JP3334375B2 - Digital signal compression method and apparatus - Google Patents

Description

DETAILED DESCRIPTION OF THE INVENTION

[0001]

BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to recording and reproducing compressed data obtained by bit-compressing digital audio signals and the like.
In particular, according to the change on the frequency axis of the input signal, the frequency size of the small block subdivided according to the time and frequency at which the floating for information compression and / or the bit allocation for compression is performed is changed. The present invention relates to a digital signal compression method and apparatus.

[0002]

2. Description of the Related Art The applicant of the present invention has previously disclosed a technique for compressing an input digital audio signal into bits and recording the digital audio signal in bursts using a predetermined data amount as a recording unit, for example, in Japanese Patent Application No. 2-221364. , Japanese Patent Application No. 2-2136
No. 5, Japanese Patent Application No. 2-222821, Japanese Patent Application No. 2-2228
No. 23 has been proposed in the specification and drawings.

This technology uses a magneto-optical disk as a recording medium and records and reproduces AD (adaptive difference) PCM audio data defined in an audio data format of a so-called CD-I (CD-interactive) or CD-ROM XA. The ADPCM data is recorded in a burst on the magneto-optical disk using, for example, 32 sectors of ADPCM data and several sectors for linking for interleave processing as a recording unit.

Several modes can be selected for the ADPCM audio in the recording / reproducing apparatus using this magneto-optical disk. For example, the sampling frequency is twice as high as the normal CD reproducing time at a compression ratio of twice. Is 37.
A level A of 8 kHz, a level B of a sampling frequency of 37.8 kHz with a 4 times compression ratio, and a level C of a sampling frequency of 18.9 kHz with an 8 times compression ratio are defined. That is, for example, in the case of the level B, the digital audio data is compressed to approximately 1/4, and the reproduction time (play time) of the disc recorded in this level B mode is a standard CD format (CD). -D
A format). Since a recording and reproducing time of the same order as a standard 12 cm can be obtained with a smaller disk, the size of the apparatus can be reduced.

However, the rotation speed of the disk is a standard C
Since it is the same as D, for example, in the case of the level B, compressed data for a reproduction time four times as long as the predetermined time is obtained. For this reason, for example, the same compressed data is read out four times in units of time such as sectors or clusters, and only one of the compressed data is used for audio reproduction. Specifically, when scanning (tracking) a spiral recording track, 1
A track jump is performed to return to the original track position for each rotation, and the reproduction operation proceeds in such a form that the same track is repeatedly tracked four times. This means that normal compressed data need only be obtained at least once out of, for example, four times of redundant reading, and is resistant to errors due to disturbances and the like, and is particularly preferable when applied to portable small devices.

Further, the present applicant has proposed a bit allocation method for realizing efficient and good compression in Japanese Patent Application No. 4-369.
No. 52 in the specification and drawings. In this technique, when allocating bits, normalization is performed by a representative value in each small block such as a so-called critical band (critical band), that is, so-called block floating is performed, and bit allocation depending on a signal size in each small block is performed.
The weighting is performed according to the band corresponding to the small block. According to this technique, when there is no extreme variation in the magnitude of the spectrum in each small block, it is possible to perform compression satisfactorily.

[0007]

However, when digital data is compressed by applying the above-mentioned technology,
Signals containing extreme variances or characteristic peak components in the spectrum size in each small block for performing block floating, for example, when compressing a single signal or a plurality of sine waves or square waves, each signal including peaks Since the deviation of the data in the small block is large, the efficiency when the block floating is performed with the representative value in each small block may be reduced. Furthermore, when the state of the input signal, that is, the variation in the magnitude of the spectrum in each small block or the characteristic peak component changes, the efficiency of the previous block floating also changes, and the processing block in the time axis direction changes. A large bias in compression efficiency may occur. Also, if bit allocation for compression is performed in a state where the efficiency of block floating is reduced, redundant bits may need to be allocated in order to reduce the generated quantization noise to an allowable noise level or less. As a result, the compression efficiency may be reduced or a large bias may be caused.

In this case, the upper limit of the usable bit rate is specified in most cases of the application using the above-described technique because of the recording medium and the transmission path. If the processing block has a low compression ratio, the processing block having a high compression efficiency has an excessive number of bits, which causes a decrease in the overall compression efficiency. Further, if the upper limit is set to a processing block having a high compression efficiency, the amount of information is insufficient in a processing block having a low compression efficiency, and there is a possibility that a problem in audibility cannot be ignored. This problem becomes more serious as the available bit rate becomes lower, and it becomes more difficult to realize a lower bit rate, in other words, a higher compression rate, as the deviation of the compression efficiency of the processing block due to the input signal becomes larger. Become.

SUMMARY OF THE INVENTION The present invention has been made in view of such circumstances, and regardless of the characteristics of an input signal, block floating and information compression without lowering the efficiency of quantization and bit allocation for information compression. It is an object of the present invention to provide a digital signal compression method and apparatus to which the above method is applied.

[0010]

SUMMARY OF THE INVENTION A digital signal compression method and apparatus according to the present invention has been proposed to achieve the above-mentioned object, and an input digital signal is divided into small blocks subdivided in time and frequency. When distributing and performing information compression by performing floating for compression, the boundary on the frequency axis of the small block to be subjected to floating is determined in accordance with the integrated value of the amount of change on the frequency axis of the input digital signal. It is characterized in that it is determined and floating is performed for each of the determined small blocks.

Further, the digital signal compression method and apparatus of the present invention obtain spectrum data on a frequency axis of an input signal, obtain an allowable noise spectrum from the spectrum data, and convert the obtained allowable noise spectrum into a time signal. And divided into small blocks subdivided for frequency,
Bit allocation for compression is performed, and the bit allocation is optimized by changing the frequency size of a small block for performing bit allocation for compression in accordance with the characteristics on the frequency axis of the input signal. I do.

Further, the digital signal compression method and apparatus according to the present invention include a step or means for calculating a characteristic on the frequency axis of the input signal, and a method for distributing the input signal into small blocks subdivided in time and frequency, and A step or means for performing floating for obtaining the spectrum data on the frequency axis; a step or means for obtaining an allowable noise spectrum from the spectrum data; and A process or means for allocating bits to the subdivided small blocks and performing bit allocation for compression; and a small block subdivided in time and frequency for performing floating for compression in accordance with the characteristics on the frequency axis of the input signal. And / or the frequency size of small blocks subdivided in time and frequency for bit allocation for compression Changing the small block for block floating and the small block for bit allocation together or independently of each other to optimize block floating and bit allocation. I do.

Further, the digital signal compression method and apparatus of the present invention provide a small block that is subdivided into time and frequency for performing block floating for compression and / or a subdivision into time and frequency that performs bit allocation for compression. The frequency size of the small block is determined in advance, and the small block and / or the floating block for compression are compressed in accordance with the compression efficiency when the predetermined size is adopted and the characteristics on the frequency axis of the input signal. Or, compare the compression efficiency when changing the frequency size of the small block for performing bit allocation for compression, and select the size of the small block that can perform information compression with higher efficiency according to the comparison result. It is characterized by doing.

Here, in each digital signal compression method and apparatus according to the present invention, a small block subdivided into time and frequency for performing block floating for compression, and time and frequency for performing bit allocation for compression. The subdivided small blocks are configured commonly or independently according to the characteristics on the frequency axis of the input signal.

