Spectral Adjustment Patents (Class 381/94.2)
  • Patent number: 6810124
    Abstract: An adaptive resonance canceller system and method for attenuating narrowband noise signals within an input signal, where the narrowband noise signals may vary significantly in frequency. The sensor comprises a notch filter, an error reference and gradient generator, and a complex correlator circuit for attenuating the narrowband noise component of the input signal. The input signal is applied to the notch filter and to the error reference and gradient generator. The latter component generates an error reference signal and an error gradient signal. These two signals are applied to the complex correlator circuit which, in turn, generates a tuning parameter signal which is applied to the notch filter to tune the filter to the center frequency of the noise component of the input signal.
    Type: Grant
    Filed: October 8, 1999
    Date of Patent: October 26, 2004
    Assignee: The Boeing Company
    Inventor: Stanley A White
  • Patent number: 6804359
    Abstract: A signal processor for reducing undesirable signal content reduces the undesirable signal content by exaggerating the undesirable signal content and then using this exaggerated undesirable signal and adaptive filter means to estimate the undesirable content in the signal and then substantially removing it from the signal. The signal processor includes a signal mapping means for exaggerating the undesirable signal content; and an adaptive filter means for reducing the undesirable signal content using the exaggerated undesirable signal content.
    Type: Grant
    Filed: August 3, 1998
    Date of Patent: October 12, 2004
    Assignee: Skyworks Solutions, Inc.
    Inventors: Li Yu, Martin Snelgrove
  • Publication number: 20040170290
    Abstract: A method and apparatus for shaping quantization noise generated when compressing audio data at a low bit rate is disclosed. A predetermined quantization noise threshold allowed during quantization of sampled audio data and quantization noise energy information of a quantized MDCT coefficient are received in all frequency bands of an audio frequency. The quantization noise energy of the quantized MDCT coefficient is attenuated in a predetermined number of frequency bands in which a difference between the predetermined quantization noise threshold and the quantization noise energy of the quantized MDCT coefficient is large.
    Type: Application
    Filed: November 25, 2003
    Publication date: September 2, 2004
    Applicant: SAMSUNG ELECTRONICS CO., LTD.
    Inventors: Tae-gyu Chang, Heung-yeop Jang
  • Publication number: 20040165736
    Abstract: The invention includes a method, apparatus, and computer program to selectively suppress wind noise while preserving narrow-band signals in acoustic data. Sound from one or several microphones is digitized into binary data. A time-frequency transform is applied to the data to produce a series of spectra. The spectra are analyzed to detect the presence of wind noise and narrow band signals. Wind noise is selectively suppressed while preserving the narrow band signals. The narrow band signal is interpolated through the times and frequencies when it is masked by the wind noise. A time series is then synthesized from the signal spectral estimate that can be listened to. This invention overcomes prior art limitations that require more than one microphone and an independent measurement of wind speed. Its application results in good-quality speech from data severely degraded by wind noise.
    Type: Application
    Filed: April 10, 2003
    Publication date: August 26, 2004
    Inventors: Phil Hetherington, Xueman Li, Pierre Zakarauskas
  • Patent number: 6754355
    Abstract: According to one embodiment of the present invention, a digital hearing device is disclosed. The digital hearing aid includes a microphone for receiving sound, which may include an analog signal. The analog signal is converted by a first converter into a digital signal. Filters are provided to divide the digital signal into multiple signal parts. A signal processor may be provided for each signal part, and performs signal processing on its respective signal part. An adder adds the output of the signal processors, which results in a processed digital signal. A second converter converts the processed digital signal back into an analog signal. A speaker then outputs the analog signal. According to another embodiment of the present invention, a method for enhancing sound is provided.
    Type: Grant
    Filed: December 7, 2000
    Date of Patent: June 22, 2004
    Assignee: Texas Instruments Incorporated
    Inventors: Trudy D. Stetzler, Pedro R. Gelabert, Tod D. Wolf
  • Patent number: 6738445
    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.
    Type: Grant
    Filed: November 26, 1999
    Date of Patent: May 18, 2004
    Assignees: IVL Technologies Ltd., Canada Inc.
