JPS60256987A - Time axis converter of acoustic signal - Google Patents

Abstract

PURPOSE:To make the audible sense constant for at both compression and expansion of a time axis and to change the scale with a low distortion factor in a wide frequency range by coding/decoding an acoustic signal with PCM and chancing the block length with a set pitch ratio. CONSTITUTION:An input acoustic signal at a terminal 1 is sampled at a prescribed period, coded and stored to a RAM7 at each block. The length of a read block from the RAM7 at time axis conversion depends on the set time of a timer circuit 10 and this set time is controlled in response to the pitch ratio between the original tone and the reproduced tone and the degree of compression or expansion of the signal by the time axis conversion with the operation of a control circuit 11. In such a case, since a pitch setting device 12 changes the pitch of the reproduced sound high or low to the input sound, the pitch ratio of the original tone to the reproduced tone is set, then the read speed of the coded data from the RAM7 is adjusted and the scale is changed without changing the length of the original tone.

Description

[Detailed description of the invention] [Industrial field of application] The present invention inputs an i-blue signal such as a musical tone signal, performs processing such as encoding using PCM (pulse coding modulation), and compresses the time axis of the signal. The present invention also relates to an audio signal time axis conversion device that performs encoded audio signal processing that changes the pitch of a musical tone or sound by outputting an expanded audio signal.

[Prior art] As an audio device that processes musical tone signals and audio signals, an audio signal (original sound) is input, converted into a digital signal through processing such as PCM encoding, and this digital audio signal is divided into blocks and - When writing these signals to the H memory, compressing or expanding the time axis of each block, and reading them out, the associated blocks are repeatedly read out or interlaced read out according to a predetermined procedure, and the input acoustic signals are Development of an audio signal time axis conversion device using encoded audio signal processing that changes the pitch without changing the length of the sound in almost real time is underway.

However, in this type of time axis conversion device, the block length of the audio signal is fixed and written in a storage means such as a memory, and the block is compressed and expanded when reading out, so for example, the length of the input sound When processing to raise the pitch without changing the pitch, the acoustic signal is read out by compressing the time of each block, and blocks with strong correlations are repeatedly read out to fill in the gaps in the resulting signals. . Therefore, in order to increase the pitch without changing the length of the acoustic signal, the number of cut points increases with the increase in the number of blocks in the output signal, so that when the output signal is decoded, The noise becomes noticeable, and there is a risk that the quality of the decoded audio signal, that is, the reproduced sound, will deteriorate. On the other hand, when processing to lower the pitch without changing the length of the input sound, the time of each block is expanded and the acoustic signal is read out from memory, and in order to compensate for the lack of time, it is necessary to Blocks are skipped, but if the length of the output audio signal read out is considerably redundant compared to the signal block before being written to the memory, that is, the input audio signal, the length of the output audio signal that has been read out is considerably redundant. A problem arises in that the reproduced sound is reproduced with a delay.

[Object of the Invention] The present invention has been made in view of the above points, and when processing to change the pitch without changing the length of the sound, that is, to perform time axis conversion of an acoustic signal, the present invention has been made in view of the above points. During readout, the number of division points of the readout block is kept as constant as possible to maintain a constant hearing sensation when compressing and expanding the time axis, and to suppress the occurrence of noise at the block division points when compressing the time axis. This prevents deterioration of the reproduced sound, and also eliminates the problem of delay in the reproduced sound due to redundant playback blocks when the time axis is expanded. It is an object of the present invention to provide a time axis converting device for an acoustic signal that can change the pitch.

[Structure of the Invention] The structure of the present invention, which has been made to achieve the above object, is as shown in FIG. Encoded information writing means that divides a digital signal into predetermined block lengths by marking them to mean cutting points, and sequentially writes them into a readable/writable storage means B as encoded information along a predetermined time axis. C; pitch ratio setting means for setting a pitch ratio for changing the pre-encoded analog audio signal to a desired pitch and outputting a pitch ratio signal; and determining the pitch from the encoded information in the storage means B. encoded signal succession means E for converting the time axis according to a predetermined procedure according to the pitch ratio signal of the ratio setting means and reading out the encoded signal for each block divided by the mark attached to the hunting information; , according to the read encoded signal,
PCM decoding means F for decoding analog audio signals
In the audio signal time axis conversion device, the encoded information writing means C writes a block length of encoded information into the storage means B based on the pitch ratio signal from the pitch ratio setting means. The gist of the present invention is an acoustic signal time axis conversion device characterized in that it is configured to change the time axis of an acoustic signal.

