JPH08317496A - Digital sound signal processor - Google Patents

Abstract

PURPOSE: To prevent the supply of excessive sound to a user even when impulsive sound is inputted by judging the impulsive sound from the situation of the change of the amplitude in short time of inputted sound signals and suppressing the signal amplitude of an input signal part. CONSTITUTION: The input sound signals are digitized and then inputted through an impulsive sound suppression means 70 to an input buffer 20. The impulsive sound suppression means 70 is constituted of an impulsive sound detection means and an amplitude suppression means. Also, the input buffer 20 is constituted of the memory of relatively large capacity, the address is annularly used and it is turned to a so-called ring buffer. Thus, for the sound signals of input, the sound in the past of a time length corresponding to one ring buffer front the present point of time remains inside the ring buffer. Then, in a speech speed conversion processing 40, a processing is executed by a certain fixed time length unit. The result of speech converted data for one frame is inputted to one of two output buffers 51 and 52 and impulsive sound suppression suited to the characteristics of the user and a using environment is performed.

Description

Detailed Description of the Invention

[0001]

BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to a digital voice signal processing device (hereinafter
The present invention relates to a speech speed conversion device), and particularly to a method of suppressing impact sounds and sounds other than target sounds.

[0002]

2. Description of the Related Art As a means for assisting the hearing of a hearing-impaired person, an analog type hearing aid which processes an amplitude and a frequency characteristic of a voice by using an analog circuit has been mainly used. On the other hand, in recent years, research and development for applying digital signal processing to compensation of hearing impairment have been actively conducted. Regarding this research and development trend, for example, the Journal of the Acoustical Society of Japan (199
1 year 47, No. 10, P760-P765) "Application of digital technology to compensation for hearing impairment" and "Speech-perception a
ids for hearing-impaired people: Current status a
nd needed reserach ", J.of Acoust. Soc. America,
Vol.90, No.2, Pt.1, Aug. 1991.

Generally, in order to compensate for the loss of hearing, the sound pressure level is amplified and the dynamic range is compressed for each frequency according to the hearing characteristics of the user. In a conventional analog hearing aid, such processing is realized by an analog circuit. In addition, in digital hearing aids that have been developed in recent years,
By implementing this processing with software such as a digital filter, the user's hearing characteristics can be adapted in more detail. In such a trend, in recent years, the digital signal processing is used to change only the speed without changing the pitch of the voice and try to perform hearing aid over the entire auditory system including the decline of higher-order language processing speed. Attempts are being made. As for such a speech speed conversion technique, for example, “Assessment of speech speed conversion speech by a hearing-impaired person” in the Acoustical Society of Japan (March 1994, 2-4-7) and Acoustical Society of Japan (1994 50)
Volume 7, No. 7, P509-P520) "Real-time speech speed conversion type listening system" and the like.

In particular, recently, a portable voice speed conversion device has been developed in which this voice speed conversion function is contained in a portable device. For this device, refer to "A portable digital spee
ch-rate converter and its evaluation by hearing-im
paired listeners ", in Proc. of Int. Conf. on Spok
en Language Processing (ICSLP94), pp.2055-2058,
Details are provided in Yokohama, Sep. 1994.

FIG. 9 shows the flow of signal processing of this conventional speech speed conversion apparatus. The thick solid line in the figure indicates the flow of the audio signal,
The dotted line indicates the flow of control signals.

In the above-mentioned conventional speech speed conversion apparatus, the voice inputted through the microphone 11 is converted into the analog amplifier 1
2. After passing through the low-pass filter 13, the A / D converter 14
Is converted into a digital value by. Digitized voice has a fixed time length (hereinafter referred to as frame) 2
The signals are alternately input to the two input frame buffers 21 and 22. While inputting to one of the input frame buffers, the voice data of the other input frame buffer is subjected to a voice speed conversion process 40 for expanding or contracting the time axis of the voice without changing its pitch. Then, the processing result is recorded in the output ring buffer 50 having a relatively large capacity. The output ring buffer 50 is a memory whose addresses are circularly used so that the end address is followed by the start address. Independently of writing the speech speed conversion processing result to the output ring buffer 50, the processed data is taken out from the output ring buffer 50 one by one, and D / A
Converter 64, low-pass filter 63, analog amplifier 6
By simultaneously converting and outputting the sound through the speaker 2 and the speaker or the earphone 61, the speech speed conversion processing and the output of the conversion result proceed at the same time, and the real-time speech speed conversion is realized. Therefore, the output voice is output with a delay time with respect to the input voice, and the delay time spreads with time. That is, the listener can listen to the voice slowly, but slowly behind the reality.

