摘要:
A sound signal picked-up by a microphone is processed in a howl suppresser including a digital filter. A frequency analyzer performs frequency analysis of the picked-up sound signal. A howl detector detects a howl contained in the sound signal from a result of frequency analysis by the frequency analyzer. An operation parts calculates coefficients to be set to the digital filter to suppress the howl according to a detection result by the howl detector, and a control part sets the calculated coefficients to the digital filter. The howl detector judges that a maximum peak power level among power levels of the sound signal in a frequency region analyzed by the frequency analyzer is a howl component when a ratio of the maximum peak power level to a mean power level of the sound signal is larger than a predetermined threshold level, preferably for a predetermined threshold time. Also, a threshold control means is provided for controlling the threshold level and/or the threshold time so as to enhance accuracy of howl detection.
摘要:
A sound field controller for generating apparent sound sources (CL, CR, LL, RR) by adjusting the amplitude and delay time of a sound signal so that the sound will be perceived by plural listeners (8, 8-1, 8-2) as sound coming from a location separated from the specific location of the front speakers (4, 6), and for additionally controlling the effect of the apparent sound sources by evaluating the attributes of the source sound signal. The controller includes FIR filters (11, 13) for generating a left sound pattern signal (hL(n)), FIR filters (12, 14) for generating a right sound pattern signal (hR(n)), a first delay circuit (15, 18) for delaying the left and right sound pattern signals by a first predetermined time and applying the delayed left and right sound pattern signals to the left and right speakers, respectively, to introduce an apparent sound source (CL) located left rear of a center listener; and a second delay circuit (16, 17) for delaying the left and right sound pattern signals by a second predetermined time and applying the delayed left and right sound pattern signals to the right and left speakers, respectively, to introduce an apparent sound source (CR) located right rear of a center listener.
摘要:
An apparatus for calculating a filter factor for a digital filter which can be made with a simple construction and in which the time period for calculation can be shortened as compared with conventional techniques. The apparatus comprises inputting means including amplitude inputting means for inputting a desirable amplitude frequency property and phase inputting means for inputting a desirable phase frequency property; reverse-Fourier transformation means for obtaining an impulse response corresponding to a transfer function having the inputted amplitude frequency property and the inputted phase and frequency property; and setting means for setting to said external filter the impulse response obtained by the reserve-Fourier transformation, as a filter factor.
摘要:
In the process of forming the effect sound for reproducing a sound field, a down-sampling part for reducing the sampling frequency of the digital signal is disposed before the step of the digital operation which becomes the central part of the forming process, by which a large amount of operation processing is carried out at a low sampling rate. With respect to the direct sound, processing such as gain adjustment is carried out without decreasing the sampling frequency. This is a sound field control system which permits to realize the higher quality and natural sound field control effect without much increasing the hardware scale.
摘要:
A sound field variable apparatus comprises direct-sound and indirect-sound collecting microphones (1,4) an adaptive filter (7) for obtaining a transfer function between the indirect-sound collecting microphone (4) and an indirect-sound reproducing speaker, and a signal generator (8) for generating a signal used for measuring a transfer function to be set in the adaptive filter (7). The transfer function between the indirect-should collecting microphone set at a desired position and the indirect-sound reproducing speaker is measured by a control circuit (20), and the transfer function thus measured is set to the adaptive filter (7) so that an echo generated caused by the acoustic combination between the indirect-sound reproducing speaker can be cancelled whereby to prevent howling from taking place.
摘要:
In the process of forming the effect sound for reproducing a sound field, a down-sampling part for reducing the sampling frequency of the digital signal is disposed before the step of the digital operation which becomes the central part of the forming process, by which a large amount of operation processing is carried out at a low sampling rate. With respect to the direct sound, processing such as gain adjustment is carried out without decreasing the sampling frequency. This is a sound field control system which permits to realize the higher quality and natural sound field control effect without much increasing the hardware scale.
摘要:
A digital equalizer for audio system applications is based on a FIR (17) (finite impulse response) digital filter whose amplitude and phase/frequency characteristics can be respectively independently established in accordance with input data representing an arbitrary amplitude/frequency characteristic (11) and input phase data (12) from which an arbitrary phase/frequency characteristic for the filter can be derived. In addition to audio frequency response equalization, the apparatus can be provided with a microphone howl suppression function.
摘要:
An apparatus for calculating a filter factor for a digital filter which can be made with a simple construction and in which the time period for calculation can be shortened as compared with conventional techniques. The apparatus comprises an inputting circuit (1) for inputting a desirable frequency property, a division circuit (2) for dividing the inputted frequency property into a plurality of frequency bands, and a calculating circuit for obtaining filter factors for realizing a frequency property of each of the divided frequency bands. The inputting circuit inputs the frequency property with a frequency resolution corresponding to the number of the filter factors. The division circuit performs a correction for a division so that the frequency property becomes zero from a frequency over the high cut-off frequency of a band-pass filter toward the Nyquist frequency of the frequency band corresponding to the transversal filter, the Nyquist frequency being 1/2 of the sampling frequency. The calculating circuit has a transformation unit (3) for performing a Hilbert transformation or a linear phase transformation with respect to the respective frequency properties divided by said division means.
摘要:
An apparatus for calculating a filter factor for a digital filter which can be made with a simple construction and in which the time period for calculation can be shortened as compared with conventional techniques. The apparatus comprises an inputting circuit (1) for inputting a desirable frequency property, a division circuit (2) for dividing the inputted frequency property into a plurality of frequency bands, and a calculating circuit for obtaining filter factors for realizing a frequency property of each of the divided frequency bands. The inputting circuit inputs the frequency property with a frequency resolution corresponding to the number of the filter factors. The division circuit performs a correction for a division so that the frequency property becomes zero from a frequency over the high cut-off frequency of a band-pass filter toward the Nyquist frequency of the frequency band corresponding to the transversal filter, the Nyquist frequency being 1/2 of the sampling frequency. The calculating circuit has a transformation unit (3) for performing a Hilbert transformation or a linear phase transformation with respect to the respective frequency properties divided by said division means.