JPH0355839B2 - - 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 Application Field] The present invention provides a method for inputting an acoustic signal such as a musical tone signal.
Encoded audio signal processing that changes the pitch of musical tones or voices by performing encoding processing using PCM (pulse coding modulation), compressing or expanding the time axis of the signal, and outputting an audio signal. The present invention relates to an apparatus for converting the time axis of an acoustic signal.

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

However, in this type of time axis conversion device, the length of the block of the audio signal is fixed and written in a storage means such as a memory, and the block is compressed and expanded when read out. 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 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 the number of blocks generated at the cut points during decoding of the output signal increases. The resulting noise becomes noticeable, and the quality of the decoded audio signal, that is, the reproduced sound, may 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 from memory, and in order to compensate for the lack of time, the sound signal with a strong correlation is 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 memory, that is, the input audio signal, the length of the output audio signal will be skipped in response to the input sound. 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 includes processing for changing the pitch without changing the length of the sound;
In other words, when performing time axis conversion of an acoustic signal, when reading encoded information from memory, the number of cut points of the read block is kept as constant as possible to keep the audibility constant both when compressing and expanding the time axis. ,
When compressing the time axis, noise generation at block separation points is suppressed to prevent deterioration of the reproduced sound, and when the time axis is expanded, the reproduced blocks become redundant and the reproduced sound is delayed. It is an object of the present invention to provide a time axis conversion device for an acoustic signal that can change the pitch with a low distortion rate over a wide frequency range.

[Configuration of the Invention] The configuration of the present invention, which has been made to achieve the above object, as shown in FIG. , Encoding in which the digital signal is divided into predetermined block lengths by adding marks indicating cutting points, and sequentially written into the read/write storage means B as encoded information along a predetermined time axis. information writing means C; pitch ratio setting means D for setting a pitch ratio for changing the pre-encoded analog audio signal to a desired pitch and outputting a pitch ratio signal; and encoding information in the storage means B. , the time axis is converted by a predetermined procedure according to the pitch ratio signal of the pitch ratio setting means D, and the encoded signal is read out for each block divided by the mark attached to the encoded information. and a PCM decoding means F that performs decoding into an analog audio signal according to the read encoded signal. Time axis conversion of an acoustic signal, characterized in that the writing means C is configured to change the block length of encoded information to be written into the storage means B based on the pitch ratio signal from the pitch ratio setting means D. The gist is the device.

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

[Example] Figure 2 shows an example of inputting an analog audio signal such as a musical tone or voice, performing PCM encoding processing, and compressing/expanding the encoded signal without changing the length of the sound, that is, the playback speed. 1 is a block diagram of an audio signal time axis conversion device that performs axis conversion to subtly change the pitch of a reproduced sound of a decoded audio signal. Reference numeral 2 designates 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 for sampling the analog audio signal sent from the compressor 3.
It is a hold circuit and the write/read controller 8
The instantaneous amplitude values of the continuous acoustic signal waveform are extracted as sample values according to the time intervals of the sampling pulses sent from the oscilloscope. 5 is an A/D that quantizes the signal sampled by the sample/hold circuit 4 and converts it into binary encoded data.
It is a converter. In this embodiment, the above filter 2 to A/D converter 5 constitute PCM encoding means. 6 sequentially inputs the coded information (data) sent from the A/D converter 5 to 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 storage, and temporarily stores data by a latch signal supplied from the write/read controller 8. When a write permission signal is sent to the RAM 7 from a multiplexer, which will be described later, the data is stored in the RAM 7 through the data bus. The encoded data is sent to the RAM 7, and the encoded data is sequentially written to the designated address of the RAM 7.

9 indicates, for example, the least significant bit (LSB) when writing encoded data to RAM 7.
This is a marking circuit that attaches a mark (for example, a code of logic "1") to designate a cutting point of a block read from the RAM 7 when performing time axis conversion by compression/expansion of a signal. This marking circuit 9
Input the most significant bit (MSB) of encoded data obtained by converting an analog audio signal into a digital signal by the A/D converter 5, and when the sign changes from 0 to 1, this is used as the signal for negative going. A zero-crossing detection section detects the zero-crossing point and outputs a marking command signal, and in response to this marking command signal, a block is cut at, for example, the least significant bit (LSB) of the encoded data output from the A/D converter 5. A marking section is provided that provides a mark (for example, a code of logic "1") indicating the equinox.

