JP4367344B2 - Loudspeaker - Google Patents

Description

  The present invention relates to a loudspeaker installed in a lecture hall, a hall or the like, and a loudspeaker incorporating a howling canceller suitable for preventing occurrence of howling.

  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 prevent the occurrence of howling as described above, it is often performed to use a filter that cuts a frequency component in which howling occurs. In this method, since the howling frequency varies depending on the installation conditions of the loudspeaker, it is indispensable to control the frequency to which the filter is assigned in order to effectively prevent howling. Thus, a howling canceller has been proposed in which acoustic characteristics between a microphone and a speaker are measured in advance and the filter frequency is controlled using the result (for example, Patent Document 1).

In addition, an adaptive filter (adaptive digital filter) is used that subtracts a signal simulating a feedback transmission system from the speaker to the microphone from the input signal of the microphone and corrects the simulated signal so that the error signal after subtraction becomes smaller. An adaptive howling canceller for preventing howling is used (for example, Non-Patent Document 1).
Japanese Patent Laid-Open No. 6-195092 Inazumi, Imai, Konishi: “Preventing howling in loudspeakers using LMS algorithm”, Proc. 417-418 (1991, 3)

  However, in the howling canceller described in Patent Document 1, it is difficult to prevent howling because the howling frequency changes when the use condition deviates from the acoustic characteristics measured in advance due to movement of the speaker. is there.

  Also, as described in Non-Patent Document 1, the adaptive howling canceller must estimate the acoustic path between the speaker and the microphone using only the information of the input signal from the microphone. Therefore, this adaptive howling canceller alone has a problem that the estimation cannot catch up when the acoustic characteristics change suddenly due to a large movement of the speaker.

  The present invention has been made in consideration of the above circumstances, and provides a loudspeaker with a built-in adaptive howling canceller capable of detecting a change in acoustic characteristics with high accuracy and effectively preventing howling. With the goal.

  The invention according to claim 1 amplifies the input signal input from the microphone and outputs the amplified audio signal to the speaker, the delay means simulating the feedback transmission system from the speaker to the microphone, and the adaptive filter And a howling canceller that subtracts a simulated signal obtained by processing an audio signal output to the speaker with a delay unit and an adaptive filter from an audio signal input from the microphone, the loudspeaker comprising: It is characterized by comprising position detection means for detecting the positional relationship of the speakers, and setting means for setting the delay time of the delay means based on the detection result of the position detection means.

  In this invention, the position detection means which detects the positional relationship of a microphone and a speaker is provided. For example, the position detecting means attaches a wireless transceiver to a microphone and a speaker, and calculates a distance from a time difference between transmission and reception. A signal input from the microphone is amplified by a loudspeaker (for example, an amplifier) and is output from a speaker. The amplified signal is given a delay time and is output to the filter. The filter estimates the transfer function of the acoustic feedback path based on this signal and the signal input from the microphone, and subtracts the feedback signal to suppress howling. Here, the delay time is determined from the positional relationship between the microphone and the speaker. That is, when the speaker and the microphone are close to each other, the delay time is shortened because the arrival time of the signal returning through the acoustic feedback path is short. When the positions of the speaker and the microphone are far from each other, the delay time is lengthened because the arrival time of the signal returning on the acoustic feedback path is long. As a result, the transfer function of the acoustic feedback path can be estimated more accurately, and howling can be effectively suppressed.

  According to a second aspect of the present invention, in the above invention, the adaptive filter includes a filter unit and a filter coefficient estimation unit, and the setting unit updates the filter coefficient based on a detection result of the position detection unit. Are set in the filter coefficient estimator.

  In this invention, the update of the filter coefficient is set from the positional relationship between the speaker and the microphone. That is, when the positions of the speaker and the microphone change, the transfer function of the acoustic feedback path also changes greatly. Therefore, the filter coefficient update parameter is set so as to update the filter more quickly. For example, the step size may be increased.

  According to a third aspect of the present invention, in the above invention, the microphone and speaker gain adjustment is further performed, and gain adjustment means for detecting gain adjusted information is further provided, and the setting means is in accordance with the gain of the microphone and speaker. The subtraction amount of the simulation signal of the adaptive filter is set.

  In the present invention, when the gain of the microphone and the speaker is adjusted, the information on the gain adjustment is notified to the filter. For example, when the gain of the microphone is increased, the signal level input from the microphone increases, so the input level of the feedback audio signal also increases. At this time, the filter increases the level of the filter output signal and subtracts it from the feedback audio signal based on information obtained by increasing the gain of the microphone. Thereby, even if the gain of the microphone or the speaker is changed, the feedback signal can be immediately subtracted, and howling can be effectively suppressed.

