JPH08292790A - Video tape recorder - Google Patents

Abstract

PURPOSE: To perform a speech speed changing so that the deviation between a video and a voice is made smaller and voice information are obtained as much as possible at the time of a double speed mode and to reproduce one part of the voice at a normal speed by a thinning processing at the time of a reproducing equal to or faster than a triple speed mode. CONSTITUTION: At the time of a double speed reproducing, this tape recorder has a double speed mode performing the speech speed changing performing a compression or elimination processing with respect to an input voice signal according to whether a reproducing signal is voice sections or silent sections. Then, at the time of a ±N (N is a natural number of >=3) multiple speed reproducing, this tape recorder is made so as to be changed over even to an N multiple speed reproducing mode performing the thinning processings of voice sections of prescribed periods of the reproducing voice signal according to the reproducing speed by the command of a system microcomputer 114.

Description

Detailed Description of the Invention

[0001]

BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to a video tape recorder (V) equipped with a voice speed conversion device for changing the voice speed of a voice signal.
TR)

[0002]

2. Description of the Related Art VTRs have been commercialized so that they can listen to sound at a normal speed even at double speed. The basic structure of the VTR is described in, for example, magazine "Electronics" 1993.
April issue, pp. 34-37.

[0003]

However, in the VTR as described above, although the voice can be heard at the normal speed during the double speed reproduction, the voice information is unconditionally pulled for half a minute and deleted. However, there are problems that it is not possible to grasp the contents of the tape recorded from the reproduced audio, and there is a large time difference between the video and audio.

Therefore, according to the present invention, the difference between the video and the audio can be reduced during the double speed reproduction, and the voice speed conversion is performed so as to limit the audio speed so that the audio information can be obtained as much as possible. For example, it is intended to provide a video tape recorder that reproduces audio by the same method as the conventional double speed reproduction when reproducing the audio speed at the speed (5 times or 9 times) at the time of fast-forwarding and rewinding reproduction. Is.

[0005]

According to the present invention, during double speed reproduction, compression / expansion processing or deletion processing is performed on an input audio signal depending on whether the reproduced audio signal is a voice section or a silent section. In the double speed audio reproduction mode in which the voice speed conversion is performed, and in the case of ± N times speed reproduction (N: a natural number of 3 or more), the processing for thinning out the audio section of the reproduction audio signal for a predetermined period according to the reproduction speed is performed. The video tape recorder is equipped with control means for setting the N-speed reproduction mode to be performed.

Further, in the above video tape recorder, in order to perform the double speed audio reproduction mode, a speech speed conversion processing means for converting the speech speed of the input audio signal, a ring memory to which the output of the speech speed conversion processing means is written, and The speech speed conversion processing means includes means for reading data from the ring memory at a constant speed, and the speech speed conversion processing means for the input voice signal according to whether the input voice signal is a voice section or a silent section and the amount of storage in the ring memory. It is characterized by having a speech speed conversion device provided with means for performing compression / expansion processing or deletion processing.

Further, in the above video tape recorder, an A / D conversion means for sampling the input analog audio signal at a sampling frequency according to a set reproduction speed multiplication factor in order to perform a double speed audio reproduction mode,
A frame memory to which the voice signals output from the D conversion means are input, and a voice speed conversion processing means that performs a voice speed conversion process on the voice signals every time a required number of voice signals is input to the frame memory, A ring memory to which the output of the speech speed conversion processing means is written, a reading means for reading data from the ring memory at a constant speed, and a storage amount calculation for calculating the storage amount of the ring memory based on the write signal and the read signal of the ring memory. The speech speed conversion processing means includes a section determining means for determining whether the input voice corresponding to the required number of voice signals input to the frame memory is a voice section or a silent section, and the section determining means. Signal processing means for performing compression / expansion processing or deletion processing on the required number of audio signals in accordance with the output and the output of the storage amount calculation means Characterized in that it has a speech speed converting device.

Further, in the above video tape recorder, in order to perform the double speed audio reproduction mode, the input digital audio signal is written into the frame memory and the required number of frame memories in which the digital audio signal is written at a speed corresponding to the set reproduction speed multiplication factor. To the frame memory at the time of the 1 × speed reproduction, the speech speed conversion processing means for performing the speech speed conversion processing on these speech signals every time they are inputted, the output of the speech speed conversion processing means are written. Read-out means for reading data from the ring memory based on a read-out signal having a frequency equal to the writing speed of the ring memory, and storage amount calculation means for calculating the storage amount in the ring memory based on the write-in signal and the read-out signal of the ring memory. The speech speed conversion processing means uses the input sound corresponding to the required number of audio signals input to the frame memory. Is a section discriminating unit for discriminating between a voice section and a silent section, and performs compression / expansion processing or deletion processing on the required number of voice signals in accordance with the output of the section discriminating unit and the output of the storage amount calculating unit. It is characterized in that it has a speech speed conversion device provided with signal processing means.

In the above video tape recorder, the memory is controlled so that the audio data is written into the memory at N times speed and the written data is read out at 1 times speed in order to perform the N times speed reproduction mode. To do.

[0010]

According to the present invention, during double speed reproduction,
The VTR is controlled so that the adaptive voice speed conversion process is performed and the simple thinning process is performed when the speed is three times or more.

According to the present invention, during double-speed reproduction, the compression / expansion process or the deletion process is performed on the input audio signal depending on whether the input audio signal is the audio section or the silent section.

According to the present invention, during the double speed reproduction, the input voice signal is subjected to the voice speed conversion processing by the voice speed conversion processing means. The output of the speech speed conversion processing means is written in the ring memory. The data written in the ring memory is read at a constant speed. In the speech speed conversion processing means, compression / expansion processing or deletion processing is performed on the input voice signal depending on whether the input voice signal is a voice section or a silent section and the amount of storage in the ring memory.

According to the present invention, during double speed reproduction, the input analog audio signal is sampled by the A / D conversion means at the sampling frequency according to the set reproduction speed multiplication factor. The audio signal output from the A / D conversion means is input to the frame memory. Each time a required number of voice signals are input to the frame memory, the voice speed conversion processing means performs the voice speed conversion process on the voice signals. The output of the speech speed conversion processing means is written in the ring memory. The data written in the ring memory is read based on a read signal having a frequency equal to the sampling frequency at the 1 × speed reproduction. The storage amount calculation means calculates the storage amount of the ring memory based on the write signal and the read signal of the ring memory.

According to the present invention, at the time of double speed reproduction, the section discrimination means discriminates the input speech for the required number of speech signals inputted to the frame memory by the section discriminating means. Then, according to the output of the section discriminating means and the output of the accumulated amount calculating means, the compression / expansion processing or the deletion processing is performed on the required number of audio signals.

According to the present invention, during double speed reproduction, the input digital audio signal is written in the frame memory at a speed corresponding to the set reproduction speed multiplication factor. Each time a required number of voice signals are input to the frame memory, the voice speed conversion processing means performs the voice speed conversion process on the voice signals. The output of the speech speed conversion processing means is written in the ring memory. The data written in the ring memory is read at a constant speed based on the read signal. The storage amount calculation means calculates the storage amount of the ring memory based on the write signal and the read signal of the ring memory.

In the speech speed conversion processing means, the section discrimination means discriminates whether the input voice for the required number of voice signals input to the frame memory is the voice section or the silent section. Then, according to the output of the section discriminating means and the output of the accumulated amount calculating means, the compression / expansion processing or the deletion processing is performed on the required number of audio signals.

According to the present invention, a part of the sound is expanded during the ± N speed reproduction, and the remaining signal is thinned out.

[0018]

Embodiments of the present invention will be described below with reference to the drawings.

FIG. 23 is a schematic block diagram of a VTR embodying the present invention. A monaural audio signal reproduced from a horizontal track of a tape T is picked up by an audio head H and input to an equalizer amplifier 111. The audio signal equalized and amplified by the equalizer amplifier 111 is connected to the terminal b of the switching circuit 115 and the speech speed conversion IC.
112. The output of the speech speed conversion IC 112 is supplied to the terminal a of the switching circuit 115. The switching circuit 115 selects the terminal a or b based on a command from the system microcomputer 114, and outputs the selected output via the mute circuit 116. That is, when the audio output from the tape T is output from the terminal b, an audio signal having a speed proportional to the tape speed is output, and the reproduced audio signal is processed from the terminal a such as compression, expansion and deletion. Signal is output. The speech speed conversion IC 112 is controlled by the system microcomputer 114 and performs the above-described processing in cooperation with the memory (dynamic RAM) 113.

Next, the operation of FIG. 23 will be described with reference to the flowchart of FIG.

First, the system microcomputer 114 determines that the VT
R determines whether the reproduction mode is set (S1). When it is determined in step 1 that the mode is the reproduction mode, it is then determined in step 2 (S) whether the double speed reproduction key 117 is pressed.
2). When this double speed playback key is pressed, the tape speed is doubled and video and audio are played back. Then, for the voice, when the double speed reproduction key 117 is pressed, the double speed adaptive talk speed process (S3).
Is done.

This adaptive voice speed processing will be described with reference to FIG. 25. The reproduced voice at the normal speed is reproduced at time T as "VTR using voice speed conversion" as shown in FIG. 25 (a). Assuming that this is applied to the double speed adaptive speech speed processing, a time of 1/2 (T / 2) as shown in FIG.
However, for this reason, the silent part between the words of the normal speed voice, that is, the part that is not speaking, is deleted, and the word part is left at the voice speed close to that of the normal speed voice. Connected. At this time, there is no complete silence due to ambient noise such as object noise, but by distinguishing the ambient noise from the target voice by adaptively changing the silence determination threshold according to the voice situation. ing. If the playback time is not halved just by removing the silence,
The playback time is halved by adaptively compressing a part of the voice according to the length of the silent section. Then, when it is judged in step 6 (S6) that the reproduction is completed, the reproduction is completed.

