Abstract: Upper and lower limits of predetermined characteristic values of products contained in a plurality of product lots are stored in accordance with the product standard for a target product. An average value of standard deviations in the characteristic values is calculated based on a control chart for the product lots. An average value of the characteristic values is calculated, and an upper limit and a lower limit of an average value of the characteristic values in a 95% confidence interval is calculated. A measurement standard deviation representing a variation in a measuring instrument with regard to the characteristic values is estimated. One of an upper limit and a lower limit of the average value of the characteristic values in the confidence interval is updated as an average value of the characteristic values. A standard deviation in the characteristic values of the product is estimated, and an upper defect rate and a lower defect rate are calculated, so that a yield rate is calculated.

Abstract: Audio recorded by a cellphone or other portable recording device (e.g., audio recorded as part of a video recording of a play or other event) is often of low quality, due to the limitations of the portable recording device. Multiple audio recordings, made during the same period of time and near the same location, can be combined to generate an improved-quality audio recording of an event. The audio recordings may be accessible to a server that selects the audio recordings and performs the combination. To protect the privacy of persons whose audio is used, more than a minimum number of recordings could be combined and/or no more than a threshold amount of any one recording could be used to generate a combined recording. Additionally, a provided ‘clean’ recording could include more than a threshold amount of the audio provided by a user or device that requests such a ‘clean’ recording.

Abstract: A semiconductor device for realizing higher-precision noise elimination includes: a decoder which decodes an encoded input signal; a determining unit which determines whether or not a voice signal is included in the input signal; a suppressor which performs a suppressing process for suppressing a noise component included in the input signal on the basis of a result of determination by the determining unit; and a first storage for storing, as a determination criterion value used for the determination, a first criterion value which specifies the proportion of a voice signal with respect to voice distortion noise.

Abstract: A noise event suppression technique for a monitoring system detects a noise event in a signal waveform when a focal sample in the waveform has an amplitude greater than an amplitude of algorithmically determined earlier and later samples in the waveform that are noncontiguous with the focal sample. When the monitoring system detects the noise event, the monitoring system reduces the amplitude of the focal sample to an amplitude between those of the earlier and later samples. The monitoring system outputs data determined using the waveform once noise events have been adequately suppressed.

Abstract: Provided are methods and systems for noise suppression within multiple time-frequency points of spectral representations. A multi-feature cluster tracker is used to track signal and noise sources and to predict signal versus noise dominance at each time-frequency point. Multiple features, such as binaural and monaural features, may be used for these purposes. A Gaussian mixture model (GMM) is developed and, in some embodiments, dynamically updated for distinguishing signal from noise and performing mask-based noise reduction. Each frequency band may use a different GMM or share a GMM with other frequency bands. A GMM may be combined from two models, with one trained to model time-frequency points in which the target dominates and another trained to model time-frequency points in which the noise dominates. Dynamic updates of a GMM may be performed using an expectation-maximization algorithm in an unsupervised fashion.

Abstract: A signal processing apparatus includes an absolute value unit configured to convert an audio signal into absolute values, a representative value calculation unit configured to calculate representative values of consecutive sample values included in blocks of the audio signal which has been converted into the absolute values using at least maximum sample values among values of the samples included in the blocks for individual blocks, an average value calculation unit configured to determine a section which includes a predetermined number of consecutive blocks as a frame and calculate a maximum value of the representative values of the blocks included in the frame and an average value of the representative values of the blocks included in the frame, and a detector configured to detect click noise in the frame on the basis of a ratio of the maximum value to the average value.

Abstract: Corrupted portions of an audio signal are detected and repaired. An audio signal, including numerous sequential frames, may be received from an audio input device. One or more corrupted frames included in the audio signal may be identified. A frame approximating an uncorrupted frame and corresponding to each corrupted frame may be constructed. Each corrupted frame may be replaced with a corresponding constructed frame to generate a repaired audio signal. The repaired audio signal may be outputted via an audio output device.

Abstract: Systems and methods improve audio signals and include means and methods of reducing stochastic noise in wideband audio signals. Multiple microphones may acquire near and far end audio signals, the audio signals may undergo transformations via a general or specialized digital signal processor.

Abstract: A system, method, and computer program product are provided for cleaning an audio segment. For a given audio segment, an offset amount is calculated where the audio segment is maximally correlated to the audio segment as offset by the offset amount. The audio segment and the audio segment as offset by the offset amount are averaged to produce a cleaned audio segment, which has had noise features reduced while having signal features (such as voiced audio) enhanced.