    Inventor: Gilbert Arthur Joseph Soulodre
  • Publication number: 20040066940
    Abstract: A method and system for inhibiting noise produced by one or more sources of undesired sound from pickup by a speech recognition unit, where a respective transducer is located proximate each source of undesired sound for converting each source of undesired sound to a corresponding electrical signal, and a noise reduction system is coupled to each of the transducers for converting each electrical signal to an equivalent anti-phase electrical signal of equal amplitude. An output of the noise reduction or a signal corresponding thereto is fed to the speech recognition unit so each of the anti-phase electrical signals cancels or reduces a corresponding electrical signal produced by the speech recognition unit upon picking up the undesired sound from the respective source.
    Type: Application
    Filed: October 3, 2002
    Publication date: April 8, 2004
    Applicant: SILENTIUM LTD.
    Inventor: Nehemia Amir
  • Publication number: 20040037439
    Abstract: A first band analyzer divides an acoustic signal received from a sound playback system through an input unit into frequency bands, and generates a first band level. An acoustic signal estimator estimates the band level of the original acoustic signal at the input unit, and generates a second band level for each band. A processor extracts an external noise component which is contained in the acoustic signal using the first band level and the second band level. The external noise can be accurately estimated with less computation than in the related art.
    Type: Application
    Filed: June 3, 2003
    Publication date: February 26, 2004
    Inventors: Tomohiko Ise, Nozomu Saito
  • Publication number: 20040028244
    Abstract: A decoding device (100) is a decoding device that generates frequency spectral data from an inputted encoded audio data stream, and includes: a core decoding unit (102) for decoding the inputted encoded data stream and generating lower frequency spectral data representing an audio signal; and an extended decoding unit (104) for generating, based on the lower frequency spectral data, extended frequency spectral data indicating a harmonic structure, which is same as an extension along the frequency axis of the harmonic structure indicated by the lower frequency spectral data, in a frequency region which is not represented by the encoded data stream.
    Type: Application
    Filed: March 7, 2003
    Publication date: February 12, 2004
    Inventors: Mineo Tsushima, Takeshi Norimatsu, Naoya Tanaka, Kosuke Nishio
  • Publication number: 20040022399
    Abstract: An excursion limiter (10) broadly comprises a voltage controlled filter (40) to suppress an audio signal according to a selected threshold at a selected frequency using a control voltage. The voltage controlled filter (40) preferably produces an inversion signal at the frequency, amplifies the inversion signal according to the control voltage to produce a suppression signal, and combines the audio signal with the suppression signal, thus suppressing the audio signal at the frequency and creating a resultant signal. A control voltage generator (42) preferably generates the control voltage using a frequency compensation filter (58), a full-wave rectifier (60), a precision half-wave rectifier (61) and a non-linear compensator (62). The frequency compensation filter (58) isolates an initial component which is rectified by the full-wave rectifier (60), shifted, and rectified again at the precision rectifier (61) before being essentially flattened in a non-linear manner by the non-linear compensator (62).
    Type: Application
    Filed: August 5, 2002
    Publication date: February 5, 2004
    Inventors: Christopher Combest, John Koval
  • Publication number: 20030236584
    Abstract: A digital audio signal has a sequence of samples. Extreme values in an audio waveform represented by the digital audio signal are detected. The extreme values include a maximum value and a minimum value. An audio frequency represented by the digital audio signal is detected in response to the number of samples between two temporally-adjacent extreme-corresponding samples. A difference between each extreme value and a value of a sample which immediately precedes the present extreme-corresponding sample is calculated. The calculated differences are multiplied by selected one of predetermined coefficients to get corrective values respectively. A decision is made as to whether or not the detected audio frequency is in one selected from predetermined frequency bands. When the detected audio frequency is in the selected frequency band, a corresponding corrective value is added to the maximum value and a corresponding corrective value is subtracted from the minimum value.
    Type: Application
    Filed: May 2, 2003
    Publication date: December 25, 2003
    Inventor: Toshiharu Kuwaoka
  • Publication number: 20030223593
    Abstract: A method of normalizing received digital audio data includes decomposing the digital audio data into a plurality of sub-bands and applying a psycho-acoustic model to the digital audio data to generate a plurality of masking thresholds. The method further includes generating a plurality of transformation adjustment parameters based on the masking thresholds and desired transformation parameters and applying the transformation adjustment parameters to the sub-bands to generate transformed sub-bands.
