JP2001110143A - Digital signal processor and recorder - Google Patents

Abstract

PROBLEM TO BE SOLVED: To correct audio data so that a higher harmonic component hardly occurs when the audio data that is subjected to AD conversion and outputted are clipped in a full scale. SOLUTION: This digital signal processor is provided with a data latching part latching sample data, a number of times measuring part measuring the number of times when the sample data become full-scale data, a variable measuring part measuring the variable of a the sample data level, a storing part storing the data of a waveform pattern corresponded to the number of times and the variable, a data selecting part selecting either the sample data or the data of the waveform pattern stored in the storing part, and a waveform pattern selecting part selecting the waveform pattern according to the combination of the number of times and the variable. The waveform pattern selecting part selects the waveform pattern from the storing part and also controls the data selecting part so as to change the sample data into the data of the waveform pattern.

Description

DETAILED DESCRIPTION OF THE INVENTION

[0001]

BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to a digital signal processing device for digitally processing sample data of a signal such as audio or video, and a recording device for recording digitally processed data on a recording medium.

[0002]

2. Description of the Related Art MPEG (Moving Picture Exclusive) is a typical audio compression method for reducing the amount of information by using the psychoacoustic characteristics and omitting the amount of information with low sensitivity to hearing.
pertsGroup) There is audio. Audio compression processing methods such as MPEG1 and MPEG2 are standardized for MPEG audio, and Layer1, Layer2 and Layer
There are three modes of audio compression, r3.

In the MPEG1 audio layer 1 audio compression processing method, an input audio signal is divided into 32 sub-bands (hereinafter, referred to as SBs) of different frequency bands using 384 sample data as one processing unit. A scale factor (scale factor; hereinafter, referred to as SF), which is a ratio when quantized and normalized so that the maximum amplitude of the waveform of the sample data of each SB is approximately 1.0, and bits are appropriately assigned to each SB. Audio compression is performed using the allocated bit allocation.

[0004] Layer 2 of MPEG1 audio is layer 1
In addition to the audio compression processing described above, sample data of 384 × 3 samples are used as one processing unit, and bit allocation is performed using a compression processing table prepared for each data transfer rate for a plurality of data transfer rates. This is a voice compression method that performs voice coding with high quality and high efficiency.

[0005] Layer 2 of MPEG2 audio is basically an extension of the stereo of MPEG1 to multi-channels, or a structure in which the sampling frequency is reduced to 1/2 so that data can be transmitted at a low speed. The method uses the same technology as MPEG1.
Therefore, description of MPEG2 is omitted.

[0006] Layer 3 of MPEG1 audio is Layer 2
In the processing of the audio compression method, the variable discrete cosine transform, which is a frequency division method that is unlikely to cause aliasing, and the entropy that makes the code length variable, focusing on the bias in the appearance probabilities of parameters appearing in the process of compression processing (Huffman) This is a speech compression method that performs encoding with higher efficiency by using processing such as encoding. An overview of MPEG audio is available in the latest MPEG textbook (first published in August 1994,
ASCII Publishing Co., Ltd., P.176
Entropy coding is described on page 17).

In a recording / reproducing apparatus which divides a frequency band such as the above-mentioned MPEG audio into a plurality of SBs and compresses and records an audio signal by a masking effect, an input audio signal is subjected to AD (Analog to Digital) conversion. It is assumed that it is converted to linear. If the level of the input audio signal is too large to be converted linearly by the AD converter,
The output of the AD converter becomes sample data clipped with full-scale digital data. In such a state, since there are harmonic components that did not exist in the actual audio signal, an FFT (Fast Fourier Transform)
The frequency spectrum obtained by the conversion has S
B has a frequency spectrum larger than the actual level.

[0008]

If the size of the frequency spectrum of the SBs in the high-frequency portion becomes larger than the actual size and the number of bits allocated to these SBs increases, more bits are originally required. The number of bits allocated to the existing SB is reduced. Further, if no bits are assigned to the SB to which bits were originally assigned, the original sound cannot be faithfully reproduced, resulting in deterioration of sound quality. The present invention has been made in order to solve the above-mentioned problems.
It is an object of the present invention to provide a digital signal processing device that corrects sample data so that harmonic components do not easily occur when the converted and output sample data is clipped at full scale, and a recording device that records the corrected data.

