JP4737758B2 - Audio signal processing method and playback apparatus - Google Patents

Description

  In the multi-channel playback system, the digital signal processing is performed on the audio signal so as to obtain an appropriate acoustic characteristic at the listening position in consideration of the speaker characteristic, the acoustic space characteristic, and the audio system characteristic. The present invention relates to an audio signal processing apparatus capable of determining a crossover frequency.

  In order to provide high-quality playback sound in a multi-channel playback system, it is necessary to set the speaker sound pressure level, speaker distance, speaker connection phase, crossover frequency, frequency characteristics, etc. of each channel to appropriate values. . Conventionally, they have been set manually, but since it was a little difficult for the user to set them appropriately, recently, devices that can automatically set them have been used. . This device is called an automatic sound field correction device, and the control unit outputs the measurement signal from the calculation unit as a reproduction sound from the speaker, and picks up the reproduction sound of the measurement signal by the microphone installed at the listening position. By calculating based on the collected sound data, the calculation unit can calculate and set appropriate values such as speaker sound pressure level, speaker distance, speaker connection phase, crossover frequency, and frequency characteristics for each channel. It is a device that can.

  The automatic sound field correction process will be described with reference to FIGS. FIG. 7 shows a plurality of speakers around the listening point, for example, a front left speaker (FL), a front right speaker (FR), a center speaker (C), a subwoofer speaker (SW), a surround left speaker (SL), and a surround light. This is a so-called 5.1-channel multi-channel system in which speakers (SR) are arranged, and automatic sound field correction processing is performed from each speaker so that an optimal acoustic environment can be obtained at a listening point in such a system. This is a process for controlling the audio signal to be output.

  Specifically, a measurement signal is converted into a reproduction sound from each speaker and output, and the reproduction sound is collected by a microphone installed at a listening point. The measurement signal is filtered using a filter coefficient that decreases the signal level toward higher frequencies on the TSP (Time Stretched Pulse) signal, which is a signal whose energy has been increased by extending the impulse on the time axis. Use the applied signal. This is to prevent the speaker (tweeter) from being destroyed when a signal having a high signal level is output at a high frequency. The reproduction sound output from the speaker is picked up by the microphone, and the filter coefficient (the filter coefficient that is the reciprocal of the filter coefficient of the signal for measurement) that reduces the signal level toward the low frequency with respect to the frequency characteristics of the picked up sound. The result is used as a measurement result. Based on the measurement result, the sound pressure level of each speaker, the distance between the speakers, the distance between the listening point and the speaker, the frequency characteristic of the speaker, and the like are calculated. Based on the calculation result, the delay time of the audio signal output to each speaker, the frequency band output from the speaker, and the like are controlled so that an optimal acoustic environment is obtained at the listening point.

  FIG. 8 shows an example of the calculated frequency characteristics of each speaker. The seven speakers described above are illustrated with the horizontal axis representing frequency and the vertical axis representing sound pressure level.

Patent Document 1 discloses a method for measuring the reproduction limit frequency of a speaker and setting a cutoff frequency of a high-pass filter. This method sequentially outputs a sine wave signal of 20 Hz to 200 Hz from a transmitter, A method in which sound output from a speaker is picked up by a microphone, and a level value is plotted by a DSP (Digital Signal Processor) on the basis of a level detection result for each frequency to determine a reproduction limit frequency of the speaker. is there. However, the method described in Patent Document 1 does not consider the unevenness of the reproduction frequency characteristic of the speaker.
JP 2003-76374 A

  FIG. 9 is an explanatory diagram of the crossover frequency. The frequency band that can be output from the speaker differs depending on the size of the speaker itself or the speaker unit. For example, in the example shown in FIG. 8, a large front speaker (FL, FR) can output an audio signal up to a low sound range, but a small surround speaker (SL, SR) is smaller than a front speaker. The bass that can be output is not low. That is, the reproduction characteristics in the low sound range are poor. In such a case, as shown in FIG. 9, the low-frequency audio signal that cannot be reproduced by the surround speaker is added to the audio signal of the front speaker or the subwoofer speaker and output by the automatic sound field correction process. . This eliminates the problem that a low-frequency audio signal is not output among the audio signals of the surround speakers, and an optimal acoustic environment can be obtained at the listening point. The crossover frequency is a frequency for specifying a band to be cut from the audio signal supplied to the surround speaker so that the surround speaker cannot output it and output it to another speaker.

