JPH11103495A - Phase compensation system for audio reproducer - Google Patents

Abstract

PROBLEM TO BE SOLVED: To improve waveform reproducibility of a sound output from a speaker. SOLUTION: This device is equipped with an audio signal source 10 for generating a pilot signal (a), a phase characteristics correction device 12A for delaying/lagging this pilot signal (a) and supplying an audio signal (b) to a power amplifier 14, a probe 16 for detecting a voltage/current waveform of a drive signal (c) supplied to a speaker 18 from the power amplifier 14 in accordance with this audio signal (b), and a microphone 20 for converting a sound output (d) generated from the speaker 18 to an electric signal in accordance with this drive signal (c). In the phase characteristics correction device 12A, waveforms of the pilot signal (a), the drive signal (c) and the sound output (d) are compared with one another, and the phase/delay amount of one or more frequency element contained in the audio signal (b) is changed so that the waveform of the sound output (d) closely resembles the waveform of the pilot signal (a).

Description

DETAILED DESCRIPTION OF THE INVENTION

[0001]

[0001] 1. Field of the Invention [0002] The present invention relates to a system for improving audio reproduction sound in total from an electric system to an acoustic system. In particular, measure the playback waveform from a speaker system using a dynamic speaker unit.
The present invention relates to a system for evaluating and improving the waveform reproducibility (improvement of physical waveform fidelity / high fidelity; or improvement in auditory perception / good reproduction).

[0002]

2. Description of the Related Art At present, an audio speaker system mainly for music reproduction is mainly composed of a dynamic speaker (mechanical coupling to a voice coil by flowing an audio signal current through a voice coil arranged in a magnetic field). Speaker that generates an acoustic output corresponding to the audio signal current from the vibrating plate. In addition, although there are minor, there is an electrostatic speaker using Coulomb force or an ion speaker that modulates ion discharge with an electric signal.

[0003] Some dynamic speakers include a direct radiator type which directly generates an acoustic output from a diaphragm.
A horn type with a horn that performs acoustic impedance matching on a direct radiator-like sound-generating part, a ribbon type with a light and ultra-thin conductor diaphragm placed in a magnetic field, and a lightweight and ultra-thin plastic film diaphragm placed in a magnetic field There are a flat type in which a conductor for flowing an audio signal current is printed on a diaphragm, and a Heile type in which a conductor for flowing an audio signal current is printed on an accordion bellows-like film diaphragm and the bellows diaphragm is arranged in a magnetic field.

As described above, there are various types of dynamic speakers, but since the quality of reproduced sound, maximum output sound pressure, ease of manufacture, ease of handling, reliability, etc. are excellent at a practical level, Direct radiator type dynamic speakers are overwhelmingly the mainstream. Further, if the handling frequency is limited to the middle range or higher, a horn type having a unique characteristic in sound quality and having a high efficiency (sensitivity) and a high maximum output sound pressure (wide dynamic range) is also used relatively frequently.

In a speaker system using a direct radiator type or horn type dynamic speaker, the responsible reproduction frequency band is divided into two or more,
A multi-way speaker system using a dedicated speaker unit for each band is widely used for music reproduction and sound reinforcement (SR).

[0006] The multi-way loudspeaker system usually has a built-in network for receiving signals from the loudspeaker units and supplying signals only in the reproduction frequency band. The phase of an audio signal transmitted to each speaker unit through this network changes according to the frequency.

In a two-way multi-speaker system, for example, the low-range speaker unit has a large mass of the vibration system, so that the acoustic output generated therefrom increases in frequency from low to middle to high. As a result, the response (sound pressure output) decreases and the phase tends to be delayed accordingly (when the middle band is used as a reference for the phase, the relatively low band has a relatively advanced phase).

That is, generally speaking, the phase characteristic of a dynamic speaker system is in the middle and low frequency range (100 Hz to 100 Hz).
On the basis of (on the order of 200 Hz), as shown by the solid line in FIG. 8, the phase tends to advance at a lower frequency side and to be delayed at a higher frequency side. (Because an actual dynamic speaker performs divided vibration, the phase characteristics on the middle and high frequency sides are not as smooth as shown in FIG. 8. The solid line in FIG. 8 indicates the approximate phase characteristic of the speaker sound output ignoring the divided vibration. Trends,
It is illustrated in a simplified manner. When the signal for driving the speaker is a sine wave of a single frequency, a smooth phase change according to the frequency does not substantially affect the sound output waveform. However, when the signal for driving the speaker is a mixed sine wave of two frequencies, the phase of the second frequency component with respect to the first frequency component fluctuates due to a change in the frequency phase of the speaker system including the network. The sound output waveform of the mixed sine wave may greatly change due to this phase change. An actual music signal is a mixed sine wave of an infinite number of complex frequencies, and a change in an acoustic output waveform (a kind of waveform distortion) due to this phase change may occur more complicatedly.

The first frequency component (fundamental wave) has a second frequency component.
If the frequency components (harmonics) of the first frequency component maintain the harmonic relationship (harmonics), the beat components of the first frequency component and the second frequency component become the harmonics of the fundamental wave, and the phase of the harmonic wave with respect to the fundamental wave becomes Even if the waveform changes to cause a change in the waveform of the reproduced sound, it is difficult for humans to perceive the change as a sound. For example, 1 kHz base signal (fundamental wave)
When an additional signal (second harmonic) of Hz is added, the beat component becomes 1 kHz of the difference component and 3 kHz of the sum component.
The difference component (1 kHz) is absorbed by the fundamental wave, and the sum component (3 kHz)
Hz) is the harmonic (third harmonic) of the fundamental wave.
Harmonic components to fundamental wave components (2 kHz, 3 kHz)
Is small (several percent or less), even if the combined waveform of the fundamental wave and the harmonic wave changes due to the phase change, the change is hardly perceived by the listener.

On the other hand, if the second frequency component (harmonic) does not maintain a harmonic relationship (harmonics) with respect to the first frequency component (fundamental wave), the first frequency component and the second frequency The beat component of the component does not become the harmonic of the fundamental wave. In this case, if the phase of the harmonic with respect to the fundamental wave changes relatively and the waveform of the reproduced sound changes, a high-quality audio system with low distortion, wide range and excellent sound field reproducibility (acoustic phase characteristic) is used. For example, a listener with good hearing may feel the waveform change.

For example, when an additional signal of 1.1 kHz is added to a base signal (fundamental wave) of 1 kHz, the beat component becomes 100 Hz of the difference component and 2.1 kHz of the sum component. The difference component (100 Hz) undulates the envelope of the fundamental wave (amplitude modulation), and the sum component (2.1 kHz) that does not become the harmonic of the fundamental wave does not blend into the fundamental wave in terms of hearing. Even if the level of the non-harmonic harmonic component (2.1 kHz) with respect to the fundamental wave component is small (several% to 10% or less), if the combined waveform of the fundamental wave and the non-harmonic harmonic changes due to the phase change, The change may be perceived as auditory (this auditory perception cannot be quantified at this time, and there is a large individual difference between listeners and testers).

In some multi-way loudspeaker systems, in order to ensure faithful waveform reproducibility, a network, a diaphragm shape, a diaphragm material, and the like are used so as to obtain as flat a phase characteristic as possible over the entire reproduction frequency band. Some voice coil positions or diaphragm positions take into account design considerations. Even with such considerations, in an actual speaker system, the mechanical mounting positions of the low-frequency and high-frequency speaker units are fixed, so that the low-frequency unit and the low-frequency unit are phase-aligned at a certain frequency point. Even if the mounting position of the high-frequency unit is adjusted, the phase often shifts between the two units if the reproduction frequency changes.