Abstract: A system and method may be provided to segment and/or classify an audio signal from transformed audio information. Transformed audio information representing a sound may be obtained. The transformed audio information may specify magnitude of a coefficient related to energy amplitude as a function of frequency for the audio signal and time. Features associated with the audio signal may be obtained from the transformed audio information. Individual ones of the features may be associated with a feature score relative to a predetermined speaker model. An aggregate score may be obtained based on the feature scores according to a weighting scheme. The weighting scheme may be associated with a noise and/or SNR estimation. The aggregate score may be used for segmentation to identify portions of the audio signal containing speech of one or more different speakers. For classification, the aggregate score may be used to determine a likely speaker model to identify a source of the sound in the audio signal.

Abstract: A noise estimation control system may limit increases of a stored background noise estimate in response to a detected noise feedback situation. The system receives an input audio signal detected within a space, and a reference audio signal that is transmitted by a speaker as an aural signal into the space. A signal processor processes the input audio signal and the reference audio signal to determine a coherence value based on an amount of the aural signal that is included in the input audio signal. The signal processor also calculates an amount to adjust the stored background noise estimate based on the coherence value and a determined background noise level of the input audio signal.

Abstract: The present technology provides robust, high quality dereverberation of an acoustic signal which can overcome or substantially alleviate the problems associated with the diverse and dynamic nature of the surrounding acoustic environment. The present technology utilizes acoustic signals received from a plurality of microphones to carry out a multi-faceted analysis which accurately identifies reverberation based on the correlation between the acoustic signals. Due to the spatial distance between the microphones and the variation in reflection paths present in the surrounding acoustic environment, the correlation between the acoustic signals can be used to accurately determine whether portions of one or more of the acoustic signals contain desired speech or undesired reverberation. These correlation characteristics are then used to generate signal modifications applied to one or more of the received acoustic signals to preserve speech and reduce reverberation.

Abstract: A signal processing apparatus includes: one or more detection means for detecting movement of a diaphragm of a speaker in correspondence with feedback methods that are different feedback methods; analog-to-digital conversion means for converting one or more detection signals acquired by the detection means into a digital form; feedback signal generating means for generating feedback signals corresponding to the feedback methods using the digital detection signals; synthesis means for combining an audio signal to be output as a driving signal of the speaker with the feedback signals; correction equalizer means for setting an equalizing characteristic to allow a sound reproduced by the speaker to have a target frequency characteristic by changing the digital audio signal; feedback operation setting means for setting feedback methods in which a feedback operation up to combining the audio signal with the feedback signal is performed and the feedback operation is not performed equalizing characteristic changing a

Abstract: A high frequency signal interpolation apparatus provides, with a simple structure, a high-quality digital audio signal through interpolation of high frequency signals missing due to compression. The high frequency signal interpolation apparatus includes a peak value detection and holding circuit configured to detect a peak value of a digital audio signal provided to an input terminal by sampling the digital audio signal and generate a square wave signal by holding the detected peak value; a high-pass filter configured to extract a higher harmonic component from the generated square wave signal; and an adder configured to add the extracted higher harmonic component to the digital audio signal provided to the input terminal.

Abstract: The present invention disclosed an all-digital speaker system device based on multi-bit ?-? Modulation, comprising an A/D converter, an interpolation filter, a ?-? modulator, a dynamic mismatch regulator, a differential buffer and a speaker array. The present invention takes advantage of the ?-? modulation technique to effectively reduce the cost and complexity of the hardware realization of the speaker system device, thus realizing the full digitalization of the whole sound transmission link and the integration of the system device with low power dissipation and volume. Furthermore, the system device according to the present invention is provided with a better control ability on the local sound field, thus providing a better practicable method for the voice secret transmission.

Abstract: Automatic recalculation of tuner filter coefficients are made in order to compensate for changes in signal properties due to processing functionality in the tuner. The architecture compensates for processing changes, such as a large continuous range of clock frequency shifts, while not sacrificing bandwidth response characteristics of the channel filter. Embodiments may calculate coefficients in order to obtain response characteristics while utilizing a completely on-chip architecture, which does not require accessing off-chip software driver programs, and does not require complex look-up tables containing filter coefficients stored in onboard memory.