  When calculating the crossover frequency, there is a method for calculating the crossover frequency by calculating the impulse response from the data collected by the measurement as described above and performing frequency analysis using the Fourier transform. . Here, as described above, the frequency characteristic obtained by simply performing the Fourier transform is often very uneven, and it is difficult to find an accurate crossover frequency. To smooth the characteristics of the uneven surface. For the signal corresponding to the smoothed frequency characteristic, first, a reference level is obtained by averaging the levels in the middle / high range or the entire band. Next, from the reference level, for example, a point that becomes 3 dB lower is searched from a higher frequency toward a lower frequency, and the frequency that becomes 3 dB lower is set as a crossover frequency.

  In addition, smoothing here is averaging a level for every specific frequency band (smoothing width). As shown in FIG. 3, smooth frequency characteristics (solid line) are obtained by performing smoothing on the measured frequency characteristics (broken line) having unevenness of the speaker. As a smoothing method, a method such as a least square method can be used in addition to averaging.

  In the above calculation method, when smoothing the frequency characteristics, if the smoothing width is performed in a narrow band such as 1/6 octave or 1/3 octave, even in the partial dip caused by the characteristic of the speaker, If the dip is deeper than 3 dB, the point is erroneously recognized as a crossover frequency. In addition, when smoothing was performed with a band width (smoothing width) of 1 octave or more, those dips were smoothed, so that erroneous recognition at that frequency was avoided, but on the contrary, it was smoothed too gently. As a result, even if the original frequency characteristic is abruptly attenuated in the crossover frequency band, it becomes a gentle attenuation characteristic, and a large error occurs in the frequency of 3 dB attenuation before and after smoothing. Therefore, there is a problem that an accurate crossover frequency cannot be obtained.

  The present invention supplies a measurement signal to a speaker to be measured and converts it into a reproduced sound, picks up the reproduced sound with a microphone as an electric signal, and corresponds to the frequency characteristics of the speaker to be measured based on the electric signal. In the audio signal processing method for determining the crossover frequency of the loudspeaker to be measured, a smoothing with a wide smoothing width and a narrowing smoothing width with respect to the signal corresponding to the frequency characteristic of the loudspeaker Smoothing with different smoothing widths in at least two stages, such as smoothing in (1), and determining a crossover frequency based on a signal corresponding to the smoothed frequency characteristic.

  In addition, the present invention is a playback device including a signal input unit, an output unit, a speaker, a measurement signal generation unit, a signal processing unit, a control unit, and an operation unit, and an instruction signal for setting a processing mode is input from the operation unit. In this case, the measurement signal generation unit generates a measurement signal under the control of the control unit, and the output unit supplies the measurement signal to the speaker under the control of the control unit to generate reproduced sound. The microphone converts the reproduced sound into an electrical signal, and the signal processing unit creates a signal corresponding to the frequency characteristic of the speaker based on the electrical signal under the control of the control unit; Smooth the signal with at least two different smoothing widths, such as smoothing with a wide band smoothing width and smoothing with a narrow band smoothing width. Is characterized in that what determines the cross-over frequency based on a signal corresponding to the grayed frequency characteristics.

  According to the present invention, it is possible to avoid misrecognition at a partial dip and to determine a highly accurate crossover frequency with no large error, and to provide an optimal sound field environment for a listener. .

  In the present invention, the crossover frequency calculation step is obtained in two stages. First, as a first step, the frequency characteristics obtained by Fourier transform are smoothed over a wide band such as 1 octave, and a rough crossover frequency that avoids misrecognition by partial dips is searched from a high frequency. It is determined at the point of 3 dB attenuation, and a fixed band around that frequency is positioned as the optimum crossover frequency band. Next, as a second step, smoothing in a narrow band such as 1/3 octave or 1/6 octave is applied to the frequency characteristics of the optimal crossover frequency band among the frequency characteristics that are also Fourier transformed. A more accurate crossover frequency is searched from a higher frequency within the above-mentioned optimum crossover frequency range, a point of 3 dB attenuation is obtained, and that frequency is set as a final crossover frequency.

