Abstract: An apparatus and method of reconstructing a linear prediction synthesis filter excitation signal, by: receiving a signal representative of output from a linear prediction synthesis filter, producing therefrom a deterministic signal comprising a magnitude spectrum (50) and a phase spectrum (52); and producing (54) the reconstructed excitation signal from the deterministic signal and a noise signal.

Abstract: A spectrum-based speech enhancement system estimates and tracks the noise spectrum of a mixed speech and noise signal. The system frames and windows a digitized signal and applies the frames to a fast Fourier transform processor to generate discrete Fourier transformed (DFT) signals representing the speech plus noise signal. The system calculates the power spectrum of each frame. The speech enhancement system employs a leaky integrator that is responsive to identified noise-only components of the signal. The leaky integrator has an adaptive time-constant which compensates for non-stationary environmental noise. In addition, the speech enhancement system identified noise-only intervals by using a technique that monitors the Teager energy of the signal. The transition between noise-only signals and speech plus noise signals is softened by being made non-binary. Once the noise spectrum has been estimated, it is used to generate gain factors that multiply the DFT signals to produce noise-reduced DFT signals.

Abstract: A pitch detection method and apparatus capable of realizing high-precision pitch detection even for speech signals in which half-pitch or double-pitch exhibits stronger autocorrelation than the pitch for detection. An input speech signal is judged as to voicedness or unvoicedness and a voiced portion and an unvoiced portion of the input speech signal are encoded by a sinusoidal analytic encoding unit 114 and by a code excitation encoding unit 120, respectively, for producing respective encoded outputs. The sinusoidal analytic encoding unit 114 performs pitch search on the encoded outputs for finding the pitch information from the input speech signal and sets the high-reliability pitch information based on the detected pitch information. The results of pitch detection are determined based on the high-reliability pitch information.

Abstract: A method and an apparatus for determining multiband voicing levels using a frequency moving method in a vocoder are provided. The method for determining the multiband voicing levels using the frequency moving method according to the present invention in the vocoder includes the steps of (a) applying a window to an input voice signal and obtaining a power spectrum from a voice spectrum obtained by Fourier converting a windowed signal, (b) moving the frequency of each subband to an origin after dividing the power spectrum into a predetermined number of subbands, (c) obtaining autocorrelation values of the respective subbands by inverse Fourier converting the power spectrum the frequency of which is moved to the origin, and (d) normalizing the respective autocorrelation values and determining the voicing levels of the subbands from the normalized autocorrelation values.

Abstract: To perform pitch analysis for encoding a speech signal, a speech signal is sampled. The sampled speech signal is spectrally whitened to produce a spectral residual signal. Samples of the spectral residual signal are collected and the collected samples are autocorrelated. Maximum values of the correlated result are determined. Gain values are determined based on at least in part the maximum values of the correlated result. The gain values are quantized using a codebook to produce a codebook index and an associated frame delay. The codebook index and the frame delay represent a pitch of the speech signal to facilitate encoding the speech signal.

Abstract: Method and apparatus for a new interface architecture which reduces the number of software components required to interface a variety of requester types coupled to a server with a variety of communications programs coupled to an on-line transaction processing system. The new interface architecture isolate attributes of the requesters and the communication programs into individual software components so that all software code associated with each requester type is included within a corresponding requester software module, and all software code associated with each communications program is included within a corresponding communications software module. Each new requester type added requires the addition of only one requester software module, and each new communications program added requires the addition of only one communications software module, thus reducing the overall number of software modules required to interface the variety of requester types to the variety of communications programs.

Abstract: A pitch determination apparatus and method using spectro-temporal autocorrelation to prevent pitch determination errors are provided.

Abstract: A method of speech coding a sampled speech signal using long term prediction (LTP). A LTP pitch-lag parameter is determined for each frame of the speech signal by first determining the autocorrelation function for the frame within the signal, between predefined maximum and minimum delays. The autocorrelation function is then weighted to emphasize the function for delays in the neighborhood of the pitch-lag parameter determined for the most recent voiced frame. The maximum value for the weighted autocorrelation function is then found and identified as the pitch-lag parameter for the frame.

