JPH056171A - Sound wave compression method - Google Patents

Abstract

(57) [Summary] [Object] The present invention, when the ADPCM method is adopted as the sound source processing method, makes it possible to minimize the noise component generated in the tone signal during the sound source processing due to the quantization error. It is an object to provide a sound wave compression technique. In the adaptive quantization processing of the musical tone signal of FIG. 20 executed by a control device such as a general-purpose computer, a cumulative peak value R (ADR) which is an actual musical tone signal that will be obtained by the sound source processing on the reproducing side. It is determined whether or not the absolute value of exceeds the maximum value that can be represented by the number of bits assigned to the tone signal (S ₂₀₀₇ ). If the determination is YES, the value of the current adaptive quantized difference data rom (ADR) is lowered to the next lower quantization level (S ₂₀₁₂ → S ₂₀₁₃ or S ₂₀₁₂ →
S ₂₀₁₄ ). As a result, the absolute value of the cumulative peak value on the reproducing side is controlled so as not to exceed the maximum value, and the generation of noise in the sound source processing is controlled.

Description

Detailed Description of the Invention

[0001]

BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention generates adaptively quantized tone signal difference data used in a tone generator for executing a sound source processing program by an adaptive differential pulse code modulation method to generate a tone signal. The present invention relates to a musical sound waveform compression method for performing the operation.

[0002]

2. Description of the Related Art Various electronic musical instruments with good performance have been realized by the development of digital signal processing technology and LSI processing technology. In particular, with the price reduction of semiconductor memory, musical tone waveforms of natural musical instruments, etc. are stored in the memory as digital signals,
The appearance of electronic musical instruments that can produce realistic musical tone waveforms by reading this with the pitches etc.
It contributes to a large increase in the music population regardless of amateur.

Examples of such a sound source system include a PCM (pulse code modulation) system and a DPCM (differential pulse code modulation) system.
And ADPCM (adaptive differential pulse code modulation). In particular, the ADPCM method is an excellent sound source method because large data compression is possible.

Since the tone wave generator that employs the tone generator system as described above requires a large amount of high-speed digital calculation, conventionally, an architecture equivalent to the tone generator algorithm based on the required tone generator system is implemented by hardware. It was composed of a dedicated sound source circuit that was realized. But,
The problem is that the scale of the hardware becomes large, and when it is realized by an LSI, in terms of yield at the time of manufacturing LSI chips, it causes an increase in cost at the manufacturing stage and leads to an increase in size of the musical tone waveform generator. Had.

On the other hand, in recent years, many high-performance microprocessors for performing general-purpose data processing have been realized, and it is possible to perform sound source processing by software using such a microprocessor. For example, by performing parallel processing with a plurality of processor configurations, it is possible to perform processing of an advanced sound source method such as the ADPCM method.

[0006]

When the ADPCM method is adopted as the sound source processing method, the adaptively quantized difference data is stored in the waveform memory or the like, and inverse quantization is performed while reading this. Thus, the difference value of the tone signal is decoded, and the difference value is accumulated at each timing to generate the tone signal.

The above-mentioned adaptively quantized difference data is P
This can be obtained by calculating a differential signal with respect to the CM tone signal and performing adaptive quantization in which the amplitude level of the differential signal is compressed and quantized.

Here, in the process of generating the adaptively quantized difference data, when the amplitude level of the difference signal is compressed and quantized, a quantization error having a maximum half width of each quantization width occurs. To do. Then, in the sound source processing, dequantization is performed based on the difference data including such a quantization error, and when the tone signal is generated by accumulating the difference values obtained thereby, the obtained tone signal is It is possible that the amplitude value slightly exceeds the maximum or minimum amplitude value that can be represented by the number of bits of the digital signal assigned to the musical tone signal by an amount corresponding to the quantization error.

In such a case, if no measures are taken,
Even if the maximum or minimum amplitude value is slightly exceeded, inversion of the sign or the like occurs in the case of a digital signal. That is,
Even if the amplitude value is slightly larger than the maximum positive value, the digital amplitude value folds back to the negative amplitude value side. On the contrary, when the amplitude value becomes slightly smaller than the negative minimum value, the digital amplitude value folds back to the positive amplitude value side.

When such a situation occurs, the waveform of the musical tone signal obtained becomes a waveform which is far from the actual change of the waveform, and when the musical tone signal is directly produced, a large noise is generated. Therefore, there is a problem that the sound quality of the musical tone signal is significantly deteriorated.

The present invention uses ADPCM as a sound source processing method.
It is an object of the present invention to provide a tone wave type compression technique capable of minimizing a noise component generated in a tone signal during a sound source processing due to a quantization error when the method is adopted.

[0012]

SUMMARY OF THE INVENTION The present invention provides a tone compression method for generating adaptively quantized difference data used in a tone generator for generating tone signals by an adaptive differential pulse code modulation method. As a premise, the following processes are executed.

First, a difference calculation process for calculating the difference between the value of the digital tone signal at the current timing and the accumulated value of reproduced difference values described later up to the previous timing to obtain the difference value at the current timing is performed. To be executed.

Next, adaptive quantization is performed while compressing the difference value at the current timing obtained in the difference calculation process to obtain the adaptive quantized difference data at the current timing. .

Further, the adaptively quantized difference data at the current timing obtained in the adaptively quantizing process is inversely quantized,
An inverse quantization process for calculating the reproduced difference value at the current timing is executed.

Subsequently, the reproduced difference value at the current timing obtained in the dequantization process is accumulated to the accumulated value of the reproduced difference values up to the previous timing, and reproduced up to the current timing. An accumulation process is performed to obtain an accumulated difference value.

Whether or not the accumulated value of the reproduced difference values up to the current timing obtained in the accumulating process exceeds the maximum change width of the number of bits assigned to the musical tone signal to generate the musical tone signal. A determination process for determining whether or not is executed.

Further, if it is determined by the determination process that the accumulated value of the reproduced difference values up to the current timing does not exceed the maximum change width, the current timing obtained by the adaptive quantization process is obtained. The output process of outputting the adaptively quantized difference data of is executed.

