JPH06177688A - Audio signal processing unit - Google Patents

Abstract

PURPOSE:To obtain an audio signal processing unit providing a suitable reproduction sound for a wide range of music. CONSTITUTION:A low pass filter 4 and a high pass filter 5 performs band- division of an audio signal into a low sound frequency signal and a high sound frequency signal. A peak value analyzer 6 analyzes a peak value of the low sound frequency signal. Furthermore, delay sections 15a to 15d delay both the low sound frequency signal and the high sound frequency signal. A voltage controlled amplifier 7 compresses a level of an output of the delay sections 15a, 15b in response to the result of analysis by the peak value analyzer 6. A mixer circuit 11 mixes the compressed low sound frequency signal with the high sound frequency signal from the delay sections 15c, 15d.

Description

Detailed Description of the Invention

[0001]

BACKGROUND OF THE INVENTION This invention is mounted on a vehicle,
The present invention relates to an audio signal processing device for optimizing an audio reproduction signal.

[0002]

2. Description of the Related Art Conventionally, various sound quality adjusting functions have been devised in an audio reproducing apparatus, and at present, the audio reproducing apparatus aims to improve the sound quality of the audio system as a whole by relating those functions to the usability of the user. In the direction. In particular, in an audio system mounted on a vehicle, a function for compensating for a lack of a sense of volume in a low sound range due to running noise is realized. For example,
The bass boost function, the graphic equalizer function to correct the acoustic characteristics of the passenger compartment, and the auto loudness function that changes the frequency characteristics in conjunction with the volume adjustment used to correct the loudness characteristics in listening psychology are put to practical use. Has been done.

FIG. 26 is a block diagram showing the structure of a conventional vehicle audio system. Here, a system with four speakers is shown. In the figure, 1 is an audio reproducing device such as a CD player or cassette tape player, 2 is a signal processing for a stereo signal from the audio reproducing device 1 having the above-mentioned low range boost function, loudness control function, graphic equalizer function and the like. An audio signal processing circuit 21 to 24 is a speaker installed in the passenger compartment 200, and 31 to 34 are audio reproducing devices 2
It is a power amplifier that amplifies the signal from and gives it to the corresponding speaker.

Next, the operation will be described. The audio reproducing device 1 reproduces a stereo signal from the audio recording medium and outputs it to the audio signal processing circuit 2. The audio signal processing circuit 2 performs signal processing on the stereo signal by each function described above, and distributes the processed signal to the power amplifiers 31 to 34. Each power amplifier 31-3
4 amplifies the power of the input signal and outputs each speaker 21 to 2
Output to 4. Each of the speakers 21 to 24 outputs a sound based on the input signal. As described above, the passenger compartment 2
Audio is played with 00 as a sound field.

However, in a vehicle audio system,
There are restrictions on the sizes, weights, and mounting positions of the speakers 21 to 24, and the restrictions limit the ability to reproduce deep bass. When a signal exceeding this limit (permissible capacity) is applied to the speakers 21 to 24, the sound is distorted and an unpleasant sound field is provided. Therefore, the audio signal processing device 2 defines the maximum output voltage, and the power amplifier 31
The output voltages of ~ 34 are controlled so as not to exceed the allowable inputs of the speakers 21-24.

Alternatively, the speakers 21 to 21 are utilized by utilizing the characteristic that the power amplifiers 31 to 34 are clipped by the power supply voltage.
24 is also designed to prevent excessive input. However, in that case, distortion in the speakers 21 to 24 is suppressed, but large distortion occurs in the power amplifiers 31 to 34, so that the audible sound quality is deteriorated.

The conventional vehicle audio system has been devised so as to bring out a well-balanced reproduction capability as a whole by the above-mentioned configuration. However, when reproducing a signal having a wide dynamic range such as a signal from a digital audio medium typified by a compact disc, there arises a problem in reproducing ability. Moreover, the problem is exacerbated when playing music of various genres, ranging from classical music to hard rock music.

For example, for a hard rock music signal,
As can be seen from the frequency spectrum shown in FIG. 27A, the components in the deep bass range of 40 Hz to 100 Hz are extremely large. Also in the time waveform shown in FIG. 27B, there is a protrusion of the wave height due to the bass component. When a music signal with such a waveform is played back by a system based on the conventional concept,
The presence of the components in the deep bass region causes distortion in the power amplifier and the speaker, and as a result, distorted reproduced sound is generated.

In order to prevent distortion, circuit design is generally performed with the overall amplification factor suppressed. However, in a system based on such a design, when a classical or pop music signal is played, the recording level is low, so even when the volume adjustment level is maximized, a powerful and unsatisfactory sound is played. Will be done.

In order to deal with such a problem, an audio signal processing apparatus that prevents excessive input to a speaker by suppressing the level of a waveform whose level is excessive by compressing the waveform when amplifying an audio signal by a non-linear amplifier. Is proposed. As a typical example of such an audio signal processing device, there is one using the power supply voltage clipping characteristic of the power amplifier described above as a non-linear characteristic. However, as described above, according to the one using the power supply voltage clip characteristic of the power amplifier, the amount of distortion changes depending on the degree of nonlinearity, but instead of suppressing the distortion in the speaker, the sound quality deteriorates in the power amplifier part. Resulting in.

Therefore, instead of utilizing the non-linear characteristic of the power amplifier, as shown in FIG. 28, the waveform compression circuit 10 as the non-linear amplification means is provided between the audio signal processing circuit 2 and the power amplifiers 31 and 32. A method has been proposed in which waveform compression is performed by the waveform compression circuit 10 at the time of amplifying an audio signal to suppress the level of a waveform having an excessive level. Note that, here, a two-speaker system is taken as an example.

As a concrete configuration example of the waveform compression circuit 10, there is one shown in FIG. In this configuration, the input signal from the audio signal processing circuit 2 is amplified by the amplifier 11 and then detected by the wave detector 12. The detection signal from the detector 12 is smoothed by the time constant circuit 13 with a predetermined time constant. Therefore, the average detection level is output from the time constant circuit 13. The voltage control type amplifier 14 changes the amplification factor according to the detection level, amplifies the input signal with the amplification factor, and supplies it to the power amplifiers 31 and 32 as an output signal.

Another concrete example of the configuration of the waveform compression circuit 10 is shown in Japanese Utility Model Laid-Open No. 63-35311 as shown in FIG. In this configuration, the output signal is amplified by the amplifier 11 and then detected by the wave detector 12. The detection signal from the detector 12 is smoothed by the time constant circuit 13 with a predetermined time constant, and the time constant circuit 13 outputs an average detection level. The voltage controlled amplifier 14 is
The amplification factor is changed according to the detection level, and the input signal is amplified and output with the amplification factor. The low pass filter circuit 17 passes the low frequency component of the output of the voltage controlled amplifier (VCA) 14. In addition, the high pass filter circuit 15
Passes the treble component of the input signal from the audio signal processing circuit 2. Then, the synthesis circuit 16 synthesizes the output of the low pass filter circuit 17 and the output of the high pass filter circuit 15 and supplies them to the power amplifiers 31 and 32 as output signals.

The waveform compression circuit 10 shown in FIG. 29 performs level detection for an input signal, and the waveform compression circuit 10 shown in FIG. 30 performs signal processing for an output signal. The audio signal of the entire band is used as the target of the level detection for. Also,
The average level smoothed with a predetermined time constant is output as the detection level.

[0015]

Since the conventional audio signal processing apparatus is constructed as described above, since the whole band signal is targeted for level detection, the high frequency range portion which originally does not require waveform compression is used. Waveform compression may occur in response to the level. Therefore, there is a problem that the sound quality is deteriorated and a sound dropout that weakens the sound more than necessary is likely to occur.

Further, since the waveform is compressed according to the average level, the nonlinear distortion at the maximum peak value of the rising edge is applied to the damping vibration that is often seen in the rock type music signal waveform shown in FIG. However, there is a problem in that it cannot be avoided by waveform compression.

As another prior art, US Pat.
There is a method disclosed in Japanese Patent No. 398061, which discloses a method of detecting a peak value between zero cross points in an audio signal waveform and compressing the waveform between the zero cross points according to the peak value. According to such a method, it is possible to sufficiently suppress the distortion due to the nonlinear characteristic of the audio signal reproducing system and the excessive wave height. However, since the full-band signal is targeted for peak detection, compression is also performed on the peak value components in the middle and high range, and as a result, the phenomenon of sound dropout may also appear. There is an unavoidable sense of discomfort.

As described above, according to the audio signal processing device in the conventional vehicle audio system, it is impossible to obtain a suitable listening sound for a wide range of audio sources. The main reason for this is that the signal processing method has not been designed based on the knowledge of the shape of the music signal and the psychoacoustic knowledge of what is the sense of distortion for the reproduced sound.

