Abstract: A method and apparatus for transient detection and time-scaling an audio signal detects transients and scales only intervals located between transients to avoid artifacts. In one embodiment, the transient detection process compares frequency characteristic energy between succeeding windows of the audio signal and calculates values of an energy curve where the energy increases. Transients are detected at maxima of the energy curve.

Abstract: A method and apparatus for an audio scrubbing system for synchronizing audio to an asynchronous clock while preserving pitch utilizes a phase-vocoder to implement time-scaling without pitch-shifting.

Abstract: A system for pitch-shifting an audio signal wherein resampling is done in the frequency domain. The system includes a method for pitch-shifting a signal by converting the signal to a frequency domain representation and then identifying a specific region in the frequency domain representation. The region being located at a first frequency location. Next, the region is shifted to a second frequency location to form a adjusted frequency domain representation. Finally, the adjusted frequency domain representation is transformed to a time domain signal representing the input signal with shifted pitch. This eliminates the expensive time domain resampling stage and allows the computational costs to become independent of the pitch modification factor.

Abstract: A method and apparatus for synchronizing audio to an asynchronous clock while preserving pitch utilizes a phase-vocoder to implement time-scaling without pitch-shifting.

Abstract: A fingerprint of an audio signal is generated based on the energy content in frequency subbands. Processing techniques assure a robust identification fingerprint that will be useful for signals altered subsequent to the generation of the fingerprint. The fingerprint is compared to a database to identify the audio signal.

Abstract: Apparatus providing a “reverse lattice” structure for IIR and FIR filters, which exhibits desirable scaling properties at no additional computational or memory cost. The reverse lattice structure implementation of an IIR filter scales down signals at sections corresponding to reflection coefficients close to 1 in magnitude. This feature provides automatic scaling for poles lying close to the unit circle which would otherwise produce large amplification factors.

Abstract: A method and apparatus for removing or amplifying voice or other signals panned to the center of a stereo recording utilizes frequency domain techniques to calculate a frequency dependent gain factor based on the difference between the frequency domain spectra of the stereo channels.

Abstract: A method for obtaining a clean speech signal in a communication system having a transducer for receiving a clean speech signal from a user and having a pair of loudspeakers for providing an output signal to the user. The output signal contains loudspeaker signals which interfere with the clean speech signal, the loudspeaker signals traveling through acoustic paths to reach the transducer. The transducer receives an input signal containing the loudspeaker signals and the clean speech signal. The method includes a number of steps, namely, performing a short time Fourier transform (STFT) on the input signal to obtain at least one frequency component, performing a short time Fourier transform (STFT) on the loudspeaker signals to obtain frequency components, summing the frequency components to obtain an interference sum, and subtracting the interference sum from the at least one frequency component to obtain the clean speech signal for translation into a time domain.

Abstract: Improved methods and systems for generating quantized filter coefficients for a desired filter response are provided. One embodiment provides a technique that can be used to quantize the coefficients of finite impulse-response filters in a way that results in better performance than would be possible using simple truncation or rounding. The technique can be applied, e.g., to highpass, lowpass, or bandpass filters.

Abstract: An apparatus and method for determining the tempo and locating the downbeats of music encoded by an audio track performs a cross-correlation between a click track and a pulse track to indicate tempo candidates and between the click track and a series of pulses to determine downbeat candidates. The rhythm of the track is modified by altering segments located between the beats before playback. Swing is added by lengthening and shortening certain segments and the time-signature is modified by deleting certain segments.

Abstract: Techniques for synthesizing a time-domain signal. The time-domain signal is partitioned into a number of time-domain frames and a waveform in generated for each time-domain frame. Each waveform includes one or more sinusoids. The waveform is generated by selecting a sinusoid for synthesis and computing a set of parameter values (e.g. the start and end amplitude, frequency, and phase values) for the selected sinusoid. A template is determined for the selected sinusoid based on the computed parameter values and a selected window function. The frequency-domain template is such that the amplitude of the selected sinusoid in the time domain matches, at a time-domain frame boundary, the amplitude of a corresponding sinusoid in an adjacent time-domain frame. The template is added to a frequency-domain frame. The process is repeated for each sinusoid in the waveform. After all sinusoids have been processed, the frequency-domain frame is transformed to a time-domain frame.