Abstract: A method of enhancing an audio signal includes the steps of: a) receiving a primary audio input signal, b) receiving a detected audio signal which comprises: A) an echo component derived from play-out of the primary audio input signal and B) a noise component, and c) estimating from the primary audio input signal and the detected audio signal: 1) a set of frequency-specific lower bound gains, such that each frequency-specific lower bound gain, when applied to a respective frequency of the primary audio input signal, would cause the noise component to just mask the echo component at that respective frequency and 2) a set of frequency-specific upper bound gains, such that each frequency-specific upper bound gain, when applied to a respective frequency of the primary audio input signal, would cause the echo component to just mask the noise component at that respective frequency; d) estimating a set of frequency-specific gains in such a way that each frequency-specific gain falls between the respective frequency-

Abstract: A method for use in a digital frequency synthesizer, the method comprising phase to amplitude conversion of an output value of a phase accumulator in said synthesizer, said conversion being carried out as an approximation (y) of a phase value (x) which corresponds to said output amplitude value, the method being characterized in that the approximation comprises a combination of a linear interpolation value and a second order sinusoidal value, the second order sinusoidal value being used as an error term to correct for errors in the linear interpolation value.

Abstract: A listening device includes an input and output transducers and a forward path defined between the transducers. The forward path may include a signal processing unit processing an SPU-input signal originating from the electric input signal in a time-frequency representation having successive time frames each having a frequency spectrum of the signal in the time frame in question. The signal processing unit includes a spectral content modification unit to modify values of the signal of one or more regions of the frequency spectrum of a given time frame so that the modified values are less correlated to the corresponding time-frequency regions of the input signal. A feedback path estimation unit is uses he improved processed output signal in the feedback estimation. The spectral content modification unit may base the modification on a model of the human auditory system so that the modifications are not perceptible by the user.

Abstract: A signal processing apparatus includes: clip detector means that detects presence/absence of a clipped part with a deformed waveform in each of N audio signals output from N microphones (where N is an integer equal to or greater than 2) based on a dynamic range of a circuit; and interpolation means that treats an audio signal in the N audio signals which has the clipped part detected by the clip detector means as an interpolation target, and other audio signals as non-interpolation targets, and interpolates the waveform of the clipped part of the interpolation target using the waveform of at least one audio signal in the non-interpolation targets.

Abstract: An active noise reduction system is provided for receiving an audio input signal and a noise interference signal and calculating an audio broadcasting signal according to a Feedback Filtered-X Least-Mean-Square (FFXLMS) algorithm, wherein the FFXLMS algorithm optimizes a (convergence factor) ? so as to decrease the numbers of divisions operated by the active noise reduction system and increase the operation speed of the active noise reduction system.

Abstract: A noise reducing apparatus includes: a voice signal inputting unit inputting an input voice signal; a noise occurrence period detecting unit detecting a noise occurrence period; a noise removing unit removing a noise for the noise occurrence period; a generation source signal acquiring unit acquiring a generation source signal with a time duration corresponding to a time duration corresponding to the noise occurrence period; a pitch calculating unit calculating a pitch of an input voice signal interval; an interval signal setting unit setting interval signals divided in each unit period interval; an interpolation signal generating unit generating an interpolation signal with the time duration corresponding to the noise occurrence period and alternately arranging the interval signal in a forward time direction and the interval signal in a backward time direction; and a combining unit combining the interpolation signal and the input voice signal, from which the noise is removed.

Abstract: This specification describes technologies relating to editing audio data. In general, one aspect of the subject matter described in this specification can be embodied in methods that include receiving an audio signal including digital audio data; receiving an input identifying particular audio data of the audio signal corresponding to a noise pulse; and replacing the audio data corresponding to the detected noise pulse using interpolation of adjacent audio data to generate an edited audio signal. Other embodiments of this aspect include corresponding systems, apparatus, and computer program products.

Abstract: A system and method may be configured to reconstruct an audio signal from transformed audio information. The audio signal may be resynthesized based on individual harmonics and corresponding pitches determined from the transformed audio information. Noise may be subtracted from the transformed audio information by interpolating across peak points and across trough points of harmonic pitch paths through the transformed audio information, and subtracting values associated with the trough point interpolations from values associated with the peak point interpolations. Noise between harmonics of the sound may be suppressed in the transformed audio information by centering functions at individual harmonics in the transformed audio information, the functions serving to suppress noise between the harmonics.

Abstract: An audio testing system is configured for receiving an audio signal from the an audio emitting device. The system samples the audio signal and obtains sampling points from the audio signal for determining if the audio signal has been distorted. A related method is also disclosed.