This speech speed conversion operation is controlled by the user through the slow button 2 provided on the apparatus, the speech speed conversion is performed while the slow switch is pressed, and the speech speed is converted while the slow switch is not pressed. The speed conversion is not applied, and the input voice is output as it is. This operation is represented by the switch means 30 controlled by the slow button 2 in FIG.

[0008]

SUMMARY OF THE INVENTION In the above-mentioned conventional speech speed converter, voice is output through an analog amplifier. Therefore, when actually using this device and using it like a hearing aid, processing for suppressing noise other than voice is
It is indispensable except for the process of listening to voice slowly. For example, it is necessary to take measures against a shocking sound generated when the door is closed vigorously or when dishes or the like are dropped. This is because, when an impact sound is input, a very loud sound is output, which not only causes great discomfort to the user, but also may damage the eardrum. In the above-mentioned conventional speech speed conversion device, no consideration has been given to a method for coping with such a problem.

It is a first object of the present invention to provide means for realizing a speech speed conversion device equipped with means for suppressing impact noise.

By the way, in a conventional hearing aid, an automatic gain control circuit and a limiter circuit have been conventionally used in order to cope with such an impact sound. FIG. 10 shows a block diagram of a conventional automatic gain control circuit.

In the conventional automatic gain control circuit having the above structure, the integrating means 312 is charged with a relatively short time constant (several milliseconds) and a long time constant (several tens to several hundreds milliseconds).
Is set to discharge. Thus, when the amplitude of the input signal becomes large, the integrating means is charged in a relatively short time, the gain of the variable gain amplifier 320 is reduced in a short time, and the output signal is prevented from becoming excessive. On the contrary, when the amplitude of the input signal becomes small, the integrating means 312 is discharged for a relatively long time, and the gain of the variable gain amplifier 320 gradually increases.

Further, such audio signal processing can also be realized by digital signal processing. A method for realizing an automatic gain control circuit by digital signal processing is described in, for example, Journal of Acoustical Society of Japan (Vol. 50, No. 4, 1994, P263 to P27).
0) "Compressor / limiter realized by DSP" and the like. According to this method, the operation can be realized with a small amount of calculation in the same manner as the automatic gain control circuit by the analog circuit.

However, when the impact sound is removed by the conventional automatic gain control circuit, the impact sound rises very quickly, so that the gain is not sufficiently lowered when the impact sound reaches the maximum sound pressure point. Therefore, the impact sound is output without being sufficiently suppressed. However, if the charging time constant of the integrating means is reduced in order to deal with the impact noise, the gain of the variable gain amplifier is lowered every time even when a relatively large voice signal other than the impact noise is input, so that the gain is reduced. However, there was a problem that ordinary sounds other than impact sounds became unnatural.

Further, once the impact sound is input, its amplitude is so large that the integrating means is charged to a very high voltage. For this reason, even after the impact noise disappears, the voltage of the integrating means does not drop easily, and it is necessary to continue listening to the voice with a low gain for a while. However, if the discharge time of the integrating means is shortened in order to solve this, the discharge will end every time between voices, and therefore the gain will increase every time the voice rises. For this reason, there is a problem that a so-called pumping phenomenon occurs in which the gain is suppressed at each rising of the voice and the gain increases between the voices.

A second object of the present invention is to suppress impact noise,
Another object of the present invention is to provide a speech speed conversion device having an automatic gain control means capable of setting an appropriate amplification gain according to the sound level at that time.

Further, since the conventional speech speed conversion device is a portable device, it is often used in a place where background noise, for example, low noise generated by an air conditioner is large. In this case, it is desired to distinguish the target voice from other noises and output only the target voice. However, in the above-mentioned conventional speech speed conversion device, no consideration has been given to means for distinguishing and outputting a target voice and other noise.

A third object of the present invention is to provide means for realizing a speech speed conversion device having a function of suppressing noise other than the intended voice.

