Patent Publication Number: US-2006004566-A1

Title: Low-bitrate encoding/decoding method and system

Description:
CROSS-REFERENCE TO RELATED APPLICATION  
      This application claims the benefit of Korean Patent Application No. 2004-48036, filed on Jun. 25, 2004, in the Korean Intellectual Property Office, the disclosure of which is incorporated herein in its entirety by reference.  
     BACKGROUND OF THE INVENTION  
      1. Field of the Invention  
      The present invention relates to an encoding/decoding method and system, and more particularly, to a low bitrate encoding/decoding method and system that can efficiently compress data at a low bitrate and thus provide a high quality audio signal.  
      2. Description of the Related Art  
      A waveform including information is originally an analog signal which is continuous in amplitude and time. Accordingly, analog-to-digital (A/D) conversion is required to represent a discrete waveform. A/D conversion comprises two distinct processes: sampling and quantizing. Sampling refers to the process of changing a signal continuous in time into a discrete signal, and quantizing refers to the process of limiting the possible number of amplitudes to a finite value, that is, the process of transforming an input amplitude x(n) at time ‘n’ into an amplitude y(n) taken from a finite set of possible amplitudes.  
      With a recent development in digital signal processing technologies, there have been developed and widely used techniques for storing/restoring an audio signal where an analog signal is converted into pulse code modulation (PCM) data, a digital signal, through sampling and quantizing processes, and the converted signal is stored in recording/storing media, such as MP3 player, a compact disc (CD) and a digital audio tape (DAT), to allow users to play back the stored signal. While this digital storing/restoring technology is much improved in terms of sound quality and storage as compared to an analog source, such as a long-play record (LP) or tape, there has been a problem in data storage and transmission due to a huge amount of digital data.  
      Here, a CD is a medium for storing data obtained by sampling an analog stereo (left/right) audio signal at a rate of 44,100 per sec and a 16 bit resolution, which can be played back. For example, in case of converting an analog audio signal for presenting a 60 sec music sequent into 2-channel digital audio data with CD audio quality, the analog audio signal is converted into digital data with a sampling rate of 44.1 kHz for 16 bits. Thus, a 60 sec music sequent requires 10.58 Mbyte (44.1 kHz*16 bits*2*60). Accordingly, transmitting a digital audio signal via a transfer channel needs a high transfer bitrate.  
      In order to overcome this problem, the differential pulse code modulation (DPCM) or adaptive differential pulse code modulation (ADPCM), which was developed to compress digital audio signals, has been employed to reduce the data size, but there has been a problem in that significant differences in efficiency occur depending on types of signals. Recently, ISO (International Standard Organization) MPEG/audio (Moving Pictures Expert Group) algorithm or Dolby AC-2/AC-3 algorithm has employed a psychoacoustic model to reduce the data size. These algorithms efficiently reduce the data size irrespective of types of signals.  
      In conventional audio compression algorithms such as MPEG-1/audio, MPEG-2/audio, or AC-2/AC-3, a time-domain signal is divided into subgroups to be transformed into a frequency-domain signal. This transformed signal is scalar-quantized using the psychoacoustic model. This quantization technique is simple but not optimum although input samples are statistically independent. Of course, statistically dependent input samples are more problematic. In order to solve this problem, a lossless encoding, such as an entropy encoding, is performed, or an encoding operation is performed including an adaptive quantization algorithm. Accordingly, such an algorithm requires very complicated procedures as compared to an algorithm where only PCM data is encoded, and a bitstream includes additional information for compressing signals as well as quantized PCM data.  
      MPEG/audio or AC-2/AC-3 standards present as high audio quality as a CD at bitrates of 64-384 Kbps, which are a sixth or an eighth of conventional digital encoding bitrates. Therefore, MPEG/audio standards will play a significant role in audio signal storage and transfer in a multimedia system, such as a Digital Audio Broadcasting (DAB), an Internet phone, or an Audio on Demand (AOD) system.  
      These conventional techniques are relatively useful for encoders. However, with the advent of mobile multimedia applications, there has been required a low bitrate audio encoding/decoding method and system that can perform both encoding and various functions at low bitrates.  
     SUMMARY OF THE INVENTION  
      Additional aspects and/or advantages of the invention will be set forth in part in the description which follows and, in part, will be apparent from the description, or may be learned by practice of the invention.  
