WO1999050825A1 - Noise reduction device and a noise reduction method - Google Patents

Noise reduction device and a noise reduction method Download PDF

Info

Publication number
WO1999050825A1
WO1999050825A1 PCT/JP1998/005512 JP9805512W WO9950825A1 WO 1999050825 A1 WO1999050825 A1 WO 1999050825A1 JP 9805512 W JP9805512 W JP 9805512W WO 9950825 A1 WO9950825 A1 WO 9950825A1
Authority
WO
WIPO (PCT)
Prior art keywords
noise
amplitude
amplitude spectrum
output
power
Prior art date
Application number
PCT/JP1998/005512
Other languages
French (fr)
Japanese (ja)
Inventor
Kazutaka Tomita
Original Assignee
Mitsubishi Denki Kabushiki Kaisha
Priority date (The priority date is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the date listed.)
Filing date
Publication date
Application filed by Mitsubishi Denki Kabushiki Kaisha filed Critical Mitsubishi Denki Kabushiki Kaisha
Priority to CA002291826A priority Critical patent/CA2291826A1/en
Priority to AU13525/99A priority patent/AU721270B2/en
Priority to EP98957196A priority patent/EP0992978A4/en
Publication of WO1999050825A1 publication Critical patent/WO1999050825A1/en

Links

Classifications

    • GPHYSICS
    • G10MUSICAL INSTRUMENTS; ACOUSTICS
    • G10LSPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L21/00Speech 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
    • G10L21/02Speech enhancement, e.g. noise reduction or echo cancellation
    • G10L21/0208Noise filtering
    • GPHYSICS
    • G10MUSICAL INSTRUMENTS; ACOUSTICS
    • G10LSPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L21/00Speech 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
    • G10L21/02Speech enhancement, e.g. noise reduction or echo cancellation
    • G10L21/0208Noise filtering
    • G10L21/0216Noise filtering characterised by the method used for estimating noise
    • G10L21/0232Processing in the frequency domain
    • GPHYSICS
    • G10MUSICAL INSTRUMENTS; ACOUSTICS
    • G10LSPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L25/00Speech or voice analysis techniques not restricted to a single one of groups G10L15/00 - G10L21/00
    • G10L25/78Detection of presence or absence of voice signals
    • G10L25/84Detection of presence or absence of voice signals for discriminating voice from noise

Definitions

  • the present invention is applied to a voice communication system or a voice recognition system used in an environment having background noise, and a noise reduction method for a voice signal that performs noise suppression by removing noise from an input voice signal and a noise section. It relates to a noise reduction method using amplitude suppression. Background art
  • a method shown in FIG. 15 below is known as a means for reducing background noise. That is, the input signal in which the background noise is superimposed on the voice is converted into an AZD (analog Z-digital) signal and divided into a fixed section (frame). (Hereinafter referred to as a frame signal) is converted to a frequency spectrum by a discrete Fourier transform, and then divided into an amplitude spectrum and a phase spectrum.
  • AZD analog Z-digital
  • the audio amplitude spectrum with reduced background noise Is estimated as the frequency spectrum of the voice with the background noise reduced along with the previous phase spectrum, and the frequency spectrum of this voice is inversely dispersive Fourier transformed and the output signal is obtained.
  • a scan Bae click Tsentralnyi sub tiger click Chillon becomes how obtained. This is S TEVEN F. BAL L "S upprssionofacusticnoi salspeechusingspectr alsubtraction," I EEE T rans. A coust., Speeshand Signal Proc., vol. AS SP-29, pp. 113-120, Apr. 1979.
  • FIG. 15 is a block diagram showing the configuration of this noise reduction method. First, the overall operation will be described with reference to FIG.
  • the input signal 107 cut out to a predetermined frame length is converted into the frequency domain by the Fourier transform unit 101, and the input phase spectrum 108 and the input amplitude spectrum 109 are output. You.
  • the input signal is determined by the noise section determination unit 102 to be a speech frame (section) if it is equal to or more than the threshold value TH, and a noise frame (section) otherwise.
  • the estimated noise amplitude spectrum calculation unit 103 weights and adds the input amplitude spectrum 109 at that time to the previous estimated noise amplitude spectrum. And outputs the estimated noise amplitude spectrum 110 updated.
  • the subtraction 'filter section 104 has a characteristic whose representative transfer function is represented by the following equation (1).
  • Equation (1) S ( ⁇ ) is the amplitude spectrum term of the input signal, ⁇ ( ⁇ ( ⁇ )) is the estimated noise amplitude spectrum term, and r is a constant representing the removal rate of the estimated noise amplitude spectrum. Increasing r increases the amount of noise suppression; conversely, decreasing r decreases the amount of noise suppression.
  • the subtraction 'filter section 104 calculates the removal rate r based on a predetermined equation.
  • the input amplitude spectrum and the estimated noise amplitude spectrum are subtracted in the same manner as in equation (1), and the difference output amplitude spectrum 111 is obtained. Output.
  • the subtraction / filter unit 104 reduces the removal rate for noise frames and low power frames such as consonants to reduce noise suppression.
  • a method in order to enhance the noise reduction effect, a method is used in which a function called a window function is defined on the time axis of the signal to be subjected to Fourier transform and multiplied and weighted.
  • a frame signal and a continuous signal with a certain length are multiplied by a window function and subjected to Fourier transform.
  • the output frame signal of the inverse Fourier transform is added to the first weighted sample below 1
  • the method of adding the overlap signal of the output of the inverse Fourier transform of the frame and outputting it is: North America CDMA mobile phone system (TIA / EIA, IS-127, "Enhanced Variable Rate Code Service Option 3 for Wideband S pread Spectrum Digital Systems") I have.
  • Figure 16 shows the window function used here.
  • a window function is used so that the sum of the weights of the samples corresponding to the addition of the frame signal and the overlap signal becomes 1, and that both ends of the Fourier transform target signal smoothly approach 0.
  • a third conventional spectral subtraction method there is a method of performing a Fourier transform on a frame signal and a signal of a fixed length (overlap signal) following the frame signal without weighting by a window function.
  • it is used in the noise reduction device disclosed in Japanese Patent Application Laid-Open No. Hei 9-134449.
  • a waveform shaping process is performed in which the overlap signal of the inverse Fourier transform output of the immediately preceding frame is superimposed on the triangular window with the frame signal of the inverse Fourier transform output. And outputs the same, and saves the inverse Fourier transform output overlap signal for overlap processing of the next frame. Equation (2) shows the waveform shaping process at this time.
  • fixed-point DSP 16-bit fixed-point digital signal processor
  • the operation precision of the fixed-point decimal point D SP can be increased by double-precision arithmetic operation or carry by shift.
  • the amount of calculation for this increases.
  • the Fourier transform and the inverse Fourier transform basically require a large amount of computation, and there are many cases where it is desired to realize the computation with a small amount of computation at the expense of a little computation accuracy.
  • this method the frame range to be Fourier-transformed is not weighted by the window function, so the dynamic range is not widened, and the accuracy of the output frame signal that has undergone limited Fourier transform and inverse Fourier transform is degraded. Less is. For this reason, a fixed-point DSP that requires a small amount of computation is used.
  • the fourth conventional noise reduction method using spectral subtraction is
  • the estimated noise amplitude spectrum is generally calculated by averaging the input signals of a plurality of frames in the section determined to be noise, that is, the average of the amplitude spectrum of the noise signal. For example, the spectral sub in Japanese Patent Application Laid-Open No. This method is used in fractionation.
  • a method of determining the state of the current frame from a predetermined finite state representing the state of the signal and determining the state of the current frame when the state corresponds to the state of a noise frame is disclosed in Japanese Unexamined Patent Publication No. — Indicated by 3 8 4 5 4.
  • the first conventional spectral subtraction method has a drawback that a noise section is deformed and becomes unpleasant residual noise called musical noise, which is the biggest practical problem.
  • the output amplitude spectrum from which the estimated noise amplitude spectrum has been subtracted has a shape in which power is concentrated in a part of the frequency, and the frequency in which the power is concentrated is not per frame. It changes by changing the rules.
  • a method of solving musical noise a method of varying the removal rate of the estimated noise amplitude spectrum is used, and the removal amount may be reduced in a noise frame, but the noise suppression amount is insufficient in a noise section. There is.
  • the noise reduction ability is improved by performing a Fourier transform on a weighted signal whose both ends approach 0 using a window function.
  • the accuracy of the signals at both ends is reduced when realizing with fixed-point DSP, so that both ends of the output signal of the inverse Fourier transform are degraded and abnormal noise occurs at frame boundaries.
  • the unweighted signal is Fourier-transformed as in the third conventional example, accuracy can be easily achieved when implementing it with a fixed-point DSP, but the noise amplitude spectrum included in the input amplitude spectrum can be improved. Since the variation of the torque component between frames becomes larger than when weighted, the noise reduction ability is inferior to that when weighted.
  • the noise section determination threshold is set high so that the noise frame can be correctly determined when the power fluctuation of the noise is large, some of the speech frames may also be noise frames. There is a problem in that the speech noise leaks into the estimated noise amplitude spectrum, causing the speech to be cut off.
  • the noise interval determination threshold is set low so that the audio frame is not determined to be a noise frame, the noise frame is determined to be a voice frame, and the estimated noise amplitude spectrum is updated.
  • the noise reduction is not performed correctly, and as a result, the noise reduction ability is deteriorated.
  • the frame state of the current frame is determined from a predetermined finite state representing the state of the signal, and this frame state is a noise frame state.
  • this frame state is a noise frame state.
  • the present invention has been made in order to solve the above-mentioned problem, and unpleasant residuals are obtained even when noise frames frequently move to or from noise frames and the spectrum of each frame is biased to some frequencies.
  • the aim is to obtain a noise reduction method with less noise. Disclosure of the invention
  • the noise reduction apparatus subtracts an estimated noise amplitude spectrum from an amplitude spectrum obtained by orthogonally transforming an input signal cut out to a predetermined frame length by a spectral subtraction filter.
  • Noise reduction device to obtain output
  • the required power is calculated by multiplying the reduced power by the spectral subtraction filter by an amplitude adjustment coefficient determined by the power of the amplitude spectrum and the power of the estimated noise amplitude spectrum.
  • An amplitude adjustment filter unit for obtaining an output is provided.
  • the spectral subtraction 'filter makes the subtraction rate of the subtraction of the estimated noise amplitude spectrum variable according to the estimated noise amplitude spectrum.
  • the amplitude adjustment filter unit is characterized in that the amplitude adjustment coefficient is made variable in accordance with the withdrawal rate.
  • the amplitude adjustment filter unit When the subtraction rate is high, the amplitude adjustment filter unit increases the amplitude adjustment coefficient to enhance noise suppression in the voice section and increases the output value of the output signal. When the rate is small, the amplitude adjustment coefficient is reduced, thereby reducing noise suppression in a noise section and reducing the output value of the output signal. It is characterized by multiplying the subtraction output by the central subtraction filter with the time-domain amplitude spectrum subjected to the inverse orthogonal transform.
  • the amplitude adjustment filter section is characterized in that the amplitude spectrum is multiplied by the amplitude adjustment coefficient in the frequency domain for each frame, and an inverse orthogonal transform is performed to obtain an output.
  • the amplitude adjustment coefficient is obtained by adding the amplitude adjustment coefficient of the current frame obtained based on the difference between the power of the amplitude spectrum of the input signal of the current frame and the power of the estimated noise amplitude spectrum of the current frame to the previous frame. It is characterized in that the value of the amplitude adjustment coefficient obtained in (1) is weighted and added.
  • a noise reduction apparatus provides an estimated noise amplitude spectrum from an amplitude spectrum obtained by orthogonally transforming an input signal cut out to a predetermined frame length.
  • a noise reduction device that obtains an output by subtracting with a spectral subtraction 'filter
  • An input signal generation unit that cuts out a set section before and after the current frame and multiplies by a weighting function of 1 or less and adds the input signal before and after the current frame so that the end is close to 0, and outputs the output of the input signal generation unit Is used as an input signal to obtain an amplitude spectrum.
  • a noise reduction apparatus includes a waveform shaping processing unit that shapes a waveform of a current frame using a set section after a previous frame multiplied by one weighting function.
  • the noise reduction apparatus subtracts an estimated noise amplitude spectrum from an amplitude spectrum obtained by orthogonally transforming an input signal cut out to a predetermined frame length by a spectral subtraction filter.
  • a noise reduction device that obtains output
  • a subtraction rate calculation unit that obtains an average noise power from a plurality of frames of the estimated noise amplitude spectrum and compares the average noise power with a determination threshold of the noise frame to obtain a removal rate
  • the subtraction rate used for the subtraction of the spectral subtraction filter is set to the above-described subtraction rate.
  • the noise reduction apparatus includes an average noise power calculation unit that calculates an average noise power from a plurality of frames of the estimated noise amplitude spectrum, and the amplitude adjustment filter unit includes a spectral sub-frame for each frame.
  • the required output is obtained by multiplying the subtraction output by the task filter by an amplitude adjustment coefficient determined by the power of the amplitude spectrum and the average noise amplitude power obtained above.
  • an estimated noise amplitude spectrum is obtained from an amplitude spectrum obtained by orthogonally transforming an input signal cut out to a predetermined frame length.
  • the required power is calculated by multiplying the reduced power by the above-mentioned spectral subtraction filter for each frame by an amplitude adjustment coefficient determined by the power of the above-mentioned amplitude spectrum and the power of the above-mentioned estimated noise amplitude spectrum. It is characterized by a process for obtaining output.
  • the noise reduction method according to the present invention is characterized in that an estimated noise amplitude spectrum is subtracted from an amplitude spectrum obtained by orthogonally transforming an input signal cut out to a predetermined frame length by a spectral subtraction 'filter.
  • an estimated noise amplitude spectrum is subtracted from an amplitude spectrum obtained by orthogonally transforming an input signal cut out to a predetermined frame length by a spectral subtraction 'filter.
  • FIG. 1 is a block diagram showing a configuration of a spectral subtraction method according to Embodiment 1 of the present invention.
  • FIG. 2 is a flowchart showing the operation performed by the amplitude adjustment filter in the noise reduction method according to the first embodiment.
  • FIG. 3 is a diagram illustrating an operation performed by the noise section determination unit according to the first embodiment.
  • FIG. 4 is a diagram for explaining the operation performed by the amplitude adjustment filter according to the first embodiment. is there.
  • FIG. 5 is a diagram illustrating an output signal of the spectral subtraction method according to the first embodiment.
  • FIG. 6 is a block diagram showing a configuration of a spectral subtraction method according to Embodiment 2 of the present invention.
  • FIG. 7 is a diagram showing a configuration of a spectral subtraction method according to Embodiment 3 of the present invention.
  • FIG. 8 is a diagram illustrating an example of an input signal in the noise reduction method according to the third embodiment.
  • FIG. 9 is a diagram illustrating an example of a weighting function multiplied by the input signal generation unit in the noise reduction method according to the third embodiment.
  • FIG. 10 is a diagram illustrating an example of an output signal of the inverse Fourier transform unit in the noise reduction method according to the third embodiment.
  • FIG. 11 is a block diagram showing a configuration of a spectral subtraction method according to Embodiment 4 of the present invention.
  • FIG. 12 is a diagram illustrating an operation in the noise reduction method according to the fourth embodiment.
  • FIG. 13 is a diagram showing a configuration of a spectral subtraction method according to the fifth embodiment of the present invention.
  • FIG. 14 is a diagram illustrating an operation in the noise reduction method according to the fifth embodiment.
  • FIG. 15 is a diagram showing a configuration of a conventional noise reduction method.
  • FIG. 16 is a diagram showing a window function used in the conventional noise reduction method.
  • FIG. 17 is a comparison diagram of the operation of the first embodiment.
  • FIG. 18 is a diagram illustrating a waveform shaping process according to the third embodiment.
  • a noise reduction method in which the subtraction rate of the spectral subtraction filter is made variable and an amplitude adjustment filter section is provided to further subtract an amplitude from a normal output depending on the extraction rate will be described.
  • FIG. 1 is a block diagram illustrating a configuration of a noise reduction method according to the present embodiment.
  • FIG. 2 is a flowchart illustrating an operation of the amplitude adjustment filter unit in FIG.
  • FIG. 17 is a diagram showing a comparison of operations in a speech section and a noise section.
  • the input signal 7 cut out to a predetermined frame length is orthogonally transformed by the Fourier transform unit 1 and transformed into the frequency domain, and the input phase spectrum 8 and the input amplitude spectrum 9 are output.
  • the input signal is judged as a speech frame (segment) if it is equal to or greater than the threshold value TH, and a noise frame (segment) if not more than the threshold value TH.
  • P in is the input signal power
  • TH is before updating the noise section determination threshold value
  • TH new new is the noise section determination threshold updated.
  • the estimated noise amplitude spectrum calculation unit 3 replaces the input amplitude spectrum 9 at that time (with the estimated noise amplitude spectrum up to that point) by, for example, the following equation (5). Then, the estimated noise amplitude spectrum 10 updated by weighting and adding is output.
  • ⁇ ( ⁇ ( ⁇ )) is the updated estimated noise amplitude spectrum
  • E old (N ( ⁇ )) is the estimated noise amplitude spectrum before updating
  • the subtraction 'filter unit 4 calculates the removal rate r based on the following equation (6), and outputs it as the noise removal strength 13.
  • POWs is the percentage of the amplitude spectrum of the input signal at all frequencies in the current frame.
  • POW N is the power of the estimated noise amplitude spectrum for all frequencies of the current frame 0 ⁇ r TH ⁇ 1
  • the input amplitude spectrum and the estimated noise amplitude spectrum are subtracted in the same manner as in equation (1), and the difference output amplitude spectrum 11 is output. I do.
  • the noise suppression is reduced by reducing the extraction rate for noise frames and low power frames such as consonants.
  • the output amplitude 12 returned to the time domain by the inverse Fourier transform unit 5 has a smaller amplitude adjustment coefficient in the next amplitude adjustment filter unit 6 in a frame having a small power, so that the amplitude suppression is enhanced.
  • the amplitude adjustment filter unit 6 is configured to perform steps 3 to 3 in FIG. Perform two actions.
  • the noise removal strength 13 (removal rate r) is input.
  • the difference between the power of the amplitude spectrum of the speech section of the input signal of the current frame and the power of the estimated noise amplitude spectrum of the current frame is obtained.
  • POWs is the power of the amplitude spectrum of all the frequencies of the input signal of the current frame
  • POWn is the power of the estimated noise amplitude spectrum of all the frequencies of the current frame
  • G is the amplitude suppression coefficient of the current frame.
  • R is the withdrawal rate. As shown in Fig. 17, when the withdrawal rate r is large, the amplitude suppression coefficient G increases. When the withdrawal rate r is small, the amplitude suppression coefficient G becomes small.
  • g (n) g (n-1) AR + G (1— AR) (8)
  • G (n) is the amplitude adjustment coefficient of the n-th sample of the current frame, and n-1 is the previous sample.
  • n-1 is the previous sample.
  • the amplitude suppression coefficient G increases, and the amplitude adjustment coefficient g (n) increases.
  • the removal ratio r is small, the amplitude suppression coefficient G becomes small, and the amplitude adjustment coefficient g (n) becomes small.
  • FIG. 4 is a diagram showing the input / output characteristics of the amplitude adjustment filter unit 6.
  • the extraction rate r is small, so that the amplitude adjustment coefficient g (n) is small, and the suppression on the output side is large, and the output value is large.
  • the amplitude adjustment coefficient g ( n ) is large because the withdrawal rate r is large in the voice section.
  • the suppression is small and the output value is relatively large.
  • the amount of noise removal in the frequency domain is reduced in a noise frame or a consonant voice frame having a small power, so that the generation of musical noise in the noise frame can be suppressed and the voice section can be prevented from being cut.
  • Figure 5 shows the amplitude spectrum of the noise frame before and after the subtraction.
  • the removal rate r When the removal rate r is set to 1.0 and the noise is largely removed, only the frequency component of the strong amplitude spectrum of the removed noise remains and becomes a source of musical noise.
  • the withdrawal rate r When the withdrawal rate r is set to 0.5 and the withdrawal is weakened, the deviation of the amplitude to a part of the frequency in the amplitude spectrum of the noise after the withdrawal is reduced, and no musical noise is generated.
  • the amplitude is suppressed accordingly, so that it is possible to prevent the noise suppression amount from becoming insufficient.
  • the amplitude adjustment coefficient is gradually changed in the time axis direction for each sample even in one frame by equation (8), so that Even if the amplitude is suddenly adjusted such as at the rising edge, a natural output can be obtained.
  • the amplitude adjustment may be performed in the frequency domain. That is, an amplitude adjustment filter section is provided after the Fourier transform, and the signal after the amplitude adjustment is subjected to the inverse Fourier transform. Configuration.
  • FIG. 6 is a block diagram showing a configuration of a spectral subtraction method according to the embodiment.
  • the configuration in FIG. 6 shows a configuration in which the amplitude adjustment filter in FIG. 1 is moved immediately after the subtraction filter 4 and the amplitude adjustment is performed in the frequency domain.
  • the operation is the same as that of mode 1.
  • the subtraction ′ filter unit 4 calculates the subtraction rate by the equation (6), and subtracts the estimated noise amplitude spectrum 10 from the input amplitude spectrum 9 based on the subtraction rate, Outputs the output amplitude spectrum.
  • the amplitude adjustment filter section 15 multiplies the amplitude adjustment coefficient calculated based on the withdrawal rate r by the output amplitude spectrum, and outputs a final output amplitude; spectrum.
  • the withdrawal strength is reduced in a noise frame or a voice frame having a low power such as a consonant
  • the generation of musical noise can be suppressed in the noise frame, and the deformation or disappearance of the consonant section can be prevented.
  • amplitude suppression is performed accordingly, so that it is possible to prevent shortage of noise suppression.
  • the amplitude adjustment coefficient need only be calculated once per frame, and it is not necessary to calculate the amplitude adjustment coefficient for each sample of the signal as in S3 of FIG. 2 of the first embodiment. .
  • the input signal since the input signal extracts the voice for each section divided by the predetermined frame, the input signal becomes discontinuous at the change point of the frame (section), which may cause abnormal noise. . In the present embodiment, this is Better to smooth signal changes between frames.
  • FIG. 7 is a block diagram showing a configuration of a spectral subtraction method according to Embodiment 3 of the present invention.
  • the input signal generation unit 19 and the waveform shaping processing unit 20 are new elements, and the other elements are the same as those in FIG. 1.
  • FIG. 8 shows an example of the input signal 7 in the noise reduction method according to the third embodiment.
  • FIG. 9 is a diagram illustrating a weighting function for multiplying the input signal performed by the input signal generation unit in the noise reduction method according to the third embodiment.
  • FIG. 10 is a diagram illustrating an example of an output signal 5 a of the inverse Fourier transform unit 5 in the noise reduction method according to the third embodiment.
  • FIG. 8 shows the time-series amplitude of the input signal 7 after the AD conversion, and this is input to the input signal generation unit 19.
  • a in Fig. 8 is the set section before the frame.
  • B and C in Fig. 8 are the set sections after the frame.
  • the weighted signal whose end approaches 0 is subjected to Fourier transform, so that the noise reduction capability is improved.
  • this system was implemented with a fixed-point arithmetic digital signal processor (fixed-point DSP (digital signal processor)), the signals at both ends (A and C in Fig. 18), whose accuracy deteriorates due to the weighting, are reduced. Since it is not used for waveform shaping, it is easy to ensure output accuracy and to prevent abnormal noise at frame boundaries.
  • fixed-point arithmetic digital signal processor fixed-point DSP (digital signal processor)
  • FIG. 11 is a block diagram showing a configuration of a spectral subtraction method according to Embodiment 4 of the present invention.
  • a withdrawal rate calculator 16 as a new element.
  • the other elements are the same as those with the same numbers in FIG.
  • FIG. 12 is a diagram illustrating an operation in the noise reduction method according to the present embodiment.
  • the solid line represents the input power P ⁇ Ws in the noise section
  • the dotted line represents the average noise power POW AVE
  • the chain line represents the noise section determination threshold POW TH .
  • POW TH is the threshold TH in the pre-noise frame described in equation (3).
  • This configuration has a configuration in which a removal rate calculation unit 16 is added to the overall configuration of the spectral subtraction of the conventional method.
  • the filter of the subtraction filter unit 4 performs the operation of equation (1).
  • the noise interval determination unit 2 calculates a noise interval determination threshold value POW TH , and determines that the noise frame is a noise frame when the input level falls below this threshold value.
  • the subtraction rate calculation unit 16 calculates the average power of a plurality of noise frames close to the current based on the determination result and sets the average power as the average noise power. Based on the following equation (12), the noise section determination threshold is determined. P ⁇ W TH , average noise power P
  • the magnitude r1 of this noise fluctuation is defined as the noise vector removal rate r in equation (1).
  • the noise section determination threshold POW TH value uses a high value that can be correctly determined as a noise section in the case of noise with large power fluctuations.
  • Figure 12 shows the changes in the average noise power and the noise section determination threshold with respect to the change in the input power in the noise section at this time.
  • the average noise power POW AVE of the noise frame takes a value smaller than the noise section determination threshold value POW TH . Therefore, the withdrawal rate r 1 is smaller than 1. As a result, the effect of suppressing the removal of the estimated noise amplitude spectrum can be obtained, the voice shaving can be reduced, and the estimated noise amplitude spectrum is updated normally.
  • FIG. 13 is a block diagram showing main components of a noise reduction method according to Embodiment 5 of the present invention.
  • FIG. 14 is a diagram illustrating an operation in the noise reduction method according to the present embodiment.
  • the noise is reduced by suppressing the amplitude in the noise section.
  • the amplitude adjustment coefficient for performing the amplitude suppression is determined depending on the current input power and the average noise power.
  • the noise interval determination unit 301 calculates the input power POWs 305 indicated by the solid line in FIG. 12 from the input signal 304, and performs the noise interval determination using the noise interval determination threshold POW TH similar to the first embodiment. Do.
  • the average noise power calculation section 302 calculates the average power of a plurality of past noise frames close to the current frame based on the result of the noise section determination, and calculates the average noise power P OW AVE 3 shown by the dotted line in FIG. 06.
  • the amplitude adjustment filter section 303 calculates the amplitude suppression coefficient G from the input power P ⁇ Ws 305 of the input signal and the average noise power P OW AVE 306 according to equation (13), and then directly adjusts the amplitude.
  • the output signal 307 is obtained by multiplying the input signal 304 by the obtained amplitude adjustment coefficient.
  • FIG. 14 shows the relationship between the input power P OWs 305 of the input signal 304 and the output power of the output signal 307 after the amplitude adjustment.
  • the dotted line in Fig. 14 shows the case where the input signal was output as it was without using the amplitude adjustment filter. That is, the dotted line indicates a case where the input power of the input signal is equal to the output power of the output signal.
  • the solid line shows the relationship between the input power POWs and the output power when the amplitude adjustment filter is used. This indicates that a value smaller than the dotted line is output by amplitude suppression. Also, when the input power is the average noise power P_ ⁇ _W AVE smaller than the output signal Pawa one indicates that zero.
  • the amplitude adjustment coefficient is determined from the result of the voice state determination. Frequently, as in the case of direct determination, the amplitude suppression does not change significantly, and the audio output stabilizes.
  • the withdrawal rate is made variable and the amplitude adjustment filter section that changes the degree of amplitude suppression based on this is provided, it is possible to prevent the voice section from being cut off.
  • the subtraction strength in a noise frame or the like is reduced, noise is suppressed by performing amplitude suppression corresponding to the strength, so that a balanced and easy-to-hear output can be obtained as a whole.
  • the amplitude adjustment coefficient gradually changes in the time axis direction even within a frame, a natural output can be obtained even when a sharp amplitude adjustment such as the rising of a voice section is performed.
  • the signals in the preceding and succeeding sections of the current frame are also multiplied by a weight function and added. Since the input signal generation unit includes the input signal generation unit, the target signal of the Fourier transform is weighted, the estimation accuracy of the noise spectrum is improved and the noise reduction effect is enhanced, and the unweighted signal of the inverse Fourier transform output is obtained. Is output as a frame signal, so that even with a fixed-point arithmetic digital signal processor, there is an effect that the arithmetic operation is small and high signal accuracy can be obtained.
  • the strength of noise removal is adjusted according to the variability of the section determined to be a noise section, so that even when the variability of noise is large, a correct noise determination threshold can be set. It also has the effect of preventing mixed speech components from squeezing the speech.
  • the output is caused by frequent changes in the noise section determination. This has the effect of avoiding adverse effects.

