JP4438632B2 - Howling canceller - Google Patents

Description

  The present invention relates to a howling canceller that suppresses howling without reducing musical sound in a loudspeaker system installed in a lecture hall or hall.

  In general, when a loudspeaker is used in a lecture hall, a hall, or the like, sound output from a speaker is input to a microphone again through an acoustic path having a certain transfer function. That is, a closed loop is formed by the path of microphone-amplifier-speaker-acoustic path-microphone. When the gain of the closed loop exceeds 1, the sound returned from the speaker to the microphone increases, and howling occurs.

  In order to efficiently prevent this howling, a howling canceller that uses an adaptive filter (adaptive digital filter) to prevent howling has been proposed (for example, see Non-Patent Document 1).

  FIG. 6 is a view showing the above howling canceller. The microphone 101 and the speaker 104 are installed in the same acoustic space such as a lecture hall or a hall. Here, the audio signal input from the microphone 101 is amplified by a front-end microphone amplifier and then converted into a digital signal y (k) by an A / D converter.

  The signal y (k) is supplied to the amplifier 103 via the adder 102 and amplified. G (z) is a transfer function of the amplifier 103. The signal x (k) output from the amplifier 103 is sounded from the speaker 104 after being converted into an audio signal by the D / A converter.

  The sound generated from the speaker 104 returns to the microphone 101 via the acoustic feedback path 105. The acoustic return path 105 is an acoustic path from the speaker 104 to the microphone 101. H (z) is a transfer function of the acoustic feedback path 105. A feedback signal d (k) fed back through the acoustic feedback path 105 is input to the microphone 101 together with a sound source signal s (k) generated by a sound source such as a speaker. The microphone 101 converts the input voice into a digital signal and outputs it as y (k).

  In such a loudspeaker, a closed loop is formed by the path of the microphone 101 -the amplifier 103 -the speaker 104 -the acoustic feedback path 105 -the microphone 101. When the gain of the closed loop exceeds 1, the feedback signal d (k) is increased and howling occurs. The loudspeaker shown in FIG. 1 has a howling canceller including a delay circuit 106, an adaptive filter 107, and an adder 102 in order to prevent the occurrence of such howling.

  The delay circuit 106 gives a delay time τ corresponding to the time delay of the acoustic feedback path 105 to the output signal x (k) of the amplifier 103 and outputs it to the adaptive filter 107 as a signal x (k−τ). The adaptive filter 107 includes a filter unit 107a and a filter coefficient estimation unit 107b as shown in FIG. 7, and the signal x (k−τ) is input to both the filter unit 107a and the filter coefficient estimation unit 107b. .

  In the filter unit 107a, the filter coefficient is set so that the signal input from the microphone 101 is attenuated by the transfer function F (z) simulating the transfer function H (z) of the acoustic feedback path 105. Therefore, the signal do (k) output from the adaptive filter 107 is a signal obtained by filtering the signal x (k−τ) with the transfer function F (z) simulating the transfer function H (z) of the acoustic feedback path 105. Therefore, the feedback signal d (k) transmitted from the speaker 104 through the acoustic feedback path 105 and re-input to the microphone 101 is simulated.

  The adder 102 is a signal do (k) simulating the feedback signal d (k) from the signal y (k) input from the microphone 101 (where y (k) is a signal obtained by adding the sound source signal and the feedback signal). ) Is subtracted. Thereby, the feedback signal is removed from the input signal, and howling can be canceled.

The filter coefficient estimating unit 107b uses an adaptive algorithm so that the transfer function F (z) matches or approximates the transfer function H (z) based on the signals x (k−τ) and e (k). Are sequentially updated. As a result, a signal do (k) simulating the signal d (k) is obtained, and howling can be prevented.
Inazumi, Imai, Konishi: "Preventing howling in loudspeakers using the LMS algorithm", Proc. 417-418 (1991, 3)

  By the way, although howling appears as a high-level signal of a sine wave (pure tone) having a constant frequency, the howling canceller described in Non-Patent Document 1 adapts a filter to this howling and attenuates the frequency peak of the signal. A simulation signal do (k) is generated. That is, the howling canceller determines that a component similar to the signal delayed by the delay circuit 106 among the signals input from the microphone is a feedback signal, and updates the filter coefficient of the adaptive filter so as to attenuate it. Therefore, the howling canceller can prevent the howling by canceling the feedback signal before the howling, and can also quickly attenuate the howling sound when the howling occurs.

