Abstract: A novel beamforming post-processor technique with enhanced noise suppression capability. The present beam forming post-processor technique is a non-linear post-processing technique for sensor arrays (e.g., microphone arrays) which improves the directivity and signal separation capabilities. The technique works in so-called instantaneous direction of arrival space, estimates the probability for sound coming from a given incident angle or look-up direction and applies a time-varying, gain based, spatio-temporal filter for suppressing sounds coming from directions other than the sound source direction resulting in minimal artifacts and musical noise.

Abstract: Provided are an apparatus and method for transmitting sound through a nonlinear medium. The apparatus includes a pre-distorter for previously distorting a sound signal to compensate for distortion to be caused by a frequency characteristic of the nonlinear medium in a transmission process of the sound signal, a sigma-delta modulator for modulating the pre-distorted sound signal into a signal having two signal levels, a high-frequency modulator for multiplying the modulated signal by a carrier wave having a higher frequency than an audio frequency band to shift the modulated signal to a carrier frequency band and generate a high-frequency modulated signal, and a transmitter for converting the high-frequency modulated signal into a sound wave signal suited to be transmitted through the nonlinear medium and transmitting the sound wave signal.

Abstract: Apparatus and a corresponding method for processing speech signals in a noisy reverberant environment, such as an automobile. An array of microphones (10) receives speech signals from a relatively fixed source (12) and noise signals from multiple sources (32) reverberated over multiple paths. One of the microphones is designated a reference microphone and the processing system includes adaptive frequency impulse response (FIR) filters (24) enabled by speech detection circuitry (21) and coupled to the other microphones to align their output signals with the reference microphone output signal. The filtered signals are then combined in a summation circuit (18). Signal components derived from the speech signal combine coherently in the summation circuit, while noise signal components combine incoherently, resulting in composite output signal with an improved signal-to-noise ratio. The composite output signal is further processed in a speech conditioning circuit (20) to reduce the effects of reverberation.

Abstract: The invention relates to a method for assessing interfering noise in motor vehicles, according to which noise occurring during a predefined measuring time is divided into different frequency ranges, the changes in level relative to the background noise are determined within said frequency ranges, and the determined changes in level are evaluated.

Abstract: An audio interface adapted to reduce a subscriber voice may receive a subscriber voice and a background noise. The subscriber voice may then be compared to the to the background noise. If the received subscriber voice is louder than the received background noise, the audio interface may output a message to the cellular telephone subscriber indicating the subscriber may reduce his speaking volume. Additionally, an audio interface may process a voice waveform that corresponds to the subscriber voice and a background waveform that corresponds to the background noise to generate a substantially opposite voice waveform and a substantially opposite background waveform respectively. The substantially opposite voice waveform and background waveform may be substantially out of phase from the voice waveform and background waveform respectively and output via one or more output ports of the audio interface.

Abstract: An embodiment includes a method that includes receiving data through a non-voice input. The method also includes translating the data into one or more numeric values. The method includes encoding the one or more numeric values into an audio stream, wherein the audio stream is to be transmitted over a transmission medium that is in use for voice communication.

Abstract: A hearing aid includes a hearing aid housing enclosing a microphone for converting sound into an audio signal, first feedback compensation means for providing a first feedback compensation signal of signals picked up by the microphone by modeling an internal mechanical feedback signal path of the hearing aid, second feedback compensation means for providing a second feedback compensation signal by modeling an external feedback signal path of the hearing aid, subtracting means for subtracting the first and second feedback compensation signals from the audio signal to form a compensated audio signal, processing means, connected to an output of the subtracting means, for processing the compensated audio signal, and a receiver, connected to an output of the processing means, for converting the processed compensated audio signal into a sound signal.

Abstract: An audio driver with reduced rub and buzz distortion that includes a digital processing module. A digital to audio converter (DAC) operable to receive a digital audio signal supplied by the digital processing module. One or more analog driver stages operable to receive an analog audio signal supplied by the DAC. A peak amplitude compressor.

Abstract: A communication system (e.g., a speakerphone) includes an array of microphones, a speaker, memory and a processor. The processor may perform a virtual broadside scan on the microphone array and analyze the resulting amplitude envelope to identify acoustic source angles. Each of the source angles may be further investigated with a directed beam (e.g., a hybrid superdirective/delay-and-sum beam) to obtain a corresponding beam signal. Each source may be classified as either intelligence or noise based on an analysis of the corresponding beam signal. The processor may design a virtual beam pointed at an intelligence source and having nulls directed at one or more of the noise sources. Thus, the virtual beam may be highly sensitive to the intelligence source and insensitive to the noise sources.