Abstract: An apparatus for providing real-time calibration for two or more microphones. A calibrator for receiving a left microphone signal and a right microphone signal and generating phase difference data. A phase and amplitude correction system for receiving one of the left microphone signal or the right microphone signal the phase difference data and generating calibration data for a beamformer. The beamformer receiving the calibration data, the left microphone signal and the right microphone signal and generating a monaural beamformed signal.

Abstract: A first sound volume calculation unit (251) obtains a length of a straight line connecting a sound emitting object and a sound detection object, and calculates a first sound volume attenuated from a predetermined reference sound volume in accordance with the length. A second volume calculation unit (252), in a case where on the straight line there is an other object that is an obstacle, calculates a second sound volume attenuated from the first sound volume by a predetermined ratio.

Abstract: A pop noise suppression apparatus for eliminating popping noise generated upon initiation or shutdown of an audio output circuit comprises a switch component and a control circuit. The switch component allows the audio output circuit to provide audio through the output of the audio output circuit. The control circuit provides a mute signal for a first period of time in to response initiation or shutdown of the audio circuit. The control circuit comprises a capacitor to be charged upon initiation of the audio output circuit or to be discharged upon shutdown of the audio output circuit. A length of the first period of time during which the mute signal is provided depends on a second period of time to charge or discharge the capacitor.

Abstract: A method and an apparatus for designing a sound compensation filter of a portable terminal are provided. The method includes synchronizing a signal input through a microphone of the system and a test signal, estimating a loss interval of the synchronized signal, compensating for a frame signal delayed by a signal loss in a time axis when the signal loss of the estimated loss interval is greater than a threshold and restoring the loss interval of the signal.

Abstract: The present invention disclosed an all-digital speaker system device based on multi-bit ?-? Modulation, comprising an A/D converter, an interpolation filter, a ?-? modulator, a dynamic mismatch regulator, a differential buffer and a speaker array. The present invention takes advantage of the ?-? modulation technique to effectively reduce the cost and complexity of the hardware realization of the speaker system device, thus realizing the full digitalization of the whole sound transmission link and the integration of the system device with low power dissipation and volume. Furthermore, the system device according to the present invention is provided with a better control ability on the local sound field, thus providing a better practicable method for the voice secret transmission.

Abstract: A speaker array and microphone arrays positioned on both sides of the speaker array are provided. A plurality of focal points each serving as a position of a talker are set in front of the microphone arrays respectively symmetrically with respect to a centerline of the speaker array, and a bundle of sound collecting beams is output toward the focal points. Difference values between sound collecting beams directed toward the focal points that are symmetrical with respect to the centerline are calculated to cancel sound components that detour from the speaker array to microphones. Then, it is estimated based on totals of squares of peak values of the difference values for a particular time period that the position of the talker is close to which one of the focal points, and the position of the talker is decided by comparing the totals of the squares of the peak values of the sound collecting beams directed to the focal points that are symmetrical mutually.

Abstract: An audio signal interpolation apparatus is configured to perform interpolation processing on the basis of audio signals preceding and/or following a predetermined segment on a time axis so as to obtain an audio signal corresponding to the predetermined segment. The audio signal interpolation apparatus includes a waveform formation unit configured to form a waveform for the predetermined segment on the basis of time-domain samples of the preceding and/or the following audio signals and a power control unit configured to control power of the waveform for the predetermined segment formed by the waveform formation unit using a non-linear model selected on the basis of the preceding audio signal when the power of the preceding audio signal is larger than that of the following audio signal, or the following audio signal when the power of the preceding audio signal is smaller than that of the following audio signal.

Abstract: A feedback calibration system and a method for controlling an electronic signal are disclosed. The feedback calibration system includes an input controller adapted to modify an input signal in response to a control signal and generate a modified input signal, a signal processing block including a signal analyzer, wherein the signal processing block is adapted to process the modified input signal to generate an output signal and the signal analyzer is adapted to detect an undesirable condition of the output signal and transmit a detection signal corresponding to the undesirable condition, a transfer function estimator adapted to model and transmit a transfer function estimate of the signal processing block in real-time in response to the detection signal, and a programmable device adapted to transmit the control signal to the input controller for modifying the input signal, wherein the control signal is based upon the transfer function estimate.

Abstract: An apparatus for processing an audio signal and method thereof applied to an audio playback system are disclosed. The apparatus comprises a decoder, an error-correcting circuit and an audio correcting module. The method for processing audio signals in accordance with the present invention decodes the audio signal to generate a decoded signal by the decoder. Then, the error-correcting circuit performs an error-correcting algorithm in the decoded signal to generate an error indication signal and an output audio signal. And the audio correcting module corrects the output audio signal to generate a corrected audio signal when the error indication signal indicates that the output audio signal has error.

