JPS6073694A - Reverberation adder - 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 device for artificially adding reverberation to a musical tone signal or the like. More specifically, the present invention relates to an improvement of a convolution-type reverberation adding device that generates a reverberation signal by convolving an input signal with a time function of reverberation characteristics.

Conventionally, as a convolution type reverberation adding device, Figures 1 to 3 are used.
The configuration shown in the figure is known.

This reverberation adding device includes a data memory 23 that sequentially stores a predetermined number of samples of an input signal X(t) obtained by digitizing an analog musical tone signal, and stores reverberation characteristics corresponding to the impulse response of a room to be virtualized. It is provided with a parameter memory 20 and a convolution calculation means for performing a convolution calculation according to the data in both memories 23 and 20 to obtain a reverberation signal OUT.

The data memory 23 enters the write mode every time the clock C2 with a period T outputted from the timing controller 32 is generated, and the output of the counter 24, which is incremented by the clock C1 with the same period T, is given as a write address ( Note that the output of the counter 24 is given via the subtracter 27, but as will be described later, the subtraction input at this time is O.
).

That is, the input signal X(t) is sequentially written into the data memory 23 at a constant sampling period T, and a predetermined number of latest sample data X1 to Xj (Xl, X2
．． X3...) are stored while being updated sequentially.

The parameter memory 22 stores impulse response characteristics as shown in FIG. 3 (A) in the data format shown in FIG.
The desired reverberation characteristics are expressed by delay time data Di at m points on the time axis and gain data Gi corresponding to each point. Data O is written at the first address 0, and parameters Di and Gi are stored in order at the subsequent addresses (note that the data at address 0 are Do and Go).

The delay time data Di here is given as an integer value indicating how many times the sampling period T of the input signal is. In other words,
D2XT is the value of the time dimension.

The writing of parameters to the memory 20 is performed by the Bontroller 22.
This is done in advance. A multiplexer 26 connected to the address input of the memory 20 is connected to the controller 22.
The switching is controlled by , and the address line is drawn into the controller 22 only when writing parameters. In other normal operating conditions, the output of the counter 26 is applied to the parameter memory 2 as a read address.

The counter 26 is cleared by a clock C4 generated immediately after the clock C1, and is incremented by a clock C5 having a period of l/(m+1) of the sampling period T, and gives the count output to the parameter memory 20 as a read address. .

Therefore, every sampling period T, the parameter memory 20
All data stored in is read out in order. The delay time data Di among the parameters read from the memory 20 serves as a subtraction input to the subtracter 270. Further, the gain data Gi is input to the multiplier 28, and is multiplied by the data Xj read from the data memory 23 as described later. The multiplication output is input to an accumulator 29 consisting of an adder 30, a register 35, and an AND circuit 31.

To be more specific, as shown in FIG. 2, the counter 24 is incremented at the rising edge of the clock C1, and at the same time, the clock C4 goes low and the counter 25 is cleared. Therefore, address O is specified in the parameter memory 20 by the output of the counter 25, and the data Do=O, Go=O stored at the address O as described above are read out.

Then, at the next rising edge of clock C5, clock C2
also rises, and the data memory 23 enters the write state.

At this time, the output of the parameter memory 20 is O, so
The output of the subtracter 27 is the output of the counter 24 itself. Therefore, the input signal X (t ) at this time is written to the address indicated by the output Ak of the counter 24. This is the latest sample data X1, and similarly, the latest past sample data X1 to Xj are stored in the data memory 23.
is stored in

When writing of one sample data to the data memory 23 is completed, the data memory 23 enters the read mode. Further, until the next rise of the clock C1, the counter 25 is incremented in synchronization with the clock C5, and the output of the counter 25 sequentially specifies the address m of the parameter memory 20, and the parameters Di and Gi are sequentially read out. .

When the delay time data Of (which is an integer value as described above) is read out from the parameter memory 20, the value Ak obtained by subtracting Di from the output Ak of the counter 24 is obtained.
-Di is output from the subtracter 27, which becomes the read address of the data memory 23, and the data corresponding to this address is read from the memory 23.