If the double speed reproduction key is not pressed in step 2 (S2), then it is judged in step 4 (S4) whether or not the mode is ± N double speed mode. This ± N speed mode corresponds to, for example, fast-forward reproduction or rewind reproduction. Then, when it is determined in step 4 that the mode is not the ± N speed mode, the process returns to step 6.

In step 4, if the mode is ± N double speed mode,
The compression process is performed by the simple thinning process (S5), and the process proceeds to step 6.

The simple thinning-out process will be described with reference to FIG. 26. For example, in the case of a reproduced audio signal at a triple speed, as shown in FIG. 26 (a), the time difference adaptive voice speed conversion to the video tape recorder is performed at T1 time. The audio signal "Circuit" is reproduced, but when the simple thinning process is performed on this reproduced signal, "Video TE" which is the T2 period is expanded and reproduced in the T1 time as shown in FIG. It That is, 3
The voice signal for double speed reproduction is converted into the standard speech speed by thinning out 2/3 (thinned period: TD). In other words, T
1: T2 = 3: 1.

Similarly, when the input signal is reproduced at 5 × speed, 4/5 is thinned out and converted to the standard speech speed (T1: T2 =
5: 1), when the input signal is 9 × speed reproduction, 8/9 is thinned out and converted to the standard speech speed (T1: T2 = 9: 1).

Further, the simple thinning-out process for the reverse N-times speed reproduction (-N times speed reproduction) will be explained with reference to FIG. 5. As shown in FIG. Convert to speed. That is, FIG. 27 (a) shows an Nx speed reverse audio signal, and FIG. 27 (b) shows a case where it is thinned out. Convert to speed (T1: T2 =
5: 1), in the case of 9 × reverse playback, 8/9 is thinned out and converted to the standard speech speed (T1: T2 = 9: 1). Incidentally, FIG.
In 7, TD indicates the period of thinning.

As described above, the adaptive voice speed conversion process is performed at the time of the double speed reproduction, and the simple thinning process is performed when the speed is higher than the triple speed. Even if conversion is performed, the amount of voices deleted will be large, the signal processing will not be much different from the simple thinning out, it will not be meaningless to make adaptive speech speed that requires complicated signal processing, and it will be rather difficult to hear. Because.
Therefore, the adaptive voice speed conversion process is performed during the double speed reproduction, and the simple thinning process is performed when the speed is higher than the triple speed, thereby simplifying the structure of the voice speed conversion IC. Not only that, it can improve the commercial value of the VTR.

FIG. 1 shows the overall structure of a speech speed conversion device corresponding to a portion for performing adaptive speech speed conversion in the speech speed conversion IC.

The input audio signal is amplified by the ALC amplifier 1 and then sent to the A / D converter 2 where it is converted into a 12-bit digital signal, for example. The standard sampling frequency of the A / D converter 2 is, for example, 8 KHz. During double speed reproduction, the sampling frequency fs of the A / D converter 2
AD becomes 16 KHz.

The output of the A / D converter 2 is a DSP (Digit
l Signal Processor) 4 and the level detector 3. The level detection unit 3 includes the A / D conversion unit 2
When the A / D converted data at the maximum value of the conversion range, ALC (automatic level control) signal is changed to A
Output to LC amplifier 1. As a result, the ALC amplifier 1
The amplifier gain of is controlled so that the input signal of the A / D converter 2 does not exceed the maximum range. That is, when the playback tape speed of the VTR changes, the input signal level of the ALC amplifier 1 also changes. Therefore, by automatically adjusting the amplifier gain based on the output of the level detection unit 3, the A / D
The input signal of the conversion unit 2 is prevented from exceeding the maximum range.

The DSP 4 is provided with a frame memory 5 having a capacity capable of storing voice signals for two frames, and a voice speed conversion unit 6 for performing voice speed conversion processing on the voice signals stored in the frame memory 5 in units of frames. ing. Here, it is assumed that one frame is composed of 200 pieces of sampling data.

Of the first half area and the second half area in the frame memory 5, the voice signal for one frame stored in one area is processed by the speech speed conversion unit 6, and at the same time, in the other area. The signal from the / D converter 2 is accumulated. Then, when a signal for one frame is accumulated in the other area, this time the data in that area is processed by the speech speed conversion unit 6 and at the same time the already processed data is stored. The signal from the A / D conversion unit 2 is accumulated in the above-described one area.

The data output from the speech speed converter 6 is written in the ring memory 7 based on the write clock. The data written in the ring memory 7 is read based on the read clock. The signal read from the ring memory 7 is converted into an analog signal by the D / A conversion unit 8, amplified by the amplifier 10, and output as an audio output signal.

Sampling frequency fs of D / A converter 8
DA is 8 KHz. The frequency of the read clock of the ring memory 7 is also 8 KHz. As the ring memory 7, a 21845 × 12 bit memory, that is, a 21845 word memory is used. Therefore, the maximum time that data can be stored in the ring memory 7 (maximum delay time of output time with respect to input signal) is 21
It becomes 845 × 1/8000 = 2.73 seconds.

The write clock for the ring memory 7 is input to the up-count input terminal (UP) of the up-down counter 9. The read clock for the ring memory 7 is input to the down-count input terminal (DOWN) of the up-down counter 9. The up / down counter 9 calculates the difference between the total number of input write clocks and the total number of input read clocks (the ring memory 7
(Accumulation amount of) is counted, and the count value is 15 bits
Output as a digital signal of. The output of the up / down counter 9 is sent to the speech speed conversion unit 6.

FIG. 2 shows the detailed structure of the speech speed converter 6.

The audio signal read from the frame memory 5 is sent to the power calculator 11 to calculate the average power value P of the audio signal for one frame. This average power value P is the amplitude of each voice signal sampled in one frame i0, i1, ... iN-1 (where N = 2).
00), it is calculated by the following formula 1.

[0039]

[Equation 1]

The average power value P obtained by the power calculation unit 11 is sent to the comparison unit 12. The threshold Th is sent from the threshold memory 13 to the comparison unit 12, and the average power value P is equal to or larger than the threshold Th (P ≧ Th) or the average power value P is smaller than the threshold Th. (P <Th) is determined. When the average power value P is greater than or equal to the threshold value Th (P ≧ Th), the comparison unit 12 outputs a signal indicating that the current frame is in the voice section, and when the average power value P is less than the threshold value Th, the current frame is detected. A signal indicating that each is a silent section is output.

The threshold value Th is, for example, 2 when the quantization bit number of the A / D converter 2 is 12 bits.
It is set to 12. The threshold value T is set as follows.
You may make it change h. That is, as shown by the dotted line in FIG. 2, the power steady state detection and threshold updating unit 14 is provided. The power steady state detection / threshold value update unit 14 uses the average power value P from the power calculation unit 11.
Is constant over a predetermined number of frames (for example, 40 frames), and when it is constant (steady state), a value twice the average power value P at that time is stored in the threshold memory 13. Write and update the threshold value Th. However, the maximum value of the updated threshold value is limited to a predetermined value, for example, 214. By doing so, it becomes possible to handle the noise that is constantly generated as a silent section.

Further, the voice section and the silent section of the input signal may be discriminated on the basis of the power cumulative value Pa of the voice signal of each frame and a given threshold value, which is expressed by the following formula 2. .

[0043]

[Equation 2]

The output of the comparison unit 12 is sent to the conditional branching unit 15. The output of the ring memory storage amount state determination unit 16 is input to the conditional branching unit 15. Also, the conditional branching unit 1
An audio signal from the frame memory 5 is sent to the frame 5 via the power calculator 11. Furthermore, the conditional branching unit 1
A pause continuation length setting memory 17 is connected to 5. In the pause duration setting memory 17, a pause duration Tdel (silence deletion start point determination value) for determining a deletion start point of a silent section is set.

The ring memory accumulated amount state discriminating unit 16 determines that the state of the ring memory 7 has reached the state immediately before the overflow, and that the state of the ring memory 7 is under, based on the accumulated amount sent from the up / down counter 9. It detects that it is in the state just before the flow.

That is, the overflow detection data memory 18 stores overflow detection data Tmax, and the underflow detection data memory 19 stores underflow detection data Tmin.
The overflow detection data Tmax is set to a value 21645 smaller than the total number of words (TOTAL) 21845 of the ring memory 7 by 200, for example. The underflow detection data Tmin is set to 200, for example.

The accumulated amount sent from the up / down counter 9 is the overflow detection data Tmax.
In the above case, the immediately preceding overflow detection signal is output from the ring memory storage amount state determination unit 16. Further, when the storage amount sent from the up / down counter 9 becomes equal to or less than the underflow detection data Tmin, the ring memory storage amount state determination unit 16 outputs a detection signal immediately before underflow. The conditional branching unit 15 determines that the ring memory 7 is in the state immediately before the overflow when the detection signal immediately before the overflow is input, and the ring memory 7 is in the state immediately before the underflow when the detection signal immediately before the underflow is input. To determine.

The conditional branching unit 15 judges whether the voice section or the silent section is sent from the comparing unit 12, the detection signal concerning the ring memory state sent from the ring memory storage amount state judging unit 16, and the pause continuation. Long setting memory 1
Based on the pause continuation length Tdel set to 7, the following 6 cases are classified. And
In response to this, the multiplexer 20 is controlled to send the audio signal to a predetermined processing unit. (1) First Case (case 1) When it is determined that the input signal is in the voice section and the ring memory 7 is not in the state immediately before overflow, the first case is performed.

In this case, the audio signal is sent to the pitch compression / expansion means 23 via the multiplexer 20.
The pitch compression / decompression means 23 performs variable speech control (VSC), and performs decompression / compression processing on the input signal at a compression rate higher than the compression rate 1 / n, where n is the reproduction speed multiplication factor. As the decompression / compression method used here, for example, the overlap addition method (Pointer Interval Control Overlap) by the pointer movement amount control is used.
and Add: PICOLA), TDHS (TimeDomain Ha
rmonic Scaling) method. Pitch extension / compression means 23
The signal subjected to the decompression / compression processing in (1) is sent to the ring memory 7 via the demultiplexer 27, and is written in the ring memory 7 in accordance with the write clock.