As described above, in an actual speaker system using a dynamic speaker, the phase characteristic of the sound output is not flat, but generally has a tendency of leading the phase from the middle band to the low band (speaker). (A schematic tendency of the phase characteristic in the middle to low frequency band excluding the divided vibration frequency region; actual phase characteristics are different for each speaker system.)

By the way, as a power amplifier (power amplifier) for driving the above speaker system at present,
Extremely wide reproduction frequency characteristics with low distortion (wide range)
A transistor type OTL / OCL (output transformerless / output capacitorless) amplifier is often used. This type of transistor amplifier is often composed of a negative feedback amplifier that has a sufficient amount of negative feedback in the DC region in order to ensure DC stability at the speaker output terminal. In this case, this type of transistor amplifier has almost flat phase characteristics in the audio region (20 Hz to 20 kHz) except for an ultra-low frequency region close to DC, as exemplified by a broken line in FIG.

When a dynamic speaker having a phase characteristic tendency as exemplified by the solid line in FIG. 8 is driven by such a transistor amplifier, even if the voltage phase characteristic at the speaker input terminal is flat, the phase characteristic of the sound output is equal. It does not become flat.

[0016]

When the sound output from the dynamic speaker driven by the above transistor amplifier is captured by a measuring microphone and the waveform is observed, the amplifier output (= speaker input) has low distortion (0.1%). Less than)
Even so, a significant waveform change may occur.

For example, when a two-frequency mixed signal (original waveform) as shown in FIG. 10A is input to a transistor OTL / OCL amplifier and its output (speaker input voltage) is observed,
As shown in FIG. 10B, a waveform substantially the same as the original waveform is observed. However, when the sound output from the speaker at that time is observed, a waveform that is significantly deformed from the original waveform may be observed as shown in FIG. (FIG. 10 (d)
The waveform itself has no significance because the waveform itself changes depending on the speaker used, the acoustic characteristics of the measurement location, the measurement frequency, its signal level, and the like. In short, even if the electrical input waveform to the speaker is faithful to the original waveform of FIG. 10A as shown in FIG. 10B, the sound output waveform may be significantly different from the original waveform. Should be recognized. Now, as the two-frequency mixed signal, the audio band 2 in which a beat (sum component and difference component) occurs in the audible frequency band.
Let us assume that a frequency signal is used and input to a dynamic speaker.

If the high-frequency component (sum component) of the beat and its harmonics are in the divided vibration region (or non-piston vibration region) of the dynamic speaker, the acoustic output of the high-frequency beat component and its harmonics is divided. Due to vibration (including many fine resonance phenomena), the phase changes complicatedly (shifts by 180 ° before and after the resonance point), and the original waveform of the two-frequency mixed signal is no longer reproduced (180 ° ± 90 ° phase shift). A cancellation component is generated for the original waveform, and the original waveform is deformed by the cancellation.) This is considered to be one of the causes of lowering the electrical / sound conversion performance (fidelity) of the dynamic speakers, and is recognized as individuality (habit) of each dynamic speaker.

Even when the high-frequency beat component is in the piston vibration range of the dynamic speaker, if the phase characteristic of the speaker is not flat, the difference between the phase shift of the fundamental wave signal and the phase shift of the additional signal and the beat component. Varies depending on the frequency, and the original waveform of the two-frequency mixed signal is not faithfully reproduced.

In a conventional dynamic dynamic speaker (especially a direct radiation type), divided vibration is considered to be a matter of course. However, it has not been possible to sufficiently prevent deterioration of the original waveform reproducibility of the dynamic speaker.

A first object of the present invention is to provide a phase compensation system and a phase compensation method for an audio apparatus in which the waveform reproducibility of an acoustic output is improved as a total system from an amplifier to a speaker by compensating the above-mentioned phase shift on the amplifier side. It is to provide.

A second object of the present invention is to provide a system and method for analyzing a phase characteristic of a reproduced waveform of an audio output as a total system of an amplifier to a dynamic speaker.

[0023]

In order to achieve the first object, a phase compensation system of a sound reproducing apparatus according to the present invention includes a predetermined speaker system and a predetermined power amplifier for driving the speaker system. An audio signal source (10) for generating a pilot signal (a) including one or more frequency components in a sound reproducing apparatus; a pilot signal (a) from the audio signal source (10) is delayed or its phase is changed. Then, the delayed or phase-changed audio signal (b) is transmitted to the power amplifier (1).
4) a phase characteristic correction means (12A) for supplying the power signal to the audio signal (b);
To detect the content (waveform) of the speaker drive signal (c) supplied from the controller to the speaker system (18).
A current detection means (16); and a sound / electric conversion means (20) for converting an acoustic output (d) generated from the speaker system (18) into an electric signal in response to the speaker drive signal (c). Have.

In the phase characteristic correction means (12A), at least two waveforms of the pilot signal (a), the speaker drive signal (c) and the sound output (d) are compared, and the sound output (d) is compared. The phase and / or the amount of delay of one or more frequency components included in the audio signal (b) are changed so that the waveform of the audio signal (b) approximates the waveform of the pilot signal (b).

In order to achieve the first object, a phase compensation method according to the present invention provides a sound reproducing apparatus including a predetermined speaker system and a predetermined power amplifier for driving the speaker system. Step (ST1) of generating a pilot signal (A) including
0 to ST12); a second step of detecting the content (waveform) of the speaker drive signal (c) supplied from the power amplifier (14) to the speaker system (18) in response to the pilot signal (a). (ST16); a third step (ST18) of converting an acoustic output (d) generated from the speaker system (18) into an electric signal in response to the speaker drive signal (c); ), A fourth step (ST20) of comparing the phases of the waveforms of the speaker drive signal (c) and the sound output (d); and the waveform of the sound output (d) approximates the waveform of the pilot signal (b). As described above, the pilot signal (a) is delayed or its phase is changed, so that the delayed or phase-changed audio signal (b)
To the power amplifier (14).
T22A).

In order to achieve the second object, a phase characteristic analysis system according to the present invention comprises a sound reproducing apparatus including a predetermined speaker system and a predetermined power amplifier for driving the speaker system. An audio signal source (10) for generating a pilot signal (a) including: a speaker system (18) from the power amplifier (14) corresponding to the pilot signal (a).
Voltage / current detection means (16) for detecting the content (waveform) of a speaker drive signal (c) supplied to the speaker; and sound generated from the speaker system (18) in response to the speaker drive signal (c) Sound / electric conversion means (20) for converting the output (d) into an electric signal; and delaying or changing the phase of the pilot signal (a) from the audio signal source (10) by delaying or changing its phase. The audio signal (b) to the power amplifier (14)
And a phase observation / analysis means (12) for supplying the power to the apparatus.

In the phase observation and analysis means (12), at least two waveforms of the pilot signal (a), the speaker drive signal (c) and the sound output (d) are compared, and the sound output (d) is compared. Is observed and evaluated by focusing on the phase and / or delay amount of one or more frequency components included in these waveforms.