Landscapes

  • Engineering & Computer Science (AREA)
  • Computational Linguistics (AREA)
  • Quality & Reliability (AREA)
  • Signal Processing (AREA)
  • Health & Medical Sciences (AREA)
  • Audiology, Speech & Language Pathology (AREA)
  • Human Computer Interaction (AREA)
  • Physics & Mathematics (AREA)
  • Acoustics & Sound (AREA)
  • Multimedia (AREA)
  • Noise Elimination (AREA)

Abstract

A noise reduction method which reduces unpleasant residual noise even when movement to or from noise frames is frequent and the spectrum of each frame unevenly includes certain frequencies. In a spectral subtraction noise reduction method for producing an output by subtracting by a spectral subtraction filter an estimated noise amplitude spectrum from an amplitude spectrum created by orthogonal-transforming an input signal segmented to a predetermined frame length, the subtraction rate of the spectral subtraction filter is variable according to the estimated noise amplitude spectrum and an amplitude adjusting filter circuit (6) is used. The amplitude adjust filter circuit (6) multiplies the output produced by subtraction by the spectral subtraction filter for each frame, by an amplitude adjusting coefficient to produce a desired output. The amplitude adjusting coefficient is determined by the power of the amplitude spectrum (9) and the power of the estimated noise amplitude spectrum (10).