The present invention has been made in order to solve the above-mentioned problems, and it is suitable for a wide range of music by utilizing the fact that the difference in music genre appears in the low range component in the music signal. It is an object to obtain an audio signal processing device that provides sound.

[0020]

An audio signal processing apparatus according to the present invention comprises band division filter means for band-dividing an audio signal into a low range signal and a high range signal, and the band division filter means. Waveform analysis means for analyzing the peak value of the bass range signal obtained by division, waveform compression means for compressing the level of the waveform of the bass range signal according to the analysis result of this waveform analysis means, and compression by this waveform compression means And a mixing means for mixing the low-frequency range signal thus generated and the high-frequency range signal from the band dividing filter means.

In the audio signal processing device according to the present invention, the band division filter means for band-dividing the audio signal into the low range signal and the high range signal, and the low level obtained by the division by the band division filter means. Waveform analyzing means for analyzing the peak value of the range signal, delay means for giving a time delay including the time required for the analyzing process of the waveform analyzing means to the low range signal and the high range signal from the band division filter means, and the waveform Waveform compression means for compressing the level of the waveform of the bass range signal delayed by the delay means according to the analysis result of the analysis means, and the bass range signal compressed by the waveform compression means and the treble range delayed by the delay means. And mixing means for mixing with the signal.

An audio signal processing apparatus according to a third aspect of the present invention is the audio signal processing apparatus according to the first or second aspect, in which the volume adjusting means for adjusting the level of the audio signal and the adjustment by the volume adjusting means are provided. The compression degree control means for reflecting the degree in the analysis result of the waveform analysis means is further provided.

An audio signal processing apparatus according to a fourth aspect of the present invention is the audio signal processing apparatus according to the first, second or third aspect, wherein the band division filter means, the waveform analysis means and the waveform compression means are provided. It is composed of a digital processing circuit.

An audio signal processing apparatus according to a fifth aspect of the present invention is the audio signal processing apparatus according to the fourth aspect, wherein the band division filter means is composed of a digital mirror filter.

An audio signal processing device according to a sixth aspect of the present invention is the audio signal processing device according to the fourth aspect, wherein the band division filter means, the waveform analysis means and the waveform compression means are constituted by a digital signal processing integrated circuit. It is a thing.

An audio signal processing device according to a seventh aspect of the present invention is the audio signal processing device according to the fourth, fifth or sixth aspect, wherein the delay amount in the delay means is 5 milliseconds to 100 milliseconds. It is set to.

According to an eighth aspect of the present invention, there is provided an audio signal processing device according to the fourth, fifth or sixth aspect, wherein the waveform analysis means performs waveform analysis with a unit between zero cross points of the waveform. Is to do.

An audio signal processing device according to the invention of claim 9 is the audio signal processing device according to claim 4, claim 5 or claim 6, wherein the waveform analysis means is a unit for each waveform within a fixed time. The waveform analysis is performed as.

[0029]

The waveform analysis means in the invention of claim 1 is
The crest value of only the low frequency range signal of the audio signal is analyzed. The waveform compression means compresses the level of the bass signal in accordance with the peak value of the bass signal of the audio signal.

The delay means in the invention according to claim 2 is
The signal from the band division filter means is delayed so that the analysis result of the waveform analysis means is reflected in the waveform to be analyzed.

According to the third aspect of the invention, the compression degree control means allows the waveform compression means to perform effective compression by adding the adjustment degree of the volume adjustment means to the analysis result of the waveform analysis means.

Each means in the invention of claim 4 is constituted by a digital processing circuit. Therefore, the band dividing filter means is composed of a digital filter and realizes more strict band division of the audio signal.

The band division filter means in the fifth aspect of the present invention is constituted by a digital mirror filter and more strictly realizes band division of the audio signal.

Each means in the invention of claim 6 is constituted by a digital signal processing integrated circuit, and simplifies the circuit configuration of the entire apparatus.

According to the seventh aspect of the invention, the delay means gives the input signal a delay amount suitable for the characteristics of the audio signal to suppress the unnatural feeling of the reproduced sound.

According to the eighth aspect of the invention, the waveform analysis means sets one analysis period between zero cross points so that waveform distortion does not occur between one analysis period and the next analysis period.

The waveform analysis means in the invention of claim 9 fixes the analysis period and realizes simplification of the circuit.

[0038]

EXAMPLES Example 1. FIG. 1 is a configuration diagram showing a configuration of a vehicle audio system including an audio signal processing device according to an embodiment to which the first invention of the present invention is applied. In the figure, 1 is an audio reproducing device such as a CD player or cassette tape player, 2 is a bass range boosting function, a loudness control function, a graphic equalizer function, etc., and performs signal processing on a stereo signal from the audio reproducing device 1. An audio signal processing circuit, 3 is a volume adjuster, 4 is a low-pass filter that passes a low frequency range of the audio signal whose volume is adjusted by the volume adjuster 3, and 5 is a high level of the audio signal whose volume is adjusted by the volume adjuster 3. A high-pass filter that passes the sound range, 6 is a wave height analyzer that analyzes the peak value of an audio signal in the low sound range, and 7 is a voltage control that amplifies the output signal of the low-pass filter 4 according to the output of the wave height analyzer 6. Type amplifier (VCA), 8 is a voltage controlled amplifier 7
And the output of the high-pass filter 5 are mixed, and the mixing / distributing circuits 21 to 24 that distribute signals to the power amplifiers 31 to 33 are speakers installed in the vehicle compartment 200.

The low-pass filter 4 and the high-pass filter 5 are examples of implementation of a band division filter, and the wave height analyzer 6 is an example of implementation of waveform analysis means. Further, the voltage control type amplifier 7 is an implementation example of the waveform compression means, and the mixing / distributing circuit 8 is an implementation example of the mixing means.

Here, the psychoacoustic knowledge which is the basis of such a configuration will be described. The human sense of volume is
It depends on both the sound frequency and the sound pressure level. Although various research results on volume perception have been reported, basically, human volume perception exhibits the characteristics shown in FIG. In FIG. 2, the vertical axis represents the sound pressure level. This figure is obtained by plotting the sound pressure levels at other frequencies that sound as loud as the sound pressure level at a given frequency (eg, 1 kHz).

According to the characteristics shown in FIG. 2, the audible sound pressure is 10
At a sound volume exceeding 0 dB, the sensitivity in the middle and low range is almost flat. At low volume, the bass sensitivity is much less sensitive.
Generally, the listening volume when listening to music in a quiet room is considered to be around 80 dB on average. 60dB ~ 1
Near 00 dB, the sensitivity at frequencies below 100 Hz is 5 dB to 20 dB slower than the sensitivity near 1 kHz. In the passenger compartment 200 of the vehicle, since the traveling noise component of 100 Hz or less is large, the volume sensitivity in the low frequency range becomes worse.

Therefore, as shown in the spectrum distribution of FIG. 3, the bass and bass music signals, in which the bass component is not particularly large, are sensitive to bass boost or the like when the vehicle is running. Correction is needed. However, as shown in the spectrum distribution of FIG. 4, it is not always necessary to perform low range boosting for a rock type music signal having a large low range component when listening at a high volume.

Based on the above psychoacoustic knowledge, it is understood that the correction of the sense of volume needs to be executed according to the genre of music and the listening level. Figure 1
The audio signal processing device shown in (1) is constructed based on such knowledge.

Next, the operation of the audio signal processing apparatus shown in FIG. 1 will be described. The audio reproducing device 1 reproduces a stereo signal from the audio recording medium and outputs it to the audio signal processing circuit 2. The audio signal processing circuit 2 has a bass boost function, as in the conventional case.
A graphic equalizer function for correcting the acoustic characteristics of the passenger compartment, an auto loudness function that changes the frequency characteristics in conjunction with the volume adjustment used for the purpose of correcting the loudness characteristics in listening psychology, perform signal processing on the stereo signal, The processed signal is output to the volume adjuster 3. The volume adjuster 3 responds to the volume setting operation by the user.
Volume adjustment processing is performed on the input signal.

The output of the volume controller 3 is sent to the low pass filter 4 and the high pass filter 5. The low-pass filter 4 passes a low-pass component of the signal input from the volume adjuster 3. The input signal generally has a bandwidth of 20 Hz.
The cut-off frequency of the low-pass filter 4 is set to 100 Hz to 200 Hz although the audio signal is -20 kHz. The high-pass filter 5 has a characteristic that is complementary to the characteristic of the low-pass filter 4, and passes the high-pass component of the signal input from the volume adjuster 3. The low pass filter 4 and the high pass filter 5 may be second-order active filters. Moreover, it is desirable that the complementarity of the cutoff characteristics of the two is ensured as much as possible.