During double speed reproduction of VTR, A / D
The sampling frequency fsAD of the conversion unit 2 is 16 KHZ, and the sampling frequency fsDA of the D / A conversion unit 8 is 8 KHZ. Therefore, the pitch is restored and output.

In the conventional general time-base decompression / compression, compression is performed at a compression rate of 1/2 during VTR double speed reproduction. In other words, the 2-pitch cycle is thinned out to the 1-pitch cycle. Therefore, the output voice becomes twice the standard voice speed. That is, in the normal reproduction of the double speed reproduction, the output sound is double the standard sound speed. However, the pitch remains unchanged.

On the other hand, in the pitch expansion / compression means 23 provided in the speech speed conversion unit 6 of FIG. 2, the compression ratio is 1/2.
Set to a larger value. Here, the compression rate is 2/3
Is set to. In other words, the 3-pitch cycle is thinned out to the 2-pitch cycle. Therefore, the output voice is
It is 3/2 times the standard voice speed. In this case as well, the pitch is
It remains as it was. In this way, when compressed at a compression rate of 2/3, it is 2 / 3-1 as compared with the case where the compression rate is 1/2.
The signal will be expanded by / 2 = 1/6. This expanded amount becomes the accumulated amount in the ring memory 7.

A method of compressing an input signal at a compression rate of 2/3 using PICOLA will be briefly described with reference to FIG. First, the pitch period is extracted from the input signal.
The extracted pitch period is Tp. A weight (weighting function K1) that linearly goes from 1 to 0 is added to the waveform A to create the waveform A ′. A weight (weight function K2) from 0 to 1 is attached to the waveform B,
Waveform B'is created.

Then, the waveforms A'and B'are added together to create a waveform A '* B' of length Tp.
These weights are added to maintain continuity at the connection points before and after the waveform A ′ * B ′. Next, the pointer is moved by 3 Tp, which is a length determined based on the compression rate, and the same operation is performed. As a result, two waveforms A ′ * B ′ and C are obtained from the three waveforms A, B, and C. In this way, the signal for 3 pitch periods becomes 2
It is compressed into a signal for a pitch period.

As the expansion / compression method by the pitch expansion / compression means 23, as shown in FIGS. 17 (a) and 17 (b), expansion / compression processing is performed in units of fixed frame length Ts of a predetermined length without pitch extraction. You may do it. The fixed frame length Ts is set to, for example, the length of 200 pieces of input data. In the example shown in FIG. 17, 3Ts is changed to 2Ts.

In the method of FIG. 17A, of the waveforms A, B, and C having the fixed frame length Ts, the waveform A is weighted linearly from 1 to 0 (weighting function K1). Waveform A "is created as a result. For waveform B, 0 to 1
A weight (weighting function K2) is applied to the waveform B ″
Is created.

Then, these waveforms A "and B" are added together to create a waveform A "* B" of length Ts.
These weights are added to maintain continuity at the connection points before and after the waveform A "* B". Then, the next waveform C
Is output as it is. This results in two waveforms A "* B" and C from the three waveforms A, B, C. In this way, the signal for 3 Ts is compressed into the signal for 2 Ts.

In the method of FIG. 17 (b), for example, the waveform A of the fixed frame length Ts of the waveforms A to C has a weight (weighting function K3) that linearly goes from 0 to 1 in the 20 pieces of data from the beginning. To obtain waveform A ″. Waveform B has 181
A waveform B ″ is obtained by adding weights (weighting function K4) that linearly goes from 1 to 0 to the first to 200th input data. Then, the waveform C is deleted. The next three waveforms D to F
The same process is performed for. In this way,
A signal composed of three waveforms A to C (or D to F) is 2
It is compressed into a signal consisting of two waveforms A "and B" (or D "and E"). In other words, the signal for 3Ts is 2
It is compressed into a signal for Ts.

When the decompression / compression process in fixed frame length units is used, the sound quality is lower than that in the case of using the decompression / compression process for each pitch period, but the processing amount is reduced.

When this speech speed converter is applied to an English learning device (during 1 × speed reproduction), the sampling frequency fsAD of the A / D converter 2 is 8 KHZ and D
The sampling frequency fsDA of the A / A converter 8 is 8 KHZ
Is. In this case, the pitch compression / expansion means 23 expands the audio signal at a compression rate of 3/2 so that the 2-pitch cycle becomes a 3-pitch cycle. That is, the voice section is expanded 1.5 times. So in this case,
3 / 2−1 = 1/2 compared to normal playback at 1 × speed
The signal is expanded only by this amount, and the expanded amount becomes the accumulated amount in the ring memory 7. (2) Second case (case 2) When it is determined that the input signal is in the voice section and the ring memory 7 is in a state immediately before overflow, the second case
It becomes a case.

In this case, the audio signal is sent to the input signal deleting section 21 via the multiplexer 20, and the audio signal is deleted. Specifically, the write operation to the ring memory 7 is stopped until the count value of the up / down counter 9 becomes equal to or less than the underflow detection data Tmin, that is, until the ring memory 7 is in the state immediately before underflow.

When the ring memory 7 is in a state immediately before underflow, 200 or less, for example, 100 mute signals (signals of value "0") are output from the mute insertion section 22, and the mute signals are output from the demultiplexer 27. Is sent to and written in the ring memory 7 via. As described above, the mute signal is written in the ring memory 7 in order to prevent a click sound from being generated at the end of the voice signal due to voice deletion. (3) Third Case (case 3) When it is determined that the input signal is in the silent section, the duration of the silent section is less than the set pause duration Tdel, and the ring memory 7 is not in a state immediately before overflow. , The third case.

In this case, the same processing as in the first case is performed. However, in the case of the third case, the decompression / compression process may be performed at a compression ratio of 1 / n, where n is the reproduction speed magnification. That is, in the case of the third case, the decompression / compression process is performed at a compression rate of 1 / n or more. (4) Fourth Case (case 4) When it is determined that the input signal is a silent section, the duration of the silent section is less than the set pause duration Tdel, and the ring memory 7 is in a state immediately before overflow. , The fourth case.

In this case, the same processing as in the second case is performed. (5) Fifth Case (case 5) It is determined that the input signal is a silent section, the duration of the silent section is equal to or longer than the set pause duration Tdel, and the ring memory 7 is not in the state immediately before underflow. Sometimes it is the fifth case.

In this case, the audio signal is sent to the input signal deleting section 25 via the multiplexer 20, and the audio signal is deleted. Specifically, the write operation to the ring memory 7 is stopped. However, in order to prevent the start portion (unvoiced section) of the voice section from being lost, and to prevent a click sound from being generated at the end of the voice due to the deletion of the voice, the waveform synthesizing and inserting unit 26 performs waveform synthesizing. Insertion processing is performed.

Waveform synthesis insertion processing by the waveform synthesis insertion unit 26 will be described with reference to FIGS. In the method according to FIG. 4 (a), the waveform synthesis insertion unit 26 uses the first
The memory 31 and the second memory 32 are provided. At the start of the input signal deleting process by the input signal deleting unit 25, a predetermined length T equal to or less than one frame length is set from the deletion start point.
s, for example, an input signal for one frame is stored in the first memory 31
Are sequentially stored in the order of address. Next, the first memory 31
The content A of the first memory 31 is multiplied by a function K1 which linearly changes from 1 to 0 as the address of becomes larger. Then, the multiplication result A ′ is written in the first memory 31 again.

Further, the input signal of the predetermined length Ts immediately before the end point of the input signal deleting section by the input signal deleting section 25 is
The data is sequentially stored in the second memory 32 in the order of addresses. next,
0 to 1 as the address of the second memory 32 increases
The content B of the second memory 32 is multiplied by the function K2 that linearly changes to. Then, the multiplication result B ′ is written in the second memory 32 again. After this, the first memory 31
The contents A ′ of the above and the contents B ′ of the second memory 32 are added to obtain the data A ′ * B ′ of the predetermined length Ts.
Then, the obtained data A ′ * B ′ of the predetermined length Ts is sent to the ring memory 7 via the demultiplexer 27 and written in the ring memory 7.

In the method according to FIG. 4B, an input signal of a predetermined length Ts equal to or less than one frame length, for example, one frame from the deletion start point is sequentially stored in the first memory 31 in the order of addresses. Next, the function K3 having a slope that linearly changes from 1 to 0 at the rear end is the content A of the first memory 31.
Is multiplied by. Then, the multiplication result A ′ is again the first
It is written in the memory 31.

Further, the input signal of the predetermined length Ts immediately before the end point of the input signal deleting section by the input signal deleting section 25 is
The data is sequentially stored in the second memory 32 in the order of addresses. next,
The content B of the second memory 32 is multiplied by a function K4 having a slope that linearly changes from 0 to 1 at the front end. Then, the multiplication result B ′ is written in the second memory 32 again. After that, the contents A ′ of the first memory 31 and the contents B ′ of the second memory 32 are mixed together to obtain 2Ts worth of data A ′ + B ′. And the obtained 2Ts
The minute data A ′ + B ′ is sent to the ring memory 7 via the demultiplexer 27 and written in the ring memory 7. In FIG. 4B, an example in which Ts has a length of one frame is shown, but data having a half length of one frame may be Ts.

When the input signal deleting unit 25 repeatedly deletes the voice signal in the silent section, the ring memory 7 may be in a state immediately before underflow. In this case, the input signal for the predetermined length Ts is stored in the second memory 32 from the time when the ring memory 7 is in the state immediately before underflow. Then, based on the data stored in the first memory 31 and the data stored in the second memory 32, the same waveform synthesis insertion processing as described above is performed. (6) Sixth case (case 6) It is determined that the input signal is a silent section, the duration of the silent section is equal to or longer than the set pause duration Tdel, and the ring memory 7 is in a state immediately before underflow. Sometimes it is the sixth case.

