Texas Instruments Corporation

Steven David Trautmann

Atsuhiro Sakurai

Ryo Tsutsui

Yoshihide Iwata

Akihiro Yonemoto

Daniel L Zelazo

Hironori Kakemizu

Judd E Heape

Christopher J Poletto

A method and apparatus for polarity detection. The method includes applying a band-pass filter to an impulse response of a loudspeaker, applying an exponential weighting to the band-pass filtered impulse response, wherein the exponential decay parameter is related to the higher corner frequency of the band-pass filter, finding the maximum peak in a waveform of sampled impulse responses, and detecting the connection polarity of the maximum peak as the polarity of the peak.

Method and apparatus for polarity detection of loudspeaker

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.

Method and system for estimating frequency and amplitude change of spectral peaks

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.

Binaural sound localization using a formant-type cascade of resonators and anti-resonators

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.

Method and system for music detection

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.

Method and system for determining a gain reduction parameter level for loudspeaker equalization

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.

Rebalancing of audio

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.

Virtualizer sweet spot expansion

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.

Method and system for determining predominant fundamental frequency

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.

Local pitch control based on seamless time scale modification and synchronized sampling rate conversion

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.

Phase locking method for frequency domain time scale modification based on a bark-scale spectral partition

A method for creating a stereophonic sound image out of a monaural signal combines two sub-methods. Comb filters decorrelate the left and right channel signals. Intra-aural difference cues, such as an Intra-Aural Time Difference (ITD) and an Intra-aural Intensity Difference (IID) separated channels. Strict complementary (SC) linear phase FIR filters divide the incoming monaural signal into three frequency band separation. The comb filters and ITD/IID applied to the low and high frequency bands create a simulated stereo sound image for the instruments other than human voice. Listening tests indicate that this invention provides a wider stereo sound image than previous methods, while retaining human voice centralization. Since the comb filter solution and ITD/IID solution can share the same filter bank, the computational cost of this method is almost the same as the previous method.

Stereo synthesizer using comb filters and intra-aural differences

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.

Virtualizer with cross-talk cancellation and reverb

A method of converting single channel audio (mono) signals to two channel audio (stereo) signals using simple filters and an Intra-aural Time Difference (ITD) is presented. This method does not distort the spectral content of the original signal very much, and has low computation requirements. A variation is proposed which also uses Intra-aural Intensity Difference (IID).

Low computation mono to stereo conversion using intra-aural differences

A system for sharing wavelet domain components among encoded signals receives a set of signals decomposed and encoded according to a wavelet transform. The decomposed and encoded signals each include a set of wavelet coefficients at each level of the decomposition of the encoded signal. Using a vector quantization technique, the system identifies one or more sets of wavelet coefficients that are sharable among two or more of the decomposed and encoded signals at a particular level of decomposition. The system then stores the sets of wavelet coefficients of the decomposed and encoded signals. Each identified sharable set of wavelet coefficients at a particular level of decomposition is stored only once and shared by two or more of the decomposed and encoded signals.

Sharing wavelet domain components among encoded signals

A new method is proposed that produces stereophonic sound image out of monaural signal within a selected frequency regions. The system employs a strictly complementary (SC) linear phase FIR filter pair that separates input signal into different frequency regions. A pair of comb filters is applied to one of the filter's output. This implementation allows a certain frequency range to be relatively localized at center while the other sounds are perceived in a wider space.

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Ibaraki, Japan

Steven David Trautmann