Embodiments of the present invention will be described below based on the drawings.

[Example] Figure 2 shows an example in which an analog audio signal such as a musical tone or voice is input.
In addition to PCM encoding processing, the pitch of the reproduced sound of the decoded audio signal is subtly changed by performing time axis conversion that involves compression and expansion of the encoded signal without changing the length of the sound, that is, the playback speed. 1 shows a block diagram of an acoustic signal time axis conversion device.

Reference numeral 2 denotes an input filter that inputs and filters an analog audio signal such as a voice or a musical tone signal input from the input terminal 1, and sets a band by removing unnecessary frequency components and noise for audio signal processing.

A compressor 3 compresses the amplitude of the input signal sent from the input filter 2, compressing the signal amplitude for noise reduction operation, and amplifying the signal in an expander of the decoding circuit described later. 4 is a sample/hold circuit that samples the analog audio signal sent from the compressor 3;
Instantaneous amplitude values of a continuous acoustic signal waveform are taken out as sample values. 5 is sample/hold circuit 4
This is an A/D converter that quantizes the sampled signal and converts it into binary coded data. In this embodiment, the above filter 2 to A/D converter 5 constitute PCM encoding means. 6 stores encoded encoded information (data) sent from the A/D converter 5 into a register provided in a read/write memory (RAM) 7 and having a storage capacity for several blocks of audio signals. This is a write latch that latches for sequential storage, and the write/continuation controller 8
Temporarily stores data using a latch signal supplied from
When a write permission signal is sent to the RAM 7 from a multiplexer, which will be described later, encoded data is sent to the RAM 7 through the data bus, and the encoded data is sequentially written to designated addresses in the RAM 7.

9 specifies, in the least significant bit (LSB), the cutting point of the block read from RAM 7 when performing time axis conversion by compression/expansion of the signal when writing encoded data to RAM 7. This is a marking circuit that attaches a mark (for example, a code of logic "1"). This marking circuit 9 is connected to an A/D converter 5 which converts analog acoustic signals into
The most significant bit (MSB) of encoded data converted into a digital signal is input, and when the sign changes from O to 1, this is detected as a zero cross point during negative going and a marking command signal is output. The detection unit receives this marking command signal, and the A/
A marking section is provided for adding a mark (for example, a code of logic "1") indicating a block division point to, for example, the least significant bit (LSB) of the encoded data output from the D converter 5.

Reference numeral 10 denotes a timer circuit that masks the marking command signal output from the zero-cross detection section of the marking circuit 9 for a preset time and stops outputting the signal. A high-level masking signal is output to the zero-cross detection section for a predetermined masking time from the rising edge. The timer circuit 10 is provided with a second timer unit that operates when a silent input signal is input, and when the silent input signal continues, that is, for a long period of time when the marking at the cutting point specifying the block circulates through the memory. In order to eliminate cases where the marking is not performed, if the marking command signal is not output from the zero-cross detection section of the marking circuit 9 even after the time set in the second timer section has elapsed, marking is forcibly applied to the marking section of the marking circuit 9. It is configured to output a command signal.

The above-mentioned write latch 6, a part of the write/read controller 8, the marking circuit 9, the timer circuit 10, etc. constitute encoded information writing means, and the digital signal converted by the A/D converter 5 is The above marks are applied to the data to divide it into predetermined block lengths, and the data is written to the storage means (in this case, the register of RAM 7) as encoded information along a predetermined time axis.In other words, the data input from terminal 1 The analog audio signal is sampled at a predetermined period, quantized, encoded, and stored in a register in the RAM 7 block by block.