In order to achieve the second object, a phase characteristic analysis method according to the present invention provides a sound reproduction system including a predetermined speaker system and a predetermined power amplifier for driving the speaker system. First Step (ST) of Generating Pilot Signal (A) Containing Component
A second step of detecting the content (waveform) of the speaker drive signal (c) supplied from the power amplifier (14) to the speaker system (18) in response to the pilot signal (a). (ST16); a third step (ST18) of converting an acoustic output (d) generated from the speaker system (18) into an electric signal in response to the speaker drive signal (c); ), A fourth step (ST20) of comparing the phases of the loudspeaker drive signal (c) and the sound output (d); and at least one level based on the result of the phase comparison in the fourth step (ST20). Fifth step (S) of measuring and recording the phase difference between the speaker drive signal (c) and the sound output (d) at a point
T22).

In the above description, "the waveforms of at least two of the pilot signal (a), the speaker drive signal (c) and the sound output (d) are compared" are as follows: In addition to the three-point comparison of (c) and (d), comparison of (a) and (d), (a) and (c)
, And (c) and (d).

Here, when the correspondence between the pilot signal (a) and the speaker drive signal (c) is known in advance (because it has been measured, etc.), the comparison between (c) and (d) gives
The similarity between the waveform of the sound output (d) and the waveform of the pilot signal (a) can be checked. If the correspondence between the speaker drive signal (c) and the waveform of the sound output (d) is known in advance (because of measurement, etc.), the sound output (d) is compared by comparing (a) and (c). And the waveform of the pilot signal (a) can be checked for similarity.

[0031]

DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENTS Hereinafter, a phase compensation system for an audio reproducing apparatus according to an embodiment of the present invention will be described with reference to the drawings. In order to avoid redundant description, common reference numerals are used for functionally common parts in a plurality of drawings.

FIG. 1 is a block diagram for explaining a system for analyzing a phase characteristic of an amplifier under a speaker load in a phase compensation system according to an embodiment of the present invention.

The audio signal source 10 generates an actual sample sound source (music, natural sound, or the like) or a pilot signal E10 whose characteristics are known. As the pilot signal E10 whose characteristics are known, a continuous sine wave of a single frequency, a single sine wave of a single cycle (see FIG. 11A; or a single raised cosine wave (not shown)), A composite sine wave obtained by combining sine waves (FIG. 10)
(See FIG. 13A), and a mixed single-shot sine wave in which a sine wave of another frequency is superimposed on a single-shot sine wave (see FIG. 13A) can be used. The pilot signal E10 can be set to various frequencies, various signal levels, and various waveforms.

The pilot signal E10 generated by the audio signal source 10 is used as an input signal Ei in the phase observation device 1
2 is supplied. The phase observation device 12 has a function of delaying the pilot signal E10 or changing its phase. The signal E10 or Ei is referred to as a signal (a). The phase observation device 12 outputs a signal E12 corresponding to the pilot signal E10 (equivalent except for the phase and / or signal level) as the signal Eo. This signal E12
Or, let Eo be a signal (b).

The signal E12 is input to the power amplifier 14 to be measured. There are various types of amplifiers 14. Here, a transistor OTL having neither an audio transformer (input transformer, interstage transformer, output transformer) nor an output capacitor in a signal transmission path.
/ OCL amplifier (extremely low distortion, both frequency characteristics and phase characteristics are flat in the audio band), and a tube amplifier having an audio transformer (at least an output transformer) in a signal transmission path having characteristics symmetric to this transistor amplifier ( (It has a moderate amount of distortion that is not harmful to the sense of hearing, and neither the frequency characteristic nor the phase characteristic is flat in the audio band.)

The amplifier 14 appropriately amplifies the input signal E12, and has a certain power (for example, 1 W to 100 W).
(W) is generated. This signal E14 is supplied to the load 18. As the load 18, a single-way full-range cone speaker, a two-way or more multi-way speaker including an LC network (only a cone speaker or a mixture of a cone speaker and a horn speaker) or the like is used.

The current and / or voltage of the speaker drive signal E14 supplied from the amplifier 14 to the load 18 is
It is detected by the current / voltage detection probe 16 and supplied to the phase observation device 12 as a detection signal E16. If the probe 16 is a current probe, the drive current E14 of the load (speaker) 18 becomes the detection signal E16, and if the probe 16 is a voltage probe, the drive voltage E1 of the load (speaker) 18
4 becomes the detection signal E16. When the probe 16 simultaneously detects the effective value of the drive current to the load (speaker) 18 and the effective value of the drive voltage in real time and multiplies them, the drive power E14 of the load (speaker) 18 is detected by a detection signal E16.
Can be detected by Such a signal E14 or E1
6 is a signal (c).

The load (speaker) 18 is driven by the speaker drive signal E14 from the amplifier 14, and generates an acoustic output A18 corresponding to the signal (c).
This sound output A18 is captured by a measurement microphone (a device having flat frequency characteristics and phase characteristics) 20, and is converted into a sound output capture signal E20. This signal E20 is referred to as a signal (d).

When the speaker 18 and the microphone 20 are installed in an anechoic room, the speaker 18 and the microphone 20
Is usually set at 50 cm to 1 m. On the other hand, when the speaker 18 and the microphone 20 are installed in a normal room with sound, the distance between the speaker 18 and the microphone 20 is set as short as possible (30c) in order to reduce the influence of the room.
m or less).

The sound output A18 usually has considerable waveform distortion (amplitude distortion and phase shift) with respect to the speaker drive signal E14 (c), but the sound output capture signal E20 (d) has this waveform distortion. Is caught. The signal (c) and the signal (d) are fed back to the phase observation device 12.

In the phase observation device 12, the pilot signal E10 (a) and the speaker drive signal E14
(C) and the waveform of the sound output capture signal E20 (d) are compared. These signal waveforms are usually very different. However, in the phase observation device 12, the amplifier 14
Is corrected by correcting the waveform (phase, amplitude, etc. of each frequency component) of the signal E12 (b) supplied to the
The approximation between the waveform of 20 (d) and the waveform of pilot signal E10 (a) can be improved.

The evaluation of the waveform approximation can be performed by obtaining a sensory evaluation based on human perception and / or obtaining a matching coefficient. Here, “determining a matching coefficient” means a signal (a) input to the phase observation device 12.
(D) means that the signal waveforms of two or more signals are numerically converted at a number of sampling points, and the frequency distribution / amplitude distribution pattern of the difference between the waveform numerical values is analyzed by computer. In this case, for example, the analyst analyzes the time change of each waveform, and pays attention to the time difference and the level difference between the two waveforms at the extreme point where the differential value is zero. (It can be said that this is similar to the work of an experimenter examining the experimental results in a physical experiment or a chemical experiment.) An algorithm for the above-mentioned "finding a matching coefficient" was discovered by accumulating a large number of results of analysis and analysis. If this is the case, the act of "determining the matching coefficient" can be completely computerized, but no practical algorithm has been found so far.

Once the above "matching coefficient" has been obtained by mobilizing the human sensory ability, this "matching coefficient" can be stored as digitized data,
The data can also be used for computer processing.

That is, in the phase observation device 12,
Sound output capture signal E20 (d) and pilot signal E
Observation / evaluation (or determination of a matching coefficient) focusing on the phase and / or delay amount of one or more frequency components included in the waveform of 10 (a) is performed.

The pilot signal E10 (A) is 120H
Various combinations of multiple frequency components (two or more), such as z and 130 Hz; 800 Hz and 8 kHz; 1 kHz and 1.1 kHz; 3 kHz and 8 kHz.
Can be included.