[0015]

[0016]

[0017]

That is, the digital signal compression method and apparatus (high-efficiency coding method and apparatus) according to the present invention calculate the characteristics of the input signal on the frequency axis and subdivide the input signal into time and frequency. The above problem is solved by performing floating for compression, controlling the frequency size of the small block to be floated according to the characteristics on the frequency axis of the input signal, and performing optimal floating. Resolve.

Further, when performing the bit allocation for performing the bit allocation for the compression, the time and the frequency for performing the bit allocation for the compression are divided into small blocks in accordance with the characteristics on the frequency axis of the input signal. The above problem is solved by changing the frequency magnitude.

Further, it is more effective to change both the small block for subdividing the time and the frequency for performing the above-mentioned block floating and the small block for bit allocation independently or independently.

On the other hand, a small block subdivided into time and frequency for block floating and / or bit allocation is determined in advance by a frequency magnitude obtained in accordance with the characteristics on the frequency axis of the input signal. It is more effective to compare the frequency size of the small blocks and select a size having a high compression efficiency overall.

[0022]

According to the digital signal compression method and apparatus of the present invention, for an input signal in which the deviation of the efficiency of block floating and / or bit allocation is large, the block floating and / or block for suppressing the deviation of the efficiency is small. Alternatively, by selecting the frequency size of a small block subdivided with respect to time and frequency for performing bit allocation, compression with a small deviation in efficiency can be realized. As a result, a decrease in compression efficiency can be prevented, and better sound quality can be obtained at the same bit rate, and recording, transmission, etc. at a lower bit rate at the same sound quality can be achieved. Can be realized.

[0023]

FIG. 1 is a block diagram showing a schematic configuration of an embodiment of a digital signal compression apparatus (compressed data recording / reproducing apparatus) of the present invention to which the digital signal compression method of the present invention is applied.

In the compressed data recording / reproducing apparatus shown in FIG. 1, a magneto-optical disk 1 driven by a spindle motor 51 is used as a recording medium. When recording data on the magneto-optical disk 1, for example, a so-called magnetic field modulation recording is performed by applying a modulation magnetic field corresponding to the recording data with the magnetic head 54 while irradiating the laser light with the optical head 53. The data is recorded along the recording track of. At the time of reproduction, the recording track of the magneto-optical disk 1 is traced by a laser beam by the optical head 53, and reproduction is performed magneto-optically.

The optical head 53 includes, for example, a laser light source such as a laser diode, a collimator lens, an objective lens,
The optical system includes optical components such as a polarizing beam splitter and a cylindrical lens, and a photodetector having a light receiving portion having a predetermined pattern. The optical head 53 is provided at a position facing the magnetic head 54 via the magneto-optical disk 1. When data is recorded on the magneto-optical disk 1, the magnetic head 54 is driven by a recording-system head drive circuit 66, which will be described later, to apply a modulation magnetic field in accordance with the recorded data. By irradiating the track with laser light, thermomagnetic recording is performed by a magnetic field modulation method. The optical head 53 detects reflected light of the laser beam applied to the target track, detects a focus error by, for example, a so-called astigmatism method, and detects a tracking error by, for example, a so-called push-pull method. When reproducing data from the magneto-optical disk 1, the optical head 53 detects the focus error and the tracking error, and at the same time, detects the difference in the polarization angle (Kerr rotation angle) of the reflected light of the laser light from the target track and reproduces the data. Generate a signal.

The output of the optical head 53 is supplied to an RF circuit 55. The RF circuit 55 extracts the focus error signal and the tracking error signal from the output of the optical head 53 and supplies the focus error signal and the tracking error signal to the servo control circuit 56.
Feed to 1.

The servo control circuit 56 comprises, for example, a focus servo control circuit, a tracking servo control circuit, a spindle motor servo control circuit, a thread servo control circuit and the like. The focus servo control circuit performs focus control of the optical system of the optical head 53 so that the focus error signal becomes zero. Further, the tracking servo control circuit performs tracking control of the optical system of the optical head 53 so that the tracking error signal becomes zero. Further, the spindle motor servo control circuit controls the spindle motor 51 so as to rotate the magneto-optical disk 1 at a predetermined rotation speed (for example, a constant linear speed). The thread servo control circuit moves the optical head 53 and the magnetic head 54 to target track positions of the magneto-optical disk 1 specified by the system controller 57. The servo control circuit 56 that performs such various control operations sends information indicating the operation state of each unit controlled by the servo control circuit 56 to the system controller 57.

A key input operation unit 58 and a display unit 59 are connected to the system controller 57. The system controller 57 controls a recording system and a reproduction system in an operation mode specified by operation input information from the key input operation unit 58. Further, the system controller 7 performs the above-described recording traced by the optical head 53 and the magnetic head 54 on the basis of the address information in the unit of sector reproduced from the recording track of the magneto-optical disk 1 by the header time and the Q data of the subcode. Manages recording and playback positions on a track. Further, the system controller 57
Control is performed to display the reproduction time on the display unit 59 based on the data compression ratio and the reproduction position information on the recording track.

This reproduction time display is based on address information (absolute time information) in sector units reproduced from a recording track of the magneto-optical disk 1 by a so-called header time or so-called subcode Q data, etc. For example, in the case of 1/4 compression, the actual time information is obtained by multiplying by 4), and this is displayed on the display unit 59. At the time of recording, if absolute time information is recorded in advance on a recording track of a magneto-optical disk or the like (preformatted), the preformatted absolute time information is read and the data compression ratio is adjusted. By multiplying the reciprocal, the current position can be displayed by the actual recording time.

Next, in the recording system of the disk recording / reproducing apparatus, the analog audio input signal AIN from the input terminal 60 is supplied to the A / D converter 6 via the low-pass filter 61.
The A / D converter 62 quantizes the analog audio input signal AIN. A / D converter 62
Digital audio signal obtained from the ATC (Ad
aptive Transform Coding) is supplied to the PCM encoder 63. The digital audio input signal DIN from the input terminal 67 is supplied to the ATC encoder 63 via the digital input interface circuit 68. A
The TC encoder 63 converts the input signal AIN into the A / D
Bit compression (data compression) processing is performed on the digital audio PCM data having a predetermined transfer rate quantized by the converter 62. Here, the compression ratio will be described as four times, but the present embodiment has a configuration that does not depend on this magnification and can be arbitrarily selected depending on the application example.

Next, writing and reading of data in the memory 64 are controlled by the system controller 57.
ATC data supplied from the ATC encoder 63 is temporarily stored, and is used as a buffer memory for recording on a disk as needed. That is, for example, the compressed audio data supplied from the ATC encoder 63 has a standard C
Data transfer rate of D-DA format (75 sectors /
Second), that is, 18.75 sectors / second, and the compressed data is continuously written to the memory 14. As described above, it is sufficient for this compressed data (ATC data) to record one sector per four sectors, but such recording every four sectors is practically impossible, so that a continuous sector as described later is used. I try to keep a record. This recording is performed in bursts at the same data transfer rate (75 sectors / second) as a standard CD-DA format by using a cluster composed of a predetermined plurality of sectors (for example, 32 sectors + several sectors) as a recording unit through a pause period. It is done on a regular basis. That is, in the memory 14, ATC audio data continuously written at a low transfer rate of 18.75 (= 75/4) sectors / second according to the bit compression rate is used as recording data. It is read out in bursts at the transfer speed. The overall data transfer speed of the read and recorded data, including the recording pause period, is as low as 18.75 sectors / sec. The instantaneous data transfer rate in the above is the standard 75 sectors / second. Therefore, when the disk rotation speed is the same speed (constant linear speed) as the standard CD-DA format, the same recording density and storage pattern as in the CD-DA format are recorded.