Description

明 細 書 雑音軽減装置及び雑音軽減方法 技術分野  Description Noise reduction device and noise reduction method
本発明は、 背景雑音のある環境下で用いられる音声通信システムや音 声認識システムで適用される、 入力音声信号から雑音除去することで雑 音抑圧を行う音声信号の雑音軽減方法及び雑音区間の振幅抑圧を用いた 雑音軽減方法に関するものである。 背景技術  The present invention is applied to a voice communication system or a voice recognition system used in an environment having background noise, and a noise reduction method for a voice signal that performs noise suppression by removing noise from an input voice signal and a noise section. It relates to a noise reduction method using amplitude suppression. Background art
従来、 背景雑音を軽減する手段として、 以下の図 1 5に示される方法 が知られている。 即ち、 音声に背景雑音の重畳した入力信号を AZD ( アナログ Zディジタル) 変換して、 一定区間 (フレーム) に分割して、 このフレーム毎に雑音軽減された信号出力を得るために、 一定区間信号 (以降フレーム信号と呼ぶ) を含む入力信号列を離散フーリェ変換して 周波数スぺク トルに一旦変換し、 これを振幅スぺク トルと位相スぺク ト ルに分割し、 この入力信号の振幅スペク トルから、 無音区間から推定し た推定雑音振幅スぺク トルをサブトラクシヨン · フィルタを用いて減算 することで、 背景雑音の軽減された音声振幅スペク トル (出力振幅スぺ ク トル) を推定して、 先の位相スぺク トルと合わせて背景雑音の軽減さ れた音声の周波数スぺク トルとし、 この音声の周波数スぺク トルを逆離 散フーリエ変換して出力信号を得るスぺク トラルサブトラクシヨンなる 方法である。 これは、 S TEVEN F . BAL L "S u p p r s s i o n o f a c u s t i c n o i s e i n s p e e c h u s i n g s p e c t r a l s u b t r a c t i o n, " I EEE T r a n s . A c o u s t . , S p e e s h a n d S i g n a l P r o c . , v o l . AS S P— 29, p p. 1 1 3— 1 20, Ap r . 1 9 79. において提案されている。 Conventionally, a method shown in FIG. 15 below is known as a means for reducing background noise. That is, the input signal in which the background noise is superimposed on the voice is converted into an AZD (analog Z-digital) signal and divided into a fixed section (frame). (Hereinafter referred to as a frame signal) is converted to a frequency spectrum by a discrete Fourier transform, and then divided into an amplitude spectrum and a phase spectrum. By subtracting the estimated noise amplitude spectrum estimated from the silent section from the amplitude spectrum using a subtraction filter, the audio amplitude spectrum with reduced background noise (output amplitude spectrum) Is estimated as the frequency spectrum of the voice with the background noise reduced along with the previous phase spectrum, and the frequency spectrum of this voice is inversely dispersive Fourier transformed and the output signal is obtained. A scan Bae click Tsentralnyi sub tiger click Chillon becomes how obtained. This is S TEVEN F. BAL L "S upprssionofacusticnoi seinspeechusingspectr alsubtraction," I EEE T rans. A coust., Speeshand Signal Proc., vol. AS SP-29, pp. 113-120, Apr. 1979.
図 1 5は、 この雑音軽減方法の構成を示したブロック図である。 まず、 図 1 5に基づいて全体の動作を説明する。  FIG. 15 is a block diagram showing the configuration of this noise reduction method. First, the overall operation will be described with reference to FIG.
所定のフレーム長に切り出された入力信号 1 0 7は、 フーリエ変換部 1 0 1で周波数領域に変換され、 入力位相スぺク トル 1 08と、 入力振 幅スぺク トル 1 09が出力される。 また、 入力信号は、 雑音区間判定部 1 02により、 しきい値 TH以上なら音声フレーム (区間) 、 以下なら 雑音フレーム (区間) と判定する。  The input signal 107 cut out to a predetermined frame length is converted into the frequency domain by the Fourier transform unit 101, and the input phase spectrum 108 and the input amplitude spectrum 109 are output. You. In addition, the input signal is determined by the noise section determination unit 102 to be a speech frame (section) if it is equal to or more than the threshold value TH, and a noise frame (section) otherwise.
推定雑音振幅スぺク トル算出部 1 03は、 現フレームが雑音フレーム と判定されると、 その時の入力振幅スペク トル 1 09を (それまでの推 定雑音振幅スぺク トルに) 重み付け加算して更新した推定雑音振幅スぺ タ トル 1 1 0を出力する。  When the current frame is determined to be a noise frame, the estimated noise amplitude spectrum calculation unit 103 weights and adds the input amplitude spectrum 109 at that time to the previous estimated noise amplitude spectrum. And outputs the estimated noise amplitude spectrum 110 updated.
サブトラクシヨン ' フィルタ部 1 04は、 その代表的な伝達関数が式 (1 ) で表される特性を持つ。
Figure imgf000004_0001
The subtraction 'filter section 104 has a characteristic whose representative transfer function is represented by the following equation (1).
Figure imgf000004_0001
F (ω) :サブトラクシヨン ' フィルタ  F (ω): Subtraction 'filter
S (ω) :入力振幅スペク トル  S (ω): Input amplitude spectrum
Ε (Ν (ω) ) :推定雑音振幅スペク トル  Ε (Ν (ω)): Estimated noise amplitude spectrum
r :引き去り率  r: removal rate
式 (1 ) において、 S (ω) は入力信号の振幅スペク トル項、 Ε (Ν (ω) ) は推定雑音振幅スペク トル項、 rは推定雑音振幅スペク トルの 引き去り率を表す定数である。 rを大きくすれば雑音抑圧量が大きくな り、 逆に小さくすれば雑音抑圧量が小さくなる。 サブトラクシヨン ' フィルタ部 1 0 4では、 所定の式に基づいて引き 去り率 rを求める。 In Equation (1), S (ω) is the amplitude spectrum term of the input signal, Ε (Ν (ω)) is the estimated noise amplitude spectrum term, and r is a constant representing the removal rate of the estimated noise amplitude spectrum. Increasing r increases the amount of noise suppression; conversely, decreasing r decreases the amount of noise suppression. The subtraction 'filter section 104 calculates the removal rate r based on a predetermined equation.
更に、 この引き去り率 rに基づいて、 式 ( 1 ) と同様にして入力振幅 スぺク トルと推定雑音振幅スぺク トルとを減算して、 差の出力振幅スぺ タ トル 1 1 1を出力する。  Further, based on the subtraction rate r, the input amplitude spectrum and the estimated noise amplitude spectrum are subtracted in the same manner as in equation (1), and the difference output amplitude spectrum 111 is obtained. Output.
即ち、 サブトラクション · フィルタ部 1 0 4では、 雑音フレームや子 音等のパワーの小さなフレームでは引き去り率を小さく して雑音抑圧を 弱める。  In other words, the subtraction / filter unit 104 reduces the removal rate for noise frames and low power frames such as consonants to reduce noise suppression.
逆フーリエ変換部 1 0 5で時間領域に戻された出力信号が出力される 第 1の従来のスぺク トラルサブトラクション方法においては、 音声の 立ち上がりや立ち下がりや子音といったパワーの小さいフレームでは、 音声が削られて品質劣化する欠点がある。 上述の rを大きくすれば音声 の削れがより大きくなる。  In the first conventional spectral subtraction method, an output signal returned to the time domain by the inverse Fourier transform unit 105 is output. However, there is a disadvantage that the quality is deteriorated due to the scraping. The larger the value of r is, the more the sound is cut.
これを解決するために、 フレーム毎にサブトラクシヨン ' フィルタの 引き去り率を可変とする方法が特開平 8— 2 2 1 0 9 2で知られている 。 この方法では、 サブトラクシヨン · フィルタとして、 式 1の F ( ω ) を用いた場合は、 入力が大きい場合を例にとると rを大きく、 入力が小 さいときは rを小さくすることで音声の削れを軽減している。  To solve this problem, a method of making the subtraction rate of the subtraction filter variable for each frame is known from Japanese Patent Application Laid-Open No. Hei 8-22092. In this method, when F (ω) in Equation 1 is used as the subtraction filter, r is increased when the input is large, and r is decreased when the input is small. Has reduced shavings.
第 2の従来のスぺク トラルサブトラクション方法として、 雑音軽減効 果を高めるために、 フーリエ変換対象信号に窓関数と呼ばれる関数を時 間軸で区間を定めて、 乗じて重み付けを行う手法が用いられている。 例 えば、 フレーム信号とこれに連続する一定長の信号 (オーバーラップ信 号) に窓関数を乗じフーリエ変換し、 逆フーリエ変換の出力のフレーム 信号は、 先頭の 1以下重み付けされたサンプルに、 直前フレームの逆フ 一リエ変換の出力のオーバーラップ信号を加算して出力とする方式が、 北米 CDMA自動車電話システム (T I A/E I A, I S— 1 2 7, " E n h a n c e d V a r i a b l e R a t e C o d e c S e r v i c e O p t i o n 3 f o r W i d e b a n d S p r e a d S p e c t r um D i g i t a l S y s t e m s " ) で実用ィ匕 されている。 ここで用いられる窓関数を図 1 6に示す。 As a second conventional spectral subtraction method, in order to enhance the noise reduction effect, a method is used in which a function called a window function is defined on the time axis of the signal to be subjected to Fourier transform and multiplied and weighted. Have been. For example, a frame signal and a continuous signal with a certain length (overlap signal) are multiplied by a window function and subjected to Fourier transform. The output frame signal of the inverse Fourier transform is added to the first weighted sample below 1 The method of adding the overlap signal of the output of the inverse Fourier transform of the frame and outputting it is: North America CDMA mobile phone system (TIA / EIA, IS-127, "Enhanced Variable Rate Code Service Option 3 for Wideband S pread Spectrum Digital Systems") I have. Figure 16 shows the window function used here.
ここで、 フレーム信号とオーバーラップ信号の加算時に対応するサン プルの重みの和が 1になるように、 また、 フーリエ変換対象信号の両端 が滑らかに 0に近付くような窓関数が用いられる。 このような窓関数に よる重み付けを行うことによって、 雑音振幅スぺク トルの推定精度が高 まるという効果が得られので、 結果的に雑音軽減効果が高めることがで さる。  Here, a window function is used so that the sum of the weights of the samples corresponding to the addition of the frame signal and the overlap signal becomes 1, and that both ends of the Fourier transform target signal smoothly approach 0. By performing weighting using such a window function, the effect of increasing the estimation accuracy of the noise amplitude spectrum is obtained, and as a result, the noise reduction effect is improved.
また、 第 3の従来のスぺク トラルサブトラクシヨン方法として、 フレ ーム信号とこれに連続する一定長の信号 (オーバーラップ信号) を窓関 数による重み付けをせずにフーリエ変換する方法があり、 特開平 9一 3 449 7に開示されたノイズ削減装置で用いられてる。 この場合は、 出 力がフレーム間で滑らかにつながるようにするために、 逆フーリェ変換 出力のフレーム信号に対して直前フレームの逆フーリェ変換出力のォ一 バーラップ信号を三角窓で重ね合わせる波形整形処理を施して出力し、 逆フーリェ変換出力のオーバーラップ信号を次フレームのオーバーラッ プ処理のために保存する。 このときの波形整形処理を式 (2) に示す。
Figure imgf000006_0001
As a third conventional spectral subtraction method, there is a method of performing a Fourier transform on a frame signal and a signal of a fixed length (overlap signal) following the frame signal without weighting by a window function. Yes, it is used in the noise reduction device disclosed in Japanese Patent Application Laid-Open No. Hei 9-134449. In this case, in order to smoothly connect the output between the frames, a waveform shaping process is performed in which the overlap signal of the inverse Fourier transform output of the immediately preceding frame is superimposed on the triangular window with the frame signal of the inverse Fourier transform output. And outputs the same, and saves the inverse Fourier transform output overlap signal for overlap processing of the next frame. Equation (2) shows the waveform shaping process at this time.
Figure imgf000006_0001
( j =0〜L一 1 )  (j = 0-L-1)
Oj
Figure imgf000006_0002
し〜 L+M— 1 )
Oj
Figure imgf000006_0002
~ L + M— 1)
Zj =〇M+j ( j =〇〜L一 1 ) Zj = 〇 M + j (j = 〇 ~ L-1)
(2)  (2)
但し、 0」· : 出力信号 However, 0 ”·: Output signal
D j : フレーム信号  D j: Frame signal
Z j : オーバーラップ信号  Z j: Overlap signal
M : フレーム長  M: Frame length
L : オーバーラップ信号長  L: overlap signal length
である。 It is.
この方法の利点を以下に説明する。  The advantages of this method are described below.
スぺク トラルサブトラクシヨン方法を実現するためには、 原理的に演 算精度に限界がある 1 6ビッ ト固定小数点演算ディジタル信号プロセッ サ (固定小数点 D S P ) を用いることが多い。 この固定少数点 D S Pは 倍精度演算や、 シフトによる桁上げによって演算精度を上げることがで きる。 でも、 そのための演算量が増えてしまう問題がある。  In order to implement the spectral subtraction method, a 16-bit fixed-point digital signal processor (fixed-point DSP), which has a theoretically limited calculation accuracy, is often used. The operation precision of the fixed-point decimal point D SP can be increased by double-precision arithmetic operation or carry by shift. However, there is a problem that the amount of calculation for this increases.
フーリェ変換と逆フーリェ変換は基本的に演算量が大きく、 演算精度 を少し犠牲にしても、 少ない演算量で実現したい場合が多くある。 この 方法を用いた場合、 フーリエ変換するフレーム信号に窓関数による重み 付けがされていないためにダイナミックレンジが広がらず、 演算精度の 限られたフーリエ変換と逆フーリェ変換を経た出力フレーム信号の精度 劣化が少ない。 この理由から少ない演算量でよい固定小数点 D S Pが用 レヽられる。  The Fourier transform and the inverse Fourier transform basically require a large amount of computation, and there are many cases where it is desired to realize the computation with a small amount of computation at the expense of a little computation accuracy. When this method is used, the frame range to be Fourier-transformed is not weighted by the window function, so the dynamic range is not widened, and the accuracy of the output frame signal that has undergone limited Fourier transform and inverse Fourier transform is degraded. Less is. For this reason, a fixed-point DSP that requires a small amount of computation is used.
第 4の従来のスぺク トラルサブトラクションを用いた雑音軽減方法は The fourth conventional noise reduction method using spectral subtraction is
、 推定雑音振幅スぺク トルの算出のために、 まず、 雑音のパワーを推定 して、 これを雑音区間判定しきい値とする。 そして、 入力パワーがこの しきい値より小さい場合には、 雑音フレームと判定する。 更に、 推定雑 音振幅スぺク トルは、 雑音と判定された区間の複数フレームの入力信号 、 つまり、 雑音信号の振幅スペク トルを平均して算出する方法が一般的 である。 例えば、 特開平 8— 2 2 1 0 9 2等におけるスぺク トラルサブ トラクシヨンで、 この方法が用いられている。 In order to calculate the estimated noise amplitude spectrum, first, the power of the noise is estimated, and this is used as a noise section determination threshold. If the input power is smaller than this threshold, it is determined that the frame is a noise frame. Further, the estimated noise amplitude spectrum is generally calculated by averaging the input signals of a plurality of frames in the section determined to be noise, that is, the average of the amplitude spectrum of the noise signal. For example, the spectral sub in Japanese Patent Application Laid-Open No. This method is used in fractionation.
第 5の従来の背景雑音を軽減する方法として、 雑音区間を振幅抑圧す ることにより雑音感を軽減する方法がある。 信号の様態を表す予め定め られた有限状態から現フレームの状態を決定して、 この状態が雑音フレ ーム状態に対応している場合、 一定の強度の振幅抑圧を行う方法が特開 平 7— 3 8 4 5 4で示されている。  As a fifth conventional method of reducing background noise, there is a method of reducing noise sensation by suppressing the amplitude of a noise section. A method of determining the state of the current frame from a predetermined finite state representing the state of the signal and determining the state of the current frame when the state corresponds to the state of a noise frame is disclosed in Japanese Unexamined Patent Publication No. — Indicated by 3 8 4 5 4.
第 1の従来のスぺク トラルサブトラタシヨン方法には、 雑音区間が変 形してミュージカルノイズと呼ばれる不快残留雑音となる欠点があり、 実用上の最大の課題となっている。  The first conventional spectral subtraction method has a drawback that a noise section is deformed and becomes unpleasant residual noise called musical noise, which is the biggest practical problem.
これは雑音区間においては、 推定雑音振幅スぺク トルを減算された出 力振幅スぺク トルが一部周波数にパワーの集中した形状となり、 また、 このパワーの集中した周波数がフレーム毎に不規則に変化することでお こる。 ミュージカルノイズを解決する方法として、 推定雑音振幅スぺク トルの引き去り率を可変とする方法を用い、 雑音フレームでは引き去り 量を小さくすればよいが、 雑音区間では雑音抑圧量が不足するという課 題がある。  This is because in the noise section, the output amplitude spectrum from which the estimated noise amplitude spectrum has been subtracted has a shape in which power is concentrated in a part of the frequency, and the frequency in which the power is concentrated is not per frame. It changes by changing the rules. As a method of solving musical noise, a method of varying the removal rate of the estimated noise amplitude spectrum is used, and the removal amount may be reduced in a noise frame, but the noise suppression amount is insufficient in a noise section. There is.
第 2の従来のスぺク トラルサブトラクション方法においては、 窓関数 によつて両端を 0に近付ける重み付けした信号をフーリェ変換すれば、 雑音軽減能力は向上する。 しかし、 固定小数点 D S Pで実現する際に、 両端の信号の精度が低下するので、 逆フーリエ変換の出力信号の両端が 劣化してフレーム境界で異音が起こるという課題がある。  In the second conventional spectral subtraction method, the noise reduction ability is improved by performing a Fourier transform on a weighted signal whose both ends approach 0 using a window function. However, there is a problem in that the accuracy of the signals at both ends is reduced when realizing with fixed-point DSP, so that both ends of the output signal of the inverse Fourier transform are degraded and abnormal noise occurs at frame boundaries.