When the plurality of frequency components constituting pilot signal E10 (a) are, for example, 1 kHz and 1.1 kHz, the fundamental frequency component (1 kHz) and the additional frequency component (1.1 kHz) are different from the fundamental frequency component (1 kHz). Harmonics (2 kHz, 3 kHz, 4 kHz, ...)
(100 Hz and 2.1 kHz) having no pulse. This beat component (100 Hz and 2.1 kHz
z) is input to the speaker 18 together with the fundamental frequency component (1 kHz) and the additional frequency component (1.1 kHz).

This beat component (100 Hz and 2.1 kHz)
z) does not fall into the harmonics of the fundamental wave (2 kHz, 3 kHz, 4 kHz,...) even if the level is relatively small (for example, 10% or less) with respect to the fundamental frequency component (1 kHz). Easy to be sensed. And the beat component (2.1 kHz) that is easy to hear like this
The waveform of the fundamental frequency component (1 kHz) on which is superimposed is greatly deformed due to the low waveform conversion performance of the speaker 18 (not a steady waveform deformation but another 100% deformation).
Is modulated by the Hz beat component and fluctuates with time).
It may appear as a deterioration in auditory sensation (whether or not this deterioration can be heard depends on the individual as well as the taste sensing ability).

Such a low-frequency side beat component (100H
z) fundamental wave + high-frequency side beat component (2.1k
The phase observation device 12 can observe and evaluate how the waveform (Hz) changes between the input and output of the amplifier 14 and how the waveform changes between the electric input and the sound output of the speaker 18.

In the observation by the phase observation device 12,
A transformer-coupled amplifier including an audio transformer in a signal path (such as a transformer drive single amplifier using a 300B output tube) is prepared as a reference for the power amplifier 14. Also, an OTL amplifier (such as a direct-coupled transistor DC amplifier) that does not include an audio transformer in the signal path is prepared as the amplifier 14 to be analyzed and evaluated.

In the phase observation device 12 of FIG. 1, the dynamic speaker 18 has a phase characteristic tendency as shown by a solid line in FIG. 8 or FIG. 9, and the OTL amplifier has a phase characteristic tendency as shown by a broken line in FIG. It is assumed that the amplifier is observed to have a phase characteristic tendency as shown by the broken line in FIG.
(The phase characteristics of the loudspeaker 18 vary greatly depending on the dynamic characteristics of each loudspeaker and the measurement frequency range. However, in FIGS. 8 and 9, the loudspeaker phase characteristics are simplified and deformed for easy understanding. Observing the phase characteristics in FIG. 9, it is found that a certain frequency (in the illustrated example, around 100 Hz to 200 Hz) is centered.
The phase characteristic of the transformer-coupled amplifier and the phase characteristic of the loudspeaker are generally observed to be in a complementary (symmetrical) relationship (whether or not such a complementary relationship is recognized for all loudspeakers is still unknown). Not confirmed).

When a loudspeaker having a phase characteristic as shown by a solid line in FIG. 9 is driven by a transformer-coupled amplifier having a phase characteristic as shown by a broken line in FIG. 9, the system total from the pilot signal (A) to the sound output capture signal (D) is obtained. Thus, a substantially flat phase characteristic is realized. That is, the waveform distortion (including the time lag) caused by the phase characteristic of the speaker 18 is compensated (neutralized) by the phase characteristic of the transformer coupling amplifier 14, and the sound which is close to the original waveform (the waveform of the pilot signal) in total. An output A18 will be obtained. An acoustic output A18 close to the original waveform reaches the listener's ear.

The phase advance characteristics of the loudspeakers shown in FIGS. 8 and 9 are usually in the middle to middle frequency range including the frequency spectrum of male vocals or contrabass or Gran Cassa (large drum).
It occurs remarkably in the low frequency range (the area that affects the sense commonly called "feel the air"). For this reason, if the dynamic speaker 18 having a tendency to lead the low-frequency phase is driven by the transistor OTL / OCL amplifier 14, the reproduced sound quality of the male vocal or contrabass or Gran Cassa may be dissatisfied.
When 8 is driven by the transformer coupled amplifier 14, this dissatisfaction can be considerably eliminated. (It seems awful, but there is a great individual difference in how we can perceive the dissatisfaction of the reproduced sound quality. Even if statistics are collected for ordinary people who are not interested in advanced music playback, this dissatisfaction
No meaningful data is available for resolution. For those who can't tell the difference between the violin of Stradivari and the violin of Garnelli, or even the work of Stradivari, who can't tell the difference in the tone of the violin due to the different production dates.
It becomes the same meaningless statistical data as the statistics of the differences in tone of various violins. FIG. 10 is a diagram for explaining the operation of compensating the above-mentioned phase characteristics. In the case where a two-frequency mixed sine signal (for example, 80 Hz and 200 Hz) is used, the output waveform of the transistor amplifier (phase flat) and the tube amplifier ( And a sound output waveform of a dynamic speaker driven by a transistor amplifier and a sound output waveform of a dynamic speaker driven by a tube amplifier when this two-frequency mixed signal is used. Is shown.

FIG. 14 shows a configuration used for obtaining the various waveforms shown in FIG. 14, the audio signal source 10 shown in FIG. 1 includes oscillators 101 and 102 and an adder 103, and the power amplifier 14 to be measured in FIG. 1 is a transistor OTL / OCL amplifier 14T and a tube with an interstage transformer and an output transformer. It is composed of an amplifier 14V.

The output of either the amplifier 14T or 14V is selected by the switch SW3 and supplied to the dynamic speaker 18. The speaker 18 driven by the transistor amplifier 14T or the tube amplifier 14V
An audio output A18 having a different waveform is generated by the amplifier. The sound output A18 is captured by the microphone 20, is appropriately amplified by the microphone amplifier MA1, and is fed back as a signal (d) to the phase observation device 12 in FIG.

That is, when the original waveform of the signal E12 (b) input to the amplifier 14 of FIG. 1 is as shown in FIG. 10 (a), the output voltage waveform of the transistor amplifier 14T having low distortion and flat phase characteristics is obtained. Is as shown in FIG. The output E14 (C) of the transistor amplifier 14T is
There is no phase shift (delay) or amplitude distortion with respect to the original waveform, and is almost the same as the input signal E12 (b) ((a) in FIG. 10).
(Note the lower waveform peaks at times t10 and t12 in (b)).

On the other hand, the output voltage waveform of the transformer-coupled amplifier 14V has a
The result is as shown in FIG. Transformer coupling amplifier 14V
The output E14 (c) has both a phase shift and an amplitude distortion with respect to the original waveform, and particularly the extreme point of the fundamental wave component (lower waveform peak).
Is temporally advanced from the corresponding position of the original waveform (note the lower waveform peaks at times t10 and t12 in FIGS. 10A and 10C).

As described above, when the speaker input waveforms are compared, the transistor amplifier 1 is more than the transformer coupled amplifier 14V.
4T seems to be much better (see (b) and (c) of FIG. 10).

However, when comparing the acoustic output A18 of the dynamic speaker 18 driven by the amplifier, it can be seen that the case where the transformer coupled amplifier 14V is used rather than the case where the transistor amplifier 14T is used (FIG. 10 (d)). ((E) of FIG. 10) is the amplifier input signal E12.
An acoustic output waveform closer to the original waveform (b) (FIG. 10A) is obtained.

