JP2014045435A - Acoustic correction device, acoustic correction method and acoustic reproducer - Google Patents

Abstract

PROBLEM TO BE SOLVED: To allow an acoustic device to provide a listener with comfortable sound, even if the listening position of a user changes.SOLUTION: An acousto-electric transducer collects sound at a plurality of measuring points in front of an acoustic device, and obtains an acoustic sound collection signal at the plurality of measuring points. When analyzing the acoustic characteristic of the acoustic sound collection signals at the plurality of measuring points, a characteristic analyzer divides each acoustic sound collection signal into at least two frequency bands, performs acoustic analysis by using a different analysis processing method for each frequency band, and obtains one measurement result by averaging the acoustic analysis results at the plurality of measuring points. A characteristic setter obtains parameters for setting the characteristics of the acoustic device based on the one measurement result.

Description

  Embodiments described herein relate generally to an acoustic correction device, an acoustic correction method, and an acoustic reproduction device.

  For example, an acoustic correction device that corrects the output characteristics of the audio device has been developed so that the sound output from the audio device of the flat-screen television receiver can be heard comfortably by the viewer.

JP 2011-239097 A JP 2012-114826 A Special table 2008-541654

  With a conventional acoustic correction device, a comfortable sound is heard at a certain distance in front of the acoustic device, but if the listening position changes back and forth, the quality of the sound may be lowered.

  According to the present embodiment, it is possible to provide an acoustic correction device, an acoustic correction method, and an acoustic device that enable the acoustic device to provide a comfortable sound to the listener even if the listening position of the user changes. With the goal.

  According to the embodiment, in the acoustic correction device that measures the sound output from the acoustic device and corrects the acoustic characteristics based on the measurement result, the sound is collected at a plurality of measurement points in front of the acoustic device, and the plurality When the acoustic characteristics of the acoustic sound collecting signals at the plurality of measurement points are analyzed, and the acoustic sound collecting signals at the plurality of measurement points are analyzed in at least two frequency bands. A characteristic analyzer that divides and performs acoustic analysis using different analysis processing methods for each frequency band, averages the results of the acoustic analysis of the plurality of measurement points, and obtains one measurement result; And a characteristic setting unit for obtaining a parameter for setting the characteristic of the acoustic device based on two measurement results.

It is a block diagram which shows the example of 1 structure of embodiment. It is a figure which shows the example of a TSP signal. It is a figure which shows the TSP synthetic | combination signal which prepared two TSP signals of FIG. It is a figure which converts the TSP synthetic | combination signal of FIG. 3 into a frequency domain, and shows an amplitude characteristic. It is a figure which shows the characteristic at the time of transforming each two TSP signals which time shifted mutually into the frequency domain. It is a figure which shows the example of the weighting function used for the synthesis | combination of the transfer characteristic of the low sound region side and the transfer characteristic of the high sound region side in this embodiment. It is a figure which shows the example which allocated the weighting count on a frequency axis, when synthesize | combining the analysis result of a low-pitched range and a mid-high range. It is a figure which shows the example of the audio equipment which has a signal correction part by which the parameter was correct | amended by this audio correction apparatus.

  Hereinafter, embodiments will be described with reference to the drawings. FIG. 1 is a block diagram illustrating a configuration example of the embodiment. A measurement signal generator (also referred to as an inspection signal generator) 101 outputs a measurement signal for measuring acoustic characteristics. An output signal (also referred to as a test signal) of the measurement signal generator 101 is supplied to an external input of the acoustic device 102 and is input to the audio output unit 102B via the switch circuit SW1. The acoustic device 102 includes a signal correction unit (equalizer) 102A that performs acoustic correction, and a sound output unit 102B that is a so-called speaker including a sound signal amplifier and / or an electroacoustic transducer or system as a speaker system.

  The acoustic output (which may be referred to as inspection sound) output from the acoustic device 102 propagates the sound wave and is collected by the microphone (that is, the acoustoelectric converter) 103. The microphone 103 collects the sound output of the sound device 102 at a plurality of measurement points in front of the sound device 102. For example, the microphone 103 is spatially moved back and forth, left and right, and up and down in front of the acoustic device 102 to collect the sound output of the acoustic device 102. As a result, sound outputs at a plurality of measurement points in the three-dimensional space are collected. Alternatively, a plurality of microphones may be used to collect sound outputs from a plurality of measurement points for each block in the three-dimensional space.

