JP2004184472A - Signal interpolation device, sound reproducing device, signal interpolation method, and program - Google Patents

Abstract

<P>PROBLEM TO BE SOLVED: To provide a signal interpolation device etc., for restoring a signal close to a source signal with small distortion from a modulated wave obtained by using a signal with the band of the source signal limited or restoring an audio signal with high sound quality. <P>SOLUTION: According to data for control made to correspond to an input audio signal, a control part 3 determines a passing band characteristic of an HPF 6 and the frequency of a local oscillation signal. A local oscillation part 4 sends the local oscillation signal of the frequency determined by the control part 3 to a mixing part 5. The mixing part 5 mixes the input audio signal and local oscillation signal with each other. The HPF 6 filters the signal generated by the mixing part 5 with the passing band characteristic determined by the control part 3. A component having passed through the HPF 6 has its amplitude adjusted by a gain control part 7 and is then added by an addition part 8 to the input audio signal to generate an output audio signal. <P>COPYRIGHT: (C)2004,JPO&NCIPI

Description

[0001]
TECHNICAL FIELD OF THE INVENTION
The present invention relates to a signal interpolation device, a sound reproduction device, a signal interpolation method, and a program.
[0002]
[Prior art]
In recent years, audio data representing audio such as music is distributed via a network such as the Internet or recorded on a recording medium such as an MD (Mini Disk) for use. Audio data distributed over a network or recorded on a recording medium generally has a certain volume of music or the like to be supplied in order to avoid an increase in the amount of data and an increase in occupied bandwidth due to an excessively wide band. Components above the frequency have been removed.
For example, in the audio data of the MP3 (MPEG1 audio layer 3) format, a frequency component of about 16 kHz or more is removed. Further, in ATRAC3 (Adaptive Transform Acoustic Coding 3) audio data, a frequency component of about 14 kHz or more is removed.
[0003]
As described above, music or the like from which a frequency component equal to or more than a certain value has been removed usually has lower sound quality than original music or the like. Therefore, it is conceivable to add a signal in place of the removed frequency component. As a technique for this, there is a technique disclosed in Patent Document 1.
[0004]
[Patent Document 1]
JP-A-7-93900
[0005]
The technique disclosed in Patent Literature 1 reduces distortion by multiplying an output audio signal obtained by passing a PCM (Pulse Code Modulation) digital audio signal through a low-pass filter by a signal including an absolute value component of the output signal. It is a technique of causing it.
[0006]
As another method, a method of extrapolating a spectrum envelope to a portion from which spectral components have been removed and adding noise along the extrapolated envelope may be considered.
[0007]
[Problems to be solved by the invention]
However, an audio signal representing music generally has harmonics of an electronic musical instrument or a human voice (vocal), and as shown in the spectrum of FIG. It has a spectrum distribution including many peaks arranged at intervals of about Hertz (100 Hertz or more and less than 1 KHz). Also, in the case of a voice signal representing voice transmitted via a telephone line, generally, components of 4 kHz or more show a spectrum distribution having similar characteristics.
[0008]
For this reason, even if the audio signal reproducing apparatus disclosed in Patent Document 1, which merely generates a harmonic by distorting the waveform of the low-frequency component of the output audio signal using an absolute value circuit or the like, the spectrum envelope cannot be obtained. Even with the above-described method of extrapolating a line and adding noise, it is not possible to obtain a signal having a spectrum close to the spectrum shown in FIG.
[0009]
SUMMARY OF THE INVENTION The present invention has been made in view of the above-mentioned circumstances, and has been made in consideration of the above-described circumstances. It is an object to provide a device, a sound reproducing device, and a signal interpolation method.
Another object of the present invention is to provide a signal interpolation device, a sound reproduction device, and a signal interpolation method for restoring an audio signal with high sound quality.
[0010]
[Means for Solving the Problems]
In order to achieve the above object, a signal interpolation device according to a first aspect of the present invention includes:
A frequency conversion unit that generates an interpolation signal by frequency-converting an interpolated signal representing a voice whose spectrum is to be interpolated,
A filter for extracting a component in a second band on a higher frequency side than the first band occupied by the signal to be interpolated among the interpolation signals;
An addition unit that generates an output signal representing the sum of the interpolated signal and the component extracted by the filter.
It is characterized.
[0011]
The frequency conversion unit may perform frequency conversion such that a component having a frequency equal to or higher than a predetermined frequency in the signal to be interpolated is frequency-converted into the second band.
[0012]
The predetermined frequency may be, for example, a frequency equal to or higher than a lower limit of a region where the distribution of the spectrum of the interpolated signal has a predetermined characteristic.