[0009]

According to the present invention, there is provided a data latch unit for latching a plurality of continuous sample data, a number measuring unit for counting the number of times the sample data is continuously full-scale data, A change amount measurement unit that measures a level change amount between the sample data that has become full scale data and the sample data immediately before or immediately after full scale, a number of times that the number of times measurement unit measures, and the change amount measurement unit A storage unit that stores data of a waveform pattern corresponding to the amount of change to be measured, and data that selects one of the sample data latched in the data latch unit and the data of the waveform pattern stored in the storage unit. A waveform pattern of the storage unit based on a combination of a selection unit, a number of times measured by the number-of-times measurement unit, and a variation measured by the variation-amount measurement unit. A waveform pattern selection unit that selects a waveform pattern from the storage unit, and controls the data selection unit to switch the sample data of the data latch unit to data of a waveform pattern. It is a digital signal processing device.

Further, the present invention provides an input sample data correction section for correcting the level of input sample data, an encoder section for encoding corrected sample data output from the input sample data correction section, and the encoder A recording unit for recording data output from the unit,
A control unit that controls the operation of the encoder unit, wherein the input sample data correction unit is a data latch unit that latches a plurality of continuous sample data, and the sample data is continuous full-scale data. A number-of-times measuring unit that measures the number of times, a change amount measuring unit that measures a level change amount between the sample data that is full-scale data and the sample data immediately before or immediately after full-scale data, and the number-of-times measuring unit. A storage unit for storing waveform pattern data corresponding to the number of times of measurement and the change amount measured by the change amount measurement unit; sample data latched by the data latch unit; and a waveform pattern stored in the storage unit. A data selection unit for selecting one of the above data, the number of times measured by the number-of-times measurement unit, and the change number measured by the change-amount measurement unit. A waveform pattern selection unit that selects a waveform pattern in the storage unit based on a combination of amounts. A recording device that controls the data selection unit to switch to data.

Further, the present invention is the recording / reproducing apparatus according to the second aspect, further comprising an AD converter for converting an analog input signal into digital data.

According to a second aspect of the present invention, in the recording / reproducing apparatus according to the second or third aspect, the encoding section divides the input signal into sub-bands of different frequency bands, quantizes the sub-bands to obtain sample data. Dividing means, masking threshold calculating means for calculating a masking threshold, level comparing means for comparing the level of the sample data with the masking threshold, and allocating bits to the sub-band based on a comparison result of the level comparing means The recording / reproducing apparatus further includes a bit allocation unit, and a requantization unit that requantizes the sample data of the number of bits allocated to the subband by the bit allocation unit.

[0013]

BEST MODE FOR CARRYING OUT THE INVENTION

DESCRIPTION OF THE PREFERRED EMBODIMENTS An embodiment of the recording / reproducing apparatus according to the present invention will be described. FIG. 1 is a block diagram illustrating the configuration of the recording / reproducing apparatus according to the present embodiment. FIG. 2 is a diagram illustrating a configuration of a data generation unit that generates sample data and corrects the sample data in the recording and reproducing apparatus according to the present embodiment.

As shown in FIG. 1, the recording / reproducing apparatus of this embodiment includes a recording / reproducing section 100 and a data generating section 200.
The recording / reproducing unit 100 compresses the sample data and records it on the recording medium 101, and further reads / compresses the compressed data from the recording medium 101 and expands it.
An audio data reproducing unit that converts a digital signal output from the encoding / decoding unit 110 into an analog signal and outputs the analog signal; a memory unit 111;
, A display unit 105, and a system microcomputer 103.

A data generator 200 converts a digital audio signal into a digital signal to generate sample data, and a sample data corrector 202 corrects the sample data generated by the sample data generator 201. Is provided.

In FIG. 2, an automatic level control (ALC) 2 detects the level of an audio signal input from an input terminal 1 and automatically variably controls the gain of the circuit.

The protection circuit section 3 includes an AD (Analog to Digita)
l) The AD converter 4 is disposed at the input of the converter 4 and is limited so that the level of the audio signal output from the ALC 2 does not become larger than the input allowable value of the AD converter 4, and the AD converter 4 is electrically destroyed. Prevent that. As a circuit component used for the protection circuit section 3, for example, a diode is used.

The A / D converter 4 converts an analog audio signal into digital audio data.

