JPH05315891A - Digital signal processing system - Google Patents

Abstract

PURPOSE:To obtain a signal processing system in which a reduction in a pitch of an audio signal is compensated by a digital signal in an excellent S/N. CONSTITUTION:A digital signal being an object of the reduction in a pitch of an audio signal is given to an oversampling filter 2 applying a multiple M of oversampling to a sample series of the digital signal. Each sequential sample of the digital signal subject to oversampling is read repetitively from a storage device for each of N sets of oversampling periods by a magnified interpolation section 3 for the sampling period and arranged on a time axis to expand the sampling period of the digital signal being a processing object into a multiple of N. Then digital data comprising a sample series subject to substantial hold interpolation are obtained at a time position applying MN division to the sampling period magnified by a multiple of N. A reflected component in the digital data is eliminated by an anti-alias filter 4.

Description

Detailed Description of the Invention

[0001]

BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to a digital signal processing system capable of reducing the pitch of an acoustic signal in the state of a digital signal.

[0002]

2. Description of the Related Art It is well known that audio signals have been edited for various purposes, and analog signals that have been widely used as audio signal editing devices have been known. In addition to the morphological sound signal editing device, a digital signal morphological signal editing device is also used. The above-described audio signal editing device is, for example, a recording / reproducing function, fade-in, in order to edit the audio signal.
It is configured to have fade-out, cross-fade, edit / playback function based on edit list, and many other functions. When detecting an edit point, the acoustic signal is greatly expanded on the time axis. That is, it has been conventionally performed to reduce the pitch of the acoustic signal.

[0003]

Conventionally, as a means for lowering the pitch of the acoustic signal of the digital signal, there is a method of lowering the sampling frequency into an analog signal. After converting to an analog signal, an anti-alias filter is used to remove aliasing noise. This conventional method requires an analog-to-digital converter for resampling when returning to a digital signal having a sampling period after the pitch is reduced and before the pitch is reduced. Since the deterioration of the signal for returning to the signal is unavoidable, there is a drawback that high-precision conversion cannot be performed. The above-mentioned drawbacks can be solved by performing the signal processing in the state of the digital signal, but in the signal processing in the state of the digital signal, aliasing noise occurs if the data is simply zero-interpolated or hold-interpolated. Therefore, when the pitch is lowered with the digital signal as it is, a signal processing method capable of removing the aliasing noise well is required.

[0004]

SUMMARY OF THE INVENTION The present invention is a digital signal processing system for reducing the pitch of an acoustic signal to 1 / N (where N is an integer of 2 or more) in the state of a digital signal,
An oversampling filter for performing M times (where M is an integer of 2 or more) oversampling a sample value sequence of a digital signal targeted for pitch reduction, and each of the oversampled digital signals in sequence. The sampled values are repeated every N consecutive oversampling periods and are arranged on the time axis, so that the sampling interval of the digital signal targeted for the above-mentioned pitch reduction is expanded N times, and the above N Means for obtaining digital data of a sample value series in a state in which substantial hold interpolation is performed at a time position where the sampling interval doubled is divided by MN, and antialiasing for removing aliasing components in the digital data A digital signal processing method including a filter is provided.

[0005]

The digital signal targeted for the reduction of the pitch of the acoustic signal is oversampled by the oversampling filter that performs M times oversampling on the sample value series of the digital signal. Each successive sample value of the oversampled digital signal is repeatedly read from the storage device every N number of oversampling periods and arranged on the time axis, and the digital signal targeted for the above-described pitch reduction is described. Of the sampling value sequence is expanded N times, and digital data consisting of a sample value series in a state in which substantial hold interpolation is performed at a time position where the sampling interval expanded N times is divided by MN is obtained. .. Then, the aliasing component in the digital data is removed by the anti-aliasing filter.

