Abstract: The invention relates to audio signal processing. More specifically, the invention relates to enhancing multichannel audio, such as television audio, by applying a gain to the audio that has been smoothed between portions of the audio. The invention relates to methods, apparatus for performing such methods, and to software stored on a computer-readable medium for causing a computer to perform such methods.

Abstract: Multi-channel noise suppression systems and methods are described that omit the traditional delay-and-sum fixed beamformer in devices that include a primary speech microphone and at least one noise reference microphone with the desired speech being in the near-field of the device. The multi-channel noise suppression systems and methods use a blocking matrix (BM) to remove desired speech in the input speech signal received by the noise reference microphone to get a “cleaner” background noise component. Then, an adaptive noise canceler (ANC) is used to remove the background noise in the input speech signal received by the primary speech microphone based on the “cleaner” background noise component to achieve noise suppression. The filters implemented by the BM and ANC are derived using closed-form solutions that require calculation of time-varying statistics of complex frequency domain signals in the noise suppression system.

Abstract: In the field of audio encoding/decoding technologies, a signal de-noising method is provided. The method includes: selecting, according to a degree of inter-frame correlation of a frame where a spectral coefficient to be adjusted resides, at least two spectral coefficients having high correlation with the spectral coefficient to be adjusted; performing weighting on the at least two selected spectral coefficients and the spectral coefficient to be adjusted to acquire a predicted value of the spectral coefficient to be adjusted; and adjusting a spectrum of a decoded signal by using the acquired predicted value, and outputting the adjusted decoded signal. A signal de-noising apparatus corresponding to the signal de-noising method and an audio decoding system using the signal de-noising apparatus are also provided.

Abstract: Systems and methods to automatically equalize coloration in speech recordings is provided. In example embodiments, a reference spectral shape based on a reference signal is determined. An estimated spectral shape for an input signal is derived. Using the estimated spectral shape and the reference spectral shape a comparison is performed to determine gain settings. The gain settings comprise a gain value for each filter of a filter system. Using gain values associated with the gain setting, automatic equalization is performed on the input signal.

Abstract: Disclosed is the system and method to remove noises in voice signals in a voice communication. The at least one embodiment of the present disclosure performs a spectral subtraction (SS) for voice signals based on a gain function by a spectral subtraction apparatus, performs clustering of voice signals consecutive on a frequency axis of a spectrogram for the voice signals in which the spectral subtraction has been already performed to designate one or more clusters, and extracts musical noises by determining continuity of each of the designated clusters on the frequency axis and a time axis of the spectrogram to extract musical noises.

Abstract: A non-spatial speech detection system includes a plurality of microphones whose output is supplied to a fixed beamformer. An adaptive beamformer is used for receiving the output of the plurality of microphones and one or more processors are used for processing an output from the fixed beamformer and identifying speech from noise though the use of an algorithm utilizing a covariance matrix.

Abstract: Systems (1600) and methods (1500) for frame synchronization. The methods involve: extracting bit sequences S0 and S1 from a Bit Stream (“BS”) of a Data Burst (“DB”); decoding S0 and S1 to obtain decoded bit sequences S?0 and S?1; using S?0 and S?1 to determine Bit Error Rate (“BER”) estimates (516, 518); combining the BER estimates to obtain a combined BER estimate; modifying S0 and S1 so that each includes at least one bit of BS which is not included in its current set of bits and so that it is absent of at least one of the bits in the current set of bits; iteratively repeating the decoding, using, combining and modifying steps to obtain more combined BER estimates; analyzing the combined BER estimates to identify a minimum combined BER estimate; and using the minimum combined BER estimate to determine a location of a vocoder voice frame within DB.

Abstract: A voice correction device includes a detector that detects a response from a user, a calculator that calculates an acoustic characteristic amount of an input voice signal, an analyzer that outputs an acoustic characteristic amount of a predetermined amount when having acquired a response signal due to the response from the detector, a storage unit that stores the acoustic characteristic amount output by the analyzer, a controller that calculates an correction amount of the voice signal on the basis of a result of a comparison between the acoustic characteristic amount calculated by the calculator and the acoustic characteristic amount stored in the storage unit, and a correction unit that corrects the voice signal on the basis of the correction amount calculated by the controller.

