JPS59195284A - Pitch adjustor - Google Patents

Abstract

(57) [Summary] This bulletin contains application data before electronic filing, so abstract data is not recorded.

Description

[Detailed description of the invention] The present invention relates to a pitch adjustment device.

When sampling the waveform of a voice or musical tone, the sampling period is Ts), and changing the pitch (the height of the sound) when playing back the digital signal obtained by AD conversion (a sequence of sample values that are discrete with respect to time). There may be cases where you want to output the

The simplest method is to perform DA conversion at every sampling period (referred to as T8) different from the sampling time, but this is not suitable when the sampling period of the system is to be fixed. In such a case, it is necessary to adjust the pitch by digital signal processing.

To do this, it is sufficient to generate a digital signal as if it were sampled at each different sampling period (referred to as TS') and output this at each sampling period Ts.

Conventionally, there is a method using linear interpolation as a means for obtaining such a signal. That is, assuming a temporally continuous signal in which discrete sample values vary linearly with respect to time, sample values are generated by sampling this signal at a period TS'. This conventional example has the advantage that the amount of calculation is small and the pitch change width can be selected almost continuously, but the error is large and depending on the shape of the spectral envelope of the original waveform, the pitch change width may be within a few percent. However, the drawback was that the sound quality deteriorated considerably.

Another known conventional example is one in which interpolation is performed not by a straight line but by a function given by the following equation.

Here, Ts is the aforementioned sampling period. In this conventional example, the sample value of the original waveform to be interpolated (x (nTs
) and X (denoted as (n+1)Ts)).
y(kTs')) is determined by the following equation.

(2) In this conventional example, in order to reduce the error, y(kTs'
) in order to obtain one sample, many sample values of the original waveform must be used, and many calculations are required. Furthermore, in order to make the range of pitch change continuous, the value of h(t) must be calculated for each value, which requires more calculations.

A third conventional example uses a digital filter. This interpolates M-1 sample values, which are value casseros, between the sample values of the original waveform (as a result, the sample value sequence becomes Ts/M), inputs this sample value sequence to a digital low-pass filter, and then output to L
It is taken out for each sample. As a result, the sunblink period becomes Ts X L/M. In this conventional example, sound quality deterioration can be prevented by using a filter with a steep cutoff characteristic, but in order to keep the pitch change width within a few percent, M must be set to about several tens to 100. The object of the present invention is to create a pitch that can be realized with a relatively small amount of calculation, has little deterioration of sound quality due to the influence of errors, and allows the pitch change width to be selected almost continuously. The object of the present invention is to provide a regulating device.

The present invention relates to a pitch adjustment device of the type that changes the pitch by interpolating between sample values of a waveform, and a means for shortening the sunblink period by interpolating a sample value whose value is zero between sample values of an original waveform. , a digital low-pass filter that filters the output sample value sequence, and a means for linearly interpolating and outputting the sample values before and after the adjusted sample value among the sample values output from the low-pass filter. be done.

According to the present invention, the amount of calculation is small, the range of change in pitch can be selected almost continuously, and at the same time, it is possible to reduce deterioration of sound quality due to errors.

Next, one embodiment of the present invention will be described using the drawings.

FIG. 1 is a block diagram showing an embodiment of the present invention, and FIG. 2 is a diagram showing an example of input/output and signals during processing.

First, an input signal, which is a sample value of an original waveform input from the input terminal 107, is stored in the input buffer 101.

An example of an input signal with a sampling period of Ts is shown in Figure 2 (
Shown in a). Control signal transmission line A11 from control circuit 106
The input signal is sequentially transferred from the input buffer 101 to the zero interpolator 102 via the input signal transmission line 110 according to a control signal sent via the input signal transmission line 110 .

In the zero interpolator 102, according to the control signal sent from the control circuit 106 via the control signal transmission line B115, sample values having a value of zero between the sample values of the input signal sent from the input buffer 101 are M- One sample is inserted, and as a result, a signal whose sampling period is multiplied by 17M is sent to the digital low-pass filter 103 through the upper interpolation signal transmission path 111. FIG. 2(b) shows an example of a signal in which a sample with a value of zero is inserted into the input signal illustrated in FIG. 2(a). This example is for M22, but in general, the larger the value of M, the more the influence of error is reduced.

The digital low-pass filter 103 is connected to the control circuit 10.
6 via the control signal transmission line C116, the signal sent from the cello interpolator 102 is filter-linked, and the resulting signal is passed through the filter output transmission line 112 to the direct M? It is sent to the cuff 104. Figure 2 (
C) shows an example of the output signal of the digital low-pass filter,
show. 'i White helix interpolator 104 outputs sample values for each period Ts' sent from the digital low-pass filter 103 in accordance with the sub-side 1 signal sent from the control circuit 106 via the control signal transmission line D 117. The signal is obtained by linear interpolation between the samples of the signal, and the resulting signal is sent to the output cover 105 via the linear interpolation signal transmission line 113. The control signal sent from the control circuit 106 to the linear interpolator 104 consists of information instructing it to receive or receive the output signal of the digital low-pass filter 103, and information instructing the time at which linear interpolation is to be performed. The latter information is obtained by the control circuit 106 from the adjustment rate α input from the adjustment rate input terminal 109 and the sampling period Ts of the system.
Ts' is calculated and generated as follows. The state of linear interpolation is shown in FIG. 2(C), and the resulting signal is shown in FIG. 2(d).

The output buffer 105 stores the signal accelerated from the linear interpolator 104, and outputs the signal to the output terminal 108 at every sampling period 'rS in accordance with a control signal sent from the control circuit 106 via the control signal transmission line E118. The signal output from the output terminal is shown in FIG. 2(e).

Note that in the embodiment, linear interpolation is performed for only the portion corresponding to the sampling period 1'S', or linear interpolation may be performed for all portions). However, in this case, the processing time will increase accordingly.

[Brief explanation of the drawing]

It is a figure showing an example of a number. In the figure, 101 is an input buffer, 102 is a zero interpolator, 103 is a vertical low-pass filter, 104 is a linear interpolator, 1
05 is an output buffer, 106 is a control circuit, 107 is an input terminal, 108 is an output terminal, and 109 is an adjustment rate input terminal.

Claims (1)

[Claims] In a pitch adjustment device of a type that changes pitch by interpolating between sample values of a waveform, means for shortening the sampling period by interpolating sample values having a value of zero between sample values of an original waveform, and It is characterized by comprising a digital low-pass filter that filters a sample value sequence, and a means for linearly interpolating and outputting sample values before and after the adjusted sample value among the output sample values of the digital low-pass filter. pitch adjustment device.