Abstract: A sample speech is analyzed by a speech analyzing unit to obtain sample characteristic parameters, and a coding distortion is calculated from the sample characteristic parameters in each of a plurality of coding modules. The sample characteristic parameters and the coding distortions are statistically processed by a statistical processing unit to obtain a coding module selecting rule. Thereafter, when a speech is analyzed by the speech analyzing unit to obtain characteristic parameters, an appropriate coding module is selected by a coding module selecting unit from the coding modules according to the coding module selecting rule on condition that a coding distortion for the characteristic parameters is minimized in the appropriate coding module. Thereafter, the characteristic parameters of the speech are coded in the appropriate coding module, and a coded speech is obtained. When the coded speech is decoded, a reproduced speech is obtained.

Abstract: A radio communication system includes a voice recognition system (221) for converting (400) a caller's voice message to a textual speech message. The textual speech message is then transmitted to an intended selective call radio (122). To perform these functions, the radio communication system includes a caller interface circuit (218), a transmitter (116), and a processor (222). To perform voice-to-text conversion, the processor is adapted to cause the caller interface circuit to sample a voice signal generated by the caller during a plurality of frame intervals, and to apply a Fourier transform thereto, thereby generating spectral data. The spectral data is subdivided into a plurality of bands. The spectral envelope of the spectral data is then filtered out to generate filtered spectral data. A Fourier transform is applied thereto to generate an autocorrelation function for each band.

Abstract: Apparatus is described for digitally encoding an input audio signal for storage or transmission. A distinguishing parameter is measure from the input signal. It is determined from the measured distinguishing parameter whether the input signal contains an audio signal of a first type or a second type. First and second coders are provided for digitally encoding the input signal using first and second coding methods respectively and a switching arrangement directs, at any particular time, the generation of an output signal by encoding the input signal using either the first or second coders according to whether the input signal contains an audio signal of the first type or the second type at that time. A method for adaptively switching between transform audio coder and CELP coder, is presented. In a preferred embodiment, the method makes use of the superior performance of CELP coders for speech signal coding, while enjoying the benefits of transform coder for other audio signals.

Abstract: On the basis of an autocorrelation coefficient calculated by an autocorrelation coefficient computation section from an input speech signal, an LSF computation section computes LSF parameters F(k) (k=1, 2, . . . , N). A modified logarithmic transformation section performs on the LSF parameters a logarithmic transformation with offset defined by f(k)=logC (1+A.times.F(k)) to obtain modified logarithmic LSF parameters f(k). The resulting modified logarithmic LSF parameters are quantized by a quantization section to provide quantized LSF parameters fq(k). Codes representing the quantized LSF parameters fq(k) are outputted. An inverse transformation defined by Fq(k)=(C.sup.fq(k) -1)/A is performed on the LSF parameters fq(k) to output LSF parameters Fq(k) on the general frequency scale.

Abstract: In a speech coding apparatus, an input device inputs a mixed speech signal of a plurality of speakers. A separating device analyzes period characteristics of the input mixed speech signal, and separates the same signal into a plurality of single speech signals each associated with a corresponding one of the speakers, based on a result of the analysis. A first extracting device extracts source speech characteristic parameters included in each of the single speech signals. A second extracting device extracts a generic vocal-tract characteristic parameter from the input mixed speech signal. In a speech decoding apparatus, a first input device inputs the source speech characteristic parameters for each of the speakers. A second input device inputs the vocal-tract characteristic parameter.

Abstract: In a frequently used speech synthesis for voice output an excitation signal is applied to a number of resonators whose frequency and amplitude are adjusted in accordance with the sound to be produced. These parameters for adjusting the resonators may be gained from natural speech signals. Such parameters gained from natural speech signals may also be used for speech recognition, in which these parameter values are compared with comparison values. According to the invention, the parameters, particularly the formant frequencies, are determined by forming the power density spectrum via discrete frequencies from which autocorrelation coefficients are formed for consecutive frequency segments of the power density spectrum from which, in turn, error values are formed, while the sum of the error values is minimized over all segments and the optimum boundary frequencies of the segments are determined for this minimum.

Abstract: A method and apparatus for extracting pitch value information from speech. The method selects at least three highest peaks from a normalized autocorrelation function and produces a plurality of frequency candidates for pitch value determination. The plurality of frequency candidates are used to identify anchor points in pitch values, and is further used to perform both forward and backward searching when an anchor point cannot be readily identified. The running mean or average of determined pitch values is maintained and used in conjunction with the identified valid pitch values in a final determination of the pitch estimation using a weighted least squares fit for identified non-valid frames.