Abstract: A method and apparatus for transcoding audio data. The method includes determining if AAC joint stereo exists, running a reference AC-3 rematrixing when the AAC joint stereo does not exist, when AAC joint stereo does exist, enabling rematrixing when the number of corresponding AAC bands is greater than half the size of the band, otherwise, running reference AC-3 rematrixing.

Abstract: An electrical apparatus a voice signal receiving method thereof are disclosed. The electrical apparatus includes a plurality of voice receivers, a voice activity detector, a voice channel switch and a noise eliminator. The voice receivers are used to receive the voice signals. The voice activity detector receives and detects the voice signals, and obtains a main voice signal from the voice signals. The voice channel switch transports the main voice signal to a voice transporting channel and transports a plurality of other voice signals of the voice signals other than the main voice signal to a noise transporting channel according to a detecting result of the voice activity detector. The noise eliminator reduces the noise in the main voice according to the voice signals from the noise transporting channel.

Abstract: A method and an apparatus for processing an audio signal in a mobile terminal, in which an audio signal that is received from a counterpart mobile terminal is classified into a voice signal and a noise signal according to respective energy. A frequency of the classified voice signal and an energy of the classified noise signal is controlled according to a predetermined criteria, then the controlled voice signal and the controlled noise signal are coupled and output to a speaker.

Abstract: By monitoring the wind noise in a location in which a cellular telephone is operating and by applying noise reduction and/or cancellation protocols at the appropriate time via analog and/or digital signal processing, it is possible to significantly reduce wind noise entering into a communication system.

Abstract: A method for noise characteristic dependent speech enhancement by an electronic device is described. The method includes determining a noise characteristic of input audio. Determining a noise characteristic of input audio includes determining whether noise is stationary noise and determining whether the noise is music noise. The method also includes determining a noise reference based on the noise characteristic. Determining the noise reference includes excluding a spatial noise reference from the noise reference when the noise is stationary noise and including the spatial noise reference in the noise reference when the noise is not music noise and is not stationary noise. The method further includes performing noise suppression based on the noise characteristic.

Abstract: Methods and apparatus for signal processing are disclosed. Source separation can be performed to extract source signals from mixtures of source signals by way of independent component analysis. Source separation described herein involves mixed multivariate probability density functions that are mixtures of component density functions having different parameters corresponding to frequency components of different sources, different time segments, or some combination thereof.

Abstract: The present technology provides techniques for transform domain reconstruction of noise-corrupted portions of an acoustic signal to emulate speech which is obscured by the noise. Replacement transform values for the noise-corrupted portions are determined utilizing the portions of the acoustic signal which contain speech.

Abstract: A method and system is disclosed for control of discontinuous transmission based on vocoder and voice activity. An access terminal (AT) may engage in a communication session via an encoder-decoder in a network device in a wireless network. During silence intervals of the communication session, when the AT has no data to transmit, the AT may transmit periodic silence frames at a silence-frame rate to the encoder-decoder. The silence frames may contain parameters for generation of audio noise by the network device. Upon determining that the encoder-decoder has ceased transmitting data to the AT in response to a prolonged absence of transmissions from the AT, the AT may increase the silence-frame rate so as to reduce the duration of the absence of transmissions from the AT, and correspondingly cause the encoder-decoder to begin transmitting audio data to the AT.

Abstract: A method for decoding an audio signal having a bandwidth that extends beyond a bandwidth of a CELP excitation signal in an audio decoder including a CELP-based decoder element. The method includes obtaining a second excitation signal having an audio bandwidth extending beyond the audio bandwidth of the CELP excitation signal, obtaining a set of signals by filtering the second excitation signal with a set of bandpass filters, scaling the set of signals using a set of energy-based parameters, and obtaining a composite output signal by combining the scaled set of signals with a signal based on the audio signal decoded by the CELP-based decoder element.

Abstract: Disclosed in the present invention is a method for cancelling echo in joint time domain and frequency domain.