[0006]

DESCRIPTION OF THE PREFERRED EMBODIMENTS The specific contents of the digital signal processing system of the present invention will be described in detail below with reference to the accompanying drawings. FIG. 1 is a block diagram showing a schematic configuration of a digital signal processing system according to the present invention, and FIGS. 2 and 3 are block diagrams when the digital signal processing system according to the present invention is applied to an audio signal jog device or a shuttle device. FIG. 6 is an explanatory diagram of expansion of a sampling section and interpolation operation. In FIG. 1 showing a schematic configuration of a digital signal processing system of the present invention, 1 is an input terminal of a digital signal which is a target of signal processing for lowering the pitch of the digital signal, and 2 is a digital signal supplied to the input terminal 1. Oversampling filter for performing M times oversampling on the sample value sequence of, 3 is a component for expanding and interpolating a sampling interval, 4 is an anti-aliasing filter, and 5 is an output terminal of a digital signal with a reduced pitch. ..

In the digital signal processing system of the present invention,
A digital signal, which is the target of signal processing that can reduce the pitch of the acoustic signal to 1 / N (where N is an integer of 2 or more) in the state of the digital signal, is audio sampled through the input terminal 1. When supplied to the filter 2, the oversampling filter 2 outputs a sample value series in which the sample value series of the digital signal supplied thereto is oversampled M times. The sample value sequence output from the above-described oversampling filter 2 is supplied to the constituent unit 3 for expanding and interpolating the sampling interval, and the constituent unit 3 for expanding and interpolating the sampling interval is shown in FIG. As has been done
For each successive sample value arranged on the time axis, it is repeatedly read from the storage device at every N consecutive oversampling periods Tns / M and arranged on the time axis. That is, in the sample value sequence output from the configuration unit 3 for expanding and interpolating the sampling interval, the sampling interval Tns of the digital signal targeted for pitch reduction is
As shown by the following equation, (Tns / M) MN = N · Tns = Ts, the section Ts is expanded N times,
In addition, the sampling interval expanded to N times is MN
The sample value series is in a state in which substantial hold interpolation is performed at the time positions to be divided.

In FIG. 4, (a) of FIG. 4 shows sample values Sp1, Sp2, S sequentially generated at every sampling period Tns of the digital signal targeted for pitch reduction.
p3 and Sp4 are shown, and FIG. 4B shows the sample value series output from the oversampling filter 2 that performs M times oversampling. Sp1, So1, Sp2, and So2 in the figure are shown. , Sp3, So3, Sp4, So4
Are sequential sample values that are sequentially generated at every oversampling period Tns / M (M = 2 in the example shown in FIG. 4). FIG. 4C shows Ts in which the sampling period is expanded to N times the sampling period Tns illustrated in FIG. 4A (N = 4 in the example shown in FIG. 4). In the interval, the sample value series output from the configuration unit 3 for expanding and interpolating the sampling interval, that is, each successive sample value output from the oversampling filter 2 is four times for each oversampling period Tns / 2. It shows a sample value series in a state in which it appears continuously and substantial hold interpolation is performed.

That is, FIG. 4 exemplifies a case where the pitch of the digital signal is lowered so that the sampling period Tns of the digital signal targeted for the pitch reduction becomes the sampling interval Ts which is expanded four times. In the example (c), the sampling period is expanded to four times the sampling period Tns, and the sampling period is expanded to four times. For every 1 / M of the sampling period Tns (oversampling period Tns / M), the first sample value Sp1 is the oversampling period Tns.
After appearing four times every / Ms, the sample value So1 appears four times every period Tns / Ms of oversampling, and the expanded sampling interval Ts following the first expanded sampling interval Ts. , The sample value Sp2 appears four times in each oversampling period Tns / Ms, and then the sample value So2 appears four times in each oversampling period Tns / Ms. In the sequentially expanded sampling Ts, the sequential sample value output from the oversampling filter 2 is the oversampling period T.
A sample value sequence is formed in a state of appearing four times (N times) every ns / Ms.

