Abstract: The system and method for virtual navigation of a sound field through interpolation of the signals from an array of microphone assemblies utilizes an array of two or more higher-order Ambisonics (HOA) microphone assemblies, which measure spherical harmonic coefficients (SHCs) of the sound field from spatially-distinct vantage points, to estimate the SHCs at an intermediate listening position. First, sound sources near to the microphone assemblies are detected and located. Simultaneously, the desired listening position is received. Only the microphone assemblies that are nearer to said desired listening position than to any near sources are considered valid for interpolation. The SHCs from these valid microphone assemblies are then interpolated using a combination of weighted averaging and linear translation filters. The result is an estimate of the SHCs that would have been captured by a HOA microphone assembly placed in the original sound field at the desired listening position.

Abstract: The system and method for virtual navigation of a sound field through interpolation of the signals from an array of microphone assemblies utilizes an array of two or more higher-order Ambisonics (HOA) microphone assemblies, which measure spherical harmonic coefficients (SHCs) of the sound field from spatially-distinct vantage points, to estimate the SHCs at an intermediate listening position. First, sound sources near to the microphone assemblies are detected and located. Simultaneously, the desired listening position is received. Only the microphone assemblies that are nearer to said desired listening position than to any near sources are considered valid for interpolation. The SHCs from these valid microphone assemblies are then interpolated using a combination of weighted averaging and linear translation filters. The result is an estimate of the SHCs that would have been captured by a HOA microphone assembly placed in the original sound field at the desired listening position.

Abstract: The method and system for measuring low-noise acoustical impulse responses at high sampling rates of the present invention utilizes two exponential sine sweeps (ESSs) to measure the impulse responses. The first ESS is a quick sweep up to the Nyquist frequency to provide an estimate of the system response and sample the ambient noise. This measurement is used to algorithmically determine an appropriate pass-band of the system. A second, slower sweep through the pass-band alone is then executed and a corresponding band-pass filter is applied to the resulting output signal to suppress noise. The result is a measured impulse response with an improved signal-to-noise ratio and a much-reduced pre-response.

Abstract: The method and system for measuring low-noise acoustical impulse responses at high sampling rates of the present invention utilizes two exponential sine sweeps (ESSs) to measure the impulse responses. The first ESS is a quick sweep up to the Nyquist frequency to provide an estimate of the system response and sample the ambient noise. This measurement is used to algorithmically determine an appropriate pass-band of the system. A second, slower sweep through the pass-band alone is then executed and a corresponding band-pass filter is applied to the resulting output signal to suppress noise. The result is a measured impulse response with an improved signal-to-noise ratio and a much-reduced pre-response.

Abstract: The system and method of the present invention rely on combining the Speakers+Room binaural Impulse Response(s) (SRbIR) with a special kind of crosstalk cancellation (XTC) filter—one that does not degrade or significantly alter the SRbIR's spectral and temporal characteristics that are required for effective head externalization. This unique combination leads to a 3D audio filter for headphones that allows the emulation of the sound of crosstalk-cancelled speakers through headphones, and allows for fixing the perceived soundstage in space using head tracking and thus solves the major problems for externalized and robust 3D audio rendering through headphones. Furthermore, by taking advantage of a well-documented psychoacoustic fact, this system and method can produce universal 3D audio filters that work for all listeners i.e. independent of the listener's head related transfer function (HRTF).

Abstract: The system and method of the present invention rely on combining the Speakers+Room binaural Impulse Response(s) (SRbIR) with a special kind of crosstalk cancellation (XTC) filter—one that does not degrade or significantly alter the SRbIR's spectral and temporal characteristics that are required for effective head externalization. This unique combination leads to a 3D audio filter for headphones that allows the emulation of the sound of crosstalk-cancelled speakers through headphones, and allows for fixing the perceived soundstage in space using head tracking and thus solves the major problems for externalized and robust 3D audio rendering through headphones. Furthermore, by taking advantage of a well-documented psychoacoustic fact, this system and method can produce universal 3D audio filters that work for all listeners i.e. independent of the listener's head related transfer function (HRTF).

Abstract: A method and system for calculating the frequency-dependent regularization parameter (FDRP) used in inverting the analytically derived or experimentally measured system transfer matrix for designing and/or producing crosstalk cancellation (XTC) filters relies on calculating the FDRP that results in a flat amplitude vs frequency response at the loudspeakers, thus forcing XTC to be effected into the phase domain only and relieving the XTC filter from the drawbacks of audible spectral coloration and dynamic range loss. When the method and system are used with any effective optimization technique, it results in XTC filters that yield optimal XTC levels over any desired portion of the audio band, impose no spectral coloration on the processed sound beyond the spectral coloration inherent in the playback hardware and/or loudspeakers, and cause no (or arbitrarily low) dynamic range loss.

Abstract: A method and system for calculating the frequency-dependent regularization parameter (FDRP) used in inverting the analytically derived or experimentally measured system transfer matrix for designing and/or producing crosstalk cancellation (XTC) filters relies on calculating the FDRP that results in a flat amplitude vs frequency response at the loudspeakers, thus forcing XTC to be effected into the phase domain only and relieving the XTC filter from the drawbacks of audible spectral coloration and dynamic range loss. When the method and system are used with any effective optimization technique, it results in XTC filters that yield optimal XTC levels over any desired portion of the audio band, impose no spectral coloration on the processed sound beyond the spectral coloration inherent in the playback hardware and/or loudspeakers, and cause no (or arbitrarily low) dynamic range loss.