Correlation Function Patents (Class 704/216)
  • Patent number: 6311153
    Abstract: An audio signal compression apparatus for compressively coding an input audio signal comprises a time-to-frequency transformation unit for transforming the input audio signal to a frequency domain signal; a spectrum envelope calculation unit for calculating a spectrum envelope having different resolutions for different frequencies, from the input audio signal, using a weighting function on frequency based on human auditory characteristics; a normalization unit for normalizing the frequency domain signal using the spectrum envelope to obtain a residual signal; a power normalization unit for normalizing the residual signal by the power; an auditory weighting calculation unit for calculating weighting coefficients on frequency, based on the spectrum of the input audio signal and human auditory characteristics; and a multi-stage quantization device having plural stages of vector quantizers connected in series, to which the normalized residual signal is input, and at least one of the vector quantizers quantizing t
    Type: Grant
    Filed: October 2, 1998
    Date of Patent: October 30, 2001
    Assignee: Matsushita Electric Industrial Co., Ltd.
    Inventors: Yoshihisa Nakatoh, Takeshi Norimatsu, Mineo Tsushima, Tomokazu Ishikawa, Mitsuhiko Serikawa, Taro Katayama, Junichi Nakahashi, Yoriko Yagi
  • Publication number: 20010014855
    Abstract: A system that provides measurements of speech distortion that correspond closely to user perceptions of speech distortion is disclosed. The system calculates and analyzes first and second discrete derivatives to detect and determine the incidence of change in the voice waveform that would not have been made by human articulation because natural voice signals change at a limited rate. Statistical analysis is performed of both the first and second discrete derivatives to detect speech distortion by looking at the distribution of the signals. For example the kurtosis of the signals is analyzed as well as the number of times these values exceed a predetermined threshold. Additionally, the number of times the first derivative data is less than a predetermined low value is analyzed to provide a level of speech distortion and clipping of the signal due to lost data packets.
    Type: Application
    Filed: April 24, 2001
    Publication date: August 16, 2001
    Inventor: William C. Hardy
  • Patent number: 6208958
    Abstract: A pitch determination apparatus and method using spectro-temporal autocorrelation to prevent pitch determination errors are provided.
    Type: Grant
    Filed: January 7, 1999
    Date of Patent: March 27, 2001
    Assignee: Samsung Electronics Co., Ltd.
    Inventors: Yong-duk Cho, Moo-Young Kim
  • Patent number: 6173255
    Abstract: Apparatus and methods for compressing an audio signal. An analog to digital converter is used to digitize the audio signal. A linear predictor processes the digitized audio signal to attenuate coherent noise and produce a residual output signal that is representative of the audio signal. An improved synchronized overlap add processor employs a one bit correlator and a smoothly-shaped window compresses the digitized audio signal. The synchronized-overlap-add processing may be used with voice or audio processing systems to change the time scale of the voice (audio) signal without changing the pitch of the processed signal. The synchronized-overlap-add processing may also be used to reduce noise in the processed signal. The present synchronized-overlap-add processing technique makes the computations required very quick, improving the utility of the processing.
    Type: Grant
    Filed: August 18, 1998
    Date of Patent: January 9, 2001
    Assignee: Lockheed Martin Corporation
    Inventors: Dennis L. Wilson, James L. Wayman
  • Patent number: 6167373
    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.
    Type: Grant
    Filed: December 30, 1999
    Date of Patent: December 26, 2000
    Assignee: Matsushita Electric Industrial Co., Ltd.
    Inventor: Toshiyuki Morii
  • Patent number: 6157830
    Abstract: A method and system for measuring the speech quality in a mobile cellular telecommunications network using available radio link parameters is disclosed herein. In a preferred embodiment, the method includes receiving a set of radio link parameters, as defined in a standard or otherwise available, such as the BER, FER, RxLev, handover statistics, soft information, and speech energy. Temporal information is obtained from the radio link parameters to create a set of temporal parameters which can be statistically analyzed, for example, for the maximum and minimum, mean, standard deviation, and autocorrelation values for a time interval. The temporal parameters are combined to yield a set of correlated parameters that are more closely related to the speech quality. An estimator then uses the correlated parameters to calculate an estimate for the speech quality. The method of the present invention takes advantage of temporal information and correlated relationships from the transmitted parameters.
    Type: Grant
    Filed: May 22, 1997
    Date of Patent: December 5, 2000
    Assignee: Telefonaktiebolaget LM Ericsson
    Inventors: Tor Bjorn Minde, Anders Tomas Uvliden, Per Anders Karlsson, Per Gunnar Heikkil.ang.