Specifically, the predetermined feature of the region is, for example, a feature that the region includes a plurality of peaks arranged at intervals of 100 Hz or more and less than 1 kHz while attenuating as the frequency increases. Should be fine.
[0013]
The predetermined frequency may be, for example, a frequency of 10 kHz or more.
[0014]
The interpolated signal may represent, for example, a voice transmitted via a telephone line.
[0015]
The filter specifies the range of the first band based on control data supplied from the outside, and changes the range of the second band to be on the higher frequency side than the range of the first band. May be performed.
[0016]
The control data may include information indicating a data format of the interpolated signal, and the filter may specify the range of the first band based on the information.
[0017]
The pass band characteristic of the filter may have a peak, and in this case, the second band may belong only to the high frequency side from the peak.
[0018]
The filter may be composed of, for example, an IIR (Infinite Impulse Response) type or FIR (Finite Impulse Response) type high-pass filter or band-pass filter.
[0019]
The filter may be such that the value of Q is determined based on control data supplied from the outside, and the passband characteristic is changed so as to have the determined value of Q.
[0020]
The control data may include genre information indicating a genre of music represented by the interpolated signal, and the filter may determine a value of Q based on the genre information.
[0021]
The filter may be such that the value of Q is determined in a range of 1 or more and 7 or less, for example.
[0022]
The adding unit may include a delay unit that delays the interpolated signal so that the component is substantially in phase with the component extracted by the filter, wherein the component extracted by the filter and the delay unit are delayed. The output signal representing the sum of the interpolated signals may be generated.
[0023]
Further, the sound reproducing device according to the second aspect of the present invention includes:
A frequency conversion unit that generates an interpolation signal by frequency-converting an interpolated signal representing a voice whose spectrum is to be interpolated,
A filter for extracting a component in a second band on a higher frequency side than the first band occupied by the signal to be interpolated among the interpolation signals;
An adding unit that generates an output signal representing a sum of the interpolated signal and a component extracted by the filter,
Sound reproducing unit that reproduces the sound represented by the output signal generated by the adding unit,
It is characterized.
[0024]
The signal interpolation method according to the third aspect of the present invention includes:
Generate an interpolation signal by frequency-converting an interpolated signal representing a voice whose spectrum is to be interpolated,
Of the interpolation signal, to extract a component in a higher frequency band than the band occupied by the signal to be interpolated,
Generating an output signal representing the sum of the interpolated signal and the extracted component;
It is characterized.
[0025]
Further, a program according to a fourth aspect of the present invention includes:
Computer
A frequency conversion unit that generates an interpolation signal by frequency-converting an interpolated signal representing a voice whose spectrum is to be interpolated,
Among the interpolation signals, a filter for extracting a component in a band on a higher frequency side than a band occupied by the interpolated signal,
An adding unit that generates an output signal representing a sum of the interpolated signal and a component extracted by the filter,
It is characterized in that it is intended to function as
[0026]
BEST MODE FOR CARRYING OUT THE INVENTION
Hereinafter, a signal interpolation device according to an embodiment of the present invention will be described with reference to the drawings, taking a high-frequency signal interpolator as an example.
[0027]
FIG. 1 is a diagram showing a configuration of a high-frequency signal interpolator according to an embodiment of the present invention. As shown in the figure, the high-frequency signal interpolator includes an input unit 1, a delay unit 2, a control unit 3, a local oscillation unit 4, a mixing unit 5, an HPF (high-pass filter) 6, a gain adjustment unit. 7 and an adder 8.
[0028]
The input unit 1 includes a recording medium drive, a processor such as a CPU (Central Processing Unit), and an input device such as a touch panel. The recording medium drive device may be, for example, an MD (Mini Disc) drive, a CD (Compact Disc) drive, or the like. The recording medium drive and the input device of the input unit 1 are connected to the processor of the input unit 1.
[0029]
The input unit 1 accesses the recording medium according to a user operation or the like while the recording medium is mounted on the recording medium, and reads data recorded on the recording medium.
Hereinafter, it is assumed that input sound data representing sound and control data associated with the input sound data are recorded in advance on a recording medium mounted on the input unit 1.
[0030]
Specifically, the input audio data is, for example, data in the form of MP3 (MPEG1 audio layer 3) or data in the form of ATRAC3 (Adaptive Transform Acoustic Coding 3).
[0031]
An audio signal represented by the input audio data (hereinafter, referred to as an input audio signal) removes, from the original audio, a component having a frequency equal to or higher than a certain value, as schematically shown in FIG. It is assumed that it has a spectrum distribution corresponding to the spectrum distribution.