The ATC audio data, ie, recorded data, read from the memory 64 in a burst at the (instantaneous) transfer rate of 75 sectors / second is transferred to the encoder 65.
Supplied to Here, from the memory 64 to the encoder 65
In the data sequence supplied to the cluster, the unit continuously recorded in one recording is a cluster including a plurality of sectors (for example, 32 sectors) and several sectors for cluster connection arranged before and after the cluster. The cluster connection sector is set to be longer than the interleave length in the encoder 65, so that even if interleaved, data in other clusters is not affected.

As described above, the encoder 65 controls the recording data supplied from the memory 64 in a burst manner.
Encoding processing (parity addition and interleaving processing) for error correction, EFM encoding processing, and the like are performed. The recording data that has been subjected to the encoding process by the encoder 65 is supplied to the magnetic head drive circuit 66. The magnetic head drive circuit 66 is connected to the magnetic head 54,
The magnetic head 54 is driven so that a modulation magnetic field corresponding to the recording data is applied to the magneto-optical disk 1.

The system controller 57 controls the memory 64 as described above,
By this memory control, the recording position is controlled so that the recording data read out from the memory 64 in a burst manner is continuously recorded on the recording track of the magneto-optical disk 2.
The recording position is controlled by controlling the recording position of the recording data read in a burst from the memory 64 by the system controller 57 and transmitting a control signal designating the recording position on the recording track of the magneto-optical disk 1 to a servo control circuit. 56.

Next, the reproducing system of the magneto-optical disk recording / reproducing unit will be described. This reproducing system is for reproducing the recorded data continuously recorded on the recording tracks of the magneto-optical disk 1 by the recording system described above.
A decoder 71 is provided, in which a reproduction output obtained by tracing a recording track of the magneto-optical disk 1 with a laser beam by the optical head 53 is binarized by the RF circuit 55 and supplied. At this time, not only a magneto-optical disk but also a read-only optical disk similar to a compact disk (CD: COMPACT DISC) can be read.

The decoder 71 corresponds to the encoder 65 in the recording system described above, and performs the above-described decoding processing for error correction and EFM decoding on the reproduced output binarized by the RF circuit 55. Processing such as processing is performed to reproduce audio data at a transfer rate of 75 sectors / sec, which is faster than the normal transfer rate. This decoder 7
1 is supplied to the memory 72.

In the memory 72, data writing and reading are controlled by the system controller 57, and reproduced data supplied from the decoder 71 at a transfer rate of 75 sectors / second is written in burst at the transfer rate of 75 sectors / second. It is. Also, in the memory 72, the reproduced data written in a burst at the transfer rate of 75 sectors / second is a regular transfer rate of 18.75 of 75 sectors / second.
It is read continuously at sectors / second.

The system controller 57 writes the reproduced data to the memory 72 at a transfer rate of 75 sectors / second, and writes the reproduced data from the memory 72 to the above 18.7.
Memory control is performed such that data is read continuously at a transfer rate of 5 sectors / second. Further, the system controller 57 performs the above-described memory control for the memory 72, and reproduces the reproduction data written in a burst from the memory 72 by the memory control from the recording track of the magneto-optical disk 1 continuously. Control the playback position. This playback position control is performed by the system controller 5.
7 controls the reproduction position of the reproduction data read out from the memory 72 in a burst manner and supplies a control signal specifying the reproduction position on the recording track of the magneto-optical disk 1 or the optical disk 1 to the servo control circuit 56. Done.

ATC audio data obtained as reproduction data continuously read from the memory 72 at a transfer rate of 18.75 sectors / second is supplied to an ATC decoder 73. The ATC decoder 73 reproduces 16-bit digital audio data by expanding the ATC data by four times (bit expansion). This ATC
The digital audio data from the decoder 73 is D
/ A converter 74.

The D / A converter 74 includes an ATC decoder 73
Is converted into an analog signal to form an analog audio output signal AOUT. The analog audio signal AOUT obtained by the D / A converter 74 is output from an output terminal 76 via a low-pass filter 75.

Next, the high-efficiency compression coding applied to the information compression of the digital signal compression apparatus of the present invention will be described in detail. That is, refer to FIG. 2 et seq. For a technique of encoding an input digital signal such as an audio PCM signal using band division coding (SBC), adaptive conversion coding (ATC), and adaptive bit allocation. I will explain while.

In the specific high-efficiency coding apparatus shown in FIG. 2, in the steady state, the input digital signal is divided into a plurality of frequency bands, and at the same time, the bandwidths of the two lowest adjacent bands are the same. In the high frequency band, the higher the frequency band, the wider the bandwidth is selected, the orthogonal transform is performed for each frequency band, and the spectrum data of the obtained frequency axis is used in the low frequency band. Bits are adaptively allocated and coded for each critical bandwidth (critical band) and for each band obtained by subdividing the critical bandwidth in consideration of block floating efficiency in the middle and high frequency bands. Usually, this block is a quantization noise generating block. The critical band is a frequency band divided in consideration of human auditory characteristics, and a band of a pure tone when the pure tone is masked by a narrow band noise near the frequency of the pure tone. That is. The bandwidth of this critical band increases as the frequency increases, and the entire frequency band of 0 to 22 kHz is divided into, for example, 25 critical bands.

Further, in the embodiment of the present invention, before the orthogonal transformation, the block size (processing block length) is adaptively changed according to the input signal, and the block floating for performing the floating process according to the input signal. Processing for changing the size of the unit is performed, and quantization is performed so that the usage efficiency of the information after compression is optimized.

That is, in FIG. 2, when the sampling frequency is 44.1 kHz, for example,
An audio PCM signal of 22 kHz is supplied.
This input signal is divided into a band of 0 to 11 kHz and a band of 11 kHz to 22 kHz by a band division filter 201 composed of a filter such as a so-called QMF.
The signal in the 1 kHz band is also converted to a signal of 0 to 5.5 k
It is divided into a Hz band and a 5.5 kHz to 11 kHz band. 11 kHz to 22 k from the band division filter 201
The signal in the Hz band is sent to the MDCT circuit 203 which is an example of the orthogonal transform circuit,
The signal in the band of 5 kHz to 11 kHz is applied to the MDCT circuit 204.
From the band division filter 202 to 0 to 5.5 k
The signals in the Hz range are sent to the MDCT circuit 205 to be subjected to MDCT processing.

Here, a QMF filter as an example of a method of dividing the input digital signal into a plurality of frequency bands is described in, for example, the document “Digital Coding.
Of Speech in Subvans "(" Digital coding
of speech in subbands "RECrochiere, Bell Syst.
Tech. J., Vol.55, No.8 1976). This QMF filter divides a band into two equal bandwidths, and is characterized in that so-called aliasing does not occur when the divided bands are combined later.