  • Patent number: 6113653
    Abstract: An open-loop delay contour estimator (204) generates delay information during coding of an information signal. The delay contour is adjusted according to an error minimization criterion on a subframe basis, which allows a more precise estimate of the true delay contour. A delay contour reconstruction block (211) uses the delay information in a decoder in reconstructing the information signal.
    Type: Grant
    Filed: September 11, 1998
    Date of Patent: September 5, 2000
    Assignee: Motorola, Inc.
    Inventors: James P. Ashley, Weimin Peng
  • Patent number: 6058359
    Abstract: Adaptive speech coding includes receiving an original speech signal, performing on the original speech signal a current coding operation, and adapting the current coding operation in response to information used in the current coding operation. Adaptive speech decoding includes receiving coded information, performing a current decoding operation on the coded information, and adapting the current decoding operation in response to information used in the current decoding operation.
    Type: Grant
    Filed: March 4, 1998
    Date of Patent: May 2, 2000
    Assignee: Telefonaktiebolaget L M Ericsson
    Inventors: Roar Hagen, Erik Ekudden
  • Patent number: 6049814
    Abstract: A system solves a problem of a low accuracy in a low-energy frequency area when spectrum feature parameters are extracted with the use of linear analysis of speech or audio signals and a problem of a low accuracy in formant extracting when a spectrum approximation is slanted, and increases the extracting accuracy of spectrum feature parameters with respect to any given frequency band.
    Type: Grant
    Filed: December 29, 1997
    Date of Patent: April 11, 2000
    Assignee: NEC Corporation
    Inventor: Masahiro Serizawa
  • Patent number: 6049766
    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.
    Type: Grant
    Filed: November 7, 1996
    Date of Patent: April 11, 2000
    Assignee: Creative Technology Ltd.
    Inventor: Jean Laroche
  • Patent number: 6041296
    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.
    Type: Grant
    Filed: April 21, 1997
    Date of Patent: March 21, 2000
    Assignee: U.S. Philips Corporation
    Inventors: Lutz Welling, Hermann Ney
  • Patent number: 6035271
    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.
    Type: Grant
    Filed: October 31, 1997
    Date of Patent: March 7, 2000
    Assignee: International Business Machines Corporation
    Inventor: Chengjun Julian Chen
  • Patent number: 6009385
    Abstract: A clipped input speech waveform is divided into a plurality of a series of signals by means of a wavelet transform such as the Daubechies wavelet transform, which are then scaled or otherwise processed to reduce the effects of clipping, prior to reconstruction of the speech waveform using the inverse transform.
    Type: Grant
    Filed: July 9, 1997
    Date of Patent: December 28, 1999
    Assignee: British Telecommunications public limited company
    Inventor: Stephen Summerfield
  • Patent number: 5974375
    Abstract: A noise codebook selects a code most suitable to the characteristics of an input speech vector from an inside quantification table. Furthermore, a codebook renewal circuit determines a correlative value between a noise code selected by the noise codebook and the input speech vector, subsequently calculates a multiplication value for each of noise codes to generate a renewal code by using the multiplication value with respect to the code selected most frequently by the coding processing at the time of voice. Renewal processing is preformed by replacing a desired code of the codebook with the renewal code. Furthermore, the renewal code is sent to a multiplexing circuit together with a renewal flag value to be sent to a decoding device by using the superfluous bit portion of an unvoice frame.
    Type: Grant
    Filed: November 25, 1997
    Date of Patent: October 26, 1999
    Assignee: Oki Electric Industry Co., Ltd.
    Inventors: Hiromi Aoyagi, Xuedong Yang, Atsushi Yokoyama
  • Patent number: 5963895
    Abstract: A transmission system contains a speech coder which utilizes a pitch detector that is arranged to select a characteristic auxiliary signal portion from the signal to be coded in order to improve the quality of the pitch detection. The pitch is found by searching in the speech signal for signal portions that correspond to the characteristics auxiliary signal portion and by calculating the time difference between the respective signal portions.
    Type: Grant
    Filed: May 10, 1996
    Date of Patent: October 5, 1999
    Assignee: U.S. Philips Corporation
    Inventors: Rakesh Taori, Robert J. Sluijter, Eric Kathmann
  • Patent number: 5963896
    Abstract: In a speech coder, an excitation quantizer 360 retrieves the positions of M non-zero amplitude pulses, which together constitute an excitation, by using spectral parameters and with a different gain for each group of the pulses less in number than M.