Abstract: A method and apparatus for discriminating between voice and voiceband data (fax/modem data) in an input signal from a voiceband channel, which is available by blocks (packets) of samples. Said discrimination is based upon the computation of two characteristics of the input signal: an autocorrelation function and a power variation function, the combination of which provides a discrimination factor which is highly accurate while requiring a low computing power.

Abstract: In a voice messaging system, as voice memory fills, voice messages are retrieved from memory, re-compressed at a higher compression ratio and re-stored to memory as necessary to maintain a balance between stored voice quality and available memory for storage of new voice messages. The messages may be compressed using ADPCM, CELP, LPC or bit robbing. By re-compressing previously-stored voice messages on an as needed basis, a balance can be maintained between the need to maximize available memory while at the same time maintain the highest voice quality possible for the newest messages. While memory utilization is low, messages can be maintained in memory using a high voice quality, low compression ratio. However, as memory utilization is shrunk due to the storage of more messages, previously stored or other selected voice messages are re-compressed at a higher compression ratio while the newer messages continue to be compressed at the lowest compression ratio.

Abstract: A narrow band code book in which parameters of a time region of a narrow band audio signal obtained from patterns of a plurality of audio signals have previously been stored and a wide band code book in which parameters of a time region of a wide band audio signal obtained from the patterns of a plurality of audio signals have previously been stored in correspondence to the code book of the narrow band, and the input narrow band audio signal is analyzed by the narrow band code book and is synthesized by the wide band code book. In this system, an autocorrelation is used on the parameters of the code books, and a signal obtained by up-sampling an linear predictive code residual is used as an exciting source at the time of audio synthesis.

Abstract: A pitch extraction method and apparatus whereby the pitch of a speech signal having various characteristics can be extracted accurately. The frame-based input speech signal, band-limited by an HPF 12 and an LPF 16, is sent to autocorrelation computing units 13, 17 where autocorrelation data is found. The pitch lag is computed and normalized in the pitch intensity/pitch lag computing units 14, 18. The pitch reliability of the input speech signals, limited by the HPF 12 and the LPF 16, is computed in elevation parameter calculation units. A selection unit 20 selects one of the parameters obtained from the input speech signal, limited by the HPF 12 and the LPF 16, using the pitch lag and the evaluation parameter.

Abstract: A method and apparatus for implementing a vocoder in a application specific integrated circuit (ASIC) is described. The apparatus contains a DSP core that performs computations in accordance with a reduced instruction set (RISC) architecture. The circuit further includes a specifically designed slave processor to the DSP core referred to as the minimization processor. The apparatus further includes a specifically designed block normalization circuitry.

Abstract: A codebook correlation matrix comprises a Toeplitz-type (diagonally symmetric) matrix which is calculated from a forty sample subframe of a speech signal, forming a 40.times.40 matrix. The resulting correlation coefficients which constitute the codes are stored within a DSP's local memory after calculation by dividing the matrix into five predefined x- and y- tracks, each track having a unique set of eight pulse positions. Using the eight pulse positions on each track, fifteen 8.times.8 sub-matrices are created which include all of the correlation coefficients in the original 40.times.40 matrix. The sub-matrices are distributed within a 5.times.5 mapping matrix which is correlated with a structure mapping matrix to determine the configuration of the resulting autocorrelation matrix for storage and searching. The sub-matrices within each column of correlated mapping matrices are searched by directing a multiplex pointer to that particular column.

Abstract: The invention relates to a method of noise suppression, a mobile station and a noise suppressor for suppressing noise in a speech signal. The suppressor comprises means (20, 50) for dividing the speech signal into a first amount of subsignals (X, P), which subsignals represent certain first frequency ranges, and suppression means (30) for suppressing noise in a subsignal (X, P) based upon a determined suppression coefficient (G). The noise suppressor further comprises recombination means (60) for recombining a second amount of subsignals (X, P) into a calculation signal (S), which represents a certain second frequency range, which is wider than the first frequency ranges and determination means (200) for determining a suppression coefficient (G) for the calculation signal (S) based upon the noise contained by it.

Abstract: An input speech signal is encoded as one or more reflection coefficients. To reduce storage requirements, the reflection coefficients are scalar quantized by storing an N-bit code rather than the entire reflection coefficient. An exemplary value for N is 8. A table is provided having 2.sup.N reflection coefficient values. The N-bit code is used to look up reflection coefficient values from the table. To reduce spectral distortion due to scalar quantization, the reflection coefficient values in the table are non-linearly scaled.