The output signal of the low pass filter 4 is sent to the pulse height analyzer 6 and the voltage controlled amplifier 7. The wave height analyzer 6 detects the peak of the output signal of the low pass filter 4 at a predetermined time interval. Then, the peak level of the input signal is compared with a predetermined level, and a control voltage corresponding to the degree of waveform compression according to the comparison result is output to the voltage controlled amplifier 7. Then, the voltage control type amplifier 7 level-compresses the signal output from the low pass filter 4 according to the control voltage according to the control voltage.

FIG. 5 is a block diagram showing an example of a concrete circuit configuration of the wave height analyzer 6. However, here, only one of the two stereo channels is shown. In the figure, reference numeral 40 is a peak hold circuit composed of operational amplifiers 41 and 42, a diode 43, a peak hold capacitor 44, and a reset analog switch 45.

The input signal from the low pass filter 4 is connected to the non-inverting input terminal of the operational amplifier 41. The output of the operational amplifier 41 is added to the peak hold capacitor 44 via the diode 43 and the operational amplifier 4
2 non-inverting input terminals. The output of the operational amplifier 42 becomes the peak value detection output, and the output of each operational amplifier 4
It is fed back to the inverting input terminals of 1, 42.

50 is an operational amplifier 51, a diode 52,
53 is a pulse generation circuit including resistance elements 54 to 57 and a capacitor 58. The series body of the two resistors 56 and 57 is connected between both poles of the power source E1. Resistance 56,5
The connection point of 7 is connected to the inverting input terminal of the operational amplifier 51. The output of the operational amplifier 51 becomes the output of the reset pulse generating circuit 50 and is fed back to the non-inverting input terminal. Further, a series body of the resistance element 55 and the diode 52 is connected between the output terminal of the operational amplifier 51 and the non-inverting input terminal, and the diode 52 is connected in antiparallel with the diode 53. The non-inverting input terminal of the operational amplifier 51 is grounded via the capacitor 58.

The operational amplifier 51, the diodes 52 and 53, the capacitor 58 and the resistance elements 54 to 57 as described above.
An astable multivibrator is configured by connecting
This unstable multivibrator outputs a pulse train having a cycle determined by the resistance value of the resistance element 55 and the capacitance of the capacitor 58. Each pulse becomes a reset pulse and drives the reset analog switch 45 of the peak hold circuit 40.

In the peak hold circuit 40, the peak hold capacitor 44 holds the peak value of the input signal, but is reset when the analog switch 45 is turned on by the reset pulse from the reset pulse generation circuit 50. Therefore, the cycle of the reset pulse is equal to the cycle of peak detection. That is, the pulse output cycle of the unstable multivibrator is set to be equal to the desired peak detection cycle.

It is desirable that this cycle is set to a value between 5 ms and 100 ms. This is because the audio signal of the crest value analysis is a bass component of the components band-divided by the low pass filter 4 having a cutoff frequency of a value between 100 Hz and 200 Hz. Therefore, the wave height analyzer 6
The upper limit frequency of the signal input to is at most 200 Hz. The cycle of the signal of the upper limit frequency is 5 ms. Since the peak value analysis only needs to target one cycle of this waveform, the peak detection cycle may be set to 5 ms or more.

The lower limit frequency of the audio signal is considered to be about 10 Hz when considering a signal from a digital medium such as a compact disc. The period of the signal of the lower limit frequency is 100 ms. Therefore, the upper limit of the peak detection cycle is 100 ms. In practice, it is set to an appropriate value between 5 ms and 100 ms in consideration of the type and cost of the music recording medium to be handled.

FIG. 6 is a block diagram showing an example of a concrete circuit configuration of the voltage control type amplifier 7. Again, only one of the two stereo channels is shown. As shown in the figure, the detection output of the wave height analyzer 6 is applied to a photoelectric conversion element 61 such as a photocoupler as an input S _C via a resistance element 63. The photoelectric conversion element 61 is
It is composed of a light emitting diode 611 and a light receiving element 612. Further, the output of the low pass filter 4 is applied to the non-inverting input terminal of the operational amplifier 62 as the input S _L. The non-inverting input terminal of the operational amplifier 62 is grounded via the resistance element 65. The inverting input terminal of the operational amplifier 62 is grounded via the resistance element 64. The light receiving element 612 of the photoelectric conversion element 61 is connected between the output terminal and the inverting input terminal of the operational amplifier 62.

The voltage control type amplifier 7 configured as described above.
At, the light amount of the light emitting diode 611 changes according to the detection output from the wave height detector 6. Along with this, the resistance value of the light receiving element 612 changes. That is, the value of the feedback resistance of the operational amplifier 62 changes. Therefore, the wave height detector 6
The amplification factor changes according to the detection output of. Specifically, the larger the peak value detection output, the smaller the amplification factor,
Waveform level compression will be performed. Then, the output of the operational amplifier 62 becomes the output of the voltage control type amplifier 7 and is given to the mixing / distributing circuit 8. The mixing and distributing circuit 8 is
The output signal of the voltage control type amplifier 7 of each of the R and L channels and the output signal of the high pass filter 5 are mixed, and mixed L
The signal is distributed to the power amplifiers 31 and 33, and the mixed R signal is distributed to the power amplifiers 32 and 34. Each of the power amplifiers 31 to 34 power-amplifies the input signal and outputs it to each of the speakers 21 to 24.

Next, the operation of the audio signal processing circuit will be described with reference to the concrete waveforms shown in FIG. FIG. 7A shows an example of a waveform changing process when a lock audio signal is processed. A characteristic of rock music is that bass instruments are frequently used, and the audio signal thereof contains a large amount of low frequency components below 200 Hz. That is, the waveform becomes, for example, the waveform a. When the signal of this waveform is input to the low-pass filter 4, the signal of waveform b is output from the low-pass filter 4. Further, the high-pass filter 5 outputs a signal having a waveform c.

The peak height analyzer 6 analyzes the waveform b and outputs the peak value, and the voltage control type amplifier 7 performs the waveform compression according to the peak value. In this example, the signal of the waveform b has its level compressed to about 1/2 and becomes the signal of the waveform d. The waveform c of the high frequency component is the unprocessed waveform e. The mixing / distributing circuit 8 mixes the waveform-compressed waveform d signal and the high-frequency waveform e signal to generate a desired waveform f signal.

FIG. 7B shows an example of the changing process of the waveform of the audio signal of pop music. A characteristic of pop music is that the vocal range occupies most of the area, and as shown by the waveform g, the deep bass portion is not extremely large. Therefore, the crest value of the signal of the waveform h that has passed through the low-pass filter 4 is not so large, and the voltage controlled amplifier 7 outputs the signal of the waveform j that is almost the same as the input signal. In addition, the high-pass filter 5 outputs the waveform i
Is output, and the unprocessed waveform k is input to the mixing and distributing circuit 8. The mixing / distributing circuit 8 mixes the signal of the waveform j and the signal of the high-frequency component waveform k to generate a signal of the waveform m. The signal waveform is input to the low-pass filter 4 and the high-pass filter 5. It is equivalent to the signal waveform g.

FIG. 7C shows an example of the changing process of the waveform of the audio signal subjected to the low range boost of pop music. In this case, the bass range is emphasized by the audio signal processing circuit 2. When the user performs an operation such as increasing the volume while the vehicle is traveling, the volume adjuster 3 outputs an audio signal whose level is raised. That is, the signal of the waveform n having a higher level than the waveform g is given to the low pass filter 4 and the high pass filter 5.

Since the low frequency range is emphasized, the low-pass filter 4 outputs a signal of waveform o. Therefore, the crest analyzer 6 analyzes the waveform o and outputs the crest value, and the voltage control type amplifier 7 performs the waveform compression according to the crest value. In this example, the signal of waveform o has its level slightly compressed,
The signal has a waveform q. In addition, the high pass filter 5
A signal of waveform p is output from the
The unprocessed waveform r is input. The mixing / distributing circuit 8 mixes the waveform-compressed waveform q signal and the high-frequency component waveform r signal to generate a desired waveform s signal.

As described above, this audio processing apparatus performs processing adapted to each input audio signal regardless of various changes in the music source and regardless of the presence or absence of preprocessing such as bass boost. To do. Therefore, unnecessary waveform compression is not applied to the pops or classicals audio signals. In the conventional device, since the signal in the entire range was subjected to waveform compression for level detection,
It was not possible to perform processing adapted to each audio signal.

Example 2. FIG. 8 is a block diagram showing the configuration of an audio signal processing apparatus according to the second embodiment of the present invention. In the device shown in FIG. 1, the signal to be amplified by the voltage-controlled amplifier 7 is at a time point delayed by one analysis period from the signal to be analyzed by the peak value analyzer 6 by the peak value analyzer 6.
If the signal targeted for amplification by the voltage controlled amplifier 7 matches the signal targeted for peak value detection by the peak height analyzer 6, the peak value analysis effect is greater.