  FIG. 1 is a block diagram showing the main part of a playback apparatus according to an embodiment of the present invention. In the figure, A is an amplifying device, B is a speaker system, 1 is a signal processing unit, 1a is a memory, 2 is an output unit, 3 is a measurement signal generation unit, 4 is a sound pickup signal input unit, 5 is an operation unit, 6 Is a control unit, 7a to 7f are speakers, and 8 is a microphone. In this embodiment, the amplification device A and the speaker system B constitute a multi-channel reproduction system, and external devices such as a DVD reproduction device, a television, a CD reproduction device, and a VTR (not shown) can be connected. .

  The signal processing unit 1 performs signal processing such as decoding processing and delay processing on an audio signal input from an external device or the like. The decoding process includes, for example, a process of converting a 2-channel audio signal into a 6-channel audio signal, a process of expanding a compressed audio signal, and the like. The delay process is a process of adding a delay time to each decoded audio signal of each channel in order to give the viewer a sense of presence when outputting audio signals from a plurality of speakers.

  An environment in which the signal processing unit 1 listens to an audio signal in a space surrounded by a plurality of speakers 7a to 7f based on a sound collection signal input from the sound collection signal input unit 4 under the control of the control unit 6 described later. In order to obtain the optimal sound effect at the listener's listening point, the values of various parameters of the sound characteristics such as frequency characteristics, delay time, sound pressure level, crossover frequency of the output signal of each channel are automatically set. To do. The operation for making this setting is called automatic sound field correction processing.

  The output unit 2 includes a digital / analog conversion unit (not shown) and an amplification unit (not shown). The digital audio signal of a plurality of channels input from the signal processing unit 1 is converted into an analog signal. The front left channel speaker (hereinafter referred to as “FL speaker”) 7a, the front right channel speaker (hereinafter referred to as “FR speaker”) 7b, and the center channel speaker connected to the output unit 2 are amplified. (Hereinafter referred to as “C speaker”) 7c, surround left channel speaker (hereinafter referred to as “SL speaker”) 7d, surround right channel speaker (hereinafter referred to as “SR speaker”) 7e, subwoofer channel speaker (hereinafter referred to as “SL speaker”). , "SW speaker") 7f it Is the output.

  The output unit 2 is controlled by the control unit 6 when the automatic sound field correction process is performed, or when the listener changes the frequency characteristic using the operation unit 5 after the automatic sound field correction process. 3 is output to the speakers 7a to 7f of the respective channels.

  The measurement signal generator 3 generates a measurement signal for measuring the values of various parameters when automatically setting the values of various parameters of acoustic characteristics in the automatic sound field correction process under the control of the control unit 6. To do. In this embodiment, a signal obtained by filtering a TSP (Time Stretched Pulse) signal is used as the measurement signal.

  The sound collection signal input unit 4 receives the sound collection signal input from the microphone 8 connected to the sound collection signal input unit 4 during the automatic sound field correction process under the control of the control unit 6. Output to.

  The operation unit 5 includes an operation button for switching on / off the automatic sound field measurement processing, an operation button for outputting a measurement signal, an operation button for inputting or changing various parameter settings of acoustic characteristics, and the like. Is provided. When the listener presses the operation button, the operation unit 5 outputs an instruction signal corresponding to the pressed operation button to the control unit 6.

  The control unit 6 comprehensively controls the entire amplification device. The control unit 6 controls the signal processing unit 1 to perform decoding processing, delay processing, and the like on the audio signal input from the external device, and output the audio signal from the output unit 2.

  When an instruction signal for turning on automatic sound field correction processing is input from the operation unit 5, the control unit 6 causes the measurement signal generation unit 3 to generate a measurement signal, and the measurement signal is transmitted via the signal processing unit 1. Then, control is performed to output from the output unit 2 to each speaker. Then, the control unit 6 causes the signal processing unit 1 to obtain values of various parameters of acoustic characteristics based on the sound collection signal input from the sound collection signal input unit 4, and stores the obtained values in the memory 1a. And control to set the value as the value of various parameters of the acoustic characteristics.

  The automatic sound field correction process will be described. As shown in FIG. 1, in a multi-channel playback system such as a home theater system having six speakers 7a to 7f, the acoustic characteristics of the room, the frequency characteristics of the speakers used, the phase characteristics of each channel, and the audio signal from the speakers. The speaker 7a of each channel at the listening point depends on the transmission characteristics of the transmission path until the listener listens to the audio signal, the type and number of speakers to be used, the installation position of the speakers, the arrangement of the listening point and each speaker, etc. Since the phase, roll-off frequency, sound pressure level, distance, and frequency characteristics of the audio signal reaching from 7f change, even if the listener listens to the audio signal of the same sound source, the difference in the above characteristics, etc. Depending on the sound, the tone, sound field, and presence can be heard differently.