  The signal is input to the characteristic analyzer 104 that analyzes the acoustic characteristics of the acoustic signal from the microphone 103. The characteristic analyzer 104 includes a measurement signal sound collection unit 104A and an acoustic characteristic analysis unit 104B. The measurement signal sound collection unit 104A collects the sound collection signals measured at the plurality of measurement points and temporarily holds them. When the acoustic characteristic analysis unit 104B analyzes the acoustic characteristics of the sound collection signals at a plurality of measurement points, for example, the sound characteristic analysis unit 104B averages the sound collection signals measured at the plurality of measurement points and combines them into one measurement result. Each acoustic sound collection signal is divided into at least two frequency bands, and acoustic analysis is performed using a different analysis processing method for each frequency band. Then, a result of acoustic analysis using different analysis processing methods is synthesized to obtain one measurement result.

  The measurement result is input to the equalizer characteristic setting unit 105. The equalizer characteristic setting unit 105 obtains a substantially reverse characteristic with respect to the measurement result. Furthermore, target characteristics that are not flat may be set as a correction result. In that case, the inverse characteristic of the ratio between the target characteristic and the measurement result may be obtained. If the target characteristic is a flat characteristic, an equalizer characteristic that is a substantially opposite characteristic of the measurement result is obtained. A parameter for the signal correction unit 102A is set based on the substantially reverse characteristic. This parameter is a parameter for canceling distortion (amplitude distortion, phase distortion, etc.) included in the measurement result.

  For example, in the case of an amplitude frequency characteristic, a reproduction sound having a flat frequency characteristic can be obtained by providing a parameter to the signal correction unit 102A.

  For example, the signal correction unit 102A corrects the amplitude frequency characteristic of the reproduced sound signal separated from the television signal received by the television receiver according to the parameter. The reproduced acoustic signal input to the signal correction unit 102A may be a signal reproduced from a recording medium or a signal taken in via a network. The signal correction unit 102A is incorporated in various devices and products that process acoustic signals.

  Here, in this apparatus, the acoustic sound collection signal is divided into at least two frequency bands, and acoustic analysis is performed using different analysis processing methods for each frequency band. Of the divided frequency bands, the low frequency range averages the sound collected signal in time synchronization, and for the middle and high range, the sound collected signal is converted to the frequency domain and divided into amplitude and phase in the frequency domain and averaged. Is being processed.

  Further, as a reference for dividing the frequency band, the so-called f0 which is the lowest resonance frequency of the speaker set by the operator in order to set the low frequency range where the reproduction sound pressure is insufficient due to the reproduction performance limit of the reproduction system is used. Yes. For example, a frequency about twice as high as f0 is used as a reference for division.

  Further, as a reference for dividing the frequency band, a low sound may be set up to a range where the influence of the phase shift becomes slight according to the synchronization accuracy of the acoustic sound collection signal. In order to synchronize the sound collected signals at different measurement points, a method of aligning the impulse response times can be considered. However, since the sound collection signals at different measurement points have different waveforms at each measurement point, it is impossible to strictly align the time. Here, in the low frequency region, since the cycle is long, the synchronization shift of the acoustic sound collection signals at different measurement points becomes a slight phase difference and the influence is slight. Therefore, in the low sound range, it is preferable to perform a synchronization process for aligning the impulse response time with respect to the averaging process of the collected sound signal. However, in the mid-high range, if there is a shift in the synchronization time of the sound collection signal at different measurement points, the signal period is short because the frequency is high, and a relatively short time shift cannot be ignored. There is a problem in that when the sound collection signal at the measurement point is totalized, the influence of phase interference enters and correct measurement characteristics cannot be obtained. Therefore, for the mid-high range, the sound collection signal is converted into the frequency domain and averaged by dividing into the amplitude and phase in the frequency domain.

  Hereinafter, the sound output sound collection and the sound collection signal analysis procedure will be described in detail.

  The measurement signal generator 101 generates a Time Stretched Pulse (TSP) signal and reproduces it from the speaker of the audio device 102. The microphone 103 collects the reproduced sound including the transfer characteristic of the measured sound field.

  In this type of conventional technology, there is known a technique in which a microphone is installed at the listening position to obtain an impulse response at the listening position. However, when the listening position deviates from the measurement position, the correction result may feel uncomfortable.

  In the present embodiment, sound is collected at a plurality of measurement points assuming the listening position in order to obtain a suitable correction result at a wide range of listening positions. Therefore, measurement is performed by the following method. While repeatedly reproducing the TSP signal from the speaker of the audio device 102, for example, holding a single microphone by hand and collecting the reproduced sound at a plurality of measurement points while appropriately moving the position.