第 3の従来例のように、 逆に、 重み付けしない信号をフーリエ変換す れば、 固定小数点 D S Pで実現する際に精度は取りやすいが、 入力振幅 スぺク トルに含まれる雑音振幅スぺク トル成分のフレーム間の変動は、 重み付けした場合より大きくなるので、 結果として雑音軽減能力は重み 付けした場合に劣る。 第 4の従来のスぺク トラルサブトラクションの雑音区間判定において 、 雑音のパワー変動が大きい場合に、 雑音フレームを正しく判定できる ように雑音区間判定しきいを高めに設定すると、 ある程度音声フレーム も雑音フレームと誤判定し、 推定雑音振幅スぺク トルには音声スぺク ト ルが漏れこみ、 音声の削れの原因となるという課題がある。 逆に、 これ を防ぐために、 音声フレームを雑音フレームと判定しないように雑音区 間判定しきい値を低めに設定すると、 雑音フレームを音声フレームと判 定してしまい、 推定雑音振幅スペク トルの更新が正しく行われずに、 結 果として雑音軽減能力が劣化するという課題がある。 Conversely, if the unweighted signal is Fourier-transformed as in the third conventional example, accuracy can be easily achieved when implementing it with a fixed-point DSP, but the noise amplitude spectrum included in the input amplitude spectrum can be improved. Since the variation of the torque component between frames becomes larger than when weighted, the noise reduction ability is inferior to that when weighted. In the noise section determination of the fourth conventional spectral subtraction, if the noise section determination threshold is set high so that the noise frame can be correctly determined when the power fluctuation of the noise is large, some of the speech frames may also be noise frames. There is a problem in that the speech noise leaks into the estimated noise amplitude spectrum, causing the speech to be cut off. Conversely, to prevent this, if the noise interval determination threshold is set low so that the audio frame is not determined to be a noise frame, the noise frame is determined to be a voice frame, and the estimated noise amplitude spectrum is updated. However, there is a problem that the noise reduction is not performed correctly, and as a result, the noise reduction ability is deteriorated.
第 5の雑音区間を振幅抑圧することにより雑音感を軽減する方法では 、 信号の様態を表す予め定められた有限状態から現フレームのフレーム 状態を決定して、 このフレーム状態が雑音フレーム状態である場合、 一 定の強度の振幅抑圧を行う方式をとると、 入力雑音種によっては短時間 の間にフレーム状態が音声フレーム状態と雑音フレーム状態の間を頻繁 に移動して、 これに伴い振幅抑圧の強度が頻繁に変化して、 結果として 出力のパワーが不安定になり聴感的に劣るという課題があった。  In the method of reducing noise by suppressing the amplitude of the fifth noise section, the frame state of the current frame is determined from a predetermined finite state representing the state of the signal, and this frame state is a noise frame state. In this case, if a method is adopted in which the amplitude is suppressed to a certain level, the frame state frequently moves between the voice frame state and the noise frame state in a short period of time depending on the type of input noise. The intensity of the power changes frequently, resulting in unstable output power and poor hearing.
本発明は、 上記の課題を解決するためになされたもので、 雑音フレー ムへの、 又は、 からの移動が頻繁で、 フレーム毎のスペク トルが一部の 周波数に偏っていても、 不快残留雑音が少ない雑音軽減方法を得ること を目的とする。 発明の開示  The present invention has been made in order to solve the above-mentioned problem, and unpleasant residuals are obtained even when noise frames frequently move to or from noise frames and the spectrum of each frame is biased to some frequencies. The aim is to obtain a noise reduction method with less noise. Disclosure of the invention
この発明に係る雑音軽減装置は、 所定のフレーム長に切り出した入力 信号を直交変換して得た振幅スぺク トルから推定雑音振幅スぺク トルを スぺク トラルサブトラクシヨン ' フィルタにより減算して出力を得る雑 音軽減装置において、 フレーム毎に上記スぺク トラルサブトラクシヨン · フィルタによる減 算出力に、 上記振幅スぺク トルのパワーと上記推定雑音振幅スぺク トル のパワーとで決まる振幅調整係数を乗算して、 所要出力を得る振幅調整 フィルタ部を備えたことを特徴とする。 The noise reduction apparatus according to the present invention subtracts an estimated noise amplitude spectrum from an amplitude spectrum obtained by orthogonally transforming an input signal cut out to a predetermined frame length by a spectral subtraction filter. Noise reduction device to obtain output For each frame, the required power is calculated by multiplying the reduced power by the spectral subtraction filter by an amplitude adjustment coefficient determined by the power of the amplitude spectrum and the power of the estimated noise amplitude spectrum. An amplitude adjustment filter unit for obtaining an output is provided.
スぺク トラルサブトラクシヨン ' フィルタは、 推定雑音振幅スぺク ト ルの減算の引き去り率を上記推定雑音振幅スぺク トルに応じて可変とし The spectral subtraction 'filter makes the subtraction rate of the subtraction of the estimated noise amplitude spectrum variable according to the estimated noise amplitude spectrum.
、 振幅調整フィルタ部は、 引き去り率に応じて振幅調整係数を可変とす ることを特徴とする。 The amplitude adjustment filter unit is characterized in that the amplitude adjustment coefficient is made variable in accordance with the withdrawal rate.
上記振幅調整フィルタ部は、 引き去り率が大きい場合、 振幅調整係数 を大きくすることにより、 音声区間で雑音抑圧を強め、 かつ、 出力信号 の出力値を大きくするとともに、 上記振幅調整フィルタ部は、 引き去り 率が小さい場合、 振幅調整係数を小さくすることにより、 雑音区間で雑 音抑圧を弱め、 かつ、 出力信号の出力値を小さくすることを特徴とする 振幅調整フィルタ部は、 振幅調整係数をスぺク トラルサブトラクショ ン · フィルタによる減算出力を逆直交変換した時間領域の振幅スぺク ト ルに乗算することを特徴とする。  When the subtraction rate is high, the amplitude adjustment filter unit increases the amplitude adjustment coefficient to enhance noise suppression in the voice section and increases the output value of the output signal. When the rate is small, the amplitude adjustment coefficient is reduced, thereby reducing noise suppression in a noise section and reducing the output value of the output signal. It is characterized by multiplying the subtraction output by the central subtraction filter with the time-domain amplitude spectrum subjected to the inverse orthogonal transform.
振幅調整フィルタ部は、 振幅調整係数をフレーム毎に周波数領域で振 幅スぺク トルに乗算し、 逆直交変換して出力を得るようにしたことを特 徴とする。  The amplitude adjustment filter section is characterized in that the amplitude spectrum is multiplied by the amplitude adjustment coefficient in the frequency domain for each frame, and an inverse orthogonal transform is performed to obtain an output.
振幅調整係数は、 現フレームの入力信号の振幅スぺク トルのパワーと 、 現フレームの推定雑音振幅スぺク トルのパワーとの差に基づいて得た 現フレームの振幅調整係数に、 前フレームで得た振幅調整係数の値を重 み付け加算した値としたことを特徴とする。  The amplitude adjustment coefficient is obtained by adding the amplitude adjustment coefficient of the current frame obtained based on the difference between the power of the amplitude spectrum of the input signal of the current frame and the power of the estimated noise amplitude spectrum of the current frame to the previous frame. It is characterized in that the value of the amplitude adjustment coefficient obtained in (1) is weighted and added.
この発明に係る雑音軽減装置は、 所定のフレーム長に切り出した入力 信号を直交変換して得た振幅スぺク トルから推定雑音振幅スぺク トルを スぺク トラルサブトラクション ' フィルタにより減算して出力を得る雑 音軽減装置において、 A noise reduction apparatus according to the present invention provides an estimated noise amplitude spectrum from an amplitude spectrum obtained by orthogonally transforming an input signal cut out to a predetermined frame length. In a noise reduction device that obtains an output by subtracting with a spectral subtraction 'filter,
現フレームの前後の設定区間を切り出して端が 0に近くなるように、 1以下の重み関数を乗算した後に現フレームの前後に付加する入力信号 生成部を備えて、 該入力信号生成部の出力を入力信号として振幅スぺク トルを求めるようにしたことを特徴とする。  An input signal generation unit that cuts out a set section before and after the current frame and multiplies by a weighting function of 1 or less and adds the input signal before and after the current frame so that the end is close to 0, and outputs the output of the input signal generation unit Is used as an input signal to obtain an amplitude spectrum.
この発明に係る雑音軽減装置は、 1の重み関数が乗算された前フレー ムの後の設定区間を用いて現フレームの波形整形を行う波形整形処理部 を備えたことを特徴とする。  A noise reduction apparatus according to the present invention includes a waveform shaping processing unit that shapes a waveform of a current frame using a set section after a previous frame multiplied by one weighting function.
この発明に係る雑音軽減装置は、 所定のフレーム長に切り出した入力 信号を直交変換して得た振幅スぺク トルから推定雑音振幅スぺク トルを スぺク トラルサブトラクション · フィルタにより減算して出力を得る雑 音軽減装置において、  The noise reduction apparatus according to the present invention subtracts an estimated noise amplitude spectrum from an amplitude spectrum obtained by orthogonally transforming an input signal cut out to a predetermined frame length by a spectral subtraction filter. In a noise reduction device that obtains output,
上記推定雑音振幅スぺク トルの複数フレーム分から平均雑音パワーを 求めて雑音フレームの判定しきい値と比較して引き去り率を得る引き去 り率算出部を備えて、  A subtraction rate calculation unit that obtains an average noise power from a plurality of frames of the estimated noise amplitude spectrum and compares the average noise power with a determination threshold of the noise frame to obtain a removal rate;
上記スぺク トラルサブトラクシヨン ' フィルタの減算に用いる引き去 り率を上記引き去り率としたことを特徴とする。  The subtraction rate used for the subtraction of the spectral subtraction filter is set to the above-described subtraction rate.
この発明に係る雑音軽減装置は、 推定雑音振幅スぺク トルの複数フレ ーム分から平均雑音パワーを求める平均雑音パワー算出部を備えて、 振幅調整フィルタ部は、 フレーム毎にスぺク トラルサブトラタシヨン • フィルタによる減算出力に、 振幅スぺク トルのパワーと上記求めた平 均雑音振幅パワーとで決まる振幅調整係数を乗算して、 所要出力を得る ようにしたことを特徴とする。  The noise reduction apparatus according to the present invention includes an average noise power calculation unit that calculates an average noise power from a plurality of frames of the estimated noise amplitude spectrum, and the amplitude adjustment filter unit includes a spectral sub-frame for each frame. The required output is obtained by multiplying the subtraction output by the task filter by an amplitude adjustment coefficient determined by the power of the amplitude spectrum and the average noise amplitude power obtained above.
この発明に係る雑音軽減方法は、 所定のフレーム長に切り出した入力 信号を直交変換して得た振幅スぺク トルから推定雑音振幅スぺク トルを スぺク トラルサブトラクシヨン · フィルタにより減算して出力を得る雑 音軽減方法において、 In the noise reduction method according to the present invention, an estimated noise amplitude spectrum is obtained from an amplitude spectrum obtained by orthogonally transforming an input signal cut out to a predetermined frame length. In the noise reduction method of obtaining an output by subtracting with a spectral subtraction filter,
上記スぺク トラルサブトラクシヨン ' フィルタの引き去り率を上記推 定雑音振幅スぺク トルに応じて可変する工程と、  Varying the removal rate of the spectral subtraction filter according to the estimated noise amplitude spectrum;
フレーム毎に上記スぺク トラルサブトラクシヨン . フィルタによる減 算出力に、 上記振幅スぺク トルのパワーと上記推定雑音振幅スぺク トル のパワーとで決まる振幅調整係数を乗算して、 所要出力を得る工程を備 えたことを特徴とする。  The required power is calculated by multiplying the reduced power by the above-mentioned spectral subtraction filter for each frame by an amplitude adjustment coefficient determined by the power of the above-mentioned amplitude spectrum and the power of the above-mentioned estimated noise amplitude spectrum. It is characterized by a process for obtaining output.
この発明に係る雑音軽減方法は、 所定のフレーム長に切り出した入力 信号を直交変換して得た振幅スぺク トルから推定雑音振幅スぺク トルを スぺク トラルサブトラクション ' フィルタにより減算して出力を得る雑 音軽減方法において、  The noise reduction method according to the present invention is characterized in that an estimated noise amplitude spectrum is subtracted from an amplitude spectrum obtained by orthogonally transforming an input signal cut out to a predetermined frame length by a spectral subtraction 'filter. In the noise reduction method to obtain output,
上記推定雑音振幅スぺク トルの複数フレーム分から平均雑音パワーを 求めて雑音フレームの判定しきい値と比較して引き去り率を得る引き去 り率算出工程と、  A subtraction rate calculation step of obtaining an average noise power from a plurality of frames of the estimated noise amplitude spectrum and comparing the average noise power with a noise frame determination threshold to obtain a removal rate;
上記スぺク トラルサブトラクシヨン · フィルタの引き去り率を上記引 き去り率とする工程を備えたことを特徴とする。 図面の簡単な説明  A step of setting the removal rate of the spectral subtraction filter to the removal rate. BRIEF DESCRIPTION OF THE FIGURES
図 1は、 本発明の実施の形態 1におけるスぺク トラルサブトラクショ ン方法の構成を示すブロック図である。  FIG. 1 is a block diagram showing a configuration of a spectral subtraction method according to Embodiment 1 of the present invention.
図 2は、 実施の形態 1の雑音軽減方法において振幅調整フィルタが行 う動作フローチヤ一ト図である。  FIG. 2 is a flowchart showing the operation performed by the amplitude adjustment filter in the noise reduction method according to the first embodiment.
図 3は、 実施の形態 1の雑音区間判定部が行う動作を説明する図であ る。  FIG. 3 is a diagram illustrating an operation performed by the noise section determination unit according to the first embodiment.
図 4は、 実施の形態 1の振幅調整フィルタが行う動作を説明する図で ある。 FIG. 4 is a diagram for explaining the operation performed by the amplitude adjustment filter according to the first embodiment. is there.
図 5は、 実施の形態 1のスぺク トラルサブトラクション方法の出力信 号を説明する図である。  FIG. 5 is a diagram illustrating an output signal of the spectral subtraction method according to the first embodiment.
図 6は、 本発明の実施の形態 2におけるスぺク トラルサブトラクショ ン方法の構成を示すブロック図である。  FIG. 6 is a block diagram showing a configuration of a spectral subtraction method according to Embodiment 2 of the present invention.
図 7は、 本発明の実施の形態 3におけるスぺク トラルサブトラクショ ン方法の構成を示す図である。  FIG. 7 is a diagram showing a configuration of a spectral subtraction method according to Embodiment 3 of the present invention.
図 8は、 実施の形態 3の雑音軽減方法における入力信号の例を示す図 である。  FIG. 8 is a diagram illustrating an example of an input signal in the noise reduction method according to the third embodiment.
図 9は、 実施の形態 3の雑音軽減方法において入力信号生成部が乗算 する重み付け関数の例を示す図である。  FIG. 9 is a diagram illustrating an example of a weighting function multiplied by the input signal generation unit in the noise reduction method according to the third embodiment.
図 1 0は、 実施の形態 3の雑音軽減方法における逆フーリエ変換部の 出力信号の例を示す図である。  FIG. 10 is a diagram illustrating an example of an output signal of the inverse Fourier transform unit in the noise reduction method according to the third embodiment.
図 1 1は、 本発明の実施の形態 4におけるスぺク トラルサブトラクシ ヨン方法の構成を示すブロック図である。  FIG. 11 is a block diagram showing a configuration of a spectral subtraction method according to Embodiment 4 of the present invention.
図 1 2は、 実施の形態 4の雑音軽減方法における動作を説明する図で ある。  FIG. 12 is a diagram illustrating an operation in the noise reduction method according to the fourth embodiment.
図 1 3は、 本発明の実施の形態 5におけるスぺク トラルサブトラタシ ョン方法の構成を示す図である。  FIG. 13 is a diagram showing a configuration of a spectral subtraction method according to the fifth embodiment of the present invention.
図 1 4は、 実施の形態 5の雑音軽減方法における動作を説明する図で ある。  FIG. 14 is a diagram illustrating an operation in the noise reduction method according to the fifth embodiment.
図 1 5は、 従来の雑音軽減方法の構成を示す図である。  FIG. 15 is a diagram showing a configuration of a conventional noise reduction method.
図 1 6は、 従来の雑音軽減方法に使用される窓関数を示す図である。 図 1 7は、 実施の形態 1の動作の比較図である。  FIG. 16 is a diagram showing a window function used in the conventional noise reduction method. FIG. 17 is a comparison diagram of the operation of the first embodiment.
図 1 8は、 実施の形態 3の波形整形処理を示す図である。 発明を実施するための最良の形態 FIG. 18 is a diagram illustrating a waveform shaping process according to the third embodiment. BEST MODE FOR CARRYING OUT THE INVENTION
実施の形態 1. Embodiment 1.
スぺク トラルサブトラクション · フィルタの引き去り率を可変とし、 振幅調整フィルタ部を備えて通常出力から更にその引き去り率に依存す る振幅減算をする本発明の実施の形態 1における雑音軽減方法を説明す る。  A noise reduction method according to the first embodiment of the present invention in which the subtraction rate of the spectral subtraction filter is made variable and an amplitude adjustment filter section is provided to further subtract an amplitude from a normal output depending on the extraction rate will be described. You.
図 1は、 本実施の形態における雑音軽減方法の構成ブロック図で、 図 2は、 図 1における振幅調整フィルタ部の動作フローチヤ一トである。 図 1 7は、 音声区間と雑音区間に対する動作の比較を示す図である。 まず、 図 1に基づいて全体の動作を説明する。  FIG. 1 is a block diagram illustrating a configuration of a noise reduction method according to the present embodiment. FIG. 2 is a flowchart illustrating an operation of the amplitude adjustment filter unit in FIG. FIG. 17 is a diagram showing a comparison of operations in a speech section and a noise section. First, the overall operation will be described with reference to FIG.