Further, in the above-described conventional speech speed conversion apparatus, the voice uttered by the user is also subjected to the speech speed conversion process, so that the user's own voice is presented to the user with a slow delay. Therefore, there is a problem in that the user's speech is significantly hindered.

A fourth object of the present invention is to provide a means for outputting the user's own voice without performing the speech speed conversion process.

[0020]

SUMMARY OF THE INVENTION A first object of the present invention is to provide a voice input section of a conventional speech speed conversion apparatus with an impact sound from a state of a change in amplitude value of an input voice signal in a short time. This is achieved by providing an impact sound detection means for making a decision and an amplitude suppressing means for suppressing a signal amplitude value of an input signal portion which is decided as an impact sound.

The second object of the present invention is, in addition to the means for achieving the first object of the present invention, a gain control means for changing the amplification factor of the audio signal based on a relatively long time change of the audio signal. Is provided in the conventional speech speed conversion device, and both means are operated simultaneously.

The third object of the present invention is achieved by providing filter means for suppressing a signal component in a frequency region other than the voice band in the conventional speech speed conversion device.

The fourth object of the present invention is to provide an utterance detecting means for detecting only the utterance of the user and an output of the utterance detecting means, independently of the means for inputting the voice to be subjected to the speech speed conversion processing. And a control means for controlling the speech speed conversion processing operation according to the above.

[0024]

In the means for achieving the first object, the impact sound detecting means detects the impact sound, and the amplitude suppressing means acts to suppress the amplitude of the input audio signal according to the detection result. . Then, the input signal in which the impact sound is suppressed acts so as to be input to the input buffer of the conventional speech speed conversion device. Therefore, the impact sound is suppressed without hindering each operation in the conventional speech speed conversion device.

In the means for achieving the second object, the impact noise is suppressed by the means for achieving the first object. Then, the gain adjusting means acts to set an appropriate gain for the output voice. Therefore, the impact sound is suppressed, and the sound amplified with an appropriate gain according to the sound level at that time is output.

In the means for achieving the third object, the filter means serves to suppress a signal component existing in a frequency region other than the voice band. Therefore, the intelligibility of the target voice is improved.

In the means for achieving the fourth object, the utterance detecting means acts so as to selectively detect the utterance of the user. When the user's utterance is detected,
The control means operates so as to output the input voice as it is without performing the voice speed conversion process. As a result, when the user speaks during the speech speed conversion operation, the voice is forcibly output as a through, so that it is possible to prevent the voice of the user himself from being delayed.

[0028]

EXAMPLES The present invention will be described in detail below with reference to examples.

FIG. 1 shows a first embodiment of the speech speed converting apparatus of the present invention. The input audio signal is digitized and then input to the input buffer 20 via the impact noise suppressing means 70.

The input buffer 20 is composed of a memory having a relatively large capacity, its address is used in a ring shape, and is a so-called ring buffer. Therefore, the input voice signal is
The past voice having a time length corresponding to one round of the ring buffer from the present time always remains in this ring buffer.

In the speech speed conversion processing means 40, processing is executed in units of a certain fixed time length (hereinafter referred to as a frame) as in the conventional method. The result of the speech speed conversion of the data for one frame is input to either one of the two output buffers 51 and 52. Therefore, the size of the output buffer is longer than one frame in consideration of the expansion rate of the voice speed conversion. The other output buffer holds the previous speech speed conversion result. While the previous speech speed conversion result data is sent to the D / A converter 64 at the sampling interval and is output through the low pass filter 63, the analog amplifier 62, and the speaker or earphone 61, the speech speed conversion processing of this frame is performed. The length of one frame and the time required for the speech speed conversion processing are adjusted so that the processing is completed.
As a result, the real-time speech speed conversion similar to that of the conventional speech speed conversion device can be realized.

Further, in this embodiment, as in the conventional speech speed conversion apparatus, the means 3 for switching the speech speed conversion operation between ON and OFF only when the user presses the slow button 2 provided on the apparatus.
With the control of 0, the speech speed conversion process is performed. When the slow button is not pressed, the output data of the shock noise suppressing means is recorded in the input buffer and is immediately output at the same time.

