JPH0449957B2 - - Google Patents

Description

[Detailed description of the invention]

(Technical Field) The present invention relates to a speech synthesizer that reads information in the waveform region of phoneme waveforms from a storage area and synthesizes speech, and in particular, performs addition to superimpose a plurality of ADPCM-encoded phoneme waveforms and outputs speech. Concerning the speech synthesizer obtained. (Prior Art) When a human voice is a voiced sound, it is emitted when airflow from the lungs becomes a quasi-periodic impulse flow through the vocal cords and resonates with a hollow resonator called the vocal tract. Figure 1 shows the waveform of a human voice. As shown in Figure 1, the human voice is “pitch”.
Almost the same waveform is repeated every cycle called . This is because, as stated earlier, the human voice is emitted by a quasi-periodic impulse stream, and the pitch period is equal to the interval of this impulse stream. When attempting to synthesize such speech, various methods can be used depending on the format in which the speech information is stored. One method is to store one-pitch period waveforms of speech (called phoneme waveforms) for various speech sounds in a storage area, and output speech by connecting these phoneme waveforms according to control information. be. Figure 2 shows a configuration using this method. 1 is a storage area for storing phoneme waveforms of 1 pitch period for each type of speech, 2 is a storage area for control information such as pitch and amplitude multiplication for connecting phoneme waveforms, and 3 is a storage area for connecting voice segments. This is the synthesis department. Speech is synthesized in the synthesizer 3 by connecting the phoneme waveforms stored in the storage area 1 using the control information stored in the storage area 2. However, in order to create a highly natural synthesized sound, it is necessary to The pitch and volume of the voice must be appropriately controlled. The loudness of the voice is controlled by multiplying the phonemes in the storage area by a certain amount using amplitude multiplication information in the storage area. Furthermore, the pitch of the voice is controlled by pitch information in the storage area 2. However, the phoneme waveform in storage area 1 has the pitch period length of the voice when the phoneme waveform was extracted, and does not necessarily match the pitch given by the control information. Therefore, if the pitch of the control information is shorter than the speech segment length, the rear end of the speech segment is cut off, and if the pitch of the control information is longer than the speech segment length, the last value of the speech segment is extended for control. It is assumed that the phoneme has the same pitch length as the information. (See Figure 3) However, if the phoneme waveform stored in storage area 1 is cut in the middle, if the phoneme waveform is not sufficiently attenuated, discontinuity will occur between the connected phoneme waveforms and the synthesized sound will be affected. This has the disadvantage of causing a shadow, and conversely, when the phoneme waveform is lengthened, the spectrum of the phoneme waveform is deformed, leading to deterioration of sound quality. (Objective and Summary of the Invention) The object of the present invention is to solve these drawbacks, and to solve the problem of multiple channels that are shifted by pitch period.
In a speech synthesizer that superimposes ADPCM encoded phoneme waveforms, by time-multiplexing ADPCM code reproduction, and by using an adder used for phoneme waveform superimposition calculation as cumulative addition/subtraction means in the ADPCM code regenerator. This is a speech synthesizer that synthesizes natural, high-quality speech with a simpler circuit configuration, and will be described in detail below. (Premise of the invention) As mentioned before, speech is produced by the resonance of impulse flows from the vocal cords, and this can be replaced by an electric circuit. That is, speech is considered to be a superposition of output waveforms (hereinafter referred to as phoneme waveforms) of a resonant filter corresponding to one impulse emitted by an excitation circuit corresponding to the vocal cords. This will be explained using FIG. 4. Reference numeral 10 in FIG. 4 is an impulse train generated by the excitation circuit. Here, the time interval between each impulse is the pitch period interval. 11 is a synthesized audio output waveform obtained by driving a resonant filter with an impulse train generated by an excitation circuit. 12 is a phoneme waveform when the resonant filter is driven by impulse _P1 of the impulse train generated by the excitation circuit. Similarly, 13 to 17 are phoneme waveforms when the resonance filter is driven by impulses _P2 to _P6 . Drive impulse train 10 emitted from the excitation circuit
Since is the addition of impulses P ₁ to P ₆ , according to the superposition theorem, the synthesized speech output waveform 11 can be obtained by adding the phoneme waveforms 12 to 17. The phoneme waveform 12 shown in FIG. 4 is zero before time point _t1 . The phoneme waveform output after time point t ₁ has the property of attenuating over time and becoming 0 at an infinite time point. In fact, in the case of a speech waveform, the phoneme waveform is considered to be almost 0 when 16 milliseconds have passed from the excitation point (t ₁ ). Therefore, 16 milliseconds from the excitation time point (t ₁ ) is sufficient for the reproduction processing of the phoneme waveform 12. Therefore, in the present invention, a phoneme waveform whose initial value is 0 and whose phoneme length is set to be sufficiently longer than the pitch period (for example, 16 milliseconds) so that the phoneme waveform is sufficiently attenuated is used as the phoneme waveform. However, the reproduction process of phoneme waveform 13 is performed at the excitation time point.
Starting from t ₂ requires multiple reproduction processing because the reproduction processing of the phoneme waveform 12 has not yet been completed. The required waveform reproduction multiplicity n is the following (1)