      The present invention provides a low bitrate audio encoding/decoding method and system that can effectively compress data at a relatively low bitrate and thus provide a high quality audio signal using algorithms for reducing and recovering frequency components.  
      According to an aspect of the present invention, there is provided a low-bitrate encoding system including: a time-frequency transform unit transforming an input time-domain audio signal into a frequency-domain audio signal; a frequency component processor unit decimating frequency components in the frequency-domain audio signal; a psychoacoustic model unit modeling the received time-domain audio signal on the basis of human auditory characteristics, and calculating encoding bit allocation information; a quantizer unit quantizing the frequency-domain audio signal input from the frequency component processor unit to have a bitrate based on the encoding bit allocation information input from the psychoacoustic model unit; and a lossless encoder unit encoding the quantized audio signal losslessly, and outputting the encoded audio signal in a bitstream format.  
      According to another aspect of the present invention, there is provided a low-bitrate encoding method including: transforming an input time-domain audio signal into a frequency-domain audio signal; decimating frequency components in the frequency-domain audio signal; modeling the received time-domain audio signal on the basis of human auditory characteristics, and calculating encoding bit allocation information; quantizing the frequency-domain audio signal input through the decimating of frequency components to have a bitrate based on the encoding bit allocation information input through the modeling of the audio signal; and encoding the quantized audio signal losslessly and outputting the encoded audio signal in a bitstream format.  
      According to another aspect of the present invention, there is provided a low-bitrate decoding system including: a lossless decoder unit decoding an input bitstream losslessly and outputting the decoded audio signal; an inverse quantizer unit recovering an original signal from the decoded audio signal; a frequency component processor unit increasing frequency coefficients of the audio signal in the inversely quantized frequency-domain audio signal; and a frequency-time transform unit transforming the frequency-domain audio signal input from the frequency component processor unit into a time-domain audio signal.  
      According to another aspect of the present invention, there is provided a low-bitrate decoding method including: decoding an input bitstream losslessly and outputting the decoded audio signal; recovering an original signal from the decoded audio signal; increasing frequency coefficients of the audio signal in the recovered frequency-domain audio signal; and transforming the frequency-domain audio signal input through the increasing of frequency coefficients into a time-domain audio signal. 
    
    
     BRIEF DESCRIPTION OF THE DRAWINGS  
      These and/or other aspects and advantages of the invention will become apparent and more readily appreciated from the following description of the embodiments taken in conjunction with the accompanying drawings in which:  
       FIG. 1  is a block diagram showing a low bitrate audio encoding system according to present invention;  
       FIG. 2  is a block diagram showing the frequency component processor unit  110  shown in  FIG. 1  according to an embodiment of the present invention;  
       FIG. 3  is a block diagram showing an embodiment of filter and decimation units shown in  FIG. 2  according to present invention;  
       FIG. 4  is a block diagram showing another embodiment of the frequency component processor unit  110  shown in  FIG. 1  according to the present invention;  
       FIG. 5  is a flowchart showing an operation of a low bitrate audio encoding system according to the present invention shown in  FIG. 1 ;  
       FIG. 6  is a flowchart showing an example of operation  510  shown in  FIG. 5 ;  
       FIG. 7  is a flowchart showing another embodiment of operation  510  shown in  FIG. 5  according to the present invention;  
       FIGS. 8A through 8D  show an example of signal variations based on frequency signal processing in an embodiment of a low bitrate audio encoding system according to the present invention;  
       FIGS. 9A through 9D  show another example of signal variations based on frequency signal processing in an embodiment of a low bitrate audio encoding system according to the present invention;  
       FIG. 10  is a block diagram showing an embodiment of a lossless audio decoding system according to the present invention;  
       FIG. 11  is a block diagram showing an aspect of the frequency component processor unit  1040  shown in  FIG. 10  according to the present invention;  
       FIG. 12  is a block diagram showing another construction of the frequency component processor unit  1040  shown in  FIG. 10 ;  
       FIG. 13  is a flowchart showing an operation of a lossless audio decoding system according to the present invention shown in  FIG. 10 ;  
       FIG. 14  is a flowchart showing an example of operation  1340  shown in  FIG. 13 ;  
       FIG. 15  is a flowchart showing another example of operation  1340  shown in  FIG. 13 ;  
       FIGS. 16A and 16B  show an example of an audio signal for a predetermined subband in an encoding operation and in a decoding operation, respectively; and  
       FIGS. 17A and 17B  show another example of an audio signal for a predetermined subband in an encoding operation and in a decoding operation, respectively. 