In addition, if it is determined in the determination process that the accumulated value of the reproduced difference values up to the above-mentioned current timing exceeds the maximum change width, the current value obtained in the adaptive quantization process is obtained. The value of the adaptive quantized difference data at the timing of is changed so that the accumulated value of the reproduced difference values up to the current timing obtained in the accumulation process based on the value does not exceed the maximum change width. Then, the difference data changing process for repeating the above-described inverse quantization process, accumulation process and determination process is executed. The changing operation in this case is, for example, an operation of changing the absolute value of the adaptively quantized difference data at the current timing to a data value that is smaller by one data value.

Then, a repeating process is performed in which the above processes are repeated at each generation timing of the tone signal.

[0021]

According to the present invention, in the inverse quantization process and the accumulation process, the reproduction process similar to the sound source process is performed on the adaptively quantized difference data at the current timing obtained in the adaptive quantization process. The accumulated value of the reproduced difference value at the current timing, which is similar to the peak value of the musical tone signal that may be obtained by the sound source processing, is obtained.

Then, by using the accumulated value corresponding to the reproducing side to execute the above-mentioned difference calculation process,
In the operation of calculating the difference value of the musical sound signal, the error due to the adaptive quantization can be absorbed.

Further, particularly in the present invention, it is obtained in the adaptive quantization process by determining whether or not the accumulated value corresponding to the reproducing side exceeds the maximum change width of the number of bits assigned to the tone signal. When sound source processing is executed based on the adaptively quantized difference data at the current timing,
It is possible to determine whether or not the sign inversion of the output value has occurred.

When it is determined in the determination process that the accumulated value exceeds the maximum change width described above, the difference data change process is the adaptive quantization of the current timing obtained in the adaptive quantization process. The value of the obtained difference data is changed so that the accumulated value obtained based on the value does not exceed the maximum change width, and the above-mentioned inverse quantization process, accumulation process, judgment process and output process are repeated. Let

By outputting the adaptively quantized difference data at the current timing thus changed, it is possible to prevent in advance the sign inversion of the output value when the sound source processing is executed. Therefore, the generation of noise can be minimized.

[0026]

Embodiments of the present invention will be described below with reference to the drawings. Configuration of this Embodiment FIG. 1 is an overall configuration diagram of an embodiment of the present invention.

In the figure, first, the entire apparatus is controlled by the microcomputer 101. In particular, not only the control input process of the musical instrument but also the process of generating a musical tone is executed by the microcomputer 101, and the tone generator circuit for generating a musical tone is not required.

A switch section 104 comprising a keyboard 102 and function keys 103 is an operation input section of a musical instrument, and performance information input from the switch section 104 is processed by the microcomputer 101.

The tone signal after the analog conversion generated by the microcomputer 101 is smoothed by the low-pass filter 105, amplified by the amplifier 106, and then the speaker 10.
Sound is emitted via 7. The power supply circuit 108 supplies necessary power to the microcomputer 101, the low pass filter 105 and the amplifier 106.

Next, FIG. 2 shows a microcomputer 101.
3 is a block diagram showing the internal configuration of FIG. The control data / waveform ROM 212 stores tone control parameters such as an envelope value target value, which will be described later, tone difference waveform data and quantized data in the ADPCM (adaptive differential pulse code modulation) system. Then, the command analysis unit 207 sequentially analyzes the contents of the program in the control ROM 201, while controlling the control data / waveform ROM 212.
Each of the above data is accessed to perform sound source processing by software.

The control ROM 201 stores a musical tone control program which will be described later, and the ROM address control unit 205 sequentially outputs program words (commands) at designated addresses via the ROM address decoder 202. Specifically, the word length of each program word is, for example, 28
Bit is a part of the program word, and RO is used as the lower part (address within page) of the address to be read next.
It is a next address method that is input to the M address control unit 205. Of course, a normal program counter type CPU may be used.

The command analysis section 207 has a control ROM 2
The operation code of the instruction output from 01 is analyzed, and a control signal is sent to each part of the circuit to execute the specified operation.

The RAM address control unit 204 has a control R
When the operand of the instruction from the OM 201 specifies a register, the address of the corresponding register in the RAM 206 is specified. The RAM 206 contains the data shown in FIGS.
For example, various tone control data to be described later are stored for eight sounding channels, and various buffers to be described later are stored for use in sound source processing to be described later.

When the instruction from the control ROM 31 is an operation instruction, the ALU unit 208 and the multiplier 209 execute addition and subtraction and logical operation, and the latter execute multiplication, based on the instruction from the command analysis unit 207.

The interrupt control unit 203, on the basis of a hard timer (not shown) inside, periodically
ROM address control unit 205 and D / A converter unit 213
Supply an interrupt signal to.

The switch section 104 of FIG. 1 is connected to the input port 210 and the output port 211. RO for control
Various data read from the M201 or the RAM 206 is sent to the ROM address control unit 205 and the ALU via the bus.
Unit 208, multiplier 209, control data and waveform ROM 2
12, the D / A converter unit 213, the input port 210 and the output port 211. In addition, the ALU unit 208,
The outputs of the multiplier 209 and the control data / waveform ROM 212 are supplied to the RAM 206 via a bus.

Next, FIG. 4 shows the D / A converter section 21 of FIG.
3 shows an internal configuration of the third example, in which one sample data of the musical sound created by the sound source processing is latched via the data bus.
01 is input. When a sound source processing end signal is input from the command analysis unit 207 of FIG. 2 to the clock input of the latch 301, one sample of musical sound data on the data bus is latched in the latch 301 as shown in FIG. .

Here, the time required for the above sound source processing is
The timing at which the tone generation process ends and the musical tone data is latched in the latch 301 is not constant because it changes depending on the execution conditions of the tone generation software. Therefore, as shown in FIG. 3, the output of the latch 301 is directly applied to the D / A converter 3.
03 cannot be entered.

Therefore, in this embodiment, as shown in FIG. 4, the output of the latch 301 is further latched by the latch 401, and the musical tone signal is latched by the interrupt signal output from the interrupt control unit 203 shown in FIG. It is made to latch by 401, and it is made to output to the D / A converter 303 at fixed intervals.