It is given by the formula, n = 16 milliseconds/minimum pitch period (1). In the case of normal voice, n of about 4 is sufficient. Therefore, the following description will be made assuming that n=4. Next, as a code for reproducing the phoneme waveform, an ADPCM code with high coding efficiency is used. series of
A regenerator for reproducing phoneme waveforms from ADPCM codes has been proposed in Japanese Patent Application No. 109800/1982, so a detailed explanation will be omitted. In differential decoding processing such as ADPCM code reproduction processing, if the reproduction processing is terminated after 16 milliseconds (128 sample periods), the final output value is retained. Therefore, the phoneme waveform condition (Condition 1) is 0 before the excitation time point (Condition 2) and is 0 at the infinite time point.In order to satisfy the condition, the reproduction initial value must be 0 and the reproduction final output value. must be 0, and even if the initial reproduction value is 0, the final reproduction value will not normally be 0. Therefore, a value (referred to as an offset PCM code) that makes the final reproduction value 0 must be added to the end to make the final reproduction value 0. (Embodiment of the invention) FIG. 5 shows an embodiment of the invention. Here, the word "channel" or "ch" is used to number the processes that are time-divisionally processed within one sample period. Each part in FIG. 5 is as follows. 10
0 is the 1st channel input register, 101 is the 2nd channel input register, 102 is the 3rd channel input register, 103 is the 4th channel input register, 104 and 105 are the selectors, 106 is the register that stores the ADPCM code data Ln, and 107 is the register 106. A pointer movement amount memory that outputs a pointer movement amount Dn using a value as an address; 108 is an adder; 109 is a selector; 110 is a selector; 111 is a first channel pointer register; 112 is a second channel pointer register;
13 is the 3rd channel pointer register, 114 is the 4th channel
A pointer register, 115 is a selector, 116 is a pointer limiter that limits the pointer value Pn to a specific range and outputs it as a pointer limiter value Pn', 117 is a quantization memory that stores the quantization step value X, 118 is a shift register, 119
is an adder/subtractor, 120 is a register, 121 is a register that outputs synthesized speech, 122 is an output terminal, 12
3 is the selector, 135 is the 1st channel down counter,
136 is the 2nd channel down counter, 137 is the 3rd channel
Down counter, 138 is the 4th channel down counter, 139 is the 1st channel decoder, 140 is the 2nd channel decoder, 141 is the 3rd channel decoder, 142 is the 4th channel down counter
4ch decoder, 143 is the BUSY signal of each channel, 144 is the END signal that gives the final point of playback,
145 is a start signal for waveform reproduction, and 150 is a controller. The controller 150 is connected to each circuit such as the first channel input register 100, the second channel input register 101, . . . , the third channel decoder 141, the fourth channel decoder 142, etc., and controls their operations, as shown in FIG. The connections are omitted to avoid complicating the diagram. FIG. 6 shows data used in the present invention to reproduce phoneme waveforms, which consists of a pointer initial value DPn, 128 ADPCM codes Ln ₁ , Ln ₂ , Ln ₃ , . . . , Ln ₁₂₈ , and an offset PCM code. The initial value data DPn of the pointer is stored in the pointer registers 111 to 114 corresponding to each channel at the start of waveform playback.
This is the data stored in . This pointer register inputs the pointer movement amount Dn corresponding to the ADPCM code data and the previous pointer value to the adder 10.
8 is updated and stored as a new pointer value. Also for offset
The PCM code is data that is read after the reproduction processing using 128 ADPCM codes, and is data that sets the final value of the waveform reproduction to 0. Further, FIG. 7 shows the relationship between the playback processing assigned to each channel and the time. In FIG. 7, 200 is a composite output waveform, 201
is the phoneme waveform reproduced by the reproduction process assigned to channel 1, 202 is the activation signal ST1 for channel 1, and 203 is the BUSY signal for channel 1.
BUSY1, 204 is the END signal END1 for channel 1
Below 205, 209, 213 are channels 2, 3,
The phoneme waveforms 206, 210, and 214 reproduced by the reproduction process assigned to channel 4 are activation signals ST2, ST3, ST4, and 20 of channels 2, 3, and 4.
7,211,215 are for channels 2, 3, and 4
BUSY signal BUSY2, BUSY3, BUSY4, 20
8,212,216 are for channels 2, 3, 4
END signals are END2, END3, and END4 signals. Embodiments of the present invention will be described in detail with reference to FIGS. 5 to 7. At time point _t1 , a starting signal ST1 is applied to the controller 150 from the outside, and the reproduction process of the phoneme waveform corresponding to FIG. 412 is started by channel 1. At this time, controller 150 sets down counter 135 to the value 128. This value corresponds to the aforementioned 16 milliseconds. (If the sampling period of the output waveform is 125 microseconds, 16 milliseconds is 128 sampling periods.) When the output of the down counter 135 is other than -1, the decoder 139 outputs "1" as the BUSY1 signal. Also, the output of the down counter 135 is 0.
When this occurs, the END1 signal is set to "1" (see FIG. 7). Thereafter, data for waveform reproduction is read from the input register 100 at every sampling period (125 microseconds) and waveform reproduction processing using the ADPCM code is performed. Down data 135 every time one playback process is completed.
is reduced by 1. At the end of the 128th playback process, the down counter 135 becomes 0.
The END1 signal becomes “1”. In the next sampling period, the END1 signal is "1", and in this case, the offset PCM code from the input register 100 is applied to the adder/subtractor 119 through the selectors 104, 105, and 123. Next, down counter 13
5 is subtracted by 1 and becomes -1. At this time, decoder 1
The END1 signal, which is the output of 39, becomes “0”,
The BUSY1 signal becomes “0”. Table 1 shows that the controller 150 selector 1
The ADPCM code Ln stored in the input registers 100, 101, 102, 103 of each channel input through 04 and the values shifted down by the shift register 118 ([X], [1/2X], [1/4 shows.