Abstract: A method and apparatus that allows a musician or disk-jockey to jump to selected beats while playing a digital music file. Numeric pads on a user interface access beat pointers which point to the location of a selected jump beat in the file. The jump beat then replaces or overlays the beat currently being played. Alternatively, a pattern of beats can be selected to overlay the song as it is played.

Abstract: In accordance with the present invention, a method and apparatus are provided wherein loop discontinuities are eliminated. In the case of amplitude discontinuities, the harmonic amplitudes contained in the loop are progressively scaled over the duration of the loop, so that for each harmonic the loop end amplitude matches the loop beginning amplitude. In the case of phase discontinuities, the harmonic phases are progressively shifted over the duration of the loop, so that for each harmonic the loop end phase matches the loop beginning phase. Shifting the phase is accomplished by slightly altering the frequency of the harmonics to produce the desired amount of phase-shift at the loop end. In accordance with the present invention, the method also provides a technique to select loop begin and end points to minimize the amount of phase adjustment.

Abstract: In accordance with the present invention, a method and apparatus are provided wherein loop discontinuities are eliminated. In the case of amplitude discontinuities, the harmonic amplitudes contained in the loop are progressively scaled over the duration of the loop, so that for each harmonic the loop end amplitude matches the loop beginning amplitude. In the case of phase discontinuities, the harmonic phases are progressively shifted over the duration of the loop, so that for each harmonic the loop end phase matches the loop beginning phase. Shifting the phase is accomplished by slightly altering the frequency of the harmonics to produce the desired amount of phase-shift at the loop end. In accordance with the present invention, the method also provides a technique to select loop begin and end points to minimize the amount of phase adjustment.

Abstract: Method and apparatus for time-scaling and/or pitch shifting by discarding and/or repeating segments of a signal. The signal is stored as a series of samples in a memory where it is readable by one or more read pointers. Periodicity of segments of the signal is determined by evaluating normalized cross-correlation over a range of possible periods. Transients are detected by monitoring changes in rms signal value. To achieve time compression or time stretching, a segment is skipped/discarded whenever a maximum time-discrepancy between the current output and an ideal output is reached or a high periodicity is detected, a jump of the optimal length would not make this time discrepancy too high, and no transient is present in the segment to be skipped/discarded.

Abstract: Estimating the time-varying spectrum envelope of a time-varying signal facilitates pitch modification and other shifting of signal content in the frequency domain. Local maxima of a spectrum of the signal are identified by applying a masking curve. The masking curve has a peak at the particular maximum and descends away therefrom the local maximum. Local maxima falling below the local maximum are eliminated. The slope of the masking curve is varied in accordance with measured parameters of the spectrum to decrease or eliminate spurious peaks. Thereafter, a smoothing procedure may be applied to smooth the spectrum in frequency.

Abstract: For various values of an integer p, short-term discrete Fourier transforms X(p,k) and X.sub..tau. (p,k) of the audio signal shifted by pR samples and of the audio signal shifted by pR+.tau. samples are respectively calculated, R denoting a predetermined integer and .tau. denoting a time shift between the echo and the portion of signal which gave rise to it, a corrected signal is calculated in the frequency domain by nonlinear transformation of X(p,k) taking account, for the various values of the frequency index k, of a value representative of an attenuation factor relating to the echo and of X.sub..tau. (p,k), and a short-term component of a corrected signal is calculated in the time domain by short-term inverse Fourier transform of the corrected signal in the frequency domain. The corrected signal in the time domain is then formed by weighted summation of its short-term components. Application to the digital reprocessing of analog recordings.