WO2015141103A1 - 信号処理装置、信号処理方法、および信号処理プログラム - Google Patents
信号処理装置、信号処理方法、および信号処理プログラム Download PDFInfo
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- 238000001228 spectrum Methods 0.000 description 7
- 230000003044 adaptive effect Effects 0.000 description 6
- 230000005236 sound signal Effects 0.000 description 4
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- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
- G10L21/00—Speech or voice signal processing techniques to produce another audible or non-audible signal, e.g. visual or tactile, in order to modify its quality or its intelligibility
- G10L21/02—Speech enhancement, e.g. noise reduction or echo cancellation
- G10L21/0208—Noise filtering
- G10L21/0264—Noise filtering characterised by the type of parameter measurement, e.g. correlation techniques, zero crossing techniques or predictive techniques
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- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
- G10L21/00—Speech or voice signal processing techniques to produce another audible or non-audible signal, e.g. visual or tactile, in order to modify its quality or its intelligibility
- G10L21/02—Speech enhancement, e.g. noise reduction or echo cancellation
- G10L21/0208—Noise filtering
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- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
- G10L21/00—Speech or voice signal processing techniques to produce another audible or non-audible signal, e.g. visual or tactile, in order to modify its quality or its intelligibility
- G10L21/02—Speech enhancement, e.g. noise reduction or echo cancellation
- G10L21/0272—Voice signal separating
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
- H04R3/00—Circuits for transducers, loudspeakers or microphones
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- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
- G10L21/00—Speech or voice signal processing techniques to produce another audible or non-audible signal, e.g. visual or tactile, in order to modify its quality or its intelligibility
- G10L21/02—Speech enhancement, e.g. noise reduction or echo cancellation
- G10L21/0208—Noise filtering
- G10L21/0216—Noise filtering characterised by the method used for estimating noise
- G10L2021/02161—Number of inputs available containing the signal or the noise to be suppressed
- G10L2021/02166—Microphone arrays; Beamforming
Definitions
- the present invention relates to a technique for acquiring a desired signal from a mixed signal in which a desired signal and noise are mixed.
- Patent Document 1 calculates a phase difference between at least two input signals out of multi-channel input signals and emphasizes the phase difference to remove a noise component included in the input signal. At the same time, a technique for reducing unerased noise is disclosed.
- phase difference is enhanced to reduce the unerased noise, but there are cases where a desired signal component is also removed together with the noise component.
- An object of the present invention is to provide a technique for solving the above-described problems.
- a signal processing apparatus includes: Correlated noise removing means for inputting at least two input signals in which a desired signal and a noise signal are mixed from at least two channels and removing a noise signal having a correlation between the at least two input signals; Residual noise removal for removing residual noise contained in the output signal of the correlation noise removal means based on the phase difference between the output signal of the correlation noise removal means and at least one input signal contained in the at least two input signals Means, Equipped with.
- a signal processing method inputs at least two input signals in which a desired signal and a noise signal are mixed from at least two channels, and correlates the at least two input signals.
- a correlated noise removal step for removing a noise signal having; Residual noise removal for removing residual noise included in the output signal in the correlation noise removal step based on a phase difference between the output signal of the correlation noise removal step and at least one input signal included in the at least two input signals Steps, including.
- a signal processing program inputs at least two input signals in which a desired signal and a noise signal are mixed from at least two channels, and correlates between the at least two input signals.
- a correlated noise removal step for removing a noise signal having; Residual noise removal for removing residual noise included in the output signal in the correlation noise removal step based on a phase difference between the output signal of the correlation noise removal step and at least one input signal included in the at least two input signals Steps, Is executed on the computer.
- only a noise component can be removed without removing a desired signal component.
- the “voice signal” is a direct electrical change that occurs in accordance with voice and other sounds, and is used to transmit voice and other sounds, and is not limited to voice.
- the signal processing apparatus 100 includes a correlation noise removing unit 101 and a residual noise removing unit 102.
- the residual noise removing unit 102 includes suppression coefficient calculation units 201 1 to 201 M and a suppression unit 202.
- the correlation noise removal unit 101 inputs at least two input signals X 1 to X M in which a desired signal and a noise signal are mixed from at least two channels. Then, a noise component that is commonly included in these input signals, that is, a noise component having a correlation between channels is removed, and X 0 is output.
- the residual noise removing unit 102 receives the output signal X 0 of the correlation noise removing unit 101 and at least one input signal among at least two input signals X 1 to X M. Based on the difference (phase difference) between the phase of the output signal X 0 and the phase of at least one input signal among X 1 to X M , the noise component contained in X 0 is removed, and S 0 is output.
- the suppression coefficient calculation units 201 1 to 201 M calculate the suppression coefficients W 1 to W M based on the phase differences between the input signals X 0 and X 1 to X M , respectively.
- the suppression unit 202 removes the residual noise component included in the input signal X 0 using at least one suppression coefficient among the suppression coefficients W 1 to W M.
- FIG. 6 is a flowchart showing processing of the signal processing apparatus according to the present embodiment.
- FIG. 3 is a diagram illustrating a configuration of the signal processing device 300 according to the present embodiment.
- the signal processing apparatus 300 is a system that acquires a desired signal from a multi-channel mixed signal in which a desired signal and noise are mixed.
- a desired signal will be described as an audio signal, but the technical scope of the present invention is not limited thereto.
- the signal processing device 300 includes a correlation noise removing unit 301 and a residual noise removing unit 302.