Also, a document “Polyphase Quadrature Filters—New Band Division Coding Technology” (“Po
lyphase Quadrature filters -A new subband coding t
echnique ", Joseph H. Rothweiler ICASSP 83, BOSTON)
Describes an equal bandwidth filter division technique.
This polyphase quadrature filter is characterized in that a signal can be divided at a time when divided into a plurality of bands of equal bandwidth.

Further, as the above-mentioned orthogonal transform, for example, an input audio signal is divided into blocks in a predetermined unit time (frame), and a fast Fourier transform (F
FT), a discrete cosine transform (DCT), a modified DCT transform (MDCT), and the like, there are orthogonal transforms that transform the time axis into the frequency axis. This MDCT is described in the literature "Subband / Filter Using Filter Bank Design Based on Time Domain Aliasing Cancellation."
Transform Coding "(" Subband / Transform Coding Using Filte
r Bank Designs Based on Time Domain Aliasing Cance
llation, "JPPrincen ABBradley, Univ. of Surrey
Royal Melbourne Inst. Of Tech. ICASSP 1987).

Here, each of the MDCT circuits 203, 204,
FIG. 3 shows a specific example of a standard input signal for a block for each band supplied to 205. In the specific example of FIG. 3, the three filter output signals each have a plurality of orthogonal transform block sizes independently for each band,
The time resolution can be switched according to the time characteristics and frequency distribution of the signal. If the signal is quasi-stationary in time, the orthogonal transform block size is 11.6 ms,
That is, the long mode (Long M) shown in FIG.
mode) and if the signal is non-stationary,
The orthogonal transform block size is further divided into two and four.
FIG. 3B shows a short mode (Short Mod).
As shown in FIG. 3E, when the whole is divided into four to make 2.9 ms, or in the middle mode A (Middle) shown in FIG.
Mode A), middle mode B (M) in FIG.
As in the case of idle mode B), a part has a time resolution of 5.8 ms in two divisions and a part has a time resolution of 2.9 ms in four divisions, thereby adapting to an actual complicated input signal. It is clear that the division of the orthogonal transform block size is more effective if more complicated division is performed if the scale of the processing device allows. The block size is determined by the block size determination circuits 206 to 208 in FIG. 2 and transmitted to the MDCT circuits 203 to 205 and output from the output terminals 216 to 218 as block size information of the corresponding block.

FIG. 4 is a block circuit diagram showing a schematic configuration of a specific example of the block size determining circuit. Here, the block determination circuit 206 of FIG. 2 will be described as an example. Among the outputs of the band division filter 201 composed of the QMF of FIG. 2, the output of 11 kHz to 22 kHz is input terminal 4 of FIG.
01 to the power calculation circuit 404. Further, among the outputs of the band division filter 202 in FIG.
The output of 5 kHz to 11 kHz is supplied to the power calculation circuit 405 via the input terminal 402 of FIG. 4, and the output of 0 to 5.5 kHz is supplied to the power calculation circuit 40 via the input terminal 403 of FIG.
6 respectively. Further, the block size determination circuits 207 and 208 in FIG.
The operation is the same, except that the signal input to 3 is different from that of the block size determination circuit 206. The input terminals 401 to 403 of each of the block size determination circuits 206 to 208 have a matrix configuration. That is, the input terminals 401 of the block size determination circuit 207 are connected to the 5.5 kHz to 11 of the band division filter 202 of FIG.
The output of 0 kHz is connected to the input terminal 402, and the output of 0 to 5.5 kHz is connected to the input terminal 402. The same applies to the block size determination circuit 208.

In FIG. 4, each power calculation circuit 404-
Reference numeral 406 determines the power of each frequency band by integrating the input time waveform for a certain period of time. At this time, the time width for integration needs to be equal to or smaller than the minimum time block among the orthogonal transform block sizes described above. Also,
Other than the above calculation method, a similar effect can be obtained by using, for example, the absolute value or the average value of the maximum amplitude within the minimum time width of the orthogonal transform block size as the representative power. The output of the power calculation circuit 404 is sent to the change extraction circuit 408 and the power comparison circuit 409, and the outputs of the power calculation circuits 405 and 406 are sent to the power comparison circuit 409. The change extraction circuit 408 obtains the differential coefficient of the power sent from the power calculation circuit 404 and sends it to the block size primary determination circuit 410 and the memory 407 as power change information. The memory 407 stores the power change information sent from the change extraction circuit 408 for the maximum time of the orthogonal transform block size or more. This is because the temporally adjacent orthogonal transform blocks mutually influence each other by window processing at the time of orthogonal transform, so that the power change information of the immediately preceding temporally adjacent block is determined by the block size primary decision circuit 410. It is necessary. The block size primary determination circuit 410 and the power change information of the corresponding block sent from the change extraction circuit 408 and the memory 40
7, the orthogonal transform block size of the corresponding frequency band is determined from the temporal displacement of the power in the corresponding frequency band based on the power change information of the block immediately before the corresponding block temporally transmitted from 7. I do. In this case, if a displacement equal to or more than a certain value is recognized, a shorter time orthogonal transform block size is selected. However, the effect can be obtained even if the displacement point (boundary value) is fixed. Furthermore, when the value is proportional to the frequency, that is, when the frequency is high, a large displacement causes a short block size in time, and when the frequency is low, the block size is determined to be short in time with a small displacement compared to that in the high case. , Is more effective.
This value (boundary value) desirably changes smoothly, but may be a stepwise change in a plurality of stages. The block size determined as described above is transmitted to the block size correction circuit 411.

On the other hand, in the power comparison circuit 409, the power information of each frequency band sent from each of the power calculation circuits 404 to 406 is compared at the same time and on the time axis with the time width at which the masking effect occurs, and the power calculation is performed. Circuit 404
The influence of another frequency band on the output frequency band is calculated and transmitted to the block size correction circuit 411. The block size correction circuit 411 is based on the masking information sent from the power comparison circuit 409 and the past block size information sent from each tap of the delay group consisting of the delays 412 to 414.
The block size sent from 0 was modified to select a longer block size in time, and the delay 412
And to the window shape determination circuit 415. The effect of the block size correction circuit 411 is that, even when a pre-echo is a problem in the relevant frequency band, if a signal having a large amplitude exists in another frequency band, particularly in a band lower than the relevant frequency band, the masking effect is obtained. Pre-echo does not cause a problem in hearing,
Alternatively, it utilizes the characteristic that the problem may be reduced. The masking refers to a phenomenon in which a certain signal masks another signal and becomes inaudible due to human auditory characteristics. The masking effect includes time-axis masking by an audio signal on a time axis. There is an effect and a simultaneous masking effect by a signal on the frequency axis. Due to these masking effects, even if there is noise in the masked portion, this noise will not be heard. For this reason, in an actual audio signal, noise within the masked range is regarded as acceptable noise.