Here, the length of the block read from the RAM 7 when performing time axis conversion is determined by the time interval of the marks indicating the cut points, that is, the set time of the timer circuit 10, and the set time of this timer circuit is By the operation of the control circuit 11, the degree of compression or expansion of the signal due to time axis conversion,
In other words, it is controlled according to the pitch ratio between the original sound and the reproduced sound.

In other words, if the length of the block during normal writing is constant, when time axis conversion is performed with the length of the reproduced sound constant, the block length of the read signal will change depending on the degree of compression or expansion of the signal. The number of cutting points changes greatly, but this control circuit 11 changes the length of the block at the time of writing so as to reduce the change in the number of cutting points in response to changes in pitch ratio. control. Therefore, the control circuit 11 detects the ratio between the period of the read clock signal i output from the variable clock signal generator 13 and the period of the write clock signal, using the pitch ratio signal 10 from the variable clock signal generator 13, When the ratio of the write/read cycles, that is, the pitch ratio between the original sound and the reproduced sound set by the pitch setting device 12, which is the pitch setting means, is large, a timer circuit is used to compress the time axis according to the degree. 10 (Tm in FIG. 4) is lengthened, and conversely, when the set pitch ratio is small, the timer set time Tl11@ is shortened accordingly. This control circuit 11 is actually a write/read controller 8. It is constituted by one gate array together with a marking circuit 9, a timer circuit 10, etc.

The pitch setting device 12 is used to set the pitch ratio between the original sound and the reproduced sound in order to change the pitch of the reproduced sound to be higher or lower than the input sound, and its operation changes the reading speed of encoded data from the RAM 7. You can adjust the pitch to change the pitch without changing the length of the original sound. This pitch setter 12 is configured as a variable resistor connected to the variable clock signal generator 13 so as to change the period of the read clock signal i applied to the read counters A14a and B14b.

15 is a clock signal generator that outputs a clock signal to the write/read controller 8 and the write counter 16;
A frequency-divided write clock signal for addressing the RAM 7 is sent to the write counter 16 . On the other hand, in the write/continuation controller 8, the clock signal generator 15
From the clock signal sent from the sample pulse signal a
, memory write enable signal d applied to A/D converter reference clock signal layer 1 write latch signal @cAM7 for A/D converter 5, memory read enable signal e1 read latch signal f1. f2, and a selection pulse signal q sent to multiplexer 1A that switches between write and read operations of RAM7.
are created respectively.

The timing of writing and reading encoded data into and from the RAM 7 is controlled so as to be performed continuously, for example, during one period of the write clock signal. The write counter 16 is a ring counter that indicates 4m corresponding to the address of the encoded data area in the RAM 7, inputs a clock signal, counts, and sends the write address to the RAM 7 via the multiplexer 17. Therefore, when writing encoded data to the RAM 7, the encoded data is first temporarily stored in the write latch 6 by the write latch signal C sent from the write/read controller 8 to the write latch 6, and then transferred to the data bus. After being set, the write enable signal d to the RAM 7 is activated, and while the multiplexer 17 is switched to the write side, the write counter 16
Encoded data is sequentially written to the addresses specified by .

Reference numeral 18 denotes a series controller for controlling the latch C19 and the latch D20, which is composed of a circuit having 7 lips and a gate of the type shown in FIG. 3, and inputs the marking signal Ma from the marking circuit 9. A latchable signal 0.0 synchronized with this marking signal Ma. q and strobe signals r and p are sent to the latch C19 and the latch D20, and the latch 019 and the latch D20 alternately temporarily store the address information sent from the write counter 16 at the timing of the block division point indicated by the marking signal Ma. go.

21 is a mark detection circuit, which is sequentially written in the RAM 7 (detects the block division point force mark signal m attached to the encoded data, and also detects the block division point force mark signal m attached to the encoded data, as shown in the timing chart of FIG. 5). At the time of reading, a mark in the code/naturalized data is detected according to the reading speed and a mark detection signal De is output, and when two mark detection signals De are generated between two mark signals m, the second mark is detected. A load signal n is generated in synchronization with the detection signal @DC.