The digital data consisting of the sample value series output from the component 3 for expanding and interpolating the sampling interval is supplied to the anti-alias filter 4, and the aliasing component in the digital data is the anti-alias filter. And is output by the output terminal 5. The curve of the one-dot chain line connecting the circles in (c) of FIG.
The present invention shows that a waveform of a signal corresponding to the digital data output from the output terminal 5 described above and a smooth signal waveform can be obtained in the present invention. As the anti-alias filter 4 described above, since the digital data input to the anti-alias filter 4 is over-sampled digital data, compared to the case where digital data in a non-over-sampled state is input, Even if the cutoff region having a gentle slope is used, noise is satisfactorily removed.
It is possible to obtain a good signal of / N.

2 and 3 are block diagrams when the digital signal processing method of the present invention is applied to an audio signal jog device or a shuttle device. In FIGS. 2 and 3, reference numeral 6 denotes an input of a digital signal that is a target of signal processing that can reduce the pitch of the acoustic signal to 1 / N (where N is an integer of 2 or more) in the state of the digital signal. 7 is a buffer memory, 8 is a recording / reproducing unit (for example, a recording / reproducing unit configured using a magneto-optical recording medium), 9 is a control unit, 10 is a signal processing unit, and 11 is an output terminal. In FIG. 3, 12 is a changeover switch and 13 is an oversampling filter. First, the apparatus shown in FIG. 2 will be described as follows. The digital signal having the sampling period Tns supplied to the signal processing unit 10 via the input terminal 6 is oversampled at the sampling period Tns / M by the oversampling filter provided in the signal processing unit 10, The data is recorded on, for example, a magneto-optical recording medium provided in the recording / reproducing unit 8 via the buffer memory 7. The digital signal recorded in the recording / reproducing section 8 is read out under the control of the control section 9, stored in the buffer memory 7, and then read out from the buffer memory 7.

The buffer memory 7 is used as a component that operates in accordance with a control command from the control unit 9 and functions as a sampling unit expansion and interpolation unit. The readout of the digital signal from that is oversampling period Tns.
The same sample value sequentially generated for each / M is read N times on the time axis and transmitted. As a result, substantial hold interpolation is performed in a state where the same sample value of the oversampled digital signal appears N times in the period Ts in which the sampling period is expanded N times the sampling period Tns. The digital data including the sampled value series of states is supplied to the signal processing unit 10. Then, the signal processing unit 1 described above
At 0, the digital data supplied from the buffer memory 7 is sent to the output terminal 11 as digital data in a state where aliasing distortion is removed by the anti-alias filter.

In the apparatus having the configuration illustrated in FIG. 2 described above,
A digital signal having a sampling cycle of Tns supplied to the signal processing unit 10 via the input terminal 6 is transferred to the signal processing unit 10
After the sampling period is oversampled by Tns / M by the oversampling filter provided in, the recording is performed on the magneto-optical recording medium provided in the recording / reproducing unit 8 via the buffer memory 7. Therefore, the magneto-optical recording medium of the recording / reproducing unit 8 is required to have a large storage capacity. FIG. 3 is FIG.
3 shows an example of the configuration of the device having the above-mentioned problems in the device having the configuration illustrated in FIG. The digital signal having the sampling period Tns supplied to the signal processing unit 10 via the input terminal 6 is transmitted to the recording / reproducing unit 8 via the signal processing unit 10 and the buffer memory 7, for example, in a magneto-optical recording medium. To be recorded.

The digital signal recorded in the recording / reproducing section 8 is read out under the control of the control section 9 and given to the movable contact v of the changeover switch 12. The movable contact v of the changeover switch 12 is fixed contact a in the operation mode in which the digital signal recorded in the recording / reproducing unit 8 is directly stored in the buffer memory 7 under the control of the control unit 9. The recording / reproducing section 8
The sample value series of the digital signal read from the M is multiplied by M by the oversampling filter 13 (where M is 2
In the case of an operation mode in which oversampling of the above integer) is performed and then the data is stored in the buffer memory 7, the fixed contact b is switched to. In the operation mode in which the movable contact v of the changeover switch 12 is switched to the fixed contact b side, the digital signal read from the recording / reproducing unit 8 is changed to the changeover switch. It is supplied to the oversampling filter 13 via 12 movable contacts v and fixed contacts b.
In the oversampling filter 13, the sample value series of the digital signal read from the recording / reproducing unit 8 is oversampled by M times (where M is an integer of 2 or more) , Digital data of which is stored in the buffer memory 7 and read from the buffer memory 7.