Abstract: An electronic device for determining a set of pitch cycle energy parameters is described. The electronic device includes a processor and executable instructions stored in memory. The electronic device obtains a frame, a set of filter coefficients and a residual signal based on the frame and the set of filter coefficients. The electronic device determines a set of peak locations based on the residual signal and segments the residual signal such that each segment includes one peak. The electronic device determines a first set of pitch cycle energy parameters based on a frame region between two consecutive peak locations and maps regions between peaks in the residual signal to regions between peaks in a synthesized excitation signal to produce a mapping. The electronic device determines a second set of pitch cycle energy parameters based on the first set of pitch cycle energy parameters and the mapping.

Abstract: An original loudness level of an audio signal is maintained for a mobile device while maintaining sound quality as good as possible and protecting the loudspeaker used in the mobile device. The loudness of an audio (e.g., speech) signal may be maximized while controlling the excursion of the diaphragm of the loudspeaker (in a mobile device) to stay within the allowed range. In an implementation, the peak excursion is predicted (e.g., estimated) using the input signal and an excursion transfer function. The signal may then be modified to limit the excursion and to maximize loudness.

Abstract: In an apparatus and method for isolating a multi-channel sound source, the probability of speaker presence calculated when noise of a sound source signal separated by GSS is estimated is used to calculate a gain. Thus, it is not necessary to additionally calculate the probability of speaker presence when calculating the gain, the speaker's voice signal can be easily and quickly separated from peripheral noise and reverb and distortion are minimized. As such, if several interference sound sources, each of which has directivity, and speakers are simultaneously present in a room with high reverb, a plurality of sound sources generated from several microphones can be separated from one another with low sound quality distortion, and the reverb can also be removed.

Abstract: In a method and system for controlling voice communication of a first person with at least a second person via a communication network a first microphone receives and converts vocal utterances from the first person to a voice signal. A first processor generates a transmission signal by processing the voice signal. A transmitter sends the transmission signal to a receiver. The receiver generates a listening signal by processing the received signal and transmits the listening signal to a speaker. The speaker converts the listening signal to an acoustic signal to be perceived by the first person. In this method a second processor generates the listening signal from the received signal by branching the voice signal and adding the branched voice signal to the received signal. The branched voice signal may be subjected to variable attenuation and/or amplification before being added to the branched voice signal to the received signal.

Abstract: A method and apparatus are provided for visualizing the latency in a conversation between a local speaker and at least one remote speaker separated from the local speaker by a communication medium. A latency estimate is obtained. A timing indication of at least the end of a conversational turn by the local speaker is obtained, and an outbound graphic is displayed, indicating the progress of at least the end-of-turn across the communication medium toward the remote speaker. The outbound graphical indication is displayed with a transit time across the medium that is derived from the latency estimate. An inbound graphic is displayed, indicating the progress across the communication medium toward the local speaker, of a start of a conversational turn by the remote speaker, which is imputed to begin when the remote speaker receives the local speaker's end-of-turn. The inbound graphical indication is displayed with a transit time across the medium that is derived from the latency estimate.

Abstract: A method for the automatic removal of speech contamination from an acoustic noise signal. The method includes the steps of: (a) receiving an input acoustic noise signal; (b) automatically detecting speech contamination in the received acoustic noise signal using a VAD; (c) automatically identifying uncontaminated segments of the received acoustic noise signal based upon a decision value output by the VAD; (d) automatically assembling a congruous uncontaminated acoustic noise signal from the identified uncontaminated segments of the received acoustic noise signal; and (e) outputting the congruous uncontaminated acoustic noise signal. Also, systems implementing such a method.

Abstract: A method of estimating noise in data containing voice information and noise includes receiving the data as a sequence of input values; transforming the data by applying a first non linear mapping to the input values wherein the derivative function of the mapping decreases in magnitude as the input values increase in magnitude smoothing the transformed data; and transforming the smoothed transformed data by applying a second non linear mapping that is opposite to the first non linear mapping, to determine an estimate of the noise in the inputted data.

Abstract: Provided are methods and systems for improving quality of speech communications. The method may be for improving quality of speech communications in a system having a speech encoder configured to encode a first audio signal using a first set of encoding parameters associated with a first noise suppressor. A method may involve receiving a second audio signal at a second noise suppressor which provides much higher quality noise suppression than the first noise suppressor. The second audio signal may be generated by a single microphone or a combination of multiple microphones. The second noise suppressor may suppress the noise in the second audio signal to generate a processed signal which may be sent to a speech encoder. A second set of encoding parameters may be provided by the second noise suppressor for use by the speech encoder when encoding the processed signal into corresponding data.

