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JPH01248798

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DESCRIPTION JPH01248798
[0001]
Industrial application field This invention is an automatic equalization apparatus (so-called
automatic equalizer) that changes the frequency characteristic of a loudspeaker according to the
acoustic space characteristic from the loudspeaker to the acoustic signal listening point in a
loudspeaker system. It is PAt thing. [Conventional art] In the conventional automatic
loudspeakers used in loudspeakers, white noise or pink noise is used as a regulation signal, and
when there is an audience, it works There was a problem. Also, the conventional automatic
equalizer passes the signal from the reference microphone through a plurality of band pass
filters for frequency analysis, compares the output signal level with the signal level in a plurality
of preset bands, and Control of the multiple band-pass filters of the equalizer so as to be the
same. [Problems to be Solved by the Invention] However, since the accuracy and resolution of
automatic equalization depend on a plurality of analysis and other band filters, it is important to
obtain high-precision amplitude and phase frequency characteristics. It was difficult. Therefore,
the present invention has been made to solve the above-mentioned problems, and the purpose of
the invention is to use a special regulation signal for desired amplitude and phase frequency
characteristics in a wide listening area. An object of the present invention is to provide a
frequency characteristic equalizer for a loudspeaker system that can be realized with high
accuracy. [Means for Solving the Problems] The gist of the present invention for achieving the
above-mentioned object is that, in a voice amplification system in which an input signal is
amplified by a power amplifier and broadcasted from a loudspeaker, the input signal is delayed
by a predetermined time. Delay circuit section, a filter section in which a coefficient
corresponding to a desired frequency characteristic for passing an output signal of the delay
circuit section is set, a plurality of sound receiving microphones distributed in the listening area
of the broadcast, and these sound receiving An error detection unit for taking out as an error
signal the difference between the sum signal obtained by applying predetermined weighting to
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each of the output signals from the sound microphones and adding them as an error signal; An
adaptive digital filter unit for self-adjustment so that the average value of the square of the error
signal received from the error detection unit is minimized. It consists in the number
characteristics equalizer. The input signal is supplied to the adaptive digital filter and the delay
circuit, and is delayed by the delay circuit for a predetermined time. The delayed input signal
passes through the filter unit having the desired amplitude and phase frequency characteristics
and is sent to the error detection unit. On the other hand, the output signals from the plurality of
sound receiving microphones are subjected to weighting and then input to the error detection
unit.
The error detection unit subtracts the weighted sum signal of the sound receiving microphone
output signal from the output signal of the filter unit, and sends it as an error signal to the
adaptive digital filter unit . The adaptive digital filter self-adjusts its amplitude and phase
frequency characteristics so that the average value of the square of the error signal is minimized.
As a result, the sound pressure frequency characteristic in the listening area where the sound
receiving microphone is installed is corrected and equalized with high accuracy until it becomes
almost equal to the frequency characteristic set in the filter section. At this time, since a plurality
of sound receiving microphones are distributed, equalization correction is performed on an
average over a wide listening area. Also, by performing adaptive signal processing in the adaptive
digital filter unit based on the error signal, a special signal for regulation becomes unnecessary,
and a normal music signal or an audio signal is used as a signal for regulation. Next, an
embodiment of the present invention will be described with reference to FIG. The main humanpower signal supply terminal unit 10, the adaptive digital filter unit 1, the power amplifier 12,
and the loudspeaker 13 are connected in series. The loud speaker 13 is provided in the loud
broadcast area 14 and receives the loud broadcast sound as a reference input signal in the
listening area in the loud broadcast area 14 as a sound receiving microphone M1. A plurality of
M2 to Mn are dispersed. Receiving microphone M1. The output signals of M2 to Mn are supplied
to the error detection unit 11 after being subjected to processing of weighting determined by the
distance from the loudspeaker 13 and the like at the sections A1 ° Δ2 to An. Further, to the
main human-power signal terminal unit 10, a filter unit G1 error detection unit 11 in which a
transfer function is set so as to have desired amplitude and phase frequency characteristics of
the delay circuit unit D1 is connected in order. The signal delay time of the delay circuit section
is obtained by adding a half impulse response time of the adaptive digital filter section l "to the
average value of the sound wave propagation time between the loudspeaker 13 and the n sound
receiving microphones M. The value is set to The error detection unit 11 subtracts the output
signal of the filter unit G and the sum of the output signals of the microphones M after weighting
to extract the error signal of the both, and this is used as an adaptive digital filter unit. It sends
out to H. In the weighting of the output signals of the plurality of sound receiving microphones M
distributed in the listening area, the output signal level of each microphone M is 1 or less
because the adaptive digital filter unit 1 operates stably. Also, processing is performed such that
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the sum of the output signals j of the microphones M is less than or equal to one.
Since a plurality of sound receiving microphones M are distributed in the listening area, the
average equalization correction of the frequency characteristic is performed over a wide range of
the listening area. Therefore, it is useful for characteristic equalization in sound space where the
listening area is relatively wide. A desired frequency vs. phase / amplitude characteristic is
realized with high accuracy by the combined overall action of the delay circuit unit D1 error
detection unit 11 and the adaptive digital filter unit H. Under the present circumstances, if it is a
signal from which the frequency band of a desired characteristic is almost)-1 even if it does not
use special regulation signals, such as white noise and pink noise, using a normal audio signal or
a music signal The characteristic equalization processing is automatically performed in a short
time that the listener does not notice. The desired characteristics are set in the filter section G.
When the transfer function 1 parameter is set in the filter section G, the imaging width and phase
frequency characteristics become flat. [Effects of the Invention: As mentioned above, according to
the present invention, in the loudspeaker system, it is possible to realize average frequency
characteristic equalization over a wide listening area, and furthermore, no special regulation
signal is used. However, the remarkable effect of being able to perform characteristic
equalization correction of high resolution is obtained.
[0002]
Brief description of the drawings
[0003]
FIG. 1 is a block diagram of a frequency characteristic equalizer for a loudspeaker system
according to the present invention.
A: weighted processing unit, D: delay circuit unit, G: filter unit, H: adaptive digital filter unit, M:
microphone for receiving sound, 10: main Human power signal supply terminal unit 11, error
detection unit 13, loudspeaker for loud-speaking, broadcast broadcast area.
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