Abstract: Disclosed herein are detectors of audio ringing feedback, that is decaying feedback with a gain of less than one, those detectors utilizing a repeated gain measurement that applied to a range of gain values characteristic of ringing-type feedback. Those gain measurements, while in the range, increase a probability measurement of feedback. When the probability of feedback reaches a threshold, a detection of feedback is made and feedback countermeasures, such as the application of a notch filter, may be applied. Optionally, the audio gain around likely frequencies of feedback may be enhanced for a time to increase the resolution of identification of a feedback frequency, which may be identified through an interpolative method. Repeated gain measurements may also identify building-type feedback. A ringing detector may include more than one range of detection, for example for building, strong-ringing and weak-ringing feedback.

Abstract: Circuits and methods for providing zero-gap playback of consecutive data streams in portable electronic devices, such as media players, are described. In some embodiments, a circuit includes a decoder circuit configured to receive encoded audio data and to output decoded audio data including data streams associated with a data file and a subsequent data file. Moreover, a predictive circuit, which is electrically coupled to the decoder circuit, is configured to selectively generate additional samples based on samples in the data file, where the additional samples correspond to times after the end of a data stream associated with the data file. Additionally, a filter circuit, which is electrically coupled to the decoder circuit and selectively electrically coupled to the predictive circuit, is configured to selectively combine or blend samples at a beginning of the subsequent data file with the additional samples. Note that the circuit may be included in an integrated circuit.

Abstract: In acoustic systems, especially with hearing aids, feedback whistling keeps occurring. To avoid this, a limit gain frequency response of the amplification device, which represents the limits of feedback whistling, is thus recorded. On the basis of the curve recorded a required gain frequency response with a number of interpolation points is created, with each interpolation point having a predetermined minimum distance in each case to the limit gain frequency response in at least two different directions. This enables feedback whistling to be largely avoided, even with shifts in resonant frequencies.

Abstract: A signal processing apparatus includes an absolute value unit configured to convert an audio signal into absolute values, a representative value calculation unit configured to calculate representative values of consecutive sample values included in blocks of the audio signal which has been converted into the absolute values using at least maximum sample values among values of the samples included in the blocks for individual blocks, an average value calculation unit configured to determine a section which includes a predetermined number of consecutive blocks as a frame and calculate a maximum value of the representative values of the blocks included in the frame and an average value of the representative values of the blocks included in the frame, and a detector configured to detect click noise in the frame on the basis of a ratio of the maximum value to the average value.

Abstract: A sampled digital audio signal is displayed on a spectrogram, in terms of frequency vs. time. An unwanted noise in the signal is visible in the spectrogram and the portion of the signal containing the unwanted noise can be selected using time and frequency constraints. An estimate for the signal within the selected portion is then interpolated on the basis of desired portions of the signal outside the time constraints defining the selected portion.

Abstract: Disclosed herein are detectors of audio ringing feedback, that is decaying feedback with a gain of less than one, those detectors utilizing a repeated gain measurement that applied to a range of gain values characteristic of ringing-type feedback. Those gain measurements, while in the range, increase a probability measurement of feedback. When the probability of feedback reaches a threshold, a detection of feedback is made and feedback countermeasures, such as the application of a notch filter, may be applied. Optionally, the audio gain around likely frequencies of feedback may be enhanced for a time to increase the resolution of identification of a feedback frequency, which may be identified through an interpolative method. Repeated gain measurements may also identify building-type feedback. A ringing detector may include more than one range of detection, for example for building, strong-ringing and weak-ringing feedback.

Abstract: Methods and systems for sample rate conversion convert a sampled signal to a higher data rate signal. Conversion pulses are received, having a conversion rate that is higher than the sample rate of the sampled signal. Sample points are then reconstructed from the sampled signal, in real time, on either side of a conversion pulse. An interpolation is performed between the reconstructed sample points, at the time of the conversion pulse. The interpolation results are outputted in real time. The process is repeated for additional conversion pulses. The outputted interpolated amplitudes form the higher data rate signal having a data rate equal to the conversion rate. Sample rate conversion is thus performed in real time according to the higher data rate clock, rather than with fixed ratios. As a result, when the higher data rate clock is affected by, for example, jitter or other frequency variations, the higher data rate samples immediately track the lower data rate samples.