- Correlated noise removing section 301 receives two or more multi-channel input signals X 1 to X M as input, and mainly removes noise components included in two or more channels, that is, noise components having a correlation between channels, X 0 is output.
- the residual noise removing unit 302 receives the output signal X 0 of the correlation noise removing unit 301 and at least one of the multi-channel input signals X 1 to X M as inputs. Based on the difference (phase difference) between the phase of X 0 and the phase of at least one signal among X 1 to X M , the noise component contained in X 0 is removed, and S 0 is output.
- the multi-channel input signals X 1 to X M are modeled as follows.
- X 1 to X M are complex spectra of the input signal, and can be obtained by performing frequency analysis such as discrete Fourier transform on the time domain signal of the corresponding channel.
- f is a frequency index
- t is a time index.
- S is the complex spectrum of the desired speech component.
- N c1 to N cM are noise components included in two or more channels among the channels 1 to M, that is, complex spectra of noise components having a correlation between the channels.
- N i1 to N iM are complex spectra of noise components independently included in each of the channels 1 to M, that is, noise components having a low correlation between channels.
- an adaptive noise canceller for example, Patent Document 2: Method described in International Publication No. 2005/024787
- an adaptive beamformer a generalized sidelobe canceller, a minimum dispersion beamformer, etc.
- the removal processing in the correlation noise removing unit 301 may be either frequency domain processing or time domain processing.
- the signal may be converted into a frequency domain signal X 0 by frequency analysis after the processing.
- the correlation noise removing unit 301 outputs X 0 represented by the following (Equation 2).
- N i0 is residual noise after the processing of the correlation noise removing unit 301, and is mainly a noise component having no correlation between channels. If the difference between N c1 to N cM (phase difference, amplitude difference) between channels is known in advance, a method that does not require adaptive operation, such as a fixed beamformer that directs null to a specific space, is used. It is also possible.
- FIG. 4 shows the configuration of the residual noise removing unit 302.
- the residual noise removing unit 302 includes a noise removing unit 421 based on the phase difference.
- the noise removal unit 421 based on the phase difference receives at least one of the output signal X 0 of the correlation noise removal unit 301 and the multi-channel input signals X 1 to X M. Based on the difference (phase difference) between the phase of X 0 and the phase of at least one signal among X 1 to X M , the noise component included in X 0 is removed, and S 1 is output. S 1 is output from the residual noise removing unit 302 as S 0 .
- FIG. 5 shows a configuration of the noise removal unit 421 based on the phase difference.
- Noise removal unit 421 based on phase difference includes suppression coefficient calculation units 501 1 to 501 M , suppression coefficient integration unit 502, and suppression unit 503.
- suppression coefficient calculator The suppression coefficient calculation units 501 1 to 501 M calculate the suppression coefficients W 1 to W M using the output signal X 0 and the multi-channel input signals X 1 to X M of the correlation noise removal unit 301, respectively. Since the operation for the channel 1 ⁇ M are the same, it will be described suppression coefficient calculation unit 501 1.
- Phase component of X 0 which is input to the suppression coefficient calculation unit 501 1 exp ⁇ -j ⁇ X0 ⁇ is, X amplitude component of 0 (Equation 2)
- ⁇ X0 is the phase of X 0 .
- ⁇ X1 is the phase of X 1 .
- the suppression coefficient calculation unit 501 M calculates the suppression coefficient W M using the following equation.
- the suppression coefficient calculation units 501 1 to 501 M output W 1 and W M calculated by (Expression 5) and (Expression 13). Since
- the suppression coefficient integration unit 502 receives the suppression coefficients W 1 to W M from the suppression coefficient calculation units 501 1 to 501 M and outputs the integrated suppression coefficient W S1 .
- the integrated suppression coefficient W S1 is obtained as follows.
- Ave is an average operator. Note that the average calculation need not be performed for all the suppression coefficients W 1 to W M , and it may be possible to omit the suppression coefficient having a large deviation from the overall average value and average again. It is also conceivable that averaging is performed using only channel suppression coefficients having values in a predetermined range, and averaging is performed using only channel suppression coefficients determined in advance.
- the channel suppression coefficient determined in advance is used without averaging, the channel suppression coefficient having the maximum value among the suppression coefficients W 1 to W M is used so that a desired audio component is not removed, etc. Can be considered.
- the suppression coefficient integration unit 502 receives the suppression coefficients W 1 to W M for each frequency f and time t. For this reason, it is conceivable to perform an average operation on the proximity frequency f and the proximity time t, instead of the average between channels as in (Equation 14).
- the suppression unit 503 receives the signal X 0 and the integrated suppression coefficient W S1 from the correlation noise removing unit 301 and removes residual noise included in X 0 .
- the amplitude component is the amplitude component of the desired audio signal
- the phase component is the phase component of the signal X 0 from the correlation noise removing unit 301.
- FIG. 6 is a flowchart for explaining the noise removal method according to the present embodiment.
- step S601 correlated noise components are removed using input signals input from a plurality of channels to obtain one output signal.
- M 2 is set for simplicity
- Nc1 and Nc2 are eliminated in (Expression 1-1) and (Expression 1-2), and S is solved. Since Nc1 and Nc2 have a correlation, Nc2 can be written as Nc1. Ni1 and Ni2 are irrelevant and remain in the output.