Abstract: Systems, methods, and apparatus for spectral contrast enhancement of speech signals, based on information from a noise reference that is derived by a spatially selective processing filter from a multichannel sensed audio signal, are disclosed.

Abstract: A noise attenuation apparatus receives a first signal comprising a desired and a noise signal component. Two codebooks (109, 111) comprise respectively desired signal candidates and noise signal candidates representing possible desired and noise signal components respectively. A noise attenuator (105) generates estimated signal candidates by for each pair of desired and noise signal candidates generating an estimated signal candidate as a combination of the desired signal candidate and the noise signal candidate. A signal candidate is then determined from the estimated signal candidates and the first signal is noise compensated based on this signal candidate. A sensor signal representing a measurement of the desired source or the noise in the environment is used to reduce the number of candidates searched thereby substantially reducing complexity and computational resource usage. The noise attenuation may specifically be audio noise attenuation.

Abstract: The embodiments described herein are directed to systems and methods for transmitting audio data and control segment in a single bitstream and reducing audio disturbance associated with the control segment when the bitstream is processed by an audio digital-to-analog converter. The system, according to one aspect, comprises a first audio unit, a transmitter coupled to the first audio unit, a receiver coupled to the transmitter, a second audio unit coupled to the receiver, a first processor coupled to at least one of the first audio unit and the transmitter, a second processor coupled to the second audio unit and the receiver, and an audio digital-to-analog converter connected to the second processor.

Abstract: An apparatus for decoding data segments representing a time-domain data stream, a data segment being encoded in the time domain or in the frequency domain, a data segment being encoded in the frequency domain having successive blocks of data representing successive and overlapping blocks of time-domain data samples. The apparatus includes a time-domain decoder for decoding a data segment being encoded in the time domain and a processor for processing the data segment being encoded in the frequency domain and output data of the time-domain decoder to obtain overlapping time-domain data blocks. The apparatus further includes an overlap/add-combiner for combining the overlapping time-domain data blocks to obtain a decoded data segment of the time-domain data stream.

Abstract: A voice processing method for use in a communication apparatus, in an embodiment, includes the following steps. A near-end audio signal is received by at least one microphone of the communication apparatus. Voice and noise energy data are generated by performing voice activity detection on the near-end audio signal. A noise amount is obtained by performing noise energy calculation with the noise energy data. Whether the noise amount exceeds a first noise amount threshold is determined. If the noise amount exceeds the first noise amount threshold, a sidetone mode of the communication apparatus is enabled to produce a sidetone signal according to the voice energy data and play the sidetone signal through a speaker thereof. A noise suppression mode is enabled to produce a far-end audio signal according to the voice energy data and transmitting the far-end audio signal by a communication module of the communication apparatus.

Abstract: An audio decoding device capable of suppressing an information amount for a lost flame compensation process and encoding efficiency is provided. A decoded sound source generator generates a lost frame's CELP decoded sound source signal. A pitch pulse information decoder CELP decodes a pitch pulse position information and a pitch pulse amplitude information. A pitch pulse waveform learner learns a pitch pulse learning waveform in a past frame in advance from the lost frame. A convolution adjuster amplitude-adjusts the pitch pulse learning waveform according to the pitch pulse amplitude information by considering a predetermined number of waveforms peripheral to a peak position of the lost frame's CELP decoded excitation signal, and convolutes a pitch pulse waveform into a time axis which has been amplitude-adjusted according to the pitch pulse position information.

Abstract: A communication system includes a front-end audio gateway or bridge and a hands-free device. An automatic speech recognition platform accessible to the hands-free device provides or makes available one or more preprocessing schemes and/or acoustic models to the front-end audio gateway or bridge. The preprocessing schemes or acoustic models can be identified by or provided before a connection is established between the front-end audio gateway and the automatic speech recognition platform, when a connection occurs between the front-end audio gateway and the automatic speech recognition platform, and/or during a speech recognition session.

Abstract: In accordance with an embodiment, a time-frequency post-processing method of improving perceptual quality of a decoded audio signal, the method includes determining a time-frequency representation (such as filter bank analysis and synthesis) of an audio signal, estimating a time-frequency energy distribution of an audio signal from a time-frequency filter bank, computing a modification gain for each time-frequency representation point to have a modified time-frequency representation, and outputting audio signal from a modified time-frequency representation.