  As described above, according to the present invention, since the delay time of the signal input to the filter is determined based on the positional relationship between the microphone and the speaker, the transfer function of the acoustic feedback path can be estimated with high accuracy. , Howling can be reliably suppressed.

  Hereinafter, an adaptive 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 a built-in adaptive howling canceller according to the first embodiment. As shown in this figure, this adaptive howling canceler built-in loudspeaker system includes a microphone 1, a microphone amplifier 2, an A / D converter 3, an equalizer amplifier 4, a speaker amplifier 6, a speaker 7, a delay circuit 8, an adaptive filter 9, and an adder. 10 and the position information control unit 11, and the microphone 1 and the speaker 7 are arranged in a lecture hall, a hall, or the like. The audio signal input via the microphone 1 is input to the microphone amplifier 2. The microphone amplifier 2 is for amplifying and adjusting the input audio signal.

  The audio signal output from the microphone amplifier 2 is converted into a digital signal by the A / D converter 3. The signal converted into the digital format is input to the equalizer amplifier 4 and subjected to frequency component adjustment and the like. Thereafter, the signal is converted into an analog audio signal by the D / A converter 5, and the final stage of output is adjusted by the speaker amplifier 6, and sound is produced from the speaker 7.

  The signal output from the equalizer amplifier 4 is also input to the delay circuit 8. The delay circuit 8 gives a time delay to the input signal and outputs it. Here, the delay circuit 8 gives a time delay that estimates the time delay of the feedback audio signal that is fed back from the speaker to the microphone. The signal output after being given a time delay by the delay circuit 8 is input to the adaptive filter 9.

  As shown in FIG. 2, the adaptive filter 9 includes a filter unit 9a and a filter coefficient estimation unit 9b, and the signals output from the delay circuit 8 are input to the filter unit 9a and the filter coefficient estimation unit 9b, respectively. The filter unit 9 a outputs a signal simulating the feedback audio signal from the speaker 7 to the microphone 1 to the adder 10, and subtracts the signal simulating the feedback audio signal from the signal transmitted to the equalizer amplifier 4 by the adder 10. Like that. A signal simulating a feedback audio signal is determined based on a signal output from the delay circuit 8 according to a certain transfer function. The filter coefficient estimator 9b uses an adaptive algorithm based on a signal output from the delay circuit 8 and a signal obtained by subtracting a signal simulating the feedback audio signal from the signal transmitted to the equalizer amplifier 4 to perform feedback. The filter coefficient of the filter unit 9a is updated so that the signal simulating the sound signal matches or approximates the actual feedback sound signal. As the adaptive algorithm, for example, a LMS (Least Mean Square) algorithm is used.

  Here, when there is no time delay, the audio signal input to the microphone 1 is immediately input to the adaptive filter 9, so that when the filter coefficient is updated, all signals input to the microphone 1 are canceled. End up. This is because the adaptive filter 9 updates the filter coefficient so as to reduce the error between the signal simulating the feedback audio signal and the actual feedback audio signal. If there is no time delay, the microphone 1 This is because the adaptive algorithm determines that all the signals input from are feedback audio signals. By giving a time delay, it becomes possible to cancel only the feedback audio signal without subtracting all the audio signals input to the microphone 1.

  On the other hand, if the time delay is applied too much, it is difficult to suppress howling because a speech signal that is greatly delayed from the time delay of the actual feedback speech signal is input to the adaptive filter and the feedback speech signal cannot be estimated. It becomes. If the time delay is too small, the time until the feedback audio signal is actually input is wasteful because nothing is output from the adaptive filter 9. Therefore, it is important for the suppression of howling that the delay circuit 8 gives the time delay accurately.

  The position information control unit 11 detects position information of the microphone 1 and the speaker 7. The position information indicates the distance between the microphone 1 and the speaker 7. Each of the microphone 1 and the speaker 7 is provided with a wireless transmitter / receiver (not shown), and the distance can be measured by transmitting and receiving wireless to each other and measuring the wireless arrival time. In addition, the structure which equips either the microphone 1 or the speaker 7 with a transmitter, and equips the other with a receiver may be sufficient.

  In addition, it is not limited to wireless transmission / reception, and any device can be used as long as it can measure the mutual distance. For example, the distance between each other may be actually measured using a laser, or the position may be measured using GPS. Alternatively, a specific pulse sound may be output from the speaker 7 and collected by the microphone 1 to measure the distance.