Abstract: A speech modification or enhancement filter, and apparatus, system and method using the same. Synthesized speech signals are filtered to generate modified synthesized speech signals. From spectral information represented as a multi-dimensional vector, a filter coefficient is determined so as to ensure that formant characteristics of the modified synthesized speech signals are enhanced in comparison with those of the synthesized speech signal and in accordance with the spectral information. The spectral information can be any one of LSP information, PARCOR information and LAR information. A degree of freedom of design of the speech modification filter used for the aural suppression of quantizing noise contained in the synthesized speech signals is thus heightened leading to the improvement of intelligibility of said synthesized speech signals. A good formant enhancement effect can be obtained without allowing any perceptible level of distortions to occur within a range of permissible spectral gradients.

Abstract: A pitch period extracting apparatus includes a microcomputer which determines a sampling frequency for an A/D converter, and a range of delay times for calculating autocorrelative values on the basis of the sampling frequency. For example, the delay times are set within a range of 20 samples.ltoreq.k.ltoreq.100 samples in a case of 8 kHz, and a range of 15 samples.ltoreq.k.ltoreq.75 samples in a case of 6 kHz. The microcomputer calculates the autocorrelative values of speech signal data stored in a buffer memory, and outputs a delay time at which a maximum autocorrelative value is obtainable as a pitch period of an inputted speech signal.

Abstract: The present invention relates to a method and to equipment for coding and decoding a sampled speech signal. It belongs to systems used in speech processing, in particular for compression of speech information. The method is based upon a time/frequency description and on a representation of the prototype as a fundamental period of a periodic waveform; moreover the excitation of the synthesis filter is carried out through a single, phase-adapted pulse.

Abstract: The periodicity of a signal from a voice channel is determined by sampling the signal, computing the log power spectrum, optionally thresholding and differencing the power spectrum, and then performing an autocorrelation function of limited order to confine the search for periodicity to spans of up to about 400 Hz to 500 Hz.

Abstract: The present invention relates to the method to receive a speech signal, to perform a recognition weighting process on it, to synthesize a synthetic speech signal, to calculate an autocorrelation of the synthetic speech signal whose delay is a predetermined value and an autocorrelation whose delay is 0, to divide the square of the former by the latter, to calculate a pitch lag and a pitch filter coefficient by calculating only the part of a positive peak with skipping over the part of a negative peak by using the results from the dividing operation, and to calculate and output the pitch lag and the pitch filter coefficient by repeating the above process Thus, real-time implementation of CELP vocoder can be achieved.

Abstract: A pitch estimation device and method utilizing a multi-resolution approach to estimate a pitch lag value of input speech. The system includes determining the LPC residual of the speech and sampling the LPC residual. A discrete Fourier transform is applied and the result is squared. A lowpass filtering step is carried out and a DFT on the squared amplitude is then performed to transform the LPC residual samples into another domain. An initial pitch lag can then be found with lower resolution. After getting the low-resolution pitch lag estimate, a refinement algorithm is applied to get a higher-resolution pitch lag. The refinement algorithm is based on minimizing the prediction error in the time domain. The refined pitch lag then can be used directly in the speech coding.

Abstract: A second spectrum parameter of which degree is lower than that of a first spectrum parameter is calculated based on the first spectrum parameter that is output from an encoder. A spectrum postfilter generates a transfer function having a denominator and a numerator wherein said first spectrum parameter is included in said denominator and said second spectrum parameter is included in said numerator, and filters the reduced signal with this transfer function. In addition, it adaptively generates a compensation coefficient based on the first and second parameters. A compensation filter generates a transfer function based the compensation coefficient and filters an output of the spectrum postfilter with this transfer function.

Abstract: A vector quantizer for a speech coder for coding speech signals at low bit rates. The vector quantizer includes an auto-correlation calculation circuit for calculating an impulse response of a weighting function for each sub-interval of an input signal vector. The vector quantizer also includes a weighted cross-correlation calculation circuit for calculating a weighted cross-correlation of the weighted input signal vector and the weighted codevector having a code length equal to that of the input signal vector. The vector quantizer further includes a weighted auto-correlation calculation circuit for calculating an auto-correlation of the weighted codevectors, by using respective auto-correlations of the impulse responses, the codevectors and the cross-correlations.