  As described above, the howling canceller with a built-in adaptive filter has a characteristic in which a component similar to a delayed signal among signals input from the microphone is regarded as a feedback signal and attenuated. For this reason, for example, the musical tone of a musical instrument that emits a continuous sound such as a violin or flute is a high-level signal close to a pure tone of a constant frequency similar to a howling signal. However, there was a problem that it was canceled by the howling canceller.

  In view of the above circumstances, the present invention has an object to provide a howling canceller including an adaptive filter that distinguishes between a continuous sound of a musical tone having a constant frequency and a howling, and attenuates only howling without attenuating input due to the musical tone. To do.

The invention according to claim 1 estimates the transfer function of the acoustic feedback path from the speaker to the microphone, generates a simulation signal by filtering the output signal to the speaker with the estimated transfer function, and outputs to cancel from the input signal a feedback sound signal, a howling canceller having an adaptive filter for updating the filter coefficients if the input signal after canceling a predetermined level or more peak component on the frequency axis when the peak signal having is input as an input signal, the input signal judgment means for judging whether or not this peak signal is the input signal by the tone, the peak signal is input from the microphone, and the input signal judgment means for this Phil but if it is determined that the input signal by the tone, of the adaptive filter in the vicinity of the detected peak frequency And sets the update of the coefficient to stop or slow down.

  In the present invention, there is provided an input signal determining means for determining whether or not the audio signal input from the microphone is an input signal based on a musical sound. The input signal determination means determines whether or not the input signal is a musical sound when a continuous sound having a constant frequency is detected. When it is determined that the input signal is a musical tone input signal, the adaptive filter is set to stop or slow down the adaptive processing at the peak frequency detected so as not to cancel the signal input from the microphone. As a result, even if a continuous tone having a constant frequency is input, the adaptive filter can operate so as not to cancel the signal.

The invention according to claim 2 estimates the transfer function of the acoustic feedback path from the speaker to the microphone, generates a simulated signal by filtering the output signal to the speaker with the estimated transfer function, and outputs to cancel from the input signal a feedback sound signal, a howling canceller having an adaptive filter for updating the filter coefficients if the input signal after canceling a predetermined level or more peak component on the frequency axis when the peak signal having is input as an input signal, the input signal judgment means for judging whether or not this peak signal is the input signal by the tone, the peak signal is input from the microphone, and the input signal judgment means for this If There it is determined that the input signal by the tone pairs in the frequency of the simulated signals in the vicinity of the detected peak frequency A filter output signal control means that the frequency characteristics change into smooth by calculating a moving average that, characterized by comprising a.

  In the present invention, there is provided an input signal determining means for determining whether or not the audio signal input from the microphone is an input signal based on a musical sound. The input signal determination means determines whether or not the input signal is a musical sound when a continuous sound having a constant frequency is detected. In addition, filter output signal control means for correcting the output signal of the adaptive filter is provided, and when the input signal determination means determines that the input signal is a musical sound, the frequency characteristic of the filter output signal is detected in the vicinity of the detected peak frequency. Smooth. For example, calculation processing is performed so as to take a moving average of the frequency characteristics of the filter output signal in the vicinity of the detected peak frequency.

The invention according to claim 3 estimates the transfer function of the acoustic feedback path from the speaker to the microphone, generates a simulated signal by filtering the output signal to the speaker with the estimated transfer function, and outputs to cancel from the input signal a feedback sound signal, a howling canceller having an adaptive filter for updating the filter coefficients if the input signal after canceling a predetermined level or more peak component on the frequency axis when the peak signal having is input as an input signal, the input signal judgment means for judging whether or not this peak signal is the input signal by the tone, the peak signal is input from the microphone, and the input signal judgment means for this If There it is determined that the input signal by the tone, attenuates the amplitude of the test signal in the detected peak frequency A filter output signal suppression means, comprising the.