  The position information control unit 11 calculates the time delay of the feedback audio signal based on the input position information. At this time, the environment such as the temperature of the space in which the microphone 1 and the speaker 7 are installed may be measured and taken into consideration in the calculation of the time delay. Thereafter, the position information control unit 11 sets the delay circuit 8 to give the calculated time delay.

  Note that the direction of the microphone 1 and the speaker 7 may be detected as direction information, and the time delay calculated based on the position information may be corrected based on the direction information. For example, when the microphone 1 and the speaker 7 are directed in opposite directions, correction is performed so that the time delay becomes longer. This is because when the microphone 1 and the speaker 7 face in opposite directions, the reflected sound component from the wall surface or the like is larger than the direct sound component, and the reflected sound from the wall surface or the like has a longer time delay than the direct sound.

  Further, the position information control unit 11 instructs the filter coefficient estimation unit 9b to change the filter coefficient based on the input position information. When the positions of the microphone 1 and the speaker 7 change, the feedback audio signal that is fed back from the speaker 7 to the microphone 1 also changes. growing. Therefore, it is necessary to change the transfer function of the filter unit 9a, and the filter coefficient estimation unit 9b sets the update parameter of the filter coefficient so that the transfer function is immediately updated based on the position information.

  Hereinafter, a signal transmitted through the above-described adaptive howling canceller built-in loudspeaker system will be described in detail.

  FIG. 3 is a diagram showing a circuit configuration of a loudspeaker with a built-in adaptive howling canceller. 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. The signal y (k) is supplied to the amplification path 13. Thereafter, the signal y (k) is amplified by the transfer function G (z) in the amplification path 13 and supplied to the speaker 7 as the signal x (k). The signal x (k) is converted into an audio signal by D / A conversion processing and output from the speaker 7. The amplification path 13 is a transfer function of a path that is transmitted from the adder 10 to the D / A converter 5 via the equalizer amplifier 4.

  The signal output from the speaker 7 returns to the microphone 1 through the acoustic feedback path 12. Here, the acoustic feedback path 12 is an acoustic path from the speaker 7 to the microphone 1, and this path is represented by a transfer function H (z). Here, the feedback voice signal d (k) fed back via the acoustic feedback path 12 is mixed with a sound source voice signal s (k) consisting of a voice signal from a sound source such as a speaker and inputted to the microphone 1. . The microphone 1 outputs the input mixed audio signal as a digital signal by A / D conversion processing.

  In the above loudspeaker system, a closed loop is formed by the route of microphone 1 -amplifier path 13 -speaker 7 -acoustic feedback path 12 -microphone 1. As a result, the feedback audio signal d (k) increases, and howling occurs. An adaptive howling canceller including the delay circuit 8, the adaptive filter 9, and the adder 10 is provided to prevent the occurrence of such howling.

  The delay circuit 8 outputs the signal x (k) with a delay τ obtained by estimating the delay time of the acoustic feedback path 12, and the output signal x (k−τ) is supplied to the adaptive filter 9. . The filter unit 9a of the adaptive filter 9 outputs a signal do (k) simulating the feedback audio signal d (k) according to the transfer function F (z). The adder 10 subtracts the signal do (k) from the signal y (k). Here, the signal y (k) is expressed by the sum of the sound source audio signal s (k) and the feedback audio signal d (k), that is, the expression y (k) = s (k) + d (k). Therefore, the output signal e (k) of the adder 10 is e (k) = y (k) −do (k) = s (k) + d (k) −do (k) = s (k) + Δ (k). (However, Δ (k) = d (k) −do (k)). Therefore, howling can be suppressed by making Δ (k) sufficiently small.

Further, the filter coefficient estimator 9b of the adaptive filter 9 uses the adaptive algorithm to match or approximate the transfer function F (z) to the transfer function H (z) based on the signals x (k−τ) and e (k). Then, the filter unit 9a is updated. As described above, the LMS algorithm or the like is used as the adaptive algorithm. 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 position information control unit 11 detects position information indicating the distance between the microphone 1 and the speaker 7 as described above, and calculates the time delay of the feedback audio signal from the speaker 7 to the microphone 1 based on the position information. That is, the change of the transfer function H (z) of the acoustic feedback path 12 is estimated in a short time based on the position information, and the delay τ is determined and the filter coefficient of the adaptive filter 9 is updated based on this.