The address Ak of the data memory 23 is the storage address of the latest sample data X1, and the address Ak
-D; is the storage address of past sample data. Let Xi be the sample data read from -Di at address 8. This data Xi is data obtained by delaying the input signal X(t) by a time DiXT.

The above delay data Xi read out from the data memory 23
is stored in the parameter memory 20 along with the delay time data Di.
The multiplier 28 multiplies (weights and numbers) the gain data Gi read from the multiplier 28.

The output G i XX i of the multiplier 28 is the output of the accumulator 2
9 is input.

The accumulator 29 sequentially accumulates the output G7XXi of the multiplier 28 using the adder 30 and the register 35 in synchronization with the clock C5. Immediately after the clock C1 rises and falls, the cumulative result for one cycle is Y=ΣGjXXf!・I is obtained, and this becomes the reverberation signal output OUT.

After that, during the period when the output of the counter 25 becomes 1, the clock C
3 becomes L level, and the AND circuit 31 of the accumulator 29 is cut off. In other words, the contents of the accumulator 29 are once cleared at this point, and accumulation for the next cycle is started again.

The above is the configuration and operation of the reverberation adding device shown in FIG. A problem with this type of convolution type reverberation adding apparatus is that the number of convolution operations that can be performed by serial sequential processing within a certain period of time is greatly limited by hardware. In the previous example, m-point convolution calculations are performed within one sampling period T, but the number of calculation points m cannot be freely increased due to constraints on calculation speed.

The fact that the number m of convolution calculation points cannot be increased means that reverberation with a very long delay time cannot be created. This is because the number of calculation points m is the number m of parameters Di and ai used in the calculation, and it is not possible to appropriately express reverberation characteristics extending to a long delay time with a small number of parameters.

This can be easily understood from Fig. 3 (A>).If the delay time data Dj is taken very roughly on the time axis,
This means that reverberation characteristics over a long period of time can be expressed with a small number of parameters. However, even if this satisfies the length of the reverberation, it does not satisfy the quality of the reverberation (this method is called parameter thinning). Achieving the desired natural reverberation generally requires a large number of parameters that are spaced sufficiently close together in time.

For these reasons, in most conventional systems, reverberation of only the initial delay time period of a desired reverberation characteristic is created by convolution calculation. As long as there is only a limited initial delay time period, the reverberation characteristics can be expressed with high density even with m parameters. However, in this case, the latter part (late delay time range) of the originally desired 10-reverberation characteristics over a large time range is ignored and omitted.

By the way, even if the reverberation in the late delay time period is omitted, if the input sound (input signal) is continuous, it will not cause much of a problem in terms of auditory sense. The reverberation level in the late delay time period is very low, so even if there is a minute level of reverberation in the latter half, it is masked by the continuous input sound and has only a slight effect on the auditory sense. Therefore, even if the reverberation in the latter half is omitted, there is no negative effect.

However, if the reverberation in the late delay time period is omitted, when the continuous input sound is interrupted, the subsequent reverberation effect will feel extremely unnatural.

Even minute level reverberations can be clearly recognized when the input sound is interrupted. As a result, the unnatural feeling that the reverberation, which should continue while attenuating, disappears midway becomes noticeable.

This invention was made against the above background, and its purpose is to provide reverberation addition using a convolution method that allows a more natural reverberation effect to be obtained under the constraint that the number of convolution calculation points is limited. The goal is to provide equipment. Specifically, the purpose is to generate natural reverberation that extends to a sufficiently large delay time when the input sound is interrupted.

- In order to achieve the objective described in F, this invention performs convolution operation on the initial delay time period of the reverberation characteristic (the latter delay time period is omitted) when the input signal is continuous. The feature is that when the sound is interrupted and the silent state continues, the delay time period that is the target of the blue-tinting calculation is gradually shifted to the latter half.

If the delay time period to be calculated is shifted to the latter half, the earliest delay time period will be loved by &IJ.

However, since the input signal during this period is O (silence), even if this is the subject of calculation, the result will be 0, and omitting this will have no practical effect at all. The amount omitted here can be effectively utilized to create more reverberations in the latter half.

Hereinafter, embodiments of the present invention will be described in detail based on the drawings.