- a suppression coefficient for suppressing the noise remaining in the output signal obtained in step S601 is calculated using the phase component of the output signal and the phase component of the input signal.
- step S605 an integrated suppression coefficient is obtained by using the average of the suppression coefficients.
- step S607 residual noise is removed using the integrated suppression coefficient.
- the correlated noise removing unit 301 removes the noise component having correlation between channels, get X 0.
- X 0 has a low correlation with the noise components included in the multi-channel input signals X 1 to X M except for audio components. Therefore, residual noise can be removed by obtaining a noise suppression coefficient based on the phase difference of X 0 and the phase difference of at least one signal among X 1 to X M.
- Equation 15 it is possible to remove only the noise component without removing the desired audio component.
- a signal processing apparatus according to a third embodiment of the present invention will be described with reference to FIGS.
- the signal processing apparatus according to the present embodiment is the same as the signal processing apparatus according to the second embodiment of FIG. 3 except that the residual noise removal unit 302 of FIG. 3 is the residual noise removal unit 702 shown in FIG. . Therefore, only the residual noise removing unit 702 will be described.
- FIG. 7 shows the configuration of the residual noise removing unit 702.
- the residual noise removing unit 702 includes correction units 722 1 to 722 M and a noise removing unit 421 based on a phase difference.
- the noise removal unit 421 based on the phase difference has the same operation as the noise removal unit based on the phase difference shown in FIG.
- Correction units 722 1 to 722 M receive the multi-channel input signals X 1 to X M , respectively, correct the input signals, and output them.
- Input signals X 1 to X M are not (Formula 1-1) to (Formula 1-M), but the following (Formula 16-1) It is assumed that (Formula 16-M).
- G 1 to G M are frequency responses to audio components included in channels 1 to M, respectively, and are complex spectra.
- the output signal X 0 of the correlation noise removing unit 301 is not (Expression 2) but the following (Expression 17).
- G 0 is the frequency response for sound component, a complex spectrum.
- the correction units 722 1 to 722 M perform the following (Expression 18-1) to (Expression) so that the sound component in (Expression 16-1) to (Expression 16-M) becomes equal to the sound component of (Expression 17), respectively. Correction is performed using correction coefficients Q 1 to Q M shown in 18-M). In other words, the input signals X 1 to X M are multiplied by the correction coefficients Q 1 to Q M. If you put it as follows, (Equation 19-1) to (Equation 19-M) and (Equation 17) can be rewritten as follows.
- the correction coefficients Q 1 to Q M shown in (Equation 18-1) to (Equation 18-M) are, for example, the arrangement of microphones for acquiring multi-channel input signals X 1 to X M Depending on the processing contents of the position and correlation noise removing unit 301, it can be determined in advance.
- the correction coefficients Q 1 to Q M can be calculated using X 0 , the uncorrected multi-channel signals X 1 to X M , and the corrected multi-channel signals X ′ 1 to X ′ M. is there. Since the operations for channels 1 to M are the same, only the case of channel 1 is illustrated in FIG.
- FIG. 8 shows a correction coefficient calculation unit 801 and a correction unit 802 for channel 1.
- the correction unit 802 is the same as the correction unit 722 1 except that the correction coefficient Q 1 is exchanged with the correction coefficient calculation unit 801.
- the correction coefficient calculation unit 801 updates the correction coefficient Q 1 so that the error between X 0 and X ′ 1 is minimized.
- X 0 and X ′ 1 are highly correlated only with the audio component contained in both signals. Therefore, an LMS (Least Mean Square) method, a normalized LMS method, or the like used when updating the adaptive filter may be used for the update.
- ⁇ is a step size parameter for adjusting the degree of update.
- the correction units 722 1 to 722 M correct the multichannel input signals X 1 to X M.
- the residual noise removing unit 302 can remove the residual noise component included in X 0. That is, the signal processing apparatus according to the present embodiment can remove only the noise component without removing the desired audio component.
- a signal processing apparatus according to the fourth embodiment of the present invention will be described with reference to FIGS.
- the signal processing apparatus according to the present embodiment is the same as the signal processing apparatus according to the second embodiment except that the residual noise removing unit 302 in FIG. 3 is replaced with a residual noise removing unit 902 shown in FIG. Therefore, only the residual noise removing unit 902 will be described.
- FIG. 9 shows the configuration of the residual noise removing unit 902.
- Residual noise removal unit 902 includes correction units 922 1 to 922 M , noise removal unit 421 based on phase difference, and re-noise removal unit 923.
- the correcting units 922 1 to 922 M operate in the same manner as the correcting units 722 1 to 722 M shown in FIG. 7, and the noise removing unit 421 based on the phase difference is a noise removing unit 421 based on the phase difference shown in FIG. Since the same operation is performed, the description is omitted.
- the re-noise removing unit 923 receives the output signal X 0 of the correlation noise removing unit and the output signal S 1 of the noise removing unit based on the phase difference from which the residual noise contained in X 0 is removed, and the residual signal contained in X 0 Remove the noise again.
- FIG. 10 shows the configuration of the re-noise removal unit 923.
- the re-noise removal unit 923 includes power calculation units 1001 and 1002, a residual noise estimation unit 1003, a re-suppression coefficient calculation unit 1004, and a suppression unit 1005.
- the power calculators 1001 and 1002 calculate and output the power of X 0 and S 1 , respectively. That is, the following X 0P and S 1P are output, respectively.