  In the present invention, there is provided an input signal determining means for determining whether or not the audio signal input from the microphone is an input signal based on a musical sound. The input signal determination means determines whether or not the input signal is a musical sound when a continuous sound having a constant frequency is detected. Furthermore, filter output signal suppression means for suppressing the output signal of the adaptive filter is provided, and when the input signal determination means determines that the input signal is a tone, the filter output signal is attenuated in the vicinity of the detected peak frequency. Attenuation also includes modifying the output signal to zero. At other frequencies, it is output as it is without being attenuated.

  As described above, according to the present invention, when the peak signal is input from the microphone, it is determined whether or not the input signal is a musical sound, and the adaptive processing of the adaptive filter is stopped or blunted at the peak frequency. It is possible to suppress only howling without suppressing the input signal due to the musical sound.

  Hereinafter, a howling canceller according to an embodiment of the present invention will be described in detail with reference to the drawings.

[First Embodiment]
FIG. 1 is a block diagram of a loudspeaker system with an adaptive howling canceller according to the first embodiment. As shown in the figure, this loudspeaker built-in adaptive howling canceller includes a microphone 1, an adder 2, an amplifier 3, a speaker 4, a delay circuit 5, an adaptive filter 6, and a waveform analysis unit 7. The microphone 1 and the speaker 4 Is placed in a lecture hall or hall.

  The audio signal input to the microphone 1 is input to the adder 2 and the waveform analysis unit 7 as a digital signal by A / D conversion processing. The adder 2 subtracts the output signal of the adaptive filter 6 from the input signal from the microphone 1 and outputs the result. The output signal of the adder 2 is amplified through the amplifier 3 and then output to the speaker 4. The signal transmitted to the speaker 4 is converted into an analog audio signal and then emitted. Here, the sound generated from the speaker 4 is re-input to the microphone 1 as a feedback signal.

  In the howling canceller in this example, the audio signal input from the microphone 1 is propagated through the acoustic space in which the amplifier 3, the speaker 4, and the microphone 1 are installed by the delay circuit 5 and the adaptive filter 6 and is input to the microphone 1 again. This simulates the transmission characteristics of a series of voice transmission paths up to the point.

  The delay circuit 5 gives a time delay that estimates the time delay of the feedback signal that is fed back from the speaker 4 to the microphone 1. The signal output after being given a time delay by the delay circuit 4 is input to the adaptive filter 6.

  The adaptive filter 6 is a filter that simulates the transfer function of the acoustic feedback path, and filters the signal delayed by the delay circuit 5. The filtered output signal is output to the adder 2 as a simulation signal.

  As shown in FIG. 2, the adaptive filter 6 includes a filter unit 6a and a filter coefficient estimation unit 6b, and a signal delayed from the delay circuit 5 is input to the filter unit 6a and the filter coefficient estimation unit 6b. The filter unit 6 a filters the input signal and outputs it to the adder 2. The adder 2 subtracts the output signal of the filter unit 6a from the input signal of the microphone.

  The filter coefficient estimator 6b detects the erasure error of the feedback signal based on the past signal delayed by the delay circuit 5 and the current signal that is the output signal of the adder 2, and matches the simulated signal with the feedback signal. The transfer function of the filter unit 6a is automatically updated to approximate it.

  An adaptive algorithm is used to update the transfer function of the filter coefficient estimation unit 6b. As the adaptive algorithm, for example, a LMS (Least Mean Square) algorithm is used.