Abstract: A charge/discharge control circuit for controlling current through an input/output audio device includes a first voltage reference; a second voltage reference and a waveform generation circuit responsive to the first and second voltage references for generating a multi-stage waveform profile which is approximately an inaudible waveform for suppressing audible artifacts in the input/output device.

Abstract: A pseudo noise superimposing unit superimposes a pseudo noise (M-sequence) to an audio signal picked up by a microphone and outputs the superimposed signal to an amplifying system. An calculating unit calculates a correlation value between the audio signal picked up by the microphone and the pseudo noise. The calculating unit estimates a gain of a closed loop based on the correlation value. A gain control unit suppresses a gain of the audio signal based on the estimated gain of the closed loop.

Abstract: The invention provides a beam forming method for a small array microphone apparatus to generate cone beam pattern by processing a combined bi-directional beam pattern of two virtual bi-directional microphones formed through at least three omni-directional microphones arranged in an L-shape. The invention also provides a small array microphone apparatus using the beam forming method according to the invention to suppress noise by processing a combined bi-directional beam pattern of two virtual bi-directional microphones formed through at least three omni-directional microphones arranged in an L-shape, thereby outputting a clear audio signal with cone beam pattern.

Abstract: A headset is constructed to generate an acoustically distinct speech signal in a noisy acoustic environment. The headset positions a pair of spaced-apart microphones near a user's mouth. The microphones each receive the user's speech, and also receive acoustic environmental noise. The microphone signals, which have both a noise and information component, are received into a separation process. The separation process generates a speech signal that has a substantial reduced noise component. The speech signal is then processed for transmission. In one example, the transmission process includes sending the speech signal to a local control module using a Bluetooth radio.

Abstract: A communication device includes: (1) a wireless adapter at which a wireless headset is communicatively connected to the communication device and at which is received a first acoustic input that includes a speech input and a first ambient noise input; (2) a microphone that receives a second acoustic input, which includes a second ambient noise input; and (3) a dual-channel adaptive noise canceller that utilizes the second ambient noise input to filter the first ambient noise input out of the first acoustic input to generate an acoustic output that primarily comprises the speech input.

Abstract: A noise suppressing device is provided for suppressing noise of a first audio signal to generate a second audio signal. In the noise suppressing device, a noise acquisition unit acquires a plurality of noise components which are different from each other. A noise suppression unit generates each suppression component by suppressing each noise component from the first audio signal, thereby providing a plurality of suppression components different from each other in correspondence to the plurality of the noise components. A signal generation unit generates the second audio signal by summing the plurality of the suppression components that are provided from the noise suppression unit.

Abstract: A portable audio device having reduced sensitivity to RF interference over a predetermined frequency range from an adjacent mobile wireless communications device may include a portable housing, a battery carried thereby, a recharging power input connected to the battery, a digital signal input, and an audio analog signal output. A digital-to-analog converter (DAC) may be carried by the portable housing and powered by the battery for converting a selected digital audio file from a memory into an analog audio signal. An audio analog amplifier may be connected between the DAC and the audio analog signal output. A first RF filter(s) may be connected to the recharging power input, and a second RF filter(s) may be connected to the audio analog signal output, both for reducing RF interference over the predetermined frequency range from the adjacent mobile wireless communications device.

Abstract: Systems, methods, and devices for noise profile determination for a voice-related feature of an electronic device are provided. In one example, an electronic device capable of such noise profile determination may include a microphone and data processing circuitry. When a voice-related feature of the electronic device is not in use, the microphone may obtain ambient sounds. The data processing circuitry may determine a noise profile based at least in part on the obtained ambient sounds. The noise profile may enable the data processing circuitry to at least partially filter other ambient sounds obtained when the voice-related feature of the electronic device is in use.

Abstract: A pop sound prevention module and a speaker apparatus thereof are provided. The pop sound prevention module is used for preventing the speaker apparatus generating a pop sound when a power shuts down. The speaker apparatus includes an audio amplifier, an audio processor, the pop sound prevention module, and a loudspeaker. The audio processor is coupled to a first operation voltage. The audio amplifier is coupled to a second operation voltage. When the power shuts down, the pop sound prevention module outputs a mute control signal to the audio amplifier according to voltage difference between the first operation voltage and the second operation voltage. According to the mute control signal, the audio amplifier prevents the loudspeaker from generating a pop sound.