In this case, the input signal is sent to the thinning processing section 24 via the multiplexer 20. In the thinning-out processing unit 24, the thinning-out processing is performed so that the compression rate becomes 1 / n, where n is the reproduction speed multiplication factor of the VTR. For example, during double-speed reproduction, the input signal is thinned out at a compression rate of 1/2, and during triple-speed reproduction, the input signal is thinned out at a compression rate of 1/3. During 1 × speed reproduction, the input signal is output as it is.

The following method is used for the thinning processing by the 1 / n thinning processing section 24. Here, 2
Description will be made by taking the case of double speed reproduction as an example.

Using the time base compression method using PICOLA or TDHS described above, the pitch of the input signal is extracted,
The pitch data portion is thinned out so that the compression rate becomes 1/2.

Further, as shown in FIGS. 5A to 5C,
The waveform may be thinned out every predetermined time Ts without performing pitch extraction.

In the method of FIG. 5A, the waveform B and the waveform D are thinned out of the waveforms A to D, and a signal composed of the waveforms A and C is obtained.

In the method of FIG. 5B, the waveform B and the waveform D among the waveforms A to D are thinned out. In addition, the waveform A has a slope (function K4) that rises from 0 to 1 at the front end.
Is multiplied by a function having a slope (function K3) that decreases from 1 to 0 at the rear end, and a waveform A ′ is created. The waveform C is created by multiplying the waveform C by a function with a slope (function K4) increasing from 0 to 1 at the front end and a slope (function K3) decreasing from 1 to 0 at the rear end. To be done. In this way, the signal composed of the four waveforms A to D is compressed into the signal composed of the two waveforms A ′ and C ′.

In the method of FIG. 5C, a weight (weighting function K1) that linearly goes from 1 to 0 is added to the waveform A to create the waveform A '. A weight (weight function K2) from 0 to 1 is attached to the waveform B,
Waveform B'is created. Then, these waveforms A'and B'are added together to create a waveform A '* B' of length Ts.

Similarly, a weight (function K1) that linearly goes from 1 to 0 is added to the waveform C, and the waveform C '
Is created. A weight (function K2) from 0 to 1 is applied to the waveform D to create the waveform D '. Then, the waveforms C'and D'are added together to create a waveform C '* D' of length Ts. In this way, the signal composed of the four waveforms A to D becomes two waveforms A ′.
Compressed to a signal consisting of * B 'and C' * D '.

As described above, in the case of the sixth case, the reproduction ratio of the VTR is n, and the thinning process is performed at the compression ratio 1 / n. However, the compression ratio is controlled as follows. You may do it.

When the thinning process is performed at the compression rate 1 / n, the ratio f between the sampling frequency fsDA of the D / A converter 8 and the sampling frequency fsAD of the A / D converter 2 is set.
When sDA / fsAD is equal to the compression ratio 1 / n,
The storage amount of the ring memory 7 does not change. However, the calculation accuracy of the compression rate 1 / n, the sampling frequency fs
Depending on the clock accuracy of AD and fsDA, fsDA /
It is possible that fsAD does not equal compression ratio 1 / n.

When fsDA / fsAD becomes larger than the compression rate 1 / n (fsDA / fsAD> 1 / n),
As fsDA / fsAD = 1 / a (a> 0), {(1
/ A)-(1 / n)}, the compression rate decreases, the degree of thinning increases, the storage amount of the ring memory 7 decreases, and the storage amount of the ring memory 7 may underflow. .

On the other hand, fsDA / fsAD is the compression ratio 1 /
When it becomes smaller than n (fsDA / fsAD <1 /
In n), fsDA / fsAD = 1 / a (a> 0), the compression ratio increases and the degree of thinning decreases by {(1 / n) − (1 / a)}, and the ring memory 7 The accumulated amount of is increasing.

Therefore, when thinning processing is performed,
After confirming the storage amount in the ring memory 7, the compression rate is controlled as follows. As fsDA / fsAD = 1 / a (a> 0), α that satisfies the condition of (1 / n) −α <1 / a <(1 / n) + α is selected. However, α is a value of 0 or more and 1 or less, for example, a value in the range of 0.001 to 0.1.

When fsDA / fsAD becomes larger than the compression rate 1 / n, that is, when the storage amount of the ring memory 7 decreases, the compression rate is changed from 1 / n to {(1 /
n) + α}. That is, the compression rate is increased and the storage amount of the ring memory 7 is increased.

When fsDA / fsAD becomes smaller than the compression rate 1 / n, that is, when the storage amount of the ring memory 7 increases, the compression rate is changed from 1 / n to {(1 /
n) -α}. That is, the compression rate is reduced and the amount of storage in the ring memory 7 is reduced.

In the above description, the compression rate is changed based on the amount stored in the ring memory 7. However, when thinning processing is performed, the compression rate is {(1 / n) -α} or {for each frame. Alternatively, it may be changed to (1 / n) + α}.

6 and 7 show the processing procedure by the speech speed conversion unit 6.

The processing by the speech speed conversion unit 6 in the VTR double speed reproduction will be described below. (1) Processing at the start of reproduction When reproduction is started and the average power value P of the first frame is calculated by the power calculation unit 11 (step 1), the calculated average power value P is equal to or greater than the threshold Th. Whether or not it is determined based on the output of the comparison unit 12 (step 2).

When the input voice signal starts from the silent section, the average power value P becomes smaller than the threshold value Th in the first frame, and the routine proceeds to step 11. Then, the duration of the silent section (the number of frames in which the silent section continues) is calculated, and it is determined whether or not the calculated duration is equal to or longer than the pause duration Tdel set in the pause duration memory 17 (step). 12). This pose duration T
del is set to a length corresponding to four frames, for example, as the number of frames.

In the processing for the first frame,
Since the duration of the silent section is less than the pause duration Tdel, it is determined based on the output of the ring memory storage amount state determination unit 16 whether or not the ring memory 7 is in the state immediately before underflow (steps 13 and 14). .

In the processing for the first frame,
Since the ring memory 7 is in the state immediately before underflow, the frame data is thinned out by the thinning processing unit 24 at a compression rate of 1/2 (step 28), and the compressed data after the thinning processing is written in the ring memory 7. . Then, the process returns to step 1. (2) Description of processing that is the first case When it is determined in step 2 that the average power value P is equal to or greater than the threshold value Th, it is determined that the current frame is the voice section, and the process proceeds to step 3. In step 3,
Whether or not the previous frame was the deletion section is determined by the first flag F.
It is determined based on the state of 1. If the previous frame is not in the deletion section, it is determined whether or not the ring memory 7 is in the state immediately before overflow based on the output of the ring memory accumulated amount state determination unit 16 (steps 6 and 7). If the previous frame is the deletion section, after the processes of steps 4 and 5 are performed, it is determined whether or not the ring memory 7 is in the state immediately before the overflow (steps 6 and 7). The processing of steps 4 and 5 will be described later.

When it is determined in step 7 that the state is not immediately before the overflow, the first case is performed, and the pitch compression / expansion means 23 temporally compresses the current frame data at a compression ratio of 2/3 ( Step 8). The compressed data is sent to and written in the ring memory 7. Then, the process returns to step 1. (2) Description of processing that is the second case When it is determined in step 2 that the average power value P is greater than or equal to the threshold value Th, it is determined that the frame sent this time is in the voice section, and step 3 Proceed to. In step 3, it is determined whether or not the previous frame is the deletion section based on the state of the first flag F1. If the previous frame is not the deletion section, it is determined whether or not the ring memory 7 is in the state immediately before overflow based on the output of the ring memory accumulated amount state determination unit 16 (step 6,
7). If the previous frame is the deletion section, after the processes of steps 4 and 5 are performed, it is determined whether or not the ring memory 7 is in the state immediately before the overflow (steps 6 and 7). The processing of steps 4 and 5 will be described later.

If it is determined in step 7 that the state is just before the overflow, the second case is established.
The input signal is deleted by the input signal deletion unit 21 until the underflow detection signal is output from the ring memory accumulation amount state determination unit 16 (step 9). That is, writing to the ring memory 7 is stopped until the ring memory 7 is in a state immediately before underflow.

When the ring memory 7 is in a state immediately before underflow, the muffling insertion section 22 writes a predetermined number of muffling signals "0" of 200 or less to the ring memory 7 (step 10). Then, the process returns to step 1.

Instead of the processing of the above step 10, FIG.
You may perform the process as shown in (a) or FIG.9 (b). The method shown in FIG. 9A will be described. For example, for the waveform A with respect to 200 input signals, the weight (1) that linearly goes from 1 to 0 since the state immediately before the overflow is determined in step 7 ( Weight function K
1) is added to obtain the waveform A ′. Further, for the waveform B for 200 input signals from immediately before underflow to 200 before, a weight (weight function K2) that goes from 0 to 1
To obtain the waveform B ′.

Then, the obtained two waveforms A'and B'are added together to form a waveform A '* having a length of 200 pieces.
Create B '. Then, 200 signals corresponding to the waveform A ′ * B ′ are written in the ring memory 7. The detection of 200 points before the underflow is performed based on the count value of the up / down counter 9. As a result, it is possible to effectively prevent the click sound from being generated at the end of the audio signal before and after the audio deletion section.

The method shown in FIG. 9B will be described. From the time when it is determined in step 7 that the state immediately before the overflow occurs, for example, the waveform A for 100 input signals is linearly changed from 1 to 0. A waveform A ′ is obtained by applying a weight (weighting function K1) to the direction. Further, for the waveform B with respect to 100 input signals from immediately before underflow to 100 before, a weight (weight function K
2) is added to obtain the waveform B ′. Then, 200 signals obtained by combining the obtained two waveforms A ′ and B ′ are written in the ring memory 7.

In step 9, when it is determined that the state immediately before the overflow occurs, the input signal is deleted by the input signal deleting section 21 until the underflow detection signal is output from the ring memory accumulated amount state judging section 16. However, the data accumulated in the ring memory 7 may be deleted so that the ring memory 7 is in a state immediately before underflow.

Specifically, the write start address of the ring memory 7 is changed from the address (point C) in the state immediately before the overflow shown in FIG. 18A to the ring memory 7 as shown in FIG. 18B. Causes the address to jump to the address (point A) where it will be in the state just before underflow. Therefore, in the process of step 9, the data accumulated at the addresses from point A to point C is deleted. Thereafter, as shown in FIG. 18C, the mute signal is written in step 10 and then the input data is written.