FIG. 2 shows the details of the impact noise suppressing means 70 in the first embodiment. This means is an impact sound detecting means 7
1 and the amplitude suppressing means 72. The impact sound detecting means 71 detects the impact sound from the change in the amplitude of the input signal in a short time. The difference calculation means 711 calculates the difference value between the previous sample value and the current sample amplitude. Judgment means 7
When the absolute value becomes larger than a predetermined threshold value once or several times in succession, it is determined that the impact sound having the abrupt amplitude change is input. Then, the output means 713 outputs 1 when the determination means 712 determines that the sound is an impact sound, and turns the output to 0 after the elapse of a certain time managed by the timer means 714. The amplitude suppressing means 72 suppresses the amplitude by performing a calculation for multiplying the input signal amplitude by the attenuation coefficient by the multiplying means 721 only while the impact sound detecting means outputs 1. As a result, the impact sound is detected, and for a certain time after the impact sound is detected,
The input amplitude will be suppressed at a constant rate.

In the above description using FIG. 2, the duration managed by the timer means 714, the judgment threshold value,
Although the damping coefficient is a constant, if the user can change these values according to the environment in which the device is used, it is possible to realize impact noise suppression that is more suitable for the environment.

FIG. 3 shows a second embodiment of the present invention.
In this embodiment, in addition to the configuration of the first embodiment, the output section is provided with a gain control means 80. The gain control means changes the amplification factor according to a change in the output signal for a relatively long time. The present embodiment is characterized in that it includes both impact noise suppressing means and gain controlling means. The impact sound suppressing unit 70 suppresses the impact sound based on a relatively short time change, as in the first embodiment of the present invention, and the gain control unit 80 changes the signal level for a relatively long time. Vary the gain for the output based on. This makes it possible to simultaneously realize both the function of suppressing the impact noise and the function of slowly changing the gain according to the signal level. In this embodiment, the gain control means 80 is installed on the output side of the structure of the speech speed converter, but the same effect can be obtained by installing it between the shock noise suppression means on the input side and the input buffer. It is possible.

FIG. 4 shows details of the gain control means 80 in the second embodiment of the present invention. The gain control means 80 of this embodiment comprises a level detection means 81, a gain setting means 82, multiplication means 83 and 84, and a coefficient generation means 85.

When the signal X (t) at time t is input, the level detection means 81 calculates the absolute value | X (t) |, and then the index calculation means 811 calculates the absolute value | X (t) | Then, the signal level index value P (t) is calculated. This calculation is expressed by the following equation.

When | X (t) | ≥ P (t-1): P (t) = | X (t) | + (P (t-1)-| X (t) |) * exp (- 1 / fs * Ta) (Equation 1) | X (t) | <P (t): P (t) = P (t-1) * exp (-1 / fs * Tr) (Equation 2) , Fs is sampling frequency, Ta and Tr are time constants.

That is, in the index calculation means 811, the signal level index value P (t-1) one hour before is compared with | X (t) |. If | X (t) | is greater than P (t-1), P is the time constant T
When a is increased to | X (t) | by a and conversely | X (t) | is smaller than P (t-1), P is decreased to 0 with a time constant Tr. , The current signal level index value P (t) is obtained. In this embodiment, the coefficient value e generated by the coefficient generating means is
The volume 3 can be changed from outside the device. Gain setting means 8 in gain control means 80 of FIG.
2 is the index value P (t) obtained by the multiplication means 84 and the coefficient
Receives the product of e as input, and gain value G corresponding to each eP (t)
(t) is output by referring to the table 821. Then, the multiplication means 83 multiplies the gain value G (t) by each data X (t) and outputs the result.

FIG. 5 shows an example of a table 821 referenced by the gain setting means 82 in the form of a graph. By setting the data as shown in FIG. 5, an operation of increasing the gain when the signal level P (t) is small and conversely decreasing the gain when P (t) is large is realized.

Furthermore, in the present embodiment, as shown in FIG. 3, the coefficient value e can be changed by adjusting the volume 3 from outside the device, so that the gain control means can be adjusted according to the usage situation of the device and the preference of the user. It is possible to change the operation of 80. For example, as shown in FIG. 5, when the coefficient value e is reduced, only the vicinity of the origin of the table 821 is referred to,
When the signal level becomes large, the gain reduction does not occur much, and conversely, when e is increased, the entire table is referred to, and the gain reduction starts before the signal level becomes too large.