[Table] Similarly, at time point t ₂ , start signal ST2 is applied externally.
is applied to the controller 150, and channel 2 starts the reproduction process of the phoneme waveform corresponding to FIG. 4, 13. At this time, the value 128 is set in the down counter 136, and the BUSY2 signal becomes "1". After that, the same playback process as for channel 1 is performed, and the down counter 136 becomes 0 after 128 sampling cycles.
The BUSY2 signal becomes "0". Thereafter, similar control is performed on channel ₃ from time point t3, channel ₄ from time point t4, and channel ₁ from time point t5. In this way, reproduction processing is performed using four channels and a synthesized speech output waveform is output at each sampling period, but the reproduction processing of each channel is performed in a time-division manner between one sampling time point. Here, the time relationship of processing within one sampling period is shown in FIG. 8a, and its flowchart is shown in FIGS. 8b and 8c. Processing within one sampling period is divided into five cycles, which are sequentially processed. In addition, most of the circuits required for ADPCM reproduction processing processed in each cycle are shared, and among the components shown in Figure 6, all components that do not have a channel number added to their names are common to each cycle. These are collectively referred to as the "shared part." (Cycle 1) Waveform reproduction processing assigned to channel 1 is performed using the input register 100, pointer register 111, down counter 135, decoder 139, and a common section, and the result is stored in the register 120. (Cycle 2) Waveform reproduction processing assigned to channel 2 is performed using the input register 101, pointer register 112, down counter 136, decoder 140, and a shared section, and the result is stored in the register 120. (Cycle 3) Waveform reproduction processing assigned to channel 3 is performed using the input register 102, pointer register 113, down counter 137, decoder 141, and a common section, and the result is stored in the register 120. (Cycle 4) The waveform reproduction process assigned to channel 4 is performed using the input register 103, pointer register 114, down counter 138, decoder 142 and the common section, and the result is stored in the register 120. (Cycle 5) The value of register 120 is stored in register 121 and used as PCM data for outputting synthesized speech. The control in the above five cycles is shown in the flowcharts of FIGS. 8b and 8c. In this way, in this embodiment, the phoneme waveform reproduction processing using the ADPCM code assigned to each channel is performed by time-division processing using the constituent elements for each channel and the commonly used shared part. . The successive addition means (register 120 and adder/subtractor 119) used for waveform reproduction of each channel also serves as addition means for superimposing phoneme waveforms, and has the advantage of being realized by a small circuit configuration. Furthermore, by changing the interval between activation signals of each channel, the pitch period of the synthesized audio output can be easily changed. Furthermore, since each phoneme waveform starts from 0 and ends at 0, it also has the advantage of not causing any deterioration in sound quality such as waveform discontinuity due to addition of each phoneme. (Effects of the Invention) As described above, according to the present invention, it is possible to construct a speech synthesis circuit that can control the pitch period and outputs high-quality synthesized speech with a simple circuit configuration.