- the residual noise estimation unit 1003 estimates the residual noise power using X 0P and S 1P and outputs the estimated noise power. That is, the following N 0P is output. However, max [] is an operator for obtaining the maximum value.
- the resuppression coefficient calculator 1004 calculates and outputs a resuppression coefficient W S0 using X 0P , S 1P , and N 0P .
- ⁇ DD is the prior SNR
- ⁇ DD may be calculated as follows. However, It is. By separately calculating the denominator and numerator of (Expression 32) using past signals as shown in (Expression 33) and (Expression 34), the value of ⁇ DD becomes more stable.
- S 1P and S 1PDD in (Equation 31) to (Equation 34) may be corrected by a pattern (model) of a desired signal (for example, sound) using the method of Patent Document 3: Japanese Patent No. 4765461. Is possible.
- the resuppression coefficient W S0 may be calculated as follows instead of (Equation 30). Where ⁇ is the posterior SNR, It is. By using the current X0P for the calculation of the re-suppression coefficient, the suppression accuracy is improved at the rising edge of the audio signal. Of course, N 0PDD of (Expression 34) may be used for N 0P of the denominator on the right side of (Expression 36).
- the MMSE STSA (Minimum Mean Square Error Short Time Spectral Amplitude) method, MMSE LSA (Minimum Mean Square Error Log Spectral Amplitude) method, etc. (Equation 30) and (Equation 35) are used to calculate the resuppression coefficient. Of course, you may.
- (Repression part) Suppression section 1005 receives signal X 0 and re-suppression coefficient W S0 from correlation noise removal section 301 and removes residual noise included in X 0 . Then, the signal S 0 is outputted.
- the re-suppression coefficient is corrected by a desired signal pattern (model) that calculates a re-suppression coefficient by combining past signals as in (Equation 31), (Equation 33), and (Equation 34).
- the current X0P is used to calculate the resuppression coefficient.
- a signal processing device according to a fifth embodiment of the present invention will be described with reference to FIGS.
- the signal processing apparatus according to the present embodiment is the same as the signal processing apparatus according to the second embodiment except that the residual noise removing unit 302 in FIG. 3 is replaced with a residual noise removing unit 1102 shown in FIG. Therefore, only the residual noise removing unit 1102 will be described.
- FIG. 11 shows the configuration of the residual noise removing unit 1102.
- Residual noise removal unit 1102 includes correction units 722 1 to 722 M , noise removal unit 421 based on phase difference, re-noise removal unit 923, and noise removal unit 1121 based on amplitude.
- the correction units 722 1 to 722 M perform the same operations as those of the correction unit described with reference to FIG.
- the noise removal unit 421 based on the phase difference performs the same operation as the noise removal unit based on the phase difference shown in FIG.
- the re-noise removing unit 923 performs the same operation as the re-noise removing unit shown in FIG.
- the noise removal unit 1121 based on amplitude receives at least the output signal S 1 of the noise removal unit 421 based on the phase difference, removes residual noise included in S 1 , and outputs S 2 .
- FIG. 12 shows the configuration of the noise removal unit 1121 based on amplitude.
- the amplitude-based noise removal unit 1121 includes a power calculation unit 1201, an amplitude-based noise estimation unit 1202, an amplitude-based suppression coefficient calculation unit 1203, and a suppression unit 1204.
- Power calculator Power calculating unit 1201 calculates the power of S 1, and outputs. That is, the following S 1P is output.
- Noise estimation unit 1202 based on amplitude estimates and outputs the power of residual noise included in S 1P using at least S 1P . That is, the following N 1P is output.
- NE [] is a noise power estimation operator, and various noise power estimation methods such as the minimum statistical method and the weighted noise estimation method disclosed in Japanese Patent No. 4282227 can be used. .
- the amplitude-based suppression coefficient calculation unit 1203 calculates and outputs the amplitude-based suppression coefficient W S2 using S 1P and N 1P .
- ⁇ DD is the prior SNR
- ⁇ DD may be calculated as follows. However, It is. By separately calculating the denominator and numerator of (Expression 42) using past signals as shown in (Expression 43) and (Expression 44), the value of ⁇ DD becomes more stable.
- the suppression coefficient W S2 based on the amplitude may be calculated as follows instead of (Equation 40). Where ⁇ is the posterior SNR, It is. By using the current S 1P for the calculation of the suppression coefficient based on the amplitude, the suppression accuracy is improved at the rising edge of the audio signal. (Equation 46) Of course, N 1PDD of (Equation 44) may be used for N 1P of the denominator on the right side.
- the suppression unit 1204 receives the signal S 1 from the noise removal unit 421 based on the phase difference and the suppression coefficient W S2 based on the amplitude, and removes residual noise contained in S 1 . Then, the signal S 2 is outputted.
- the noise removal unit 1121 based on the amplitude is used not in the subsequent stage of the re-noise removal unit 923 but in the previous stage.
- the noise removing unit 421 based on the phase difference (Equation 9)
- E S1 and E N1 shown in (Equation 10) is not zero, it can be removed without removing the desired audio components, only a higher accuracy noise component.
- the present invention may be applied to a system composed of a plurality of devices, or may be applied to a single device. Furthermore, the present invention can also be applied to a case where a multi-channel noise removal program that implements the functions of the embodiments is supplied directly or remotely to a system or apparatus. Therefore, in order to realize the functions of the present invention on a computer, a program installed on the computer, a medium storing the program, and a WWW (World Wide Web) server that downloads the program are also included in the scope of the present invention. . In particular, at least a non-transitory computer readable medium storing a program for causing a computer to execute the processing steps included in the above-described embodiments is included in the scope of the present invention.