This constant value indicating the upper limit of the occupied band of the input audio signal (ie, “f” in FIG. 2A)_INIs about 16 kHz when the input audio data is composed of MP3 format data, and is about 14 kHz when the input audio data is composed of ATRAC3 format data.
[0032]
The control data is information indicating the format (for example, MP3 format, ATRAC3 format, etc.) of the input audio data associated with the control data, and it is assumed that the input audio data represents music. It contains information indicating the genre of music (eg, classical, rock, jazz, etc.). When the recording medium on which the control data is recorded is an MD, a part or all of the control data may be included in, for example, UTOC (User Table Of Contents).
[0033]
When the input unit 1 reads the input audio data and the control data associated with each other from the recording medium, the input unit 1 supplies the input audio signal represented by the read input audio data to the delay unit 2 and the mixing unit 5, The read control data is supplied to the control unit 3.
[0034]
Each of the delay unit 2, the control unit 3, the local oscillation unit 4, the mixing unit 5, the HPF 6, the gain adjustment unit 7, and the addition unit 8 is a processor such as a DSP (Digital Signal Processor) or a CPU, or a dedicated integrated circuit. It is composed of However, part or all of the functions of the delay unit 2, the control unit 3, the local oscillation unit 4, the mixing unit 5, the HPF 6, the gain adjustment unit 7, and the addition unit 8 may be performed by a single processor.
[0035]
FIG. 3 is a diagram illustrating a logical configuration of the delay unit 2, the local oscillation unit 4, and the HPF 6. FIG. 3 also shows a mutual connection relationship between the delay unit 2, the local oscillation unit 4, the mixing unit 5, the HPF 6, the gain adjustment unit 7, and the addition unit 8.
[0036]
When the input audio signal is supplied from the input unit 1, the delay unit 2 delays the input audio signal and supplies the input audio signal to the adding unit 8.
The length of time for which the delay unit 2 delays the signal elapses until the input audio signal supplied to the mixing unit 5 is supplied to the addition unit 8 through the processing in the mixing unit 5, the HPF 6, and the gain adjustment unit 7. To be substantially equal to the length of time that The phase of the delayed input audio signal supplied from the delay unit 2 to the addition unit 8 and the phase of the signal supplied from the gain adjustment unit 7 to the addition unit 8 are supplied to the addition unit 8 at the same time. It is assumed that the phases are substantially the same.
[0037]
The control unit 3 determines the frequency of a local oscillation signal, which will be described later, generated by the local oscillation unit 4 based on the content of the control data supplied from the input unit 1, and outputs information indicating the determined frequency to the local oscillation unit 4. To supply.
Further, the control unit 3 determines the passband characteristics of the HPF 6 based on the content of the control data, and supplies information indicating the determined passband characteristics to the HPF 6.
[0038]
Specifically, the control unit 3 further includes, for example, a nonvolatile memory such as an EEPROM (Electrically Erasable / Programmable Read Only Memory), and the nonvolatile memory stores a bandwidth table and a Q value table in advance. .
The bandwidth table is a table that stores data indicating the upper limit value of the occupied band of the input audio signal for each of the possible formats of the input audio data. The Q value table is a table that stores data specifying the Q value of the HPF 6 for each music genre represented by the input audio data. The value of Q is in a range from about 1 to about 7.
Then, the control unit 3 specifically performs the processes described below as (1) to (4).
[0039]
(1) First, the control unit 3 specifies the upper limit value of the occupied bandwidth of the input audio data in the format indicated by the control data by searching the bandwidth table.
[0040]
(2) Next, the control unit 3 determines the passband of the HPF 6 and the frequency of the local oscillation signal so as to satisfy the following conditions (a) and (b). That is, as shown in FIGS. 2B and 2C,
(A) Frequency f of local oscillation signal_OSCIs the upper limit frequency f of the occupied band of the input audio signal._INLower,
(B) Peak frequency f of gain of HPF 6_PIs the frequency f_INLower and frequency (f_IN−f_OSC)taller than,
The pass band of the HPF 6 and the frequency of the local oscillation signal are determined so that the following relationship is satisfied. That is, f_IN, F_OSCAnd f_PHave substantially the relationship shown in Equations 1 and 2.
[0041]
(Equation 1)
f_OSC  <F_IN
(Equation 2)
(F_IN  −f_OSC) <F_P  <F_IN
[0042]
However, as described with reference to FIG. 8, a signal representing music generally has a spectral distribution including many peaks in which components of 10 kHz or more are attenuated as the frequency becomes higher and arranged at intervals of about several hundred hertz. It has the feature of having.