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. In addition, timer circuit 1
0 is equipped with a second timer section that operates when a silent input signal is input, and when the silent input signal continues, that is, the marking of the cutting point specifying a block is performed for a long time such that it goes around the memory. In order to eliminate cases where 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, the marking command signal is forcibly sent to the marking section of the marking circuit 9. is configured to output.

The above-mentioned write latch 6, part of the write/read controller 8, marking circuit 9 and timer circuit 10
etc. constitute encoded information writing means, and A/D
The digital signal converted by the converter 5 is marked with the above marks, divided into predetermined block lengths, and written into the storage means (in this case, the register of the RAM 7) as encoded information along a predetermined time axis. It works like that. That is, the analog audio signal input from terminal 1 is sampled at a predetermined period, quantized, and encoded, and then stored in RAM 7 for each block.
It is stored in the internal register.

Here, the RAM when performing time axis conversion is
The length of the read block from 7 is determined by the time interval of the mark indicating the cutting point, that is, the set time of the timer circuit 10. It is controlled according to the degree of compression or expansion of the signal, that is, the pitch ratio between the original sound and the reproduced sound. In other words, if the block length during normal writing is constant, when time axis conversion is performed with the length of the reproduced sound constant, the read signal will change depending on the degree of compression or expansion of the signal. The number of cutting points for each block changes greatly, but this control circuit 11 adjusts the number of cutting points during writing to reduce the change in the number of cutting points in response to changes in pitch ratio. control the length of Therefore, the control circuit 11
The ratio between the period of the read clock signal i outputted from the variable block signal generator 13 and the period of the write clock signal h is detected by the pitch ratio signal io from the variable clock signal generator 13, and the period of this write/read is determined. In other words, when the pitch ratio between the original sound and the reproduced sound set by the pitch setting device 12, which is a pitch setting means, is large, the timer circuit 10 is set according to the degree of compression of the time axis. The timer setting time (Tm in FIG. 4) is lengthened, and conversely, when the set pitch ratio is small, the timer setting time Tm is shortened accordingly. This control circuit 11 is actually constituted by one gate array together with a write/read controller 8, a marking circuit 9, a timer circuit 10, etc.
The pitch setting device 12 sets the pitch ratio between the original sound and the reproduced sound in order to change the pitch of the reproduced sound higher or lower with respect to the input sound, and by its operation, the reading speed of encoded data from the RAM 7 is adjusted. You can adjust the pitch to change the pitch without changing the length of the original sound. This pitch setter 12 is a read counter A.
Read clock signal i applied to 14a and B14b
The clock signal generator 13 is configured as a variable resistor provided in the variable clock signal generator 13 so as to change the period of the clock signal.

15 is a clock signal generator that outputs a clock signal to the write/read controller 8 and the write counter 16, and the write counter 16 receives a divided write clock signal h for addressing to the RAM 7. It will be done. On the other hand, in the write/read controller 8, from the clock signal sent from the clock signal generator 15, the sample pulse signal a, the A/D
A/D conversion reference clock signal b for converter 5, write latch signal c, memory write enable signal d applied to RAM 7, memory read enable signal e, read latch signals f ₁ , f ₂ , and write to RAM 7 and a selection pulse signal g to be sent to the multiplexer 17 for switching the read operation. The timing of writing and reading encoded data to and from the RAM 7 is controlled so as to be performed continuously, for example, during one period of the write clock signal h. The write counter 16 is a ring counter that indicates a value corresponding to the address of the encoded data area in the RAM 7, and counts by inputting the clock signal h, and sends the write address to the multiplexer 1.
7 to RAM7. Therefore, when writing encoded data to the RAM 7, the encoded data is first transferred from the write/read controller 8 to the write latch 6.
Write latch 6 is activated by write latch signal C sent to
After being temporarily stored in the data bus and set on the data bus,
While the write enable signal d to the RAM 7 is activated and the multiplexer 17 is switched to the write side, encoded data is sequentially written to the addresses designated by the write counter 16.

A readout controller 18 controls the latch C19 and the latch D20, and is composed of a circuit equipped with a D-type flip-flop and a gate as shown in FIG. Send latch enable signals o, q and strobe signals r, p synchronized with signal Ma to latch C19 and latch D20,
Latch C19 and latch D20 alternately temporarily store the address information sent from write counter 16 at the timing of the block division point indicated by marking signal Ma.