The digital filter unit 5 performs an operation of oversampling the audio data output from the AD conversion unit 4, shifts the quantization noise out of the audible band, and thins out the oversampled data to serialize the audio data. Output as data.

The SP (Serial to Parallel) conversion unit 6
The serial audio data output from the digital filter 5 is converted into parallel audio data.

The sample data holding unit 7 includes a plurality of latch circuits 7-1, 7-2,.
And holds N pieces of parallel audio data output from. Hereinafter, the parallel audio data will be referred to as sample data.

The continuous full-scale counting section 8 counts the number of times that the sample data latched by the latch circuit 7-1 is continuously full-scale. In this embodiment,
The ranks are classified according to the number of sample data that continuously become full scale, and each rank is named a count flag. For example, when the number of full scales is less than 5 samples, the count flag is 0, the count flag is 1 for 5 to 10 samples, and the count flag is 2 for 11 to 20 samples. FIG. 5A is a table showing the relationship between the number of times flag described above and the number of continuous full scales.

The change amount measuring section 10 compares, for example, sample data of the latch circuits 7-1 and 7-2,
The amount of change in continuous sample data is measured. When the latch circuit 7-2 is the full-scale sample data and the sample data of the latch circuit 7-1 is the sample data smaller than the full scale, the change amount measuring unit 10 calculates the difference between the two sample data. It is measured and stored as a level change. In the present embodiment, the level change amount is divided into a plurality of ranks, and each rank is named a change flag. For example, if the magnitude of the change in the level of the sample data is less than 2 to 4 steps, the change flag is 0; if the magnitude of the change is less than 4 to 10 steps, the change flag is 1, and the magnitude of the change is Is 10 steps ~
If less than 20 steps, change flag 2 is set. FIG. 5B is a table showing the relationship between the above-described change flag and the amount of change in the level of the sample data.

The pattern selector 9 selects a waveform pattern similar to a sine wave stored in the memory unit 11 in advance, based on the count flag and the change flag.

The output control unit 12 holds the sample data of the waveform pattern selected and input from the memory unit 11 and outputs it to the sample data selection unit 13.

The sample data selection section 13 switches between the sample data latched by the latch circuit 7-N and the sample data of the waveform pattern held by the output control section 12 and outputs them. The switching operation of the sample data selection unit 13 is performed according to an instruction from the pattern selector 9.

FIG. 3 is a diagram showing the sample data output from the SP converter 6 in a time-series manner. FIG. 4 is a diagram showing waveform patterns stored in the memory unit 11 corresponding to the values of the change flag and the number of times flag. 3 and 4, the vertical axis represents the value of the change flag indicating the rank of the amount of change in the level of the sample data, and the horizontal axis represents the value of the number of times flag indicating the rank of the number of consecutive full-scale sample data. Show. FIG. 5 is a diagram showing the relationship between the number of times flag and the number of continuous full scales and the relationship between the change flag and the amount of change in the level of the sample data.

In FIG. 3, for example, the waveform of the sample data of the change flag 1 and the number of times flag 1 is such that the number of continuous full-scale sample data is six.
The amount of change in level between the full scale and the sample data before and after the full scale is four steps.

Here, the waveform pattern corresponding to the change flag 1 and the number-of-times flag 1 is as shown in the waveform pattern of the waveform pattern A portion of the change flag 1 and the number-of-times flag 1 in FIG.
The amount of change in the level of the sample data is 2 to 4 steps, and the number of continuous full-scale sample data is 1 sample. In FIG. 4, the hatched portion indicates the level at which the sample data has been corrected, and the corrected waveform pattern is a waveform approximating a sine wave.

FIG. 6 is a flowchart showing an example of an operation for selecting a waveform pattern from the memory unit 11 and correcting the generated sample data.