- Correlated noise removing means for inputting at least two input signals in which a desired signal and a noise signal are mixed from at least two channels and removing a noise signal having a correlation between the at least two input signals; Residual noise removal for removing residual noise contained in the output signal of the correlation noise removal means based on the phase difference between the output signal of the correlation noise removal means and at least one input signal contained in the at least two input signals Means,
- a signal processing apparatus comprising: (Appendix 2) The signal processing apparatus according to appendix 1, wherein the residual noise removing unit includes a noise removing unit based on a phase difference.
- the noise removing means based on the phase difference is: Suppression coefficient calculating means for calculating a suppression coefficient based on a phase difference between the output signal of the correlation noise removing means and at least one of the input signals; Suppression coefficient integration means for receiving a suppression coefficient from at least one suppression coefficient calculation means and outputting an integrated suppression coefficient;
- the signal processing apparatus according to claim 2, further comprising: suppression means that suppresses residual noise contained in the output signal of the correlation noise removing means using an integrated suppression coefficient from the suppression coefficient integrating means.
- the signal processing apparatus includes a re-noise removing unit downstream of the noise removing unit based on the phase difference.
- the re-noise removing means includes Residual noise estimating means for estimating the power of residual noise from the output signal power of the correlation noise removing means and the output signal power of the noise removing means based on the phase difference;
- a resuppression coefficient calculating means for calculating a resuppression coefficient using the power of the output signal of the correlation noise removing means and the output signal power of the noise removing means based on the phase difference and the power of the estimated residual noise;
- Suppression means for suppressing residual noise contained in the output signal of the correlation noise removing means using the resuppression coefficient from the resuppression coefficient calculating means;
- the signal processing apparatus including (Appendix 7) The signal processing apparatus according to appendix 5, wherein the residual noise removing
- the noise removal means based on the amplitude is: An amplitude-based noise estimation means for estimating the power of noise included in the output signal of the noise removal means based on the phase difference; A suppression coefficient calculation means based on amplitude for calculating a suppression coefficient based on amplitude using the power of the output signal of the noise removal means based on the phase difference and the estimated noise power from the noise estimation means based on the amplitude;