所定のフレーム長に切り出された入力信号 7は、 フーリエ変換部 1で 直交変換され、 周波数領域に変換され、 入力位相スぺク トル 8と、 入力 振幅スぺク トル 9が出力される。 また入力信号は雑音区間判定部 2によ り、 しきい値 TH以上なら音声フレーム (区間) 、 以下なら雑音フレー ム (区間) と判定する。 ここで、 次式 (3) が成立するならば、 新しい しきい値 ΤΗ に置き換える。  The input signal 7 cut out to a predetermined frame length is orthogonally transformed by the Fourier transform unit 1 and transformed into the frequency domain, and the input phase spectrum 8 and the input amplitude spectrum 9 are output. In addition, the input signal is judged as a speech frame (segment) if it is equal to or greater than the threshold value TH, and a noise frame (segment) if not more than the threshold value TH. Here, if the following equation (3) holds, replace it with a new threshold ΤΗ.
Pin< 2. 0 TH (3) P in <2.0 TH (3)
THnew = 0. 9 TH+ 0. l Pin (4) TH new = 0.9 TH + 0. l P in (4)
ここで、 Pinは入力信号パワー、 THは更新前の雑音区間判定しきい 値、 THnew は更新後の雑音区間判定しきい値である。 Here, P in is the input signal power, TH is before updating the noise section determination threshold value, TH new new is the noise section determination threshold updated.
こうして、 図 3に示す関係が得られる。 推定雑音振幅スペク トル算出 部 3は、 現フレームが雑音フレームと判定されると、 その時の入力振幅 スぺク トル 9を (それまでの推定雑音振幅スぺク トルに) 例えば、 式 ( 5) のような重み付け加算して更新した推定雑音振幅スぺク トル 1 0を 出力する。  Thus, the relationship shown in FIG. 3 is obtained. When the current frame is determined to be a noise frame, the estimated noise amplitude spectrum calculation unit 3 replaces the input amplitude spectrum 9 at that time (with the estimated noise amplitude spectrum up to that point) by, for example, the following equation (5). Then, the estimated noise amplitude spectrum 10 updated by weighting and adding is output.
E (N (ω) ) = Eold (Ν (ω) ) α + S (ω) α) (5) E (N (ω)) = E old (Ν (ω)) α + S (ω) α) (5)
Ε (Ν (ω) ) は更新後の推定雑音振幅スぺク トル、 Ε (Ν (ω)) is the updated estimated noise amplitude spectrum,
Eold (N (ω) ) は更新前の推定雑音振幅スぺク トル、 E old (N (ω)) is the estimated noise amplitude spectrum before updating,
S (ω) は入力信号の振幅スぺク トル、  S (ω) is the amplitude spectrum of the input signal,
0 < α < 1  0 <α <1
である。 It is.
サブトラクシヨン ' フィルタ部 4では、 次の式 ( 6 ) に基づいて引き 去り率 rを求め、 雑音引き去り強度 1 3として出力する。 The subtraction 'filter unit 4 calculates the removal rate r based on the following equation (6), and outputs it as the noise removal strength 13.
r =m i n { 1. 0, r TH · P OWs /P OWN } (6) r = min {1.0, r TH · P OWs / P OW N } (6)
P OWs は現フレームの全周波数の入力信号の振幅スぺク トルのパヮ POWs is the percentage of the amplitude spectrum of the input signal at all frequencies in the current frame.
P OWNは現フレームの全周波数の推定雑音振幅スぺク トルのパワー 0 < rTH< 1 POW N is the power of the estimated noise amplitude spectrum for all frequencies of the current frame 0 <r TH <1
更に、 この引き去り率 rに基づいて、 式 ( 1 ) と同様にして入力振幅 スぺク トルと推定雑音振幅スぺク トルとを減算して、 差の出力振幅スぺ タ トル 1 1を出力する。  Further, based on the subtraction rate r, the input amplitude spectrum and the estimated noise amplitude spectrum are subtracted in the same manner as in equation (1), and the difference output amplitude spectrum 11 is output. I do.
即ち、 サブトラクシヨン ' フィルタ部 4 (スベタ トラルサブトラクシ ヨン ' フィルタ) では、 雑音フレームや子音等のパワーの小さなフレー ムでは引き去り率を小さく して雑音抑圧を弱める。  In other words, in the subtraction 'filter unit 4 (smooth subtraction' filter), the noise suppression is reduced by reducing the extraction rate for noise frames and low power frames such as consonants.
逆フーリエ変換部 5で時間領域に戻された出力振幅 1 2は、 パワーの 小さなフレームでは次の振幅調整フィルタ部 6で振幅調整係数が小さく なるため、 振幅抑圧が強まることになる。  The output amplitude 12 returned to the time domain by the inverse Fourier transform unit 5 has a smaller amplitude adjustment coefficient in the next amplitude adjustment filter unit 6 in a frame having a small power, so that the amplitude suppression is enhanced.
即ち、 振幅調整フィルタ部 6は、 図 2のステップ S 1ないし S 3の 3 つの動作を行う。 最初の S 1では、 雑音引き去り強度 1 3 (引き去り率 r ) を入力する。 そして、 現フレームの入力信号の音声区間の振幅スぺ ク トルのパワーと現フレームの推定雑音振幅スぺク トルのパワーとの差
Figure imgf000016_0001
That is, the amplitude adjustment filter unit 6 is configured to perform steps 3 to 3 in FIG. Perform two actions. In the first step S1, the noise removal strength 13 (removal rate r) is input. Then, the difference between the power of the amplitude spectrum of the speech section of the input signal of the current frame and the power of the estimated noise amplitude spectrum of the current frame is obtained.
Figure imgf000016_0001
(POWs-POWN≥ 0 かつ、 P〇WS— r - POWN> 0) (POW s -POW N ≥ 0 and P〇W S — r-POW N > 0)
G= 0 G = 0
(P〇WS— POWNく 0、 或いは、 P〇WS— r · POWN≤ 0) から次式 (7) でフレーム毎の振幅抑圧係数 Gを求める。 From (P〇W S — POW N 0 0 or P〇W S — r · POW N ≤ 0), the amplitude suppression coefficient G for each frame is obtained by the following equation (7).
(7)  (7)
ここで、 POWs は現フレームの入力信号の全周波数の振幅スぺタ ト ルのパワー、 POWn は現フレームの全周波数の推定雑音振幅スぺク ト ルのパワー、 Gは現フレームの振幅抑圧係数、 rは引き去り率である。 図 1 7に示すように、 引き去り率 rが大きいと振幅抑圧係数 Gは大きく なる。 引き去り率 rが小さいと振幅抑圧係数 Gは小さくなる。  Here, POWs is the power of the amplitude spectrum of all the frequencies of the input signal of the current frame, POWn is the power of the estimated noise amplitude spectrum of all the frequencies of the current frame, and G is the amplitude suppression coefficient of the current frame. , R is the withdrawal rate. As shown in Fig. 17, when the withdrawal rate r is large, the amplitude suppression coefficient G increases. When the withdrawal rate r is small, the amplitude suppression coefficient G becomes small.
更に、 S 2では、 次式 (8) によりサンプル毎の振幅調整係数 g (n Further, in S2, the amplitude adjustment coefficient g (n
) を求める。 ).
g (n) = g (n - 1 ) AR + G ( 1— AR) (8)  g (n) = g (n-1) AR + G (1— AR) (8)
但し、 0く ARく 1である。  However, it is 0 and AR is 1.
また、 g (n) は現フレームの第 nサンプルの振幅調整係数、 n— 1 はその前のサンプルを表わす。 図 1 7に示すように、 引き去り率 rが大 きいと振幅抑圧係数 Gが大きくなり、 振幅調整係数 g (n) は大きくな る。 引き去り率 rが小さいと振幅抑圧係数 Gが小さくなり、 振幅調整係 数 g (n) は小さくなる。  G (n) is the amplitude adjustment coefficient of the n-th sample of the current frame, and n-1 is the previous sample. As shown in Fig. 17, when the withdrawal rate r is large, the amplitude suppression coefficient G increases, and the amplitude adjustment coefficient g (n) increases. When the removal ratio r is small, the amplitude suppression coefficient G becomes small, and the amplitude adjustment coefficient g (n) becomes small.
更に、 S 3では、 逆フーリエ変換部 5からの出力振幅 1 2を Sinで表 わして、 次式 (9) により、 振幅調整係数 g (n) を乗算して出力信号 1 4を得る。 Further, in S 3, the output amplitude 1 2 from the inverse Fourier transform unit 5 and Table Wa at S in, by the following equation (9), the output signal is multiplied by the amplitude adjustment coefficient g (n) Get 1 4
Sout (n) =Sin (n) X g (n) (9) S out (n) = S in (n) X g (n) (9)
図 4は、 振幅調整フィルタ部 6の入出力特性をを示す図であり、 雑音 区間では引き去り率 rが小さいため振幅調整係数 g (n) が小さくて、 出力側では抑圧が大きくなって出力値が少なくなり、 音声区間では逆に 引き去り率 rが大きいため振幅調整係数 g (n) が大きくて、 出力側で は抑圧が小さくなって出力値が相対的に多くなる。 FIG. 4 is a diagram showing the input / output characteristics of the amplitude adjustment filter unit 6.In the noise section, the extraction rate r is small, so that the amplitude adjustment coefficient g (n) is small, and the suppression on the output side is large, and the output value is large. On the other hand, the amplitude adjustment coefficient g ( n ) is large because the withdrawal rate r is large in the voice section. On the output side, the suppression is small and the output value is relatively large.
この実施の形態によれば、 雑音フレームや子音等パワーの小さい音声 フレームでは周波数領域における雑音の引き去り量を小さくするので、 雑音フレームでミュージカルノイズの発生を押さえ、 音声区間の削れを 防ぐことができる。 このときの引き去り前、 引き去り後の雑音フレーム の振幅スペク トルの様子を表したのが、 図 5である。  According to this embodiment, the amount of noise removal in the frequency domain is reduced in a noise frame or a consonant voice frame having a small power, so that the generation of musical noise in the noise frame can be suppressed and the voice section can be prevented from being cut. . Figure 5 shows the amplitude spectrum of the noise frame before and after the subtraction.
引き去り率 rを 1. 0として大きく雑音を引き去った場合は、 引き去 り後の雑音の振幅スぺク トルの強勢な周波数の成分だけが残留しミュー ジカルノイズのもととなる。 引き去り率 rを 0. 5として引き去りを弱 めた場合は、 引き去り後の雑音の振幅スぺク トルにおける一部周波数へ の振幅の片寄りが軽減され、 ミュージカルノイズは発生しない。 この引 き去り量を減らしたフレームでは、 それに応じて振幅抑圧を行っている ので、 雑音抑圧量が不足することも防ぐことができる。 また、 本実施の 形態における振幅調整フィルタ部 6では、 式 (8) によって振幅調整係 数が 1つのフレーム内でもサンプル毎に順に時間軸方向で徐々に変化す るようにしたので、 音声区間の立ち上がり等急激な振幅調整を行っても 、 自然な出力が得られる。 When the removal rate r is set to 1.0 and the noise is largely removed, only the frequency component of the strong amplitude spectrum of the removed noise remains and becomes a source of musical noise. When the withdrawal rate r is set to 0.5 and the withdrawal is weakened, the deviation of the amplitude to a part of the frequency in the amplitude spectrum of the noise after the withdrawal is reduced, and no musical noise is generated. In the frame in which the removal amount is reduced, the amplitude is suppressed accordingly, so that it is possible to prevent the noise suppression amount from becoming insufficient. In addition, in the amplitude adjustment filter unit 6 according to the present embodiment, the amplitude adjustment coefficient is gradually changed in the time axis direction for each sample even in one frame by equation (8), so that Even if the amplitude is suddenly adjusted such as at the rising edge, a natural output can be obtained.
実施の形態 2. Embodiment 2.
振幅調整は、 周波数領域で実施してもよい。 即ち、 フーリエ変換後に 振幅調整フィルタ部を設けて、 振幅調整後の信号を逆フーリエ変換する 構成とする。 The amplitude adjustment may be performed in the frequency domain. That is, an amplitude adjustment filter section is provided after the Fourier transform, and the signal after the amplitude adjustment is subjected to the inverse Fourier transform. Configuration.
本発明の実施の形態 2の方法を、 図に基づいて説明する。  The method according to the second embodiment of the present invention will be described with reference to the drawings.
図 6は、 実施の形態に係わるスぺク トラルサブトラクシヨン方法の構 成を示すブロック図である。  FIG. 6 is a block diagram showing a configuration of a spectral subtraction method according to the embodiment.
図 6に基づいて、 本方法の動作を説明する。  The operation of the method will be described with reference to FIG.
図 6の構成は、 図 1における振幅調整フィルタを、 サブトラクシヨン • フィルタ 4の直後に移動して、 振幅調整を周波数領域で行う構成を表 しており、 その他の部分の動作は、 実施の形態 1の動作と同一である。 図において、 サブトラクション ' フィルタ部 4は、 式 (6 ) によって 引き去り率を算出して、 この引き去り率に基づいて入力振幅スぺク トル 9から推定雑音振幅スぺク トル 1 0を減算して、 出力振幅スぺク トルを 出力する。 振幅調整フィルタ部 1 5は、 この引き去り率 rに基づいて算 出された振幅調整係数を出力振幅スぺク トルに掛けて最終的な出力振巾; スぺク トルを出力する。  The configuration in FIG. 6 shows a configuration in which the amplitude adjustment filter in FIG. 1 is moved immediately after the subtraction filter 4 and the amplitude adjustment is performed in the frequency domain. The operation is the same as that of mode 1. In the figure, the subtraction ′ filter unit 4 calculates the subtraction rate by the equation (6), and subtracts the estimated noise amplitude spectrum 10 from the input amplitude spectrum 9 based on the subtraction rate, Outputs the output amplitude spectrum. The amplitude adjustment filter section 15 multiplies the amplitude adjustment coefficient calculated based on the withdrawal rate r by the output amplitude spectrum, and outputs a final output amplitude; spectrum.
この実施の形態によれば、 雑音フレームや子音等パワーの小さい音声 フレームでは引き去り強度を小さくするので、 雑音フレームでミユージ カルノイズの発生を押さえ、 子音区間の変形や消滅を防ぐことができ、 また、 引き去り量を減らしたフレームでは、 それに応じて振幅抑圧を行 つているので、 雑音抑圧量が不足することも防ぐことができる。  According to this embodiment, since the withdrawal strength is reduced in a noise frame or a voice frame having a low power such as a consonant, the generation of musical noise can be suppressed in the noise frame, and the deformation or disappearance of the consonant section can be prevented. In frames with a reduced amount of subtraction, amplitude suppression is performed accordingly, so that it is possible to prevent shortage of noise suppression.
また、 振幅調整係数はフレームに一回算出すればよく、 実施の形態 1 の図 2の S 3ように信号 1サンプル毎に振幅調整係数を計算しなくても よいので、 少ない演算量で実現できる。  Also, the amplitude adjustment coefficient need only be calculated once per frame, and it is not necessary to calculate the amplitude adjustment coefficient for each sample of the signal as in S3 of FIG. 2 of the first embodiment. .
実施の形態 3 . Embodiment 3.
先の実施の形態では、 入力信号は、 所定フレームで区切られた区間毎 に音声を抽出していたので、 フレーム (区間) の変化点で入力信号が不 連続となり、 異音感が生じることがある。 本実施の形態では、 これを改 善してフレーム間の信号変化を円滑化する。 In the above embodiment, since the input signal extracts the voice for each section divided by the predetermined frame, the input signal becomes discontinuous at the change point of the frame (section), which may cause abnormal noise. . In the present embodiment, this is Better to smooth signal changes between frames.
以下、 本発明の実施の形態の方法を、 図に基づいて説明する。  Hereinafter, a method according to an embodiment of the present invention will be described with reference to the drawings.
図 7は、 本発明の実施の形態 3におけるスぺク トラルサブトラクショ ン方法の構成を示すブロック図である。 入力信号生成部 1 9と波形整形 処理部 20が新しい要素であり、 その他は、 図 1の各要素と同じである 図 8は、 本実施の形態 3の雑音軽減方法における入力信号 7の例を示 す図で、 図 9は、 本実施の形態 3の雑音軽減方法における入力信号生成 部が行う入力信号に対して乗算する重み付け関数を示した図である。 また、 図 1 0は、 本実施の形態 3の雑音軽減方法における逆フーリエ 変換部 5の出力信号である 5 aの例を示す図である。  FIG. 7 is a block diagram showing a configuration of a spectral subtraction method according to Embodiment 3 of the present invention. The input signal generation unit 19 and the waveform shaping processing unit 20 are new elements, and the other elements are the same as those in FIG. 1. FIG. 8 shows an example of the input signal 7 in the noise reduction method according to the third embodiment. FIG. 9 is a diagram illustrating a weighting function for multiplying the input signal performed by the input signal generation unit in the noise reduction method according to the third embodiment. FIG. 10 is a diagram illustrating an example of an output signal 5 a of the inverse Fourier transform unit 5 in the noise reduction method according to the third embodiment.
図 7の構成図に基づいて、 本実施の形態における雑音軽減方法の動作 を説明する。  The operation of the noise reduction method according to the present embodiment will be described based on the configuration diagram of FIG.
A D変換後の入力信号 7の時系列振幅を表したのが図 8で、 これが入 力信号生成部 1 9に入力される。 入力信号生成部 1 9は、 現フレームで 雑音抑圧処理するフレーム信号を s ( i ) ( i =n, n + 1 , . . . , η + f r ) とした場合、 フレーム信号に前後の信号を加えた s ( i ) ( i = n - a , n - a + 1 , . . . , n + f r + s m+ a) を切出す。 図 8の Aは、 フレームの前の設定区間である。 図 8の Bと Cは、 フレーム の後の設定区間である。 次に、 図 9の重み付け関数を掛けて、 次式 (1 0) のように重み付けられた信号 s ' ( i ) ( i = n - a , n - a + 1 , . . . , n+ f r + s m+ a ) を算出する。 図 8の Aと Cの設定区間 では、 端が 0に近づく重み付けがされる。 図 8の Bの設定区間では、 フ レーム信号と同じ 1の重み付けがされる。  FIG. 8 shows the time-series amplitude of the input signal 7 after the AD conversion, and this is input to the input signal generation unit 19. When the frame signal to be subjected to noise suppression processing in the current frame is s (i) (i = n, n + 1,..., Η + fr), the input signal Cut out the added s (i) (i = n−a, n−a + 1,..., N + fr + sm + a). A in Fig. 8 is the set section before the frame. B and C in Fig. 8 are the set sections after the frame. Next, the weighting function of FIG. 9 is multiplied to obtain a signal s ′ (i) (i = n−a, n−a + 1,..., N + fr + s m + a) is calculated. In the setting section of A and C in Fig. 8, the weight is set so that the end approaches 0. In the setting section B in Fig. 8, the same 1 weight as the frame signal is applied.
s ' ( i ) = S ( i ) - W ( i -n + a)  s' (i) = S (i)-W (i -n + a)
( l = n— a , n— a + 1 , . . . , n + f r + s m+ a ) ( 1 0) (l = n—a, n—a + 1,..., n + fr + sm + a) ( Ten)