The reason why the transformer-coupled amplifier 14V is closer to the original waveform is mainly due to the transformer-coupled amplifier 1V.
This is probably due to the lower phase lead of 4V (relative to transistor amplifier 14T). Because
When the waveform of the signal (b) input to the transistor amplifier 14 is distorted by advancing the phase of the low-frequency component of the pilot signal (a) by the phase observation device 12, the sound that is distorted as shown in FIG. This is because it is observed that the output waveform approaches the acoustic output waveform using the transformer coupled amplifier of FIG. By appropriately adjusting not only the phase but also the amplitude, the output waveform shown in FIG.

FIG. 2 is a flowchart illustrating a method of observing the phase characteristics in the phase characteristic analysis system of FIG. This method can be realized by connecting a personal computer (not shown) to the system shown in FIG. 1 via an appropriate interface and performing software processing on the personal computer.

In FIG. 2, a pilot signal for measurement (A) is supplied by a CPU of a personal computer (not shown).
Signal type and its frequency, for example, 120H
A continuous oscillation sine wave of z and 130 Hz is preset (step ST10). At this time, the relative signal level of the 130 Hz sine wave with respect to the 120 Hz sine wave is also preset to a predetermined value. Then, the audio signal source 1 of FIG.
0 generates a combined signal E10 of a 120 Hz sine wave and a 130 Hz sine wave = pilot signal (a) (step ST12).

If the frequency preset in step ST10 is only a single frequency, step ST12
Is a simple continuous sine wave, a single sine wave of a single frequency, a tone burst wave of a single frequency, or the like.

If the frequency preset in step ST10 is a plurality of frequencies, the measurement signal = pilot signal (a) generated in step ST12 is modulated by a continuous sine wave including a plurality of frequency components and a certain frequency component. In addition, it becomes a single-shot sine wave of another frequency or a tone burst wave of a plurality of frequencies.

The generated pilot signal (a) is input to the amplifier under test 14 (tube amplifier using a transformer or transistor OTL / OCL amplifier) (step S).
T14). At this time, the level of the pilot signal (A) input to the amplifier under test 14 is adjusted so that the amplifier output becomes a predetermined value (for example, 1 Watt).

A predetermined amplifier output (1 watt) = speaker drive signal (c) is detected by the current / voltage detection probe 16, and the detected signal E 16 is
Supplied to The phase observation device 12 measures the waveform of the load current, the load voltage, or the load power from the signal E16 (step ST16).

Note that the effective value is calculated for one cycle of the measured current waveform, the effective value is calculated for one cycle of the voltage waveform measured at the same time, and the effective values of the two are multiplied. Then, the measured power can be obtained.

The amplifier load (dynamic speaker) 18 receiving the speaker drive signal (c) produces an acoustic output (d) corresponding to the speaker drive signal (c). When the sound output waveform is captured by the microphone 20, a microphone output signal E20 indicating the sound output (d) waveform is obtained.
Is obtained. The microphone output signal E20 = sound output (d) is supplied to the phase observation device 12. The phase observation device 12 measures the waveform of the sound output (d) from the speaker 18 based on the signal E20 (step ST16).

Subsequently, the phase observation device 12 outputs the load current,
The phases of the waveform (c) of the load voltage or load power and the waveform (d) of the sound output are compared (step ST20). This comparison is performed at one or more points of the waveform to be measured (for example, FIG.
0 at t10, t20; t10 *, t20 *).

The comparison result is stored in the memory of the personal computer using the used pilot signal (a) (120 Hz and 130 Hz) and the measurement level (1 watt) as parameters (step ST22).

When it is desired to acquire more data at another frequency (Yes in step ST24), the pilot signal (A)
Is changed (step ST26), and step S26 is performed.
The processing from T14 to ST22 is repeated.

When the necessary data has been obtained (No in step ST24), the phase observation processing of FIG. 2 is completed.

As for a plurality of frequency components of the pilot signal (a) preset in step ST10 of FIG. 2, at least one of the plurality of frequency components is changed and steps ST14 to ST26 are repeated. good.

The plurality of frequency components of the pilot signal (a) preset in step ST10 of FIG. 2 are within the audio band (about 20 Hz to 20 kHz).
In the above, a fundamental frequency component (for example, 1 kHz) and an additional frequency component (for example, 1.1 kHz) that creates a beat having no harmonic relationship with the fundamental frequency component may be included.

FIG. 11 shows output waveforms of a transistor amplifier (phase flat) and a tube amplifier (phase advance) when a single-shot sine signal (about 100 to 200 Hz) is generated from the signal source 10 of FIG. FIG. 7 is a diagram showing, in a deformed manner, an acoustic output waveform of a dynamic speaker driven by a transistor amplifier and an acoustic output waveform of a dynamic speaker driven by a tube amplifier when this signal is used. Even when a single sine wave is used, it is observed that the phase of the speaker output waveform of the transformer coupled tube amplifier advances in comparison with the transistor amplifier (note points P30 and P40).

FIG. 12 shows an output waveform of a transistor amplifier (phase flat) and a tube amplifier (phase advance) when a single frequency sine wave (about 100 to 200 Hz) is continuously generated from the signal source 10 of FIG. FIG. 7 is a diagram showing, in a deformed manner, an output waveform of a dynamic speaker driven by a transistor amplifier and an acoustic output waveform of a dynamic speaker driven by a tube amplifier when this signal is used. Even when a single-frequency continuous sine wave is used, it is observed that the phase of the speaker output waveform of the transformer-coupled tube amplifier advances in comparison with the transistor amplifier (note points P50 and P60).

FIG. 13 shows a case where a two-frequency mixed single-shot sine signal (a single-shot sine of about 100 Hz and a tone burst wave of about 1 kHz superposed at a low level) is used.
The output waveform of the transistor amplifier (phase flat) and the output waveform of the tube amplifier (phase advance); and the acoustic output waveform of the dynamic speaker driven by the transistor amplifier and the tube amplifier when this signal is used. It is a figure which shows the sound output waveform of a dynamic speaker deformed. Even when a single-frequency two-frequency sine signal is used, it is observed that the phase of the speaker output waveform of the transformer coupled tube amplifier advances in comparison with the transistor amplifier (note points P70 and P80).

FIG. 3 is a block diagram for explaining a system for improving the phase characteristics of an amplifier under speaker load in the phase compensation system according to one embodiment of the present invention. The configuration in FIG. 3 is substantially the same except that the phase observation device 12 in FIG. 1 is changed to a phase characteristic correction device 12A.

In the system shown in FIG. 3, the phase characteristic correction device 12A delays or changes the phase of the pilot signal (a) from the audio signal source 10, and converts the delayed or phase-changed audio signal (b). It has a function of supplying power to the power amplifier 14. The detailed internal configuration of the phase characteristic correction device 12A will be described later with reference to FIG.

In short, the system shown in FIG. 3 uses the amplifier 14 based on the observation result of the phase observation device 12 shown in FIG.
Irrespective of the characteristics (particularly the phase characteristics), the sound output A18 waveform (d) of the speaker 18 generates a signal E12A (b) that approaches the pilot signal E10 waveform (a).

FIG. 4 is a flowchart for explaining a phase characteristic correcting method in the phase characteristic improving system of FIG. The processing steps in FIG. 4 are substantially the same except for step ST22 in FIG.

That is, from the phase comparison result (comparison between the pilot signal waveform or the speaker drive waveform and the speaker output waveform) in step ST20, at least a part of the waveform (in the example of FIG. 10, the negative peak of the fundamental wave component) point)
, The signal is delayed such that the phase difference between the compared waveforms falls within a predetermined range (eg, about 0 ° ± 15 °) (step ST22A in FIG. 4).