Since the change in the processing time in the sound source system is absorbed by using the two latches in this manner, a complicated timing control program for outputting the musical sound data to the D / A converter becomes unnecessary. Overall Operation of this Embodiment Next, the overall operation of this embodiment will be described.

In this embodiment, the microcomputer 10
1 as shown in the main flowchart of FIG.
₆₀₂ repeatedly performs a series of processes to S _610. Then, the actual sound source processing is performed by interrupt processing. Specifically, the program executed as the main flowchart of FIG. 6 is interrupted at regular intervals, and the program of the sound source processing for producing the tone signal of 8 channels is executed based on the interrupt. When the processing is completed, the musical tone waveforms for 8 channels are added and output from the D / A converter unit 213 connected to the microcomputer 101. After that, the process returns from the interrupt state to the main flow. It should be noted that the above-mentioned interrupt is periodically performed based on the hard timer in the interrupt control unit 203 in FIG. This cycle is equal to the sampling cycle when outputting a musical sound.

The above is the outline of the operation of this embodiment. Next, the overall operation of this embodiment will be described in detail with reference to FIGS. The main flowchart of FIG. 6 shows the flow of processes other than the sound source process, which is executed by the microcomputer 101 in a state where the interrupt is not interrupted by the interrupt control unit 203.

First, the power is turned on, and the initialization of the contents of the RAM 206 (see FIG. 2) in the microcomputer 101 (S ₆₀₁ ). Next, each switch of the function key 103 (see FIG. 1) connected to the outside of the microcomputer 101 is scanned ( _S602 ), and the state of each switch is fetched from the input port 210 into the key buffer area in the RAM 206. As a result of the scanning, the function key whose state has changed is identified, and the corresponding function is processed ( _S603 ). For example, the tone number is set, the envelope number is set, and if the additional function is rhythm performance, the rhythm number is set.

After that, the keyboard key pressed on the keyboard 102 of FIG. 1 is taken in as in the case of the above-mentioned function key (S ₆₀₄ ), and the changed key is identified to perform the key assigning process ( S ₆₀₅ ).

Next, when a demo performance key (not shown) is pressed with the function key 103 (FIG. 1), demo performance data (sequencer data) is sequentially read from the control data / waveform ROM 212 of FIG. Then, key assignment processing and the like are performed (S ₆₀₆ ). When the rhythm start key is pressed, the rhythm data is the control data and waveform R
The data is sequentially read from the OM 212, and key assignment processing and the like are performed (S ₆₀₇ ).

After that, the timer process described below is performed (S ₆₀₈ ). That is, the time value of the time data incremented in the interrupt timer process (S ₇₀₂ ) to be described later is determined, and the sequencer data for time control is sequentially read for demo performance control or the time control is read for rhythm performance control. The time control for performing the demo performance of _S606 or the rhythm performance of _S607 is performed by comparing with the rhythm data of _S607 .

Further, in the tone generation process _S609 , a pitch envelope process is performed in which an envelope is added to the pitch of the musical tone to be tone-generated and pitch data is set in the corresponding tone generation channel.

Furthermore, a flow one round preparation process is executed.
(S ₆₁₀ ). In this process, alter or change the state of the sound channel of the note number that became a key-start in keyboard key processing of S ₆₀₅ in the key depression, the state of the sound channel of the note number that became a key release in the silencer Etc. are processed.

Next, the interrupt processing of FIG. 7 will be described. By the interrupt control unit 203 of FIG.
When the program corresponding to the main flow of 1 is interrupted, the processing of the program is interrupted and the execution of the interrupt processing program of FIG. 7 is started. In this case, in the interrupt processing program, the registers and the like to be written by the main flow program of FIG. 6 are controlled so that the contents are not rewritten. Therefore, it is not necessary to save and restore the registers at the start and end of the normal interrupt processing.
As a result, the transition between the processing of the main flowchart of FIG. 6 and the interrupt processing is performed quickly.

Subsequently, sound source processing is started in the interrupt processing (S ₇₀₁ ). This sound source processing is shown in FIG. As a result, musical tone waveform data obtained by accumulating eight sounding channels is obtained in a buffer B, which will be described later, in the RAM 206 of FIG.

[0051] In addition, in S _702, interrupt timer processing is performed. Here, the value of the time data (not shown) on the RAM 206 (FIG. 2) is incremented by utilizing the fact that the interrupt processing of FIG. 7 is executed at constant sampling intervals. That is, the elapsed time can be known by looking at the value of this time data. The time data thus obtained is used for the time control in the timer process S ₆₀₈ of the main flow of FIG. 6, as described above.

Then, in _S703 , the contents of the buffer area are latched in the latch 301 (FIG. 4) of the D / A converter unit 213. Next, the operation of the sound source process executed in step _S701 of the interrupt process will be described with reference to the flowchart of FIG.

First, the area of the RAM 206 for waveform data addition is cleared (S ₈₀₁ ). Next, sound source processing is performed for each of the sound generation channels (S _{802 to} S ₈₀₉ ),
Finally, when the sound source processing of the 8th channel is completed, waveform data in which 8 channels are added to the predetermined buffer area B is obtained. These detailed processes will be described later.

Next, FIG. 9 is a flow chart conceptually showing the relationship of the processes in the flow charts of FIGS. 6 and 7. First, a process A (hereinafter, the same applies to B, C, ..., F) is performed (S ₉₀₁ ). This "processing" corresponds to, for example, "function key processing" or "keyboard key processing" in the main flowchart of FIG. After that, the interrupt processing is started, and the sound source processing is started ( _S902 ). This allows
A musical tone signal obtained by collecting eight sounding channels for one sample is obtained and output to the D / A converter unit 213. Then, the process returns to some process B in the main flowchart. Above operation is, the sound source processing for all eight sound channel is repeated while made (S ₉₀₄ ~S _911). Then, this repetitive processing is continued while the musical tone is being sounded. Data Structure in Sound Source Processing Next, a specific example of the sound source processing executed in S ₇₀₁ of FIG. 7 will be described.

