Abstract: Methods, digital systems, and computer readable media are provided for estimating change of amplitude and frequency in a digital audio signal by transforming a frame of the digital audio signal to the frequency domain, locating a frequency peak in the transformed frame, determining an interpolated peak of the located frequency peak, computing inner products of a portion of the transformed frame about the interpolated peak with a plurality of test signals, and estimating change of amplitude and change of frequency for the frequency peak from results of the inner products.

Abstract: This invention is a method for binaural localization using a cascade of resonators and anti-resonators to implement an HRTF (head-related transfer function). The spectrum of the cascade reproduces the magnitude spectrum of a desired HRTF. The proposed method provides a considerably more computationally efficient implementation of HRTF filters with no detectable deterioration of output quality while saving memory when storing a large quantity of HRTFs due to the parameterization of its resonators and anti-resonators. Finally, the method offers additional flexibility since the resonators and anti-resonators can be manipulated individually during the design process, making it possible to interpolate smoothly between HRTFs, reduce spectral coloring or achieve higher accuracy at perceptually relevant frequency regions. These HRTF are useful in stereo enhancement and multi-channel virtual surround simulation.

Abstract: This invention involves time-scale modification of audio signals. The invention describes overlap and add time scale modification with variable input and output buffer sizes. Seamless speed change is achieved by keeping track of previously processed data to avoid discontinuities during playback speed transitions.

Abstract: Methods, digital systems, and computer readable media are provided for detection of music in an audio signal. Music is detected by partitioning the audio signal into overlapping frames, determining a fundamental frequency of a frame in the overlapping frames, including the fundamental frequency of the frame in a histogram of fundamental frequency values of frames occurring in the audio signal prior to the frame, and indicating that music is present in the audio signal when a number of occurrences of a fundamental frequency value in the histogram exceeds a threshold.

Abstract: Rebalancing of an audio signal refers to achieving a balance of perceived loudness, typically of right and left channels, given an unbalanced input. A flexible method to automatically rebalance an audio input signal is robust against noise in extreme cases through the individual channels combined in various ways as a function of the loudness ratio between input channels.

Abstract: Methods, digital systems, and computer readable media are provided for determining a gain reduction parameter level for loudspeaker equalization by determining a noise score, an equalization effectiveness score, and an equalization non-effectiveness score for a candidate gain reduction parameter level, determining a composite quality score using the three scores, and designing a compensating filter for the loudspeaker using the candidate gain reduction parameter level if the composite quality score is better than composite quality scores of all other candidate gain reduction parameter levels.

Abstract: Audio loudspeaker virtualizers and cross-talk cancellers and methods use a combination of interaural intensity difference and interaural time difference to define virtualizing filters. This allows enlargement of a listener's sweet spot based on psychoacoustic effects.

Abstract: Methods, digital systems, and computer readable media are provided for determining a predominant fundamental frequency of a frame of an audio signal by finding a maximum absolute signal value in history data for the frame, determining a number of bits for downshifting based on the maximum absolute signal value, computing autocorrelations for the frame using signal values downshifted by the number of bits, and determining the predominant fundamental frequency using the computed autocorrelations.

Abstract: This invention locally controls the pitch of speech and audio signals. The invention is based on a seamless time scale modification (S-TSM) scheme connected to a synchronized sampling rate converter that switches between different time scale factors in a seamless manner and controls pitch during playback in a nearly continuous way.

Abstract: This invention improves the perceived quality of frequency-domain time scale modification by selection of spectral bands used in phase locking based upon a Bark scale according to the variation in human hearing frequency response. A spectral peak is identified for each band. At these peaks the phases are rotated using the phase vocoder algorithm. For a few spectral lines near these peaks, the phase differences are copied from the non-rotated spectrum. The number selected is preferably 4. Remaining spectral lines within each spectral band located farther from the peak are phase rotated using the phase vocoder algorithm. The boundaries of the spectral bands may be adjusted based upon the digital audio data to maintain important frequency groups within the same spectral band.

Abstract: Audio loudspeaker and headphone virtualizers and cross-talk cancellers and methods use separate virtual speaker locations for different Bark frequency bands and a single reverberation filter for multi-channel virtualizer inputs.

Abstract: Audio loudspeaker and headphone virtualizers and cross-talk cancellers and methods use separate virtual speaker locations for different Bark frequency bands and a single reverberation filter for multi-channel virtualizer inputs.

Abstract: Audio loudspeaker and headphone virtualizers and cross-talk cancellers and methods use separate virtual speaker locations for different Bark frequency bands and a single reverberation filter for multi-channel virtualizer inputs.

Abstract: Methods, digital systems, and computer readable media are provided for determining a gain reduction parameter level for loudspeaker equalization by determining a noise score, an equalization effectiveness score, and an equalization non-effectiveness score for a candidate gain reduction parameter level, determining a composite quality score using the three scores, and designing a compensating filter for the loudspeaker using the candidate gain reduction parameter level if the composite quality score is better than composite quality scores of all other candidate gain reduction parameter levels.

Abstract: This invention is a method for binaural localization using a cascade of resonators and anti-resonators to implement an HRTF (head-related transfer function). The spectrum of the cascade reproduces the magnitude spectrum of a desired HRTF. The proposed method provides a considerably more computationally efficient implementation of HRTF filters with no detectable deterioration of output quality while saving memory when storing a large quantity of HRTFs due to the parameterization of its resonators and anti-resonators. Finally, the method offers additional flexibility since the resonators and anti-resonators can be manipulated individually during the design process, making it possible to interpolate smoothly between HRTFs, reduce spectral coloring or achieve higher accuracy at perceptually relevant frequency regions. These HRTF are useful in stereo enhancement and multi-channel virtual surround simulation.

Abstract: Audio loudspeaker and headphone virtualizers and methods use room impulse responses with modified individual head-related transfer functions prior to superposition including middle truncation; and perform convolutions in the frequency domain with zero-padded sections to avoid circular convolution overlap.

Abstract: This invention is a method for binaural localization using a cascade of resonators and anti-resonators to implement an HRTF (head-related transfer function). The spectrum of the cascade reproduces the magnitude spectrum of a desired HRTF. The proposed method provides a considerably more computationally efficient implementation of HRTF filters with no detectable deterioration of output quality while saving memory when storing a large quantity of HRTFs due to the parameterization of its resonators and anti-resonators. Finally, the method offers additional flexibility since the resonators and anti-resonators can be manipulated individually during the design process, making it possible to interpolate smoothly between HRTFs, reduce spectral coloring or achieve higher accuracy at perceptually relevant frequency regions. These HRTF are useful in stereo enhancement and multi-channel virtual surround simulation.

Abstract: A time-domain time-scale modification method based on the synchronous overlap-and-add method consists of a generalization of the envelope-matching time-scale modification method. The cross-correlation function employs a fixed-size cross-correlation buffer to eliminate the need for normalization inside the search loop. This fixed-size cross-correlation buffer is the center of the overlap region corresponding to the case where the fine overlap adjustment value is set to zero. The computational cost of this invention is lower than any other method with a comparable quality.

Abstract: This invention locally controls the pitch of speech and audio signals. The invention is based on a seamless time scale modification (S-TSM) scheme connected to a synchronized sampling rate converter that switches between different time scale factors in a seamless manner and controls pitch during playback in a nearly continuous way.

Abstract: Audio cross-talk cancellation by inverse HRTF matrix only for low frequencies; high frequencies rely upon the natural barrier of a listener's head. The low frequency cutoff is determined by a peak in the inverse matrix of the head-related transfer functions.