    
    
     DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENTS  
      Reference will now be made in detail to the embodiments of the present invention, examples of which are illustrated in the accompanying drawings, wherein like reference numerals refer to the like elements throughout. The embodiments are described below to explain the present invention by referring to the figures.  
       FIG. 1  is a block diagram showing an embodiment of a low bitrate audio encoding system according to an aspect of the present invention. The system comprises a time-frequency transform unit  100 , a frequency component processor unit  110 , a quantizer unit  120 , a lossless encoder unit  130 , a psychoacoustic model unit  140 , and a bitrate control unit  150 .  
      The time-frequency transform unit  100  transforms a time-domain audio signal into a frequency-domain audio signal. A modified discrete cosine transform (MDCT) may be used to transform a time-domain signal into a frequency-domain signal.  
      The frequency component processor unit  110  receives a frequency-domain audio signal from the time-frequency transform unit  100 , and transforms N frequency coefficients into N′ frequency coefficients in the frequency-domain audio signal, where N′ is less than N. This transform may be regarded as a non-linear and non-invertible transform. The frequency component processor unit  110  divides frequency components into subbands. An integer MDCT may be used to divide frequency components into subbands.  
      In order to remove perceptual redundancy based on human auditory characteristics, the psychoacoustic model unit  140  transforms an input audio signal into a frequency-domain spectrum, and determines encoding bit allocation information on signals not to be perceived by the human ear with respect to each of the subbands in the frequency component processor unit  110 . The psychoacoustic model unit  140  calculates a masking threshold for each of the subbands, which is encoding bit allocation information, using a masking phenomenon resulting from interaction between predetermined subbands signals divided in the frequency component processor unit  110 . The psychoacoustic model unit  140  outputs the calculated encoding bit allocation information to the quantizer unit  120 . In addition, the psychoacoustic model unit  140  determines window switching based on a perceptual energy, and outputs the window switching information to the time-frequency transform unit  100 .  
      The quantizer unit  120  quantizes a frequency-domain audio signal, which is input from the frequency component processor unit  110  and transformed into N′ frequency components, to have a bitrate based on the encoding bit allocation information input from the psychoacoustic model unit  140 . That is, frequency signals in each of the subbands are scalar-quantized so that a quantization noise amplitude in each of the subbands is less than a masking threshold, which is encoding bit allocation information, and thus the human ear cannot perceive the signals. Using a noise-to-mask ratio (NMR), which is the ratio between the masking threshold calculated in the psychoacoustic model unit  140  and noise generated in each of the subbands, a quantization is performed so that an NMR value in each of the subbands is not greater than 0 dB. The NMR value not greater than 0 dB indicates that the masking value is higher than the quantization noise, which means that the human ear cannot perceive the quantization noise.  
      The lossless encoder unit  130  losslessly encodes the quantized audio signal received from the quantizer unit  120 , and outputs the encoded signal in a bitstream format. The lossless encoder unit  130  can efficiently compress signals using a lossless coding algorithm, such as a Huffman coding or arithmetic coding algorithm.  
      The bitrate control unit  150  receives information on the bitrate of the bitstream from the lossless encoder unit  130 , and outputs a bit allocation parameter suitable for the bitrate of the bitstream to be output to the quantizer unit  120 . That is, the bitrate control unit  150  controls the bitrate of the bitstream to be output and outputs the bitstream at a desired bitrate.  
       FIG. 2  is a block diagram showing an embodiment of the frequency component processor unit  110  shown in  FIG. 1 .  FIG. 3  is a block diagram showing embodiments of filter and decimation units shown in  FIG. 2 . The frequency component processor unit  110  comprises a subband division unit  200 , a time-domain transform unit  210 , a filter unit  220 , a decimation unit  230 , an output-energy selection unit  240 , and a frequency-domain transform unit  250 .  
      Regarding the relationship between  FIG. 3  and filter  220  and decimation unit  230  of  FIG. 2 , Filtering/Decimation is the method using the band split or sub-band filter, and decimation refers to choosing one of the N samples.  FIG. 3  illustrates specific examples of the filter unit  220  and decimation unit  230  in  FIG. 2 . Therefore, the filter unit  220  includes a low-pass filter  300  and a high-pass filter  320 . The decimation unit  230  includes of a time-domain decimation unit  340  which reduces by half, in a time domain, the reference signal input from the low-pass filter  300 , and a time-domain decimation unit  360  which reduces by half, in a time domain, the references signal input from the high-pass filter  320 .  