  The waveform analysis unit 7 performs fast Fourier transform (hereinafter referred to as FFT) on the signal input from the microphone 1 and analyzes the frequency characteristics of the input signal. As a result, it is determined whether or not the input signal is an input signal based on a musical sound. Whether or not the input signal is a musical sound is determined by comparing the input waveform with a previously stored musical sound waveform pattern and determining that the input signal is a musical sound when the correlation with the musical sound waveform pattern is high. As a comparison method, a cross-correlation function or a speech recognition method such as an HMM (Hidden Markov Model) method may be used. If it is determined that the input signal is a musical sound, the adaptive filter 6 is instructed not to cancel the signal input from the microphone. Note that the frequency characteristic of the input signal may be compared with the frequency characteristic of the musical sound waveform stored in advance, and the input signal may be determined as a musical sound. In addition, a howling waveform is stored instead of a musical sound waveform, and the input signal is compared with a stored howling waveform. May be.

Hereinafter, the operation of the above-described sounding system with a howling canceller will be described in detail.
FIG. 3 is a block diagram showing transfer characteristics of the loudspeaker system with built-in adaptive howling canceller according to the first embodiment. As shown in the figure, the audio signal input via the microphone 1 is converted into a digital signal y (k) by A / D conversion processing and input to the amplifier 3 via the adder 2. The amplifier 3 is for amplifying the input signal y (k). G (z) is a transfer function of the amplifier 3.

  The signal x (k) output from the amplifier 3 is converted into an analog audio signal by D / A conversion processing, and a sound is produced from the speaker 4. The sound generated from the speaker 4 returns to the microphone 1 through the acoustic return path 8. The acoustic return path 8 is an acoustic path from the speaker 4 to the microphone 1. H (z) is a transfer function of the acoustic feedback path 8. A feedback signal d (k) fed back through the acoustic feedback path 8 is input to the microphone 1 together with a sound source signal s (k) generated by a sound source such as a speaker.

  The signal x (k) output from the amplifier 3 is also input to the delay circuit 5. The delay circuit 5 gives a time delay to the input signal x (k) and outputs it. Here, a time delay τ that estimates the time delay of the feedback audio signal fed back from the speaker 4 to the microphone 1 is obtained. It is given. The signal x (k−τ) output with the time delay τ given by the delay circuit 5 is input to the adaptive filter 6.

As shown in FIG. 2, the adaptive filter 6 includes a filter unit 6a and a filter coefficient estimation unit 6b. The filter unit 6a and the filter coefficient estimation unit 6b each receive a signal x (k− τ) is input. The filter unit 6a outputs a signal do (k) simulating the feedback signal d (k) from the speaker 4 to the microphone 1, and the feedback signal d from the signal y (k) re-input from the microphone 1 by the adder 2. A signal do (k) simulating (k) is subtracted. A signal do (k) simulating the feedback signal d (k) is determined based on the signal x (k−τ) output from the delay circuit 5 according to the transfer function F (z). The filter coefficient estimation unit 6b uses the signal x (k−τ) output from the delay circuit 5 and the signal do (k) simulating the feedback signal d (k) among the signals transmitted from the microphone 1 to the amplifier 3. Based on the subtracted signal e (k), an adaptive algorithm is used to filter the filter coefficient of the filter unit 6a so that the signal do (k) simulating the feedback signal matches or approximates the actual feedback signal d (k). Is to be updated. As the adaptive algorithm, for example, an LMS algorithm is used. If the root mean square value of the signal e (k) is J = E [e (k) ² ] (where E [•] is an expected value), a filter coefficient that minimizes J is estimated by calculation. The filter coefficient of the filter unit 6a is updated using the estimated filter coefficient.

  As described above, the adaptive filter 6 simulates the feedback signal d (k) among the signal x (k−τ) output from the delay circuit 5 and the signal y (k) transmitted from the microphone 1 to the amplifier 3. Since the filter coefficient is updated based on the signal e (k) obtained by subtracting the signal do (k), it is possible to cancel the signal d (k) among the signals input from the microphone 1. When the gain of the closed loop formed by the microphone 1 -amplifier 3 -speaker 4 -acoustic feedback path 8 -microphone 1 path exceeds 1, the amplitude value of a certain frequency of the feedback signal d (k) increases with time. Although it increases and forms a peak on the frequency spectrum, and howling occurs, the adaptive filter 6 updates the filter coefficient so as to attenuate this peak, and the generated simulated signal do (k) is used as the input signal y ( Subtract from k). Accordingly, when the signal y (k) obtained by adding the sound source signal s (k) input from the microphone and the feedback signal d (k) is a continuous sound having a constant frequency, the filter coefficient is sequentially updated to obtain the continuous sound. Can be attenuated.