In this embodiment, the microcomputer 10
It has been described above that 1 is responsible for sound source processing for 8 sounding channels. The data for the sound source processing for eight channels has a data format as shown in FIG.
It is set in an area for each sound generation channel in 206.

A buffer B for waveform accumulation as shown in FIG. 11 is secured in the RAM 206. In this case, various control data for sound source processing based on the ADPCM system is stored in each sound generation channel area of FIG.
In the figure, A represents an address designated when adaptive quantized difference data (described later) is read during sound source processing, and A _I is an integer part of the current address, and is also used in the control data / waveform ROM 212 (see FIG. 2). ) Directly corresponds to the address in which the adaptive quantized difference data of () is stored. Also, A _F
Is a fractional part of the current address, which is used for interpolation of the dequantized difference value read from the ROM 412. Next, P _I represents the integer part of the pitch data, and P _F represents the fractional part of the pitch data. As an example, P _I = 1 and P _F = 0 are pitches of the original sound, P _I = 2 and P _F = 0 are pitches one octave higher, and P _I = 0 and P _F = 0.5. Represent pitches one octave below. Various other control data will be described in detail when the ADPCM method is described later.

In the present embodiment, when the main flow of FIG. 6 is executed, control data necessary for sound source processing, such as pitch data and envelope data, is set in the corresponding sounding channel area. Then, in the sound source processing corresponding to each channel in FIG. 8 which is executed as the sound source processing in the interrupt processing in FIG. 7, the musical tone generation processing is executed while using the various control data set in the sound generation channel area. It As described above, the data communication between the main flow program and the sound source processing program is performed by RA
It is performed via the control data (tone generation data) of the sound generation channel area on the M206, and since the sound generation channel area in each program need only be accessed regardless of the execution state of the other program, both programs are effectively executed. The module structure can be independent, and the program structure can be simple and efficient.

Further, the control data / waveform ROM 212 is used.
17 stores adaptive quantized difference data (described later) in a data format not shown in FIG.
A decompression table as shown in FIGS. 18A and 18B is stored.
As will be described later, this table is data used for inverse quantization of the adaptive quantized difference data read from the control data / waveform amount ROM 212. In addition, the ROM 212 stores control data for tone generation corresponding to each tone color, and when a certain tone color is set by the performer, the ROM 212 stores the RAM 206 in the RAM 206.
The above control data is transferred to each of the above-mentioned sounding channel regions,
Is set. Principle of Sound Source Processing by ADPCM Method Hereinafter, the processing is performed using such a data structure as shown in FIG.
The sound source processing of the ADPCM method, which is the sound source processing for each channel (any one of S _{802 to} S ₈₀₉ ) will be described. The sound source processing is performed by the microcomputer 10
The first command analysis unit 207 is realized by interpreting and executing the sound source processing program stored in the control ROM 201. Below, unless otherwise stated,
The processing is performed under this premise.

First, an outline of the operating principle of the ADPCM system will be described. First, in FIG. 12, the sample data X _P corresponding to the address A _I for control data and the waveform ROM 212 (Fig. 2) is, prior to one of the addresses A _I, the address (A _I -1) which is not shown in FIG. It is a value obtained from the difference value from the corresponding sample data.

[0060] The address A _I for control data and a waveform ROM 212, and is written adaptive quantized differential data for obtaining a difference value D between the next sample data, the difference value D is determined from this data. Thus, the sample data of the next address Motomari in X _P + D, which replaces the new sample data X _P.

In this case, if the current address is A _F as shown in the figure, the sample data corresponding to the current address A _F can be obtained by X _P + D × A _F. In this way, AD
In the PCM system, the adaptive quantized difference data for obtaining the difference value D between the current address and the sample data corresponding to the next address is the control data / waveform ROM 212.
(FIG. 2), the difference value D is calculated based on it, and the difference value D is added to the current sample data to obtain the next sample data, whereby the waveform data is sequentially created.

When such an ADPCM system is adopted,
When quantizing a waveform such as a voice or a musical sound whose difference value between adjacent samples is generally small, it is obvious that the quantization can be performed with a smaller number of bits as compared with the normal PCM method.

Further, in the ADPCM system, the following subtle principles of adaptive quantization are adopted.
Now, for example, when storing a difference value of a musical tone waveform in a memory, in order to quantize a difference value that can take an amplitude value (for example, a voltage value) of maximum ± E while securing a constant S / N, n
It is assumed that the number of quantized bits is required. On the other hand, when the data amount of m bits, which is less than n bits per sample data, can be assigned due to the memory capacity limitation, the difference value must be quantized by m bits per sample.

Therefore, normally, the range of the amplitude value of ± E is divided into 2 ^m, but this increases the quantization width and deteriorates the S / N. Therefore, in the ADPCM method, when a difference value such as a tone waveform is stored in a memory, a certain S / N can be secured, and the absolute value of the amplitude value that can be expressed in m bits is smaller than | ± E | Define the range of | ± e |. Then, the difference value whose absolute value of the amplitude value is larger than | ± e | is divided by a normalization coefficient of 1 or more and compressed so as to be within the range, and then quantized by m bits,
Store it in memory. As a result, it is possible to reduce the number of quantization bits per sample data while securing a constant S / N, and it is possible to reduce the memory capacity.
Such an operation is called adaptive quantization.

Here, in order to reproduce the original difference value from the difference data adaptively quantized as described above, the normalization coefficient at the time of adaptive quantization of each sample data is also stored. Incidentally, when the adaptively quantized difference data is read from the memory, the corresponding normalization coefficient must be read together, and the adaptively quantized difference data must be multiplied to reproduce the original difference value. . This operation is called inverse quantization.

However, if the above-mentioned normalization coefficient is stored for each sample data, after all, a storage capacity equivalent to that when the difference value in the range of the amplitude value ± E is quantized by n bits is required. However, data compression cannot be realized.