      For example, the case of dividing the signal “X” into a low band signal and a high band signal is explained. Low-pass filter unit  300  receives and low-pass filters the signal “X”, and high-pass filter unit  320  receives and high-pass filters the signal “X”. Time-domain decimation unit  340  receives the low-pass filtered samples, chooses and odd numbered sample, and performs decimation. Time-domain decimation unit  360  receives high-pass filtered samples, chooses an even numbered sample, and performs decimation.  
      The subband division unit  200  divides an audio signal, which is input from the time-frequency transform unit  100  and transformed into a frequency, into subbands.  
      The time-domain transform unit  210  transforms the audio signal, which is divided into subbands, into a time-domain audio signal corresponding to each of the subbands.  
      The filter unit  220  filters the time-domain audio signal input from the time-domain transform unit  210 . The filter unit  220  includes of a lowpass filter  300  and a highpass filter  320 . The lowpass filter  300  extracts a reference signal composed of low frequency components from the time-domain audio signal, and the highpass filter  320  extracts a detailed signal composed of high frequency components from the time-domain audio signal.  
      The time-domain audio signal, which is filtered in the filter unit  220 , is decimated by a predetermined range in the decimation unit  230 . The decimation unit  230  includes of a time-domain decimation unit  340 , which reduces by half in a time domain the reference signal input from the lowpass filter  300 , and a time-domain decimation unit  360 , which reduces by half in a time domain the reference signal input from the highpass filter  320 . While the example in  FIG. 3  illustrates a time-domain signal reduced by half in a time domain, the decimation range of a time-domain signal may be differently set.  
      The output-energy selection unit  240  determines which one has higher output energy among the reference and detailed signals which are reduced in a time domain being the decimation unit  230 . That is, the output-energy selection unit  240  compares the output energy between the reference and detailed signals, and selects only a signal with higher output energy.  
      The frequency-domain transform unit  250  receives the selected time-domain audio signal from the output-energy selection unit  240 , and transforms the received signal into a frequency-domain audio signal.  
      Accordingly, it is possible to reduce frequency components since only any one of the reference and detailed signals is selected.  
       FIG. 4  is a block diagram showing another embodiment of the frequency component processor unit  110  shown in  FIG. 1 .  
      A subband division unit  400  divides an audio signal, which is input from the time-frequency transform unit  100  and transformed into a frequency, into subbands.  
      A representative value extraction information unit  420  provides prior information on how to extract a representative value from each of the subbands divided in the subband division unit  400 . For instance, a representative value is selected for five frequency components in each of the subbands, and there is provided representative value extraction information on whether a maximum value is to be determined to be the representative value.  
      A representative value extracting unit  440  receives each of the subbands signals divided in the subband division unit  400  and information on the representative value extraction from the representative value extraction information unit  420 , and extracts only a representative value corresponding to the information. Accordingly, it is possible to reduce frequency components since only a frequency component corresponding to a certain representative value is selected from each of the subbands.  
      While the example in  FIG. 4  illustrates an audio signal, the frequency component processor unit  110  can handle a data signal including an image signal as well as an audio signal using the aforementioned embodiments.  
       FIG. 5  is a flowchart showing an operation of a low bitrate audio encoding system according to an aspect of the present invention shown in  FIG. 1 .  
      In operation  500 , a time-domain audio signal input is transformed into a frequency-domain audio signal. In operation  510 , a portion of frequency components is reduced in a frequency-domain audio signal. That is, N frequency coefficients are transformed into N′ frequency coefficients in the frequency-domain audio signal, where N′ is less than N. In operation  520 , encoding bit allocation information is calculated using a psychoacoustic model. In operation  530 , reduced frequency components are quantized according to the encoding bit allocation information. In operation  540 , the quantized audio signal is encoded.  
       FIG. 6  is a flowchart showing an example of operation  510  shown in  FIG. 5 .  
      In operation  600 , the frequency-domain audio signal input through operation  500  is divided into subbands.  
      In operation  610 , the audio signal divided into subbands is transformed into a time-domain audio signal corresponding to each of the subbands.  