And this load signal n is every other load signal j1
and load signal j2, and the OR gates 21a and 21
The load signals j1 . j2 presets the address data that will be the start of reading by latch C19 or latch D20. The reason why such two reading counters A1 days are used and the encoded data is read out in two systems is that when the encoded data is skipped or repeatedly read out at the block division point of the signal. This is to reduce noise caused by waveform deviations due to differences in signal pitch and amplitude, and the envelope circuit 22a,
The mutually inverted envelope signals EnA and EnB output from the readout latch A22b are input to the encoded data from the readout latch A23a and the readout latch B23b, respectively.
applied to the reference inputs of two D/A converters 24a and 24b, and each block is not immediately connected;
When the previous data block passes the cutoff point, its output is gradually reduced, and the output of the new data block is gradually increased from the cutoff point, thereby connecting signals at the block cutoff point.

The above-mentioned 23a and 23b are read latches of two systems, respectively, and the read latch signal f1 or f2 from the write/read controller 8 is sequentially input to the multiplexer 17 in synchronization with the read selection of the selection pulse signal g. When switched to the read side, it latches the encoded data addressed by the read counter A14a or B14b and read from the RAM 7 onto the data bus.

The above read counters A14a, B14b, multiplexer 17, successive controller 18, latch 19.20
, mark detection circuit 21 and readout latch 23a, 23
b constitutes encoded signal successive output means together with a part of the write/read controller 8.

24a and 24b are D for inputting the encoded signals latched by read latches A and B and converting them into analog signals.
The signal output from the D/A converters 24a and 24b is sent to an integrator 25 for integration, and then an expander is used to restore the signal amplitude compressed by the compressor 3 at the input. 26, further sent to an output filter 27, and is configured so that an analog music cost signal whose time axis has been converted and whose pitch has been changed is obtained from an output terminal 28. The above D/A converters 24a and 24b.

The integrator 25, expander 26, and output filter 27 correspond to PCM decoding means.

Although not shown in the block diagram of Figure 2, it is not only possible to input the original sound such as a musical tone signal and simply change the pitch by time axis conversion processing and output it, but also to provide a mixer circuit on the output side to convert the original sound signal. By mixing and outputting a time-axis converted reproduced sound signal with a specific pitch ratio that forms a consonant interval with respect to the original sound, for example, a consonant sound with a harmonic third or fifth degree can be produced from a single sound source. can also be generated.

In addition, by simply regenerating the digital signal encoded and stored in the RAMT by a predetermined period of time, and mixing this with the original sound signal and outputting it, it is converted into the original analog sound signal. It is also easy to apply an echo.

Next, the operation of the FR-to-FR axis conversion device will be explained with reference to the waveform diagrams and timing charts of FIGS. 4 to 8. Here, FIG. 4 is a timing chart showing the marking operation of adding a mark to the encoded signal and setting the block length.
The figure shows a graph showing the process of compressing an audio signal as an example of time axis conversion of the audio signal and the process of repeatedly reading out some of the ticks in the encoded signal, and a timing chart of each signal showing the operation. 6 is a timing chart of each signal (a to h) showing the operation of writing and reading encoded data to the RAM 7, and FIG. 7 is a process when expanding an audio signal. A graph showing the process of skipping some blocks of the encoded signal accompanying this, and each signal (De to j2) showing the operation.
An explanatory diagram showing correspondence with the timing chart of
The figure is a diagram schematically showing the envelope of the input waveform of the integrator 25. Each figure will be appropriately cited in the following description to further clarify the operation of the acoustic signal time axis converting device according to the embodiment of the present invention.

When using an audio signal time axis conversion device, first, the pitch setting device 12 sets the pitch ratio (tone interval ratio) between the input original sound and the output reproduced sound. For example, if the pitch ratio is set to 1:2, this means that the frequency of the reproduced sound is converted to twice that of the original sound.For example, if the input musical sound signal is the sound of "C", the reproduced sound is a "Do J" sound, which is an octave higher. By setting this pitch ratio, the period of the clock signal of the variable clock signal generator 13 is changed from the fixed period of the clock signal generator 15 sent to the writing side circuit. It is adjusted in inverse proportion to the pitch ratio with the period of the clock signal as a reference, and the read speed increases in proportion to the pitch ratio with the signal writing speed as the reference.

