Abstract: There is disclosed a method and apparatus for changing the frequency content of an input spectrum and a method and apparatus for reducing the perceptibility of a component of an input signal. The first aspect involves adjusting frequency components of the input spectrum in response to a time varying adjustment frequency spectrum to produce an output frequency spectrum including adjusted frequency components of the input spectrum. The time varying input spectrum may be produced by selectively addressing a number of individual sub-spectra at different times. In addition, the input spectrum may be divided into a plurality of sub-spectra and each sub-spectrum may be operated on separately by a different adjustment frequency spectrum at different times. In addition, a perceptual model may be used to enhance the adjustment of the input spectrum or sub-spectra.

Abstract: The invention is a method for transforming a source individual's voice so as to adopt the characteristics of a target individual's voice. The excitation signal component of the target individual's voice is extracted and the spectral envelope of the source individual's voice is extracted. The transformed voice is synthesized by applying the spectral envelope of the source individual to the excitation signal component of the voice of the target individual. A higher quality transformation is achieved using an enhanced excitation signal created by replacing unvoiced regions of the signal with interpolated data from adjacent voiced regions. Various methods of transforming the spectral characteristics of the source individual's voice are also disclosed.

Abstract: A method for shifting the timbre and/or pitch of an input signal samples the input signal at a first rate and stores the samples in a memory buffer. A digital signal processor resamples the stored input signal at a rate that differs from the first rate at which the input note is originally sampled and stores the resampled input signal in a second memory buffer. A pitch shifter shifts the pitch of the input signal by periodically scaling the resampled input signal by a window function to create an output signal. The rate at which the resampled data is replicated by the window function determines the pitch of the output signal.

Abstract: A method for shifting the timbre and/or pitch of an input signal samples the input signal at a first rate and stores the samples in a memory buffer. A digital signal processor resamples the stored input signal at a rate that differs from the first rate at which the input note is originally sampled and stores the resampled input signal in a second memory buffer. A pitch shifter shifts the pitch of the input signal by periodically scaling the resampled input signal by a window function to create an output signal. The rate at which the resampled data is replicated by the window function determines the pitch of the output signal.

Abstract: A method for shifting the timbre and/or pitch of an input signal samples the input signal at a first rate and stores the samples in a memory buffer. A digital signal processor resamples the stored input signal at a rate that differs from the first rate at which the input note is originally sampled and stores the resampled input signal in a second memory buffer. A pitch shifter shifts the pitch of the input signal by periodically scaling the resampled input signal by a window function to create an output signal. The rate at which the resampled data is replicated by the window function determines the pitch of the output signal.

Abstract: A karaoke type system allows a participant to sing on key with a prerecorded song. A microphone produces an input signal that corresponds to a singer's voice, and a pitch corrector samples the input vocal signal and determines its pitch. The pitch corrector reads a series of codes that are stored with the prerecorded song that indicates the pitch at which the input vocal signal is to be sung in order to be on key with the prerecorded song. The pitch corrector shifts the pitch of the input vocal signal to be on key.

Abstract: Disclosed are a method and apparatus for analyzing an input vocal signal to produce a plurality of harmony signals that are combined with the input vocal signal to produce a multivoice signal. The method makes a current estimate of the fundamental frequency of the input vocal signal and determines if the current estimate is the correct estimate of the fundamental frequency. If the current estimate is correct, a reference note is assigned to correspond to the current estimate and a plurality of harmony notes are selected to correspond to the reference note. The method then generates a plurality of harmony signals by scaling the input vocal signal with a piecewise linear approximation of a Hanning window to extract a portion of the input vocal signal and by replicating the extracted portion at a plurality of rates equal to the fundamental frequencies of each of the harmony notes. The plurality of harmony signals and the input vocal signal are combined to produce the multivoice signal.

Abstract: Disclosed are a method and apparatus for analyzing an input vocal signal to produce a plurality of harmony signals that are combined with the input vocal signal to produce a multivoice signal. The method makes a current estimate of the fundamental frequency of the input vocal signal and determines if the current estimate is the correct estimate of the fundamental frequency. If the current estimate is correct, a reference note is assigned to correspond to the current estimate and a plurality of harmony notes are selected to correspond to the reference note. The method then generates a plurality of harmony signals by scaling the input vocal signal with a piecewise linear approximation of a Hanning window to extract a portion of the input vocal signal and by replicating the extracted portion at a plurality of rates equal to the fundamental frequencies of each of the harmony notes. The plurality of harmony signals and the input vocal signal are combined to produce the multivoice signal.

Abstract: A pitch detector is disclosed that automatically recognizes the pitch of musical notes quickly and outputs the pitch information in a variety of formats. The detector employs a microprocessor that samples the signal from a musical instrument or voice at regular intervals using an analog-to-digital converter and then utilizes both amplitude and time information from the waveform to determine the fundamental period of the note, while rejecting the harmonic components. The microprocessor analyzes the waveform looking for peaks that are approximately equal in amplitude separated by opposite polarity peaks. The time intervals between the peaks must be approximately equal too. Timing information is measured using more than one point on the waveform to avoid inaccuracies caused by temporary distortions of the waveform. The timing points are chosen at points where the slope of the waveform is high for substantially optimal accuracy.