Therefore, an embodiment of the sound source processing by the ADPCM system which can suppress the increase of the storage capacity will be described below. Example of Sound Source Processing by ADPCM Method In this example, the fact that the distribution of the amplitudes of the difference values of the musical tone waveform is a distribution peculiar to the musical tone waveform in advance is utilized, and instead of the normalization coefficient, each amplitude of the difference value 17 (a) and FIG. 18 (a) showing which compression values are to be compressed, a compression characteristic table as described later is determined, and when the difference value is stored in the memory (212 in FIG. 2), this compression table It is assumed that the adaptive quantization is performed and the data is stored while being compressed. At the time of sound source processing, dequantization is performed while decompressing the difference value that has been adaptively quantized by the expansion characteristic that is the inverse characteristic of the compression characteristic, the original difference value is reproduced, and the original difference value is accumulated to form the tone waveform. You can play. Therefore, the control data / waveform ROM 212 is used.
(FIG. 2) is shown in FIG.
(b) A decompression table, which will be described later with reference to FIG. 18 (b), is stored.

The operation of the embodiment of the sound source processing of the ADPCM system will be described with reference to the operation flowcharts of FIGS. Each variable in the flow is secured in the corresponding sounding channel area on the RAM 206 in the microcomputer 101 of FIG. 2 in the data format of FIG.

The sound source processing based on FIGS. 13 to 15 is roughly divided into envelope processing (S _{1301 to} S ₁₃₀₇ ) and waveform processing (S _{1308 to} S ₁₃₂₄ ). First, FIG.
Also, the envelope processing will be described before the waveform processing based on the principle of the ADPCM system of FIG.

FIG. 16 is a diagram showing an envelope generated by the envelope processing. The envelope value E generated by the processing of S _{1301 to} S ₁₃₀₇ is multiplied by the musical tone waveform output O in step S ₁₃₂₃ , which will be described later, to add an envelope to each musical tone waveform sample data.

The envelope added to each tone waveform data is composed of several steps (segments) in terms of time. In the figure, an example of four steps is shown. In the figure, Δx is the envelope sampling period, and Δy is the change width of the envelope value.

In the envelope processing S _{1301 to} S ₁₃₀₇ , the envelope value E is calculated for each sampling timing,
A check is made as to whether that value has reached the target envelope value OE for the current step. Then, when E reaches OE, it is detected in the sound generation processing of _S609 in the main flow of FIG. 6, and the data (Δx, Δy and the target envelope value OE) for the envelope of the next step is displayed. Control data and waveform ROM 2
It is read out from the memory 12 and set in the corresponding tone generation channel area (see FIG. 10) on the RAM 206 (FIG. 2).

Specifically, in S ₁₃₀₁ , the timer value Δx _t for comparison with the envelope calculation cycle Δx is incremented at each interrupt timing. Then S
_{At 1302} , it is determined whether Δx matches Δx _t .

If they do not match, the envelope process is not performed. When it is determined in S ₁₃₀₂ that Δx matches Δx _t , the sign bit of the envelope value change width Δy is determined in S ₁₃₀₃ .

If the sign bit is positive and the determination in S ₁₃₀₃ is YES, that is, if the envelope is rising, the change width Δy is added to the current envelope value E in step S ₁₃₀₄ .

On the contrary, when the sign bit is negative and the determination in S ₁₃₀₃ is N
O, see i.e., envelope in the case of descending, in step S _1305, the variation width Δy from a current envelope value E is subtracted.

Thereafter, in S ₁₃₀₆ , it is judged if the current envelope value E has become equal to or larger than the target envelope value OE. When E becomes equal to or larger than OE, the current envelope value E is replaced with the target envelope value OE.

As described above, this is detected by the sound generation processing of _S609 in the main flow of FIG. 6, and the data for the envelope of the next step is set on the RAM 206. When 0 is detected as the current envelope value E in the tone generation process, the tone generation ends.

Next, the waveform processing of S _{1308 to} S ₁₃₂₄ based on the principle of the ADPCM system of FIG. 12 will be described. Among the addresses in the control data / waveform ROM 212 (FIG. 2) in which the adaptive quantized difference data is stored, the address in which the data to be currently processed is stored is (A _I, A) in FIG. _F ).

First, the pitch data (P _I, P _F ) is added to the current address (A _I, A _F ) (S ₁₃₀₈ ). This pitch data corresponds to the key number that is pressed on the keyboard 102 of FIG.

Then, the integer part A _I of the added address
It is determined whether there is a change in (S ₁₃₀₉ ). Judgment is N
If it is O, the difference data D at the address A _I in FIG. 12 is used to calculate the interpolation data value O corresponding to the fractional part A _F of the address by the calculation process of D × A _F (S ₁₃₂₁ ). The difference value D is obtained by the sound source processing at the interrupt timing before this time (see S ₁₃₁₄ or S ₁₃₁₇ described later).

Next, the sample data X _P corresponding to the integer part A _I address to the interpolation data value O is added, the current address (A _I, A _F) new sample data corresponding to O (in FIG. 12 X (Corresponding to _Q ) is obtained ( _S1322 ).

Thereafter, this sample data is multiplied by the envelope value E obtained by the envelope processing described above (S ₁₃₂₃ ), and the obtained contents of O are stored in the waveform data buffer B (FIG. 11) in the RAM 206 (FIG. 2). (See ₁₃ ). In this buffer B, the musical tone waveform outputs generated by the sound source processing (FIGS. 8S _{802 to} S ₈₀₉ ) for the other sounding channels are accumulated, and finally, one channel worth of data is accumulated for one sample. The musical tone waveform data of is generated.

Then, returning to the main flow of FIG.
Interrupt is applied at the sampling cycle of
~ The operation flow chart of the sound source processing in Fig. 15 is again real.
The current address (A_I,A_F) To pitch data
(P_I,P_F) Is added (S ₁₃₀₈).

The above operation is repeated until a change occurs in the integer part A _{I of the} address. During this time, the sample data _XP and the difference value D are not updated, only the interpolation data O is updated according to the address _AF, and new sample data _XQ is obtained each time.