Abstract: A sampled digital audio signal is displayed on a spectrogram, in terms of frequency vs. time. An unwanted noise in the signal is visible in the spectrogram and the portion of the signal containing the unwanted noise can be selected using time and frequency constraints. An estimate for the signal within the selected portion is then interpolated on the basis of desired portions of the signal outside the time constraints defining the selected portion. The interpolated estimate can then be used to attenuate or remove the unwanted sound.

Abstract: Corrupted portions of an audio signal are detected and repaired. An audio signal may be received from an audio input device. The audio signal may include numerous sequential frames. One or more corrupted frames included in the audio signal may be identified. A frame approximating an uncorrupted frame and corresponding to each corrupted frame may be constructed. Each corrupted frame may be replaced with a corresponding constructed frame to generate a repaired audio signal. The repaired audio signal may be outputted via an audio output device.

Abstract: A noise reducing apparatus includes: a voice signal inputting unit inputting an input voice signal; a noise occurrence period detecting unit detecting a noise occurrence period; a noise removing unit removing a noise for the noise occurrence period; a generation source signal acquiring unit acquiring a generation source signal with a time duration corresponding to a time duration corresponding to the noise occurrence period; a pitch calculating unit calculating a pitch of an input voice signal interval; an interval signal setting unit setting interval signals divided in each unit period interval; an interpolation signal generating unit generating an interpolation signal with the time duration corresponding to the noise occurrence period and alternately arranging the interval signal in a forward time direction and the interval signal in a backward time direction; and a combining unit combining the interpolation signal and the input voice signal, from which the noise is removed.

Abstract: A signal processing apparatus includes: clip detector means that detects presence/absence of a clipped part with a deformed waveform in each of N audio signals output from N microphones (where N is an integer equal to or greater than 2) based on a dynamic range of a circuit; and interpolation means that treats an audio signal in the N audio signals which has the clipped part detected by the clip detector means as an interpolation target, and other audio signals as non-interpolation targets, and interpolates the waveform of the clipped part of the interpolation target using the waveform of at least one audio signal in the non-interpolation targets.

Abstract: Methods, systems, and apparatus, including computer program products, for restoring audio signals. A data sequence of samples representing an audio signal is received. Multiple filter coefficients are defined for a filter, and a current sample in the data sequence is selected to be processed. The filter coefficients are updated based on a previous sample preceding the current sample in the data sequence and a filtered value determined by the filter for the previous sample. A filtered value for the current sample is determined using the filter with the updated filter coefficients. The filtered value of the current sample is used to determine whether the current sample has been corrupted by impulsive noise, for example, a crackle.

Abstract: The invention relates to a method for processing a digital input signal (x(i)) in a digital domain, comprising: -sampling a wideband of input frequencies of said digital input signal (x(i)) with a sampling frequency (fs), which decimates with a decimation factor (D), -linear shaping said sampled input frequencies with a configurable delay, -producing an output signal (y(i)) containing said linear shaped input frequencies, wherein the output signal (y(i)) has the same sampling frequency (fs) as said input signal (x(i)).

Abstract: A noise eliminating circuit is disclosed which comprises a noise elimination processing unit that interpolates a generation period of pulse noise overlapped with a received signal depending on a first detection signal acquired by level detection of an intermediate frequency signal of the received signal, the first detection signal indicating the generation of the pulse noise, wherein the noise eliminating circuit comprises: a predicting unit that predicts a value of the intermediate frequency signal at a predetermined clock time based on an intermediate frequency signal generated a predetermined time earlier than the intermediate frequency signal; a detecting unit that compares a difference between the value of the predicted intermediate frequency signal and the value of the generated intermediate frequency signal, at the predetermined clock time, with a predetermined threshold, to output a second detection signal indicating the generation of the pulse noise; and a noise elimination controlling unit that sele

Abstract: An adaptive noise reduction method and apparatus capable of reducing efficiently variable period noise from a main input is provided. The pitch of a noise waveform to be reduced can be variable with a change in a period of motor noise occurring when the revolution period is changed by disc motor control of DVD-RAM, by revolution speed control of other motors, and revolution period change on starting a motor and the like. Therefore, the renewal of adaptive filter coefficient becomes almost unnecessary, thus allowing noise reduction to be performed without degrading a noise canceling effect.

Abstract: The present invention is characterized in performing noise suppression immediately before or after mixing signals received from a plurality of terminals. Thus, in multi-point connection for a plurality of terminal devices, a mixed signal can be supplied with high sound quality to a receiver terminal, regardless of the presence and performance of the noise suppression function in a transmitter terminal.