Abstract: A method for operating a speech recognition system is described in which a speech signal (S1) of a user is detected and analyzed so as to recognize speech information contained in the speech signal (S1). The speech recognition system determines a reception quality value (SQ) or a noise value which represents a current reception quality. The speech recognition system is switched over to a mode of operation which is less sensitive to noise and/or outputs an alert signal (SW) to the user when the reception quality value (SQ) drops below a given reception quality threshold or when the noise value exceeds a noise threshold. An appropriate speech recognition system is also described.

Abstract: A method is disclosed for controlling a voice-activated device by interpreting a spoken command as a series of voiced and non-voiced intervals. A responsive action is then performed according to the number of voiced intervals in the command. The method is well-suited to applications having a small number of specific voice-activated response functions. Applications using the inventive method offer numerous advantages over traditional speech recognition systems including speaker universality, language independence, no training or calibration needed, implementation with simple microcontrollers, and extremely low cost. For time-critical applications such as pulsers and measurement devices, where fast reaction is crucial to catch a transient event, the method provides near-instantaneous command response, yet versatile voice control.

Abstract: A voice band enhancement apparatus is used that includes a frequency transform unit to perform frequency transform on an input signal to calculate a spectrum, a mapping function calculating unit to calculate, by use of the spectrum, a mapping function for generating high-range components from low-range components of the spectrum, a wide-band spectrum generating unit to generate, in a higher range than a band of the spectrum, a high-range spectrum based on the mapping function and to integrate the generated high-range spectrum and the spectrum calculated by the frequency transform unit, thereby generating a wide-band spectrum wider than the band of the spectrum calculated by the frequency transform unit, and an inverse frequency transform unit to perform inverse frequency transform on the wide-band spectrum to calculate an output signal.

Abstract: A Yule-Walker based, low-complexity voice activity detector (VAD) is disclosed. An input signal is typically noisy speech (i.e., corrupted with, for example, babble noise). In one embodiment, a first initialization stage of the VAD computes an occurrence of a silent period within the input signal and the AR parameters. The VAD could accordingly compute a tentative adaptive threshold and output hypothesis H1 (which means speech is present) during this stage. During the second initialization stage, the VAD generally builds a database of associated values and computes the adaptive threshold accordingly. The second initialization stage could also output tentative VAD decisions based on the tentative threshold computed in the first initialization stage. Finally, the VAD periodically retrains or updates AR parameters, threshold values and/or the database and outputs VAD decisions accordingly.

Abstract: An encoding method includes: extracting core layer characteristic parameters and enhancement layer characteristic parameters of a background noise signal, encoding the core layer characteristic parameters and enhancement layer characteristic parameters to obtain a core layer codestream and an enhancement layer codestream. The disclosure also provides an encoding device, a decoding device and method, an encapsulating method, a reconstructing method, an encoding-decoding system and an encoding-decoding method. By describing the background noise signal with the enhancement layer characteristic parameters, the background noise signal can be processed by using more accurate encoding and decoding method, so as to improve the quality of encoding and decoding the background noise signal.

Abstract: A method and computing system for suppressing noise in an audio signal, comprising: receiving the audio signal at signal processing means; determining that another signal is input to the signal processing means, the input signal resulting from an activity which generates noise in the audio signal; and selectively suppressing noise in the audio signal in dependence on the determination that the input signal is input to the signal processing means to thereby suppress the generated noise in the 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 method and apparatus for generating synthesis audio signals are provided. The method includes decoding a bitstream; splitting the decoded bitstream into n sub-band signals; generating n transformed sub-band signals by transforming the n sub-band signals in a frequency domain; and generating synthesis audio signals by respectively multiplying the n transformed sub-band signals by values corresponding to synthesis filter bank coefficients.