Abstract: An audio signal compensation device includes: a signal processor configured to perform filtering on an input audio signal; a filter coefficients storage module configured to store a plurality of filter coefficients; a user interface configured to provide options for a determination of filter coefficients to a user and to obtain a selection result from the user; and a filter coefficients determining module configured to determine a set of filter coefficients among the plurality of filter coefficients based on the selection result.

Abstract: A signal processing apparatus includes: two sound input units, an orthogonal transformer to transform two sound signals input from the two sound input units into respective spectral signals in a frequency domain, a phase difference calculator to calculate a phase difference between the spectral signals in the frequency domain, a range determiner to determine a coefficient responsive to a frequency in the phase difference as a function of frequency, and determine a suppression range related to a phase on a per frequency basis of the frequency responsive to the coefficient; and a filter to phase-shift a component of one of the spectral signals on a per frequency basis in order to generate a phase-shifted spectral signal when the phase difference at each frequency falls within the suppression range, synthesizing the phase-shifted spectral signal and the other of the spectral signals in order to generate a filtered spectral signal.

Abstract: A sound distortion suppression control method not causing sound distortion even if the volume of an amplifier is increased to nearly its maximum value when the user adjusts bass and treble to obtain desired tone. Sound to be output is specified by a volume setting value, a bass setting value and a treble setting value, and these values are temporarily stored in a memory section. Furthermore, a maximum allowable bass value and a maximum allowable treble value are respectively stored in the memory section in advance. In the case that the bass setting value and the treble setting value become larger than the maximum allowable bass value and the maximum allowable treble value, respectively, the bass setting value and the treble setting value are changed to a value not more than the maximum allowable bass value and a value not more than the maximum allowable treble value, respectively.

Abstract: A communication system (e.g., a speakerphone) includes an array of microphones, a speaker, memory and a processor. The processor may be configured to perform acoustic echo cancellation, to track multiple talkers with highly directed beams, to design beams with nulls pointed at noise sources, to generate a 3D model of the physical environment, to compensate for the proximity effect, and to perform dereverberation of a talker's voice signal. The processor may also be configured to use a standard codec in non-standard ways. The processor may perform a virtual broadside scan on the microphone array, analyze the resulting amplitude envelope for acoustic source angles, examine each of the source angles with a directed beam, combine the beam outputs that show the characteristics of intelligence or speech.

Abstract: A system such as a speakerphone may include a processor, memory and an array of microphones. The processor may be configured (via program instructions stored in the memory) to perform automatic echo cancellation, self calibration and beam forming. In particular, the processor may receive input signals from the microphone array and operate on the input signals with a highly directed virtual beam which is a composite of two or more beams which are restricted to respective frequency ranges. The two or more beams may include beams of different kinds, e.g., superdirective beams and delay-and-sum beams.

Abstract: A receiving apparatus includes: a local oscillator to output first- and second-local-oscillator signals whose phases are orthogonal to each other; a mixer to output first- and second-intermediate-frequency signals; a first filter to allow a component from a desired signal to pass therethrough, and eliminate a component from an image signal having a frequency symmetrical with that of the desired signal, in the first- and second-intermediate-frequency signals; a second filter to allow a component from the image signal to pass therethrough, and eliminate a component from the desired signal, in the first- and second-intermediate-frequency signals; a comparator to compare levels between output signals of the first and second filters; and a control unit to switch a frequency of the first- and second-local-oscillator signals to a difference frequency between a frequency of the desired signal and the intermediate frequency or a sum frequency thereof, according to a comparison result of the comparator.

Abstract: An embodiment of the invention involves a display device and method for actively managing playback of demo content within a partially-assisted or non-assisted commercial environment. The playback of demo content is activated through detection of an audio triggering event in which playback of the streaming advertising content is temporarily halted. According to this embodiment, after the demo content is displayed, playback of the streaming advertising content resumes and cannot be interrupted for a selected period of time that is normally greater and will not be less than the playback time for the demo content. Other embodiments are described and claimed.