21 is a mark detection circuit, which detects the mark signal m at the block division point added to the encoded data sequentially written in the RAM 7, and also detects the mark signal m at the block division point.
As shown in the timing chart in the figure, when reading the encoded data, marks in the encoded data are detected according to the reading speed, and a mark detection signal is sent.
De is output, and when two mark detection signals De are generated between two mark signals m, a load signal n is generated in synchronization with the second mark detection signal De.
Then, this load signal n is divided into every other load signal j ₁ and load signal j ₂ , and the OR gate 21
The data is sent to the two read counters A14a and B14b via the load signals _j1 and _j2 , respectively, and the read counters A and B use the latch C19 or the latch D20 to read out the address data at the beginning of the readout. Preset.
The reason why such two read counters A and B are used to read encoded data in two systems is to prevent skipping or repeated reading of encoded data at the signal block division point. This is to reduce noise caused by deviations in waveforms due to differences in signal pitch and amplitude, and the mutually inverted envelope signals EnA and EnB output from the envelope circuits 22a and 22b are output from the readout latch A23a and the readout latch B23b. is applied to the reference inputs of two D/A converters 24a and 24b to which the encoded data is respectively input,
Each block is not connected immediately; instead, when the previous data block passes the cut point, its output is gradually throttled down, and the new data block gradually increases its output from the cut point, so that Signal connections are being made.

The above-mentioned 23a and 23b are two systems of read latches, respectively, and the read latch signal f ₁ or f ₂ from the write/read controller 8 is sent to the multiplexer 17.
When the multiplexer 17 is switched to the read side, a selection pulse signal g is sequentially inputted to the readout side in synchronization with the readout selection. Latch the read encoded data.

The read counters A14a, B14b, multiplexer 17, read controller 18, latches 19, 20, mark detection circuit 21, and read latches 23a, 23b together with a part of the write/read controller 8 constitute encoded signal reading means. There is.

24a and 24b are D/A converters that input the encoded signals latched by read latches A and B and convert them into analog signals;
The signals output from 4a and 24b are sent to an integrator 25 for integration, then sent to an expander 26 to restore the signal amplitude compressed by the compressor 3 at the time of input, and further sent to an output filter 27. , so that an analog audio signal whose time axis has been converted and whose pitch has been changed can be obtained from the output terminal 28. The above D/A converters 24a, 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 tone with a harmonic third or fifth degree can be produced from a single sound source. can also be generated.

Also, by simply regenerating the digital signal encoded and stored in the RAM 7 by a predetermined time delay, mixing this with the original sound signal and outputting it, an echo is applied to the original analog sound signal. It is also easy.

Next, the operation of the time 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, and Fig. 5 shows the processing when compressing the audio signal as an example of time axis conversion of the audio signal. An explanatory diagram showing a graph showing the process of repeatedly reading out some blocks of the coded signal associated with this and a timing chart of each signal showing the operation, FIG. 6 shows the writing of the coded data to the RAM 7 FIG. 7 is a timing chart of each signal (a to h) showing the operation of reading and reading, and FIG. FIG. 8 is a diagram schematically showing the envelope of the input waveform of the integrator 25. FIG. 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 pitch 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.
If it is the sound of , the reproduced sound will be the sound of ``C,'' which is one octave higher. By setting this pitch ratio, the period of the clock signal of the variable clock signal generator 13 is adjusted in inverse proportion to the pitch ratio, with the period of the fixed clock signal of the clock signal generator 15 sent to the writing side circuit as a reference. Based on the signal writing speed, the reading speed increases in proportion to the pitch ratio.

After setting the pitch ratio and starting the operation of the device, the control circuit 11 controls the clock signal generator 15.
The period ratio of the fixed write clock signal h outputted from the variable clock signal generator 13 and the read clock signal i outputted from the variable clock signal generator 13 is detected by the pitch ratio signal io, 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 signal compression is performed, the timer circuit 1
The setting time (Tm) of the timer circuit 10 is lengthened, and when the pitch ratio is 1 or less and the signal is expanded, the setting time (Tm) of the timer circuit 10 is shortened accordingly. By controlling the set time of the timer circuit 10, the block length of the read signal becomes almost constant regardless of the compression or expansion processing, and the block length of the read signal becomes almost constant regardless of the compression processing or expansion processing. It is possible to eliminate deterioration in sound quality and the skipping of reproduced sound that occurs during expansion processing with a small pitch ratio.