      In operation  620 , the time-domain audio signal is filtered into two signal components. The two signal components refer to a reference signal, which is composed of low frequency components extracted from the time-domain audio signal using a lowpass filter, and a detailed signal, which is composed of high frequency components extracted from the time-domain audio signal using a highpass filter.  
      In operation  630 , each of the time-domain audio signal components divided in operation  620  is decimated by a predetermined range. For instance, as shown in the decimation unit  230  of  FIG. 3 , the reference signal input from the lowpass filter is reduced by half in a time domain, and a detailed signal input from the highpass filter is reduced by half in a time-domain. While the present example illustrates the reference and detailed signals reduced by half in a time domain, the decimation range of the reference and detailed signals may be set differently.  
      In operation  640 , it is determined which one has higher output energy among the reference and detailed signals which are reduced in a time domain in operation  630 . That is, the output energy is compared between the reference and detailed signals, and only a signal with a higher output energy is selected.  
      In operation  650 , the selected time-domain audio signal is received and transformed into a frequency-domain audio signal. That is, only any one of the reference and detailed signals is selected and transformed into a frequency-domain signal, whereby frequency components of a first input audio signal can be reduced.  
       FIG. 7  is a flowchart showing another example of operation  510  shown in  FIG. 5 .  
      In operation  700 , a frequency-domain audio signal input through operation  500  is divided into subbands.  
      In operation  720 , information on how to extract a representative value from each of the subbands divided in operation  700  is retrieved. For instance, a representative value is selected for five frequency components in each of the subbands, and there is provided representative value extraction information on whether a maximum value is to be determined to be the representative value.  
      In operation  740 , each of the subbands signals divided in operation  700  and information on the representative value extraction in operation  720  are received, and only a representative value corresponding to the information is extracted. Accordingly, it is possible to reduce frequency components since only a frequency component corresponding to a certain representative value is selected from each of the subbands.  
      While the example in  FIG. 7  illustrates an audio signal, the frequency component processor unit  110  can handle a data signal including an image signal as well as an audio signal by using the aforementioned embodiments.  
       FIGS. 8A through 8D  show an example of signal variations based on frequency signal processing in an embodiment of a low bitrate audio encoding system according to the present invention.  
       FIG. 8A  shows a time-domain input audio signal,  FIG. 8B  shows an audio signal in a range of 2.5 to 5 kHz which is divided in the subband division unit  200  of a frequency component processor unit  110 ,  FIG. 8C  shows a reference signal divided in the filter unit  220  of the frequency component processor unit  110 , and  FIG. 8D  shows a detailed signal divided in the filter unit  220  of the frequency component processor unit  110 .  
      In  FIG. 8D , EL/(EL+EH)=0.70 means that the reference signal occupies 70% of the whole signals. That is, in this case, since the reference signal has a higher energy ratio than the detailed signal, the reference signal will be selected in the output-energy selection unit  230  of the frequency component processor unit  110 .  
       FIGS. 9A through 9D  show another example of signal variations based on frequency signal processing in an embodiment of a low bitrate audio encoding system according to the present invention.  
       FIG. 9A  shows a time-domain input audio signal,  FIG. 9B  shows an audio signal in a range of 5 to 10 kHz which is divided in the subband division unit  200  of a frequency component processor unit  110 ,  FIG. 9C  shows a reference signal divided in the filter unit  220  of the frequency component processor unit  110 , and  FIG. 9D  shows a detailed signal divided in the filter unit  220  of the frequency component processor unit  110 .  
      In  FIG. 9D , EL/(EL+EH)=0.80 means that the reference signal occupies 80% of the whole signals. That is, in this case, since the reference signal has a higher energy ratio than the detailed signal, the reference signal will be selected in the output-energy selection unit  230  of the frequency component processor unit  110 .  
       FIG. 10  is a block diagram showing an embodiment of a lossless audio decoding system according to the present invention. The system includes a lossless decoder unit  1000 , an inverse quantizer unit  1020 , a frequency component processor unit  1040 , and a frequency-time transform unit  1060 .  
      The lossless decoder unit  1000  performs a process reverse to that of the lossless encoder unit  130 . Accordingly, a received encoded bitstream is decoded, and the decoded audio signal is output to the inverse quantizer unit  1020 . That is, the lossless decoder unit  1000  decodes additional information, which includes the quantization step size and a bitrate allocated to each band, and the quantized data in a layered bitstream according to the order in which the layer is generated. The lossless decoder unit  1000  can decode signals using an arithmetic decoding or Huffman decoding algorithm.  