この信号 s ' ( i ) ( i = n— a , n— a + 1 , . . . , n + f r + s m+ a ) をフーリエ変換部 1がフーリエ変換する。 逆フーリエ変換部 5の出力信号を u ( i ) ( i = 0 , 1, . . . , 2 a + f r + s m) と すると、 図 1 0は、 u ( i ) を表しており、 時間的には u ( i ) ( i = 0, 1, . . . , 2 a + f r + s m) は、 s (n - a + i ) ( i = 0, 1 , . . . , 2 a + f r + s m) に対応する。  The signal s ′ (i) (i = n−a, n−a + 1,..., N + fr + sm + a) is subjected to Fourier transform by the Fourier transform unit 1. Assuming that the output signal of the inverse Fourier transform unit 5 is u (i) (i = 0, 1,..., 2a + fr + sm), FIG. 10 shows u (i), and U (i) (i = 0, 1, ..., 2a + fr + sm) is s (n-a + i) (i = 0, 1, ..., 2a + fr + sm).
波形整形処理部 20は、 図 1 8に示すように、 逆フーリエ変換された 信号でフレーム信号に対応する u ( i ) ( i = a , a + 1 , . . . , a + f r ) がフレーム間で不連続にならないようにする波形整形処理をし て出力する。 次式 (1 1 ) は、 前フレームの後の設定区間 Bの逆フーリ ェ変換出力の u ( i ) ( i = a + f r, a + f r + 1 , . . . , a + f r + s m) を u p ( i ) ( i = 0 , 1, . . . , s m) としたときの波 形整形処理後の出力信号を表す式である。  As shown in FIG. 18, the waveform shaping processing unit 20 converts the inverse Fourier-transformed signal corresponding to the frame signal into u (i) (i = a, a + 1,..., A + fr). Waveform shaping processing to prevent discontinuity between the signals is output. The following equation (11) gives u (i) (i = a + fr, a + fr + 1, ..., a + fr + sm) of the inverse Fourier transform output of the set section B after the previous frame. This is the expression representing the output signal after waveform shaping processing when is set to up (i) (i = 0, 1,..., Sm).
u ' ( i ) - (u ( i ) - i + u p ( i - a)  u '(i)-(u (i)-i + up (i-a)
• s m— 1ノ ) / s m  • s m— 1 no) / s m
( i = a , a + 1 , . . . , a + s m) ( 1 1 )  (i = a, a + 1,..., a + s m) (1 1)
u ' ( i ) :波形整形処理後の信号  u '(i): Signal after waveform shaping processing
この実施の形態によれば、 端を 0に近付ける重み付けした信号をフー リエ変換しているので、 雑音軽減能力が向上する。 また、 固定小数点演 算ディジタル信号プ口セッサ (固定小数点 D S P (ディジタルシグナル プロセッサ) ) で本システムを実現した際に、 重み付けしたために精度 が劣化する両端の信号 (図 1 8の Aと C) を波形整形処理に使用しない ので、 出力の精度の確保が容易になり、 フレーム境界での異音を防ぐこ とができる。  According to this embodiment, the weighted signal whose end approaches 0 is subjected to Fourier transform, so that the noise reduction capability is improved. Also, when this system was implemented with a fixed-point arithmetic digital signal processor (fixed-point DSP (digital signal processor)), the signals at both ends (A and C in Fig. 18), whose accuracy deteriorates due to the weighting, are reduced. Since it is not used for waveform shaping, it is easy to ensure output accuracy and to prevent abnormal noise at frame boundaries.
D S Pを用いると 1 6 b i t程度の短い語長の固定小数点の演算を行 うため、 ダイナミックレンジが大きい信号を処理すると、 精度劣化が生 じる。 端を 0に近づけるように重み付けすると、 信号のダイナミックレ ンジが拡大するので、 D S Pを使うと精度劣化が起きる。 この実施の形 態では、 精度劣化が起きた端の信号を波形整形処理に用いないので、 波 形整形処理の精度が向上する。 Using DSP, fixed-point arithmetic with a word length as short as 16 bits is performed. Therefore, when processing signals with a large dynamic range, accuracy degradation occurs. Weighting the edges closer to zero will increase the dynamic range of the signal, and the use of DSPs will degrade accuracy. In this embodiment, the accuracy of the waveform shaping process is improved because the signal at the end where the accuracy degradation has occurred is not used for the waveform shaping process.
実施の形態 4 . Embodiment 4.
スぺク トラルサブトラクションの雑音軽減効果を高める方法として、 雑音区間の判別を容易にして、 かつ、 引きすぎによる音声の削れを軽減 する方法が考える。  As a method to enhance the noise reduction effect of spectral subtraction, we consider a method that makes it easy to identify noise sections and that reduces speech shaving due to overdrawing.
以下、 本発明の実施の形態 4の方法を図に基づいて説明する。  Hereinafter, the method according to the fourth embodiment of the present invention will be described with reference to the drawings.
図 1 1は、 本発明の実施の形態 4におけるスぺク トラルサブトラタシ ヨン方法の構成を示すブロック図である。 図において、 新規な要素とし て引き去り率算出部 1 6がある。 他の要素は、 図 1の同番号のそれと同 じものである。  FIG. 11 is a block diagram showing a configuration of a spectral subtraction method according to Embodiment 4 of the present invention. In the figure, there is a withdrawal rate calculator 16 as a new element. The other elements are the same as those with the same numbers in FIG.
図 1 2は、 本実施の形態の雑音軽減方法における動作を説明する図で ある。 図において、 実線は雑音区間の入力パワー P〇Ws を、 点線は平 均雑音パワー P O WAVEを、 鎖線は雑音区間判定しきい値 P O WTHを表す 。 なお、 P O WTHとは、 式 (3 ) で述べた前雑音フレームでのしきい値 T Hである。 FIG. 12 is a diagram illustrating an operation in the noise reduction method according to the present embodiment. In the figure, the solid line represents the input power P〇Ws in the noise section, the dotted line represents the average noise power POW AVE , and the chain line represents the noise section determination threshold POW TH . Note that POW TH is the threshold TH in the pre-noise frame described in equation (3).
図 1 1の全体構成図に基づいて、 本実施の形態における雑音軽減方法 の動作を説明する。  The operation of the noise reduction method according to the present embodiment will be described based on the overall configuration diagram of FIG.
本構成は、 従来法のスぺク トラルサブトラクションの全体構成に引き 去り率算出部 1 6を追加した構成となっている。 図 1 1において、 サブ トラクシヨン . フィルタ部 4のフィルタは、 式 ( 1 ) の演算を行う。 雑 音区間判定部 2では、 雑音区間判定しきい値 P O WTHを算出し、 入力パ ヮ一がこのしきい値を下回った場合、 雑音フレームと判定する。 引き去り率算出部 1 6は、 この判定結果に基づいて現在に近い複数雑 音フレームの平均パワーを算出して平均雑音パワーとし、 下記の式 (1 2 ) に基づいて雑音区間判定しきい値を P〇WTH、 平均雑音パワーを P This configuration has a configuration in which a removal rate calculation unit 16 is added to the overall configuration of the spectral subtraction of the conventional method. In FIG. 11, the filter of the subtraction filter unit 4 performs the operation of equation (1). The noise interval determination unit 2 calculates a noise interval determination threshold value POW TH , and determines that the noise frame is a noise frame when the input level falls below this threshold value. The subtraction rate calculation unit 16 calculates the average power of a plurality of noise frames close to the current based on the determination result and sets the average power as the average noise power. Based on the following equation (12), the noise section determination threshold is determined. P〇W TH , average noise power P
R 1 (! 2 ) R 1 (! 2 )
Figure imgf000022_0001
Figure imgf000022_0001
O WAVE としたときの雑音区間判定しきい値 1 7と平均雑音パワーとの 比 r lを計算し、 これを雑音変動の大きさ 1 8として出力する。 Calculate the ratio rl between the noise section judgment threshold value 17 and the average noise power when OW AVE is set, and output this as the noise fluctuation magnitude 18.
この雑音変動の大きさ r 1を、 式 ( 1 ) の雑音スぺク トルの引き去り 率 rとする。 雑音区間判定しきい P O WTH値は、 パワー変動の大きい雑 音の場合に、 正しく雑音区間と判定できる高めの値を用いる。 このとき の雑音区間における入力パワーの変化に対して、 平均雑音パワーと雑音 区間判定しきい値の変化を図 1 2に示す。 The magnitude r1 of this noise fluctuation is defined as the noise vector removal rate r in equation (1). The noise section determination threshold POW TH value uses a high value that can be correctly determined as a noise section in the case of noise with large power fluctuations. Figure 12 shows the changes in the average noise power and the noise section determination threshold with respect to the change in the input power in the noise section at this time.
この実施の形態によれば、 雑音のパワー変動が大きい場合でも、 図 1 2のように、 雑音フレームの平均雑音パワー P O WAVE は、 雑音区間判 定しきい値 P O WTHよりも小さい値をとるので、 引き去り率 r 1は 1よ り小さい値をとることになる。 結果として、 推定雑音振幅スペク トルの 引き去りを抑制する効果が得られ、 音声の削れを軽減でき、 かつ、 推定 雑音振幅スぺク トルの更新も正常に行われる。 According to this embodiment, even when the power fluctuation of the noise is large, as shown in FIG. 12, the average noise power POW AVE of the noise frame takes a value smaller than the noise section determination threshold value POW TH . Therefore, the withdrawal rate r 1 is smaller than 1. As a result, the effect of suppressing the removal of the estimated noise amplitude spectrum can be obtained, the voice shaving can be reduced, and the estimated noise amplitude spectrum is updated normally.
実施の形態 5 . Embodiment 5
雑音区間の雑音とみなされる入力信号を振幅抑圧して、 不快残留雑音 を軽減する方法を説明する。  A method for reducing the unpleasant residual noise by suppressing the amplitude of the input signal regarded as noise in the noise section will be described.
以下、 本発明の実施の形態 5の方法を、 図に基づいて説明する。  Hereinafter, the method according to the fifth embodiment of the present invention will be described with reference to the drawings.
図 1 3は、 本発明の実施の形態 5における雑音軽減方法の主要構成部 を示すブロック図である。  FIG. 13 is a block diagram showing main components of a noise reduction method according to Embodiment 5 of the present invention.
また、 図 1 4は、 本実施の形態の雑音軽減方法における動作を説明す る図である。 本構成は、 雑音区間の振幅抑圧によって雑音を軽減する方法で、 振幅 抑圧を行う振幅調整係数を現在の入力パワーと平均雑音パワーに依存し て決定する。 FIG. 14 is a diagram illustrating an operation in the noise reduction method according to the present embodiment. In this configuration, the noise is reduced by suppressing the amplitude in the noise section. The amplitude adjustment coefficient for performing the amplitude suppression is determined depending on the current input power and the average noise power.
図 1 3を用いて動作を説明する。  The operation will be described with reference to FIGS.
雑音区間判定部 30 1は、 入力信号 304から図 1 2の実線で示され る入力パワー POWs 30 5を算出し、 実施の形態 1同様の雑音区間判 定しきい値 POWTHによる雑音区間判定を行う。 平均雑音パワー算出部 3 02は、 この雑音区間判定結果により、 現在のフレームに近い過去の 複数の雑音フレームの平均パワーを算出して、 図 1 2の点線で示される 平均雑音パワー P OWAVE 3 06とする。 The noise interval determination unit 301 calculates the input power POWs 305 indicated by the solid line in FIG. 12 from the input signal 304, and performs the noise interval determination using the noise interval determination threshold POW TH similar to the first embodiment. Do. The average noise power calculation section 302 calculates the average power of a plurality of past noise frames close to the current frame based on the result of the noise section determination, and calculates the average noise power P OW AVE 3 shown by the dotted line in FIG. 06.
振幅調整フィルタ部 3 0 3は、 入力信号の入力パワー P〇Ws 30 5 と平均雑音パワー P OWAVE 3 0 6から式 ( 1 3) によって振幅抑圧 係数 Gを算出し、 これを、 そのまま振幅調整係数とし、 入力信号 3 04 に得た振幅調整係数を掛けて出力信号 30 7とする。 The amplitude adjustment filter section 303 calculates the amplitude suppression coefficient G from the input power P〇Ws 305 of the input signal and the average noise power P OW AVE 306 according to equation (13), and then directly adjusts the amplitude. The output signal 307 is obtained by multiplying the input signal 304 by the obtained amplitude adjustment coefficient.
G : POWs POWAVE G : POWs POWAVE
POWs  POWs
(POWs-POWAVE≥ 0) (POW s -POW AVE ≥ 0)
G= 0 G = 0
(POWs-POWAVE< 0) (POW s -POW AVE <0)
( 1 3)  ( 13)
また、 式 (1 3) で算出した振幅抑圧係数 Gを用いて、 フレーム間で 振幅調整係数が滑らかに変化するようにするために、 式 (8) のように 、 Gに 1サンプル毎に平滑操作を行い最終的な振幅調整係数 g (n) と 図 1 4は、 入力信号 304の入力パワー P OWs 30 5と振幅調整後 の出力信号 30 7の出力パワーの関係を図示したものである。 図 1 4の点線は、 振幅調整フィルタを使わず、 入力信号をそのまま出 力した場合である。 即ち、 点線は、 入力信号の入力パワー =出力信号の 出力パワーとなる場合を示している。 実線は、 振幅調整フィルタを使つ た場合の入力パワー P O Ws と出力パワーの関係を示している。 振幅抑 圧により、 点線より小さい値が出力されることを示している。 また、 入 力パワーが平均雑音パワー P〇WAVE より小さい場合は、 出力信号パヮ 一はゼロになることを示している。 Also, in order to make the amplitude adjustment coefficient change smoothly between frames using the amplitude suppression coefficient G calculated by equation (13), as shown in equation (8), G is smoothed for each sample. FIG. 14 shows the relationship between the input power P OWs 305 of the input signal 304 and the output power of the output signal 307 after the amplitude adjustment. The dotted line in Fig. 14 shows the case where the input signal was output as it was without using the amplitude adjustment filter. That is, the dotted line indicates a case where the input power of the input signal is equal to the output power of the output signal. The solid line shows the relationship between the input power POWs and the output power when the amplitude adjustment filter is used. This indicates that a value smaller than the dotted line is output by amplitude suppression. Also, when the input power is the average noise power P_〇_W AVE smaller than the output signal Pawa one indicates that zero.
この実施の形態によれば、 引き去り率を可変にした上に、 これに基づ レ、て振幅抑圧度を変える振幅調整フィルタ部を備えたので、 振幅調整係 数を音声の状態判定の結果から直接決定した場合のように頻繁に、 しか も強度に振幅抑圧を変化させず、 音声出力が安定する。 産業上の利用可能性  According to this embodiment, since the withdrawal rate is made variable and the amplitude adjustment filter section for changing the amplitude suppression degree based on the variable is provided, the amplitude adjustment coefficient is determined from the result of the voice state determination. Frequently, as in the case of direct determination, the amplitude suppression does not change significantly, and the audio output stabilizes. Industrial applicability
以上のように、 この発明によれば、 引き去り率を可変にした上に、 こ れに基づいて振幅抑圧度を変える振幅調整フィルタ部を備えたので、 音 声区間の削れを防ぐため等の目的で、 雑音フレーム等での引き去り強度 を小さく しても、 それに対応した振幅抑圧を行うことによって雑音を抑 圧するため、 全体として、 バランスがとれた聞き易い出力が得られる効 果がある。  As described above, according to the present invention, since the withdrawal rate is made variable and the amplitude adjustment filter section that changes the degree of amplitude suppression based on this is provided, it is possible to prevent the voice section from being cut off. Thus, even if the subtraction strength in a noise frame or the like is reduced, noise is suppressed by performing amplitude suppression corresponding to the strength, so that a balanced and easy-to-hear output can be obtained as a whole.
また更に、 振幅調整を周波数領域で行うようにしたので、 信号 1サン プル毎に振幅調整係数を計算する必要がなく、 演算量を少なくできる効 果がある。  Furthermore, since the amplitude adjustment is performed in the frequency domain, there is no need to calculate the amplitude adjustment coefficient for each signal sample, and the amount of calculation can be reduced.
また更に、 振幅調整係数がフレーム内でも時間軸方向で徐々に変化す るので、 音声区間の立ち上がり等急激な振幅調整を行っても自然な出力 が得られる。  Further, since the amplitude adjustment coefficient gradually changes in the time axis direction even within a frame, a natural output can be obtained even when a sharp amplitude adjustment such as the rising of a voice section is performed.
また更に、 現フレームの前後区間の信号も重み関数を乗算して加算す る入力信号生成部を備えたので、 フーリエ変換の対象信号が重み付けさ れて雑音スぺク トルの推定精度が向上して雑音軽減効果が高まり、 また 、 逆フーリエ変換出力の重み付けされていない信号をフレーム信号とし て出力して、 固定小数点演算ディジタル信号プロセッサでも演算が少な くて高い信号精度が得られる効果がある。 Further, the signals in the preceding and succeeding sections of the current frame are also multiplied by a weight function and added. Since the input signal generation unit includes the input signal generation unit, the target signal of the Fourier transform is weighted, the estimation accuracy of the noise spectrum is improved and the noise reduction effect is enhanced, and the unweighted signal of the inverse Fourier transform output is obtained. Is output as a frame signal, so that even with a fixed-point arithmetic digital signal processor, there is an effect that the arithmetic operation is small and high signal accuracy can be obtained.
または、 雑音区間と判定された区間の変動性に応じて雑音除去の強弱 を調整するようにしたので、 雑音の変動性が大きい場合も追従して正し い雑音判定しきい値を設定できて、 かつ、 混入音声成分が音声を削るこ とを防ぐ効果もある。  Alternatively, the strength of noise removal is adjusted according to the variability of the section determined to be a noise section, so that even when the variability of noise is large, a correct noise determination threshold can be set. It also has the effect of preventing mixed speech components from squeezing the speech.
また更に、 雑音区間の判定結果から直接振幅抑圧量を定めずに、 徐々 に変化する雑音区間の平均パワーを用いて振幅抑圧量を定めるようにし たので、 出力が雑音区間判定の頻繁な変化による悪影響が避けられる効 果がある。  Furthermore, since the amplitude suppression amount is determined using the average power of the noise section that changes gradually without directly determining the amplitude suppression amount from the determination result of the noise section, the output is caused by frequent changes in the noise section determination. This has the effect of avoiding adverse effects.