Next, in the delays 412 to 414, the past orthogonal transform block sizes are sequentially recorded, and are output to the block size determination circuit 411 from the outputs of the taps, ie, the outputs of the delays 412 to 414. At the same time, the output of the delay 412 is sent to the output terminal 417 and the delay 41
The outputs of 2, 413 are connected to a window shape determination circuit 415. The outputs from the delays 412 to 414 serve to help the block size correction circuit 411 determine the block size of the corresponding block based on a change in the block size over a longer time period. For example, a block size that is shorter in time in the past is frequently selected. If it is, it is possible to increase the selection of the block size that is short in time, and if the selection of the block size that is short in time has not been made in the past, it is possible to determine that the selection of the block size that is long in time is increased. And It should be noted that the number of taps of this delay group may be increased or decreased depending on the actual configuration and scale of the apparatus, except for the delays 412 and 413 required for the window determination circuit 415 and the output terminal 417. In the window shape determination circuit 415, the output of the block size correction circuit 411, that is, the block size of the next block which is temporally adjacent to the corresponding block and the delay 41
2 from the output of the block 2, ie, the block size of the block, and the output of the delay 413, ie, the immediately preceding block size of the block in question.
The shape of the window used in the CT circuits 203 to 205 is determined and output to the output terminal 416. The output terminal 417 shown in FIG.
16, ie, the window shape information is connected to each unit as an output of the block size determination circuits 206 to 208 in FIG.

Here, the window shape determined by the window shape determining circuit 415 will be described. FIG. 5 shows the shape of the adjacent block and window. As can be seen from FIGS. 6 (a) to 6 (c), the window used for the orthogonal transformation has a portion overlapping with the temporally adjacent block as shown by the dotted line and the solid line in FIG. In the example, since the shape overlapping the center of the adjacent block is adopted, the shape of the window changes depending on the orthogonal transformation size of the adjacent block.

FIG. 6 shows details of the window shape. In FIG. 6, window functions f (n) and g (n +
N) is given as a function satisfying the following equation (1).

F (n) × f (L-1-n) = g (n) × g (L-1-n) f (n) × f (n) + g (n) × g (n) = 1 ... (1) 0≤n≤L-1.

L in the equation (1) is the conversion block length if the adjacent conversion block lengths are the same, but if the adjacent conversion block lengths are different, the shorter conversion block length is set to L. , K is a longer conversion block length, and in a region where the windows do not overlap, it is given by the following equation (2).

F (n) = g (n) = 1 K ≦ n ≦ 3K / 2−L / 2 f (n) = g (n) = 0 3K / 2 + L ≦ n ≦ 2K (2) In this way, by making the overlapping portion of the window as long as possible, the frequency resolution of the spectrum at the time of orthogonal transformation is improved. As is clear from the above explanation,
The shape of the window used for the orthogonal transform is determined after the orthogonal transform size of three temporally continuous blocks is determined. Therefore, in the present embodiment, there is a difference of one block between the signal blocks input from the input terminals 401 to 403 and the signal blocks output from the output terminals 416 and 417 in FIG.

The power calculation circuits 405 and 40 shown in FIG.
6 and the power comparison circuit 409 can be omitted to configure the block size determination circuits 206 to 208 in FIG. Further, by fixing the shape of the window to the minimum temporal block size that the orthogonal transform block can take, the type is made one type, and the type of the window is determined by the delay 412 shown in FIG.
414, the block size correction circuit 411, and the window shape determination circuit 415 may be omitted. Particularly, in an application example in which a delay in the processing time is not desired, the above-described omission causes a configuration with a small delay, which effectively operates.

Referring again to FIG. 2, each MDCT circuit 203
The spectrum data or MDCT coefficient data on the frequency axis obtained by performing the MDCT processing in steps 205 to 205 are transmitted to adaptive bit allocation coding circuits 210 to 212, block floating unit determination circuits 219 to 221 and bit allocation calculation circuit 209. ing. The block floating unit determination circuits 219 to 221 determine a block floating unit, which is a unit for performing a floating process in accordance with the degree of concentration in energy or power, from the previous spectrum data or MDCT coefficient data,
The information is transmitted to the adaptive bit allocation encoding circuits 210 to 212 and output from the output terminals 216 to 218 as auxiliary information for decoding together with the above-described block size information and window shape information.

Here, FIG. 7 is a block circuit diagram showing a schematic configuration of one specific example of the block floating unit determination circuits 219 to 221 of FIG. The operation of the block floating unit determination circuit will be described with reference to FIG. In FIG. 7, an input terminal 301 is supplied with spectrum data or MDCT coefficient data on the frequency axis from the MDCT circuits 203 to 205. This spectrum data or MDCT coefficient data is sent to the variation calculating circuit 303. The variation calculating circuit 303 calculates the variation for each frequency by differentiating the intensity of the spectrum data or MDCT coefficient data with respect to frequency and obtaining a differential coefficient.
4 is transmitted. The integration and comparison circuit 304 integrates the variation data for each frequency in ascending order of frequency and compares it with the threshold value output from the threshold value output circuit 308.
Further, the frequency at the time when the integrated value exceeds the threshold value is transmitted to the block floating unit determination circuit 305 as the frequency of the unit boundary, the integration data is set to zero, the integration is started again, and all the frequencies are started. The above-described operation is performed for the change data. The threshold value output circuit 308 functions to output a threshold value so that the amount of change in the block floating unit is equal to or less than a certain value. And a good result has been obtained by applying a gradient distribution such that is large. If this threshold value is made variable depending on an application example, an input signal or the like, better results can be obtained, and depending on the application example, a constant can be used to simplify the configuration.

Next, in the block floating unit determination circuit 305, the frequency which becomes the boundary of the unit transmitted from the integration comparison circuit 304 and the output of the bit allocation calculation circuit 209 of FIG. The block floating unit is determined on the basis of the bit allocation, and transmitted to the unit correction circuit 306 and the energy calculation circuit 307. At this time, the boundary of the block floating unit is basically determined by the frequency serving as the boundary of the unit transmitted from the integration and comparison circuit 304, but the output of the bit allocation becomes zero in both adjacent units. In this case, since the advantage of separating the block floating unit disappears, the modification is performed so that the same block floating unit is used. Further, in the unit correction circuit 306, the boundary of the block floating unit transmitted from the block floating unit determination circuit 305 is corrected from the output of the energy calculation circuit 307 and the output of the standard unit output circuit 307, and output from the output terminal 310. The adaptive bit allocation encoding circuits 210 to 212 in FIG.
And output terminals 216-218. The energy calculation circuit 307 calculates the energy for each unit by integrating the spectrum data or MDCT coefficient data in the unit transmitted from the block floating unit determination circuit 305 for each unit,
The data is transmitted to the unit correction circuit 306. In the unit correction circuit 306, when the transmitted energy of each unit is equal to or less than a certain value, the corresponding unit is not made independent and is set as a common unit with an adjacent unit. This has the function of preventing high-intensity data from forming an independent unit in a limited portion with a narrow width in frequency, and similarly preventing small data from forming an independent unit in a narrow portion with a narrow width.

Further, in the standard unit output circuit 309, in the configuration of the standard block floating unit and in the applied example, the bandwidths of the two lowest bands adjacent to each other are the same, and the higher the frequency band, the higher the bandwidth. Widely selected, perform orthogonal transformation for each frequency band,
The obtained spectrum data on the frequency axis is subdivided into the critical band (critical band) in the low band for each so-called critical band in consideration of the human auditory characteristics described above, and the critical band in the middle and high band in consideration of the block floating efficiency. The unit divided for each divided band is output. In an application example, the standard block floating unit is common to a band for quantization, and has an advantage that data (auxiliary data) other than the compressed spectrum data or MDCT coefficient data portion can be reduced. have. On the other hand, focusing only on the efficiency of block floating, it is best if all the parts having large changes are configured as independent units. However, in this case, the number of auxiliary data increases. Therefore, the effective use of comprehensive information is determined in consideration of the amount of auxiliary data.
Therefore, the unit correction circuit 306 in FIG. 7 evaluates the amount of auxiliary data, the efficiency of block floating, and the amount of information required at the time of quantization in the unit after correction based on the output of the energy calculation circuit 307 described above. Only when the efficiency is higher than when the unit is adopted, the unit obtained earlier is adopted.