  Next, transfer characteristics representing the entire target sound collection space are calculated from the sound collection results. If the microphone 103 is fixed, the sound propagation time from the speaker of the acoustic device 102 to the microphone 103 is constant. For this reason, noise that is asynchronous with the TSP signal that is the measurement signal is canceled by synchronously adding the collected sound signal at regular intervals. By using this canceling effect, the S / N can be improved and the transfer characteristic can be obtained.

  However, in the present embodiment, the distance from the speaker to the microphone changes from moment to moment by moving the microphone while holding it in the hand, so that the propagation delay time changes every moment. In addition, since the sound propagation direction from the speaker to the microphone also changes, the sound collection signal is not constant in amplitude characteristics and phase characteristics due to the influence of the directivity characteristics of the speakers.

  Therefore, when attempting to synchronously add the sound collection results at a plurality of measurement points, at a frequency causing phase interference due to a difference in transmission delay time or a phase characteristic, as shown in FIG. Valley) will appear.

  2 to 4 show examples for explaining the principle of phase interference. FIG. 2 is a TSP signal created with 256 samples. The horizontal axis is the number of samples, and the vertical axis is the amplitude. FIG. 3 shows a case where two of the TSP signals are synthesized by shifting by three samples, but it can be seen that in the frequency component portion where the two signals are in opposite phases, the signals cancel each other and the amplitude is reduced. . FIG. 4 shows the amplitude characteristics by Fourier transforming the time domain TSP synthesized signal of FIG. 3 and converting it to the frequency domain, that is, the frequency domain.

  Similarly, in actual measurement, for each measurement point, the sound collected signal is not constant in both amplitude and phase characteristics, so there is a portion that is synthesized in the opposite phase when averaging multiple signals in the time domain. In this case, phase interference with a small amplitude occurs.

  In the present embodiment, it is aimed to avoid an error in the amplitude characteristic caused by the phase interference. In other words, the band causing phase interference is converted into the frequency domain for each measurement point, divided into amplitude characteristics and phase characteristics, and the averaging process of amplitude characteristics and the averaging process of phase characteristics are calculated separately, The acoustic characteristics are not affected by phase interference.

  For example, in contrast to the example of FIG. 3 in which the principle of interference is described above, the two TSP signals are shifted in time from each other, but both of the characteristics of FIG. 5 converted to the frequency domain have flat amplitude characteristics. It is. Therefore, (1) even if a plurality of signals converted to the frequency domain are averaged, flat characteristics are maintained. On the other hand, (2) phase characteristics can be calculated while maintaining a relative phase relationship by performing an averaging process in a complex region where a phase delay occurs.

  On the other hand, in the low frequency region, in the frequency band below the limit frequency of the reproduction capability of the speaker, the sound pressure of the reproduced sound may drop greatly and the reproduced sound may be buried in background noise. In this case, in the method for obtaining the average of the amplitude characteristics, noise and signal cannot be distinguished, and the measurement result is affected by noise. For this event, if the noise is uncorrelated with the generation period of the measurement signal sound TSP signal, the S / N can be improved by canceling the noises by phase interference by synchronous addition.

  Therefore, in this embodiment, transfer characteristics obtained by synchronous addition are used in a frequency band where the phase interference between the signals does not occur due to a change in the transfer delay time that occurs according to the change in the microphone position or the frequency is less than the frequency. Is adopted. In other words, it is possible to obtain a measurement result in which a false dip characteristic due to phase interference between signals does not appear while suppressing the influence of noise.

  The transfer characteristic on the low frequency side subjected to the synchronous addition in the time domain is converted into the frequency domain by Fourier transform, and is synthesized in the frequency domain with the transfer characteristic on the high frequency side as described below.

As a reference for dividing the frequency band, one method may be determined in accordance with the limit of the reproduction capability of the speaker. Specifically, for the lowest resonance frequency f0 of the speaker system
Based on a × f0 (where a = 1 or 1 or more), the low-frequency-side transfer characteristic and the high-frequency-side transfer characteristic are synthesized.

FIG. 6 shows an example of a weighting function used for synthesizing the low frequency range transfer characteristic and the high frequency range transfer characteristic.
(1-cosα) / 2 and (1 + cosα) / 2
The synthesis can be realized by multiplying each transfer characteristic by the coefficient calculated in the range of 0 ≦ α ≦ π and adding. The combining method is not limited to this coefficient, and may be determined so that the sum is 1.

  As another criterion for dividing the frequency domain, it is possible to use the time accuracy of synchronous addition as a criterion.