According to the second embodiment of the present invention described with reference to FIGS. 3 to 5, since the impact noise suppressing means 70 performs the impact noise suppressing operation, the time constant of the gain control means 80 is increased. Ta and Tr can be set to appropriate values for sounds other than impact noise. This makes it possible to avoid problems caused by improper setting of the time constant, such as a pumping phenomenon, and at the same time suppress impact noise. Although not shown in FIG. 3, the time constants Ta and Tr used when calculating the P (t) can be changed from outside the device, depending on the use environment and user preference. Gain control can be realized.

FIG. 6 shows a third embodiment of the present invention.
In this embodiment, filter means 100 is provided in addition to the configuration of the second embodiment. Filter means 100
For example, a filter having a characteristic of suppressing noise other than the intended human voice, such as a band limiting filter, is used as the filter.

In this embodiment, the filter operation can be controlled from outside the device. The user can operate the filter according to the usage environment of the apparatus to improve the clarity in a situation where there is much noise other than voice.

FIG. 7 shows a fourth embodiment of the present invention.
In this embodiment, in addition to the configuration of the third embodiment, the utterance detecting means 200 for detecting the utterance of the user is provided. This is the second microphone 210, the integrating means 22.
0, comparing means 230. The second microphone 210 is used to detect the utterance of the user of the apparatus, and is installed closer to the user's mouth than the first microphone 11. The integrating means 220 integrates the output of the second microphone 210 and outputs an output corresponding to the utterance situation of the user. That is, the output of the integrator 220 maintains a high value when the user speaks loudly and maintains a low value while the user is silent. The comparing means 230 compares the integrator 220 with the threshold value, and outputs 1 when the integrator output is higher than the threshold value and outputs 0 when the integrator output is lower than the threshold value. The output of the comparison means 230 is connected to the switch means 30, and when the output of the comparison means 230 becomes 1, the switch means 30 operates so as to be forcibly connected to the through side.

With this operation, when the user utters during the speech speed conversion operation, the input from the first microphone is immediately output as a through signal, which causes a problem in the conventional speech speed conversion device. In addition, it is possible to prevent a phenomenon in which the voice of the user is fed back to the user with a time delay and the speech of the user is significantly disturbed.

The threshold value required for the comparison means 230 to perform the comparison operation can be set in advance, but in the present embodiment, the threshold value setting means 240 is provided to adjust the volume from outside the apparatus. The above threshold value can be changed by turning 4. This makes it possible to set the threshold value according to the loudness of the user's voice and the surrounding noise, so that malfunction can be reduced.

FIG. 8 shows the external appearance of the speech speed conversion apparatus according to the fourth embodiment. By installing the second microphone 210 in the vicinity of the earphone 61 worn by the user, it is devised so that only the voice of the user is mainly detected. The first microphone 11 is built in the main body of the apparatus, and the user can freely orient the voice in a desired direction.

[0049]

According to the present invention, it is possible to prevent an excessive sound from being given to the user even when an impact sound is input in the speech speed conversion device. Furthermore, by adjusting the threshold value used when determining the impact sound, the user can achieve the effect of suppressing the impact sound that is suitable for the usage environment and the characteristics of the user.

Further, according to the present invention, in the speech speed conversion device, it is possible to suppress the impact sound and at the same time control the output gain according to the level fluctuation of the input voice observed for a time longer than the impact sound. There is an effect that. Further, by adjusting the time constant used for gain control by the user, there is an effect that the output gain control suitable for the use environment and the characteristics of the user can be realized.

Further, according to the present invention, in the speech speed conversion device, there is an effect that noise other than the target voice can be suppressed and the clarity of the target voice can be improved. Furthermore, there is an effect that the user can operate the noise suppression function according to the usage environment and the characteristics of the user.

Further, according to the present invention, in the speech speed conversion apparatus, even if the user speaks during the speech speed conversion operation, the user's own voice is delayed and given to the user. This has the effect of preventing interference with speech. Further, by adjusting the threshold value used in the means for detecting the utterance of the user according to the usage environment, there is an effect that the malfunction at the time of detecting the utterance of the user can be reduced.