      The inverse quantizer unit  1020  recovers an original signal from the decoded quantization step size and quantized data.  
      The frequency component processor unit  1040  transforms N′ frequency coefficients, which were reduced in the frequency component processor unit  110  as described in  FIG. 1 , into the original N frequency coefficients through frequency component processing.  
      The frequency-time transform unit  1060  transforms the frequency-domain audio signal back into the time-domain signal to allow a user to play the audio signal.  
       FIG. 11  is a block diagram showing an embodiment of the frequency component processor unit  1040  shown in  FIG. 10 . The frequency component processor unit  1040  includes a subband division unit  1100 , a time-domain transform unit  1110 , an interpolation unit  1120 , a filter unit  1130 , and a frequency-domain transform unit  1140 .  
      The subband division unit  1100  divides an audio signal, which is input from the lossless decoder unit  1000  and transformed into a frequency, into subbands.  
      The time-domain transform unit  1110  transforms the audio signal, which is divided into subbands, into a time-domain audio signal corresponding to each of the subbands.  
      The interpolation unit  1120  receives the time-domain audio signal from the time-domain transform unit  1110 , and interpolates the signal, which is decimated by a predetermined range in the decimation unit  230  of  FIG. 2 , by the decimated range. For instance, since the decimation unit  230  in  FIG. 3  reduces the reference or detailed signal by half, the interpolation unit  1120  increases the time-domain signal by double. While the example in  FIG. 11  illustrates a time-domain signal interpolated by double, the interpolation range of a time-domain signal may be differently set. In addition, the interpolation unit  1120  may interpolate signals using additional information of an interpolation factor.  
      The filter unit  1130  detects whether the time-domain audio signal input from the interpolation unit  1120  is the reference signal composed of low frequency components within the time-domain audio signal in  FIG. 3 , or the detailed signal composed of high frequency components within the time-domain audio signal in  FIG. 3 . The filter unit  1130  detects using additional information whether it is a reference signal or a detailed signal.  
      The frequency-domain transform unit  1140  receives the reference or detailed signal from the filter unit  1130 , and transforms the input time-domain audio signal into a frequency-domain signal.  
       FIG. 12  is a block diagram showing another construction of the frequency component processor unit  1040  shown in  FIG. 10 . The frequency component processor unit  1040  comprises a subband division unit  1200 , a representative value extracting unit  1220 , and an interpolation unit  1240 .  
      The subband division unit  1200  divides an audio signal, which is input from the lossless decoder unit  1000  and transformed into a frequency, into subbands.  
      The representative value extracting unit  1220  extracts a representative value from an audio signal divided into subbands.  
      The interpolation unit  1240  receives a representative value from the representative value extracting unit  1220 , and interpolates frequency components into each of the subbands divided in the subband division unit  1200 . The interpolation unit  1240  performs an interpolating operation by using a predetermined parameter or additional information in a bitstream received from a low bitrate audio encoding system. Referring to the example in  FIG. 4 , in case of selecting a representative value for five frequency components in each of the subbands, four unselected frequency components in each of the subbands may be set to have the same value as the representative value. In addition, the four unselected frequency components may be interpolated differently depending on distances from the frequency component having the representative value. The representative value may be determined to be the maximum value or the mean value of the frequency components.  
      While the example in  FIG. 12  illustrates an audio signal, the frequency component processor unit  110  can handle a data signal including an image signal as well as an audio signal using the aforementioned embodiments.  
       FIG. 13  is a flowchart showing an operation of a lossless audio decoding system according to the present invention shown in  FIG. 10 .  
      Operation  1300  performs a process reverse to that of operation  540  of  FIG. 5 , where the quantized audio signal is losslessly encoded. Accordingly, a received encoded bitstream is decoded, and the decoded audio signal is output. That is, in operation  1300 , additional information, which includes the quantization step size and a bitrate allocated to each band, and the quantized data are decoded in a layered bitstream according to the order in which the layer is generated. In operation  1300 , a decoding process is performed using an arithmetic decoding or Huffman decoding algorithm.  
      In operation  1320 , an original signal is recovered from the decoded quantization step size and quantized data.  
      In operation  1340 , the inversely quantized signal is increased from N′ frequency coefficients reduced in operation  510  of  FIG. 5  to the original N frequency coefficients through frequency component processing.  