In order to make the waveform of FIG. 10 (b) into a phase-leading waveform as shown in FIG. 10 (c), instead of advancing the waveform of interest (for example, a fundamental wave component), the waveform of interest is used as a reference for the phase. Then, the phases of other signal components (additional signal components) may be delayed.

The process of step ST22A in FIG. 4 is appropriately performed at a plurality of frequencies (step ST24 YES), and the phase adjustment is performed at several frequency points in step ST22A.
Made in Although the contents of this phase adjustment change on a case-by-case basis, when the signal (a) obtained as a result of this phase adjustment is given to the transistor amplifier 14T, FIG.
The sound output waveform close to the original waveform (a) of FIG.
0 (e)).

FIG. 5 is a block diagram illustrating the internal configuration of the phase characteristic correction device 12A in the phase characteristic improvement system of FIG.

First oscillator 101 and second oscillator 1
02 generates a continuous sine wave, a single sine wave, a tone burst wave, a triangular wave, a rectangular wave, etc., of various frequencies and various signal levels. The output E101 of the oscillator 101 and the output E102 of the oscillator 102 are analog-combined in an adder (mixing amplifier / waveform combiner) 103 to become a pilot signal (a).

One of a signal obtained by buffering a high-quality audio source such as a compact disk (CD) in a buffer BA1 and a pilot signal (a) from an adder 103 is selected by a switch SW1. The selected signal (a) is used as an input signal Ei as a delay circuit (analog processing or digital processing) 12 with a variable delay amount.
Input to 0. The delay circuit 120 converts the input signal Ei into
An audio source output or a phase shift correction output E is appropriately delayed (phase shift or phase shift) for each frequency component.
o (b) is generated.

The amount of delay (the amount of phase shift) in the delay circuit 120
Is determined by the following configuration. That is, the adder 1
03 is supplied to the switch SW2 via the buffer BA2. The drive signal (c) of the speaker 18 detected by the probe 16 of FIG. 1 or FIG. 3 is adjusted to an appropriate level by the attenuator 126, and the switch SW2 is switched via the buffer BA3.
Supplied to The waveform (analog) of the signal (a) or the signal (c) selected by the switch SW2 is, for example, an 8-bit resolution analog / digital converter (AD).
C) The signal is converted into an 8-bit digital signal E124 by 124.

On the other hand, the microphone output signal (d) indicating the sound output of the speaker 18 is appropriately amplified by the microphone amplifier MA1, adjusted to an appropriate level by the attenuator 127, and supplied to the buffer BA4 via the buffer BA4, for example, with an 8-bit resolution analog signal. / Digital converter (ADC) 128. The ADC 128 converts the microphone output signal (d) into an 8-bit digital signal E128.

The digital signals E124 and E128 from the ADCs 124 and 128 are supplied to a phase comparison / phase coincidence detection unit 125. The detection unit 125 includes a digital signal input / output interface, a memory, a DSP (digital signal processor), a CPU, and the like (CPU
If the processing capability of the DSP is high, the DSP function can be performed by software processing of the CPU.) This CPU is programmed to support the above-described operation of obtaining the “matching coefficient”.

The CPU of the detection unit 125 compares the probe signal waveform (c) and the speaker sound output waveform (d) corresponding to the various pilot signal waveforms (a) with various frequencies, and obtains the waveform (d). Is closer to the waveform (A) so that the delay parameter E12
Set 5. This parameter setting operation corresponds to the operation of obtaining the “matching coefficient” (mainly, the operation of obtaining the phase match of the comparison target waveform).

The parameter (here, 8-bit data) E125 is sent from the detection unit 125 to the delay amount switching unit 129.
Is set to Then, the delay amount switching unit 129 latches each bit data of the setting parameter E125, and supplies the latched 8-bit data E129 to the delay circuit 120. Thereby, the delay circuit 120 sets the setting parameter E
A signal delay corresponding to 125 is performed on the signal Ei for each predetermined frequency range, and the result Eo is output.

FIG. 6 is a block diagram illustrating a case where the variable delay circuit 120 in the phase characteristic correction device 12A of FIG.

An input Ei (an audio source such as a pilot signal (a) or a CD) to be subjected to the phase correction is 1
Or more analog filters 1201a-1212
04a. If the input Ei is 80 Hz and 2
In the case of a 00 Hz mixed sine wave, two analog filters may be used. That is, the analog band-pass filter (or analog low-pass filter) 1201a extracts the 80 Hz component of the input Ei, and the analog band-pass filter (or analog high-pass filter) 1204a extracts the 200 Hz component of the input Ei.

The frequency components of the input Ei extracted by the filter in this manner are individually individually corresponding to delay circuits (or phase shift circuits; phase shifters) 1205a to 1205a to
08a. Each delay circuit (phase shift circuit) 1205
The signal delay amounts (or phase shift amounts) of the signals a to 1208a are shown in FIG.
E129 latched by the delay amount switching unit 129 of FIG.
(Delay amount setting parameter E125 obtained before that)
Is determined individually.

According to the data E129, for example, the delay circuit 1205a does not delay the 80 Hz component (or makes the delay amount small), and
If the 0 Hz component is delayed (or the delay amount is made relatively large), as a result, 80H of the input Ei
A waveform in which the phase of the z component is advanced is obtained.

The frequency components of the input Ei that have been variously delayed in accordance with the delay parameter E129 in this way are synthesized in the analog mixer 1209a, and become the output Eo after the phase shift correction corresponding to the input Ei. 14).

FIG. 7 is a block diagram exemplifying a case in which the delay amount variable delay circuit 120 in the phase characteristic correction device 12A of FIG. The circuit function of the digital delay circuit of FIG. 7 is basically the same as that of the analog delay circuit of FIG. However, an analog / digital converter (ADC) 1200d is arranged at the first stage receiving an input Ei for performing digital processing internally, and a DAC is provided in a digital mixer 1209d that synthesizes each signal after delay.

The ADC 1200d has a sampling frequency of 44 to 9 when importance is placed on signal quality.
Multi-bit analog / digital converter with 6 kHz and resolution of 16 to 24 bits or bit stream type analog / digital converter with sampling frequency of 100 kHz or more and resolution of 16 to 24 bits (clock of 64 fs when sampling frequency is fs) And a ΔΣ type 1- to 4-bit ADC operating at high speed.

Also, the DA in the digital mixer 1209d
As C, a high-speed and high-resolution DAC that matches the digital signal quality with a sampling frequency of 44 to 96 kHz and a resolution of 16 to 24 bits is used.

In the digital mixer 1209d of FIG. 7, the mixing ratio of each input frequency component can be designated by the mixing ratio setting signal E125 from the CPU in the detection unit 125 of FIG. The mixture ratio designation data is stored in a register (not shown) inside the mixer 1209d, but the content of the stored data is not necessarily a fixed value that is not fixed, but changes according to the operation level of the speaker 18 or a change in the use environment. May be changed as appropriate.

Each digital delay 1205d to 1208
d each has its own register (not shown), and the delay amount setting signal E12 corresponding to each of these registers.
9 (for example, 8-bit data) are written.

In the configuration shown in FIG.
As 201d to 1204d, a digital graphic equalizer (or parametric equalizer) having a １ ／ octave band to a ６ octave band may be used. That is, the digital type graphic equalizer includes digital delays 1205d to 1208 shown in FIG.
d, these digital delays 1205d-1
208d can be controlled by the configuration of FIG.