  The waveform analysis unit 7 performs FFT on the signal y (k) input from the microphone 1 as described above, and detects the peak frequency. For example, the peak frequency may be detected by setting the maximum frequency in the frequency band in which the amplitude value is equal to or greater than a predetermined threshold as the peak frequency. Further, the lowest frequency peak among the detected peak frequencies is set as a primary peak of the input signal, and a peak near a frequency that is an integral multiple of the primary peak is set as a harmonic overtone peak.

  When the detected peak frequency is continuously input for a predetermined time or more, it is determined whether or not the input signal is a musical sound from the spectrum shape near each peak frequency. Specifically, the waveform of the input signal is compared with a previously stored musical sound waveform pattern, and when the correlation with the musical sound waveform pattern is high, it is determined that the input signal is a musical sound. A cross-correlation function between the waveform of the input signal and the musical sound waveform is obtained to determine whether the signal is a musical sound signal. A pattern matching technique such as HMM may be used instead of the cross correlation function. In addition, a howling waveform is stored instead of a musical sound waveform, and the input signal is compared with a stored howling waveform. May be. Alternatively, the frequency width of the detected peak shape may be calculated, a predetermined threshold value may be set for the peak width with this frequency width as the peak width, and an input signal based on a musical sound may be determined when the frequency is equal to or greater than the threshold value. As described above, any method may be used as long as it can be determined whether or not the input signal is a musical sound.

  When it is determined that the input signal is not an input signal due to a musical sound but an input signal due to howling, the adaptive filter 6 suppresses howling. When the peak frequency is detected and it is determined that the input signal is a musical sound, the adaptive filter 6 is instructed not to cancel the signal input from the microphone 1.

  Here, the filter coefficient estimation unit 6b is instructed to stop or slow down the adaptive processing at the detected peak frequency. The transfer function of the adaptive filter 6 is represented by F (z), and this transfer function is updated using coefficients called forgetting coefficient λ and step size α. The forgetting factor λ is a factor that multiplies the transfer function so far, and is set in the range of 0-1. If the forgetting factor λ is reduced, the transfer function up to that point is deleted and the update is promoted. The step size α is a coefficient representing the magnitude of correction. When the step size α is increased, the corrected transfer function is used more and the update is promoted. The corrected transfer function F ′ (z) is expressed by the following formula: F ′ (z) = λF (z) + αΔF (z) (where ΔF (z) is a filter difference).

  Accordingly, when the waveform analysis unit 7 detects the peak frequency and determines that the input signal is a musical sound, the waveform analysis unit 7 sets the step size α of the adaptive filter 6 so as not to update the transfer function F (z) at the detected peak frequency. Set to be smaller. Note that the step size α may be reduced not only for the detected primary peak frequency but also for the harmonic peak frequency. Alternatively, the step size α may be set to be small and the forgetting factor λ may be set to be large. As a result, it is possible to avoid canceling the peak frequency band of the input signal determined as a musical sound among the new input signals from the microphone 1.

[Second Embodiment]
FIG. 4 is a block diagram of a speech enhancement system with an adaptive howling canceller according to the second embodiment. In addition, the same code | symbol is attached | subjected to the part similar to the sounding system with a built-in adaptive howling canceller which concerns on 1st Embodiment, and detailed description is abbreviate | omitted. As shown in the figure, the loudspeaker system in this example further includes a signal correction unit 9 between the adaptive filter 6 and the adder 2 in addition to the howling canceller of the first embodiment.