    Type: Grant
    Filed: August 26, 1997
    Date of Patent: October 5, 1999
    Assignee: NEC Corporation
    Inventor: Kazunori Ozawa
  • Patent number: 5953411
    Abstract: Input/output sample correlation is achieved through the use of first and second correlation tags appended to input and output buffers, respectively. As an output buffer of output samples is prepared, the first correlation tag identifies one of tie output samples in that buffer. As input samples are input and placed into an input buffer, a second correlation tag is appended to the input buffer which identifies the input sample that is input at the time the output sample identified by the first correlation tag is output. Accordingly, a correlation between input and output samples can be used in an echo cancellation operation or the like.
    Type: Grant
    Filed: December 18, 1996
    Date of Patent: September 14, 1999
    Assignee: Intel Corporation
    Inventor: Robert L. Farrell
  • Patent number: 5943645
    Abstract: In methods and apparatus for computing measures of echo of a far end signal in a near end signal, the evolution over time of frequency spectra of the near end and far end signals are compared to compute a measure of the echo. A far end spectrum of the far end signal and a near end spectrum of the near end signal are determined for each of a plurality of successive time intervals. A respective measure of correlation is determined for each of a plurality of spectrum pairs, each spectrum pair comprising a respective near end spectrum and a respective far end spectrum, the far end spectrum corresponding to a time interval which lags a time interval corresponding to the near end spectrum by a respective time lag. The measures of correlation are compared to determine a maximum measure of correlation which can be used as a measure of echo. The echo measure computation technique is particularly suitable for use in echo suppressors for digital cellular telephony systems, but has other applications.
    Type: Grant
    Filed: April 24, 1997
    Date of Patent: August 24, 1999
    Assignee: Northern Telecom Limited
    Inventors: Dominic Ho, Rafi Rabipour, Majid Foodeei
  • Patent number: 5933803
    Abstract: The invention is related digital speech encoding. In a speech codec according to the invention, for modeling a speech signal (301) both prediction parameters (321, 322, 331) modeling a speech signal in a short term and prediction parameters (341, 342, 351) modeling a speech signal in a long term are used. Each prediction parameter (321, 322, 331, 341, 342, 351) is presented using a certain accuracy, in a digital system with a certain number of bits. In speech encoding according to the invention the number of bits used for presenting prediction parameters (321, 322, 331, 341, 342, 351) is adjusted based upon information parameters (321, 322, 331, 341, 342, 351) obtained from a short-term LPC-analysis (32) and from a long-term LTP-analysis (31, 34, 35). The invention is particularly suitable for use at low data transfer speeds, because it offers a speech encoding method of even quality and low average bit rate.
    Type: Grant
    Filed: December 5, 1997
    Date of Patent: August 3, 1999
    Assignee: Nokia Mobile Phones Limited
    Inventor: Pasi Ojala
  • Patent number: 5878387
    Abstract: The coding apparatus comprises an adaptive codebook storing excitation signals as vectors, a synthesis filter for forming a synthesis signal, referring to the vectors stored in the adaptive codebook, a similarity computation circuit for computing a similarity between the synthesis signal obtained by the synthesis filter and a target signal, and a coding scheme determining circuit for deciding one coding scheme from a plurality of coding schemes respectively having coding bit rates different from each other, on the basis of the similarity obtained by the similarity computation circuit.
    Type: Grant
    Filed: September 29, 1995
    Date of Patent: March 2, 1999
    Assignee: Kabushiki Kaisha Toshiba
    Inventors: Masahiro Oshikiri, Kimio Miseki, Masami Akamine, Tadashi Amada
  • Patent number: 5867814
    Abstract: A speech coder, formed with a digital speech encoder and a digital speech decoder, utilizes fast excitation coding to reduce the computation power needed for compressing digital samples of an input speech signal to produce a compressed digital speech datastream that is subsequently decompressed to synthesize digital output speech samples. Much of the fast excitation coding is furnished by an excitation search unit in the encoder. The search unit determines excitation information that defines a non-periodic group of excitation pulses The optimal location of each pulse in the non-periodic pulse group is chosen from a corresponding set of pulse positions stored in the encoder. The search unit ascertains the optimal pulse positions by maximizing the correlation between (a) a target group of filtered versions of digital input speech samples provided to the encoder for compression and (b) a corresponding group of synthesized digital speech samples.