The above-mentioned buffer memory 7 is used as one that operates according to a control command from the control unit 9 and functions as the component unit 3 for expanding and interpolating the sampling interval. The readout of the digital signal from that is oversampling period Tns.
The same sample value sequentially generated for each / M is read N times on the time axis and transmitted. As a result, substantial hold interpolation is performed in a state where the same sample value of the oversampled digital signal appears N times in the period Ts in which the sampling period is expanded N times the sampling period Tns. The digital data including the sampled value series of states is supplied to the signal processing unit 10. Then, the signal processing unit 1 described above
At 0, the digital data supplied from the buffer memory 7 is sent to the output terminal 11 as digital data in a state where aliasing distortion is removed by the anti-alias filter.

[0016]

As is apparent from the above description in detail, the digital signal processing system of the present invention converts the digital signal targeted for the pitch reduction of the acoustic signal into the sample value series of the digital signal. Oversampling is performed by an oversampling filter that performs M times oversampling, and each successive sample value of the oversampled digital signal is repeatedly read from the storage device at every N oversampling periods, Along the axis, the sampling interval of the digital signal which is the subject of the pitch reduction is expanded N times, and the sampling interval expanded N times is substantially divided by MN at a time position. Obtain the digital data consisting of the sample value series in the state of hold interpolation and Because of aliasing components in the digital data so as to remove the anti-alias filter, the present invention shows that it is possible to obtain an output of the smoother the signal waveform. As the anti-alias filter 4 described above, since the digital data input to the anti-alias filter 4 is over-sampled digital data, compared to the case where digital data in a non-over-sampled state is input, Since noise is satisfactorily removed even if the cutoff region having a gradual slope is used, a signal with a good S / N can be easily obtained.

[Brief description of drawings]

FIG. 1 is a block diagram showing a schematic configuration of a digital signal processing system of the present invention.

FIG. 2 is a block diagram when the digital signal processing system of the present invention is applied to an audio signal jog device or a shuttle device.

FIG. 3 is a block diagram when the digital signal processing method of the present invention is applied to an audio signal jog device or a shuttle device.

FIG. 4 is an explanatory diagram of expansion of a sampling section and interpolation operation.

[Explanation of symbols]

1, 6 ... Input terminal of a digital signal which is a target of signal processing for reducing the pitch of the digital signal, 2 ... Oversampling for performing M times oversampling on a sample value series of the digital signal supplied to the input terminal 1. A filter, a component for expanding and interpolating a sampling interval, an antialias filter, an output terminal of a digital signal with a reduced pitch, a buffer memory, a ...
8 ... Recording / reproducing unit, 9 ... Control unit, 10 ... Signal processing unit, 11
... output terminal, 12 ... changeover switch, 13 ... oversampling filter,

Claims (1)

[Claims] 1. A digital signal processing method for reducing the pitch of an acoustic signal to 1 / N (where N is an integer of 2 or more) in the state of a digital signal, the digital signal being the target of the pitch reduction. Oversampling filter that performs oversampling M times (where M is an integer of 2 or more) on the sample value series of, and successive sample values of the oversampled digital signal are consecutive N oversampled. The sampling interval of the digital signal which is the target of the pitch reduction is expanded N times and arranged on the time axis by repeating every cycle, and the sampling interval expanded N times is divided into MN. Means for obtaining digital data of a sample value series in a state where substantial hold interpolation is performed at the time position; Digital signal processing system consisting of an anti-alias filter for removing aliasing components in the data.