The apparatus shown in FIG. 8 has a structure that makes the peak value analysis more effective. That is, in comparison with the configuration shown in FIG. 1, the low pass filter 4 and the voltage control type amplifier 7 are provided.
And the delay units 15a and 15b are added between and. Further, delay units 15c and 15d are added between the high pass filter 5 and the mixing circuit 11. Each delay unit 15a-1
The delay amount of 5d is an amount corresponding to one analysis period in the wave height analyzer 6. Here, a two-speaker system is taken as an example. Therefore, a mixing circuit 11 is provided in place of the mixing / distributing circuit 8 shown in FIG. Also, the delay unit 15
a to 15d are implementation examples of the delay means described in claim 2.

Next, the operation will be described. The audio reproducing device 1 reproduces a stereo signal from the audio recording medium and outputs it to the audio signal processing circuit 2. The audio signal processing circuit 2 has a bass range boost function, a graphic equalizer function for correcting the acoustic characteristics of the passenger compartment, and an automatic frequency characteristic that changes in conjunction with the volume adjustment used for the purpose of correcting the loudness characteristics in listening psychology. The loudness function performs signal processing on the stereo signal and outputs the processed signal to the volume adjuster 3. The volume adjuster 3 performs volume adjustment processing on the input signal according to the volume setting operation by the user.

The output of the volume adjuster 3 is sent to the low pass filter 4 and the high pass filter 5. The low-pass filter 4 passes a low-pass component of the signal input from the volume adjuster 3. The cutoff frequency of the low-pass filter 4 is
As in the case of the first embodiment, it is set to 100 Hz to 200 Hz. The high-pass filter 5 has a characteristic that is complementary to the characteristic of the low-pass filter 4, and passes the high-pass component of the signal input from the volume adjuster 3.

The L signal of the output signals of the low pass filter 4 is sent to the wave height analyzer 6 and the delay section 15a. Further, the R signal is sent to the wave height analyzer 6 and the delay unit 15b. As in the case of the first embodiment, the wave height analyzer 6 has 5
The peak of the output signal of the low pass filter 4 is detected at time intervals of 100 ms to 100 ms. Then, the peak level of the input signal is compared with a predetermined level, and a voltage corresponding to the degree of waveform compression according to the comparison result is supplied to the voltage control type amplifier 7.
Output to. The detailed configuration and operation of the wave height analyzer 6 are the same as in the case of the first embodiment, so a detailed description is omitted here.

The delay units 15a and 15b delay the output signal of the low pass filter 4 by one analysis period and output it to the voltage control type amplifier 7. The voltage-controlled amplifier 7 level-compresses the signals output from the delay units 15a and 15b according to the control voltage according to the control voltage. Then, the compressed signal is output to the mixing circuit 11. The detailed configuration and operation of the voltage control type amplifier 7 are similar to those in the first embodiment, and therefore detailed description thereof will be omitted here.

The delay units 15c and 15d delay the output signal of the high-pass filter 5 by one analysis period and output it to the mixing circuit 11. The mixing circuit 11 includes the output signals of the voltage controlled amplifier 7 for the R and L channels and the delay units 15c and 15
The output signal of d is mixed, the mixed L signal is outputted to the power amplifier 31, and the mixed R signal is mixed with the power amplifier 3
Distribute into 2. Each power amplifier 31, 32 power-amplifies the input signal and outputs it to each speaker 21, 22.

As described above, by delaying the audio signal by the waveform analysis time, the analysis result and the audio signal to which the result is applied can be matched, and a reproduced sound with less abnormal sound can be provided. .

In the audio signal processing device shown in FIG. 8 and the audio signal processing device shown in FIG. 1, the wave height analyzer 6 and the voltage control type amplifier 7 are arranged in the subsequent stage of the audio signal processing circuit 2 and the volume controller 3. Therefore, the SN characteristic is slightly deteriorated. In order to prevent the deterioration, it is conceivable to dispose the wave height analyzer 6 and the voltage control type amplifier 7 in the preceding stage of the audio signal processing circuit 2 and the sound volume adjuster 3. However, in such a case, the listening sound volume condition is set. Crest value detection and waveform compression with consideration cannot be performed,
The object of the invention is not achieved.

Example 3. FIG. 9 is a configuration diagram showing a configuration of a vehicle audio system including an audio signal processing device according to a third embodiment of the present invention. In the figure, 3A
Is an electronic volume that amplifies the audio signal and outputs it to the distribution circuit 13 under the control of the microcomputer. Reference numeral 9 is a built-in A-D converter, which is a volume adjustment signal Cv and an adjustment signal Vr of a corresponding voltage. , A multiplier 10 that multiplies the signal Vp from the peak value analyzer 6 and the adjustment signal Vr from the microcomputer 9, 11 is a mixing circuit, and 12 is a volume adjustment signal Cv from the microcomputer 9. A bass range boost circuit that performs bass range boost processing on the output signal of the mixing circuit 11 and outputs the processed signal to the electronic volume 3A, and 14 is an operation unit for the user to input a volume adjustment instruction or the like.

It should be noted that the microcomputer 9 and the electronic volume 3A are an implementation example of the volume control means described in claim 3, and the multiplier 10 is an implementation example of the compression degree control means.

Next, the operation will be described. The audio reproducing device 1 reproduces a stereo signal from the audio recording medium and outputs it to the low pass filter 4 and the high pass filter 5. The low-pass filter 4 passes the low-frequency component of the input signal. High pass filter 5
Passes the high frequency component of the input signal.

The wave height analyzer 6 operates in the same manner as in the case of the first embodiment, and outputs the signal Vp of the voltage corresponding to the wave height value.
This signal Vp is input to the multiplier 10 in this case. The multiplier 10 multiplies the signal Vp from the wave height analyzer 6 and the adjustment signal Vr from the microcomputer 9 and outputs the multiplication result to the voltage control type amplifier 7 as a control voltage corresponding to the degree of waveform compression. The voltage controlled amplifier 7 is
The operation is performed in the same manner as in the first embodiment to compress the waveform of the audio signal in the low range.

As can be seen from the specific configuration shown in FIG. 6, the voltage control type amplifier 7 has a lower amplification factor as the control voltage increases. The adjustment signal Vr from the microcomputer 9 corresponds to the volume instruction amount input by the user. Therefore, even if there is no change in the signal Vp from the wave height analyzer 6, the degree of waveform compression is large if the volume instruction amount by the user is large. Even if there is no change in the volume instruction amount by the user, if the signal Vp from the wave height analyzer 6 is large, that is, if the peak value of the audio signal in the bass range is large, the degree of waveform compression is large. Then, if the crest value of the audio signal in the low frequency range is large and the volume instruction amount by the user is large, the degree of waveform compression is large.

The output of the voltage control type amplifier 7 and the output of the high pass filter 5 are mixed by the mixing circuit 11 and then sent to the bass boost circuit 12. Bass boost circuit 1
2 is configured as shown in FIG. 10, for example. Figure 10
Shown in is a bass boost circuit including an auto loudness correction function. Although only one channel is shown here, this circuit is actually provided for the left and right two channels.

In the bass boost circuit 12 shown in FIG. 10, the audio signal Lin from the mixing circuit 11 is applied to the non-inverting input terminal of the operational amplifier 71 which functions as a so-called buffer amplifier. The output of the operational amplifier 71 is
It is applied to the non-inverting input terminal of the operational amplifier 72 via the resistance element 75, and is also fed back to the inverting input terminal of the operational amplifier 71.

The output of the operational amplifier 72 is supplied to the electronic volume control 3A as the output of the bass boost circuit 12, and is fed back to the inverting input terminal through the resistance element 77. Further, between the inverting input terminal and the non-inverting input terminal of the operational amplifier 72, an analog switch 73 for receiving the volume adjustment signal Cv from the microcomputer 9 as a control input.
Is provided. The analog switch 73 has a resistance element 74.
And a movable terminal 73a grounded via a series body of an inductor 76 and fixed terminals 73b drawn out from a plurality of taps of a resistance element inserted between an inverting input terminal and a non-inverting input terminal. Has been done. The audio signal applied to the operational amplifier 72 is provided with a bass range boost characteristic determined by the resistance value, the inductance, and the negative feedback resistance value of the resistance element 77 according to the switch state of the analog switch 73.

The microcomputer 9 outputs the volume adjustment signal Cv indicating the volume instruction amount input by the user to the bass range boost circuit 12. The analog switch 73 of the bass boost circuit 12 changes the position of the movable terminal 73a according to the value indicated by the volume adjustment signal Cv. Therefore, since the resonance resistance value changes, the bass range boost characteristic changes. The output of the bass boost circuit 12 is given to the electronic volume 3A.