[0053]

[Brief description of drawings]

FIG. 1 is a configuration diagram of a first embodiment of the present invention.

FIG. 2 is a configuration diagram of impact noise suppressing means of the present invention.

FIG. 3 is a configuration diagram of a second embodiment of the present invention.

FIG. 4 is a block diagram of a gain control means of the present invention.

FIG. 5 shows an example of a table used in the gain control means of the present invention.

FIG. 6 is a configuration diagram of a third embodiment of the present invention.

FIG. 7 is a configuration diagram of a fourth embodiment of the present invention.

FIG. 8 is an external view of a user wearing the fourth embodiment of the present invention.

FIG. 9 is a configuration diagram of a conventional speech speed conversion device.

FIG. 10 is a block diagram of a conventional shock noise suppression circuit.

[Explanation of symbols]

DESCRIPTION OF SYMBOLS 1 ... Speak speed converter, 2 ... Slow button, 3, 4 ... Volume, 10 ... Voice input part, 11 ... 1st microphone,
12 ... Analog amplifier, 13 ... Low pass filter, 14
... A / D converter, 20 ... Input buffer, 21, 22 ... Frame buffer, 30 ... Means for switching on / off of speech speed conversion operation, 40 ... Speech speed conversion processing means, 50 ... Output buffer,
51, 52 ... Frame buffer, 60 ... Audio output unit, 6
1 ... Speaker or earphone, 62 ... Analog amplifier, 63 ... Low pass filter, 64 ... D / A converter, 70
... impact sound suppressing means, 71 ... impact sound detecting means, 711 ... difference calculating means, 712 ... judging means, 713 ... output means, 7
14 ... Timer means, 715 ... Threshold value, 72 ... Amplitude suppressing means, 721 ... Multiplying means, 722 ... Attenuation coefficient, 80
... gain control means, 81 ... level detection means, 82 ... gain setting means, 821 ... gain table, 83, 84 ... multiplication means, 85 ... coefficient generation means, 100 ... filter means, 20
0 ... User voice detection unit, 210 ... Second microphone,
220 ... Integrator, 230 ... Comparison means, 240 ... Threshold setting means, 300 ... Conventional automatic gain control circuit, 310 ...
Short-time average amplitude extraction means, 311 ... Absolute value calculation means, 3
12 ... Integrating means, 320 ... Gain control amplifier.

Claims (15)