  The audio signal reproduced by an external device and input to the amplification device is based on the premise that the environment for listening to the audio signal is appropriate. For example, the sound pressure level of the output signal of each channel is equal, the distance between the speakers 7a to 7f, the frequency characteristics of each speaker are appropriate, and listening is performed in an environment where an appropriate type of speaker is used as the speaker configuration. Therefore, in order to reproduce the audio signal output from the amplification device with an optimum sound, an appropriate environment must be prepared.

  In order to realize such an environment, the conventional amplifying apparatus is equipped with a sound field correction function, and before the listener listens to the audio signal, the speaker configuration, the speaker size, the distance between the speakers, and the output signal of each channel. Values such as the sound pressure level and the frequency characteristic of the output signal of each channel are input using the operation unit, and the sound field is corrected so that an appropriate environment is obtained. However, these sound field correction operations are difficult to set appropriate values unless they are skilled persons, and are troublesome and require a lot of labor and time. Therefore, some conventional amplifying devices are provided with an automatic sound field correcting function for automatically performing the above-described sound field correction, and the amplifying device of this embodiment also has an automatic sound field correcting function. With this function, sound field correction is easy for the listener, and various parameter values can be set with high accuracy.

  The automatic sound field correction process in the amplification device A of this embodiment will be specifically described. The automatic sound field correction process includes a measurement process and a correction process, and the measurement process is first executed. The measurement process is a process of measuring the current acoustic characteristic at the listening point. For this process, a measurement microphone 8 is arranged at the listening point for the speakers 7a to 7f arranged as shown in FIG.

  In the measurement process, based on the control of the control unit 6, the measurement signal generation unit 3 generates a TSP signal that is an original signal of the measurement signal, and the signal processing unit (for example, a digital signal processor (Digital Signal Processor) : DSP) output to 1. The TSP signal is a signal that is expanded on the time axis by passing the impulse signal through a virtual filter having a delay time that varies depending on the frequency, and is flat from a low frequency to a high frequency. The signal processing unit 1 performs filter processing (first processing) on the TSP signal generated by the measurement signal generation unit 3 using a filter coefficient that decreases the signal level from a low frequency toward a high frequency. This filter process creates a signal whose signal level gradually decreases from a low frequency to a high frequency. This is used to prevent a speaker (tweeter) from being damaged when a signal having a high signal level is output at a high frequency, and the measurement signal is stored in the memory 1a. The signal processing unit 1 reads out a measurement signal from the memory 1a based on the control of the control unit 6, is digital-analog converted by the output unit 2, and sequentially outputs the signals from the speakers 7a to 7f. The measurement signals output from 7a to 7f are collected by the microphone 8.

  The collected signal is input to the collected signal input unit 4. The collected sound signal input unit 4 includes an analog-digital converter that converts an analog signal into a digital signal, converts the collected sound signal into a digital signal, and outputs the converted digital collected sound signal to the signal processing unit 1. The signal processing unit 1 temporarily stores the digital sound pickup signal input from the sound pickup signal input unit 4 in the memory 1a. The signal processing unit 1 uses a filter coefficient (a filter coefficient that is the reciprocal of the filter coefficient of the measurement signal) whose signal level decreases from a high frequency to a low frequency with respect to the digital sound pickup signal stored in the memory 1a. A filter process (second filter process) is performed. The first filter processing applied to the measurement signal is canceled by the second filter processing, and the measurement result of the speaker that outputs the measurement signal is obtained. An impulse response is obtained by performing convolution processing on the measurement result using data obtained by inverting the TSP signal on the time axis. Signals output from the amplifiers are output from the speakers 7a to 7f, and the transfer characteristics until the signals are picked up by the microphone 8 correspond to the impulse responses obtained here. A signal (measurement result signal) as a result of the second filter processing and the convolution processing is stored in the memory 1a. And the signal processing part 1 performs the process which calculates | requires the frequency characteristic etc. of each speaker based on the said measurement result signal memorize | stored in the memory 1a.