In step 9, as described above, when the data accumulated in the ring memory 7 is deleted so that the ring memory 7 is in the state immediately before underflow, the mute signal is written in the ring memory 7 in step 10. 19 (a) or 19 (b) may be performed.

Now, as shown in FIG. 18B, the ring memory 7 underflows from the address (point C) when the write start address of the ring memory 7 is in the state immediately before the overflow shown in FIG. 18A. It is assumed that a jump has been made to the address (point A) that is in the immediately preceding state. For the data S accumulated from this point A to a predetermined number, for example, an address 200 points ahead (point B in FIG. 19A),
As shown in (a), a weight (weight function K1) that linearly goes from 1 to 0 is added to obtain a waveform S '. Further, as shown in FIG. 19A, a weight (weight function K2) from 0 to 1 is added to 200 pieces of input data (waveform T) written in the ring memory 7 thereafter. ,
Obtain the waveform T '.

Then, the two obtained waveforms S'and T'are added together to form a waveform S '* of 200 lengths.
Create T '. Then, 200 signals for this waveform S ′ * T ′ are written in the ring memory 7 from the point A. As a result, it is possible to effectively prevent the click sound from being generated at the end of the audio signal before and after the accumulated data deletion section.

Explaining the method shown in FIG. 19 (b), the data accumulated from point A in FIG. 18 (b) to a predetermined number, for example, 100 addresses ahead (point B in FIG. 19 (b)). For S, a weight (weighting function K1) that linearly goes from 1 to 0 is added to obtain a waveform S ′. Also,
For 100 pieces of input data (waveform T) written in the ring memory 7 thereafter, a weight (weight function K2) from 0 to 1 is applied to obtain a waveform T '. Then, 200 signals obtained by combining the obtained two waveforms S ′ and T ′ are written into the ring memory 7 from the point A. (3) Description of processing that is the third case When it is determined in step 2 that the average power value P is smaller than the threshold value Th, the duration of the silent section up to this time is calculated (step 11) and calculated. The pause length is set in the pause duration memory 17 and the pause duration Tde is set.
It is determined whether or not it is 1 or more (step 12). And
When it is determined that the duration of the silent section is less than the pause duration Tdel, the ring memory storage amount state determination unit 1
Based on the output of 6, it is determined whether or not the state is immediately before underflow (steps 13 and 14).

When the ring memory 7 is not in the state immediately before the underflow, it is determined whether or not it is in the state immediately before the overflow based on the output of the ring memory accumulated amount state determination unit 16 (steps 6 and 7). If it is not the state immediately before the overflow, the third case occurs, and the pitch compression / expansion means 23 sets the current frame data to 2/3.
The time axis is compressed at the compression ratio of (step 8). The compressed data is sent to and written in the ring memory 7. Then, the process returns to step 1. (4) Description of processing that is the fourth case When it is determined in step 2 that the average power value P is smaller than the threshold value Th, the duration of the silent section up to this time is calculated (step 11) and calculated. The pause length is set in the pause duration memory 17 and the pause duration Tde is set.
It is determined whether or not it is 1 or more (step 12). And
When it is determined that the duration of the silent section is less than the pause duration Tdel, the ring memory storage amount state determination unit 1
Based on the output of 6, it is determined whether or not the state is immediately before underflow (steps 13 and 14).

When the ring memory 7 is not in the state immediately before underflow, it is determined whether or not it is in the state immediately before overflow based on the output of the ring memory accumulation amount state determination unit 16 (steps 6 and 7). In the case of the state immediately before the overflow, the fourth case occurs, and the input signal is deleted by the input signal deletion unit 21 until the underflow detection signal is output from the ring memory accumulation amount state determination unit 16 (step 9). That is, the ring memory 7
Ring memory 7 until
Writing to is interrupted.

When the ring memory 7 is in a state immediately before underflow, the muffling insertion section 22 writes a predetermined number of muffling signals "0" of 200 or less to the ring memory 7 (step 10). Then, the process returns to step 1. (5) Description of the Process that is the Fifth Case When it is determined in step 2 that the average power value P is smaller than the threshold value Th, the duration of the silent section up to this time is calculated (step 11) and calculated. The pause length is set in the pause duration memory 17 and the pause duration Tde is set.
It is determined whether or not it is 1 or more (step 12). And
When it is determined that the duration of the silent section is equal to or longer than the pause duration Tdel, the ring memory storage amount state determination unit 1
Based on the output of 6, it is determined whether or not the state is immediately before underflow (steps 15 and 16).

When the ring memory 7 is not in the state immediately before underflow, the fifth case is set, and the first flag F1 indicating that the current frame is the deletion section by the input signal deletion unit 25 is set (step 17). The first flag F1 is reset (F1 = 0) in the initial setting when the power is turned on. Then, it is determined whether or not the second flag F2 indicating whether or not the current frame is the first frame of the deletion section by the input signal deletion unit 25 is reset (step 18).

The second flag F2 is reset (F2 = 0) in the initial setting when the power is turned on. Then, it is set (F2 = 1) when the processing for the first frame of the deletion section by the input signal deletion unit 25 is completed. Then, it is reset (F2 =
0).

Therefore, when the current frame is the first frame of the deletion section by the input signal deletion unit 25, the second flag F2 is reset (F2 = 0). When the second flag F2 is reset, the waveform synthesis insertion section 26 stores the current frame data in the first memory 31 (step 19). Further, the input signal deleting unit 25 stops the writing of the current frame data to the ring memory 7 (step 20).
That is, the current frame data is deleted. And
After the second flag F2 is set (F2 = 1) (step 21), the process returns to step 1.

Further, when there is a silent section,
After passing through steps 2, 11, 12, and 15, the process proceeds to step 16, where it is determined based on the output of the ring memory accumulated amount state determination unit 16 whether or not the ring memory 7 is in the state immediately before underflow.

When the ring memory 7 is not in the state immediately before underflow, the current frame is the input signal deleting section 25.
The first flag F1 indicating that the section is a deletion section is set (step 17). Then, it is determined whether or not the second flag F2 indicating whether or not the current frame is the first frame of the deletion section by the input signal deletion unit 25 is reset (step 18).

In this case, since the second flag F2 is set (F2 = 1), it is determined that the current frame is not the first frame of the deletion section by the input signal deletion unit 25. In this case, the waveform synthesis insertion unit 26 stores the current frame data in the second memory 32 (step 22). Further, the input signal deleting unit 25 stops the writing of the current frame data to the ring memory 7 (step 23). Then, the process returns to step 1.

Further, when the silent section continues and the ring memory 7 is not in the state immediately before underflow, steps 2, 11, 12, 15, 16, 17,
The processing of 18, 22 and 23 is repeated. That is,
The frame data in the second memory 32 is updated, and the writing of the frame data to the ring memory 7 is stopped.

Thereafter, when the frame data of the voice section is input, in step 2, the average power value P
Is equal to or greater than the threshold Th, it is determined based on the state of the first flag F1 whether or not the previous frame is the deletion section by the input signal deletion unit 25 (step 3). In this case, since the first flag F1 is set (F1 = 1), it is determined that the previous frame is the deletion section by the input signal deletion unit 25, and the process proceeds to step 4. In step 4, the deletion processing by the input signal deletion unit 25 is stopped and the waveform synthesis insertion processing by the waveform synthesis insertion unit 26 is performed.

That is, as already described with reference to FIG. 4A, the contents of the first memory 31 are multiplied by the function that linearly changes from 1 to 0, and the contents of the second memory 32 are changed from 0 to 1. Is multiplied by a linearly varying function and the results of both multiplications are added together. The addition result (corresponding to A ′ * B ′ in FIG. 4A) is sent to the ring memory 7 via the demultiplexer 27 and written in the ring memory 7.

After that, the first flag F1 and the second flag F2 are reset (F1 = F2 = 0) (step 5). Then, the process proceeds to step 6.

By the way, in the case where the deletion processing by the input signal deletion unit 25 as described above is repeatedly performed on the continuous silent section, the ring memory 7
May be in a state just before underflow. In this case, YES is obtained in the above step 16 and step 24
Move on to. In step 24, it is determined whether or not the previous frame is the deletion section by the input signal deletion unit 25, the first flag F.
It is determined based on the state of 1.

In this case, since the first flag F1 is set (F1 = 1), the routine proceeds to step 25, where the current frame data is stored in the second memory 32. Then, the deletion processing by the input signal deletion unit 25 is stopped, and the waveform synthesis insertion processing is performed by the waveform synthesis insertion unit 26 (step 26). Then, the first flag F1 and the second flag F2 are reset (F1 = F2
= 0) (step 27), the process proceeds to step 1.

The waveform synthesizing / inserting process by the waveform synthesizing / inserting unit 26 in the above step 26 is almost the same as the waveform synthesizing / inserting process described in the above step 4, except that the frame data stored in the second memory 32 is This is different from the case of the processing described in step 4 above in that it is the frame data after the ring memory 7 is in the state immediately before underflow.

The process of step 25 is omitted and
If YES in step 24, the second memory 32
It is also possible to move to step 26 without storing the current frame data. In this case, in the waveform synthesizing / inserting process performed in step 26, as in the waveform synthesizing / inserting process described in step 4, the second
The frame data (previous frame data) before the underflow state stored in the memory 32 is used.

Further, the processing of step 22 may be omitted, and a step of storing the frame data in the second memory 32 may be added between steps 3 and 4. In this case, step 4
In step 19, the first memory 31
The contents recorded in step 3 and step 4 above.
Based on the contents recorded in the second memory 32 in the steps added between and, the waveform synthesis insertion processing is performed. (6) Description of Processing Being the Sixth Case When it is determined in step 2 that the average power value P is smaller than the threshold value Th, the duration of the silent section up to this time is calculated (step 11) and calculated. The pause length is set in the pause duration memory 17 and the pause duration Tde is set.
It is determined whether or not it is 1 or more (step 12). And
When it is determined that the duration of the silent section is equal to or longer than the pause duration Tdel, the ring memory storage amount state determination unit 1
Based on the output of 6, it is determined whether or not the state is immediately before underflow (steps 15 and 16).