  The signal correction unit 9 corrects a signal in an arbitrary frequency range among the output signals of the adaptive filter 6. Here, the output signal of the adaptive filter 6 is subjected to a correction operation so as to smooth the frequency characteristic of the output signal of the adaptive filter 6 for the frequency band detected by the waveform analysis unit 7. The correction operation of the output signal of the adaptive filter 6 will be described below.

  FIG. 5 is a diagram showing transfer characteristics of the loudspeaking system with a built-in adaptive howling canceller according to the second embodiment. As shown in the figure, the audio signal input via the microphone 1 is converted into a digital signal y (k) by A / D conversion processing and input to the amplifier 3 via the adder 2. The amplifier 3 is for amplifying the input signal y (k). G (z) is a transfer function of the amplifier 3.

  The signal x (k) output from the amplifier 3 is converted into an analog audio signal by D / A conversion processing, and a sound is produced from the speaker 4. The sound generated from the speaker 4 returns to the microphone 1 through the acoustic return path 8. The acoustic return path 8 is an acoustic path from the speaker 4 to the microphone 1. H (z) is a transfer function of the acoustic feedback path 8. A feedback signal d (k) fed back through the acoustic feedback path 8 is input to the microphone 1 together with a sound source signal s (k) generated by a sound source such as a speaker. The microphone 1 converts the input voice into a digital signal and outputs it as y (k).

  The signal x (k) output from the amplifier 3 is also input to the delay circuit 5. The delay circuit 5 gives a time delay to the input signal x (k) and outputs it. Here, a time delay τ that estimates the time delay of the feedback audio signal fed back from the speaker 4 to the microphone 1 is obtained. It is given. The signal x (k−τ) output with the time delay τ given by the delay circuit 5 is input to the adaptive filter 6.

  The adaptive filter 6 outputs a signal do (k) simulating the feedback audio signal d (k) according to the transfer function F (z). Further, the signal correction unit 9 outputs the signal d1 (k) according to the transfer function F1 (z). The adder 2 subtracts the signal d1 (k) from the signal y (k) re-input from the microphone 1.

  The signal d1 (k) is converted so as to replace the frequency characteristic around the peak frequency detected by the waveform analysis unit 7 with the frequency characteristic of the signal do (k) with a moving average of the frequency characteristic. It is a thing. As a result, the frequency characteristic of the output signal do (k) of the adaptive filter 7 is smoothed, and the input signal y (k) due to the musical sound can be prevented from being canceled. In this example as well, the moving average may be calculated not only in the vicinity of the detected peak frequency but also in the vicinity of its harmonic frequency, and the frequency characteristics of the output signal may be smoothed.

  Further, the signal correction unit 9 may cut the signal at the peak frequency detected by the waveform analysis unit 7 from the simulated signal do (k). Of the simulated signal do (k), the output signal of the frequency detected by the waveform analysis unit 7 is cut out, corrected to attenuation or 0, and synthesized with other frequencies not subjected to the correction calculation and output. Thus, the signal correction unit 9 may function like a band cut filter. In this case, correction calculation may be performed not only in the vicinity of the detected peak frequency but also in the vicinity of the harmonic frequency.

  Also in the second embodiment, when the waveform analysis unit 7 detects the peak frequency and determines that it is not an input signal due to howling, the step size of the adaptive filter 6 is not updated so as to not update the transfer function F (z). You may make it set so that (alpha) may be made small. Thereby, a new input signal from the microphone 1 is not canceled.

  As described above, the howling canceller in the present invention performs FFT on the audio signal input from the microphone, determines whether the input signal is due to howling from the waveform of the peak frequency, and inputs due to howling. When it is determined that the signal is not a signal, the adaptive filter output signal is set so as not to cancel the microphone input signal at the detected peak frequency. It is possible to prevent the input signal from being canceled by stopping or slowing down the adaptive processing of the adaptive filter at the detected peak frequency. Further, in the frequency characteristic of the output signal of the adaptive filter, it is possible to avoid canceling the input signal by calculating a moving average near the detected peak frequency.

  As a result, when a continuous sound having a constant frequency such as a violin is input, it is possible to suppress only howling without canceling the musical sound.