Therefore, in order to generate an output audio signal by adding such a characteristic component to the input audio signal by the processing described below, as shown in FIG. The component of 10 kHz or more (or the component having the characteristic even if it is not 10 kHz or more) is processed by the mixing unit 5 so that the frequency f_INIt is desirable that the frequency is converted so as to occupy the above band.
In order to obtain such a result of the frequency conversion, for example, if the portion having the characteristic in the input audio signal is a frequency component of about 10 kHz or more, the frequency f of the local oscillation signal_OSCIs (f_IN(−10 kHz).
[0043]
(3) While performing the process of (2), the control unit 3 determines the Q value to be taken by the HPF 6 by searching the Q value table using the music genre indicated by the control data as a search key.
[0044]
(4) Then, the control unit 3 sets f to satisfy the above condition (a)._OSCIs supplied to the local oscillator 4. F satisfying the condition (b) described above._PIs supplied to the HPF 6 with the information designating the value of as the peak frequency of the passband characteristic of the HPF 6 and the information designating the value of Q determined in the process (3).
[0045]
When the information indicating the frequency of the local oscillation signal is supplied from the control unit 3 to the local oscillation unit 4, the local oscillation unit 4 generates a local oscillation signal having the frequency indicated by the information and supplies the signal to the mixing unit 5.
[0046]
The local oscillating unit 4 includes, for example, an IIR (Infinite Impulse Response) type low-pass filter as shown in FIG. However, it is assumed that the value of Q of this low-pass filter is set to substantially infinity.
The local oscillator 4 shown in FIG. 3 is excited and oscillated by the trigger pulse when, for example, the controller 3 generates and supplies a trigger pulse to instruct the local oscillator 4 to start generating a local oscillation signal. Thus, a local oscillation signal that is substantially a sine wave is generated.
When the local oscillation unit 4 has the configuration shown in FIG. 3, the local oscillation unit 4 adjusts the oscillation frequency of the local oscillation unit 4 by, for example, adjusting the amplification factor of each processing block that amplifies a signal. Is adjusted to the value indicated by the information supplied from the control unit 3.
[0047]
In FIG. 3, "D" represents a processing block for delaying a signal, "A" represents a processing block for amplifying a signal, and "+" represents a processing block for adding a signal. .
[0048]
The mixing unit 5 is supplied with the same input audio signal as that supplied to the delay unit 2 at the same time as the delay unit 2. Then, by mixing the input audio signal and the local oscillation signal generated by the local oscillation unit 4, a signal representing a product of the input audio signal and the local oscillation signal is generated, and the generated signal is supplied to the HPF 6.
[0049]
The signal supplied from the mixing unit 5 to the HPF 6 includes a component having a frequency corresponding to the sum of the frequency of the input voice signal and the frequency of the local oscillation signal (sum component), and a frequency corresponding to the difference between the frequency of the input voice signal and the frequency of the local oscillation signal. Component (difference component). The spectrum of the sum component has a frequency f as shown in FIG._OSCAnd the frequency (f_IN+ F_OSC). Further, the spectrum of the difference component has a frequency (f) as shown in FIG._IN−f_OSC).
[0050]
The HPF 6 filters the component supplied from the mixing unit 5 and supplies the filtered component to the gain adjustment unit 7.
The HPF 6 has a pass band characteristic corresponding to an IIR type high-pass filter, as shown in a graph of an example of the pass band characteristic in FIG. 4, and has a peak, and the frequency increases on the higher frequency side than the peak. Accordingly, the attenuation rate of the HPF 6 increases. Then, the HPF 6 adjusts the peak frequency and the Q value to the values indicated by the information supplied from the control unit 3. As the value of Q is larger, the peak of the pass band characteristic of HPF 6 becomes steeper.
[0051]
FIG. 3 exemplifies a logical configuration in a case where the HPF 6 includes two-stage cascade-connected IIR high-pass filters.
When the HPF 6 has the configuration shown in FIG. 3, the HPF 6 adjusts the peak frequency and the Q value of the pass band characteristic of the HPF 6 by adjusting the amplification factor of each processing block that amplifies a signal, for example. Is adjusted to the value indicated by the information supplied from the control unit 3.
[0052]
As described above, the sum component and the difference component of the signal supplied to the HPF 6 by the mixing unit 5 occupy a band as illustrated in FIG. On the other hand, the peak frequency f of the pass band characteristic of the HPF 6_PIs (f_IN−f_OSCF)_INIs less than.