When the ring memory 7 is in the state immediately before underflow, it is determined whether or not the previous frame is the deletion section by the input signal deletion unit 25 based on the state of the first flag F1 (step 24). First flag F1
Is reset (F1 = 0), that is, when the previous frame is not the deletion section by the input signal deletion unit 25, the sixth case is performed, and the process proceeds to step 28.
In step 28, the thinning processing section 24 thins the current frame data at a compression rate of 1/2. Then, the thinned data is sent to the ring memory 7 and written. Then, the process returns to step 1.

That is, even if the duration of the silent section is equal to or longer than the pause duration Tdel, if the ring memory 7 is in the state immediately before underflow and the previous frame is not the section deleted by the input signal deleting unit 25, The data is not deleted, is thinned out at a compression rate of 1/2, and then written in the ring memory 7.

In FIG. 7, in step 12,
It is determined whether or not the duration of the silent section is longer than the set pause duration Tdel, but step 12 in FIG.
As shown in A, the first period in which the duration T of the silent section is set.
Is it less than the reference length T1 (T <T1) or less than the second reference length T2 (where T1 <T2) which is equal to or more than the first reference length T1 where the duration T of the silent section is set (T1 ≦ T <T2 ),
Alternatively, the second reference length T2 in which the duration T of the silent section is set
It may be determined whether or not (T ≧ T2). As the first reference length, for example, a length of 4 frames is
For example, a length of 40 frames is set as the second reference length.

Then, as shown in FIG. 8, it is possible to proceed to the following steps according to each determination result.
That is, when the duration T of the silent section is less than the set first reference length T1 (T <T1), the process proceeds to step 13. The first reference length T1 in which the duration T of the silent section is set
When the length is less than the second reference length T2 (T1 <T2) set as described above (T1 ≦ T <T2), the process proceeds to step 28 to perform thinning by the 1 / n thinning process. The second reference length T2 or more (T ≧ T
When it is 2), the process proceeds to step 15.

FIG. 10 shows the relationship between the input signal and the output signal at the time of double-speed reproduction, and particularly shows how the input signal in the silent section is deleted. 11 and 12 are data writing start points to the ring memory 7, data reading start points from the ring memory 7, and points A to H in FIG.
The state of the ring memory 7 in FIG.

In FIG. 10, since the input signal is in the silent section and the ring memory 7 is in the empty state at the start of the double speed reproduction (see FIG. 11A), the frame data is thinned out. The data is thinned out at a compression rate of 1/2 by 24 and then written in the ring memory 7.

When the accumulated amount Tm of the ring memory 7 reaches the underflow detection data Tmin, the reading of data from the ring memory 7 is started (FIG. 11).
(B)).

When frame data for the voice section a of the input signal is sent (point A), the frame data is compressed by the pitch compression / expansion means 23 at a compression rate of 2/3. The frame data is expanded on the basis of compression at a compression rate of 1/2 in which the lengths of the input signal and the output signal match. In this sense, the extension processing is described in FIG. Then, this compressed data is written in the ring memory 7. At the point A, as shown in FIG. 11C, the accumulated amount TmA remains Tmin.

Output signal a for the voice section a of the input signal
1 is read out with a delay of the accumulated amount TmA at the point A. Then, at the time when the voice section a of the input signal has been input (point B), as shown in FIG. 11D, the accumulated amount Tmin at the point A, which is the start point of the current compression section,
The sum of the compressed data of the voice section a from the point A to the point B and the expanded amount StB for the compression of the compression rate 1/2 becomes the storage amount TmB (= StB + Tmin) of the ring memory 7. Therefore, the output signal a1 for the voice section a of the input signal
Ends when TmB (= StB + Tmin) has elapsed from point B.

The frame data of the silent section less than the pause duration Tdel following the voice section a of the input signal is also compressed by the pitch compression / expansion means 23 at a compression ratio of ⅔. When the voice section b is input subsequently to the silent section, the frame data of the voice section b is also compressed by the pitch compression / expansion means 23 at a compression rate of 2/3.

Then, at the time when the voice section b of the input signal has been input (point C), as shown in FIG. 11E, the accumulated amount Tm at the point A, which is the start point of the current compression section.
The sum of in and the expanded amount StC of the compressed data corresponding to the input signals from the points A to C for 1/2 compression is the storage amount TmC (= StC + Tmin) of the ring memory 7. Therefore, the output signal b1 for the voice section b of the input signal ends being output at a point after TmC (= StC + Tmin) has elapsed from the point C.

When a signal of a silent section having a length equal to or longer than the pause duration Tdel is sent following the voice section b of the input signal, the pause duration Tdel is reached (D
The point data is compressed by the pitch compression / expansion means 23 at a compression rate of ⅔.

At point D, as shown in FIG. 11 (f), the accumulated amount Tmin at point A, which is the start point of the current compression section.
Then, the sum of the compressed data corresponding to the input signals from the points A to D and the decompressed amount StD for 1/2 compression is the storage amount TmD (= StD + Tmin) of the ring memory 7.
Therefore, the output signal for the silent section between the voice section b of the input signal and the point D is TmD (= StD +) from the point D.
The output ends when Tmin minutes have passed.

The frame data in the silent section after the pause duration Tdel is deleted by the input signal deleting section 25 until the accumulated amount in the ring memory 7 becomes the underflow detection data Tmin or less. The length Std of the pause deletion portion is from the point A, which is the start point of the current compression section, to the point D.
The compressed data corresponding to the input signal up to the point becomes equal to the expansion amount StD for 1/2 compression. Input signal deletion unit 2
After the deletion processing is performed by 5, the waveform synthesis insertion unit 22 inserts a synthesized waveform for preventing click sound, but the inserted synthesized waveform portion is omitted in FIG.

The end point (E
12 (g), the accumulated amount TmE of the ring memory 7 is less than or equal to the underflow detection data Tmin. Here, an example is shown in which the accumulated amount TmE is equal to the underflow detection data Tmin.

The frame data for the silent section from the point E is thinned out by the thinning processing section 24 at a compression rate of 1/2 and then written in the frame memory 7. And
When the signal of the voice section c is input (point F), the frame data of the voice section c is compressed by the pitch compression / expansion means 23 at a compression rate of 2/3. That is, a new compression section is started. Then, the compressed data is the ring memory 7
Is written to.

At point F, as shown in FIG. 12 (h), the accumulated amount TmF of the ring memory 7 is the same Tm as at point E.
It is in.

Output signal c for voice section c of input signal
1 is output with a delay of the accumulated amount Tmin at the point F. Pause duration Tde following the voice section c of the input signal
The frame data in the silent section less than 1 (the silent section from the voice section c to the point G) is also compressed by the pitch compression / expansion means 23 at a compression rate of 2/3.

At point G, as shown in FIG. 12 (i), the accumulated amount Tmin at point F, which is the start point of the current compression section.
Then, the sum of the compressed data corresponding to the input signals from the F point to the G point and the decompressed amount StG for 1/2 compression is the storage amount TmG (= StG + Tmin) of the ring memory 7.
Therefore, the output signal for the silent section from the voice section c of the input signal to the G point is TmG (= StG +) from the G point.
The output ends when Tmin minutes have passed.

The frame data in the silent section after the pause duration Tdel is deleted by the input signal deleting section 25 until the accumulated amount in the ring memory 7 reaches the underflow detection data Tmin. The length S of this pose deletion part
td becomes equal to the extension amount StG of the compressed data corresponding to the input signal from the point F to the point G, which is the start point of the current compression section, for 1/2 compression.

The final point (H
12 (j), the accumulated amount TmH of the ring memory 7 is equal to or less than the underflow detection data Tmin. Here, an example is shown in which the accumulated amount TmH is equal to the underflow detection data Tmin.

The frame data for the silent section from the point H is thinned out by the thinning processing section 24 at a compression rate of 1/2 and then written in the frame memory 7. And
When the signal of the voice section d is input, the frame data of the voice section d is compressed by the pitch compression / expansion means 23 at a compression rate of 2/3. Then, the expanded data is written in the ring memory 7.

FIG. 13 shows the relationship between the input signal and the output signal at the time of double speed reproduction, and particularly shows how the input signal is deleted when the state immediately before the overflow occurs.
FIG. 14 shows the ring memory 7 at points S to U in FIG.
Shows the state of.

The voice section a from a certain point to the point T,
Frame data for a series of input signals including b, c, etc. and a silent section is compressed by the pitch compression / expansion means 23 at a compression ratio of 2/3 (compressed for compression at a compression ratio of 1/2). And In this case, the ring memory 7
The amount of extension is accumulated in.

At the input start point (point S) of the voice section b, as shown in FIG. 14A, the accumulated amount Tmin at the start point of the compression processing of the series of input signals and the compression processing of the above compression processing. The sum of the compressed data corresponding to the input signal from the start point to the point S and the decompressed amount StS for 1/2 compression is the storage amount TmS (= StS + Tmin) of the ring memory 7.
Therefore, the output signal b1 for the voice section b is started to be output when TmS (= StS + Tmin) minutes have elapsed from the point S.

It is assumed that the ring memory 7 is in a state immediately before overflow at the time (point T) when the compressed data corresponding to the input signal of the voice section c is written in the ring memory 7. That is, it is assumed that the accumulated amount in the ring memory 7 becomes equal to or larger than the overflow detection data Tmax at the point T.

At the point T, as shown in FIG. 14B, the accumulated amount Tmin at the start point of the compression process for the series of input signals and the input signal from the compression process start point to the point T. The sum of the corresponding compressed data and the expansion amount StT for 1/2 compression is the storage amount TmT in the ring memory 7.
(= StT + Tmin). In other words, the total number of words in the ring memory 7 is set to TOTAL, the overflow detection data is set to Tmax, and TOTAL and Tmax are set.
If the difference between and is Dmin, the accumulated amount Tmt at the point T is
Since it is equal to Tmax, it becomes TOTAL-Dmin.

