatus is capable of accurately deriving acoustic transfer functions such as head-related transfer functions (HRTF) at low cost. Various aspects of the invention include constraining the reflection geometry of a measurement system to facilitate removal of reflection effects, establishing ambient noise level and ambient reverberation time to calibrate test signals, generating soundfields using Golay code test signals, invalidating measurements by detecting test subject movement and short-duration ambient sounds, deriving distance and/or interaural time difference (ITD) using minimum-phase forms of impulse responses, and deriving equalized HRTF suitable for use in acoustic displays without knowing output or input transducer acoustical properties. Spatial resampling of derived HRTF and spectral shaping of test signals are discussed.