Accordingly, as shown in FIG. 2C, the HPF 6 has a frequency f_PAnd f_IN+ F_OSCThe following sum components are passed and the other components are substantially blocked.
However, as described above, the attenuation rate of the HPF 6 increases as the frequency increases on the high frequency side from the peak of the passband characteristic, and therefore, the sum component passing through the HPF 6 increases as the frequency increases.
[0053]
The gain adjuster 7 amplifies the signal supplied from the HPF 6 and supplies the amplified signal to the adder 8.
The gain of the gain adjuster 7 is determined by the following equation: the envelope of the spectrum of the output signal generated by the adder 8 is the frequency f_INIt is assumed that the value is set so as to be smooth in the vicinity of. This value may be empirically determined in advance by, for example, the manufacturer of the high-frequency signal interpolator.
[0054]
The adder 8 generates a signal representing the sum of the delayed input audio signal supplied from the delay unit 2 and the signal supplied from the gain adjuster 7, and outputs the signal as an output audio signal. The format of the output audio signal is arbitrary, and may be, for example, an analog signal or a digital signal such as a PCM (Pulse Code Modulation) format.
[0055]
The output audio signal includes, as shown in FIG. 2D, a component corresponding to the input audio signal and an added component. The added component frequency-converts the component of the input audio signal whose frequency is equal to or higher than the predetermined value to the high frequency side by an amount corresponding to the difference between the predetermined value and the upper limit frequency of the occupied band of the input audio signal. It is obtained by that.
[0056]
When the input audio signal is a signal whose band is limited, it is highly likely that the component removed from the original audio signal is composed of harmonic components of 10 kHz or more of the original audio signal. Therefore, when the input audio signal is a signal whose band is limited, the output audio signal is close to the original audio signal before the band is limited.
[0057]
The component added to the input audio signal in the output audio signal is a component on the higher frequency side than the peak of the pass band characteristic of the HPF 6, and the higher the frequency, the more the component is attenuated. For this reason, the spectrum of the output audio signal has a natural distribution close to the typical spectrum of audio, such that the higher the frequency, the lower the intensity.
[0058]
In addition, since the frequency of the local oscillation signal and the pass band characteristics of the HPF 6 change according to the occupied bandwidth of the input audio signal, it is possible to perform appropriate interpolation for input audio signals having various occupied bandwidths.
[0059]
Also, by changing the value of Q of the HPF 6, the shape of the envelope of the high-frequency component added to the input audio signal in the output audio signal changes, and this is used to suit the music genre. It is possible to obtain a spectral distribution.
Specifically, in the case of a rock where a sound source that emits a high sound such as a cymbal is used frequently, the value of Q is lowered so that a high frequency component does not attenuate and remains much. In the case of a classic where such a sound source is not frequently used, A method of increasing the value of Q to greatly attenuate high-frequency components can be considered.
[0060]
Note that the configuration of this high-frequency signal interpolator is not limited to the above.
For example, the format of the input audio data does not need to be the MP3 format or the ATRAC3 format, but may be the AAC (Advanced Audio Coding) format or any other format.
Further, the high-frequency signal interpolator may further include an audio amplifier, a speaker, and the like for reproducing the sound represented by the output signal generated by the adding unit 8.
[0061]
Further, the input unit 1 may include a communication control device including an interface (for example, a modem or the like) for connecting to an external communication line in place of or together with the recording medium driver. In this case, the input unit 1 may acquire the input audio data and control data from an external communication line instead of reading the data from the recording medium.
[0062]
Note that, as described above, in the case of an audio signal representing an audio transmitted via a telephone line, generally, a component having a frequency of 4 kHz or more attenuates as the frequency increases and peaks at intervals of about several hundred hertz while attenuating. It has the characteristic that it has a spectrum distribution including many.
For this reason, when the input audio signal is transmitted via a telephone line (for example, this high-frequency signal interpolator is incorporated in a telephone terminal such as a cellular phone, and the telephone terminal interpolates the voice received by the telephone terminal. In such a case, in order to generate an output audio signal close to the original audio signal, a component having a frequency of about 4 kHz or more in the input audio signal is processed by the mixing unit 5 so that the frequency f_INIt is desirable that the frequency is converted so as to occupy the above band. In order to obtain the result of such frequency conversion, the frequency f_OSCIs (f_IN-4 kHz).
[0063]
Further, the control data may directly indicate the value of the upper limit frequency of the occupied band of the input audio data.