Abstract: A noise estimation method for a noisy speech signal according to an embodiment of the present invention includes the steps of approximating a transformation spectrum by transforming an input noisy speech signal to a frequency domain, calculating a smoothed magnitude spectrum having a decreased difference in a magnitude of the transformation spectrum between neighboring frames, calculating a search spectrum to represent an estimated noise component of the smoothed magnitude spectrum, and estimating a noise spectrum by using a recursive average method using an adaptive forgetting factor defined by using the search spectrum. According to an embodiment of the present invention, the amount of calculation for noise estimation is small, and large-capacity memory is not required. Accordingly, the present invention can be easily implemented in hardware or software. Further, the accuracy of noise estimation can be increase because an adaptive procedure can be performed on each frequency sub-band.

Abstract: In a signal processing method and apparatus, a predetermined correcting signal having a same frame length as a second frame signal in which predetermined processing is performed to a frequency spectrum of a first frame signal of a frame length to which a predetermined window function is performed and is converted into a time domain is adjusted so that amplitudes of both ends of the correcting signal become equal to amplitudes of both or one of frame ends of the second frame signal, and a corrected frame signal is obtained by subtracting an adjusted correcting signal from the second frame signal.

Abstract: A method for overlap-adding signals useful for performing frame loss concealment (FLC) in an audio decoder as well as in other applications. The method uses a dynamic mix of windows to overlap two signals whose normalized cross-correlation may vary from zero to one. If the overlapping signals are decomposed into a correlated component and an uncorrelated component, they are overlap-added separately using the appropriate window, and then added together. If the overlapping signals are not decomposed, a weighted mix of windows is used. The mix is determined by a measure estimating the amount of cross-correlation between overlapping signals, or the relative amount of correlated to uncorrelated signals.

Abstract: The present invention relates to a postfilter and a postfilter control to be associated with a postfilter for improving perceived quality of speech reconstructed at a speech decoder. The postfilter control comprises means for measuring stationarity of a speech signal reconstructed at a decoder, means for determining a coefficient to a postfilter control parameter based on the measured stationarity, and means for transmitting the determined coefficient to a postfilter, such that the postfilter can process the reconstructed speech signal by applying the determined coefficient to the postfilter control parameter to obtain an enhanced speech signal.

Abstract: A method of removing noise includes detecting a frequency spectrum of a noise signal around the transmitting terminal, when an input signal which is a mixture of a voice signal and the noise signal is received, detecting a frequency spectrum of the input signal and an energy level of the voice signal, multiplying the frequency spectrum of the noise signal by a weight value that is determined based on the energy level of the voice signal to obtain a weighted noise frequency spectrum, and subtracting the weighted noise frequency spectrum from the frequency spectrum of the input signal.

Abstract: The invention concerns a method and apparatus for performing packet loss or Frame Erasure Concealment (FEC) for a speech coder that does not have a built-in or standard FEC process. A receiver with a decoder receives encoded frames of compressed speech information transmitted from an encoder. A lost frame detector at the receiver determines if an encoded frame has been lost or corrupted in transmission, or erased. If the encoded frame is not erased, the encoded frame is decoded by a decoder and a temporary memory is updated with the decoder's output. A predetermined delay period is applied and the audio frame is then output. If the lost frame detector determines that the encoded frame is erased, a FEC module applies a frame concealment process to the signal. The FEC processing produces natural sounding synthetic speech for the erased frames.

Abstract: There is disclosed an audio decoding device capable of improving audio quality of a decoded signal by considering the energy change of a past signal in eracure concealment processing. In this device, an energy change calculation unit (143) calculates an average energy of an audio source signal of one-pitch cycle from the end of the ACB vector outputted from an adaptive codebook (106). Moreover, the energy change calculation unit (143) calculates a ratio of the average energy of the current sub-frame and the sub-frame immediately before and outputs the ratio to an ACB gain generation unit (135). The ACB gain generation unit (135) outputs a conceal processing ACB gain defined by the ACB gain decoded in the past or information on the energy change ratio outputted from the energy change calculation unit (143) to a multiplier (132).

Abstract: A correction spectrum calculation unit 6 obtains a correction spectrum by smoothing an estimated noise spectrum in accordance with the degree of its variations, and a suppression quantity limiting coefficient calculation unit 7 decides a suppression quantity limiting coefficient from the correction spectrum. A suppression quantity calculation unit 9 obtains a suppression coefficient based on the suppression quantity limiting coefficient, and the spectrum suppression unit 10 carries out amplitude suppression of spectral components of an input signal.