First, the polarity of the sample data latched by the latch circuit 7-1 is detected, and the polarity is compared with the immediately preceding sample data. (ST1), (ST2),
(ST3), (ST4)

It is detected whether the sample data latched by the latch circuit 7-1 is at full scale. If it is at full scale, the number of continuous full-scale samples is measured and added to the full-scale counter of the full-scale continuous number measurement unit 8, and when the sample data is no longer at full scale, the counter value is output. S
The process moves to T8. If the polarity of the continuous sample data or the sample data is not at full scale, the full scale number counter is reset to 0 and the process proceeds to ST1. (ST5), (ST6), (ST7)

When the number of full scale counters is three or more, the change amount measuring unit 13 calculates the magnitude of the level change of the two samples of the sample data at full scale and the sample data before and after full scale. calculate.
(ST8), (ST9), (ST9 ')

If the sample data is 16 bits, the total number of steps of the sample data is 65536 steps. However, there are cases where the level change exceeds 16384 steps in which the step changes suddenly, or a minute change of 2 steps or less. When there is a significant change, or when the number of full scales is less than three, the input original sample data is output as it is without correcting the sample data. (ST10), (S
T24)

If the level change is between two stages and less than 16384 stages, the process from ST11 to ST22 is performed to determine the values of the change flag and the number of times flag. (ST11)
~ (ST22)

When the polarity of the sample data, the value of the change flag, and the value of the number flag are determined, the waveform pattern stored in the memory 11 is read based on the polarity, the change flag, and the number flag, and the sample data is switched and output. (ST23)

FIG. 7 is a diagram for explaining the configuration of the encoding / decoding section of the recording / reproducing apparatus of this embodiment.

Sample data generated by the data generator 200 is input from the input unit 300. The band division unit 301 includes a plurality of bandpass filters that divide the audio frequency band into a plurality of frequency bands, and divides the input sample data into 32 different frequency spectrums for each SB.

The masking threshold calculator 302 calculates the power level of each SB by fast Fourier transform (FFT, Fast Fourier Transform) processing, and obtains a masking threshold using psychoacoustic characteristics. Masking means, for example, that in a quiet environment, you can hear the sound of a babbling, but in a storm, you may not be able to hear the sound. A state in which you cannot hear. The masking threshold is a maximum audio level at which an audio signal in a certain SB is masked by adjacent audio.

The bit allocation unit 303 allocates an appropriate number of bits to each SB based on the masking threshold output from the masking threshold calculation unit 302 and the level of the sample data of each SB. After allocating an appropriate number of bits to each SB, the requantization unit 304 re-quantizes the sample data with the allocated number of bits and outputs compressed sample data. Also, the bit allocation unit 303 represents the maximum amplitude value of the sample data waveform when the waveform of the sample data in the SB is normalized so that the maximum amplitude is approximately 1.0, and the requantized and compressed sample data. Scale factor (SF, scale) indicating the ratio of audio level to maximum amplitude value (magnification of compressed data)
factor) and output. The sample data is represented as an accurate audio signal by the requantized data and the calculated SF data.

The formatting unit 305 includes a synchronization signal,
Header information including identifiers of various modes of MPEG audio, each S determined by the bit allocation unit 303
Number of bits allocated to B, SF, and requantization unit 30
The sample data re-quantized in step 4 is converted into a signal of a frame format of the MPEG1 audio standard and output.

The level comparison section 306 compares the level of quantization noise generated when the sample data is requantized with the number of bits allocated to each SB with a masking threshold, and furthermore, the signal level and masking of each SB. The difference between the threshold levels is compared with a predetermined reference value. The comparison result is sent to the bit allocation unit 303, and the bit allocation unit 303 determines the number of bits to be allocated to each SB based on the comparison result.

The sample data input from the input unit 300 is input to the band division unit 301 and is converted into sample data for each SB divided into 32. At the same time, the sample data input from the input unit 300 is input to the masking threshold value calculation unit 302 and subjected to FFT processing,
A power level is calculated for each B. By calculating the power level for each SB, the masking threshold calculator 3
02 outputs a masking threshold of each SB by the masking effect.

In the bit allocation section 303,
The masking threshold is determined by each SB output from the band division unit 301.
And the number of bits to be assigned to each SB is determined so as to be within a predetermined total number of assignable bits. In the processing procedure for assigning the number of bits to each SB, first, the SB having the largest dynamic range is selected from the 32 SBs. Scan each SB to find the quantization noise to masking threshold ratio (NM
The SB having the largest R) is selected, and a part of the number of bits is assigned to the SB. Since the quantization noise changes due to the number of bits allocated to the SB and the NMR changes, the SB is scanned again after the NMR is calculated again and the NMR is calculated.
, The process of allocating a part of the number of bits to the SB and calculating the NMR again is repeated until all the assignable bits are assigned. After that, according to the number of bits allocated to each SB, the requantization unit 304 resamples and compresses the sample data allocated to each SB into sample data.