Therefore, the output signal corresponding to the series of input signals is stored at the point T from the accumulated amount TmT (= StT + Tmi).
The output is finished at the point of n) delay.

At the point T, when the ring memory 7 is in the state immediately before the overflow, the input signal thereafter is unconditionally deleted by the input signal deleting section 21 until the ring memory 7 is in the state immediately before the underflow. It
After the deletion processing is performed by the input signal deletion unit 21, the muffling insertion unit 22 inserts muffling, but the inserted muffling portion is omitted in FIG. 13. After the ring memory 7 is about to overflow (T
Point), the frame data is deleted, and FIG.
It is assumed that the ring memory 7 is in the state immediately before underflow (accumulation amount TmU = Tmin) at point U as shown in FIG. In this case, the input signal composed of four silent sections from point T to point U and three voice sections d, e, f is deleted. Therefore, the input signal from point T to point U does not appear as an output signal.

When the signal of the voice section g is input after point U, the frame data for this voice section is compressed by the pitch compression / expansion means 23 at a compression rate of ⅔ (for compression at a compression rate of ½). Are decompressed) and then written in the ring memory 7. The output signal g for the voice section g is started to be output with a delay by the storage amount Tmin of the ring memory 7 at the point U.

In the above embodiment, the voice section and the silent section of the input signal are discriminated based on the average power value P of each frame, but they may be discriminated based on the average amplitude of each frame. . In this case, as shown in FIG. 15, an average amplitude calculator 11A for calculating an average amplitude value in frame units is provided in place of the power calculator 11 of FIG. When the quantization bit number of the D conversion unit 2 is 12 bits, for example, the value 26
Threshold is set. Then, the average amplitude calculation unit 1
By comparing the average amplitude value calculated by 1A with the threshold value of the threshold value memory 13A by the comparison unit 12A, it is determined whether it is the voice section or the silent section.

That is, if the average amplitude value is greater than or equal to the threshold value, it is determined to be a voice section, and if the average amplitude value is less than the threshold value, it is determined to be a silent section. The average amplitude value W on a frame-by-frame basis is i0, i1, ... iN-1 (where N = 2) for the amplitude of each audio signal sampled in one frame.
00), it is calculated based on the following Equation 3.

[0154]

(Equation 3)

The other processing is the same as the processing by the speech speed conversion unit 6 in FIG. 2, and therefore its explanation is omitted.

Even in this case, the threshold value may be changed as follows. That is,
As shown by the dotted line in FIG. 15, an average amplitude steady state detecting and threshold updating unit 14A is provided. The average amplitude steady state detection / threshold updating unit 14A includes an average amplitude calculating unit 11
It is determined whether or not the average amplitude value W from A is constant over a predetermined number of frames, and when it is constant (steady state), a value twice the average amplitude value W at that time is set as the threshold memory 13A. To update the threshold. However, the maximum value of the updated threshold value is limited to a predetermined value, for example, 28 2.

Further, the voice section and the silent section of the input signal may be discriminated on the basis of the amplitude cumulative value Wa of the voice signal of each frame and a given threshold value shown in the following formula 4. .

[0158]

[Equation 4]

Also, the voice section and the silent section of the input signal are detected as the voice section if the periodicity of the signal of each frame is detected and the detected period is within a predetermined pitch period range of the voice signal. If it is determined that the detected period is outside the predetermined pitch period range of the audio signal, it may be determined to be a silent section.

In this case, as shown in FIG.
Instead of the power calculation unit 11 of FIG. 1, a pitch period detection unit 11B for detecting the periodicity for each frame based on the autocorrelation method is provided, and the pitch period range of the audio signal is set in the threshold value memory 13B. It Then, the cycle detected by the pitch cycle detection unit 11B and the threshold memory 13
The comparison unit 12B compares the pitch period range of the audio signal set to B.

The pitch period range of the audio signal to be set is
It depends on the reproduction speed, and is set to, for example, a range of 66 × n (Hz) to 320 × n (Hz) during n-fold speed reproduction. Therefore, during double speed reproduction, the pitch period range of the audio signal is set to the range of 132 Hz to 640 Hz. The other processing is the same as the processing by the speech speed conversion unit 6 in FIG. 2, and thus the description thereof will be omitted.

Alternatively, the voice section and the silent section of the input signal may be discriminated by comparing the power spectrum of the signal of each frame with the power spectrum of the steady state.

In this case, as shown in FIG.
In place of the power calculation unit 11 of, a power spectrum calculation unit 11C that calculates a power spectrum for a predetermined one or a plurality of frequency bands for each frame is provided.
In addition, the power spectrum in the steady state for the predetermined one or more frequency bands is stored in the power spectrum storage unit 1.
It is stored in 3C.

The contents of the power spectrum storage section 13C are as follows:
When the power spectrum steady state detection unit 14B detects that the power spectrum is in the steady state based on the change state of the power spectrum calculated by the power spectrum calculation unit 11C, the power spectrum in the detected steady state is updated.

The input signal is the power spectrum calculation unit 11C.
Then, the power spectrum for a predetermined one or a plurality of frequency bands is calculated for each frame.
Then, the calculated power spectrum and the steady-state power spectrum stored in the power spectrum storage unit 13C are compared by the comparison unit 12C.

If the calculated power spectrum fluctuates with respect to the steady-state power spectrum, the frame is discriminated as a voice section. On the contrary, if the calculated power spectrum does not fluctuate with respect to the power spectrum in the steady state, the frame is determined to be a silent section.

Specifically, the power spectrum storage unit 13
C is the predetermined one based on the steady-state power spectrum for the one or more predetermined frequency bands.
Alternatively, threshold values for a plurality of frequency bands are stored. Then, it is stored in the power spectrum storage unit 13C. By comparing the power spectrum for the predetermined one or a plurality of frequency bands calculated by the power spectrum calculation unit 11C with the corresponding threshold value stored in the power spectrum storage unit 13C, the input signal is converted into a voice signal. It is determined whether it is a section or a silent section.

For example, it is assumed that the power spectrum in the steady state is a noise-only power spectrum as shown in FIG. In addition, it is assumed that the power spectrum of voice that does not include noise is shown in FIG. In the steady state, the noise shown in the power spectrum of FIG.
When an audio signal having the power spectrum shown in 1 (b) is input, the power spectrum becomes a combination of the power spectra of both as shown in FIG. 21 (c).

Therefore, for example, the frequency band f in which the power is relatively small in the power spectrum in the steady state is
The powers for a and fb increase significantly in the power spectrum of the voice section. That is, by comparing the steady state power in one or more frequency bands in which the power is relatively small in the steady state power spectrum with the power in the one or more frequency bands of the power spectrum of the input signal, It is possible to determine whether is a voice section or a silent section.

When it is known that the noise in the steady state is noise in a high frequency band, a power spectrum for a low frequency band (for example, a frequency band of 4 KHz or less) that is less affected by noise is calculated, Depending on whether the calculated power spectrum is greater than or equal to a predetermined threshold,
It is also possible to determine whether the input signal is a voice section or a silent section.

When the voice section and the silent section are discriminated by comparing the power average value P of each frame with the threshold value Th, the threshold value is calculated based on the accumulated amount in the ring memory 7. You may make it change Th. That is, the threshold value T is set so that as the storage amount of the ring memory 7 decreases, in other words, as the empty area of the ring memory 7 increases, the missing parts of the voice section decrease.
h is reduced. As a result, the output voice is naturally closer to the sound.

That is, as shown in FIG. 22, threshold value adjusting means 51 is provided. The threshold value adjusting means 51 obtains the storage amount of the ring memory 7 from the ring memory storage amount state determination unit 16. Then, the storage time Tm is calculated by dividing the obtained storage amount of the ring memory 7 by the sampling frequency of the D / A conversion unit 8. Then, the threshold Th is determined based on the calculated accumulation time Tm, and the contents of the threshold memory 13 are updated.

More specifically, the storage amount of the ring memory 7 obtained from the ring memory storage amount state determination unit 16 is the sampling frequency of the D / A conversion unit 800.
By dividing by 0, the accumulation time Tm is obtained.
Then, the threshold Th for the accumulation time Tm is obtained based on the data of the threshold Th for the accumulation time Tm created in advance.

The following table shows an example of the data of the threshold Th with respect to the accumulation time Tm when the quantization bit number of the A / D converter 2 is 12 bits.

[0175]

[Table 1]

Further, when the voice section and the silent section are discriminated by comparing the power cumulative value Pa of each frame with the threshold value, the average amplitude value W of each frame is compared with the threshold value. When discriminating between the voice section and the silent section, the amplitude cumulative value Wa of each frame is compared with a threshold value, and the power threshold of each frame is compared with the threshold value to determine the voice section and the silent section. Also in the case of determining, the ring memory 7
You may make it change a threshold value based on the accumulation amount of.

Further, the pause duration Tdel for determining the deletion start point of the silent section may be changed based on the accumulated amount in the ring memory 7. That is,
The pause duration Tdel is made longer so that the amount of storage in the ring memory 7 becomes smaller, in other words, the more the empty area of the ring memory 7 becomes, the less the deleted portion of the silent section becomes. As a result, the output voice is naturally closer to the sound.

That is, as shown in FIG. 22, the pause duration adjusting means 52 is provided. Pose duration adjusting means 52
Obtains the storage amount of the ring memory 7 from the ring memory storage amount state determination unit 16. And the obtained ring memory 7
The storage time Tm is calculated by dividing the storage amount of 1 by the sampling frequency of the D / A conversion unit 8. Then, based on the calculated accumulation time Tm, the pause duration Tde
1 is determined, and the content of the pause duration setting memory 17 is updated.

More specifically, the storage amount of the ring memory 7 obtained from the ring memory storage amount state determination unit 16 is the sampling frequency of the D / A conversion unit 800.
By dividing by 0, the accumulation time Tm is obtained.
Then, the pause duration Tdel for the accumulation time Tm is obtained based on the data of the pause duration Tdel for the accumulation time Tm created in advance.