  For example, assume that the time delay of the feedback audio signal is 1000 taps. Here, assuming that the delay τ is set to 800 taps, a deviation of 200 taps occurs from the actual time delay. Therefore, not only the noise component is output from the adaptive filter 9 for the 200 taps until the feedback audio signal is input to the microphone 1, but also the number of taps used by the adaptive filter in the transfer function H (z) ( For example, the feedback audio signal corresponding to the portion exceeding 2048 taps) cannot be canceled. On the other hand, if the delay τ is set to 1200 taps, the output of the adaptive filter 9 is delayed by 200 taps from the feedback audio signal, and therefore corresponds to the transfer function of the first 200 tap portion of the transfer function H (z). The feedback audio signal cannot be canceled. That is, during that time, it is difficult to suppress howling because the feedback audio signal d (k) cannot be subtracted from the signal y (k) input from the microphone 1. If the delay τ is set to 950 taps here, the deviation from the actual time delay is only 50 taps, so that howling can be suppressed with high accuracy.

  In the loudspeaker system with built-in adaptive howling canceller according to the first embodiment of the present invention, the position information control unit 11 measures the distance between the microphone 1 and the speaker 7, calculates the time delay of the feedback audio signal from the sound speed, and sets the delay τ. To do. If the speed of sound is c and the detection distance between the microphone 1 and the speaker 7 is L, the delay τ can be calculated by the calculation of L / c. Since the speed of sound changes depending on the installation environment of the microphone 1 and the speaker 7, the position control unit 11 may further include a function of detecting temperature, atmospheric pressure, etc., and may be considered in the calculation. Note that the direction information of the microphone 1 and the speaker 7 may be considered in the calculation of the delay τ. The tap can be obtained by dividing the delay τ thus calculated by the sampling period time.

  Further, the position information control unit 11 instructs the adaptive filter 9 to update the filter coefficient based on the detected position information. The transfer function of the adaptive filter 9 is represented by F (z), and this transfer function is updated with 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).

  The loudspeaker system with built-in adaptive howling canceller according to the first embodiment of the present invention reduces the forgetting factor λ when the position information control unit 11 detects the position information of the microphone 1 and the speaker 7 and their positions change. In addition, the adaptive filter 9 is instructed to increase the step size α. As a result, the adaptive filter 9 can be updated faster than the original coefficient update, and howling can be suppressed more effectively.

  In addition, this invention is not limited to embodiment mentioned above, The deformation | transformation which is illustrated below is possible.

  FIG. 4 is a block diagram showing a configuration of a loudspeaker system with an adaptive howling canceller according to a modification of the first embodiment. Parts similar to those in FIG. 1 are denoted by the same reference numerals, and detailed description thereof is omitted. As shown in the figure, in this example, the gain information control unit 14 is connected to the microphone amplifier 2, the speaker amplifier 6, and the adaptive filter 9. The gain information control unit 14 acquires gain information from the microphone amplifier 2 and the speaker amplifier 6 and instructs the adaptive filter 9 to set the output level of the simulation signal do (k) according to the gain information.

  When the gain of the microphone amplifier 2 or the gain of the speaker amplifier 6 is changed, the level of the feedback audio signal also changes. For example, if the gain of the speaker amplifier 6 is changed to twice, the level of the feedback audio signal d (k) is about twice. However, the level of the simulation signal do (k) output from the adaptive filter 9 is not doubled at that time. Therefore, Δk increases and the possibility of howling increases. When the gain of the microphone amplifier 2 or the gain of the speaker amplifier 6 is changed in the adaptive howling canceller built-in loudspeaker system according to this modification, the information is transmitted to the gain information control unit 14, and the gain information control unit 14 is changed. Since the level of the output signal of the adaptive filter 9 is set to change according to the gain, the level of the output signal do (k) of the adaptive filter is immediately adapted after the level of the feedback audio signal d (k) changes. It becomes possible to change to a level appropriate for updating the coefficient of the filter. Control of increasing the output signal level of the adaptive filter when the gain of the microphone amplifier 2 or the speaker amplifier 6 is increased is effective even when the gain of the microphone amplifier 2 or the gain of the speaker amplifier 6 is suddenly changed. It is possible to suppress howling.