When the pitch ratio is set and the device starts operating, the control circuit 11 outputs a fixed write clock signal output from the clock signal generator 15 and a continuous write clock signal output from the variable clock signal generator 13. Then, the period ratio of the clock signal i is detected by the pitch ratio signal 10, and based on the reciprocal of this period ratio, that is, the pitch ratio, if the pitch ratio is 1 or more, that is, when the signal is compressed, the timer circuit is activated according to the degree. The setting time ('Tm) of the timer circuit 10 is increased, and when the signal is expanded with a pitch ratio of 1 or less, the setting time ('Tm) of the timer circuit 10 is shortened accordingly. . By controlling the set time of this timer circuit 10,
Regardless of compression or decompression processing, the block length of the read signal is almost constant, and this may cause deterioration in sound quality due to an increase in block separation points during playback that occurs when compression processing is performed with a large pitch ratio, or when the pitch ratio is reduced. It is possible to eliminate delays in reproduced sound that occur during decompression processing.

An acoustic signal such as a musical tone signal is input from an input terminal 1 , passes through an input filter 2 to remove frequency components and noise unnecessary for signal processing from the signal, and is sent to a compressor 3 . The compressor 3 compresses the amplitude of the acoustic signal for noise reduction, etc. The amplitude-compressed analog acoustic signal is sent to the PCM encoding processing circuit, and first,
The signal is input to the sample/hold circuit 4 and sampling is performed. Sampling is done by write/read controller 8
This is performed at the time interval of the sampling pulse signal a sent from the sampling pulse signal a, and each amplitude value is held as a sample value. The signal sampled by the sample-and-hold circuit 4 is then sent to the A/D converter 5, where it is encoded into binary numbers as quantized signal data. The encoded data of the acoustic signal converted into PCM code by the A/D converter 5 is then transferred to the write latch 6.
and is latched by a latch signal C supplied from the write/read controller 8. The encoded data latched by the write latch 6 is sequentially sent to the RAM 7, and is stored in the RAM 7 at the encoding period while being addressed based on the address signal supplied from the write counter 16 via the multiplexer 17. will be written in sync with.

On the other hand, encoded data is transferred from the A/D converter 5 to the write latch 6.
When the encoded data is temporarily stored, a mark indicating the block division point of the signal is placed in the least significant bit of the encoded data at intervals based on the set time of the timer circuit controlled by the control circuit 11, and the acoustic signal is It is given as follows at the zero cross position of .

That is, as shown in FIG. 4, the acoustic signal S input from the input terminal 1 is converted into coded data by the A/D converter 5, but it is not binary coded data using complement representation. The most significant bit (MSB>) represents the positive or negative sign of the value of the encoded data. Therefore, when the most significant bit (MSB) of the encoded data changes from O to 1, the marking section of the marking circuit is set to the negative sign. A marking command signal U is generated as a going zero-crossing point, and a mark indicating a block division point is attached to, for example, the least significant bit of the encoded data latched by the write latch 6, as a code of logic "1", for example. .

Note that if the silent signal continues, the zero-crossing point of the acoustic signal cannot be detected and marking cannot be performed.
In such a case, the second timer section in the timer circuit 10 outputs the output signal to the zero cross detection section after a set delay time from the falling edge of the mask signal V of the timer circuit 10, and marks the signal unconditionally. A command signal is generated, and a mark indicating a block cutting point is added to a part of the encoded data in the same manner as above. Of course, the set delay time of the second timer section is also controlled by the control circuit 11 in the same way as the mask time of the timer circuit 10.

On the other hand, the encoded data stored in the RAM 7 is read out in real time according to a predetermined procedure while compressing or expanding the signal and at a reading speed based on the pitch ratio set by the pitch setting device 12 by the operation of the reading side circuit. The data is read out from the RAM 7 as follows.