Abstract: A signal processing circuit includes: an AD converter configured to quantize an input signal, whose amplitude changes in accordance with temperature, within a set voltage range and convert the quantized input signal into a digital signal; and a setting circuit configured to set the voltage range so as to be wider when the input signal is greater in amplitude in accordance with the temperature and so as to be narrower when the input signal is smaller in amplitude in accordance with the temperature.

Abstract: A signal processing device includes a non-inverting amplifier, an inverting amplifier, a converter, and a controller. The non-inverting amplifier amplifies a level of an analog sound signal input from outside with a first gain whose value is variable. The inverting amplifier amplifies a level of the analog sound signal amplified by the non-inverting amplifier with a second gain whose value is variable. The converter converts the analog sound signal amplified by the inverting amplifier to a digital sound signal. The controller detects a level of the digital sound signal converted by the converter and, in accordance with the detected level of the digital sound signal converted by the converter, controls the first gain and the second gain such that a level of the analog sound signal input to the converter is at a pre-specified level.

Abstract: A sound processing apparatus according to the present invention sets a volume value to a default value, outputs a pure tone signal of a predetermined frequency as a test signal from a loudspeaker, and acquires the output test signal using a microphone. When a signal level of a harmonic tone component of the acquired signal is equal to or greater than a threshold, the apparatus reduces the volume value so that the signal level equal to or greater than the threshold falls below the threshold and stores the reduced volume value. When outputting an acoustic signal, the apparatus adjusts a signal level of the acoustic signal at a stored frequency so that a product of a signal level at the stored frequency and a current volume value does not exceed a product of a signal level of the test signal and the stored volume value.

Abstract: A system for noise removal is coupled to a signal unit that provides a digital signal. The noise removal system includes a transformation module to transform the digital signal into an f-digital signal, a threshold filter to generate a noiseless signal from the f-digital signal based on a threshold profile, and a signal synthesizer to provide a gain to the noiseless signal and to transform the noiseless signal into an output signal.

Abstract: A device and method are provided that reduce interference between a wireless communication device and a speaker. Generally speaking, a microphone input is monitored for detecting noise created by the interference. If noise is detected, a power transmission level of the wireless device is reduced from a standard power transmission level.

Abstract: The adaptive beamformer unit (191) comprises: a filtered sum beamformer (107) arranged to process input audio signals (u 1, u2) from an array of respective microphones (101, 103), and arranged to yield as an output a first audio signal (z) predominantly corresponding to sound from a desired audio source (160) by filtering with a first adaptive filter (f1(-t)) a first one of the input audio signals (u1) and with a second adaptive filter (f2(-t)) a second one of the input audio signals (u2), the coefficients of the first filter (f1(-t)) and the second filter (f2(-t)) being adaptable with a first step size (a1) and a second step size ((x2) respectively; noise measure derivation means (111) arranged to derive from the input audio signals (u1, u2) a first noise measure (x1) and a second noise measure (x2); and an updating unit (192) arranged to determine the first and second step size (a1, (x2) with an equation comprising in a denominator the first noise measure (x1) for the first step size (a1), respectively the

Abstract: A speaker arrangement is described, suitable for improving the sound quality of a small size first speaker. The arrangement comprises a cavity, to which the first speaker is acoustically coupled. A second speaker is also acoustically coupled to the cavity, and the first and second speakers are excited in phase with each other. The first speaker may be an earphone speaker tuned to be held to a user's ear, whereas the second speaker may be a larger ring alert speaker with lower compliance than the first speaker, both of which are incorporated in the same mobile phone. By exciting the second speaker in phase with the first speaker, the second speaker will aid the smaller first speaker's reproduction of sound, particularly in the lower frequency range.

Abstract: A method of increasing the intelligibility of an audio broadcast in an at least partially enclosed space from at least one amplified audio source. An input microphone receives an incident audio wavefront at a first position in the at least partially enclosed space. An active noise control system is employed to generate a cancelling audio wavefront having a magnitude substantially equal to the magnitude of incident audio wavefront and a phase substantially opposite to the phase of the incident audio wavefront. The cancelling audio wavefront is broadcast at a second position in the at least partially enclosed space adjacent to a reflective surface of the at least partially enclosed space so as to attenuate the incident audio wavefront substantially at or near the reflective surface in order to reduce reverberations of the incident audio wavefront. In this manner, reverberations which could reduce the intelligibility of the audio broadcast to an audience is reduced.