[Second Embodiment]
FIG. 5 is a block diagram showing the configuration of the adaptive howling canceller built-in loudspeaker system according to the second embodiment of the present invention. 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, in the loudspeaker system with a built-in adaptive howling canceller in this example, an audio signal input via the microphone 1 is converted into a digital signal by the A / D converter 3 via the microphone amplifier 2, and the equalizer amplifier 4 After the frequency component adjustment in step S4, the D / A converter 5 converts the signal into an analog signal, which is the same as that in the first embodiment, but is then input to the speaker amplifiers A1 to An. Speakers S1 to Sn are connected to the respective speaker amplifiers A1 to An, and the final stage of output is adjusted by the speaker amplifiers A1 to An, and sounds are produced from the speakers S1 to Sn.

  In this example, the plurality of speakers S1 to Sn all output the same signal in parallel, but the feedback audio signal fed back to the microphone 1 is input with a single transfer function in which the audio output from each speaker is integrated. Will be.

  Also in this example, each speaker S1-Sn and the positional information control part 11 are connected, and the positional information control part 11 detects the position of each speaker S1-Sn, and the positional information on the microphone 1. FIG. The microphone 1 and each of the speakers S1 to Sn can transmit and receive radio waves and measure the distance by measuring the radio arrival time. Note that, in this example as well, not only wireless transmission / reception, but also any device that can measure the distance to each other may be used. For example, the distance between each other may be actually measured using a laser, or the position may be measured using GPS. Alternatively, a specific pulse sound may be output from the speakers S1 to Sn and collected by the microphone 1 to measure the distance.

  In this example, the feedback audio signal fed back to the microphone 1 can be considered as one transfer function in which the voices output from the respective speakers are integrated, but the time delay of the feedback audio signal fed back to the microphone 1 is Each of the speakers S1 to Sn is different. Accordingly, even if the time delay of the feedback audio signal of the single speaker 7 and the microphone 1 is calculated and the delay τ is input as in the first embodiment, the acoustic feedback path is changed when the positions of the other speakers and the microphone are changed. There is a possibility that howling occurs due to a change in the transfer function.

  The position information control unit 11 in the second embodiment calculates the time delay of the feedback audio signal based on the input position information, but detects all the position information of the speakers S1 to Sn and the microphone 1, and from among them, The time delay of the speaker S closest to the microphone 1 is calculated. It is the speaker S closest to the microphone 1 that substantially affects the feedback audio signal, and howling can be effectively suppressed by calculating the time delay between the speaker S closest to the microphone 1 and the microphone 1. .

  For example, when the speaker S closest to the microphone 1 changes from the speaker S1 to the speaker S2, the position information control unit 11 calculates the time delay between the speaker S2 and the microphone 1 from the state in which the time delay between the speaker S1 and the microphone 1 is calculated. Change to the state to calculate.

FIG. 6 is a diagram showing a circuit configuration of a voice enhancement system with a built-in adaptive howling canceller according to the second embodiment of the present invention. In addition, the same code | symbol is attached | subjected to the component 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 audio signal input via the microphone 1 is converted into a digital signal y (k) by A / D conversion processing. The signal y (k) is supplied to the amplification path 16. Thereafter, the signal y (k) is amplified by the transfer function G ₁ (z) in the amplification path 16 and supplied to the speaker S as the signal x ₁ (k). The signal x ₁ (k) is converted into an audio signal by D / A conversion processing and output from the speaker S. The amplification path 16 is a transfer function of a path that is transmitted from the adder 10 to the D / A converter 5 via the equalizer amplifier 4.

Signals output from the speakers S <b> 1 to Sn are returned to the microphone 1 through the acoustic feedback path 15. Here, the acoustic feedback path 15 is an integrated acoustic path from the speakers S1 to Sn to the microphone 1, and this path is represented by a transfer function H ₁ (z). The feedback audio signal d ₁ (k) fed back via the acoustic feedback path 15 is mixed with the sound source audio signal s (k) and input to the microphone 1. The mixed audio signal input to the microphone 1 is output to the adder 10 as a digital signal by A / D conversion processing.

The delay circuit 8 outputs a signal x ₁ (k) with a delay τ obtained by estimating the delay time of the acoustic feedback path 15, and supplies the output signal x ₁ (k−τ) to the adaptive filter 9. Is done. Adaptive filter 9 outputs according transmit a signal simulating the feedback audio signal _{d 1 (k) d2 (k} ) function F ₁ (z). The adder 10 subtracts the signal d ₂ (k) from the signal y (k). Here, the signal y (k) is expressed by the sum of the sound source sound signal s (k) and the feedback sound signal d ₁ (k), that is, the expression y (k) = s (k) + d ₁ (k). . Therefore, the output signal e ₁ (k) of the adder 10 is e ₁ (k) = y (k) −d ₂ (k) = s (k) + d ₁ (k) −d ₂ (k) = s (k ) + Δ ₁ (k) (where Δ (k) = d ₁ (k) −d ₂ (k)). Therefore, howling can be suppressed by making Δ ₁ (k) sufficiently small.