  Based on this impulse response, the sound pressure level, peak level, etc. of the output signal of each channel are observed, or FFT (Fast Fourier Transform) analysis is performed, so that the speaker configuration (presence / absence of speaker, speaker size), between speakers The phase relationship, the distance between the speakers, the distance of each speaker relative to the listening point, the sound pressure level and frequency characteristics of the output signal from each speaker, and the crossover frequency are obtained.

  Then, the process proceeds to correction processing. In the correction process, various parameters are set and corrected based on the measurement result obtained in the measurement process.

  The signal processing unit 1 sets the number of speakers based on the presence / absence of speakers based on the measurement result obtained by the measurement process, corrects the delay time between the channels based on the distance between the speakers or the distance from each speaker to the listening point, Correction of the difference in sound pressure level of the output signal between the channels based on the sound pressure level of the output signal, correction of the frequency characteristic of the output signal of each channel based on the frequency characteristic of the output signal of each speaker, output signal of each speaker The coefficient for setting the crossover frequency of the output signal of each channel based on the frequency characteristics is calculated by calculation.

  The coefficient obtained by the calculation is stored in the memory 1a, and when outputting the audio signal input from the external device, the frequency characteristic correction and the phase correction for the audio signal are performed using the coefficient stored in the memory 1a. Signal processing such as sound pressure level correction, delay time addition, and crossover frequency setting is performed and output.

  Here, the determination of the crossover frequency will be described with reference to FIGS. 2 to 6 with reference to FIG. FIG. 2 is a flowchart. The signal processing unit 1 performs an FFT (Fast Fourier Transform) analysis (Step 2) based on the measured impulse response (Step 1). When the signal processing unit 1 performs FFT analysis on the signal corresponding to the frequency characteristic obtained from the signal collected from the microphone 8, a signal (dotted line) corresponding to the frequency characteristic as shown in FIG. 3 is obtained. This signal is a signal whose frequency is a parameter and whose level value is an output value. In this description, the frequency characteristics of the surround speaker will be described as an example.

  Next, the signal is smoothed with a smoothing width such as one octave, for example, with a wide band smoothing width (step 3), and attenuated by 3 dB from the reference level obtained by averaging the sound pressure levels in the middle and high bands. The frequency is searched from a high frequency, and a band with a certain width of the frequency is set as the optimum crossover frequency range (step 4). In this case, when the signal corresponding to the frequency characteristic obtained by the measurement is smoothed with a smoothing width of one octave, the signal corresponding to the frequency characteristic in which the level unevenness is smoothed as shown in FIG. ) Is obtained. As shown in FIG. 8, the level of the reference frequency band (1 kHz to 2 kHz in this embodiment) of the signal corresponding to the frequency characteristic is used as a reference level, and the frequency characteristic is attenuated by 3 dB or more from the reference level. To do. As shown in FIG. 4, when there are a plurality of frequency points attenuated by 3 dB or more from the reference level, a low frequency point is selected.

  After that, smoothing with a smoothing width of a narrow band such as a 1/6 octave smoothing width is performed on the signal (dotted line) corresponding to the frequency characteristics as shown in FIG. 3 obtained by performing the FFT analysis (step) 5). In this case, at the frequency of the selected point, the range of one octave centered on the frequency is set as the optimal crossover frequency range, and the optimal crossover frequency range is, for example, 1/1 with respect to the signal (dotted line) described above. Smoothing in a narrow band such as 6-octave smoothing width.

  The frequency that is attenuated by 3 dB from the reference level obtained by averaging the sound pressure levels in the middle and high bands is searched again from the highest frequency in the optimum crossover frequency range obtained in step 4, and the frequency is finally determined. The crossover frequency is set (step 6). As a result of smoothing at 1/6 octave, a frequency characteristic (solid line) as shown in FIG. 5 is obtained. In the frequency characteristic, a frequency attenuated by 3 dB from the reference level is detected. As shown in FIG. 6, even if there are a plurality of frequencies that are attenuated by 3 dB from the reference level, those that are not in the optimum crossover frequency range (such as points indicated as “false recognition” in the figure) are erroneously recognized. It will not be done. When there are a plurality of frequencies attenuated by 3 dB in the optimum crossover frequency range, the lowest frequency is set as the crossover frequency.