Here, the operation of the block floating unit will be described with reference to FIG. FIG. 8A illustrates a part of the spectrum data or MDCT coefficient data input from the input terminal 301 of FIG. Here, in the configuration of the standard block floating unit, in the above-mentioned frequency portion, three block floating units N-1, N, and N + 1 having a uniform frequency width as indicated by broken lines are formed. Shall be. In this state, in the N-1 unit, only the data of N-1 (2) is larger than the other data, and the block floating coefficient is determined by this data. Is normalized by the preceding block floating coefficient,
That is, the result is that some of the most significant digits of the normalized data are zero.

In FIG. 8B, a solid line graph represents a differential coefficient, dP / df, obtained by differentiating the spectrum data or MDCT coefficient data of FIG. 8A with respect to frequency. Further, the bar graph in the figure shows a value obtained by integrating the above-described differential coefficient and dP / df, Σ (dP / df). here,
Assuming that the output value of the threshold value output circuit 308 in FIG. 7 is a value indicated by a dashed line in FIG. 7, N-1 (2) and N-1 (3)
Since the integrated value, Σ (dP / df), exceeds the above-mentioned threshold value between the above data, the boundary with the block floating unit is set here. FIG. 8C is a diagram showing a block floating unit constituted by the boundaries obtained as described above. As is clear from the figure, in FIG. 8C, since the frequency width of the first unit, N′−1, becomes narrow and the deviation of each data in the unit decreases, FIG. The efficiency of block floating is improved as compared with. In addition, as described above, since the integration of the change from low-frequency data to high-frequency data is performed, the boundary of the unit immediately before a large change in the data often becomes a unit boundary. , Block floating of the data in the lower neighborhood and quantization control are facilitated. Therefore, masking to lower frequency sounds is less effective than higher frequency sounds, which is consistent with the auditory characteristic, and good compression can be realized.

Referring again to FIG. 2, bit allocation calculating circuit 2
09 is based on the spectrum data divided in consideration of the block floating transmitted by the above-described critical band and block floating unit determination circuits 219 to 221, and takes into account the so-called masking effect and the like, and takes the critical band and the block floating into consideration. The masking amount for each divided band is obtained, and the number of bits allocated to each band is obtained based on the masking amount and the energy or peak value of each divided band in consideration of the critical band and block floating.
It is transmitted to the adaptive bit allocation coding circuits 210 to 212. Adaptive bit allocation encoding circuits 210 to 212 quantize each spectrum data (or MDCT coefficient data) according to the number of bits allocated to each band. The data encoded in this manner is output to output terminal 2
13 to 215.

Next, FIG. 9 shows the bit allocation calculating circuit 20.
9 is a block circuit diagram illustrating a schematic configuration of one specific example of Ninth Embodiment 9. FIG.
The operation of the bit allocation calculation circuit will be described with reference to FIG. In FIG. 9, the input terminal 701 is
The spectrum data or MDCT coefficient data on the frequency axis is supplied from the MDCT circuits 203 to 205. The input data on the frequency axis is sent to the energy calculation circuit 702 for each band, and the energy of each divided band in consideration of the masking amount and the critical band and the block floating is, for example, the value of each amplitude value in the band. It is obtained by calculating the sum. Instead of the energy for each band, a peak value or an average value of the amplitude value may be used. This energy calculation circuit 7
As an output from 02, for example, the spectrum of the sum value of each band is shown as SB in FIG. However, in FIG. 10, for simplicity of illustration, the number of divided bands in consideration of the masking amount, the critical band, and the block floating is represented by 12 bands (B1 to B12).

Here, in order to consider the effect of the spectrum SB on so-called masking, a convolution process is performed such that the spectrum SB is multiplied by a predetermined weighting function and added. Therefore, the output of the energy calculation circuit 702 for each band, that is, each value of the spectrum SB is sent to the convolution filter circuit 703. The convolution filter circuit 703 includes, for example, a plurality of delay elements for sequentially delaying input data and a plurality of multipliers (for example, 25 corresponding to each band) for multiplying an output from these delay elements by a filter coefficient (weighting function). Multipliers) and a sum adder for summing the outputs of the multipliers. By this convolution processing, FIG.
The sum of the parts indicated by the dotted lines is calculated.

Here, a specific example of the multiplication coefficient (filter coefficient) of each multiplier of the convolution filter circuit 703 will be described. When the coefficient of the multiplier M corresponding to an arbitrary band is 1, the multiplier M-1 is a coefficient of 0.15, multiplier M-2 is a coefficient of 0.0019, and multiplier M-3 is a coefficient of 0.00000.
086, a coefficient 0.4 by a multiplier M + 1, and a multiplier M + 2
Is multiplied by the coefficient 0.06 by the multiplier M + 3 and the coefficient 0.007 by the multiplier M + 3, thereby performing the convolution processing of the spectrum SB. Here, M is an arbitrary integer of 1 to 25.

Next, the output of the convolution filter circuit 703 is sent to a subtractor 704. The subtractor 704 calculates a level α corresponding to an allowable noise level described later in the convolved area. The level α corresponding to the permissible noise level (permissible noise level) is, as described later, a level at which the permissible noise level of each band of the critical band is obtained by performing inverse convolution processing. It is. here,
The subtractor 704 is supplied with an allowance function (a function expressing a masking level) for obtaining the level α. The level α is controlled by increasing or decreasing the allowable function. The permissible function is supplied from the (n-ai) function generation circuit 705 described below.

That is, the level α corresponding to the allowable noise level can be obtained by the following equation (3), where i is a number sequentially given from the lower band of the critical band. α = S− (n−ai) (3) In the equation (3), n and a are constants, a> 0, S is the intensity of the convolution-processed bark spectrum, and (3)
In the expression, (n-ai) is the allowable function. In the present embodiment, n = 38 and a = 1, and there was no deterioration in sound quality at this time, and good coding could be performed.

In this way, the level α is obtained, and this data is transmitted to the divider 706. The divider 706 is for inversely convolving the level α in the convolved area. Therefore, by performing the inverse convolution processing, a masking spectrum can be obtained from the level α. That is, this masking spectrum becomes an allowable noise spectrum. Note that the above inverse convolution process requires a complicated operation, but in the present embodiment, inverse convolution is performed using a simplified divider 706.

Next, the masking spectrum is transmitted to a subtractor 708 via a synthesis circuit 707. Here, the output from the energy detection circuit 702 for each band, that is, the spectrum S
B is supplied via a delay circuit 709. Therefore, the subtracter 708 performs a subtraction operation between the masking spectrum and the spectrum SB, thereby obtaining the signal shown in FIG.
As shown, the spectrum SB is masked below the level indicated by the level of the masking spectrum MS.