    Type: Application
    Filed: June 3, 2002
    Publication date: December 4, 2003
    Inventor: Alex A. Lopez-Estrada
  • Publication number: 20030219131
    Abstract: A phase-locked loop (PLL) circuit generates a sine wave signal synchronized with an input signal. The sine wave signal is directly supplied to a first multiplier. The sine wave signal is also supplied to a second multiplier after shifting the phase by 90 degrees by an all-pass filter (APF). The first and second multipliers multiply the corresponding input signals by corresponding predetermined gains. An adder sums the products. Sound corresponding to the sum is produced from a speaker to a sound field. A filter-controlling unit controls the gains for the first and second multipliers, respectively, to minimize an error signal “e” for the output level of a microphone installed at a listening position.
    Type: Application
    Filed: February 11, 2003
    Publication date: November 27, 2003
    Inventor: Masaichi Akiho
  • Patent number: 6639997
    Abstract: An object of the present invention is to provide a simple apparatus for and a simple method of embedding and extracting digital information with little clue to a third party as to embedded digital information with less effort, and the embedded information is securely reconstructed thereby. To embed digital information, a band division portion receives a digital image signal, and then divides the same into ten frequency band signals through discrete wavelet transform so as to compute wavelet coefficients. A mapping portion maps inherent digital information to a pseudo-random number string. An information embedding portion embeds the mapped pseudo-random number string in a string structured by every or some of the computed wavelet coefficients in MRR (signals exclusive of an LL3 signal). A band synthesis portion synthesizes the embedded LL3 digital image signal.
    Type: Grant
    Filed: February 3, 2000
    Date of Patent: October 28, 2003
    Assignee: Matsushita Electric Industrial Co., Ltd.
    Inventors: Takashi Katsura, Hisashi Inoue
  • Publication number: 20030198357
    Abstract: A sound intelligibility enhancement (SIE) system is disclosed. The SIE system uses a psychoacoustic model and preferably an oversampled filterbank wherein the level of a signal-of-interest that falls below the environmental noise is selectively amplified as a function of the input level and frequency so that it is audible above the noise but never exceeds a predetermined maximum output level as a function of frequency. The SIE system can be combined with active noise cancellation.
    Type: Application
    Filed: August 7, 2002
    Publication date: October 23, 2003
    Inventors: Todd Schneider, David Coode, Robert L. Brennan, Peter Olijnyk
  • Publication number: 20030185408
    Abstract: A method and apparatus for de-noising weak bio-signals having a relatively low signal to noise ratio utilizes an iterative process of wavelet de-noising a data set comprised of a new set of frames of wavelet coefficients partially generated through a cyclic shift algorithm. The method preferably operates on a data set having 2N frames, and the iteration is performed N−1 times. The resultant wavelet coefficients are then linearly averaged and an inverse discrete wavelet transform is performed to arrive at the de-noised original signal. The method is preferably carried out in a digital processor.
    Type: Application
    Filed: March 29, 2002
    Publication date: October 2, 2003
    Inventors: Elvir Causevic, Eldar Causevic, Mladen Victor Wickerhauser
  • Publication number: 20030138116
    Abstract: System (10) is disclosed including an acoustic sensor array (20) coupled to processor (42). System (10) processes inputs from array (20) to extract a desired acoustic signal through the suppression of interfering signals. The extraction/suppression is performed by modifying the array (20) inputs in the frequency domain with weights selected to minimize variance of the resulting output signal while maintaining unity gain of signals received in the direction of the desired acoustic signal. System (10) may be utilized in hearing aids, voice input devices, surveillance devices, and other applications.
    Type: Application
    Filed: November 7, 2002
    Publication date: July 24, 2003
    Inventors: Douglas L. Jones, Michael E. Lockwood, Robert C. Bilger, Albert S. Feng, Charissa R. Lansing, William D. O'Brien, Bruce C. Wheeler, Mark Elledge, Chen Liu, Carolyn T. Bilger
  • Patent number: 6594365
    Abstract: A system for identifying a model of an acoustic system in the presence of an external noise signal is disclosed. The system includes an acoustic actuator for generating controlled sound within the acoustic system. A sensor receives the controlled sound and the external noise signal and produces a sensed signal. A control system generates a control signal in response to an error signal. The control system includes a system model for generating an estimated response signal. The control system also generates the error signal representing the difference between the sensed signal and the estimated response signal. A masking threshold generator receives the sensed signal and the error signal and produces spectral shaping parameters. A shaped signal generator for receives the spectral shaping parameters and produces a test signal which is provided as an input to the control system.