FIG. 4 shows the configuration of a first embodiment of the invention. Many parts of the configuration of this reverberation adding apparatus are exactly the same as the apparatus shown in FIG. 1, and the common parts are given the same reference numerals. The structure and operation common to the apparatus shown in FIG. 1 will be referred to above, and the following description will focus on the parts of the apparatus shown in FIG. 4 that are different from those shown in FIG. 1. Each clock C5 is also generated at the timing shown in FIG.

Further, it is assumed that the number of convolution calculation points that can be performed by serial sequential processing within one sampling period T is m points, which is the same as that in FIG. 1 in the apparatus of the present invention.

In the device of the present invention, one parameter Df, Gi (i=1~m~
2) is stored. Parameters Di, G of this data
The reverberation characteristics extending to a large delay time are expressed by i. This situation is shown in FIGS. 5(A) and 5(B). In other words, the reverberation characteristics over a wide time period necessary to obtain the desired natural reverberation are expressed with a sufficiently high density by a large number of parameters, and are stored in the parameter memory 20. Further, in this embodiment, the input signal
As a level discrimination means for comparing the level of (t) with a minute reference level, there is an absolute value detection circuit 41 that detects the absolute value of the input signal x (1,). A comparator 42 for comparing the value N is provided. The comparator 42 clears the counter 43 when the input signal level is above the reference value N, and cancels the clearing when the input signal level is below the reference value N.

When the counter 7I3 is cleared by the output of the comparator 42, its output is 1 (not 0). When the counter 43 is cleared, the counter 43 is incremented by the clock C1, and its output increases in the order of 1→2→3→.

The counter 25 which provides the read address of the parameter memory 20 is cleared by the clock C4 (its output becomes 0), and the output of the counter 43 is read (preset) by the clock C2 after one period of the clock C5. ). and then clock C
5 is sequentially incremented.

In other words, the counter 43 is cleared and its output is 1.
Then, the output of the counter 25 becomes 0 in synchronization with clock C4, becomes 1 in synchronization with clock C2, and m -1 clocks C5 are generated until the plurality of clocks c1 are generated. It increases to m. The output of the counter 25 changes in the order of 0→1→2−>...m. This is exactly the same operation as the device shown in FIG.

Therefore, in the above state, among the seven parameters stored in the parameter memory 20, m parameters in the earliest delay time period are used to perform the convolution operation. Therefore, the reverberation signal output OUT at this time becomes exactly the same as the operation of the apparatus shown in FIG.

The above operation is an operation when the level of the input signal is continuously equal to or higher than the reference value N. In this operation, -14= ignores the second half of the reverberation characteristics stored in the parameter memory 20, but this is a continuous input point]
The reverberation effect is not significantly affected by the sound masking effect.

When the input sound stops, that is, when the input signal level becomes smaller than the reference value N, the counter 43 is cleared, and the counter 43 is incremented by the clock C1. When the counter 43 is incremented by 1, its output becomes 2, and in this state, the output of the counter 25 is 0→2→
It changes as 3→4→...m+1. In other words, parameter D
2. G2~Dm +1. The m parameters of Gm +1 are used to perform the convolution operation.

If the silence continues further, the output of the counter 43 becomes 3,
The output of the counter 25 changes as O→3→4→5→...m+2. This end is D3, 03~Dm+2. Gm
m parameters up to +2 are used to perform the convolution operation.

In this way, when the input signal level continues to be below the reference level, the delay time period for selecting the parameter ll1lla is sequentially shifted to the side where the delay time is larger. Then, the parameters of the earliest delay time period will be ignored in order from D+ and G+.

However, at this time, the input signal level continues to be silent, so the sample data of the input signal corresponding to the ignored parameter is almost O, and the result of 0xGi is also O, so this calculation is performed from the beginning. There is no need.

Instead of initially ignoring the IW delay time parameter,
Since the convolution operation is performed by sequentially using the parameters of the delay time period of parameters [)m, G'm C1, and descending, the parameters stored in the parameter memory 2o in the silent state after the incoming horn sound has stopped. According to the reverberation characteristics, reverberation that extends over a long delay time period is created, and natural reverberation that gradually decays over a long period of time is generated.