The electronic volume 3 A amplifies the input signal according to the value indicated by the volume adjustment signal Cv from the microcomputer 9 and outputs the amplified signal to the distribution circuit 13.
The distribution circuit 13 distributes the L signal from the electronic volume 3A to the power amplifiers 31 and 33, and distributes the R signal to the power amplifiers 32 and 34. Each power amplifier 31-34
Power-amplifies the input signal and outputs each speaker 21 to 24.
Output to.

As described above, the signal V from the wave height analyzer 6 is
When the product of p and the adjustment signal Vr is large, the degree of decrease in the amplification factor of the voltage controlled amplifier 7 is large, and when the product is small, the degree of decrease in the amplification factor is small. That is, when the crest value of the low frequency component of the audio signal is large and the level of the volume adjustment signal is high, the degree of waveform compression becomes higher. The audio signal processing device according to this embodiment also
Waveform compression is performed for music sources with excessive low-frequency components such as rock music sources, and waveform compression is not performed for classical music sources, thereby realizing processing suited to the music genre. Further, in this case, the processing is realized in consideration of the user's volume adjustment instruction. Further, in this case, since the electronic volume 3A for adjusting the volume is arranged in the latter stage of the entire circuit,
The deterioration of the SN ratio of the audio signal due to the presence of the audio signal processing device is small.

Example 4. FIG. 11 is a block diagram showing the configuration of an audio signal processing device according to the fourth embodiment of the present invention. In this case, delay units 15a and 15b are added between the low-pass filter 4 and the voltage controlled amplifier 7 in addition to the configuration shown in FIG. Further, delay units 15c and 15d are added between the high pass filter 5 and the mixing circuit 11. The delay units 15a to 15d are equivalent to those shown in the second embodiment, and the delay amount is an amount corresponding to one analysis period in the wave height analyzer 6. Here, a two-speaker system is taken as an example. Therefore, the distribution circuit 13 shown in FIG. 9 is not provided here.

Next, the operation will be described. The audio reproducing device 1 reproduces a stereo signal from the audio recording medium and outputs it to the low pass filter 4 and the high pass filter 5. The low-pass filter 4 passes the low-frequency component of the input signal. High pass filter 5
Passes the high frequency component of the input signal.

The wave height analyzer 6, the microcomputer 9 and the multiplier 10 operate in the same manner as in the third embodiment, and the multiplier 10 uses the voltage control type amplifier 7 as a control voltage corresponding to the degree of waveform compression. Output to. The voltage-controlled amplifier 7 operates in the same manner as in each of the above-described embodiments to compress the waveform of the bass audio signal.

The delay section 15a and the delay section 15b are
As in the case of the above embodiment, the output signal of the low-pass filter 4 is delayed by one analysis period and output to the voltage control type amplifier 7. The voltage-controlled amplifier 7 level-compresses the signals output by the delay units 15a and 15b according to the control voltage from the multiplier 10 according to the control voltage. In addition, the delay units 15c and 1
5 d delays the output signal of the high-pass filter 5 by one analysis cycle and outputs it to the mixing circuit 11.

Since the subsequent operation is the same as that of the third embodiment, the description of the subsequent operation will be omitted here. The audio signal processing device according to this embodiment exhibits the effects obtained by the device according to the third embodiment, and by delaying the audio signal by the time of waveform analysis,
The analysis result and the audio signal to which the analysis result is applied are matched to each other to provide a reproduced sound with less abnormal sound.

Example 5. FIG. 12 is a configuration diagram showing the configuration of an audio signal processing device according to the fifth embodiment of the present invention. In principle, in the audio signal processing device according to each of the above-mentioned embodiments, the waveform distortion is small and the sound quality does not change in the sound range other than the low range. However, when the low-pass filter 4 and the high-pass filter 5 that are analog circuits are used, the frequency band division accuracy may deteriorate due to variations in the constants of circuit components, and as a result, the sound quality may deteriorate. . On the other hand, if frequency band division or waveform compression is performed by digital signal processing using a digital circuit such as a digital filter, the processing accuracy depends on the operation word length etc. it can. The audio signal processing device according to this embodiment is constructed based on such an idea.

In FIG. 12, 80 is an AD converter for AD converting an audio signal from the audio reproducing apparatus 1, 81 is two quadratic mirror filters (QM).
F) A digital band division circuit having 81a and 81b for performing frequency band division. Here, the QMF 81a performs band division of the L signal, and the QMF 81b performs band division of the R signal. Reference numeral 88 is a digital wave height detector for detecting the peak value of the low frequency range component by digital processing, and 89 is a digital waveform compressor for performing waveform compression by digital processing.

Further, 83 is two QMFs 83a and 83b.
The output of the digital waveform compressor 89 and the QMF 81
a digital mixing circuit for mixing the outputs of a and 81b, 8
5 is a volume control for the output of the digital mixing circuit 83,
Digital processing circuits for performing processing such as distribution, and 86 and 87 are DA converters for converting the output of the digital processing circuit 85 into a power amplifier (not shown). 9
Reference numeral 0 is a controller that performs processing such as giving a volume adjustment signal to the digital processing circuit 85. Here, 4
A speaker system is assumed. This audio signal processing apparatus is a digital version of the audio signal processing apparatus according to the first embodiment, but it is important to use QMF for band division processing and mixing processing.

The digital band division circuit 81 is an implementation example of band division filter means, and the digital wave height detector 88 is an implementation example of waveform analysis means. The digital waveform compressor 89 is an implementation example of the waveform compression means, and the digital mixing circuit 83 is an implementation example of the mixing means.

Next, the operation will be described. The audio signal from the audio reproducing apparatus 1 is AD converted by the AD converter 80 and then applied to the digital band division circuit 81. In the digital band division circuit 81, the QM
The F81a performs band division of the L signal of the audio signal. Then, the low frequency output is sent to the digital wave height detector 88, and the high frequency output is sent to the digital mixing circuit 83. Also,
The QMF 81b performs band division of the R signal of the audio signal.

FIG. 13 shows a concrete configuration example of each QMF 81a, 81b. As shown in the figure, QMF 81a, 81
Reference numeral b is a target coefficient FIR filter that divides the band into two equal parts, and includes n (n is an odd number) delay devices 101 connected in cascade.
_{1 to} 101 _n , multipliers 103 ₀ to respectively connected to the output side and input terminal of each delay device 101 _{1 to} 101 _n
103 _n , adders 102 ₃ , 102 ₅ , ... For adding odd-numbered multiplication outputs (odd-numbered tap outputs)
102 _n , adders 102 ₂ , 102 ₄ , for adding even-numbered multiplication outputs (even-numbered tap outputs)
.., 102 _(n-1) , and the output of the adder 102 _n is the adder 1
02 _(n-1) subtractor 102 _H is subtracted from the output to generate a high frequency output of, as well as added to the output and the adder 102 _n outputs of the adder 102 _(n-1) generating a low-frequency output And an adder 102 _L for

[0093] In this case, the multiplication factor of the multiplier 103 ₀ ~103 _n is in turn _{a 0, a 1, ···,} a n,

The relationship of a _ni = a _i (i is a natural number) is satisfied.

In the QMF configured as described above, the high-frequency output and the low-frequency output are obtained by adding the odd-numbered tap outputs and the even-numbered tap outputs separately and taking out the results one by one. And can be obtained at the same time. And
The low band output is sent to the digital wave height detector 88, and the high band output is sent to the digital mixing circuit 83.

The digital wave height detector 88 outputs a control signal by the same processing as that of the wave height analyzer 6 shown in FIG. However, the processing is digital processing. Further, the digital waveform compressor 89 outputs a low frequency component compressed by the same process as the voltage control type amplifier 7 shown in FIG. However, the processing is digital processing.

Then, in the digital mixer 83,
The QMF 83a outputs the low-frequency component of the L signal output from the digital waveform compressor 89 and the L signal output from the QMF 81a.
Mix with the high frequency components of the signal. In addition, QMF83b
The low frequency component of the R signal output from the digital waveform compressor 89 and the high frequency component of the R signal output from the QMF 81b are mixed.

FIG. 14 shows a concrete configuration example of each QMF 83a, 83b. As shown in the figure, QMF83a, 83
b is a subtracter 101 _H for subtracting the high band output from the digital band division circuit 81 from the low band output from the digital waveform compressor 89, and delay units 101 ₁ connected in cascade at the output side of the adder 101 _H. ˜101 _{n −1} , a digital band division circuit 81 for the low frequency output from the digital waveform compressor 89.
Adder 101 _L and adder 10 for adding the high frequency output from the
Each delay device 101 ₁ to ₁ 1 connected in cascade at the output side of 1 _L
01 _n , the output of the adder 101 _H and each delay device 101 ₁
Multipliers 103 _{0 to} 103 _n respectively connected to every other output of ˜101 _n , adders 102 _{1 to} 102 _n for adding each multiplication output, and adder 102 ₁
It is composed of a coefficient unit 104 that halves the added value by 102 _n .