The upper limit of the occupied band of the input audio signal is affected by the quality of the line (for example, telephone line) through which the input audio signal is transmitted, regardless of the format of the input audio data representing the input audio signal. Therefore, it may be limited to a certain value. For this reason, the control data may indicate the type and quality of the line on which the input audio data is transmitted. In this case, the bandwidth table may include, for example, data indicating the upper limit value of the occupied band of the input audio signal in association with the type or quality of the line on which the input audio signal is transmitted.
[0064]
Further, the control data does not have to be associated with the input audio data in a one-to-one correspondence. For example, the control data recorded on the recording medium may be all input audio data recorded on the recording medium. It may represent a format or a music genre.
[0065]
If the upper limit of the occupied band of the input voice data is fixed in advance, the local oscillator 4 does not need to change the frequency of the local oscillation signal, and the HPF 6 also changes the peak frequency of the pass band characteristic. You don't have to. Also, the value of Q of the HPF 6 need not necessarily be variable. When the value of Q is a fixed value, if the value of Q is about 2.3, an aurally favorable result can be obtained for music in general.
If the frequency of the local oscillation signal, the peak frequency of the pass band characteristic of the HPF 6, and the value of Q are all fixed, the control unit 3 is unnecessary, and the input unit 1 does not need to acquire the control data. .
[0066]
In addition, the control unit 3 may set the frequency of the local oscillation signal, the peak frequency of the pass band characteristic of the HPF 6, and / or the value of Q according to a user instruction. In this case, the control unit 3 may include, for example, an input device such as a touch panel, and may follow an instruction of an operator who operates the input device. Note that the input device of the input unit 1 may also function as the input device.
[0067]
Although it is difficult to obtain an output audio signal having a natural spectrum distribution close to a typical spectrum of audio, the pass band characteristics of the HPF 6 need not necessarily have a peak, and the HPF 6 is not necessarily an IIR type. It is not necessary. Therefore, the HPF 6 may be composed of, for example, a FIR (Finite Impulse Response) type high-pass filter.
When the pass band characteristic of the HPF 6 does not have a peak, for example, the cutoff frequency may satisfy the above condition (b) instead of the peak frequency.
[0068]
Further, the HPF 6 may be composed of an IIR or FIR type band pass filter. In this case, the peak frequency on the low frequency side or the cutoff frequency on the low frequency side of the HPF 6 may satisfy the above condition (b).
[0069]
Also, if the phase shift between the portion corresponding to the input voice signal and the added portion of the output voice signal is negligible perceptually, the high-frequency signal interpolator does not necessarily need to include the delay unit 2. In this case, the input unit 1 may directly supply the input audio signal to the adding unit 8.
[0070]
When the HPF 6 has the configuration shown in FIG. 3, after determining the pass band characteristics of the HPF 6, the control unit 3 determines the amplification factor of each processing block in the HPF 6 that amplifies a signal, by using this pass band. It may be determined so that the characteristics are realized, and the determined amplification factor may be notified to the HPF 6. In this case, the HPF 6 may adjust the amplification factor of each processing block that performs signal amplification to a value as notified from the control unit 3.
[0071]
Further, the high-frequency signal interpolator may be constituted by an analog circuit. Specifically, for example, the local oscillation unit 4 may be configured by a known oscillation circuit. Further, the mixing section 5 may be constituted by a known multiplication circuit or a known amplitude modulation circuit. Further, the HPF 6 may be constituted by a known buffer circuit or a known high-pass filter circuit such as a Chebyshev type, a Butterworth type and the like including an operational amplifier, a resistor and a capacitor. Further, the delay unit 2 may be configured by a known delay element such as a delay line or a buffer circuit. Further, the gain adjustment unit 7 may be configured by a known amplifier circuit. Further, the adding unit 8 may be configured by a known adding circuit including an operational amplifier and a resistor.
[0072]
FIG. 5 shows a case where the mixing section is formed of an amplitude modulation circuit, the HPF 6 is formed of a Chebyshev type high-pass filter circuit, the delay section 2 is formed of a buffer circuit, and the addition section 8 is formed of an operational amplifier addition circuit. FIG. 3 is a diagram showing an example of the configuration of this high-frequency signal interpolator. However, in FIG. 5, "B" represents a buffer circuit, and "OPA" represents an operational amplifier. In the configuration of FIG. 5, it is assumed that the buffer circuit constituting the HPF 6 performs the function of the gain adjustment unit 7.
[0073]
When the local oscillating unit 4 is configured by a variable frequency oscillating circuit such as an analog VCO (Voltage Controlled Oscillator) that can control the oscillating frequency by supplying a control signal from the outside, the control unit 3 performs the above-mentioned (a). The oscillation frequency of the local oscillation signal may be controlled by supplying the local oscillation unit 4 with a control signal that generates a local oscillation signal having a frequency satisfying the condition (2).