[0086] Next, in S ₁₃₀₈ , the current address (A
As a result of adding the pitch data (P _I, P _F ) to _I, A _F ) and the integer part A _I of the current address changes (S ₁₃₀₉ ), the address A _I reaches or exceeds the end address A _E. It is determined whether or not ( _S1310 ).

When the determination is NO, the following S _{1311 to} S ₁₃₁₄
By the loop processing of, the sample data corresponding to the integer part A _I of the current address is calculated. That is, first, the variable before the old A _I (see FIG. 10) stores the value before the integer part A _I of the current address changes. this is,
It is realized by repeating the process of S ₁₃₁₃ or S ₁₃₂₀ described later.

While the value of the old A _I is sequentially incremented in S ₁₃₁₃ , the adaptive quantized difference data on the control data / waveform ROM 212 (FIG. 2) designated by the old A _I is read out in S ₁₃₁₄ . , The difference value D is calculated by the inverse quantization process. As described above, this dequantization is performed by using the adaptive quantized difference data as an input and the ROM 2
This is realized by directly decompressing the decompression table having the characteristics shown in FIGS. 17 (b) and 18 (b) stored on the memory 12, and the difference value D is calculated.

In the case of a musical tone waveform signal, its waveform characteristics are known in advance, so that FIG.
By preparing a plurality of sets of decompression tables as in the example of (a) or FIG. 18 (b) and storing them in the control data / waveform ROM 212, it is possible to store them based on the current size of the adaptive quantized difference data. It can be directly dequantized by the expansion table corresponding to the musical sound to be generated. For this reason, it is possible to perform much more accurate adaptive quantization as compared with the case of applying it to a signal such as a normal communication voice signal in which only the long-term average characteristics are known, and it is possible to perform high quality and high compression. It becomes possible to record / play back the tone signal at a certain rate.

In this case, the table selection data TBL indicating which decompression table is used for decompression corresponding to the tone color setting by the player is stored in the RAM 206 (see FIG. 6) in the function key processing S603 of the main flow of FIG. 6, for example. 2) Figure 10 above
It is set in each tone generation channel area shown in. And
During the decompression process, the decompression table corresponding to the table selection data TBL is the control data / waveform ROM 212.
Read from.

As described above, the difference value D is calculated in step S ₁₃₁₄ . Then, this difference value D is sequentially accumulated in the sample data X _P in step S ₁₃₁₂ .
The results above operation is repeated, when the value of the old A _I is equal to the integer portion A _I of the current address after the change, the sample data X _P value integer part A of the current address after change
_The value corresponds to _I.

In this way, the integer part A of the current address
If the sample data X _P corresponding to _I is obtained, the determination is YES in S _1311, proceeds to the processing of the aforementioned interpolated value (S _1321).

The sound source process described above is repeated at each interrupt timing, and if the determination in S ₁₃₁₀ changes to YES, the process for the next loop reproduction is started. The loop reproduction is a process executed when a predetermined waveform region of a musical tone signal is repeatedly and continuously sounded, and a preset muting instruction from the loop address A _L to the end address A _E is issued by the performer. It is a process that is repeatedly sounded until it occurs. As a result, it becomes possible to produce a sound for a long time without storing all the musical tone signals.

First, the address (A _I −A _E ) that exceeds the end address A _E is added to the loop address A _L , and the obtained address is the integer part A of the new current address.
Are _I (S _1315).

Hereinafter, depending on how much the address has advanced from the loop address A _{L, the} operation of calculating and accumulating the difference value D is repeated several times to correspond to the integer part A _I of the new current address. Sample data X _P is calculated.

That is, first, as the initial setting, the sample data X _P is set as the value of the sample data X _PL (see FIG. 10) at the preset loop address A _L , and the old A
_I is the value of the loop address A _L (S _1316).

A _L and X _PL are read in advance from the control data / waveform ROM 212 to each tone generation channel area of the RAM 206 (see FIG. 10). It should be noted that these may be set by the performer by means not particularly shown.

Then, the following processing of S _{1317 to} S ₁₃₂₀ is repeated. That is, while the value of the old A _I is sequentially incremented in S ₁₃₂₀ , the adaptive quantized difference data on the control data / waveform ROM 212 (FIG. 2) instructed by the old A _I is read out in S ₁₃₁₇ . _The difference value D is calculated by the inverse quantization (expansion) process similar to the case of ₁₃₁₄ .

Then, this difference value D is sequentially accumulated in the sample data X _P in S ₁₃₁₉ . As a result of repeating the above operation, when the value of the old A _I becomes equal to the integer part A _I of the current address after the change, the sample data X _P
Is the integer part A _I of the new current address after loop processing
Is the value corresponding to.

In this way, when the sample data X _P corresponding to the new integer part A _I of the current address is obtained, S
The determination of ₁₃₁₈ is YES, and the process proceeds to the above-described interpolation value calculation process ( _S1321 ).

As described above, A for one sound generation channel
Waveform data based on the DPCM method is generated. Example of Adaptive Quantization Processing of Musical Sound by ADPCM Method Next, in order to enable the above-described sound source processing, adaptive quantization (compression processing) is performed on a PCM musical sound signal to obtain adaptive quantized difference data. An example of processing will be described.

In the adaptive quantization processing adopted in this embodiment,
As described above, a compression table that determines which value each amplitude of the difference value of the PCM tone signal is compressed is determined, and the difference value is adaptively quantized and stored while being compressed by this compression table. Then, at the time of sound source processing, dequantization is performed while expanding the adaptively quantized difference value using the decompression table having the inverse characteristic of the compression characteristic, the original difference value is reproduced, and the original difference value is accumulated to form the tone waveform. To play.

Therefore, for each characteristic of the musical tone signal, for example, for each type of musical tone such as flute, cymbal, shakuhachi, for example, as shown in FIG.
7 (a) and a compression table as shown in FIG. 18 (a) are defined. In these figures, d (ADR) on the vertical axis is the difference value of the PCM tone signal, and G {d obtained by table conversion
(ADR)} is the adaptively quantized difference data.

For example, FIG. 17A shows a characteristic that the compression rate is large, and is used for compressing a musical tone signal such as a cymbal in which the PCM difference signal can have a large value. On the other hand, FIG. 18A shows a characteristic that the compression rate is small, and is used for compressing a musical tone signal such as a flute in which the PCM difference signal cannot have a very large value.