The multiplication coefficients of the multipliers 103 _{0 to} 103 _n are a ₀ , a ₁ , ..., An in _order , and satisfy the relationship of a _ni = a _i (i is a natural number).

The QMFs 83a and 83 configured as described above
If a signal obtained by interpolating "0" every other input digital signal and returning to the original sampling frequency is input to b, the digital signal in which the high frequency component and the low frequency component are combined is restored. It

The digital processing circuit 85, after adjusting the volume of the signal from the digital mixing circuit 83,
The signal is distributed to the DA converter 86 and the DA converter 87. The D-A converters 86 and 87 are the digital processing circuit 8
5 output is D-A converted and supplied to a power amplifier (not shown)

As described above, since the symmetric QMF is used for the filter for dividing the frequency band of the audio signal and the filter for synthesizing the high frequency band and the low frequency band, signal processing with high restoration accuracy is realized. . According to such a configuration, for a signal having no extreme fluctuation in the frequency spectrum, such as a pop type or classic type audio signal, the compression process is not executed and the deterioration in band division and synthesis does not occur. Faithful reproduction is realized as if no signal processing circuit were present. When an analog filter is used as a filter for frequency band division, even if compression processing is not performed, signal deterioration during band division and synthesis is often unavoidable, so reproduced sound is slightly degraded. I often end up doing it.

Example 6. In the audio signal processing device according to the fifth embodiment, as shown in FIGS. 13 and 14, since a large number of delay devices and multipliers are required,
Filtering takes a considerable amount of time. For example, it is expected that the processed signal will be delayed by 100 ms or more with respect to the input signal.

Therefore, when giving priority to obtaining a simplified circuit even if the fidelity of waveform transmission is slightly sacrificed,
The IIR filter shown in FIG. 15 may be used instead of the QMFs 81a and 81b. In addition, the overall configuration is the same as that shown in FIG. 12, except that the QMFs 81a, 81b, 83a, 8
3b is removed.

As shown in FIG. 15, delay devices 101 ₀ to 1 ₀
01 ₄ , multipliers 103 _{1 to} 103 ₄ and adder 102
The number of is significantly reduced. Therefore, the calculation time is significantly shortened, and can be shortened to 100 ms or less, for example. Further, the digital mixing circuit 85 is realized by a simple adder. In this way, the scale and cost of the digital signal processing circuit is significantly reduced.

Example 7. FIG. 16 is a block diagram showing the arrangement of an audio signal processing apparatus according to the seventh embodiment of the present invention. This audio signal processing circuit differs from that shown in FIG. 12 in that digital delay devices 82a to 82d are inserted between a digital band division circuit 81, a digital waveform compressor 89 and a digital mixing circuit 83. is there. The delay amount of the digital delay devices 82a to 82d is
The processing time of the digital wave height detector 88 (time from signal input to processing result output) is set to be equal.

The frequency band division by the digital band division circuit 81, the wave height detection processing by the digital wave height detector 88, the waveform compression processing by the digital waveform compressor 89, the mixing processing by the digital mixer 83, etc. are the same as those in the fifth embodiment. This is the same as the operation in the device. The digital delay devices 82a to 82d are the delay devices 15 in the device according to the second embodiment and the device according to the fourth embodiment.
It operates similarly to a to 15d. Therefore, as in the case of the second and fourth embodiments, the analysis result and the audio signal to which the result is applied are matched, and it is possible to provide a reproduced sound with less abnormal sound.

Example 8. FIG. 17 is a block diagram showing the arrangement of an audio signal processing apparatus according to the eighth embodiment of the present invention. In the apparatus shown in FIG. 12 and the apparatus shown in FIG. 16, the digital band dividing circuit 81, the digital wave height detector 88, the digital waveform compressor 89, and the digital mixing circuit 85 were each constituted by individual digital circuits. Further, in the device shown in FIG. 16, the digital delay devices 82a to 82d are each composed of individual digital circuits. On the other hand, in the audio signal processing device according to this embodiment, those circuits are realized by a digital signal processing integrated circuit (hereinafter referred to as DSP).

The DSP 100 realizes the functions of the digital band division circuit 81, the digital wave height detector 88, the digital waveform compressor 89, the digital mixing circuit 85, and the digital delay devices 82a to 82d by a processing procedure according to a program. Therefore, the operation of the audio signal processing device is the same as the operation of the audio signal processing device according to the fifth embodiment shown in FIG. 12 or the operation of the audio signal processing device according to the seventh embodiment shown in FIG. Detailed description of the operation is omitted here.

As described above, the digital processing is performed by the DSP 10
By integrating 0, the circuit is simplified. Further, the controller 90 of the microcomputer controls the adjustment processing such as volume adjustment and equalizing other than the waveform compression, and the adjustment signal processing under the control is also performed by the DSP 1.
If 00 is performed, the circuit can be further simplified. When the digital signal delay devices 82a to 82d are not used, that is, when the audio signal processing device according to the fifth embodiment shown in FIG. 12 is integrated in the DSP 100, the digital signal delay devices 82a to
No program is required to implement the functionality of 82d.

Example 9. FIG. 18 is a block diagram showing the arrangement of an audio signal processing apparatus according to the ninth embodiment of the present invention. In this audio signal processing device, the digital delay devices 82a to 82d are realized by the signal delay processing means 101 which is an external memory of the DSP 100. Therefore, DS
The P100 executes only the functions of the digital band division circuit 81, the digital wave height detector 88, the digital waveform compressor 89, and the digital mixing circuit 85, that is, the arithmetic processing function (processing other than the delay function).

In this way, by separating the part that implements the arithmetic processing function and the part that implements the delay function, D
It is possible to reduce the program of the SP100, and as a result, it is possible to prevent the DSP100 from increasing in size. Therefore, the circuit scale as a whole becomes small, and a more suitable one can be obtained. Note that the capacity of the memory that constitutes the signal delay processing means (memory) 101 is the capacity of storing the waveform samples in one analysis period, which is the time length required for the crest value analysis, and in the range of 5 ms to 100 ms in practice. Good thing.

Next, the specific contents of the digital signal processing from the start of the crest value analysis to the completion of the waveform compression processing will be described. The following processing is performed by the DSP shown in FIG.
Assuming that 100 and the memory 101 are used, FIG.
It can also be applied to the one using the DSP 100 shown in FIG. Further, it can be applied to the audio signal device shown in FIG.
Furthermore, temporary storage processing (delay processing) in the following processing
Can be applied to the audio signal processing device shown in FIG.

First, the overall flow of digital signal processing will be described with reference to the flowchart of FIG. When the audio reproducing device 1 starts reproduction, AD
The converter 80 performs AD conversion of the audio signal and outputs the audio signal as sample waveform data. DS
P100 takes in the waveform data (step ST1
1), band division processing by QMF or the like is performed (step S
T12). Then, the low-pitched waveform data and the high-pitched waveform data after the band division are written in the memory 101 (steps ST13 and ST14). In the audio signal processing device shown in FIG. 16, the digital band division circuit 81
Performs steps ST11 and ST12. Also, A
The waveform data from the -D converter 80 is sent to the digital delay unit 8
2a, 82d.

The DSP 100 analyzes the peak value of each waveform data of the bass component (step ST15), and calculates the compression rate according to the peak value which is the analysis result (step ST16). In the audio signal processing device shown in FIG. 16, the digital crest value detector 88 operates in step ST.
15, the processing of ST16 is performed.

Next, the DSP 100 performs a compression process on the waveform data in the low tone range in the memory 101 according to the compression rate (step ST17). In the audio signal processing device shown in FIG. 16, the digital waveform compressor 83 performs the process of step ST17 on the waveform data from the digital delay devices 82a and 82b.

Then, the DSP 100 outputs the waveform data after compression and the waveform data in the high frequency range in the memory 101 to the QMF.
And the like (step ST18), and the combined waveform data is output to the DA converter 86 (step ST1).
9). In the audio signal processing device shown in FIG. 16, the digital mixing circuit 83 performs steps ST18 and ST19 on the compressed waveform data and the treble waveform data from the digital delay units 82c and 82d.

Here, each waveform data after band division is temporarily stored in the memory 101 or delayed by the digital delay devices 82a to 82d, and waits for the time from the peak value analysis start to the waveform compression start. Will be done. Therefore, the waveform data to be compressed is surely compressed. Such a method of waiting for data to be processed is well known in the field of speech analysis and synthesis as a kind of predictive analysis method. However, if the predictive division method is applied to the signal processing method that leads to the system optimization of the audio reproduction system, a unique effect that cannot be obtained by the conventional system is produced.