The position information control unit 11 detects the position information indicating the distance between the microphone 1 and the speakers S1 to Sn as described above, and the time delay of the feedback audio signal to the microphone 1 based on the position information of the speaker S closest to the microphone 1. Is calculated. That is, the change of the transfer function H ₁ (z) of the acoustic feedback path 15 is estimated in a short time based on the position information of the speaker S closest to the microphone 1, and based on this, the delay τ is determined and the adaptive filter 9 filter coefficient is updated.

The transfer function of the adaptive filter 9 is represented by F ₁ (z), and this transfer function is also updated with coefficients called forgetting factor λ and step size α, as in the first embodiment. Also in this embodiment, when the positions of the speaker S and the microphone 1 change and a sudden change in the acoustic feedback path occurs, the adaptive filter 9 is instructed to decrease the forgetting factor λ and increase the step size α. . As a result, the adaptive filter 9 can be updated faster than the original coefficient update, and howling can be suppressed more effectively.

  The second embodiment can be modified as exemplified below.

FIG. 7 is a block diagram showing a configuration of a voice enhancement system with a built-in adaptive howling canceller according to a modification of the second embodiment. As described above, the gain information control unit 14 is connected to the microphone amplifier 2, the speaker amplifiers A <b> 1 to An, and the adaptive filter 9. The gain information control unit 14 acquires gain information from the microphone amplifier 2 and the speaker amplifiers A1 to An, and instructs the adaptive filter 9 to set the output level of the simulation signal d ₂ (k) according to the gain information.

Also in this example, since the gain information control unit 14 is set so as to change the level of the output signal of the adaptive filter 9 according to the changed gain, immediately after the level of d ₁ (k) changes, the adaptive filter The level of the output signal d ₂ (k) can be changed to a level suitable for updating the coefficient of the adaptive filter. Thus, howling can be effectively suppressed even when the gain of the microphone amplifier 2 or the gains of the speaker amplifiers A1 to An changes abruptly.

  As described above, the loudspeaker system with built-in adaptive howling canceller according to the present invention detects the position information of the microphone and the speaker, and sets the delay time of the signal input to the adaptive howling canceller according to the distance between them, so that the sound can be accurately obtained A change in characteristics can be detected, and howling can be effectively prevented.

1 is a block diagram showing a configuration of a speech enhancement system with an adaptive howling canceller according to a first embodiment of the present invention. 1 is a block diagram showing in detail the configuration of an adaptive howling canceller according to a first embodiment of the present invention. The figure which shows the circuit structure 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 the modification of 1st Embodiment. 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 circuit structure of the voice enhancement system with a built-in adaptive howling canceller which concerns on 2nd 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 the modification of 2nd Embodiment.

Explanation of symbols

1-microphone 2-microphone amplifier 3-A / D converter 4-equalizer amplifier 5-D / A converter 6-speaker amplifier 7-speaker 8-delay circuit 9-adaptive filter 9a-filter unit 9b-filter coefficient estimation unit 10- Adder 11-position information control unit 12-acoustic feedback path 13-amplification path 14-gain information control unit 15-acoustic feedback path 16-amplification path S-speaker A-speaker amplifier

Claims (3)

Amplifying means for amplifying an input signal input from a microphone and outputting the amplified audio signal to a speaker;
A delay unit and an adaptive filter for simulating a feedback transmission system from the speaker to the microphone are included, and a simulated signal obtained by processing the audio signal output to the speaker by the delay unit and the adaptive filter is subtracted from the audio signal input from the microphone. And howling canceller
A loudspeaker comprising:
Position detecting means for detecting a positional relationship between the microphone and the speaker;
Setting means for setting a delay time of the delay means based on a detection result of the position detection means;
Loudspeaker with The adaptive filter has a filter unit and a filter coefficient estimation unit,
The setting unit sets the filter coefficient estimation unit to update the filter coefficient based on the detection result of the position detection unit.
The loudspeaker according to claim 1. Gain adjustment means for adjusting the gain of the microphone and the speaker and detecting information on the gain adjustment is further provided.
The setting means sets the level of the simulation signal of the adaptive filter according to the gains of the microphone and the speaker.
The loudspeaker according to claim 1 or 2.

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