Claims

請求の範囲 The scope of the claims
1 . 所定のフレーム長に切り出した入力信号を直交変換して 得た振幅スぺク トルから推定雑音振幅スぺク トルをスぺク トラルサブト ラクシヨン ' フィルタにより減算して出力を得る雑音軽減装置において フレーム毎に上記スぺク トラルサブトラクシヨン ' フィルタによる減 算出力に、 上記振幅スぺク トルのパワーと上記推定雑音振幅スぺク トル のパワーとで決まる振幅調整係数を乗算して、 所要出力を得る振幅調整 フィルタ部を備えたことを特徴とする雑音軽減装置。 1. A noise reduction apparatus that obtains an output by subtracting an estimated noise amplitude spectrum from an amplitude spectrum obtained by orthogonally transforming an input signal cut out to a predetermined frame length by a spectral sub-fraction filter. For each frame, the required power is calculated by multiplying the reduced power by the spectral subtraction filter by an amplitude adjustment coefficient determined by the power of the amplitude spectrum and the power of the estimated noise amplitude spectrum. A noise reduction device comprising an amplitude adjustment filter for obtaining an output.
2 . スぺク トラルサブトラクション · フィルタは、 推定雑音 振幅スぺク トルの減算の引き去り率を上記推定雑音振幅スぺク トルに応 じて可変とし、 振幅調整フィルタ部は、 引き去り率に応じて振幅調整係 数を可変とすることを特徴とする請求項 1記載の雑音軽減装置。  2. The spectral subtraction filter makes the subtraction rate of the subtraction of the estimated noise amplitude spectrum variable according to the above-mentioned estimated noise amplitude spectrum, and the amplitude adjustment filter section changes the subtraction rate according to the subtraction rate. 2. The noise reduction device according to claim 1, wherein the amplitude adjustment coefficient is variable.
3 . 上記振幅調整フィルタ部は、 引き去り率が大きい場合、 振幅調整係数を大きくすることにより、 音声区間で雑音抑圧を強め、 か つ、 出力信号の出力値を大きくするとともに、 上記振幅調整フィルタ部 は、 引き去り率が小さい場合、 振幅調整係数を小さくすることにより、 雑音区間で雑音抑圧を弱め、 かつ、 出力信号の出力値を小さくすること を特徴とする請求項 2記載の雑音軽減装置。  3. When the withdrawal rate is large, the amplitude adjustment filter unit increases the amplitude adjustment coefficient to enhance noise suppression in a voice section, increases the output value of an output signal, and increases the output value of the output signal. 3. The noise reduction apparatus according to claim 2, wherein when the withdrawal rate is small, the noise suppression is reduced in a noise section and the output value of the output signal is reduced by reducing the amplitude adjustment coefficient.
4 . 振幅調整フィルタ部は、 振幅調整係数をスぺク トラルサ ブトラタシヨン · フィルタによる減算出力を逆直交変換した時間領域の 振幅スぺク トルに乗算することを特徴とする請求項 1記載の雑音軽減装 置。  4. The noise reduction according to claim 1, wherein the amplitude adjustment filter section multiplies an amplitude spectrum in a time domain obtained by performing an inverse orthogonal transform on an output obtained by subtracting the amplitude adjustment coefficient from the spectral subtralation filter. Equipment.
5 . 振幅調整フィルタ部は、 振幅調整係数をフレーム毎に周 波数領域で振幅スぺク トルに乗算し、 逆直交変換して出力を得るように したことを特徴とする請求項 1記載の雑音軽減装置。 5. The amplitude adjustment filter section multiplies the amplitude spectrum by the amplitude adjustment coefficient in the frequency domain for each frame, and performs inverse orthogonal transform to obtain an output. The noise reduction device according to claim 1, wherein
6 . 振幅調整係数は、 現フレームの入力信号の振幅スぺタ ト ルのパワーと、 現フレームの推定雑音振幅スぺク トルのパワーとの差に 基づいて得た現フレームの振幅調整係数に、 前フレームで得た振幅調整 係数の値を重み付け加算した値としたことを特徴とする請求項 1記載の 雑音軽減装置。  6. The amplitude adjustment coefficient is the amplitude adjustment coefficient of the current frame obtained based on the difference between the power of the amplitude signal of the input signal of the current frame and the power of the estimated noise amplitude spectrum of the current frame. 2. The noise reduction apparatus according to claim 1, wherein the value of the amplitude adjustment coefficient obtained in the previous frame is a value obtained by weighting and adding.
7 . 所定のフレーム長に切り出した入力信号を直交変換して 得た振幅スぺク トルから推定雑音振幅スぺク トルをスぺク トラルサブト ラタシヨン · フィルタにより減算して出力を得る雑音軽減装置において 現フレームの前後の設定区間を切り出して端が 0に近くなるように、 1以下の重み関数を乗算した後に現フレームの前後に付加する入力信号 生成部を備えて、 該入力信号生成部の出力を入力信号として振幅スぺク トルを求めるようにしたことを特徴とする雑音軽減装置。  7. A noise reduction device that obtains an output by subtracting an estimated noise amplitude spectrum from an amplitude spectrum obtained by orthogonally transforming an input signal cut out to a predetermined frame length using a spectral subtralation filter. An input signal generation unit that cuts out a set section before and after the current frame and multiplies by a weighting function of 1 or less and adds the input signal before and after the current frame so that the end is close to 0, and outputs the output signal of the input signal generation unit A noise reduction apparatus characterized in that an amplitude spectrum is obtained by using the input signal as an input signal.
8 . 1の重み関数が乗算された前フレームの後の設定区間を 用いて現フレームの波形整形を行う波形整形処理部を備えたことを特徴 とする請求項 7記載の雑音軽減装置。  8. The noise reduction apparatus according to claim 7, further comprising a waveform shaping processing unit that shapes a waveform of the current frame using a set section after the previous frame multiplied by the weight function of 8.1.
9 . 所定のフレーム長に切り出した入力信号を直交変換して 得た振幅スぺク トルから推定雑音振幅スぺク トルをスぺク トラルサブト ラタシヨン · フィルタにより減算して出力を得る雑音軽減装置において 上記推定雑音振幅スぺク トルの複数フレーム分から平均雑音パワーを 求めて雑音フレームの判定しきい値と比較して引き去り率を得る引き去 り率算出部を備えて、  9. A noise reduction device that obtains an output by subtracting an estimated noise amplitude spectrum from an amplitude spectrum obtained by orthogonally transforming an input signal cut out to a predetermined frame length by a spectral subtralation filter. A subtraction rate calculation unit that obtains an average noise power from a plurality of frames of the estimated noise amplitude spectrum and compares the average noise power with a determination threshold of the noise frame to obtain a removal rate;
上記スぺク トラルサブトラクシヨン ' フィルタの減算に用いる引き去 り率を上記引き去り率としたことを特徴とする雑音軽減装置。 A noise reduction apparatus characterized in that a subtraction rate used for subtraction of the spectral subtraction filter is defined as the subtraction rate.
1 0 . 推定雑音振幅スぺク トルの複数フレーム分から平均雑音 パワーを求める平均雑音パワー算出部を備えて、 10. An average noise power calculator that calculates the average noise power from a plurality of frames of the estimated noise amplitude spectrum is provided.
振幅調整フィルタ部は、 フレーム毎にスぺク トラルサブトラクシヨン • フィルタによる減算出力に、 振幅スぺク トルのパワーと上記求めた平 均雑音振幅パワーとで決まる振幅調整係数を乗算して、 所要出力を得る ようにしたことを特徴とする請求項 1記載の雑音軽減装置。  The amplitude adjustment filter section multiplies the subtraction output by the spectral subtraction filter for each frame by an amplitude adjustment coefficient determined by the power of the amplitude spectrum and the average noise amplitude power obtained as described above. The noise reduction device according to claim 1, wherein a required output is obtained.
1 1 . 所定のフレーム長に切り出した入力信号を直交変換して 得た振幅スぺク トルから推定雑音振幅スぺク トルをスぺク トラルサブト ラクシヨ ン · フィルタにより減算して出力を得る雑音軽減方法において 上記スぺク トラルサブトラクシヨン ' フィルタの引き去り率を上記推 定雑音振幅スぺク トルに応じて可変する工程と、  1 1. Noise reduction to obtain the output by subtracting the estimated noise amplitude spectrum from the amplitude spectrum obtained by orthogonally transforming the input signal cut out to a predetermined frame length by using a spectral subtrac tion filter Varying the removal rate of the spectral subtraction 'filter in accordance with the estimated noise amplitude spectrum.
フレーム毎に上記スぺク トラルサブトラクシヨン · フィルタによる減 算出力に、 上記振幅スぺク トルのパワーと上記推定雑音振幅スぺク トル のパワーとで決まる振幅調整係数を乗算して、 所要出力を得る工程を備 えたことを特徴とする雑音軽減方法。  For each frame, the required power is calculated by multiplying the reduced power by the spectral subtraction filter by an amplitude adjustment coefficient determined by the power of the amplitude spectrum and the power of the estimated noise amplitude spectrum. A noise reduction method comprising a step of obtaining an output.
1 2 . 所定のフレーム長に切り出した入力信号を直交変換して 得た振幅スぺク トルから推定雑音振幅スぺク トルをスぺク トラルサブト ラクシヨン · フィルタにより減算して出力を得る雑音軽減方法において 上記推定雑音振幅スぺク トルの複数フレーム分から平均雑音パワーを 求めて雑音フレームの判定しきい値と比較して引き去り率を得る引き去 り率算出工程と、  1 2. A noise reduction method in which an estimated noise amplitude spectrum is subtracted from an amplitude spectrum obtained by orthogonally transforming an input signal cut out to a predetermined frame length by a spectral subtraction filter to obtain an output. A subtraction rate calculating step of obtaining an average noise power from a plurality of frames of the estimated noise amplitude spectrum and comparing the average noise power with a noise frame determination threshold to obtain a removal rate;
上記スぺク トラルサブトラクシヨン · フィルタの引き去り率を上記引 き去り率とする工程を備えたことを特徴とする雑音軽減方法。  A noise reduction method, comprising a step of setting the removal rate of the spectral subtraction filter to the removal rate.
PCT/JP1998/005512 1998-03-30 1998-12-07 Noise reduction device and a noise reduction method WO1999050825A1 (en)