An acoustic signal such as a musical tone signal is input from an input terminal 1, and passes through an input filter 2 to remove frequency components and noise unnecessary for signal processing from the signal.
Sent to compressor 3. The compressor 3 compresses the amplitude of the audio signal for noise reduction, etc. The amplitude-compressed analog audio signal is sent to the PCM encoding processing circuit, and first input to the sample/hold circuit 4 where it is sampled. It will be done. The sampling ring is a sampling pulse signal a sent from the write/read controller 8.
This is done at time intervals of , 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 a PCM code by the A/D converter 5 is then sent to the write latch 6.
and is latched by the latch signal C supplied from the write/read controller 8. Then, the encoded data latched by the write latch 6 is sequentially sent to the RAM 7 and the write counter 16
RAM 7 is being addressed based on an address signal supplied via multiplexer 17 from
The data is written in synchronization with the encoding cycle.

On the other hand, when the encoded data is sent from the A/D converter 5 to the write latch 6 and temporarily stored, the control circuit 11 adds a mark indicating the block division point of the signal to the least significant bit of the encoded data. It is applied at intervals based on the set time of the controlled timer circuit and at the zero-crossing position of the acoustic signal as follows.

That is, as shown in FIG. 4, the acoustic signal s input from the input terminal 1 is converted into encoded data by the A/D converter 5, but it is converted into encoded data using complement representation.
In the case of encoded data in a base number, the most significant bit (MSB) represents the sign of the value of the encoded data. Therefore, when the most significant bit (MSB) of the encoded data changes from 0 to 1, the marking section of the marking circuit generates the marking command signal u as a negative going zero crossing point,
A mark indicating a block cutting 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, but in such a case, the timer circuit 1
The second timer section in 0 outputs an 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 unconditionally generates a marking command signal, similar to the above. A mark indicating a block cutting point is added to a part of the encoded data. 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 transferred to the pitch setter 12 by the operation of the reading side circuit.
The signal is read out from the RAM 7 in real time as follows at a reading speed based on the pitch ratio set in , and according to a predetermined procedure while compressing or expanding the signal.

First, the mark detection circuit 21 detects the RAM 7
The mark signal m of the block division point attached to the encoded data sequentially written in the block is detected, and
As shown in the timing chart in the figure, the mark detection signal De detected according to the reading speed of the encoded data (the slope of the graph line B) is detected, and two mark detection signals De are detected between the two mark signals m. When this occurs, a load signal n is generated in synchronization with the second mark detection signal De. And every other load signal n is load signal j ₁ and load signal j ₂
are divided into two read counters A14a and B14b via OR gates 21a and 21b, and are sent to two read counters A14a and B14b respectively.
4a and B14b use these load signals _j1 and _j2 to preset the address data that is the beginning of the readout that is latched by the latches C19 and D20. 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 g sent from the write/read controller 8 to the multiplexer 17, the read enable signal e to the RAM 7 is When activated, encoded data is sequentially read from the address specified by the read counter 14a or 14b.

At this time, when the slope of the graph line B indicating the read speed is greater than the slope of the graph line A indicating the data write speed, that is, when compression of the time axis is performed, two marks are generated between the mark signals m. When the detection signal De is generated, the previous block is read out again to prevent the read processing from overtaking the write processing.

On the other hand, when the signal is expanded by setting the pitch ratio to 1 or less, two mark signals m are generated between two mark detection signals De, as shown in the graph of FIG. In this case, a load signal n is generated in the mark detection circuit 21 in synchronization with the second mark detection signal, and this is used as the two readout signals.
The read counters 14a and 14 are divided into j ₁ and j ₂ .
b, and blocks are skipped along a graph line D indicating a lower read speed with respect to a graph line C indicating a write speed.

The encoded data outputted from the RAM 7 is latched in read latches A23a and B23b and sent to two D/A converters 24a and 24b. The D/A converters 24a, 24b convert the read encoded signal into an analog signal, and the envelope signals EnA, EnB to which the reference input is applied cause the D/A converters 24a, 24b to
The amplitude is adjusted so that the peak of the waveform of the analog signal output from 24b, that is, the envelope, has a smooth slope. 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. An acoustic signal whose pitch has been raised by the signal compression process of axis conversion is output.