  The smoothing will be described more specifically. Smoothing is smoothing the level of frequency characteristics for each specific frequency band. The measurement result signal obtained by the measurement (the signal stored in the memory 1a) is level data using the frequency as a parameter, and can be said to be a signal corresponding to the frequency characteristic. This signal is subjected to FFT processing (frequency analysis) for N points (points), and a level is obtained for each fs / N [Hz]. N points are the number of center frequencies in the frequency band to be averaged. In this embodiment, a smoothing width of 1 octave is set as the smoothing width of the wide band, and the level of the frequency band is averaged (smoothing). Specifically, for each center frequency (for example, 63, 125, 250, 500, 1k, 2k, 4k, 8k, 16k, ... [Hz]) from the low frequency to the high frequency, Levels in a specific frequency band centered on the center frequency are averaged, and the level in the frequency band is calculated (smoothing). For example, when the center frequency is 1 kHz, the level of the band from 707 Hz to 1.41 kHz centering on 1 kHz is averaged, and the average value is set to the level of the center frequency 1 kHz. Similarly, levels are obtained at other center frequencies. A characteristic obtained by connecting these levels with a line is set to a frequency characteristic that is smoothed by one octave.

  After smoothing in one octave is completed in this way, the optimum crossover frequency range is detected, and the level of the frequency band in a narrow band within the optimum crossover frequency range (smoothing) is performed. Specifically, when a frequency band of 50 Hz to 100 Hz (center frequency 75 Hz) is detected as the optimum crossover frequency band, a smoothing width of a narrow band (1/6 octave) within the frequency band of the optimum crossover frequency band. Smoothing. That is, for each center frequency (53, 59, 67, 75, 84, 94 [Hz]), the level in a specific frequency band centered on each center frequency is averaged, and the level in the frequency band is calculated. calculate. For example, when the center frequency is 53 Hz, the level of the band from 50 Hz to 56 Hz centering on 53 Hz is averaged, and the average value is set as the level of the center frequency 53 Hz. Similarly, the level is found at other center frequencies. A characteristic obtained by connecting these levels with a line is a frequency characteristic obtained by smoothing with a smoothing width of 1/6 octave. For this frequency characteristic, the crossover frequency is determined based on a level attenuated by 3 dB from the reference level.

  In addition, when the frequency of 3 dB attenuation is not found in the optimal crossover frequency band, the frequency that is closest to the 3 dB attenuation is set as the crossover frequency. Based on the crossover frequency obtained as a result of this correction processing, the control unit 6 sets a coefficient of the crossover frequency in the signal processing unit 1. The signal processing unit 1 performs a filtering process for allowing an audio signal having a frequency equal to or higher than the crossover frequency to pass through to an audio signal of a channel (surround channel audio signal) output from the surround speaker. Thereby, the audio signal of the frequency band more than a crossover frequency among the audio signals of a surround channel is output from a surround speaker.

  Further, the signal processing unit 1 performs addition processing so as to add an audio signal having a frequency band lower than the crossover frequency of the surround channel audio signal to another speaker (for example, a front speaker or a subwoofer speaker). As a result, among the surround channel audio signals, an audio signal in a frequency band equal to or lower than the crossover frequency is output from a speaker other than the surround speakers (for example, a front speaker or a subwoofer speaker).

  In this embodiment, the crossover frequency is determined based on a level attenuated by 3 dB from the reference level, but other values may be selected. Further, at the time of wide band (1 octave) smoothing and narrow band (1/6 octave) smoothing, the level attenuated from the reference level may be set to a different level in each case. Further, although one octave is exemplified as the broadband smoothing width, it may be a smoothing width of about 1/3 to 1.5 octaves, and the narrowband smoothing width is 1/2 to 1/10 of the wideband smoothing width. A smoothing width of about an octave may be used.

  In the present invention, the crossover frequency of the speaker system can be measured, and the audibility characteristics in the multi-channel reproduction system can be set satisfactorily.

It is a block diagram which shows the principal part of the reproducing | regenerating apparatus of one Example of this invention. It is a flowchart which shows the operation | movement of the crossover frequency calculation process of this invention. It is explanatory drawing about the smoothing of crossover frequency calculation. It is explanatory drawing about the smoothing of crossover frequency calculation. It is explanatory drawing about the smoothing of crossover frequency calculation. It is explanatory drawing about the smoothing of crossover frequency calculation. It is explanatory drawing of an example of arrangement | positioning of the speaker of a multichannel system. It is explanatory drawing which shows an example of the frequency characteristic of each calculated speaker. It is explanatory drawing about a crossover frequency.