[Brief explanation of drawings]

Figure 1 is a diagram showing speech waveforms, Figure 2 is a block diagram of a conventional speech synthesizer, Figure 3 is a diagram showing waveforms used when changing pitch, and Figure 4 is a diagram showing the superposition of phoneme waveforms. Explanatory diagram, Fig. 5 is a diagram showing one embodiment of the present invention, Fig. 6 is a diagram showing the data format of the phoneme waveform, and Fig. 7 is a diagram showing the phoneme waveform, ST signal, BUSY
Figure 8a shows the time relationship between the signals and the END signal. Figure 8a shows the time relationship of processing within one sample cycle time. Figures 8b and c show the processing within one sample cycle time. This is a flowchart. 100...1st channel input register, 101...1st channel input register
2ch input register, 102...3rd channel input register, 103...4th channel input register, 104...
Selector, 105... Selector, 106... Register, 107... Pointer movement amount memory, 108
... Adder, 109 ... Selector, 110 ... Selector, 111 ... 1st channel pointer register, 1
12... 2nd channel pointer register, 113... 2nd channel pointer register
3ch pointer register, 114... 4th channel pointer register, 115... selector, 116... pointer limiter, 117... quantization memory, 11
8...Shift register, 119...Additional/subtractor, 1
20...Register, 121...Register, 122
...Output terminal, 123...Selector, 135...
1st channel down counter, 136... 2nd channel down counter, 137... 3rd channel down counter, 1
38...4th channel down counter, 139...th
1ch decoder, 140...2nd channel decoder, 14
1... 3rd channel decoder, 142... 4th channel decoder, 143... BUSY signal, 144... END signal, 145... Start signal, 150... Controller.

Claims (1)

[Claims] Corresponding to a plurality of channels, a pointer initial value, a plurality of ADPCM code data, and an offset PCM
The unit data for phoneme waveform reproduction, including encoded data, is input sequentially to the reproduction means, the phoneme waveform is reproduced for each channel, and the phoneme waveforms of each channel shifted by the pitch period are added by the addition means and decoded. In the speech synthesizer that obtains the output, the phoneme waveform input to each channel is a phoneme waveform whose initial value is 0 and whose phoneme length is set sufficiently longer than the pitch period, and the offset PCM code data is reproduced. This is code data set to set the final value of the phoneme waveform to 0, and the reproduction means includes a plurality of input registers 100 to 103 that store ADPCM code data of the phoneme waveform corresponding to each channel, and A pointer movement amount memory 107 that outputs a pointer movement amount according to ADPCM code data, and a register that stores a pointer value, which stores the pointer initial value as an initial value and stores the pointer movement amount corresponding to the ADPCM code data. and the previous pointer value to the adder 108
A plurality of pointer registers 111 to 114 corresponding to each channel update and store the result of addition as a new pointer value, and a plurality of predetermined quantization steps are stored and a quantization step value corresponding to the pointer value is calculated. Output quantization memory (11
7), a shift register 118 capable of storing and shifting the quantization step value, and means 119, 1 for cumulatively adding and subtracting the output of the shift register 118.
20, is a means for time-divisionally performing phoneme waveform reproduction processing in each channel within one sampling period, and the cumulative addition/subtraction means 119, 120 add the phoneme waveforms of each channel shifted by a pitch period. A speech synthesizer, characterized in that the speech synthesizer also serves as the addition means.