- a signal processing method including: (Appendix 10) A correlation noise removing step of inputting at least two input signals in which a desired signal and a noise signal are mixed from at least two channels, and removing a noise signal having a correlation between the at least two input signals; Residual noise removal for removing residual noise included in the output signal in the correlation noise removal step based on a phase difference between the output signal of the correlation noise removal step and at least one input signal included in the at least two input signals Steps,
- a signal processing program for causing a computer to execute.
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Abstract
Description
少なくとも2つのチャンネルから、所望信号と雑音信号が混在する少なくとも2つの入力信号を入力し、前記少なくとも2つの入力信号の間で相関を有する雑音信号を除去する相関雑音除去手段と、
前記相関雑音除去手段の出力信号と前記少なくとも2つの入力信号に含まれる少なくとも1つの入力信号との位相差に基づいて、前記相関雑音除去手段の出力信号に含まれる残留雑音を除去する残留雑音除去手段と、
を備えた。
前記相関雑音除去ステップの出力信号と前記少なくとも2つの入力信号に含まれる少なくとも1つの入力信号との位相差に基づいて、前記相関雑音除去ステップによる出力信号に含まれる残留雑音を除去する残留雑音除去ステップと、
を含む。
前記相関雑音除去ステップの出力信号と前記少なくとも2つの入力信号に含まれる少なくとも1つの入力信号との位相差に基づいて、前記相関雑音除去ステップによる出力信号に含まれる残留雑音を除去する残留雑音除去ステップと、
をコンピュータに実行させる。
本発明の第1実施形態としての信号処理装置100について、図1と図2を用いて説明する。図1に示すように、信号処理装置100は、相関雑音除去部101と、残留雑音除去部102とを含む。また図2に示すように、残留雑音除去部102は、抑圧係数算出部2011~201Mと、抑圧部202とを含む。
次に、図3乃至図6を参照して、本発明の第2実施形態に係る信号処理装置300について説明する。なお図6は、本実施形態の信号処理装置の処理を示すフローチャートである。
図3は、本実施形態に係る信号処理装置300の構成を示す図である。本実施形態において、信号処理装置300は所望の信号と雑音とが混在する多チャンネルの混在信号から所望の信号を取得するシステムである。以降では所望の信号を音声信号として説明するが、本発明の技術範囲をそれらのみに限定する趣旨のものではない。
多チャンネルの入力信号X1~XMを次のようにモデル化する。
X1~XMは入力信号の複素スペクトルであり、対応するチャンネルの時間領域の信号に対して離散フーリエ変換などの周波数分析を行うことにより得られる。fは周波数のインデックス、tは時間のインデックスである。以降、fとtは必要な場合を除いて適宜省略する。Sは所望の音声成分の複素スペクトルである。Nc1~NcMは各チャンネル1~Mのうち2以上の複数のチャンネルに含まれる雑音成分、すなわちチャンネル間で相関を有する雑音成分の複素スペクトルである。Ni1~NiMは各チャンネル1~Mそれぞれに独立して含まれる雑音成分、すなわちチャンネル間で相関が低い雑音成分の複素スペクトルである。
Ni0は相関雑音除去部301の処理後の残留雑音であり、主にチャンネル間で相関を有さない雑音成分である。なお、チャンネル間でのNc1~NcMの違い(位相差、振幅差)があらかじめわかっている場合には、ある特定の空間にヌルを向ける固定のビームフォーマなど適応動作が不要な方法を用いることも可能である。
図4は、残留雑音除去部302の構成を示す。残留雑音除去部302は、位相差に基づく雑音除去部421を含む。位相差に基づく雑音除去部421は、相関雑音除去部301の出力信号X0と、多チャンネル入力信号X1~XMのうち少なくとも1つの信号を入力とする。X0の位相と、X1~XMのうち少なくとも1つの信号の位相との差分(位相差)に基づいてX0に含まれる雑音成分を除去し、S1を出力する。S1はS0として残留雑音除去部302から出力される。
図5は、位相差に基づく雑音除去部421の構成を示す。位相差に基づく雑音除去部421は、抑圧係数算出部5011~501M、抑圧係数統合部502、抑圧部503を含む。
抑圧係数算出部5011~501Mは、それぞれ、相関雑音除去部301の出力信号X0と多チャンネル入力信号X1~XMを用いて抑圧係数W1~WMを算出する。チャンネル1~Mに対する動作は同じであるため、抑圧係数算出部5011について説明する。
抑圧係数算出部5011~501Mは、(式5)、(式13)で計算したW1、WMを出力する。なお、|S|と|X0|が共に正の数であること、また|S|≦|X0|の仮定から、W1~WMを0~1の範囲になるように制限をかけて出力してもよい。
抑圧係数統合部502は、抑圧係数算出部5011~501Mからの抑圧係数W1~WMを受けて、統合抑圧係数WS1を出力する。例えば、統合抑圧係数WS1を次のように求める。