Explanation of symbols

  A ... amplifier, B ... speaker system, 1 ... signal processing unit, 1a ... memory, 2 ... output unit, 3 ... measurement signal generation unit, 4 ... sound pickup signal input unit, 5 ... operation unit, 6 ... control unit 7a ... FL speaker, 7b ... FR speaker, 7c ... C speaker, 7d ... SL speaker, 7e ... SR speaker, 7f ... SW speaker, 8 ... microphone.

Claims (6)

A measurement signal is supplied to the speaker to be measured and converted into reproduced sound. The reproduced sound is picked up by a microphone to be an electric signal, and a signal corresponding to the frequency characteristic of the speaker to be measured is obtained based on the electric signal. In the audio signal processing method for determining the crossover frequency of the measured speaker,
The signal corresponding to the frequency characteristics of the measured loudspeaker is subjected to smoothing with different smoothing widths of at least two stages, such as smoothing with a wide band smoothing width and smoothing with a narrow band smoothing width, to obtain a smoothed frequency characteristic. An audio signal processing method, wherein a crossover frequency is determined based on a corresponding signal. A measurement signal is supplied to a speaker to be measured and converted into a reproduced sound. The reproduced sound is picked up by a microphone to be an electric signal, and a first characteristic corresponding to the frequency characteristic of the measured speaker is based on the electric signal. In an audio signal processing method for obtaining a signal and determining a crossover frequency of the measured speaker,
The first signal is smoothed with a wide band smoothing width, and the first crossover frequency is determined based on the second signal corresponding to the frequency characteristic subjected to the smoothing with the wide band smoothing width. Thereafter, a frequency range including the first crossover frequency is determined,
The first signal in the determined frequency band is subjected to smoothing with a narrow-band smoothing width that is narrower than the wide-band smoothing width, and the smoothing is performed with the narrow-band smoothing width. A second crossover frequency is determined based on a third signal corresponding to the audio signal processing method, and the second crossover frequency is used as the crossover frequency of the speaker under measurement. The smoothing width in the wide band is about 1 to 3 to 1.5 octaves, and the smoothing width in the narrow band is a smoothing width about octave of 1/2 to 1/10 of the smoothing width in the wide band. The audio signal processing method according to claim 1, wherein A reproduction apparatus having a signal input unit, an output unit, a speaker, a measurement signal generation unit, a signal processing unit, a control unit, and an operation unit,
When an instruction signal to enter the processing mode is input from the operation unit,
The measurement signal generator generates a measurement signal under the control of the controller,
The output unit supplies the measurement signal to the speaker and converts it into reproduced sound under the control of the control unit,
The microphone converts the reproduced sound into an electric signal,
The signal processing unit creates a signal corresponding to the frequency characteristic of the speaker based on the electrical signal under the control of the control unit, and performs smoothing with a wide band smoothing width and narrow band smoothing width with respect to the signal. A reproduction apparatus characterized by performing smoothing with different smoothing widths in at least two stages, such as smoothing in (1), and determining a crossover frequency based on a signal corresponding to the smoothed frequency characteristic. A reproduction apparatus having a signal input unit, an output unit, a speaker, a measurement signal generation unit, a signal processing unit, a control unit, and an operation unit,
When an instruction signal to enter the processing mode is input from the operation unit,
The measurement signal generator generates a measurement signal under the control of the controller,
The output unit supplies the measurement signal to the speaker and converts it into reproduced sound under the control of the control unit,
The microphone converts the reproduced sound into an electric signal,
The signal processing unit creates a first signal corresponding to the frequency characteristic of the speaker based on the electrical signal under the control of the control unit, and smoothes the first signal with a wide band smoothing width. And determining a first crossover frequency based on a second signal corresponding to the frequency characteristic subjected to smoothing with the wideband smoothing width, and determining a frequency range including the first crossover frequency Then, the first signal in the determined frequency band is smoothed with a narrow-band smoothing width that is narrower than the wide-band smoothing width, and smoothed with the narrow-band smoothing width. A second crossover frequency is determined based on a third signal corresponding to the frequency characteristic, and the scan signal is obtained with the second crossover frequency. Reproducing apparatus, characterized in that it is an crossover frequency over mosquitoes. The smoothing width in the wide band is about 1 to 3 to 1.5 octaves, and the smoothing width in the narrow band is a smoothing width about octave of 1/2 to 1/10 of the smoothing width in the wide band. The playback apparatus according to claim 3 or 4, characterized in that:

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