[0074]
In addition, as an element such as a resistor or a capacitor that determines a pass band characteristic of the high-pass filter circuit constituting the HPF 6, an element (for example, a field-effect transistor, If a varicap is used, the control unit 3 can control the peak frequency and the Q value of the pass band characteristic of the HPF 6 by supplying the control signal to these elements of the HPF 6.
[0075]
The high-frequency signal interpolator may be provided to a computer such as a microcomputer or a personal computer by inputting the above-described input unit 1, delay unit 2, control unit 3, local oscillation unit 4, mixing unit 5, HPF 6, gain adjustment unit. The program may be installed from a recording medium (such as a CD-ROM or a floppy (registered trademark) disk) storing a program for executing the operation of the adder 7 and the adder 8.
[0076]
This program may be any program that causes a computer to perform the processing shown in FIG. 6, for example.
[0077]
That is, the processor of the computer that executes this program loads the input voice data and the control data that are associated with each other from the recording medium, and stores them in the main storage device. Then, based on the loaded control data, the frequency of the local oscillation signal and the passband characteristic to be obtained by the filtering in step S5 described later are determined in the same manner as in the control unit 3 (step S1).
[0078]
Next, the processor of the computer acquires one sample of the input audio signal from the loaded input audio data (step S2). Further, data for one sample of the local oscillation signal having the frequency determined in step S1 is generated (step S3). In step S2, the first sample of the data not yet obtained in step S2 is obtained, and in step S3, data whose phase is delayed by one sample from the data generated last in step S3 (however, In the first step S3, data having an arbitrary initial phase is generated.
[0079]
Next, the processor calculates the product of the value of the input voice data obtained in step S2 and the value of the sample of the local oscillation signal generated in step S3 (step S4), and filters the obtained product (step S5). ).
[0080]
The filtering in step S5 may be performed, for example, by calling the function whose source code is shown in FIG. 7 once or a plurality of times.
FIG. 7 shows an example of a source code in which a process for causing a computer to perform the function of a secondary IIR filter is described in C language.
[0081]
The source code shown in FIG. 7 is generated by an IIR type filter having pass band characteristics determined by the values of the variables “a0”, “a1”, “a2”, “b0”, “b1”, and “b2”. And “xr” are individually filtered, the result of filtering the variable “xl” is stored in the storage location indicated by the pointer “yl”, and the result of filtering the variable “xr” is stored in the storage location indicated by the pointer “yr”. This represents the process of storing. (That is, the source code shown in FIG. 7 represents a process of filtering two input audio signals in parallel. However, the number of input audio signals to be processed in parallel may be arbitrary. is there.)
[0082]
Then, the processor adds the value of the input audio data acquired in step S2 to the value obtained as a result of the filtering in step S5 (step S6), and outputs the addition result as an output audio signal (step S7). The process returns to step S2.
[0083]
For example, this program may be uploaded to a bulletin board (BBS) of a communication line and distributed via the communication line. Alternatively, a carrier wave is modulated by a signal representing the program, and the obtained modulated wave is transmitted. A device that has transmitted and received this modulated wave may demodulate the modulated wave and restore the program.
Then, by starting this program and executing it in the same manner as other application programs under the control of the OS, the above-described processing can be executed.
[0084]
When the OS shares a part of the processing, or when the OS constitutes a part of one component of the present invention, a program excluding the part is stored in the recording medium. You may. Also in this case, in the present invention, it is assumed that the recording medium stores a program for executing each function or step executed by the computer.
[0085]
【The invention's effect】
As described above, according to the present invention, there is provided a signal interpolation device for restoring a signal close to the original signal with less distortion from a modulated wave obtained by using a signal whose band of the original signal is limited, A reproduction device and a signal interpolation method are realized.
Further, according to the present invention, a signal interpolation device, a sound reproduction device, and a signal interpolation method for restoring an audio signal with high sound quality are realized.
[Brief description of the drawings]
FIG. 1 is a diagram showing a configuration of a high-frequency signal interpolator according to an embodiment of the present invention.
2A is a graph showing a spectrum of an input audio signal, FIG. 2B is a graph showing a spectrum of a sum component and a difference component generated by a mixing unit, and FIG. It is a graph showing the spectrum of the component to be output and the pass band characteristic of the HPF, and (d) is a graph showing the spectrum of the output audio signal.
FIG. 3 is a diagram illustrating a logical configuration of a delay unit, a local oscillation unit, and an HPF.