[Claims] 1. An input analog audio signal is digitized, the digital audio signal is recorded in an input buffer, the digital audio signal recorded in the input buffer is subjected to signal processing in frame units, and the signal-processed digital signal is processed. Record a plurality of audio signals in any one of the prepared output buffers, and at the same time, convert the past audio signals that have already been subjected to the above signal processing and recorded in another output buffer into analog audio signals. In the digital voice signal processing apparatus for outputting the speech signal, the means for performing the above signal processing is a speech rate conversion processing means for expanding and contracting the time axis without changing the pitch of the voice, and it is set as the change amount of the input signal amplitude in a short time. When an impact sound is detected by comparing it with a threshold value, the impact sound is suppressed by applying an attenuation coefficient to the input signal amplitude for a certain period of time. At least one of a shock noise suppression means for controlling the sound noise and a gain control means for adjusting the gain in accordance with the fluctuation of the audio signal level in the observation time longer than the observation time at the time of detecting the shock noise. A digital audio signal processing device characterized by: 2. An input analog audio signal is digitized, the digital audio signal is recorded in an input buffer, the digital audio signal recorded in the input buffer is processed on a frame-by-frame basis, and the signal-processed digital signal is processed. Record a plurality of audio signals in any one of the prepared output buffers, and at the same time, convert the past audio signals that have already been subjected to the above signal processing and recorded in another output buffer into analog audio signals. In the digital audio signal processing device that outputs the output, the impact sound is detected by comparing the change amount of the input signal amplitude in a short time with a threshold value, and when the impact sound is detected, the attenuation coefficient is applied to the input signal amplitude for a certain period of time. And a sound signal level for an observation time longer than the observation time for detecting the impact sound. And a gain control means for adjusting the gain in accordance with the fluctuation of the digital audio signal processing apparatus. 3. The impact sound suppressing means comprises impact sound detecting means and amplitude suppressing means, and the percussion sound detecting means calculates a difference value for each sample of an input digital audio signal and the difference value. And a difference threshold, the output value is changed from 0 to 1 when the difference value exceeds the difference threshold value once or continuously several times as a result of the comparison. Furthermore, the amplitude control means is characterized in that the amplitude control means multiplies an input sound signal by an attenuation coefficient and outputs the same while the output of the impact sound detection means is one. The digital audio signal processing device according to claim 1. 4. The difference threshold and the damping coefficient are:
4. The digital audio signal processing device according to claim 3, wherein the user can adjust the signal from outside the device. 5. The gain control means, the level detection means for outputting a level index value that changes at a rising time and a falling time with different time constants according to the input signal level, means for generating a coefficient, and the level. A multiplication means for multiplying the output of the detection means by the coefficient, a gain setting means for outputting a gain value by referring to a table based on the output of the multiplication means, and a multiplication of the gain value output by the gain setting means and an input signal. 3. The digital audio signal processing device according to claim 1, further comprising a second multiplication means for performing. 6. The speech speed conversion device and the digital voice signal processing device according to claim 5, wherein the rising and falling time constants can be set by the user from outside the device. 7. The level detecting means includes means for calculating an absolute value of an input signal, and index calculating means for calculating a current level index value based on the absolute value and a level index value one time before. 6. The digital audio signal processing device according to claim 5, comprising delay means for delaying the current level index value by one time. 8. The index calculating means is such that the input signal at time t is X (t), the level index value is P (t), fs is the sampling frequency, Ta is the rising time constant, and Tr is the falling time. Given that | X (t) | ≥ P (t-1) with time constant: P (t) = | X (t) | + (P (t-1)-| X (t) |) * If exp (-1 / fs * Ta) | X (t) | <P (t): P (t) = P (t-1) * exp (-1 / fs * Tr) 8. The digital audio signal processing device according to claim 7, wherein P (t) is calculated by calculation. 9. In the table referred to by the gain setting means, a maximum gain value is associated with an input value in the range of 0 to a certain range, and an input value exceeding the above range is
6. The digital audio signal processing device according to claim 5, wherein the corresponding gain value decreases in proportion to the increase of the input value in proportion to the exponential function. 10. An input analog audio signal is digitized, the digital audio signal is recorded in an input buffer, the digital audio signal recorded in the input buffer is subjected to signal processing in frame units, and the signal processing is performed. The digital audio signal is recorded in any one of the prepared output buffers, and at the same time, the past audio signal which has already been subjected to the signal processing and recorded in another output buffer is converted into an analog audio signal. In the digital audio signal processing device for converting and outputting, the signal processing is a speech speed conversion processing for expanding and contracting the time axis of the audio without changing the pitch of the audio, and for processing other than a certain frequency range including the audio signal band. A digital audio signal processing device having a filter means for suppressing frequency components. 11. A digital audio signal processing apparatus according to claim 10, wherein the operation of the filter means and the non-operation of the filter means can be set by a user from outside the apparatus. 12. An input analog voice signal is digitized, the digital voice signal is recorded in an input buffer, the digital voice signal recorded in the input buffer is subjected to signal processing on a frame-by-frame basis, and the processed digital voice is processed. One of output buffers with multiple signals
, And at the same time, the past audio signal that has already been subjected to the above signal processing and recorded in another output buffer,
In the digital audio signal processing device for converting and outputting to an analog audio signal, the signal processing is a speech speed conversion processing for expanding and contracting the time axis without changing the pitch of the audio,
In addition, the digital signal processing device has means for detecting the utterance of the user, and when the utterance of the user is detected, the speech speed conversion process is stopped and the input signal is directly output. Audio signal processing device. 13. The means for detecting the utterance of the user comprises: a utterance detecting microphone; an integrator for integrating a voice signal input by the microphone; and means for setting a threshold value used for utterance detection. 13. The digital audio signal processing device according to claim 12, further comprising: a comparison unit that compares the threshold value with the output of the integrator. 14. The digital audio signal processing device according to claim 13, wherein the speech detection microphone is installed near an earphone worn by a user. 15. The threshold value setting means can be controlled by a user from the outside of the apparatus.
The digital audio signal processing device according to 1.