First, the mark detection circuit 21 detects the mark signal m at the cutoff point attached to the encoded data that is sequentially written into the RAM 7, and the mark signal m of the encoded data is detected as shown in the timing chart of FIG. Mark detection signal De detected according to reading speed ° (slope of graph line B)
is detected and two mark detection signals De are generated between the two mark signals m, a load signal n is generated in synchronization with the second mark detection signal De. Then, every other load signal n is divided into a load signal j1 and a load signal j2, which are sent to two read counters A14a and B14 via OR gates 21a and 21b, respectively.
The read counters A14a and B14b are latched C19 and D20 by the load signals j1 and j2.
Presets the address data that is latched at the beginning of the read. At this time, as shown in the timing chart of FIG. 6, when the multiplexer 17 is switched to the read side by the selection pulse signal 9 sent from the write/read controller 8 to the multiplexer 17,
A read enable signal e to the RAM 7 is activated, and encoded data is sequentially read from the address specified by the read counter 14a1 or 14b.

At this time, graph line A indicating the data writing speed
When the slope of the graph line B indicating the reading speed is large with respect to the slope of Blocks are read repeatedly to prevent read operations from overtaking write operations.

On the other hand, when the signal is expanded by setting the pitch ratio to 1 or less, as shown in the graph of FIG.
If there is, a load signal n is generated in the mark detection circuit 21 in synchronization with the second mark detection signal, and this is divided into two read signals J11J2 and sent to the read counter 14a.

14b, blocks are skipped along a graph line with a lower read speed than the graph line C indicating the write speed.

Then, the encoded data output from the RAM 7 is latched into read latches A23a and B23b, and two D/D/
The signals are sent to A converters 24a and '24b. D/A converter 2
At 4a and 24b, the read encoded signal is converted into an analog signal, and the waveform of the analog signal output from each D/A* converter 24a and 24b is determined by the envelope signals EnA and EnB to which the reference input is applied. The amplitude is adjusted so that the peak, that is, the envelope, has a smooth slope. Then, the two output signals are combined and sent to an integrator 25 to be decoded into a complete analog signal, and then returned to the original signal amplitude by an expander 26, passed through an output filter 27, and is output from an output terminal 28. An acoustic signal whose pitch has been raised by signal compression processing of time axis conversion is output.

To summarize the main points of the above explanation, in this embodiment, when the time axis is compressed by 172 times, that is, when the pitch ratio is set to 2, the length of one block during encoding is, for example, approximately 60 m5ec.
The block length during decoding is set to approximately 30m5ec, while the time axis is doubled, that is, the pitch ratio is set to 17.
2, set the block length during encoding to about 1, for example.
The block length when decoding is approximately 30 m5e.
ci.

Therefore, even when an acoustic signal is compressed and expanded, the block length at the time of decoding is controlled to approximately a constant length, so that the hearing sensation is approximately constant, and the noise at the block separation point is suppressed at a short time on the time axis. This also solves the problem that when the occurrence of deterioration of the reproduced sound is suppressed and the time axis is extended, the reproduction block becomes redundant and the reproduced sound is delayed.

In addition, since the acoustic signal is encoded by PCM, the acoustic signal level at the block joint during repeated block reading and reading skipping during decoding is almost zero, so other encoding When compared to coding methods such as DPCM and 6M that take the difference between audio signals, the seam is located at the peak value of the audio signal.
It is possible to perform time axis transformation of an acoustic signal with a low distortion rate over a wider frequency band. For example, in audio signal time axis converters that have the same configuration except for whether the encoding means uses PCM or DPCM, the frequency band can be down by 4 dB in the DPCM method. Even 40H2~8K)-jZ
However, in the PCM system, even a 3 dB reduction covers the band from 20H2 to 8°5KH2. Also, the total harmonic distortion rate is 1.7% in the DPCM method.
However, in the PCM method, it was less than 0.5%. This is because in the DPCM method, although the peak value of the acoustic signal is not necessarily the same, the rate of change of the acoustic signal is zero and the signals can be connected most smoothly. This is thought to be because the signals are connected at irregular peak points, and the difference between them causes signal distortion, which limits the frequency band.

This problem was solved by adopting the PCM method.

Also, when comparing the PCM method with the 6M and DPCM methods,
Since no differences are taken, the sharp attack of the acoustic signal can be reproduced more faithfully, and there is also the advantage that there is less delay in the auditory sense of sound.