    Type: Grant
    Filed: November 18, 1998
    Date of Patent: July 15, 2003
    Assignee: Tenneco Automotive Operating Company Inc.
    Inventor: Graham P. Eatwell
  • Publication number: 20030128851
    Abstract: An amplitude suppression quantity denoting a noise suppression level of a current frame is calculated in an amplitude suppression quantity calculating unit (20), a perceptual weight distributing pattern of both a spectral subtraction quantity and a spectral amplitude suppression quantity is determined in a perceptual weight pattern adjusting unit (21), the spectral subtraction quantity and the spectral amplitude suppression quantity given by the perceptual weight distributing pattern are corrected according to a frequency band SN ratio in a perceptual weight correcting unit (7), a noise subtracted spectrum is calculated from an amplitude spectrum, a noise spectrum and a corrected spectral subtraction quantity in a spectrum subtracting unit (8), and a noise suppressed spectrum is calculated from the noise subtracted spectrum and a corrected spectral amplitude suppression quantity in a spectrum suppressing unit (9).
    Type: Application
    Filed: February 6, 2003
    Publication date: July 10, 2003
    Inventor: Satoru Furuta
  • Publication number: 20030125823
    Abstract: A signal processing method divides a first signal of two signals to be compared in similarity into smaller regions, selects one of the regions, and calculates the correlation of the selected one with the other second signal. The method finds a time difference, an expansion factor, and a similarity in one region in which the maximum similarity as the square of the correlation is obtained, and performs integration in the position represented by the time difference and the expansion factor of values based on similarities. The method performs similar processing on all the regions, and evaluates similarity by, in a peak where the integrated value of similarities is a maximum, compares its magnitude with a threshold value. The region corresponding to the peak can be extracted.
    Type: Application
    Filed: October 18, 2002
    Publication date: July 3, 2003
    Inventors: Mototsugu Abe, Masayuki Nishiguchi
  • Publication number: 20030086575
    Abstract: Disclosed is an apparatus for and a method of filtering noise from a mixed sound signal to obtained a filtered target signal, comprising the steps of inputting (100) the mixed signal through a pair of microphones (10) into a first channel (15a) and a second channel (15b), separately Fourier transforming (110) each said mixed signal into the frequency domain, computing (130) a signal short-time spectral amplitude |Ŝ| from said transformed signals, computing (140) a signal short-time spectral complex exponential ei arg(S) from said transformed signals, where arg(S) is the phase of the target signal in the frequency domain, computing (150) said target signal S in the frequency domain from said spectral amplitude and said complex exponential.
    Type: Application
    Filed: December 5, 2001
    Publication date: May 8, 2003
    Inventors: Radu Victor Balan, Justinian Rosca
  • Publication number: 20030063759
    Abstract: A directional signal processing system for beamforming information signals. The system includes an oversampled filterbank, which has an analysis filterbank for transforming the information signals in time domain into channel signals in transform domain, a synthesis filterbank and a signal processor. The signal processor processes the outputs of the analysis filterbank for beamforming the information signals. The synthesis filterbank transforms the outputs of the signal processor to a single information signal in time domain.
    Type: Application
    Filed: August 7, 2002
    Publication date: April 3, 2003
    Inventors: Robert L. Brennan, Edward Chau, Hamid Sheikhzadeh Nadjar, Todd Schneider
  • Publication number: 20030053640
    Abstract: An input signal is applied to a notch filter having a transfer function that is the inverse of the expected noise signal. The filtered signal is coupled to a first amplifier and the input signal is coupled to a second amplifier. The outputs of the amplifiers are summed. The gains of the amplifiers are oppositely adjusted in response to the magnitude of the input signal. At low amplitude, the filtered signal is amplified more than the unfiltered signal. At high amplitude, the unfiltered signal is amplified more than the filtered signal.