  Now, for example, it is assumed that the timing of the sound collection signal can be adjusted with an accuracy of about 5 samples in a signal having a sampling frequency of 48 kHz. Since the frequency at which the number of samples is 15 ° in phase error is (360/15) × 5 = 120, the frequency is 120 samples per period. That is, at a frequency of 48000 ÷ 120 = 400 [Hz] or less, the error at the time of synchronous addition falls within 20 × log (cos (15 °)) ≈−0.3 [dB] or less. This makes it possible to determine the frequency domain division criterion.

  FIG. 7 shows an example in which a weighting coefficient is assigned on the frequency axis when combining the analysis results of the low sound range and the mid-high sound range.

  With the above-described sound correction method and sound correction device, the sound is received comfortably by the viewer. Furthermore, in the present embodiment, when setting the acoustic characteristics, the plurality of measurement points are spatial positions including front, rear, left, right, upper and lower in front of the acoustic device. For this reason, even if a listener who listens to the acoustic device moves to various positions, the sound can be heard comfortably.

  Usually, the above-described work of collecting sound at a plurality of measurement points is performed at the time of factory shipment of a product equipped with an acoustic device. However, the device of this embodiment may be incorporated in the product itself on which the acoustic device is mounted. Then, when the arrangement of the product is determined, the acoustic characteristic correction mode of the product may be set, and the correction parameter of the signal correction unit (equalizer) of the product may be obtained.

  FIG. 8 schematically shows an audio apparatus having a signal correction unit whose parameters are corrected by the audio correction apparatus, for example, a television receiver as the product.

  In FIG. 8, reference numeral 201 denotes a digital tuner device that includes a plurality of tuners, and can simultaneously receive a plurality of channels and demodulate a plurality of broadcast programs. The tuner device 201 receives a high frequency signal received by an antenna via an input terminal. The tuner device 201 may include, for example, a tuner (not shown) for receiving BS / CS digital broadcasts.

  A broadcast program signal received and demodulated by the tuner device 201 is input to the signal processing unit 202. In the signal processing unit 202, TSs (Transport Streams) of a plurality of channels (CH) are remultiplexed into one TS. The multiplexed TS includes a packet sequence of broadcast programs for each channel. Identification information for channel and packet identification is added to the packet of each channel. The multiplexed TS is input to the recording apparatus 201 under the control of the control unit 203. Of the TS input to the TS processing unit 122, a packet including control information is input to the control unit 203 and processed.

  Here, the recording apparatus 111 is shown as a block in a lump, and includes, for example, a plurality of hard disk drives (HDD), an optical disk recording / reproducing apparatus, a USB memory, and the like.

  A packet sent from the signal processing 202 to the control unit 203 is, for example, a table describing ECM (Entitlement Control Message) that is encryption information of a broadcast program and event information such as a program name, a performer, and a program start time. It includes information such as EIT (Event Information Table) and EPG (Electric Program Guide) information. The EPG information may include sponsor information.

  The video data included in the packet is encoded by, for example, MPEG (Moving Picture Expert Group) method, AVC (Advanced Video Coding) method or the like. The audio data in the audio packet is encoded by, for example, a PCM (Pulse Code Modulation) method, a Dolby method, an MPEG method, or the like.

  Therefore, the audio packet separated by the signal processing unit 202 is decoded by the audio decoder. The audio data decoded by the audio decoder 123 is subjected to synchronization processing, volume adjustment, etc., and supplied to the acoustic device 102. The operation of the acoustic device 102 is as described above.

  The video packet separated by the signal processing unit 202 is decoded by a video decoder. The video data decoded by the video decoder is subjected to synchronization processing, luminance adjustment, color adjustment, and the like. The adjusted video data is input to the display signal output device 205. The display signal output unit 205 is provided with a video output unit 205A, which sets the scale, resolution, number of lines, aspect ratio, etc. according to the display 205B and outputs the output video signal to the display 205B. .

  Since the above-described parameters are set in the signal correction unit 102A, the listener can enjoy comfortable sound.

  Although the above example has been described as a sound reproduction device for a television receiver, it is needless to say that this embodiment can be applied to a sound reproduction device in audio equipment specialized for audio.

  In the above-described embodiment, the name described in the block of the drawing is not limited to such a name. The signal generator may be referred to as a signal generator, a signal generation module, and means, and the signal correction unit may be referred to as a signal corrector, an equalizer, a signal correction module, and means. Furthermore, the characteristic analyzer may also be referred to as a characteristic analysis module, a characteristic analysis device, a characteristic analysis unit, a means, or the like.