In the configuration of FIG. 6 or FIG. 7, when the audio band is not frequency-divided and the whole band is phase-corrected, or the phase is corrected only for a specific band (for example, 20 to 200 Hz) in the audio band. In this case, the combination of the filter (1201 to 1204) and the delay circuit / phase shift circuit (1205 to 1208) is one set (for example, 1201a + 1205a or 1201d + 12).
05d) may be used. In this case, the mixer (1209a or 1209d) can be omitted. However, in the configuration of FIG. 7, when it is necessary to return the signal after the digital phase correction to analog, an appropriate DAC is used.

In the above description, the target loudspeaker system is a dynamic loudspeaker. However, the present invention can be applied to phase correction of another type of loudspeaker system having similar phase characteristics.

The summary of the present invention is as follows: (1) A two-frequency mixed wave is input to a power amplifier as a pilot signal (a), and an output signal (c) from the power amplifier to a dynamic speaker is obtained. Detection and measurement system 12 / 12A
The sound output signal (d) from the dynamic speaker is collected by a microphone and input to the measurement system 12 / 12A.

(2) The waveforms of the three input signals (a, c, d) are measured and compared to determine a correction parameter (phase shift / delay and signal level for each frequency band) or a matching coefficient.

(3) The phase shift amount / delay amount and the signal level for each frequency band are set from the obtained parameters (particularly, the phase shift amount / delay amount on the low frequency side).

(4) The above three input signals (A, C, D)
Are measured and compared to evaluate the suitability of the parameter (3). If the sound output (d) from the dynamic speaker has not reached the desired quality level, steps (1) to (3) are repeated until the sound output reaches the desired level.

The desired level is determined by determining whether the absolute value of the phase difference between the signals (a) or (c) and (d) for each frequency band is equal to or less than a predetermined value (can be automatically determined by CPU processing). And a final hearing (subjective judgment by the system user). Hearing is
This is performed using a high-quality music source under the parameters set in (3).

[0110]

According to the present invention, once a music signal whose phase or the like is corrected is supplied to a power amplifier, the waveform reproducibility including the speaker system is improved as a whole, so that the tone quality of the reproduced sound is improved. It is better than the case where is not used.

[Brief description of the drawings]

FIG. 1 is a block diagram illustrating a system for analyzing a phase characteristic of an amplifier when a speaker is loaded in a phase compensation system according to an embodiment of the present invention.

FIG. 2 is a flowchart illustrating a method for observing phase characteristics in the phase characteristic analysis system of FIG. 1;

FIG. 3 is a block diagram illustrating a system for improving a phase characteristic of an amplifier when a speaker is loaded in the phase compensation system according to the embodiment of the present invention;

FIG. 4 is a flowchart illustrating a phase characteristic correction method in the phase characteristic improving system of FIG. 3;

FIG. 5 is a block diagram illustrating an internal configuration of a phase characteristic correction device 12A in the phase characteristic improvement system of FIG. 3;

FIG. 6 is a block diagram illustrating a case where a delay amount variable delay circuit 120 in the phase characteristic correction device 12A of FIG. 5 is configured by an analog method;

FIG. 7 is a block diagram illustrating a case where a delay amount variable delay circuit 120 in the phase characteristic correction device 12A of FIG.

FIG. 8 shows a phase characteristic (flat) of a power amplifier (OTL amplifier or OTL / OCL transistor DC amplifier) having no low-frequency phase shift element such as an output transformer, and a phase characteristic of a dynamic speaker (phase lag toward high frequency). FIG.

FIG. 9 shows a phase characteristic of a power amplifier (tube amplifier using a transformer) having a low-frequency phase shift element such as an output transformer or an input transformer (a tendency of phase advance toward a high frequency) and a phase characteristic of a dynamic speaker (high). FIG. 4 is a diagram schematically showing a tendency of phase delay toward a region).

FIG. 10 shows an output waveform of a transistor amplifier (phase flat) and an output waveform of a tube amplifier (phase advance) when a two-frequency mixed sine signal is used; and a transistor amplifier when the two-frequency mixed signal is used. FIG. 4 is a diagram showing, in a distorted form, an acoustic output waveform of a dynamic speaker driven by a tube amplifier and an acoustic output waveform of a dynamic speaker driven by a tube amplifier.

FIG. 11 shows an output waveform of a transistor amplifier (phase flat) and an output waveform of a tube amplifier (phase advance) when a single-shot sine signal is used;
The figure which deformed and shows the sound output waveform of the dynamic speaker driven by the transistor amplifier, and the sound output waveform of the dynamic speaker driven by the tube amplifier.

FIG. 12 shows an output waveform of a transistor amplifier (phase flat) and an output waveform of a tube amplifier (phase advance) when a single frequency sine signal is used; driven by a transistor amplifier when this signal is used. FIG. 6 is a diagram showing, in a deformed manner, an acoustic output waveform of a dynamic speaker and an acoustic output waveform of a dynamic speaker driven by a tube amplifier.

FIG. 13 shows an output waveform of a transistor amplifier (phase flat) and an output waveform of a tube amplifier (phase advance) when a two-frequency mixed single-shot sine signal is used; driven by a transistor amplifier when this signal is used. The figure which deformed and showed the acoustic output waveform of the dynamic speaker and the acoustic output waveform of the dynamic speaker driven by the tube amplifier.

FIG. 14 is a block diagram illustrating a configuration used when obtaining the waveform example of FIG. 10;

[Explanation of symbols]

Reference Signs List 10: audio signal source; 12: phase observation device (phase observation analyzer); 12A: phase characteristic correction device (phase characteristic correction device); 14: power amplifier; 14T: transistor OT
L / OCL amplifier; 14V ... tube amplifier using interstage transformer and output transformer (300 in output tube)
B: Single amplifier using B); 16: voltage / current detection probe (voltage / current detection means); 18: load (dynamic speaker or speaker system); 20: microphone (sound / electric conversion means); 101: oscillator 1; 102 ...
Oscillator 2; 103 adder (mixing amplifier / waveform synthesizer); 120 delay variable delay circuit (variable delay means); 124, 128 ADC (8-bit analog /
Digital converter; ADC means); 125: phase comparison / phase coincidence detection unit (internally 8- to 64-bit CPU processing; phase comparison means); 126, 127 ... ATT (attenuator /
Signal attenuator); 129 delay amount switching section (setting means); 1
201a to 1204a ... analog filter (low-pass,
Bandpass or highpass; analog filter means); 1205a to 1208a ... analog delay circuit or analog phase shift circuit (analog delay / phase shift means);
09a: Analog mixer (one whose level can be adjusted for each input); 1200d: ADC (sampling frequency 44
A multi-bit analog / digital converter having a resolution of 16 to 24 bits at a frequency of ~ 96 kHz or a bit stream type analog / digital converter having a resolution of 16 to 24 bits at a sampling frequency of 100 kHz or more); 1201
d to 1204d: digital filter (low-pass, band-pass, or high-pass; digital filter means);
05d to 1208d: digital delay circuit (digital delay means); 1209d: digital mixer (resolution: 16 to 2)
4 bit multi-bit or bitstream type D
Has AC; digital filters 1201d to 1204d
Does not have a digital attenuator function, if a level is adjusted for each input, a digital attenuator for level adjustment is provided for each input); SW1, SW2, SW
3. Changeover switch; BA1 to BA4: Buffer amplifier;
MA: Microphone amplifier.