    Type: Application
    Filed: September 14, 2001
    Publication date: March 20, 2003
    Applicant: Fender Musical Instruments Corporation
    Inventors: Dale Vernon Curtis, Charles Clifford Adams
  • Publication number: 20030035553
    Abstract: Perceptual coding of spatial cues (PCSC) is used to convert two or more input audio signals into a combined audio signal that is embedded with two or more sets of one or more auditory scene parameters, where each set of auditory scene parameters (e.g., one or more spatial cues such as an inter-ear level difference (ILD), inter-ear time difference (ITD), and/or head-related transfer function (HRTF)) corresponds to a different frequency band in the combined audio signal. A PCSC-based receiver is able to extract the auditory scene parameters and apply them to the corresponding frequency bands of the combined audio signal to synthesize an auditory scene. The technique used to embed the auditory scene parameters into the combined signal enables a legacy receiver that is unaware of the embedded auditory scene parameters to play back the combined audio signal in a conventional manner, thereby providing backwards compatibility.
    Type: Application
    Filed: November 7, 2001
    Publication date: February 20, 2003
    Inventors: Frank Baumgarte, Jiashu Chen, Christof Faller
  • Publication number: 20020186852
    Abstract: There are provided a method and an apparatus for reducing random, continuous, non-stationary noise in audio signals, the noisy audio signal being filtered by means of a predetermined filter function. The filter function is determined dynamically having regard to the current properties of the noisy audio signal and/or its constituent parts, and the filter function is also limited dynamically having regard to the current properties of the noise component contained in the noisy audio signal.
    Type: Application
    Filed: December 13, 2001
    Publication date: December 12, 2002
    Inventors: Jan Rademacher, Jorg Bitzer
  • Publication number: 20020150265
    Abstract: In an apparatus which estimates characteristics of a surrounding noise only when an input signal is soundless and performs a noise reduction or suppression of the input signal based on the estimated result, a signal noise ratio is estimated from the input signal, and an automatic switch or an automatic adjustment is performed so as to execute a noise reduction only when the signal noise ratio is good, otherwise to avoid the noise reduction or make the noise reduction degree smaller.
    Type: Application
    Filed: March 27, 2002
    Publication date: October 17, 2002
    Inventors: Hitoshi Matsuzawa, Yasushi Yamazaki
  • Publication number: 20020150264
    Abstract: Method for eliminating spurious signal components (SS) from an input signal (ES), said method including the characterization, in a signal analysis phase (I), of the spurious signal components (SS) and of the information signal (NS) contained in the input signal (ES), and the determination or generation, in a signal processing phase (II), of the information signal (NS) or estimated information signal (NS′) on the basis of the characterization obtained in the signal analysis phase (I), said characterization of the signal components (SS, NS) being performed under utilization at least of auditory-based features (M1 to Mj).
    Type: Application
    Filed: April 11, 2001
    Publication date: October 17, 2002
    Inventors: Silvia Allegro, Hans-Ueli Roeck
  • Patent number: 6445801
    Abstract: The disclosed method uses the Wiener frequency filtering to suppress noise in noisy sound signals (u(t)). This method includes a preliminary step in which the sound signals (u(t)) to be noise-suppressed are digitized by sampling and subdivided into frames. The method then includes a first series of steps including the creation of a noise model on N frames, the estimating of the spectral density of the noise and of the energy of the noise model and the computing of a coefficient that reflects the statistical dispersion of the noise. It also includes a second series of steps including the computation of the spectral density of the signals to be noise-suppressed fore each frame. The coefficients of the Wiener filter are modified for each successively processed frame, by the parameters determined at the end of the two series of steps, so as to introduce an energy compensation and an adaptive overestimation of the noise.
    Type: Grant
    Filed: November 20, 1998
    Date of Patent: September 3, 2002
    Assignee: Sextant Avionique
    Inventors: Dominique Pastor, GĂ©rard Reynaud, Pierre-Albert Breton
  • Publication number: 20020118844
    Abstract: A noise or vibration control system reduces a sampling rate and reduces a control rate to improve computation efficiency. The present invention permits the use of a sample frequency (fs) that is less than twice the frequency of interest (fd). The sensed signals are filtered to extract a particular frequency range with a lower bound given by (2n−1)*fs/2 and an upper bound given by (2n+1)*fs/2, where n is an integer chosen so that the frequency of interest (fd) is within the extracted frequency range. The control commands are also calculated at a reduced rate, which is dependent upon the bandwidth of the tone, rather than the absolute frequency of the tone. Rather than updating the control signals directly on the sampled sensor data yk as it enters the computer, the control computations are done on the harmonic components ak and bk, or equivalently on the magnitude and phase.