FIG. 4 is a graph showing pass band characteristics of an HPF.
FIG. 5 is a diagram showing a configuration of a modification of the high-frequency signal interpolator according to the embodiment of the present invention.
FIG. 6 is a flowchart showing a flow of processing performed when the computer performs the function of the high-frequency signal interpolator according to the embodiment of the present invention.
FIG. 7 is an example of a source code describing a filtering process shown in FIG. 6;
FIG. 8 is a graph illustrating a typical spectrum of an audio signal representing music.
[Explanation of symbols]
1 Input section
2 Delay unit
3 control part
4 Local oscillator
5 Mixing section
6 HPF
7 Gain adjustment section
8 Addition unit

Claims (17)

A frequency conversion unit that generates an interpolation signal by frequency-converting an interpolated signal representing a voice whose spectrum is to be interpolated,
A filter for extracting a component in a second band on a higher frequency side than the first band occupied by the signal to be interpolated among the interpolation signals;
An addition unit that generates an output signal representing the sum of the interpolated signal and the component extracted by the filter.
A signal interpolation device characterized by the above-mentioned. The frequency conversion unit performs frequency conversion such that a component having a frequency equal to or higher than a predetermined frequency in the interpolated signal is frequency-converted into the second band.
The signal interpolation device according to claim 1, wherein: The predetermined frequency is a frequency equal to or higher than a lower limit of a region in which the distribution of the spectrum of the interpolated signal has a predetermined characteristic.
3. The signal interpolation device according to claim 2, wherein: The predetermined feature of the region is characterized in that the region includes a plurality of peaks arranged at intervals of 100 Hz or more and less than 1 kHz while attenuating as the frequency increases.
The signal interpolation device according to claim 3, wherein: The predetermined frequency is a frequency of 10 kHz or more;
3. The signal interpolation device according to claim 2, wherein: The interpolated signal represents a sound transmitted via a telephone line,
3. The signal interpolation device according to claim 2, wherein: The filter specifies the range of the first band based on control data supplied from the outside, and changes the range of the second band to be on the higher frequency side than the range of the first band. Let
The signal interpolation device according to claim 1, wherein: The control data includes information indicating a data format of the interpolated signal, and the filter specifies the range of the first band based on the information.
The signal interpolation device according to claim 7, wherein: The pass band characteristic of the filter has a peak, the second band belongs only to the higher frequency side than the peak,
The signal interpolation device according to any one of claims 1 to 8, wherein: The filter is composed of an IIR (Infinite Impulse Response) type or a FIR (Finite Impulse Response) type high-pass filter or a band-pass filter.
The signal interpolation device according to claim 1, wherein: In the filter, the value of Q is determined based on control data supplied from the outside, and the passband characteristic is changed so as to have the determined value of Q.
The signal interpolation device according to claim 9, wherein: The control data includes genre information indicating a genre of music represented by the interpolated signal, and the filter determines a value of Q based on the genre information.
The signal interpolation device according to claim 11, wherein: The filter determines a value of Q in a range of 1 to 7.
The signal interpolation device according to claim 11, wherein: The adder includes a delay unit that delays the interpolated signal so that the extracted signal has substantially the same phase as the component extracted by the filter, the component extracted by the filter and the interpolated signal delayed by the delay unit. Generating the output signal representing the sum of
The signal interpolation device according to claim 1, wherein: A frequency conversion unit that generates an interpolation signal by frequency-converting an interpolated signal representing a voice whose spectrum is to be interpolated,
A filter for extracting a component in a second band on a higher frequency side than the first band occupied by the signal to be interpolated among the interpolation signals;
An adding unit that generates an output signal representing a sum of the interpolated signal and a component extracted by the filter,
Sound reproducing unit that reproduces the sound represented by the output signal generated by the adding unit,
A sound reproducing device characterized by the above-mentioned. Generate an interpolation signal by frequency-converting an interpolated signal representing a voice whose spectrum is to be interpolated,
Of the interpolation signal, to extract a component in a higher frequency band than the band occupied by the signal to be interpolated,
Generating an output signal representing the sum of the interpolated signal and the extracted component;
A signal interpolation method characterized in that: Computer
A frequency conversion unit that generates an interpolation signal by frequency-converting an interpolated signal representing a voice whose spectrum is to be interpolated,
Among the interpolation signals, a filter for extracting a component in a band on a higher frequency side than a band occupied by the signal to be interpolated,
An adding unit that generates an output signal representing a sum of the interpolated signal and a component extracted by the filter,
Program to make it work.

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