Priority Applications (3)

Application Number Priority Date Filing Date Title
CA002291826A CA2291826A1 (en) 1998-03-30 1998-12-07 Noise reduction device and a noise reduction method
AU13525/99A AU721270B2 (en) 1998-03-30 1998-12-07 Noise reduction apparatus and noise reduction method
EP98957196A EP0992978A4 (en) 1998-03-30 1998-12-07 Noise reduction device and a noise reduction method

Applications Claiming Priority (2)

Application Number Priority Date Filing Date Title
JP10/84174 1998-03-30
JP8417498 1998-03-30

Publications (1)

Publication Number Publication Date
WO1999050825A1 true WO1999050825A1 (en) 1999-10-07

Family

ID=13823136

Family Applications (1)

Application Number Title Priority Date Filing Date
PCT/JP1998/005512 WO1999050825A1 (en) 1998-03-30 1998-12-07 Noise reduction device and a noise reduction method

Country Status (6)

Country Link
EP (1) EP0992978A4 (en)
KR (1) KR100314332B1 (en)
CN (1) CN1258368A (en)
AU (1) AU721270B2 (en)
CA (1) CA2291826A1 (en)
WO (1) WO1999050825A1 (en)

Cited By (6)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
WO2006064699A1 (en) * 2004-12-17 2006-06-22 Waseda University Sound source separation system, sound source separation method, and acoustic signal acquisition device
WO2006123721A1 (en) * 2005-05-17 2006-11-23 Yamaha Corporation Noise suppression method and device thereof
US7706550B2 (en) 2004-01-08 2010-04-27 Kabushiki Kaisha Toshiba Noise suppression apparatus and method
JP2015002505A (en) * 2013-06-18 2015-01-05 パイオニア株式会社 Noise reducer, broadcast receiver and noise reduction method
US9318122B2 (en) 2014-04-18 2016-04-19 Fujitsu Limited Audio signal processing apparatus and audio signal processing method
US9754606B2 (en) 2012-05-01 2017-09-05 Ricoh Company, Ltd. Processing apparatus, processing method, program, computer readable information recording medium and processing system

Families Citing this family (10)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
WO2006046293A1 (en) * 2004-10-28 2006-05-04 Fujitsu Limited Noise suppressor
GB2422237A (en) * 2004-12-21 2006-07-19 Fluency Voice Technology Ltd Dynamic coefficients determined from temporally adjacent speech frames
KR100686382B1 (en) 2005-07-08 2007-02-22 엔에이치엔(주) Messenger Notification System and Method Using Synchronization Server
CN1822092B (en) * 2006-03-28 2010-05-26 北京中星微电子有限公司 Method and its device for elliminating background noise in speech input
CN101166232B (en) * 2006-10-19 2010-05-12 华晶科技股份有限公司 Method for eliminating motor sound and digital image picking apparatus
WO2009038136A1 (en) * 2007-09-19 2009-03-26 Nec Corporation Noise suppression device, its method, and program
US20120143604A1 (en) * 2010-12-07 2012-06-07 Rita Singh Method for Restoring Spectral Components in Denoised Speech Signals
JP6687453B2 (en) * 2016-04-12 2020-04-22 パナソニック インテレクチュアル プロパティ コーポレーション オブ アメリカPanasonic Intellectual Property Corporation of America Stereo playback device
JP6668995B2 (en) * 2016-07-27 2020-03-18 富士通株式会社 Noise suppression device, noise suppression method, and computer program for noise suppression
CN108831500B (en) * 2018-05-29 2023-04-28 平安科技(深圳)有限公司 Speech enhancement method, device, computer equipment and storage medium

Citations (2)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
JPH0822297A (en) * 1994-07-07 1996-01-23 Matsushita Commun Ind Co Ltd Noise suppression device
JPH08221092A (en) * 1995-02-17 1996-08-30 Hitachi Ltd Nose eliminating system using spectral subtraction

Family Cites Families (6)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
GB8801014D0 (en) * 1988-01-18 1988-02-17 British Telecomm Noise reduction
US5710862A (en) * 1993-06-30 1998-01-20 Motorola, Inc. Method and apparatus for reducing an undesirable characteristic of a spectral estimate of a noise signal between occurrences of voice signals
CA2153170C (en) * 1993-11-30 2000-12-19 At&T Corp. Transmitted noise reduction in communications systems
DE69420705T2 (en) * 1993-12-06 2000-07-06 Koninkl Philips Electronics Nv SYSTEM AND DEVICE FOR NOISE REDUCTION AND MOBILE RADIO
SE505156C2 (en) * 1995-01-30 1997-07-07 Ericsson Telefon Ab L M Procedure for noise suppression by spectral subtraction
JP3484801B2 (en) * 1995-02-17 2004-01-06 ソニー株式会社 Method and apparatus for reducing noise of audio signal

Patent Citations (2)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
JPH0822297A (en) * 1994-07-07 1996-01-23 Matsushita Commun Ind Co Ltd Noise suppression device
JPH08221092A (en) * 1995-02-17 1996-08-30 Hitachi Ltd Nose eliminating system using spectral subtraction

Non-Patent Citations (3)

* Cited by examiner, † Cited by third party
Title
ITOH K., ET AL.: "REDUCTION OF ENVIRONMENTAL NOISE BASED ON SPEECH/NON-SPEECH IDENTIFICATION.", IEICE TRANSACTIONS ON COMMUNICATIONS., COMMUNICATIONS SOCIETY, TOKYO., JP, vol. 95., no. 03., 1 November 1995 (1995-11-01), JP, pages 17 - 24., XP002921513, ISSN: 0916-8516 *
See also references of EP0992978A4 *
YAMAMOTO H., ET AL.: "ESTIMATED SEGMENTAL SNR BASE ADAPTIVE SPECTRAL SUBTRACTION APPROACH FOR SPEECH RECOGNITION.", IEICE TRANSACTIONS ON COMMUNICATIONS., COMMUNICATIONS SOCIETY, TOKYO., JP, vol. 94., no. 271., 1 October 1994 (1994-10-01), JP, pages 17 - 24., XP002921514, ISSN: 0916-8516 *

Cited By (10)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US7706550B2 (en) 2004-01-08 2010-04-27 Kabushiki Kaisha Toshiba Noise suppression apparatus and method
WO2006064699A1 (en) * 2004-12-17 2006-06-22 Waseda University Sound source separation system, sound source separation method, and acoustic signal acquisition device
JP2006197552A (en) * 2004-12-17 2006-07-27 Univ Waseda Sound source separation system and method, and acoustic signal acquisition device
US8213633B2 (en) 2004-12-17 2012-07-03 Waseda University Sound source separation system, sound source separation method, and acoustic signal acquisition device
WO2006123721A1 (en) * 2005-05-17 2006-11-23 Yamaha Corporation Noise suppression method and device thereof
US8160732B2 (en) 2005-05-17 2012-04-17 Yamaha Corporation Noise suppressing method and noise suppressing apparatus
JP4958303B2 (en) * 2005-05-17 2012-06-20 ヤマハ株式会社 Noise suppression method and apparatus
US9754606B2 (en) 2012-05-01 2017-09-05 Ricoh Company, Ltd. Processing apparatus, processing method, program, computer readable information recording medium and processing system
JP2015002505A (en) * 2013-06-18 2015-01-05 パイオニア株式会社 Noise reducer, broadcast receiver and noise reduction method
US9318122B2 (en) 2014-04-18 2016-04-19 Fujitsu Limited Audio signal processing apparatus and audio signal processing method

Also Published As

Publication number Publication date
CA2291826A1 (en) 1999-10-07
AU721270B2 (en) 2000-06-29
AU1352599A (en) 1999-10-18
KR20000076037A (en) 2000-12-26
KR100314332B1 (en) 2001-11-16
CN1258368A (en) 2000-06-28
EP0992978A4 (en) 2002-01-16
EP0992978A1 (en) 2000-04-12

Similar Documents

Publication Publication Date Title
WO1999050825A1 (en) Noise reduction device and a noise reduction method
EP1557827B1 (en) Voice intensifier
EP0809842B1 (en) Adaptive speech filter
JP3457293B2 (en) Noise suppression device and noise suppression method
JP4863713B2 (en) Noise suppression device, noise suppression method, and computer program
JP2000347688A (en) Noise suppressor
JP5483000B2 (en) Noise suppression device, method and program thereof
EP1729286A2 (en) Method and apparatus for noise suppression
US20010005822A1 (en) Noise suppression apparatus realized by linear prediction analyzing circuit
JP2001134287A (en) Noise suppressing device
WO1999030315A1 (en) Sound signal processing method and sound signal processing device
WO2002080148A1 (en) Noise suppressor
JPH07248794A (en) Method for processing voice signal
JPH0916194A (en) Noise reduction for voice signal
WO2005124739A1 (en) Noise suppression device and noise suppression method
CN104067339A (en) Noise suppression device
JP2004341339A (en) Noise restriction device
JP2000330597A (en) Noise suppressing device
JP2004272292A (en) Sound signal processing method
JP2001159899A (en) Noise suppressor
JPH06208395A (en) Formant detecting device and sound processing device
US20030033139A1 (en) Method and circuit arrangement for reducing noise during voice communication in communications systems
JP2002140100A (en) Noise suppressing device
US20030065509A1 (en) Method for improving noise reduction in speech transmission in communication systems
JP3360423B2 (en) Voice enhancement device

Legal Events

Date Code Title Description
WWE Wipo information: entry into national phase

Ref document number: 98805620.8

Country of ref document: CN

WWE Wipo information: entry into national phase

Ref document number: 09367487

Country of ref document: US

WWE Wipo information: entry into national phase

Ref document number: 1998957196

Country of ref document: EP

WWE Wipo information: entry into national phase

Ref document number: 1019997008125

Country of ref document: KR

WWE Wipo information: entry into national phase

Ref document number: IN/PCT/1999/35/KOL

Country of ref document: IN

AK Designated states

Kind code of ref document: A1

Designated state(s): AL AM AT AU AZ BA BB BG BR BY CA CH CN CU CZ DE DK EE ES FI GB GE GH GM HU ID IL IN IS JP KE KG KR KZ LC LK LR LS LT LU LV MD MG MK MN MW MX NO NZ PL PT RO RU SD SE SG SI SK SL TJ TM TR TT UA UG US UZ VN YU ZW

AL Designated countries for regional patents

Kind code of ref document: A1

Designated state(s): GH GM KE LS MW SD SZ UG ZW AM AZ BY KG KZ MD RU TJ TM AT BE CH CY DE DK ES FI FR GB GR IE IT LU MC NL PT SE BF BJ CF CG CI CM GA GN GW ML MR NE SN TD TG

WWE Wipo information: entry into national phase

Ref document number: 13525/99

Country of ref document: AU

ENP Entry into the national phase

Ref document number: 2291826

Country of ref document: CA

Ref document number: 2291826

Country of ref document: CA

Kind code of ref document: A

121 Ep: the epo has been informed by wipo that ep was designated in this application
WWP Wipo information: published in national office

Ref document number: 1998957196

Country of ref document: EP

REG Reference to national code

Ref country code: DE

Ref legal event code: 8642

WWG Wipo information: grant in national office

Ref document number: 13525/99

Country of ref document: AU

WWP Wipo information: published in national office

Ref document number: 1019997008125

Country of ref document: KR

WWG Wipo information: grant in national office

Ref document number: 1019997008125

Country of ref document: KR

WWW Wipo information: withdrawn in national office

Ref document number: 1998957196

Country of ref document: EP