Further, a decompression table as shown in FIGS. 17 (b) and 18 (b) is determined in correspondence with each compression table described above. This decompression table has the inverse characteristics of the compression tables of FIGS. 17 (a) and 18 (a), and the adaptive quantized difference data rom (ADR) on the horizontal axis is decompressed and converted to the reproduction difference value J.
{Rom (ADR)} is obtained. As described above, this decompression table is the same as the control data / waveform ROM of FIG.
Although stored in 212 and used for sound source processing, it is also used in the following adaptive quantization processing.

After selecting the compression table as described above, the adaptive quantization process shown in the operation flowchart of FIG. 19 is executed to obtain the adaptive quantized difference data.
Note that this flow chart is for control data and waveform RO
Since it is a process for obtaining the adaptive quantized difference data stored in M212 (FIG. 2), it is executed by the control device such as a general-purpose computer (not shown), not by the instrument main body, and the compression table and decompression described above are performed. A table is also prepared on a memory (not shown). Further, it is assumed that a PCM musical tone signal as shown in FIG. 20 (a) is selectively stored in advance in a waveform memory (not shown). Further, for simplification of description, when a PCM tone signal of a certain musical instrument is selected, the signal is stored from address 0 to FFF (hexadecimal notation) on the waveform memory. Further, the symbol value used in the following description actually indicates the content of the variable provided on the RAM, but in the description, it will be described simply as a symbol indicating the value.

In FIG. 19, first, P on the memory is
The storage address value ADR of the CM tone signal is initially set to the leading address 0 of the tone signal (S ₁₉₀₁ ). Then, while the address value ADR is incremented by +1 in step S ₁₉₁₀ , the PCM musical tone signal is sequentially read until it is determined to be the final address FFF (S ₁₉₁₁ ), and the adaptive quantum in steps S _{1902 to} S ₁₉₁₄ below. The conversion process is executed.

That is, the PCM tone signal value PCM (AD is read from the address corresponding to the address value ADR.
R) is set as the input value H (ADR) (S ₁₉₀₂ ). Next, the cumulative peak value R (ADR-1) up to the previous time is subtracted from the value of the input value H (ADR) to obtain the current PCM difference value d (ADR) ( _S1903 ). Cumulative peak value R (A
DR-1) will be described later.

Next, the PCM difference value d (ADR) is
It is compressed and converted by the compression table selected in advance from FIG. 17 (a), FIG. 18 (a), etc., and the compression value G {d (AD
R)} is the current adaptive quantized difference data rom (AD) stored in the control data / waveform ROM 212 (FIG. 2).
R) (S ₁₉₀₄ ).

Then, the above-mentioned adaptive quantized difference data ro
m (ADR) is decompressed and dequantized by the decompression table corresponding to the compression table, and the converted value is set as the reproduction difference value K (ADR) (S ₁₉₀₅ ). Then, this value is added to the accumulator peak value R (ADR-1) up to the previous, current accumulated peak value R (ADR) is calculated (S _1906).

In these processes, a reproduction process similar to the sound source process is performed on the above-mentioned adaptive quantized difference data rom (ADR), and the current accumulated peak value R ( This is a process for obtaining ADR). And
Using the accumulated peak value R (ADR) corresponding to this reproduction side, in step S ₁₉₀₃ described above, the next PCM difference value d
By obtaining (ADR), it is possible to absorb the error due to the adaptive quantization.

Accumulated peak value R obtained as described above
In this embodiment, (ADR) is obtained as 12-bit data including positive and negative signs in sound source processing. Then, it is determined whether or not this absolute value exceeds the maximum value of 2047 ( _S1907 ).

If it does not exceed, it is further determined whether or not the current address ADR is equal to the preset loop address A _L for the above-mentioned loop reproduction (see FIG. 14) (S ₁₉₀₈ ).

If they are not equal, the address ADR is incremented by 1 (S ₁₉₁₀ ), and the process returns to the step S 2002 through the judgment of the step S ₁₉₁₁ . Here, if it is determined that the current address ADR is equal to the loop address A _L (S ₁₉₀₈
Is YES), the cumulative peak value R (ADR) at that time
Is set as the loop crest value X _PL and stored in the control data / waveform ROM 212 of FIG. 2 together with the loop address A _L. These data are used in the loop reproduction process of FIG. 14 described above.

In step S ₁₉₀₇ , the cumulative peak value R
The absolute value of (ADR) is 20 which is the maximum value of 12 bits.
If it is determined that the value exceeds 47, the current value of the adaptive quantized difference data rom (ADR) is lowered to the quantization level one level lower. That is, if rom (ADR) is a positive value, it is decremented by 1 (S ₁₉₁₂ → S ₁₉₁₃ ), and if it is a negative value, it is decremented by 1 (S ₁₉₁₂ → S ₁₉₁₄ ).

As described above, the accumulated peak value R (ADR) is an actual musical tone signal that will be obtained by the sound source processing on the reproducing side (musical instrument side). When accumulating while dequantizing, the absolute value of the accumulated peak value is 1 of the specified number of bits on the reproducing side due to the quantization error.
It is possible for the maximum value of 2 bits to be exceeded. Then, when this occurs, inversion of the sign or the like occurs and large noise occurs. Therefore, when the absolute value of the cumulative peak value R (ADR) that may be obtained on the reproduction side exceeds the maximum value, the current adaptive quantized difference data rom ( By lowering the quantization level of (ADR), the absolute value of the cumulative peak value on the reproducing side is controlled so as not to exceed the maximum value.

After the above processing, the above-mentioned step S is performed again.
_The above processing is repeated until ₁₉₀₅ and S ₁₉₀₆ are recalculated and the determination in step S ₁₉₀₇ becomes NO. Normally, the absolute value of the cumulative peak value R (ADR) can be made equal to or less than the above-mentioned maximum value only by lowering the quantization level by one level.