To summarize the main points of the above explanation, in this embodiment, when compressing the time axis by 1/2, that is, when the pitch ratio is set to 2, the length of one block during encoding is set to about 60 msec, for example, when decoding. The block length is approximately 30msec.
On the other hand, if you want to double the time axis, that is, reduce the pitch ratio to 1/2, the block length during encoding will be approximately 15 msec, and the block length during decoding will be approximately 30 msec. controlled as follows. Therefore, even when an audio signal is compressed and expanded, the block length during decoding is controlled to approximately a constant length.
When the time axis is shortened, the auditory sensation is kept almost constant, noise is suppressed at the block separation point to prevent deterioration of the reproduced sound, and when the time axis is further extended, the reproduction blocks become redundant. The problem with the playback sound being delayed has also been resolved.

In addition, since the audio signal is encoded using PCM, the audio signal level at the block joint during repeated readout of blocks during decoding and read skipping is almost zero, so other encoding method, e.g. taking the difference between acoustic signals
Compared to encoding methods such as DPCM and ΔM in which the seam occurs at the peak value of the acoustic signal, it is possible to perform time axis conversion of the acoustic signal with a low distortion rate over a wider frequency band. For example, in acoustic 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. However, in the PCM system, even a 3dB reduction covers the band from 20Hz to 8.5KHz. Also, in terms of total harmonic distortion,
In the DPCM method, it was 1.7%, but in the PCM method, it was 0.5%.
% or less. This is because in the DPCM method, although the peak values of the acoustic signals are not necessarily the same, the rate of change of the acoustic signals is zero, and the signals can be connected most smoothly.
This is thought to be because each block is connected at peak points where the magnitude of the acoustic signal is uneven, and the difference appears as signal distortion, which is the reason for limiting the frequency band. This problem was resolved by adopting the PCM method. Also, the PCM method has ΔM and
Compared to the DPCM method, it has the advantage that sharp attacks in acoustic signals can be reproduced more faithfully, and there is less delay in the auditory sense of sound, since no differences are calculated.

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. For example, a signal as shown in parentheses is output from the D/A converter 24a.
The converter 24b outputs a signal as shown in parentheses, for example, and as a result, the input signal to the integrator 25 has a shape shown in parentheses. Therefore, the joints between the decoding blocks are eliminated so that they are not noticeable to the listener.

In this embodiment, the write/read controller 8, the marking circuit 9, the timer circuit 10, and the control circuit 11, along with other peripheral gates, etc., are configured by one gate array in order to increase the control speed. However, it is also easy to configure it using 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.
It goes without saying that the invention can be implemented in various ways without departing from the gist of the invention.

[Effects of the Invention] As detailed above, the time axis conversion of an audio signal according to the present invention is configured to encode and decode an audio signal using PCM and change the block length according to a set pitch ratio. According to the device, even when compressing the signal by shortening the decoding cycle when performing time axis conversion that subtly changes the pitch without changing the length of the input original sound, the write Since the actual block length is controlled to be long, the number of division points for each block does not increase, and it is possible to prevent noise and deterioration of reproduced sound due to an increase in the number of division points. Furthermore, even when the decoding cycle is lengthened to expand the signal, the block length at the time of writing is controlled to be shortened, 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 transformation of the acoustic signal can be performed with low distortion over a wide frequency band. Also plays.

[Brief explanation of drawings]

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 the encoded signal. Fig. 5 is a timing chart showing the marking operation for setting the block length with . FIG. 6 is an explanatory diagram showing a graph showing the repeated readout process and a timing chart of each signal showing its operation in correspondence.
A timing chart of each signal showing the operation of writing and reading encoded data to RAM 7. Figure 7 shows the process of expanding the audio signal and the associated process of skipping reading some blocks of the encoded signal. FIG. 8 is a diagram schematically showing the envelope of the input waveform of the integrator 25. 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 ...
Write counter, 14a, 14b...read counter, 24a, 24b...D/A converter.

Claims (1)

[Claims] 1. PCM encoding means for inputting an analog audio signal and encoding it into a digital signal using PCM; a pitch ratio for changing the unencoded analog audio signal to a desired pitch; pitch ratio setting means for setting a pitch ratio signal and outputting a pitch 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 outputting the pitch ratio signal; encoded signal reading means for reading encoded signals for each block divided by marks attached to encoded information; and PCM for decoding into analog audio signals according to the read encoded signals. In the audio signal time axis converting device, the encoded information writing means writes a block length of the encoded information to the storage means based on the pitch ratio signal from the pitch ratio setting means. A time axis conversion device for an acoustic signal, characterized in that it is configured to change the time axis of an acoustic signal.