    Type: Grant
    Filed: November 17, 1995
    Date of Patent: February 2, 1999
    Assignee: National Semiconductor Corporation
    Inventor: Mei Yong
  • Patent number: 5864795
    Abstract: An improved vocoder system and method for estimating pitch in a speech waveform. The vocoder receives digital samples of a speech waveform and generates a plurality of parameters based on the speech waveform, including a pitch parameter. The present invention comprises an improved method for estimating and correcting the pitch parameter using correlation techniques. The method comprises first performing a correlation calculation on a frame of the speech waveform, which produces one or more correlation peaks at respective numbers of delay samples. The vocoder then compares the one or more correlation peaks with a clipping threshold value. If a single peak at location P.sub.d is greater than the clipping threshold, then the vocoder performs additional calculations to ensure that this single correlation peak is not a second or higher multiple of the true pitch. In the preferred embodiment, the vocoder assumes the peak at location P.sub.
    Type: Grant
    Filed: February 20, 1996
    Date of Patent: January 26, 1999
    Assignee: Advanced Micro Devices, Inc.
    Inventor: John G. Bartkowiak
  • Patent number: 5854998
    Abstract: An improved speech processing system has a short-term analyzer, a target vector generator and a maximum likelihood, multi-pulse analyzer. The multi-pulse analyzer generates a plurality of sequences of equal amplitude, variable sign, variably spaced pulses. Each of the sequences have a different amplitude value and each of the pulses within each sequence have equal amplitudes but variable signs. The multi-pulse analyzer generates a signal corresponding to the sequence of equal amplitude, variable sign, variably spaced pulses which, according to maximum likelihood criteria, most closely represents the target vector. The maximum likelihood criteria are based on the cross-correlation of the target vector with an impulse response for the pulses in each sequence and on either a covariance matrix or an autocorrelation vector of the impulse response.
    Type: Grant
    Filed: October 18, 1996
    Date of Patent: December 29, 1998
    Assignee: AudioCodes Ltd.
    Inventors: Felix Flomen, Leon Bialik
  • Patent number: 5819209
    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.
    Type: Grant
    Filed: May 23, 1995
    Date of Patent: October 6, 1998
    Assignee: Sanyo Electric Co., Ltd.
    Inventor: Takeo Inoue
  • Patent number: 5797119
    Abstract: In a code excited speech encoder, an input speech signal is segmented into speech samples at first intervals and a spectral parameter is derived from the speech samples that occur at second intervals longer than the first intervals, the spectral parameter representing the characteristic spectral feature. Each speech sample is weighted with the spectral parameter for producing weighted speech samples. The pitch period of the speech signal is determined from the weighted speech samples. A predetermined number of excitation code vectors having smaller amounts of distortion are selected from excitation codebooks as candidate code vectors. The candidate vectors are comb-filtered with a delay time set equal to the pitch period. One of the filtered code vectors having a minimum distortion is selected. The selected filtered code vector is calculated for minimum distortion and, in response thereto, a gain code vector is selected from a gain codebook.
    Type: Grant
    Filed: February 3, 1997
    Date of Patent: August 18, 1998
    Assignee: NEC Corporation
    Inventor: Kazunori Ozawa
  • Patent number: 5774836
    Abstract: An improved vocoder system and method for estimating pitch in a speech waveform which more accurately disregards false pitch estimates resulting from secondary excitations. The vocoder system first performs a correlation calculation on a speech frame and generates an estimated pitch value. The present invention then compares the estimated or determined pitch with a threshold value to determine if the determined or estimated pitch has a suspiciously low pitch value. If so, the present invention performs error checking to disregard pitch estimates that are the result of the First Formant frequency's contribution to the pitch estimation process. The error checking involves examining the higher multiples of the determined pitch value to ascertain whether the determined pitch value might be incorrect. The present invention determines whether one or more higher multiples are missing, whether the higher multiples are related by a common factor, and whether adjacent multiples have missing peaks.
    Type: Grant
    Filed: April 1, 1996
    Date of Patent: June 30, 1998
    Assignee: Advanced Micro Devices, Inc.
    Inventors: John G. Bartkowiak, Mark Ireton
  • Patent number: 5774835
    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.
    Type: Grant
    Filed: August 21, 1995
    Date of Patent: June 30, 1998
    Assignee: NEC Corporation
    Inventor: Kazunori Ozawa
  • Patent number: 5761632
    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.
    Type: Grant
    Filed: May 16, 1997
    Date of Patent: June 2, 1998
    Assignee: NEC Corporation
    Inventor: Masahiro Serizawa