By the adaptive quantization processing described above, for example, the 12-bit PCM musical tone signal of FIG. 20 (a) can be compressed into a 4-bit ADPCM musical tone signal.

[0119]

According to the present invention, by the dequantization process and the accumulator, the accumulated value of the reproduced difference value at the present timing similar to the crest value of the musical tone signal which will be obtained by the sound source processing. Can be obtained, and by determining whether or not the accumulated value corresponding to the reproduction side exceeds the maximum change width of the number of bits assigned to the musical tone signal, the adaptation of the current timing obtained in the adaptive quantization process can be obtained. When the sound source processing is executed based on the quantized difference data, it is possible to determine whether or not the sign inversion of the output value has occurred.

Then, based on the determination result of this determination process, the difference data changing process changes the value of the adaptively quantized difference data at the current timing, so that the output value of the output value is changed when the sound source processing is executed. The occurrence of sign inversion can be prevented in advance, and the generation of noise can be minimized.

[Brief description of drawings]

FIG. 1 is an overall configuration diagram of a present embodiment.

FIG. 2 is an internal configuration diagram of a microcomputer.

FIG. 3 is a configuration diagram of a conventional D / A converter unit.

FIG. 4 is a configuration diagram of a D / A converter unit according to the present embodiment.

FIG. 5 is a timing chart in D / A conversion.

FIG. 6 is an overall operation flowchart of the present embodiment.

FIG. 7 is an operation flowchart of interface processing.

FIG. 8 is an operation flowchart of sound source processing.

FIG. 9 is a conceptual diagram showing the relationship between a main operation flowchart and interrupt processing.

FIG. 10 is a diagram showing a storage area for each sound generation channel on a RAM.

FIG. 11 is a diagram showing a buffer area on a RAM.

FIG. 12 is a principle explanatory diagram in the case of obtaining an interpolation value X _Q using the difference value D and the current address _AF in the ADPCM method.

FIG. 13 is an operation flowchart (No. 1) of sound source processing by the ADPCM method.

FIG. 14 is an operation flowchart (No. 2) of sound source processing by the ADPCM method.

FIG. 15 is an operational flowchart (No. 3) of the embodiment of the sound source processing by the ADPCM method.

FIG. 16 is an explanatory diagram of envelope characteristics.

FIG. 17 is a diagram showing a first example of a compression table and a decompression table.

FIG. 18 is a diagram showing a second example of a compression table and a decompression table.

FIG. 19 is an operation flowchart of adaptive quantization processing.

FIG. 20 is a diagram showing an example of a PCM musical tone master and an ADPCM musical tone signal.

[Explanation of symbols]

101 Microcomputer 102 keyboard 103 Function key 104 Switch part 105 low-pass filter 106 amplifier 107 speaker 108 power supply circuit 201 Control ROM 202 ROM address decoder 203 Interrupt control unit 204 RAM address control unit 205 ROM address controller 206 RAM 207 Command parser 208 ALU section 209 multiplier 210 input ports 211 Output port 212 Control data and waveform ROM 213 D / A converter section 301, 401 Latch 303 D / A converter

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[Procedure amendment]

[Submission date] July 24, 1992

[Procedure Amendment 1]

[Document name to be corrected] Drawing

[Name of item to be corrected] Figure 2

[Correction method] Change

[Correction content]

[Fig. 2]

   ─────────────────────────────────────────────────── ─── Continued front page    (72) Inventor Kosuke Shiba             3-2-1 Sakaemachi, Hamura-cho, Nishitama-gun, Tokyo               Casio Computer Co., Ltd. Hamura Technology Center             Within (72) Inventor Koichiro Taiji             3-2-1 Sakaemachi, Hamura-cho, Nishitama-gun, Tokyo               Casio Computer Co., Ltd. Hamura Technology Center             Within (72) Inventor Kazuo Ogura             3-2-1 Sakaemachi, Hamura-cho, Nishitama-gun, Tokyo               Casio Computer Co., Ltd. Hamura Technology Center             Within

Claims (2)

[Claims] 1. A musical tone compression method for generating adaptively quantized difference data used in a musical tone waveform generator for generating a musical tone signal by an adaptive differential pulse code modulation method, wherein a digital musical tone signal at a current timing. Difference value and the accumulated value of the reproduced difference values up to the previous timing are calculated to obtain a difference value at the current timing, and a difference between the current timing obtained in the difference calculation step. Adaptive quantization process that performs adaptive quantization while compressing values to obtain adaptive quantized difference data at the current timing, and adaptive quantized difference data at the current timing obtained in the adaptive quantization process Is dequantized to calculate the regenerated difference value at the current timing, and the regenerated difference value at the current timing obtained in the dequantization process The accumulation process of accumulating the accumulated difference value up to the timing to obtain the accumulated value of the reproduced difference value up to the current timing, and the current timing obtained in the accumulation process A judgment step of judging whether or not the accumulated value of the reproduced difference values exceeds the maximum change width of the number of bits assigned to the musical sound signal for generating the musical sound signal; and the present timing according to the judging step. When it is determined that the accumulated value of the reproduced difference values up to the maximum change width does not exceed the maximum change width, the adaptive quantized difference data at the current timing obtained in the adaptive quantization process is output. If it is determined in the output step and the determination step that the accumulated value of the reproduced difference values up to the current timing exceeds the maximum change width, the current timing obtained in the adaptive quantization step. Adaptive quantum of The value of the reproduced difference data is changed based on the value so that the accumulated value of the reproduced difference values up to the present timing obtained in the accumulating process does not exceed the maximum change width. A musical tone characterized by comprising a difference data changing process for repeating the quantizing process, the accumulating process, the judging process and the outputting process, and a repeating process for repeating each of the above processes at each generation timing of a musical tone signal. Waveform compression method. 2. The adaptive quantization in the difference data changing step, if the accumulated value of the reproduced difference values up to the current timing exceeds the maximum change width in the judging step. Changing the absolute value of the adaptively quantized difference data at the present timing obtained in the process to a data value smaller by one data value, and repeating the dequantization process, the accumulation process, and the determination process. The method for compressing a musical tone waveform according to claim 1, characterized in that