Abstract: An active vibration/noise control device which is provided with a plurality of cancel signal generation parts for generating output signals for respectively cancelling noises generated at a plurality of vibration/noise generation sources. The effect of the suspension of either of first and second cancel signal generation parts on the other is reduced. According to the operating state of the first cancel signal generation part, the simulated transmission properties of the second cancel signal generation part are adjusted. Consequently, without regard to the operating state of the first cancel signal generation part, the noise control performance of the second cancel signal generation part can be maintained.

Abstract: A processed component calculating unit 14 obtains a transformed noise suppressed spectrum 18a based on the ratio between a noise suppressed spectrum 18 and an estimated noise spectrum 17, and a phase disturbing unit 15 performs phase disturbance to obtain a processed spectrum 19 consisting of smoothed components that make deterioration components in the noise suppressed spectrum 18 subjectively imperceptible. A signal addition unit 11 adds the processed spectrum 19 to the frequency components of the noise suppressed spectrum 18 deteriorated through the noise suppression of a noise suppressing unit 3 to suppress the deterioration components.

Abstract: This disclosure describes circuit configurations that may be used for active noise cancellation in the digital domain. In particular, this disclosure proposes the use a down sample unit and an up sample unit, rather than memory-based delay circuits, to achieve one or more desired delays in digital adaptive noise cancellation circuits or other circuits that use delay for signal processing. The delay achieved by the down sample unit and the up sample unit may be tunable so as to allow flexibility in producing the necessary delay for different active noise cancellation circuit configurations. Many different adaptive noise cancellation circuit configurations are discussed, and the techniques may also be useful for other types of circuits, such as low-latency equalization circuits.

Abstract: Sound quality is enhanced in a sound system including handsets and headsets. Handset sound enhancing algorithms are implemented in a handset. The handset automatically determines which, if any, of a plurality of headset sound enhancing algorithms are active in a headset in communication with the handset. The handset determines how to use the handset sound enhancing algorithms in a sound processing channel based on which of the headset sound enhancing algorithms are active in the headset.

Abstract: A noise reduction system is used in a BTSC system to reduce noise of an audio signal. The noise reduction system has an audio spectral compressing unit that has a filter and a memory in the approach of the digital processing. The filter is arranged to filter an input signal according to a transfer function, a variable d, and several parameters b0/a0, a0/b0, b1/b0 and a1/a0. The memory is arranged to store the parameters.

Abstract: A method of suppressing wind noise in a voice signal determines an upper frequency limit that lies within the frequency spectrum of the voice signal, and for each of a plurality of frequency bands below the upper frequency limit, compares the average power of signal components in a first portion of the signal to the average power of signal components in a second portion of the signal, where the second portion is successive to the first portion. Signal components are identified in at least one of the plurality of frequency bands as containing impulsive wind noise in dependence on the comparison, and the identified signal components are attenuated.

Abstract: Methods and systems to produce audio output signals from audio input signals. In one embodiment, a first portion of the audio input signals can be pre-processed, with the output used to modulate ultrasonic carrier signals, thereby producing modulated ultrasonic signals. The modulated ultrasonic signals can be transformed into a first portion of the audio output signals, which is directional. Based on a second portion of the audio input signals, a standard audio speaker can output a second portion of the audio output signals. Another embodiment further produces distortion compensated signals based on the pre-processed signals. The distortion compensated signals can be subtracted from the second portion of the audio input signals to generate inputs for the standard audio speaker to output the second portion of the audio output signals. In yet another embodiment, noise can be added during pre-processing of the first portion of the audio input signals.

Abstract: An audio output apparatus including a digital operation unit, a digital-to-analog converter (DAC), a left channel unit, a right channel unit and a common unit is provided. During an initial setting, the digital operation unit detects a cross talk voltage on a left channel earphone unit in the left channel unit or a right channel earphone unit in the right channel unit, calculates a first ratio and a second ratio, and then performs an arithmetic operation on a left and a right channel signals according to the first and the second ratios, so as to obtain a compensated left channel signal and a compensated right channel signal respectively for eliminating a cross talk phenomenon.

Abstract: Systems and methods are described for applying noise suppression to one or more audio signals to generate a noise-suppressed audio signal therefrom. In a single-channel implementation, an input signal is received that comprises a desired audio signal and an additive noise signal. Noise suppression is then applied to the input signal to generate a noise-suppressed signal in a manner that is controlled by at least a parameter that specifies a degree of balance between distortion of the desired audio signal and unnaturalness of a residual noise signal included in the noise-suppressed signal. In an alternative single-channel implementation, a plurality of sub-band signals obtained by applying a frequency conversion process to a time domain representation of an input signal is received. Noise suppression is then applied to each of the sub-band signals by passing each of the sub-band signals through a time direction filter. Multi-channel noise suppression variants are also described.