In the formatting section 305, MPEG1
Data such as a scale factor (SF) and sample data are arranged and output so as to be a signal of a frame format determined by the audio standard.

FIG. 8 is a diagram for explaining a state in which the input audio signal is divided into SBs and becomes sample data to which bits are assigned in the recording / reproducing apparatus of this embodiment.

The vertical axis of FIG. 8A shows the audio level and the level of the masking threshold (hatched portion) obtained by using the audio level and the psychoacoustic characteristics in each SB of the input audio data. The horizontal axis indicates the number of each SB.
The B0 side has a low frequency and the SB31 side has a high frequency.

FIG. 8B shows a state where the bit allocation unit 303 requantizes the sample data with the number of bits allocated in each SB. The vertical axis represents the number of bits of the sample data allocated to each SB, which is the number of quantization bits when the requantization unit 204 requantizes the sample data of each SB. The horizontal axis indicates the SB number, and the SB0 side has a lower frequency and the SB3
One side is a high frequency. FIG. 8B shows that in each SB, as the audio level of the input audio data is larger than the masking threshold, the number of bits is more allocated,
This indicates that the dynamic range increases.

Bits are not assigned to SB26, SB29, and SB31 because the audio level is lower than the masking threshold. Further, although the sound level of the SB 27 is higher than the masking threshold, no bit is allocated to the SB 27 because the algorithm preferentially allocates bits to SBs whose voice level is higher than the masking threshold.

FIG. 9 is a diagram showing the number of bits allocated to each SB before and after switching the sample data to the correction waveform data in the recording / reproducing apparatus of this embodiment. FIG. 9A is a diagram showing the number of bits allocated to each SB when the sample data of the SB 15 is clipped. FIG. 9B is a diagram showing the number of bits allocated to each SB when the sample data of the SB 15 is switched to the waveform data for correction.

In FIG. 9A, SB26 to SB3
1 is assigned the number of bits indicated by the hatched portion.
From SB21 to SB21, the number of bits is reduced. Also,
No bits are assigned to SB19 and SB20. When bits are assigned to the high-band SBs in this way, the sample data may be lost from the originally required SBs, such as the SB19 and SB20, thereby deteriorating the sound quality.

In this embodiment, as described above, the clipped sample data and the sample data before and after the clipped sample data are replaced with the sample data of the original sample data approximated by a sine wave. By performing this process, the number of bits allocated to SB26 to SB31 can be reduced, and the appropriate number of bits can be allocated to SB18 to SB21. In this way, the distribution of the number of bits to each SB shown in FIG. 9A changes to the distribution of the number of bits to each SB shown in FIG. 9B.

If the distribution of the number of bits to each SB is performed as shown in FIG.
Even at a large level at which the conversion unit becomes full scale, it is possible to obtain sound quality with little distortion.

The recording medium used in the recording apparatus described above includes a CD (Compact Disc), an M
D (Mini Disc), DVD (Digital Versatile Disc), P
An optical disk using D (Phase Change Disc), a magneto-optical disk or a phase change optical disk may be used.
It may be a magnetic hard disk.

[0055]

According to the present invention, even if the input audio or video signal is at such a large level that it becomes full scale in the A / D converter, the number of sample data that continuously becomes full scale and the number of sample data before and after the full scale are obtained. By replacing the sample data with the sample data determined in advance according to the magnitude of the change amount of the sample data, it is possible to allocate bits to the SB without causing distortion, and to obtain data with little distortion.

[Brief description of the drawings]

FIG. 1 is a block diagram illustrating a configuration of a recording / reproducing apparatus according to an embodiment of the present invention.

FIG. 2 is a diagram illustrating a configuration of a data generation unit that generates sample data and corrects the sample data in the recording / reproducing apparatus of the present embodiment.

FIG. 3 is a diagram showing, in chronological order, sample data output from an SP converter in the recording / reproducing apparatus of the present embodiment.

FIG. 4 is a diagram showing waveform patterns stored in a memory unit corresponding to values of a change flag and a number-of-times flag in the recording / reproducing apparatus of the present embodiment.