ただし、Aveは平均演算子である。なお、平均演算は全ての抑圧係数W1~WMで行う必要はなく、全体の平均値からのずれが大きい抑圧係数を省いて、再度平均することも考えられる。また事前に定めた範囲の値をとるチャンネルの抑圧係数だけを用いて平均する、事前に決めたチャンネルの抑圧係数だけを用いて平均することも考えられる。さらに、平均せずに、事前に決めたチャンネルの抑圧係数を使用する、所望の音声成分が除去されないように抑圧係数W1~WMのうち値が最大となるチャンネルの抑圧係数を使用するなどが考えられる。
抑圧係数統合部502は、周波数f、時間tごとに、抑圧係数W1~WMを受ける。そのため、(式14)のようなチャンネル間だけの平均ではなく、近接周波数f、近接時間tに対して平均演算を行うことも考えられる。
抑圧部503は、相関雑音除去部301からの信号X0と統合抑圧係数WS1を受けて、X0に含まれる残留雑音を除去する。
次にステップS603において、ステップS601で求めた出力信号に残っている雑音を抑圧するための抑圧係数を、出力信号の位相成分と入力信号の位相成分を用いて算出する。
さらにステップS605では、抑圧係数の平均を用いることにより、統合抑圧係数を求める。
そして、ステップS607に進み、統合抑圧係数を用いて、残留雑音を除去する。
図7、図8を参照して、本発明の第3実施形態に係る信号処理装置について説明する。本実施形態に係る信号処理装置では、図3の残留雑音除去部302が図7で示す残留雑音除去部702である点を除き、図3の第2実施形態に係る信号処理装置と同じである。したがって、残留雑音除去部702についてだけ説明する。
補正部7221~722Mは、多チャンネル入力信号X1~XMをそれぞれ受けて、入力信号を補正し、出力する。入力信号X1~XMが、(式1-1)~(式1-M)ではなく、以下の(式16-1)
~(式16-M)であるとする。
ただし、G1~GMは、それぞれチャンネル1~Mに含まれる音声成分に対する周波数応答であり、複素スペクトルである。また、相関雑音除去部301の出力信号X0が、(式2)ではなく、以下の(式17)であるとする。
ただし、G0は音声成分に対する周波数応答であり、複素スペクトルである。補正部7221~722Mは、それぞれ(式16-1)~(式16-M)における音声成分が(式17)の音声成分と等しくなるように以下の(式18-1)~(式18-M)で示す補正係数Q1~QMを用いて補正する。
すなわち、補正係数Q1~QMを入力信号X1~XMに乗算する。
補正係数計算部801は、X0とX'1の誤差が最小になるように、補正係数Q1を更新する。X0とX'1は、両信号に含まれる音声成分だけの相関が高い。そのため、更新には適応フィルタを更新する際に用いられるLMS(Least Mean Square)法、正規化LMS法などを用いればよい。
図9、図10を参照して、本発明の第4実施形態に係る信号処理装置について説明する。本実施形態に係る信号処理装置では、図3の残留雑音除去部302が図9で示す残留雑音除去部902に置き換わる点を除き、第2実施形態に係る信号処理装置と同じである。したがって、残留雑音除去部902についてだけ説明する。
再雑音除去部923は、相関雑音除去部の出力信号X0とそのX0に含まれる残留雑音を除去した位相差に基づく雑音除去部の出力信号S1を受けて、X0に含まれる残留雑音を再度除去する。図10は、再雑音除去部923の構成を示す。再雑音除去部923は、パワー計算部1001、1002、残留雑音推定部1003、再抑圧係数計算部1004、抑圧部1005を含む。
残留雑音推定部1003は、X0PとS1Pを用いて、残留雑音のパワーを推定し、推定雑音パワーとして出力する。すなわち、以下のN0Pを出力する。
再抑圧係数計算部1004は、X0P、S1P、N0Pを用いて、再抑圧係数WS0を計算し、出力する。例えば、
図11、図12を参照して、本発明の第5実施形態に係る信号処理装置について説明する。本実施形態に係る信号処理装置では、図3の残留雑音除去部302を図11で示す残留雑音除去部1102に置き換えた点を除き、第2実施形態に係る信号処理装置と同じである。したがって、残留雑音除去部1102についてだけ説明する。
振幅に基づく雑音除去部1121は、位相差に基づく雑音除去部421の出力信号S1を少なくとも受けて、S1に含まれる残留雑音を除去し、S2を出力する。図12は、振幅に基づく雑音除去部1121の構成を示す。振幅に基づく雑音除去部1121は、パワー計算部1201、振幅に基づく雑音推定部1202、振幅に基づく抑圧係数計算部1203、抑圧部1204を含む。
振幅に基づく雑音推定部1202は、少なくともS1Pを用いて、S1Pに含まれる残留雑音のパワーを推定し、出力する。すなわち、以下のN1Pを出力する。
振幅に基づく抑圧係数計算部1203は、S1P、N1Pを用いて、振幅に基づく抑圧係数WS2を計算し、出力する。例えば、
と(式10)に示すES1とEN1がゼロでない場合であっても、所望の音声成分を除去せずに、雑音成分だけをさらに高精度に除去することができる。
以上、実施形態を参照して本願発明を説明したが、本願発明は上記実施形態に限定されるものではない。本願発明の構成や詳細には、本願発明のスコープ内で当業者が理解し得る様々な変更をすることができる。また、それぞれの実施形態に含まれる別々の特徴を如何様に組み合わせたシステムまたは装置も、本発明の範疇に含まれる。例えば、上記実施形態に記載した信号処理装置を含むマイクユニットも本発明の範疇に含まれる。
上記の実施形態の一部または全部は、以下の付記のようにも記載されうるが、以下には限られない。
(付記1)
少なくとも2つのチャンネルから、所望信号と雑音信号が混在する少なくとも2つの入力信号を入力し、前記少なくとも2つの入力信号の間で相関を有する雑音信号を除去する相関雑音除去手段と、
前記相関雑音除去手段の出力信号と前記少なくとも2つの入力信号に含まれる少なくとも1つの入力信号との位相差に基づいて、前記相関雑音除去手段の出力信号に含まれる残留雑音を除去する残留雑音除去手段と、
を備えた信号処理装置。
(付記2)
前記残留雑音除去手段が、位相差に基づく雑音除去手段を有する付記1に記載の信号処理装置。
(付記3)
前記位相差に基づく雑音除去手段は、
前記相関雑音除去手段の出力信号と少なくとも1つの前記入力信号の位相差に基づいて抑圧係数を算出する抑圧係数算出手段と、
少なくとも1つの前記抑圧係数算出手段からの抑圧係数を受けて統合抑圧係数を出力する抑圧係数統合手段と、
抑圧係数統合手段からの統合抑圧係数を用いて、前記相関雑音除去手段の出力信号に含まれる残留雑音を抑圧する抑圧手段と、を含む付記2に記載の信号処理装置。
(付記4)
前記残留雑音除去手段は、前記位相差に基づく雑音除去手段の前段に、各チャンネルの前記入力信号を補正する補正手段を有する付記2または3に記載の信号処理装置。
(付記5)