A temporal transition of a signal having a specific waveform example along the above processing will be described with reference to the timing chart of FIG. From the AD converter 80, the sample waveform acquisition start time (t = 0) shown in (h) is used as a starting point (a).
It is assumed that the waveform data shown in is output. Then, by the band division processing, waveform data in the low range shown in (b) and waveform data in the high range shown in (c) are obtained. At this time, it is assumed that the band division process takes τ ₁ .

Then, as shown in (i) and (j), 1
The peak value analysis is performed for the waveform data in the low frequency range of the frame. Therefore, the processing time is τ ₁ to τ ₂ .
The compression ratio is calculated at the timing shown in (k) according to the analysis result.

Memory 101 or digital delay device 8
From 2a and 82b, as shown in (d), low-frequency waveform data delayed by one frame is output. Waveform compression is performed on the waveform data at the timing shown in (l). Waveform compression is executed for each waveform data. Therefore, the compressed waveform data is output as shown in (f). The memory 101 or the digital delay devices 82c and 82d output waveform data in the high frequency range delayed by one frame as shown in (e). Therefore, waveform-compressed low-pitched waveform data and high-pitched waveform data are combined at the timing shown in (m),
The combined data is output as shown in (g).

Thereafter, the above processing is repeated for each frame. Here, the time τ ₁ required for band division is preferably as short as possible, but it takes a considerable time when the QMF shown in FIGS. 13 and 14 is used. When the IIR filter is used, it can be reduced to 1/10 or less as compared with the case where the QMF is used. Therefore, in that case, the delay of the audio signal in the path from the audio reproducing device 1 to the power amplifier is substantially equal to the delay in the memory 101, and is about 100 ms at the maximum.

Example 10. Next, a more effective predictive analysis method for waveform compression will be described with reference to specific examples shown in FIGS. 21 and 22. FIG. 21 is a flowchart showing the analysis procedure, which corresponds to the analysis compression processing in FIG. 22A and 22B are waveform diagrams showing examples of audio signal waveforms. FIG. 22A shows a waveform before processing and FIG. 22B shows a waveform after processing. It is assumed here that the audio signal processing device shown in FIG. 18 is used, but this method is also applicable to the device shown in FIG. 16 or FIG.

The DSP 100 sets the waveform data in the low tone range from the zero point (or a point close to it) of the peak value to the next zero point as one frame of data. That is, the analysis time length is set between adjacent zero points in the digital audio waveform. Therefore, when the DSP 100 detects the zero point from the waveform data in the low frequency range, the DSP 100 starts the analysis of the peak value of one frame.

The zero point is easily obtained as a point where the sign of the sample waveform data is inverted. In the example shown in FIG.
The analysis time length is between T _{0 and} T _1, between T _{1 and} T ₂ , and so on. The DSP 100 determines whether there is waveform data that exceeds the threshold level V _S for the waveform data within one analysis time (step ST22). The threshold level V _S is a predetermined value based on the linear region characteristic of the audio system and the like. The point in time at which waveform data that exceeds the threshold level V _S is generated is designated as T _S point.

[0126] When the T _S point is present, a peak value from among these points are extracted point of maximum (maximum absolute value),
The peak value V _P at that point is obtained (step ST23,
ST24). Then, the gain coefficient G corresponding to the compression rate is calculated as | V _S / V _P | (step ST25).
On the other hand, if the T _S point does not exist, it means that the compression is not necessary, and thus the gain coefficient G = 1 is set (steps ST23 and ST26). Note that steps ST22 to ST
In the audio signal processing device shown in FIG. 16, the processing of 26 is executed by the digital wave height detector 88.

The DSP 100 reads out the waveform data of the low frequency range for one frame from the memory 101 after the lapse of the staying time described later. Then, each waveform data is multiplied by the gain coefficient G to create compressed low-pitched waveform data. Note that the data in the memory 101 may be multiplied by the gain coefficient G to read the compressed low-pitched waveform data after the residence time has elapsed. Further, in the audio signal processing device shown in FIG. 16, the process of step ST27 is executed by the digital waveform compressor 83.

In the waveform shown in FIG. 22A, T _{2 to} T ₃
There is waveform data that exceeds the threshold level V _S between
The maximum value V _P occurs at time T _P. There is no waveform data exceeding the threshold level V _S between the other zero points. Accordingly, the waveform compression is performed only between T ₂ through T _3, as shown in FIG. 22 (b), T ₂ ~T
Waveform data whose maximum peak value during ₃ is V _S is output.

By thus setting the analysis time length between the zero points, continuity is maintained and waveform distortion is suppressed in the compressed waveform. Note that, in reality, the waveform shown in FIG. 22 (b) is better than the waveform shown in FIG.
1 dwell time or digital delay 8
It is delayed by the delay amount of 2a to 82d. Here, for example, the lowest frequency component included in the audio signal is 20 Hz.
Assuming that, the zero point of the sine wave of that frequency is 25 ms. Therefore, if the delay amount is set to 25 ms, there is no frame that extends for a period exceeding the delay amount, so that there is no problem in waveform analysis.

Example 11. In the waveform compression process, in order to further reduce the waveform distortion in the compression process, the analysis time length may be set from one zero point to the next next zero point, as shown in FIG. That is, of the points where the waveform crosses the zero level, one frame is defined between adjacent zero points having the same cross direction (positive → negative or negative → positive). Note that FIG. 23A shows a waveform before processing, and FIG.
Shows the waveform after processing. In this case, the DSP 100 or the digital wave height detector 88 targets the waveform data from the zero point to the next and next zero point for the peak value detection and the gain coefficient calculation. The specific processing procedure is the tenth
This is the same as the case of the above embodiment.

In this case, one section to be subjected to waveform compression includes the positive and negative regions of the waveform, that is, a section corresponding to one wavelength. Therefore, as compared with the case of the tenth embodiment. Is spreading. However, by setting the analysis time in this way, harmonic distortion due to waveform compression can be further reduced. However, the memory 101
The capacity is increased about twice as much as that of the tenth embodiment.

Example 12 The waveform compression processing according to each of the above-described embodiments is processing by a so-called variable length frame in which the analysis time length changes depending on the difference in waveform form. However, the analysis time can also be according to a fixed length frame. The fixed length analysis time for fixed length frames is
As described above, the value is preferably 5 ms to 100 ms. The processing contents will be described below with reference to the flowchart of FIG. 24 and the waveform diagram of FIG. 25. Here, description will be made assuming the audio signal processing device shown in FIG. However, this method can also be applied to the devices shown in FIGS.

When the audio reproducing apparatus 1 starts reproduction, the DSP 100 starts band division processing, peak value analysis processing and the like. The DSP 100 sets the threshold level V _S and sets the previous frame gain coefficient G ₋₁ to “1” at the time of initial setting. In peak value analysis processing, DS
P100 receives the waveform data for one frame from the band processing portion (step ST31). Then, the zero point is detected from each waveform data (step ST32). The start point of one frame is called the start point and the end point is called the end point. The zero point is set to T _i (i ≧ 0). And D
The SP 100 multiplies the waveform data from the start point to T ₀ by the gain coefficient G _-1 (step ST33).

Next, the DSP 100 detects whether or not the waveform data from the point T ₀ to the terminal point exceeds the threshold level V _S (step ST34). The time point at which waveform data exceeding the threshold level V _S exists is T _S
And When there is a T _S point, extracts a V _P (the maximum value of the absolute value) maximum value of the waveform data at those points (step ST35,37). Then, | V _S / V _P | is set as the gain coefficient G corresponding to the compression rate (step ST38). If there is no T _S point, “1” is set, which means that compression is not performed (step ST
36).

Then, each waveform data from the point T ₀ to the terminal point is multiplied by the gain coefficient G (step ST39). Further, the gain coefficient G of the current frame is set as the frame gain coefficient G _-1 used in the next frame (step ST40). For example, when the waveform shown in FIG. 25 (a) is present, the waveform shown in FIG. 25 (b) is obtained by the above processing. That is, the waveform from the T ₀ point to the terminal point is compressed to about 80%. Then, the compression rate applied from the point T ₀ to the terminal point is applied up to the first zero point T ₁ of the next frame. Note that, in reality, FIG.
The waveform shown in (b) lags behind the waveform shown in FIG.

As described above, by detecting the zero points and processing the zero points with the same compression rate, the continuity of the waveform can be maintained and the waveform distortion can be minimized, as in the case of the above embodiments. . Therefore, the deterioration of sound quality is small. Furthermore, since the analysis time length is fixed, the delay time of the waveform data in the DSP 100 is the same as the analysis time length. Therefore, the control regarding the input / output of the sample waveform is simplified, and the circuit configuration is simplified.