Furthermore, since two A/D converters are used in this embodiment, the envelope of the signal sent to the integrator 25 during decoding is determined by the D/A conversion, as schematically shown in FIG. The D/A converter 24a outputs a signal as shown by I>, and the D/A converter 24b outputs a signal as shown by
) is output, and the input signal to the integrated integrator 25 has a shape as shown in (III). Therefore, the seams between blocks during decoding are eliminated so that they are not noticeable to the listener.

In this embodiment, the write/read controller 8.
Marking circuit 9. Although the timer circuit 10 and control circuit 11 are constructed from a single gate array along with other peripheral gates, etc. in order to increase the III control speed, they can also easily be constructed from a microcomputer that performs processing at a sufficiently high speed. .

Although the embodiments of the present invention have been described above, the present invention is not limited to these embodiments in any way, and it goes without saying that it can be implemented in various forms without departing from the gist of the present invention. It is.

[Effect of the invention] As detailed above, PCM encodes and encodes acoustic signals.
According to the audio signal time axis conversion device of the present invention, which is configured to decode and change the block length according to a set pitch ratio, the pitch of the input original sound can be slightly changed without changing the length of the sound. When performing time axis conversion, even if the decoding period is shortened to compress the signal, the block length during writing is controlled to be long, so the number of cutting points for each block increases. Therefore, it is possible to prevent noise and deterioration of reproduced sound due to an increase in the number of cutting points. In addition, even when the decoding cycle is lengthened to expand the signal, the block length at the time of writing is controlled to be short, so that the next playback sound after a silent state is played back without delay. This makes it possible to maintain a constant sense of hearing at all times. In addition, since the acoustic signal is encoded and decoded using PCM, the attack can be faithfully reproduced, and the time axis and conversion of the acoustic signal can be performed at a low (X distortion rate) over a wide frequency band. (The effect of 1 is also Qin.

[Brief explanation of the drawing]

Fig. 1 is a basic configuration diagram of the present invention, Fig. 2 is a block diagram of an audio signal time axis conversion device as an embodiment, Fig. 3 is a circuit diagram of the readout controller 18, and Fig. 4 is a mark on an encoded signal. A timing chart showing the marking operation for setting the block length by adding . Figure 5 shows the process of compressing an audio signal as an example of time axis conversion of an audio signal, and some blocks of the encoded signal associated with this process. An explanatory diagram showing a graph showing the repeated read processing and a timing chart of each signal showing the operation, and FIG. 6 shows the timing of each signal showing the operation of writing and reading encoded data to the RAM 7. The chart in FIG. 7 shows a graph showing the process of expanding the audio signal and the accompanying process of skipping some blocks of the encoded signal, and a timing chart of each signal showing the operation. An explanatory diagram, FIG. 8 is a diagram schematically showing the envelope of the input waveform of the integrator 25,
It is. 4...Sample/hold circuit 5...A/D converter 6...Write latch 8...RAM 9...Marking circuit 10...Timer circuit 11...Control circuit 12... Pitch setting device 16...Writing counters 14a, 14b...Reading counters 24a, 24b-' D/A converter agent Patent attorney Tsutomu Adachi and one other person Figure 3 I8 Figure 5 Figure 6 @ Figure 7 figure

Claims (1)

[Claims] A PCM encoding means for inputting an analog audio signal and encoding it into a digital signal using PCM; encoded information writing means for sequentially writing encoded information along a predetermined time axis into a readable/writable storage means; and a pitch ratio for changing the unencoded analog audio signal to a desired pitch; pitch ratio setting means for outputting a ratio signal; converting the time axis from the encoded information in the storage means according to a predetermined procedure according to the pitch ratio signal of the pitch ratio setting means and adding the encoded information to the encoded information; An audio system comprising: encoded signal successive output means for reading encoded signals for each block divided by marks; and PCM decoding means for decoding into analog audio signals according to the read encoded signals. In the signal time axis conversion device, the encoded information writing means is configured to change the block length of the encoded information written in the storage means based on the pitch ratio signal from the pitch ratio setting means. Features: A time axis conversion device for acoustic signals.