前記残留雑音除去手段は、前記位相差に基づく雑音除去手段の後段に、再雑音除去手段を有する付記2乃至4のいずれか1項に記載の信号処理装置。
(付記6)
前記再雑音除去手段は、
前記相関雑音除去手段の出力信号のパワーと前記位相差に基づく雑音除去手段の出力信号のパワーから残留雑音のパワーを推定する残留雑音推定手段と、
前記相関雑音除去手段の出力信号のパワーと前記位相差に基づく雑音除去手段の出力信号のパワーと前記推定した残留雑音のパワーを用いて再抑圧係数を計算する再抑圧係数計算手段と、
前記再抑圧係数計算手段からの再抑圧係数を用いて、前記相関雑音除去手段の出力信号に含まれる残留雑音を抑圧する抑圧手段と、
を含む付記5に記載の信号処理装置。
(付記7)
前記残留雑音除去手段は、前記位相差に基づく雑音除去手段の後段、かつ前記再雑音除去手段の前段に、振幅に基づく雑音除去手段を有する付記5に記載の信号処理装置。
(付記8)
前記振幅に基づく雑音除去手段は、
前記位相差に基づく雑音除去手段の出力信号に含まれる雑音のパワーを推定する、振幅に基づく雑音推定手段と、
前記位相差に基づく雑音除去手段の出力信号のパワーと振幅に基づく雑音推定手段からの推定雑音パワーを用いて振幅に基づく抑圧係数を計算する振幅に基づく抑圧係数計算手段と、
前記振幅に基づく抑圧係数計算手段からの振幅に基づく抑圧係数を用いて、前記位相差に基づく雑音除去手段の出力信号に含まれる雑音を抑圧する抑圧手段と、を備えた付記7に記載の信号処理装置(付記9)
少なくとも2つのチャンネルから、所望信号と雑音信号が混在する少なくとも2つの入力信号を入力し、前記少なくとも2つの入力信号の間で相関を有する雑音信号を除去する相関雑音除去ステップと、
前記相関雑音除去ステップの出力信号と前記少なくとも2つの入力信号に含まれる少なくとも1つの入力信号との位相差に基づいて、前記相関雑音除去ステップによる出力信号に含まれる残留雑音を除去する残留雑音除去ステップと、
を含む信号処理方法。
(付記10)
少なくとも2つのチャンネルから、所望信号と雑音信号が混在する少なくとも2つの入力信号を入力し、前記少なくとも2つの入力信号の間で相関を有する雑音信号を除去する相関雑音除去ステップと、
前記相関雑音除去ステップの出力信号と前記少なくとも2つの入力信号に含まれる少なくとも1つの入力信号との位相差に基づいて、前記相関雑音除去ステップによる出力信号に含まれる残留雑音を除去する残留雑音除去ステップと、
をコンピュータに実行させる信号処理プログラム。
Claims (10)
- 少なくとも2つのチャンネルから、所望信号と雑音信号が混在する少なくとも2つの入力信号を入力し、前記少なくとも2つの入力信号の間で相関を有する雑音信号を除去する相関雑音除去手段と、
前記相関雑音除去手段の出力信号と前記少なくとも2つの入力信号に含まれる少なくとも1つの入力信号との位相差に基づいて、前記相関雑音除去手段の出力信号に含まれる残留雑音を除去する残留雑音除去手段と、
を備えた信号処理装置。 - 前記残留雑音除去手段が、位相差に基づく雑音除去手段を有する請求項1に記載の信号処理装置。
- 前記位相差に基づく雑音除去手段は、
前記相関雑音除去手段の出力信号と少なくとも1つの前記入力信号の位相差に基づいて抑圧係数を算出する抑圧係数算出手段と、
少なくとも1つの前記抑圧係数算出手段からの抑圧係数を受けて統合抑圧係数を出力する抑圧係数統合手段と、
抑圧係数統合手段からの統合抑圧係数を用いて、前記相関雑音除去手段の出力信号に含まれる残留雑音を抑圧する抑圧手段と、を含む請求項2に記載の信号処理装置。 - 前記残留雑音除去手段は、前記位相差に基づく雑音除去手段の前段に、各チャンネルの前記入力信号を補正する補正手段を有する請求項2または3に記載の信号処理装置。
- 前記残留雑音除去手段は、前記位相差に基づく雑音除去手段の後段に、再雑音除去手段を有する請求項2乃至4のいずれか1項に記載の信号処理装置。
- 前記再雑音除去手段は、
前記相関雑音除去手段の出力信号のパワーと前記位相差に基づく雑音除去手段の出力信号のパワーから残留雑音のパワーを推定する残留雑音推定手段と、
前記相関雑音除去手段の出力信号のパワーと前記位相差に基づく雑音除去手段の出力信号のパワーと前記推定した残留雑音のパワーを用いて再抑圧係数を計算する再抑圧係数計算手段と、
前記再抑圧係数計算手段からの再抑圧係数を用いて、前記相関雑音除去手段の出力信号に含まれる残留雑音を抑圧する抑圧手段と、
を含む請求項5に記載の信号処理装置。 - 前記残留雑音除去手段は、前記位相差に基づく雑音除去手段の後段、かつ前記再雑音除去手段の前段に、振幅に基づく雑音除去手段を有する請求項5に記載の信号処理装置。
- 前記振幅に基づく雑音除去手段は、
前記位相差に基づく雑音除去手段の出力信号に含まれる雑音のパワーを推定する、振幅に基づく雑音推定手段と、
前記位相差に基づく雑音除去手段の出力信号のパワーと振幅に基づく雑音推定手段からの推定雑音パワーを用いて振幅に基づく抑圧係数を計算する振幅に基づく抑圧係数計算手段と、
前記振幅に基づく抑圧係数計算手段からの振幅に基づく抑圧係数を用いて、前記位相差に基づく雑音除去手段の出力信号に含まれる雑音を抑圧する抑圧手段と、を備えた請求項7に記載の信号処理装置。 - 少なくとも2つのチャンネルから、所望信号と雑音信号が混在する少なくとも2つの入力信号を入力し、前記少なくとも2つの入力信号の間で相関を有する雑音信号を除去する相関雑音除去ステップと、
前記相関雑音除去ステップの出力信号と前記少なくとも2つの入力信号に含まれる少なくとも1つの入力信号との位相差に基づいて、前記相関雑音除去ステップによる出力信号に含まれる残留雑音を除去する残留雑音除去ステップと、
を含む信号処理方法。 - 少なくとも2つのチャンネルから、所望信号と雑音信号が混在する少なくとも2つの入力信号を入力し、前記少なくとも2つの入力信号の間で相関を有する雑音信号を除去する相関雑音除去ステップと、
前記相関雑音除去ステップの出力信号と前記少なくとも2つの入力信号に含まれる少なくとも1つの入力信号との位相差に基づいて、前記相関雑音除去ステップによる出力信号に含まれる残留雑音を除去する残留雑音除去ステップと、
をコンピュータに実行させる信号処理プログラム。
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JPWO2015141103A1 (ja) | 2017-04-06 |
US10043532B2 (en) | 2018-08-07 |
JP6432597B2 (ja) | 2018-12-05 |
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