The following table shows an example of the data of the pause duration Tdel with respect to the accumulation time Tm during the double speed reproduction of the VTR.

[0181]

[Table 2]

Although the case where the input signal is an analog signal has been described above, the present invention can be applied to a case where the input signal is digital data. For example,
When a compressed digital audio signal is sent from an IC memory, a magnetic disk, a digital communication line, etc., the compressed digital audio signal is expanded and PC
The converted PCM audio signal is converted into an M audio signal, and the obtained PCM audio signal is temporarily stored in a buffer. After that, the PCM audio data is read from the buffer at a speed according to the set reproduction speed magnification and sent to the frame memory 5 in FIG.

The adaptive speech rate conversion processing has been described in detail above. Next, a concrete memory control operation for speech rate conversion by the simple thinning method will be described. That is, the speech speed conversion IC 112 controls the memory 113 as follows to convert the speech speed.

FIG. 28 shows the memory control operation during the 3 × speed reproduction. The audio signal is written at the 3 × speed, the reading is started at the same time as the writing, and the reading is performed so that the audio becomes the same speed as the 1 × speed reproduction. finish. That is, the memory is controlled so that data is written in a cycle T / 3, which is ⅓ of the read cycle T. Therefore, the T0 period in the figure is thinned out. Similarly, the writing time is T / 5 during 5 × speed reproduction, and T is during 9 × speed reproduction.
/ 9.

FIG. 29 shows the read / write timing of the memory during reverse 5 × speed reproduction, where T is 1
Double speed cycle, W is writing period, * 2 is T / 5, * is T /
It is 6. The timing of the write address of the memory is quintuple speed, the write cycle period is 5/6 of the quintuple-speed normal direction reproduction, and the write cycle counter, which is set to six T periods, circulates at five times, Writing is performed only when the value is 0. For this reason, as shown in the figure, the writing and the reading circulate with a slight shift, and the reading returns to the original after five readings, so that the reproduced sound can be heard smoothly without any change in the contents at the point α. As shown in the drawing, writing and reading are performed in the order of d and D, e and E, F1 for f1, and f2 for F2. Further, in FIGS. 28 and 29, the broken line shows the progress state of the write address, and the solid line shows the progress state of the read address.

[0186]

As described above, in the VTR of the present invention, the adaptive voice speed conversion process is performed at the time of double speed reproduction, and the simple thinning process is performed at the time of triple speed reproduction or more, thereby converting the voice speed. Not only is the structure of the IC for use simplified, but an appropriate speech rate conversion process is automatically selected, so that the commercial value of the VTR can be improved. Further, according to the present invention, it is possible to reduce the processing load at the time of double speed reproduction, reduce the deviation between the video and the audio, and obtain the advantage that the memory capacity for accumulating the audio signal does not become enormous.

[Brief description of drawings]

FIG. 1 is a block diagram showing an overall configuration of a speech speed conversion device.

FIG. 2 is a block diagram showing a configuration of a speech speed conversion unit.

FIG. 3 shows a compression ratio of an input signal of 2 / using PICOLA.
6 is an explanatory diagram showing a method of compression in FIG.

FIG. 4 is an explanatory diagram illustrating a process performed by a waveform synthesis processing unit.

FIG. 5 is an explanatory diagram for explaining various thinning-out processing methods performed by a thinning-out processing unit.

FIG. 6 is a flowchart showing a processing procedure by a speech speed conversion unit.

FIG. 7 is a flowchart showing a processing procedure by a speech speed conversion unit.

8 is a flowchart corresponding to FIG. 7, showing a modification of the processing procedure by the speech speed conversion unit.

9 is an explanatory diagram for explaining a process that can be replaced with the process of step 10 of FIG.

FIG. 10 is a time chart showing a relationship between an input signal and an output signal at the time of double speed reproduction, in particular, a state in which the input signal in a silent section is deleted.

FIG. 11 is a data writing start point to the ring memory 7,
FIG. 11 is a schematic diagram showing a state of the ring memory 7 at a data read start point from the ring memory 7 and points A to D in FIG. 10.

12 is a schematic diagram showing a state of the ring memory 7 at points E to H in FIG.

FIG. 13 shows a relationship between an input signal and an output signal at the time of double speed reproduction, particularly when a state immediately before overflow occurs,
It is a time chart which shows a mode that an input signal is deleted.

FIG. 14 is a ring memory 7 at points S to U in FIG.
It is a schematic diagram which shows the state of.

FIG. 15 is a block diagram corresponding to FIG. 2, showing a modified example of a circuit for discriminating between a voice section and a silent section.

16 is a block diagram corresponding to FIG. 2, showing another modified example of the circuit for discriminating between the voice section and the silent section.

FIG. 17 shows a compression rate of 2 / for an input signal in units of fixed frames.
6 is an explanatory diagram showing a method of compression in FIG.

FIG. 18 is an explanatory diagram for explaining a process that can be replaced with the process of step 9 of FIG.

FIG. 19 is an explanatory diagram for explaining a process that can be replaced with the process of step 10 of FIG. 6 when the process of FIG. 18 is adopted as the process of step 9 of FIG.

20 is a block diagram corresponding to FIG. 2, showing still another modified example of the circuit for discriminating between the voice section and the silent section.

FIG. 21 is a graph showing a steady-state power spectrum, a power spectrum of noise-free speech, and a power spectrum of a speech section.

FIG. 22 is a block diagram showing a speech speed conversion unit to which threshold value adjusting means and pause duration adjusting means are added.

FIG. 23 is a circuit block diagram of an essential part of a video tape recorder embodying the present invention.

24 is a diagram showing a flowchart for explaining the operation of the circuit block diagram of FIG. 23;

FIG. 25 is a diagram for explaining the concept of adaptive speech rate conversion processing.

FIG. 26 is a diagram for explaining the concept of speech speed conversion by simple thinning processing.

[Fig. 27] Fig. 27 is a diagram for describing the concept of voice speed conversion by simple thinning processing during reverse reproduction.

FIG. 28 is a diagram for explaining a memory control method for realizing a simple thinning process.

FIG. 29 is a diagram for explaining a memory control method for realizing a simple thinning process during reverse playback.

[Explanation of symbols]

 2 A / D conversion section 4 DSP 5 Frame memory 6 Speech rate conversion section 7 Ring memory 8 D / A conversion section 9 Up-down counter 11 Power calculation section 11A Average amplitude calculation section 11B Pitch cycle detection section 11C Power spectrum calculation section 12, 12A, 12B, 12C Comparing unit 15 Conditional branching unit 16 Ring memory accumulated amount state determining unit 21, 25 Input signal deleting unit 23 Pitch compression / expansion unit 24 Decimation processing unit 51 Threshold adjusting unit 52 Pause duration adjusting unit 112 Speech speed Conversion IC 114 Microcomputer

─────────────────────────────────────────────────── ─── Continuation of the front page (51) Int.Cl. ⁶ Identification number Office reference number FI technical display location H04N 5/93 H04N 5/93 G

Claims (5)

[Claims] 1. At the time of double speed reproduction, a voice speed conversion for performing compression / expansion processing or deletion processing on an input audio signal is performed depending on whether the reproduced audio signal is a voice section or a silent section. A double speed audio reproduction mode and an N times speed reproduction mode in which, during ± N times speed reproduction (N: a natural number of 3 or more), the audio section of a reproduction audio signal for a predetermined period is thinned according to the reproduction speed. A video tape recorder comprising control means for setting. 2. A voice speed conversion processing means for converting a voice speed of an input voice signal, a ring memory in which an output of the voice speed conversion processing means is written, and a ring memory so as to perform a double speed voice reproduction mode. The voice speed conversion processing means compresses and expands the input voice signal according to whether the input voice signal is a voice section or a silent section and the amount of storage in the ring memory. A video tape recorder having a speech speed conversion device having means for performing processing or deletion processing. 3. The A / D conversion means and the A / D conversion means for sampling the input analog audio signal at a sampling frequency according to a set reproduction speed magnification in order to perform a double speed audio reproduction mode. A frame memory to which the voice signal output from the means is input, and a voice speed conversion processing means for performing a voice speed conversion process on the voice signal every time a required number of voice signals are input to the frame memory,
A ring memory to which the output of the speech speed conversion processing means is written,
The reading means for reading data from the ring memory at a constant speed, and the storage amount calculating means for calculating the storage amount of the ring memory based on the write signal and the read signal of the ring memory are provided. According to the section discriminating means for discriminating whether the input voice corresponding to the required number of voice signals input to the frame memory is the voice section or the silent section, and the output of the section discriminating means and the accumulated amount calculating means, A video tape recorder having a speech speed conversion device equipped with signal processing means for performing compression / expansion processing or deletion processing on a required number of audio signals. 4. A frame memory in which an input digital audio signal is written at a speed corresponding to a set reproduction speed multiplication rate in order to perform a double speed audio reproduction mode, and a required number of audio signals are stored in the frame memory. Each time a signal is input, a voice speed conversion processing unit for performing a voice speed conversion process on those audio signals, a ring memory to which the output of the voice speed conversion processing unit is written, and a writing to a frame memory at the time of 1 × speed reproduction A read means for reading data from the ring memory based on a read signal having a frequency equal to the speed and a storage amount calculation means for calculating the storage amount of the ring memory based on the write signal and the read signal of the ring memory are provided. The speech speed conversion processing means is such that the input voice corresponding to the required number of voice signals input to the frame memory is a voice section or a silent section. And a signal processing means for performing compression / expansion processing or deletion processing on the required number of audio signals in accordance with the output of the section determination means and the output of the accumulated amount calculation means. A video tape recorder having a speech speed conversion device. 5. The memory control means according to claim 1, further comprising a memory control means for controlling the memory so that the audio data is written in the memory at N times speed and the written data is read out at 1 times speed in order to perform the N times speed reproduction mode. A video tape recorder that is characterized.