FIG. 6 shows the configuration of a second embodiment of the invention. This device differs from FIG. 4 in that the absolute value detection circuit 41 in FIG. 4 is replaced by a power calculation circuit 44. This power calculation circuit 44 squares the input signal X(t) using a squaring circuit, integrates its output in synchronization with a clock C5 using an integrating circuit consisting of an adder 46 and a register 7I7, and outputs the output via a register 48. The signal is configured to be applied to the comparator 42 in synchronization with the clock C1.

Note that when the latest input signal is received, it is necessary to recognize the power at that time, so in this embodiment, input data for a certain period in the past is temporarily stored in the flash memory
The power is calculated in advance based on the contents of this holding memory. These pre-memories 49 delay the number of tJ input data necessary for power calculation, and during this time, power is calculated from these data, and at the same time, input data corresponding to the data memory is sent out via the buffer register 50 to synchronize the timing. . For these controls, the aforementioned control signal C2 and the address designation signal line to the data memory are also used.

In other words, it is determined whether the input signal is interrupted (sound state or no sound state) based on the average value of the power of the input signal X'(t). According to this example,
More accurate sound detection according to the statistical properties of musical tone signals, etc.
Can perform silent state discrimination.

Note that the frequency band of the incoming J signal may be divided into several parts, the reverberation adding device of the present invention may be provided for each band, and the outputs thereof may be combined to form the R final reverberation signal.

In this case, appropriate reverberation characteristics can be set for each frequency band. As is well known, high-frequency components of reverberant sound decay quickly and have a short reverberation time, while low-frequency components decay slowly and have a long reverberation time. A reverberation effect that matches the actual reverberation characteristics can be achieved by providing a reverberation load device for each of a plurality of bands.

In particular, when the present invention described above is applied to a system in which reverberation devices are provided separately for a plurality of frequency bands, the following effects are added. In other words, it is assumed that the low-frequency component of the musical tone signal has disappeared, or that the high-frequency component continues or is intermittent. Even in this case, according to the above configuration, the discontinued low frequency component 18- is not affected by the continuing high frequency component at all,
The low-frequency component alone produces a long-lasting natural reverberation.

As described in detail above, according to the present invention, even under the constraint that the number of convolution calculation points is limited in terms of hardware, a more natural reverberation effect over a wide reverberation time zone can be obtained.

[Brief explanation of the drawing]

Fig. 1 is a block diagram of a conventional reverberation adding device, Fig. 2 is a timing chart showing the operation of the device shown in Fig. 1, and Fig. 3 is an explanatory diagram of the reverberation characteristics stored in the parameter memory in the device shown in Fig. 1. , FIG. 4 is a block diagram of the first embodiment of the reverberation adding device according to the present invention, and FIG. 5 is an explanatory diagram of the reverberation characteristics stored in the parameter memory in the device of FIG. 4.
FIG. 6 is a block diagram showing a second embodiment of the reverberation adding device according to the present invention. 20...Parameter memory 23...Data memory 29...Accumulator 41...Absolute value detection circuit 42...Comparator 43...Counter 44...Power calculation circuit Patent applicant Nippon Musical Instrument Manufacturing Co., Ltd. 708

Claims (1)

[Claims] (1) A data memory that reads input signal data at a constant sampling period T and stores data for a predetermined period of time while sequentially updating; a parameter memory that stores reverberation characteristics expressed by multi-point delay data and gain data corresponding to each point; each time one sample data is written to the data memory, data is transferred from the parameter memory within the time T; Read out a predetermined number m of parameters that are part of it,
Among the input signal data stored in the data memory, a convolution operation is performed in which m pieces of data specified by the parameter are given a weight specified by the parameter and added, and the result is used as a reverberation signal. Convolution calculation means for outputting an output every time T; Level discrimination means for comparing the input signal level with a minute reference level; When the input signal level is equal to or higher than the reference level, the processing target of the convolution operation means is The above m parameters are selected from the earliest delay time period, and when the input signal level continues to be below the reference level, the delay time periods from which the m parameters are selected are sequentially moved to the side with larger delay time. A reverberation adding device comprising: means for shifting a processing target time period; and;