    Type: Application
    Filed: February 27, 2002
    Publication date: August 29, 2002
    Inventors: William Arthur Welsh, Douglas G. MacMartin, Alan M. Finn
  • Patent number: 6415253
    Abstract: A noise suppression device receives data representative of a noise-corrupted signal which contains a speech signal and a noise signal, divides the received data into data frames, and then passes the data frames through a pre-filter to remove a dc-component and the minimum phase aspect of the noise-corrupted signal. The noise suppression device appends adjacent data frames to eliminate boundary discontinuities, and applies fast Fourier transform to the appended data frames. A voice activity detector of the noise suppression device determines if the noise-corrupted signal contains the speech signal based on components in the time domain and the frequency domain. A smoothed Wiener filter of the noise suppression device filters the data frames in the frequency domain using different sizes of a window based on the existence of the speech signal. Filter coefficients used for Wiener filter are smoothed before filtering.
    Type: Grant
    Filed: February 19, 1999
    Date of Patent: July 2, 2002
    Assignee: Meta-C Corporation
    Inventor: Steven A. Johnson
  • Patent number: 6343268
    Abstract: A system that reconstructs independent signals from degenerate mixtures estimates independent Auto Regressive (AR) processes from their sum. The system includes an identification system and an estimator. A mixture of two signals is inputted into the system and through the identifying and filtering processes, two estimates of the original signals are outputted. The identification system includes an ARMA identifier, a computation of autocovariance coefficients, an initializer and a gradient descent system. The estimator includes filtering.
    Type: Grant
    Filed: December 1, 1998
    Date of Patent: January 29, 2002
    Assignee: Siemens Corporation Research, Inc.
    Inventors: Radu Balan, Alexander Jourjine, Justinian Rosca
  • Publication number: 20020009204
    Abstract: With a view toward properly combining a fundamental echo with a harmonics echo according to the quality of a signal, the ratio between two frequency components of the fundamental echo is determined (704), and the ratio between two frequency components of the harmonics echo is determined (702). Further, a component ratio between a fundamental component and a harmonics component in an echo receive signal is adjusted based on these two ratios (706, 708).
    Type: Application
    Filed: June 6, 2001
    Publication date: January 24, 2002
    Inventor: Shigeru Matsumura
  • Publication number: 20020006207
    Abstract: The invention relates to a method of providing a user with information on the operation of the portable device and to a portable device. In the device, such a tone is produced that, due to a tone feature, can be distinguished from background noise. This feature can be tone frequency, duration, volume or moment of time. The device can analyse background noise automatically, and based on this, it adjusts at least one feature of the tone automatically such that the tone can be distinguished from background noise, and the background noise does not mask out the tone. Alternatively, the user himself can adjust the tone frequency or duration in a desired way so that it would be distinguished from background noise more clearly.
    Type: Application
    Filed: June 26, 2001
    Publication date: January 17, 2002
    Inventors: Juha Matero, Sami Ronkainen
  • Patent number: 6314394
    Abstract: A method of reducing undesired components from a signal that includes a desired component and undesirable components utilizes an autoregressive model technique. An autoregressive module determines a power spectral density approximation of the signal. An error component of the power spectral density approximation includes the desired component. Portions of the error component having frequencies outside of the expected range of the desired component preferably are filtered so that the result is the desired component with the undesired component removed. The invention is useful, for example, for reducing undesirable noise components from sound signals.