  The input signal is not limited to the input from the microphone, but may be any signal as long as it collects a sound source signal of a speaker such as a vibration sensor. Further, the frequency analysis method is not limited to the FFT described above, and any method may be used as long as it is a method capable of calculating a frequency peak. For example, a band pass filter or the like may be used.

Block diagram of a loudspeaker system with built-in adaptive howling canceller according to the first embodiment of the present invention. The block diagram which shows the structure of the adaptive filter concerning 1st Embodiment of this invention in detail The figure which shows the transfer characteristic of the sounding system with a built-in adaptive howling canceller which concerns on 1st Embodiment of this invention. The block diagram which shows the structure of the sounding system with a built-in adaptive howling canceller which concerns on 2nd Embodiment of this invention. The figure which shows the transfer characteristic of the sounding system with a built-in adaptive howling canceller which concerns on 2nd Embodiment of this invention. Block diagram showing the circuit configuration of a conventional loudspeaker built-in adaptive howling canceller Block diagram showing the configuration of a conventional adaptive filter in detail

Explanation of symbols

1-microphone 2-adder 3-amplifier 4-speaker 5-delay circuit 6-adaptive filter 6 a-filter unit 6 b-filter coefficient estimation unit 7-waveform analysis unit 8-acoustic feedback path 9-signal correction unit 101-conventional Microphone 102 in loudspeaker system-Adder 103 in conventional loudspeaker system-Amplifier 104 in conventional loudspeaker system-Speaker 105 in conventional loudspeaker system-Acoustic feedback path 106 in conventional loudspeaker system-Delay circuit 107 in conventional loudspeaker system- Adaptive filter 107a in conventional loudspeaker system-filter unit 107b in conventional adaptive filter-filter coefficient estimator in conventional adaptive filter

Claims (3)

Estimate the transfer function of the acoustic feedback path from the speaker to the microphone, filter the output signal to the speaker with the estimated transfer function to generate a simulated signal, and cancel the simulated signal from the feedback audio signal that is the input signal A canceling canceller having an adaptive filter that inputs a signal after cancellation and updates a filter coefficient ,
When a peak signal having a peak component of a predetermined level or higher on the frequency axis is input as an input signal, input signal determination means for determining whether or not the peak signal is an input signal based on a musical sound;
Peak signal from the microphone is inputted, and when the input signal judgment means which determines that the input signal by tone, by setting so as to stop or slow down the update of the filter coefficient of the adaptive filter in the vicinity of the detected peak frequency Howling canceller characterized by. Estimate the transfer function of the acoustic feedback path from the speaker to the microphone, filter the output signal to the speaker with the estimated transfer function to generate a simulated signal, and cancel the simulated signal from the feedback audio signal that is the input signal A canceling canceller having an adaptive filter that inputs a signal after cancellation and updates a filter coefficient ,
When a peak signal having a peak component of a predetermined level or higher on the frequency axis is input as an input signal, input signal determination means for determining whether or not the peak signal is an input signal based on a musical sound;
When a peak signal is input from a microphone and the input signal determination means determines that the input signal is a musical sound, the frequency characteristic is smoothed by calculating a moving average with respect to the frequency of the simulated signal in the vicinity of the detected peak frequency . Filter output signal control means,
A howling canceller characterized by comprising Estimate the transfer function of the acoustic feedback path from the speaker to the microphone, filter the output signal to the speaker with the estimated transfer function to generate a simulated signal, and cancel the simulated signal from the feedback audio signal that is the input signal A canceling canceller having an adaptive filter that inputs a signal after cancellation and updates a filter coefficient ,
When a peak signal having a peak component of a predetermined level or higher on the frequency axis is input as an input signal, input signal determination means for determining whether or not the peak signal is an input signal based on a musical sound;
A filter output signal suppression means for attenuating the amplitude of the simulated signal at the detected peak frequency when a peak signal is input from the microphone and the input signal determination means determines that the input signal is a musical sound;
A howling canceller characterized by comprising

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