Claims (19)

[Claims] An audio reproducing apparatus including a predetermined speaker system and a predetermined power amplifier for driving the speaker system, comprising: an audio signal source for generating a pilot signal including one or more frequency components; Phase characteristic correcting means for delaying or changing the phase of a pilot signal and supplying a delayed or phase-changed audio signal to the power amplifier; and supplying the audio signal from the power amplifier to the speaker system in response to the audio signal Voltage for detecting the content of the speaker drive signal
Current detection means; sound / electricity conversion means for converting sound output generated from the speaker system into an electric signal in response to the speaker drive signal;
Comparing at least two waveforms of the speaker drive signal and the audio output, and phase and / or phase of one or more frequency components included in the audio signal so that the waveform of the audio output approximates the waveform of the pilot signal.
Alternatively, a phase compensation system for a sound reproducing apparatus, wherein the delay amount is changed. 2. The method according to claim 1, wherein the at least one frequency component of the pilot signal includes, in an audio band, a fundamental frequency component and an additional frequency component that forms a beat having no harmonic relation to the fundamental frequency component. The system of claim 1, wherein: 3. The method according to claim 1, wherein the signal detected by the voltage / current detection means includes a voltage component, a current component, or a power component of the speaker drive signal. The described system. 4. A first digital signal providing a first digital signal by digitizing a waveform of the pilot signal or the speaker drive signal.
ADC means; second ADC means for digitizing the waveform of the sound output to provide a second digital signal;
Phase comparing means for comparing the phase of the digital signal with the second digital signal to detect the presence / absence of a phase match and providing a phase comparison result indicating the presence / absence of a phase match; and providing a setting signal corresponding to the phase comparison result Variable delay means for delaying the pilot signal or the speaker drive signal by an amount determined by the setting signal or delaying the phase of the pilot signal or the speaker drive signal to provide the audio signal; The system according to any one of claims 1 to 3, comprising: 5. The variable delay means: filter means for extracting a predetermined frequency component from the pilot signal; and delaying or delaying the signal waveform of the frequency component extracted by the filter means by an amount determined by the setting signal. 5. The system of claim 4 including phase and delay shifting means. 6. The filter means includes a plurality of filter circuits for dividing the pilot signal into a plurality of frequency bands, and the delay / phase shift means includes a plurality of frequency components extracted from the plurality of filter circuits. A plurality of delay / phase shift circuits for individually delaying or individually delaying the waveforms by an amount determined by the setting signal, wherein the variable delay means is individually delayed by the plurality of delay / phase shift circuits; 6. The system according to claim 5, further comprising a mixer circuit for combining the delayed signals at a predetermined ratio to provide the audio signal. 7. The variable delay means: a digital filter means for extracting a predetermined frequency component from the pilot signal; and a signal waveform of the frequency component extracted by the digital filter means is delayed by an amount determined by the setting signal. 5. The system of claim 4 including digital delay means. 8. The digital filter means includes a plurality of digital filter circuits for dividing the pilot signal into a plurality of frequency bands, and the delayed digital delay means includes a plurality of frequencies extracted from the plurality of digital filter circuits. A plurality of digital delay circuits for individually delaying each of the digital data of the components by an amount determined by the setting signal, wherein the variable delay unit converts a digital signal individually delayed by the plurality of digital delay circuits into a predetermined signal; Digital mixer DA for converting the synthesized digital signal into an analog signal and providing the audio signal
The system of claim 7, including a C circuit. 9. A sound reproducing apparatus including a predetermined speaker system and a predetermined power amplifier for driving the speaker system, a first step of generating a pilot signal including one or more frequency components; A second step of detecting the content of a speaker drive signal supplied from the power amplifier to the speaker system; and converting a sound output generated from the speaker system into an electric signal in response to the speaker drive signal. A fourth step of comparing the phases of the pilot signal, the speaker drive signal, and the sound output waveform; and delaying the pilot signal so that the sound output waveform approximates the pilot signal waveform or By changing its phase, the delayed or phase-changed Phase compensation method of sound reproduction apparatus characterized by comprising a; a I o signal and a fifth step for supplying to said power amplifier. 10. The method according to claim 9, wherein the pilot signal includes a plurality of frequency components, and at least one of the plurality of frequency components is changed to repeat the processing of the first to fifth steps. The described method. 11. The pilot signal includes a plurality of frequency components, wherein the plurality of frequency components form an additional frequency component that forms a fundamental frequency component and a beat having no harmonic relationship with the fundamental frequency component in an audio band. The method according to claim 9 or claim 10, comprising: 12. The signal detected in the second step includes a voltage component and a current component of the speaker drive signal.
12. The method according to any one of claims 9 to 11, comprising including a power component. 13. An audio reproducing apparatus including a predetermined speaker system and a predetermined power amplifier for driving the speaker system, wherein: an audio signal source for generating a pilot signal including one or more frequency components; Voltage / current detecting means for detecting the content of a speaker drive signal supplied from the power amplifier to the speaker system; and converting an acoustic output generated from the speaker system into an electric signal in response to the speaker drive signal. Sound / electric conversion means; delaying a pilot signal from the audio signal source or changing its phase,
Phase observation analysis means for supplying a delayed or phase-changed audio signal to the power amplifier, wherein the phase observation analysis means compares at least two waveforms among the pilot signal, the speaker drive signal, and the sound output. In addition, the similarity between the waveform of the sound output and the waveform of the pilot signal is observed and evaluated by focusing on the phase and / or delay amount of one or more frequency components included in these waveforms. A phase characteristic analysis system for a sound reproducing apparatus, characterized by: 14. The pilot signal includes a plurality of frequency components, wherein the plurality of frequency components form a fundamental frequency component and an additional frequency component that forms a beat having no harmonic relation to the fundamental frequency component in an audio band. The system of claim 13, comprising: 15. A transformer-coupled amplifier including an audio transformer in a signal path as a reference of the power amplifier, and a transformerless amplifier not including an audio transformer in a signal path as an object of analysis / evaluation. In the means, the delay amount or the phase change amount of the audio signal source with respect to the pilot signal is adjusted so that the evaluation result of the waveform approximation by the transformerless amplifier approaches the evaluation result of the waveform approximation by the transformer coupling amplifier. The system according to claim 13 or 14, wherein the system is configured as follows. 16. A sound reproducing system including a predetermined speaker system and a predetermined power amplifier for driving the speaker system, wherein a first step of generating a pilot signal including one or more frequency components; A second step of detecting the content of a speaker drive signal supplied from the power amplifier to the speaker system; and converting a sound output generated from the speaker system into an electric signal in response to the speaker drive signal. And a fourth step of comparing the phases of the pilot signal, the speaker drive signal, and the waveform of the sound output; and
A fifth step of measuring and recording the phase difference between the speaker drive signal and the sound output at the above-mentioned level points. 17. The method according to claim 16, wherein the pilot signal includes a plurality of frequency components, and at least one of the plurality of frequency components is changed to repeat the processing of the first to fifth steps. The described method. 18. The pilot signal includes a plurality of frequency components, wherein the plurality of frequency components form an additional frequency component that forms a fundamental frequency component and a beat having no harmonic relationship with the fundamental frequency component in an audio band. 18. The method according to claim 16 or claim 17, comprising: 19. A signal detected in the second step includes a voltage component, a current component,
19. The method according to any one of claims 16 to 18, comprising including a power component.