The output from the subtracter 708 is taken out via an allowable noise correction circuit 710 and an output terminal 711.
M and the like (not shown). The ROM or the like assigns each band in accordance with the output (the level of the difference between the energy of each band and the output of the noise level setting means) obtained from the subtraction circuit 708 via the allowable noise correction circuit 710. Output bit number information. By transmitting the allocated bit number information to the adaptive bit allocation coding circuits 210 to 212 in FIG. 2, each spectrum data on the frequency axis from the MDCT circuits 203 to 205 in FIG. 2 is allocated to each band. It is quantized by the number of bits.

That is, in summary, in the adaptive bit allocation coding circuits 210 to 212 of FIG. 2, the masking amount, the energy of each divided band in consideration of the critical band and block floating, and the output of the noise level setting means are output. The spectral data of each band is quantized by the number of bits allocated according to the level of the difference. Note that the delay circuit 709 in FIG.
It is provided to delay the spectrum SB from the energy detection circuit 702 in consideration of the amount of delay in each circuit before 07.

By the way, at the time of synthesizing by the synthesizing circuit 707, data indicating a so-called minimum audible curve RC which is a human auditory characteristic as shown in FIG. The masking spectrum MS can be synthesized. At this minimum audible curve, if the absolute noise level is below this minimum audible curve, the noise will not be heard. Although the minimum audible curve is different depending on the reproduction volume at the time of reproduction, for example, even if the coding is the same, in a realistic digital system, for example, the method of entering music into the 16-bit dynamic range is not considered. Since there is not much difference, if quantization noise in the most audible frequency band around 4 kHz is not heard, for example, it is considered that quantization noise below the level of the minimum audible curve is not heard in other frequency bands. . Therefore, assuming that the system is used in such a manner that noise around 4 kHz of the word length of the system cannot be heard, and an allowable noise level is obtained by synthesizing the minimum audible curve RC and the masking spectrum MS together, In this case, the allowable noise level can be set up to the shaded portion in FIG. In this embodiment, the 4 kHz level of the minimum audible curve is adjusted to the lowest level corresponding to, for example, 20 bits. FIG.
2 also shows the signal spectrum SS at the same time.

In the allowable noise correction circuit 710,
The permissible noise level in the output from the subtracter 708 is corrected based on, for example, information on the equal loudness curve sent from the correction information output circuit 713. Here, the equal loudness curve is a characteristic curve relating to human auditory characteristics. For example, the loudness curve is obtained by calculating the sound pressure of sound at each frequency that sounds as loud as the pure tone of 1 kHz, and connecting the curves with each other. Also called a sensitivity curve. This equal loudness curve is the minimum audible curve R shown in FIG.
It draws substantially the same curve as C. In this equal loudness curve, for example, at around 4 kHz, even if the sound pressure falls by 8 to 10 dB below 1 kHz, it sounds as large as 1 kHz, and conversely, at around 50 Hz, the sound pressure must be about 15 dB higher than the sound pressure at 1 kHz. It doesn't sound the same size. For this reason, it can be seen that noise exceeding the level of the minimum audible curve (allowable noise level) should have a frequency characteristic given by a curve corresponding to the equal loudness curve. From this, it can be seen that correcting the allowable noise level in consideration of the equal loudness curve is suitable for human auditory characteristics.

Further, the correction information output circuit 713 detects the output information amount (data amount) at the time of quantization in the adaptive bit allocation encoding circuits 210 to 212, and outputs the bit rate target value of the final encoded data. The permissible noise level is corrected based on the information on the error between. This is because the total number of bits obtained by previously performing temporary adaptive bit allocation for all bit allocation unit blocks is a fixed number of bits (target value) determined by the bit rate of the final encoded output data. May have an error, and the bits are allocated again so that the error becomes zero. That is, when the total number of allocated bits is smaller than the target value, the difference bit number is allocated to each unit block and added. When the total allocated bit number is larger than the target value, the difference bit number is set to each unit block. It works by allocating to and shaving.

In order to perform the above operation, an error of the total allocated bit number from the target value is detected, and the correction information output circuit 713 corrects each of the allocated bit numbers according to the error data. Output data. Here, when the error data indicates that the number of bits is insufficient, it is possible to consider a case where the data amount is larger than the target value by using a large number of bits per unit block. Further, when the error data is data indicating the remainder of the number of bits, a case where the number of bits per unit block is small and the data amount is smaller than the target value can be considered. Therefore,
From the correction information output circuit 713, in accordance with the error data, the data of the correction value for correcting the allowable noise level in the output from the subtractor 708 based on, for example, the information data of the equal loudness curve is output. Will be output. By transmitting the correction value as described above to the allowable noise correction circuit 710, the subtractor 70
8 is corrected. In the system as described above, the data obtained by processing the orthogonal transform output spectrum with the sub-information as the main information and the scale factor indicating the block floating state as the sub-information, and the word length indicating the word length are obtained.
Sent from encoder to decoder.

FIG. 13 shows the ATC decoder 73 shown in FIG. 1, that is, a reproduced signal obtained by reproducing a signal recorded on a recording medium after being highly efficient coded as described above, or transmitted and received via a transmission medium. 2 shows a decoding circuit for decoding the received signal again. Quantized MD of each band
The CT coefficients, that is, data equivalent to the output signals of the output terminals 213 to 215 in FIG. 2 are supplied to the decoding circuit input 107, and the used block size information and the frequency of the small block for block floating and quantization are used. 2, ie, the output terminals 216 to 218 of FIG.
Is provided to an input terminal 108. The adaptive bit allocation decoding circuit 106 releases bit allocation using the adaptive bit allocation information. Next, in the inverse orthogonal transform (IMDCT) circuits 103 to 105, signals on the frequency axis are converted into signals on the time axis. The signals on the time axis of these partial bands are decoded into full-band signals by band synthesis filter (IQMF) circuits 102 and 101.

The present invention is not limited to the above embodiment. For example, the above-described recording / reproducing medium and a signal compression device or a decompression device, or a signal compression device and a decompression device are integrated. It is not necessary to connect the data, and it is also possible to connect between them by a data transfer line, an optical cable, communication by light or radio waves, etc., without using a recording medium. Further, for example, the present invention is applicable not only to audio PCM signals but also to signal processing devices for digital audio (speech) signals and digital video signals.

Further, by recording the data compressed by the digital signal compression device on a recording medium, the recording capacity can be effectively used. In addition, as this recording medium,
Not only the above-described optical disk, but also various types of recording media such as a magnetic disk, an IC memory, a card incorporating the memory, and a magnetic tape can be used.

[0082]

As is apparent from the above description, according to the digital signal compression method and apparatus of the present invention, an input signal that increases the deviation of the efficiency of block floating and / or bit allocation, in other words, For signals containing extreme variations in the size of data on the frequency axis in each small block subdivided in time and frequency for block floating and / or bit allocation and peak components that stand out, By selecting the frequency size of the small block so as to keep the deviation small, it is possible to realize the distribution of bits with a small deviation in compression efficiency. As a result, a decrease in compression efficiency can be prevented, and better sound quality can be obtained at the same bit rate, and recording, transmission, etc. at a lower bit rate at the same sound quality can be achieved. Can be realized. Therefore, high-efficiency compression and decompression with excellent sound quality can be performed in terms of hearing.

[0083]

[Brief description of the drawings]

FIG. 1 is a block circuit diagram illustrating a configuration example of a compressed data recording / reproducing device (disk recording / reproducing device) as an embodiment of a digital signal compression device of the present invention.