[0137]

As described above, according to the first aspect of the present invention, the audio signal processing device analyzes the peak value of only the low-pitched signal of the audio signal, and determines the wave length of the low-pitched signal of the audio signal. Since the level of the bass signal is compressed according to the high value, there is an effect that a suitable listening sound can be provided to each of various music sources of different genres. Furthermore, since the processing adapted to the audio signal waveform is performed, there is an effect that faithful reproduction is guaranteed for a music source that does not require compression or under a listening condition that does not require compression.

According to the second aspect of the present invention, the audio signal processing device analyzes the peak value of only the low-pitched signal of the audio signal, and lowers the peak value of the low-pitched region of the audio signal according to the peak value. The invention according to claim 1, wherein the signal from the band-splitting filter means is delayed so that the analysis result is reflected in the waveform to be analyzed, in contrast to the configuration in which the level of the signal in the range is compressed. In addition to the effect of, the effect that more strict waveform compression processing can be performed is also obtained.

According to the third aspect of the invention, the audio signal processing device adjusts the volume of the audio signal, and the degree of compression that reflects the adjustment degree of the volume adjusting means in the analysis result of the waveform analyzing means. Since the control means is further provided, there is an effect that the waveform compression processing in consideration of the volume adjustment command of the user can be performed. Further, a volume adjusting portion, which is an element for greatly changing the audio signal level, can be arranged in the latter stage of the device, and there is an effect that the deterioration of the SN ratio due to the presence of this device can be suppressed.

According to the invention described in claim 4, since the audio signal processing device has a configuration in which the band division filter means, the waveform analysis means, and the waveform compression means are constituted by a digital processing circuit, each of the above effects is achieved. In addition to,
The effect that band division and waveform compression can be performed accurately can also be obtained.

According to the fifth aspect of the invention, the audio signal processing device has a structure in which the band division filter means is a digital mirror filter.
Band division is performed more precisely, and when processing signals with small fluctuations in the frequency spectrum, such as pop and classical audio signals, and when compression processing is not necessary, the fidelity as if this device does not exist. There is an effect that various signal reproduction is executed.

According to the sixth aspect of the invention, the audio signal processing device has a structure in which the band division filter means, the waveform analysis means, and the waveform compression means are digital signal processing integrated circuits. There is an effect that a small scale and low price can be obtained.

According to the invention described in claim 7, since the audio signal processing device has a configuration in which the delay amount in the delay means is set to 5 milliseconds to 100 milliseconds, reproduction due to waveform compression or signal delay is performed. There is an effect that a thing that can suppress the unnaturalness of sound can be obtained.

According to the eighth aspect of the invention, in the audio signal processing device, the waveform analysis means performs the waveform analysis in units of zero-cross points of the waveform. There is an effect that the continuity is maintained in the waveform of the signal and the waveform distortion can be suppressed.

According to the ninth aspect of the invention, the audio signal processing device has a configuration in which the waveform analysis means performs waveform analysis in units of each waveform within a fixed time. Is simplified, and as a result, the circuit configuration is simplified.

[Brief description of drawings]

FIG. 1 is a configuration diagram showing a configuration of a vehicle audio system including an audio signal processing device according to a first embodiment of the present invention.

FIG. 2 is an explanatory diagram showing a frequency-sound pressure level curve.

FIG. 3 is a distribution diagram showing a spectrum distribution of a classic type or pop type audio signal.

FIG. 4 is a distribution diagram showing a spectrum distribution of a rock music signal.

FIG. 5 is a circuit diagram showing a configuration example of a wave height analyzer.

FIG. 6 is a circuit diagram showing a configuration example of a voltage control type amplifier.

FIG. 7 is a waveform diagram showing a changing process of a waveform of an audio signal.

FIG. 8 is a block diagram showing a configuration of an audio signal processing device according to a second embodiment of the present invention.

FIG. 9 is a configuration diagram showing a configuration of a vehicle audio system including an audio signal processing device according to a third embodiment of the present invention.

FIG. 10 is a circuit diagram showing a configuration example of a bass boost circuit.

FIG. 11 is a block diagram showing a configuration of an audio signal processing device according to a fourth embodiment of the present invention.

FIG. 12 is a block diagram showing a configuration of an audio signal processing device according to a fifth embodiment of the present invention.

FIG. 13 is a circuit diagram showing a configuration example of a quadratic mirror filter used for band division.

FIG. 14 is a circuit diagram showing a configuration example of a quadratic mirror filter used for signal combination.

FIG. 15 is a circuit diagram showing a configuration example of an IIR filter.

FIG. 16 is a block diagram showing the structure of an audio signal processing device according to a seventh embodiment of the present invention.

FIG. 17 is a block diagram showing the structure of an audio signal processing device according to an eighth embodiment of the present invention.

FIG. 18 is a block diagram showing the structure of an audio signal processing device according to a ninth embodiment of the present invention.

FIG. 19 is a flowchart showing processing of waveform analysis and waveform compression.

FIG. 20 is a waveform diagram showing a process of waveform analysis and waveform compression.

FIG. 21 is a flow chart showing waveform analysis processing according to the tenth embodiment of the present invention.

FIG. 22 is a waveform chart showing the result of waveform analysis processing in the tenth embodiment of the present invention.

FIG. 23 is a waveform chart showing the result of waveform analysis processing in the eleventh embodiment of the present invention.

FIG. 24 is a flowchart showing waveform analysis processing according to the twelfth embodiment of the present invention.

FIG. 25 is a waveform chart showing the result of waveform analysis processing in the twelfth embodiment of the present invention.

FIG. 26 is a configuration diagram showing a configuration of a vehicle audio system including a conventional audio signal processing device.

FIG. 27A is a spectrum diagram showing a frequency spectrum of a hard rock music signal. (B) is a waveform diagram showing a time waveform of a hard rock music signal.

FIG. 28 is a block diagram showing a conventional improved audio signal processing device.

FIG. 29 is a block diagram showing a configuration of a conventional waveform compression circuit.

FIG. 30 is a block diagram showing the configuration of another conventional waveform compression circuit.

[Explanation of symbols]

3A electronic volume (volume control means) 4 low pass filter (band division filter means) 5 high pass filter (band division filter means) 6 wave height analyzer (waveform analysis means) 7 voltage control type amplifier (waveform compression means) 8 Mixing / distributing circuit (mixing means) 9 Microcomputer (volume adjusting means) 10 Multiplier (compression degree controlling means) 15a to 15d Delay unit (delaying means) 81 Digital band division circuit (band division filter means) 81a to 81b QMF (digital) Mirror filter) 83 Digital mixing circuit (mixing means) 88 Digital wave height detector (waveform analysis means) 89 Digital waveform compressor (waveform compression means) 100 DSP (digital signal processing integrated circuit)

─────────────────────────────────────────────────── ─── Continuation of the front page (51) Int.Cl. ⁵ Identification code Office reference number FI technical display location H04S 1/00 D 8421-5H

Claims (9)

[Claims] 1. A band division filter means for band-dividing an audio signal into a bass range signal and a treble range signal, and a waveform analysis means for analyzing a peak value of the bass range signal obtained by the division by the band division filter means. A waveform compression means for compressing the level of the waveform of the bass range signal according to the analysis result of the waveform analysis means, and a mixing means for mixing the bass range signal compressed by the waveform compression means and the treble range signal. Equipped audio signal processing device. 2. A band division filter means for band-dividing an audio signal into a bass range signal and a treble range signal, and a waveform analysis means for analyzing a peak value of the bass range signal obtained by the division by the band division filter means. A delay means for giving a delay of a time including a time required for the analysis processing of the waveform analysis means to the low range signal and the high range signal, and a low level delayed by the delay means according to an analysis result of the waveform analysis means. An audio signal processing apparatus comprising waveform compression means for compressing the level of the waveform of a tone range signal, and mixing means for mixing the low tone range signal compressed by this waveform compression means and the high tone range signal delayed by the delay means. . 3. The volume adjusting means for adjusting the level of an audio signal, and the compression degree controlling means for reflecting the degree of adjustment by the volume adjusting means on the analysis result of the waveform analyzing means. Audio signal processor. 4. Band division filter means, waveform analysis means,
The audio signal processing device according to any one of claims 1, 2 and 3, wherein the waveform compression means is composed of a digital processing circuit. 5. The audio signal processing apparatus according to claim 4, wherein the band division filter means is a digital mirror filter. 6. Band division filter means, waveform analysis means,
The audio signal processing apparatus according to claim 4, wherein the waveform compression means is a digital signal processing integrated circuit. 7. The audio signal processing device according to claim 4, wherein the delay amount in the delay means is 5 milliseconds to 100 milliseconds. 8. The audio signal processing apparatus according to claim 4, wherein the waveform analysis means performs waveform analysis with a unit between zero cross points of the waveform as a unit. 9. The audio signal processing device according to claim 4, wherein the waveform analysis means performs waveform analysis using each waveform within a fixed time as a unit.