FIG. 5 is a diagram showing the relationship between the number of times flag and the number of continuous full scales and the relationship between the change flag and the amount of change in the level of sample data in the recording / reproducing apparatus of this embodiment.

FIG. 6 is a flowchart illustrating an example of an operation of selecting a waveform pattern from a memory unit and correcting sample data to be reproduced in the recording and reproducing apparatus of the present embodiment.

FIG. 7 is an outline of processing performed when audio data is compressed by the MPEG audio system in the recording / reproducing apparatus of the present embodiment.

FIG. 8 is a diagram illustrating a state in which an input audio signal is sample data divided into different SBs and assigned bits in the recording / reproducing apparatus of the present embodiment.

FIG. 9 is a diagram showing the number of bits allocated to each SB before and after correcting sample data with correction waveform data in the recording / reproducing apparatus of the present embodiment.

[Explanation of symbols]

1 input terminal, 2 ALC, 3 protection circuit section, 4
ADC, 5 digital filter, 6 SP conversion unit, 7 sample data holding unit, 7-1 to 7-N latch circuit, 8 continuous count measurement unit, 9 pattern selector, 10 change amount measurement unit, 11 memory unit, 12
Output control section, 13 sample data selection section, 14 digital compressed data generation section, 100 recording / reproduction section, 1
01 recording medium (IC memory), 103 system microcomputer, 104 operation unit, 105 display unit, 110
Encoding / decoding unit, 111 memory, 112
Audio playback unit, 200 data playback unit, 201
Sample data generation section, 202 sample data correction section, 300 input section, 301 band division section, 30
2 Masking threshold calculation section, 303 bit allocation section, 304 requantization section, 305 formatting section

Claims (4)

[Claims] A data latch unit for latching a plurality of continuous sample data; a number measurement unit for measuring the number of times the sample data is continuously full-scale data; A change amount measurement unit that measures the amount of change in level between the data and the sample data immediately before or immediately after the full scale,
A storage unit that stores data of a waveform pattern corresponding to the number of times measured by the number-of-times measurement unit and the amount of change measured by the amount-of-change measurement unit, and a sample data latched by the data latch unit and the storage unit. A data selection unit for selecting any of the stored waveform pattern data, and a waveform for selecting the waveform pattern of the storage unit based on a combination of the number of times measured by the number-of-times measurement unit and the change amount measured by the change-amount measurement unit A pattern selection unit, wherein the waveform pattern selection unit selects a waveform pattern from the storage unit and controls the data selection unit to switch the sample data of the data latch unit to data of a waveform pattern. Digital signal processing device. 2. An input sample data corrector for correcting the level of input sample data, an encoder for encoding corrected sample data output from the input sample data corrector, and an encoder output from the encoder. A recording unit that records data to be read, and a control unit that controls the operation of the encoder unit.The input sample data correction unit includes: a data latch unit that latches a plurality of continuous sample data; A frequency measuring unit for measuring the number of times that the data is full-scale data, and a change amount for measuring a level change amount between the sample data that is the full-scale data and the sample data immediately before or immediately after the full-scale data. A measuring unit, the number of times measured by the number-of-times measuring unit, and the amount of change measured by the amount-of-change measuring unit. A storage unit for storing data of the associated waveform pattern; a data selection unit for selecting either the sample data latched by the data latch unit or the data of the waveform pattern stored in the storage unit; A waveform pattern selection unit that selects a waveform pattern in the storage unit based on a combination of the number of times the unit measures and the change amount measured by the change amount measurement unit, wherein the waveform pattern selection unit converts the waveform pattern from the storage unit. A recording apparatus for selecting and controlling the data selection unit to switch the sample data of the data latch unit to data of a waveform pattern. 3. The recording / reproducing apparatus according to claim 2, further comprising an AD converter for converting an analog input signal into digital data. 4. The recording and reproducing apparatus according to claim 2, wherein the encoding unit divides the input signal into sub-bands of different frequency bands, quantizes the sub-band into sample data, and quantizes the sub-band into sample data. Masking threshold calculating means for calculating a masking threshold, level comparing means for comparing the level of the sample data with the masking threshold, and bit allocation means for allocating bits to the sub-band based on a comparison result of the level comparing means. Recording / reproducing apparatus, further comprising requantization means for requantizing sample data of the number of bits allocated to the subband by the bit allocation means.