  Although several embodiments of the present invention have been described, these embodiments have been presented by way of example and are not intended to limit the scope of the invention. These novel embodiments can be implemented in various other forms, and various omissions, replacements, and changes can be made without departing from the scope of the invention. These embodiments and modifications thereof are included in the scope and gist of the invention, and are included in the invention described in the claims and the equivalents thereof.

  DESCRIPTION OF SYMBOLS 101 ... Measurement signal generator, 102 ... Speaker (electroacoustic transducer), 103 ... Microphone (acoustoelectric transducer), 104 ... Characteristic analyzer, 105 ... Equalizer characteristic setting part, 200: Signal correction unit.

Claims (11)

In the sound correction device that measures the sound output from the sound device and corrects the sound characteristics based on the measurement result,
An acoustoelectric transducer that collects sound at a plurality of measurement points in front of the acoustic device and obtains a sound collection signal at the plurality of measurement points;
When analyzing the acoustic characteristics of the sound collected signals at the plurality of measurement points, each of the sound collected signals is divided into at least two frequency bands, and different analysis processing methods are used for the respective frequency bands. A characteristic analyzer that performs acoustic analysis and averages the results of the acoustic analysis of the plurality of measurement points to obtain one measurement result;
And a characteristic setting unit for obtaining a parameter for setting the characteristic of the acoustic device based on the one measurement result. The acoustic analysis in the characteristic analyzer is
Of the divided bands, the low frequency range is obtained by averaging the sound collection signal in time synchronization, and for the middle and high range, the sound collection signal is converted into the frequency domain and divided into amplitude and phase in the frequency domain. The sound correction apparatus according to claim 1, which performs an averaging process. The characteristic analyzer is used as a reference for dividing the frequency band.
3. The sound correction apparatus according to claim 1, wherein the lowest resonance frequency of the speaker set by the operator is used to set a low frequency range in which the reproduction sound pressure is insufficient due to a reproduction performance limit of the reproduction system. The characteristic analyzer is used as a reference for dividing the frequency band.
The sound correction device according to claim 1 or 2, wherein a low sound range is set up to a range in which the influence of the phase shift becomes slight according to the synchronization accuracy of the sound collection signal.   The sound correction device according to claim 1, wherein the sound device is built in a television receiver. In the acoustic correction method of measuring the sound output from the acoustic device and correcting the acoustic characteristics based on the measurement result,
Sound output is collected at a plurality of measurement points in front of the acoustic device, and sound collection signals at the plurality of measurement points are acquired,
When analyzing the acoustic characteristics of the sound collected signals at the plurality of measurement points, each of the sound collected signals is divided into at least two frequency bands, and different analysis processing methods are used for the respective frequency bands. Performing an acoustic analysis and averaging the results of the acoustic analysis of the plurality of measurement points to generate one measurement result;
An acoustic correction method for obtaining a parameter for setting a characteristic of the acoustic device based on the one measurement result.   In the acoustic analysis, in the low frequency range of the divided bands, the sound collection signal is averaged in time synchronization, and for the middle and high sound range, the sound collection signal is converted into the frequency domain, and the amplitude in the frequency domain. The acoustic correction method according to claim 6, wherein the averaging processing is performed by dividing the phase into phases.   The minimum resonance frequency of a speaker set by an operator is used as a reference for dividing the frequency band to set a low frequency range in which a reproduction sound pressure is insufficient due to a reproduction performance limit of a reproduction system. The acoustic correction method according to item.   The acoustic correction method according to claim 5 or 6, wherein, as a reference for dividing the frequency band, a low sound range is set up to a range where the influence of the phase shift becomes slight according to the synchronization accuracy of the acoustic sound collection signal. A sound reproducing device having a sound device,
The acoustic device includes a signal correction unit and a speaker,
The correction parameter setting of the signal correction unit is
Sound output is collected at a plurality of measurement points on the front, top, bottom, left and right of the acoustic device, and acoustic collection signals at the plurality of measurement points are acquired,
When analyzing the acoustic characteristics of the sound collected signals at the plurality of measurement points, each of the sound collected signals is divided into at least two frequency bands, and different analysis processing methods are used for the respective frequency bands. Performing an acoustic analysis and averaging the results of the acoustic analysis of the plurality of measurement points to generate one measurement result;
The sound reproducing device, wherein the sound is generated as a parameter in a direction for canceling distortion (amplitude distortion, phase distortion) included in the measurement result based on the one measurement result.   The sound reproducing device according to claim 10, wherein the sound device is provided in a television receiver.

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