      In operation  1360 , the frequency-domain audio signal is transformed back into the time-domain signal to allow a user to play the audio signal.  
       FIG. 14  is a flowchart showing an example of operation  1340  shown in  FIG. 13 .  
      In operation  1400 , the frequency-domain audio signal input through operation  1300  is divided into subbands.  
      In operation  1410 , the audio signal divided into subbands is transformed into a time-domain audio signal corresponding to each of the subbands.  
      In operation  1420 , the time-domain audio signal is received, and the signal decimated by a predetermined range in operation  630  of  FIG. 6  is interpolated by the decimated range. The interpolation is performed using a parameter previously set in a low bitrate audio decoding system or additional information in a bitstream received from a low bitrate audio encoding system. For example, since the reference or detailed signal is reduced by half in  FIG. 6 , the time-domain signal is increased by double in operation  1420 . While the example in  FIG. 14  illustrates a time-domain signal interpolated by double, the interpolation range of a time-domain signal may be set differently. In addition, the interpolation may be performed using additional information of an interpolation factor in operation  1420 .  
      In operation  1430 , it is detected whether the time-domain audio signal input through operation  1420  is the reference signal composed of low frequency components within the time-domain audio signal, or the detailed signal composed of high frequency components within the time-domain audio signal. A reference signal or a detailed signal may be detected according to additional information.  
      In operation  1440 , the reference or detailed signal is input through operation  1430 , and the input time-domain audio signal is transformed into a frequency-domain signal.  
       FIG. 15  is a flowchart showing another example of operation  1340  shown in  FIG. 13 .  
      In operation  1500 , a frequency-domain audio signal input through operation  1300  of  FIG. 13  is divided into subbands.  
      In operation  1520 , a representative value is extracted from the audio signal divided into subbands.  
      In operation  1540 , frequency components are interpolated into each of the subbands divided in operation  1500  using a representative value input through operation  1520 . Referring to the example in  FIG. 4 , in case of selecting a representative value for five frequency components in each of the subbands, four unselected frequency components in each of the subbands may be set to have the same value as the representative value. In addition, the four unselected frequency components may be interpolated differently depending on distances from the frequency component having the representative value. The representative value may be determined to be the maximum value or the mean value of frequency components.  
      While the example in  FIG. 15  illustrates an audio signal, the frequency component processor unit  110  can handle a data signal including an image signal as well as an audio signal by using the aforementioned embodiments.  
       FIGS. 16A and 16B  show an example of an audio signal for a predetermined subband in an encoding operation and in a decoding operation, respectively.  
      According to an aspect of the present invention,  FIG. 16A  shows an audio signal in a range of 2.5 to 5 kHz in an encoding operation, and  FIG. 16B  shows an audio signal in a range of 2.5 to 5 kHz in a decoding operation.  
       FIGS. 17A and 17B  show another example of an audio signal for a predetermined subband in an encoding operation and in a decoding operation, respectively.  
       FIG. 17A  shows an audio signal in a range of 5 to 10 kHz in an encoding operation, and  FIG. 17B  shows an audio signal in a range of 5 to 10 kHz in a decoding operation.  
      The present invention can be recorded on computer-readable recording media with computer-readable codes. Examples of the computer include all kinds of apparatuses with an information processing function. Examples of the computer-readable recording media include all kinds of recording devices for storing computer-readable data, such as ROM, RAM, CD-ROM, magnetic tape, floppy disk, optical data storage system, etc.  
      While the present invention has been described with reference to exemplary embodiments thereof, it will be understood by those skilled in the art that various changes in form and details may be made therein without departing from the scope of the present invention as defined by the following claims.  
      There is provided a low-bitrate encoding/decoding method and system. According to the present invention, it is possible to efficiently compress data at a low bitrate and thus provide a high quality audio signal in storing and recovering audio signals in a variety of audio systems, such as Digital Audio Broadcasting (DAB), internet phone, and Audio on Demand (AOD), and multimedia systems including software. In addition, it is possible to provide an encoding/decoding method and system that can efficiently compress data signals including image signals as well as audio signals.  
      Although a few embodiments of the present invention have been shown and described, it would be appreciated by those skilled in the art that changes may be made in these embodiments without departing from the principles and spirit of the invention, the scope of which is defined in the claims and their equivalents.