Abstract: A crosstalk suppression arrangement for a stereo audio system, wherein the stereo audio system comprises a left stereo channel and a right stereo channel, each comprising a driving circuit, an output impedance of the driving circuit comprising at least a first output impedance part, and a load impedance, and wherein the load impedances of the left and the right stereo channels are connected to a reference voltage via a common reference voltage impedance, the crosstalk suppression arrangement comprising: a crosstalk suppression impedance having an impedance value based on the reference voltage impedance, at least one of the load impedances and at least one of the output impedances, wherein the crosstalk suppression impedance is connected at one end to a point between the load impedance and the output impedance of the left stereo channel and at another end to a point between the load impedance and the output impedance of the right stereo channel.

Abstract: A method for controlling a noise cancellation system having an adaptive control portion is provided. The noise cancellation system is operable to generate a cancellation noise configured to at least partially cancel an unwanted noise in a defined environment. The adaptive control portion is operable to adjust the operation of the noise cancellation system based on a level of unwanted noise that remains when the cancellation noise and the unwanted noise are combined. The method includes receiving an error signal representing a portion of a noise not cancelled by a cancellation noise, where the cancellation noise is generated from the noise cancellation system. The method also includes determining whether the level of the error signal exceeds a first threshold value for a first predetermined period of time. The method also includes calculating a crest factor using the error signal. The method also includes determining whether the crest factor exceeds a second threshold value.

Abstract: A digital demodulation device for demodulating an amplitude modulation (AM) signal whose carrier has a first frequency includes: a processing circuit for performing first path digital processing and second path digital processing according to a second frequency and digital values of the AM signal, where the first path digital processing represents performing down conversion by mixing the AM signal with a first sinusoidal signal whose frequency is equal to the second frequency, the second path digital processing represents performing down conversion by mixing the AM signal with a second sinusoidal signal whose frequency is equal to the second frequency, the second frequency is equal to the first frequency plus a predetermined frequency shift, and the second sinusoidal signal is orthogonal to the first sinusoidal signal; and an output stage for outputting an output signal according to processing results of the first path digital processing and the second path digital processing.

Abstract: A sound processing apparatus includes a power spectrum operation unit obtaining a power spectrum of an audio signal, an envelope component removal unit removing an envelope component of the power spectrum and generating a signal characteristic that represents a peakness of the power spectrum, a filter characteristic calculation unit calculating a filter characteristic suppressing the signal characteristic by using the signal characteristic, and a suppress filter filtering the audio signal by using the filter characteristic.

Abstract: An apparatus and method for processing a signal are provided.

Abstract: The present invention relates to a sound source separation and display method and a system thereof, and provides in particular a sound source separation and display method and a system thereof that are intended to eliminate a specific sound source. In order to separate a plurality of sound sources by using a single set of microphone array, the result of processing of sound source identification is utilized. More specifically, a signal in a direction is extracted from the result of the processing of the sound source identification, and a field limited to/eliminated of the effect of the signal is calculated and displayed. Such an operation can be repeated. A virtual reference signal can be created in a time domain.

Abstract: A transient in a digital audio signal can be detected by generating a first set of spectral characteristics associated with a first portion of the digital audio signal and a second set of spectral characteristics associated with a second portion of the digital audio signal, wherein the first and second portions of the digital audio signal partially overlap, comparing values in the first set of spectral characteristics with corresponding values in the second set of spectral characteristics to generate a set of ratios, weighting the set of ratios, and analyzing at least a portion of the weighted set of ratios to detect a transient associated with the first portion of the digital audio signal. Further, an indicator identifying the presence of a detected transient can be output. Additionally, one or more ratios in the set of ratios can be weighted based on amplitude, frequency, or a power function.

Abstract: In a gain adjusting method and a gain adjusting device for adjusting gain of a processed voice signal that is obtained by signal processing of an input voice signal, a masking power of the processed voice signal is computed, and gain is adjusted for every frequency if the frequency is masked according to the masking power, such that a difference between the processed voice signal and the input voice signal where the frequency is not masked is canceled.