    Type: Grant
    Filed: May 27, 1999
    Date of Patent: November 6, 2001
    Assignee: Lear Corporation
    Inventor: Alan M. Finn
  • Publication number: 20010036284
    Abstract: The circuit for the adaptive suppression of noise is a component part of a digital hearing aid, consisting of two microphones (1, 2), two AD-converters (3, 4), two compensating filters (5, 6), two retarding elements (7, 8), two subtractors (9, 10), a processing unit (11), a DA-converter (13), an earphone (15) as well as the two filters (17, 18). The method for the adaptive suppression of noise can be implemented with the indicated circuit. The two microphones (1, 2), dependent on their spatial arrangement or their directional characteristics and dependent on the location of the acoustic signal sources, provide two differing electric signals (d1(t), d2(t)), which are digitalized in the two AD-converters (3, 4) and pre-processed together with the two fixed compensation filters (5, 6). Following subsequently are the two filters (17, 18) arranged symmetrically crosswise in forward direction with the adaptive filter coefficients (w1, w2).
    Type: Application
    Filed: February 1, 2001
    Publication date: November 1, 2001
    Inventor: Remo Leber
  • Publication number: 20010031055
    Abstract: An audio signal processing device comprises signal supply means to supply over more than one input channel and per input channel over separate frequency subbands domain subchannels coded audio signals.
    Type: Application
    Filed: December 20, 2000
    Publication date: October 18, 2001
    Inventors: Ronaldus Maria Aarts, Fransiscus Marinus Jozephus De Bont, Paulus Henricus Antonius Dillen, Augustus Josephus Elizabeth Maria Janssen
  • Patent number: 6278786
    Abstract: An active noise cancellation aircraft headset system. A speaker is mounted within each earcup of a headset for receiving and acoustically transducing a composite noise cancellation signal. A microphone is also mounted within each earcup for transducing acoustic pressure within the earcup to a corresponding analog error signal. An analog filter receives the analog error signal and inverts it to generate an analog broadband noise cancellation signal. The analog error signal is also provided to an analog to digital converter, which receives the analog microphone error signal and converts it to a digital error signal. A DSP takes the digital error signal and, using an adaptive digital feedback filter, generates a digital tonal noise cancellation signal. A digital to analog converter then converts the digital tonal noise cancellation signal to an analog tonal noise cancellation signal so that it can be combined with the analog broadband noise cancellation signal.
    Type: Grant
    Filed: July 29, 1998
    Date of Patent: August 21, 2001
    Assignee: Telex Communications, Inc.
    Inventor: Jason D. McIntosh
  • Patent number: 6222927
    Abstract: A desired acoustic signal is extracted from a noisy environment by generating a signal representative of the desired signal with a processor for a hearing aid device. The processor receives binaural signals from two microphones at different locations. The binaural inputs to the processor are converted from analog to digital format and then submitted to a discrete Fourier transform process to generate discrete spectral signal representations. The spectral signals are delayed by a number of time intervals in a dual delay line to provide a number of intermediate signals, each corresponding to a different position relative to a desired signal source. Location of the noise source is determined and the spectral content of the desired signal is determined from the intermediate signal corresponding to the noise source location. Inverse transformation of the selected intermediate signal followed by digital to analog conversion provides an output signal representative of the desired signal.
    Type: Grant
    Filed: June 19, 1996
    Date of Patent: April 24, 2001
    Assignee: The University of Illinois
    Inventors: Albert S. Feng, Charissa R. Lansing, Chen Liu, William O'Brien, Bruce C. Wheeler
  • Patent number: 6185309
    Abstract: A method and apparatus for separating signals from instantaneous and convolutive mixtures of signals. A plurality of sensors or detectors detect signals generated by a plurality of signal generating sources. The detected signals are processed in time blocks to find a separating filter, which when applied to the detected signals produces output signals that are statistically independent.
    Type: Grant
    Filed: July 11, 1997
    Date of Patent: February 6, 2001
    Assignee: The Regents of the University of California
    Inventor: Hagai Attias
  • Patent number: 6178248
    Abstract: A dual-processing interference cancelling system and method for processing a broadband input in a computationally efficient manner. Dual processing divides the input into higher and lower frequency bands and applies adaptive filter processing to the lower frequency band while applying non-adaptive filter processing to the higher frequency band. Various embodiments are shown including those based on sub-bands, broadband processing with band-limited adaptation, and broadband processing with an external main-channel generator.
    Type: Grant
    Filed: April 14, 1997
    Date of Patent: January 23, 2001
    Assignee: Andrea Electronics Corporation
    Inventor: Joseph Marash