FIG. 2 is a block circuit diagram showing a specific example of a high-efficiency compression encoding encoder that can be used for bit rate compression encoding according to the embodiment.

FIG. 3 is a diagram illustrating a structure of an orthogonal transform block at the time of bit compression.

FIG. 4 is a block circuit diagram illustrating a configuration example of a circuit that determines an orthogonal transform block size.

FIG. 5 is a diagram illustrating a relationship between a change in temporal length of orthogonally adjacent orthogonal transform blocks and a window shape used at the time of orthogonal transform.

FIG. 6 is a diagram illustrating a detailed example of a window shape used at the time of orthogonal transformation.

FIG. 7 is a block circuit diagram illustrating a configuration example of a circuit that determines a block floating unit.

FIG. 8 is a diagram showing a spectrum clearly showing the progress of the operation of the block floating decision circuit in the present embodiment, as well as differential and integrated data.

FIG. 9 is a block circuit diagram showing an example of a bit allocation calculation circuit using a convolution operation for realizing a bit allocation operation function.

FIG. 10 is a diagram illustrating spectra of respective critical bands and bands divided in consideration of block floating.

FIG. 11 is a diagram showing a masking spectrum.

FIG. 12 is a diagram in which a minimum audible curve and a masking spectrum are combined.

FIG. 13 is a block circuit diagram showing a specific example of a high-efficiency compression encoding decoder that can be used for the bit rate compression encoding of the embodiment.

[Explanation of symbols]

 201, 202 Band division filter 203-205 Orthogonal transformation circuit (MDCT) 206-208 Block determination circuit 209 Bit allocation calculation circuit 210-212 Adaptive bit allocation coding circuit 303 Change calculation circuit 304 Integration comparison circuit 305 Block floating unit determination circuit 306 Unit correction circuit 307 Energy calculation circuit 308 Threshold output circuit 309 Standard unit output circuit 404 to 406 Power calculation circuit 407 Memory 408 Change extraction circuit 409 Power comparison circuit 410 Primary block size determination circuit 411 Block size correction circuit 412 414 delay 415 window shape determination circuit

Claims (10)

(57) [Claims] 1. A digital signal compression method for distributing an input digital signal into small blocks subdivided with respect to time and frequency, and performing floating for compression to compress information, wherein a change of the input digital signal on a frequency axis is provided. A digital signal compression method, comprising: determining a boundary on the frequency axis of a small block to be floated in accordance with an integrated value of an amount, and performing floating for each of the determined small blocks. 2. The method according to claim 1, wherein a frequency of which the integrated value of the change in frequency on the frequency axis exceeds a predetermined threshold determined in accordance with a band is set as a boundary of the small block to be floated. 2. The digital signal compression method according to claim 1, wherein: 3. Obtaining spectrum data on a frequency axis of an input digital signal, obtaining an allowable noise spectrum from the spectrum data, and distributing the obtained allowable noise spectrum into small blocks subdivided in time and frequency. In the digital signal compression method of performing information allocation by performing bit allocation for compression, a small block for performing bit allocation for compression in accordance with an integrated value of a change amount on the frequency axis of the input digital signal. A digital signal compression method, comprising determining a boundary on the frequency axis of (i) and assigning bits to each of the determined small blocks. 4. The integrated value of the amount of change on the frequency axis is:
3. The digital signal compression method according to claim 2, wherein a frequency exceeding a predetermined threshold value predetermined according to a band is set as a boundary of the small block to which the bit allocation for the compression is performed. 5. A digital signal compression method for compressing information of a digital signal, comprising the steps of: calculating a characteristic on a frequency axis of the input signal; distributing the input signal into small blocks subdivided in time and frequency; Performing a floating process, obtaining spectrum data on a frequency axis, obtaining an allowable noise spectrum from the spectrum data, and converting the allowable noise spectrum into small blocks subdivided with respect to time and frequency. A step of distributing and allocating bits for compression, and, according to characteristics on the frequency axis of the input signal, small blocks and / or bits for compression subdivided into time and frequency for performing floating for compression. Changing the frequency size of the small blocks subdivided for the time and frequency at which the allocation is performed, A digital signal compression method characterized by optimizing block floating and bit allocation by changing a small block for block floating and a small block for bit allocation together or independently. 6. A digital signal compression method for compressing information of an input digital signal, wherein a time and a frequency for performing block floating for compression are divided into small blocks and / or a time and frequency for performing bit allocation for compression. The frequency size of the subdivided small blocks is determined in advance, and compression is performed in accordance with the compression efficiency when the predetermined size is adopted and the integrated value of the variation of the input digital signal on the frequency axis. Small blocks with floating of
Or, by comparing the compression efficiency when the boundary on the frequency axis of the small block for which bit allocation for compression is determined is determined, the boundary of the small block that can perform information compression with higher efficiency is determined according to the comparison result. A digital signal compression method characterized by selecting. 7. A digital signal compressor for distributing an input digital signal into small blocks subdivided with respect to time and frequency, and performing floating for compression to compress information, wherein a change of the input digital signal on a frequency axis is provided. A digital signal compression apparatus characterized in that a boundary on the frequency axis of a small block to be floated is determined according to an integrated value of the amount, and floating is performed for each of the determined small blocks. 8. A spectrum data forming means for obtaining spectrum data on the frequency axis of the input signal, an allowable noise spectrum calculating means for obtaining an allowable noise spectrum from the spectrum data, and an allowable noise spectrum calculated by the allowable noise spectrum calculating means. A bit allocation means for distributing a possible noise spectrum into small blocks subdivided in time and frequency and allocating bits for compression, wherein the amount of change of the input digital signal on the frequency axis A digital signal compression apparatus for deciding a boundary on the frequency axis of a small block for performing bit allocation for compression in accordance with an integrated value of the small block, and optimizing bit allocation for each of the determined small blocks. . 9. A digital signal compressing apparatus for compressing information of a digital signal, wherein a characteristic calculating means for calculating a characteristic on a frequency axis of the input signal, and the input signal is subdivided with respect to time and frequency. Block floating means for distributing to small blocks and performing floating for compression, spectrum data forming means for obtaining spectrum data on the frequency axis, allowable noise spectrum calculating means for obtaining an allowable noise spectrum from the spectrum data, Bit allocation means for distributing the permissible noise spectrum obtained by the permissible noise spectrum calculation means into small blocks subdivided with respect to time and frequency, and performing bit allocation for compression; and The time and frequency of floating for compression Size control means for changing the frequency size of the divided small blocks and / or the subdivided small blocks with respect to time and frequency for performing bit allocation for compression; A digital signal compression apparatus characterized in that block floating and bit allocation are optimized by changing small blocks for bit allocation together or independently of each other. 10. A digital signal compression apparatus for compressing information of an input digital signal, wherein a time and a frequency for performing block floating for compression are divided into small blocks and / or a time and frequency for performing bit allocation for compression. The frequency size of the subdivided small block is determined in advance, and the compression efficiency when this predetermined size is adopted and the floating for compression according to the integrated value of the change amount of the input signal on the frequency axis are determined. The compression efficiency when the boundary on the frequency axis of the small block to be subjected to the compression and / or the small block for performing the bit allocation for the compression is determined is compared, and the information compression is performed with higher efficiency according to the comparison result. A digital signal compression apparatus comprising a block determining means for selecting a size of a small block that can be performed.