Adaptive digital audio encoding system

An encoding system employing a novel perceptual spectrum difference estimation device improves the coding efficiency and audio quality of a digitized audio signal. The system comprises M number of encoding means arranged in parallel for encoding the input digital audio signal of a current frame, respectively; M number of decoding means arranged in parallel for decoding each of the encoded digital audio signals; a first estimator for estimating a power density spectrum for a difference signal between the input digital audio signal and each of the decoded digital audio signals; a second estimator for estimating a power density spectrum for the input digital audio signal of the current frame and for determining a masking threshold therefor based on the power density spectrum for the input digital audio signal; a third estimator for estimating a set of perceptual spectrum distances based on the power density spectrum for each of the difference signals and the masking threshold; and a circuit for selecting an encoded digital audio signal having a smallest perceptual spectrum distance.

FIELD OF THE INVENTION 
The present invention relates to a digital audio encoding system; and, more 
particularly, to an improved digital audio encoding system capable of 
providing an encoded audio signal with a minimum distortion measured in 
accordance with the human auditory perception. 
DESCRIPTION OF THE PRIOR ART 
Transmission of digitized audio signals makes it possible to deliver high 
quality audio signals comparable to those of standard compact disc/or 
digital audio tape. When an audio signal is expressed in a digital form, a 
substantial amount of data need be transmitted especially in the case of 
high definition television system. Since, however, the available frequency 
bandwidth assigned to such audio signals is limited, in order to transmit 
the substantial amounts of digital data, e.g., 768 Kbits per second for 16 
bit PCM(Pulse Code Modulation) audio signal with 48 KHz sampling 
frequency, through the limited audio bandwidth of, e.g., about 128 KHz, it 
is inevitable to compress the audio signal. At the receiving end of the 
digital transmission, the compressed audio signal is decoded. 
The quality of the decoded audio signal is largely dictated by the 
compression technique employed for the encoding thereof. Sometimes, in 
order to selectively generate an audio signal with a least distortion, the 
digital audio encoding system is provided with a plurality of encoders 
employing different compression techniques, a corresponding number of 
decoders and an audio distortion measuring device. In such a case, the 
encoders are arranged in a parallel fashion in order to carry out the 
encoding of the input digital audio signal simultaneously; and each of the 
decoders is coupled to its corresponding encoder for the decoding of the 
encoded digital audio signal therefrom. In such an arrangement, the 
digital audio encoding system selectively generates one of the encoded 
digital audio signals which causes a least audio distortion. 
Audio distortions are usually measured in terms of "Total Harmonic 
Distortion(THD)" and "Signal to Noise Ratios(SNR)", wherein said THD is a 
RMS(root-mean-square) sum of all the individual harmonic-distortion 
components and/or IMD's(Intermodulation Distortions) which consist of sum 
and difference products generated when two or more signals pass through an 
encoder; and said SNR represents the ratio between the amplitude of an 
input digital signal and the amplitude of an error signal. 
Such THD or SNR measurement, however, is a physical value which has no 
direct bearing on the human auditory faculty or and, accordingly, the 
conventional digital audio encoding system having such audio distortion 
measuring device has a limited ability to provide an encoded digital audio 
signal which best reflects the human auditory perception. 
SUMMARY OF THE INVENTION 
It is, a primary object of the invention to provide a novel digital audio 
encoding system for adaptively encoding an input digital audio signal 
closely matching the human auditory perception. 
In accordance with the present invention, there is provided a novel system 
for encoding an input digital audio signal having a plurality of frames, 
which comprises: M number of encoding means arranged in parallel for 
encoding the input digital audio signal in a current frame, respectively; 
M number of decoding means arranged in parallel for decoding each of the 
encoded digital audio signals, respectively; first estimation means for 
estimating a power density spectrum of a difference signal between the 
input digital audio signal and each of the decoded digital audio signals; 
second estimation means for estimating a power density spectrum of the 
input digital audio signal in the current frame and for determining a 
masking threshold thereof based on the power density spectrum of the input 
digital audio signal; third estimation means for estimating a set of 
perceptual spectrum distances based on the power density spectrum for each 
of the difference signals and the frequency masking threshold; and means 
for selecting an encoded digital audio signal having a smallest perceptual 
spectrum distance.

DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENTS 
Referring to FIG. 1, there is shown a block diagram illustrating a digital 
audio encoding system 100 in accordance with the present invention. 
The encoding system 100 comprises an encoding device 10, a decoding device 
20, a first and a second power density spectrum estimation units 30 and 
34, a masking threshold estimation unit 40, a perceptual spectrum distance 
estimator 50, a comparator 60, a selector 70 and a formatting circuit 80. 
An input digital audio signal x(n,i) of an ith frame, or a current frame, 
which includes N samples, i.e., n=0, 1, 2, . . ,N-1, is applied to the 
encoding device 10 which is adapted to perform an encoding operation of 
the input digital audio signal at a predetermined bit rate, wherein N is a 
positive integer and one frame includes L, e.g., 32, subbands. A "frame" 
used herein denotes a part of the digital audio signal which corresponds 
to a fixed number of audio samples and is a processing unit for the 
encoding and decoding of the digital audio signal. 
As shown, the encoding device 10 includes a plurality of encoders, e.g., 
two encoders 10A and 10B, which are coupled in a parallel manner in order 
to simultaneously receive the input digital audio signal of the current 
frame and encode the input digital audio signal by using various 
compression techniques. For instance, the encoder 10A may carry out an 
encoding operation of the input digital audio signal of the ith frame by 
employing an intra-frame bit allocation technique which adaptively assigns 
bits to each subband included within one frame based on a perceptual 
entropy for each of the subbands therein, and the encoder 10B may perform 
an encoding operation of the input digital audio signal by using an 
inter-frame bit allocation technique which adaptively assigns bits to each 
frame included within a predetermined group of frames based on a 
perceptual entropy for each frame; and, alternatively, the encoders 10A 
and 10B may include non-uniform and uniform quantizers, respectively. 
The perceptual entropy PE(i) for the ith frame, as is well known in the 
art, may be represented as: 
##EQU1## 
wherein m is a subband index with m=0,1, . . . ,L-1, L being the total 
number of subbands in a frame; P(m), a sound pressure level in subband m 
estimated from a Fast Fourier Transform(FFT) technique; and M(m), a 
masking threshold in subband m. 
The encoded digital audio signal from each of the encoders is applied to 
the selector 70 and the decoding device 20 which includes a plurality of 
decoders, e.g., 20A and 20B. Each of the decoders is adapted to decode a 
corresponding encoded digital audio signal from the encoders. The decoded 
digital audio signals y1(n,i) and y2(n,i) from the decoders 20A and 20B 
are applied to the first and second power density spectrum estimation 
units 30 and 34, respectively, wherein each of said power density spectrum 
estimation units includes a subtractor 31(35) and a power density spectrum 
estimator 32(36), respectively. The subtractor 31 included in the first 
power density spectrum estimation unit 30 generates a difference signal 
e1(n,i) representative of the difference between the input digital audio 
signal x(n,i) to the system and the decoded digital audio signal y1(n,i) 
from the decoder 20A, which may be represented as: 
EQU e1(n, i)=x(n,i)-y1(n,i) Eq. (2) 
wherein both x(n,i) and y1(n,i) are P(e.g., 16)-bit pulse code 
modulation(PCM) audio signals. 
Subsequently, the difference signal is provided to the power density 
spectrum estimator 32 which serves to carry out the Fast Fourier Transform 
conversion thereof from the time domain to the frequency domain. 
Turning now to FIG. 2, the power density spectrum estimator 32 includes a 
windowing circuit 32A and a Fast Fourior Transform(FFT) circuit 32B. 
The windowing circuit 32A receives the difference signal e1(n,i) from the 
subtracter 31; and performs the windowing process by multiplying the 
difference signal e1(n,i) with a predetermined hanning window. The 
predetermined harming window h(n) may be represented as: 
##EQU2## 
wherein N is a positive integer and n=0, 1, 2, . . , N-1. 
Accordingly, the output w1(n,i) from the windowing circuit 32A may be 
represented as: 
EQU w1(n,i)=e1(n,i).multidot.h(n) Eq. (4) 
wherein i is a frame index and n has the same meaning as previously 
defined. 
The output w1(n,i) from the windowing circuit 32A is then provided to the 
FFT circuit 32B which estimates the power density spectrum thereof; and, 
as a preferred embodiment of the present invention, includes a 512 point 
FFT for Psychoacoustic Model I[or MPEG(moving pictures expert group)-Audio 
Layer I]. Accordingly, the power density spectrum E1(k,i) for the 
difference signal e1(n,i) of the ith frame, as is well known in the art, 
may be calculated as follows: 
##EQU3## 
wherein k=0, 1, . . ,(N/2)-1, N and n have the same meanings as previously 
defined. 
Referring back to FIG. 1, the second power density spectrum unit 34 is 
substantially identical to the first power density spectrum unit 30 
excepting that the power density spectrum E2(k,i) for a difference signal 
e2(n,i) representative of the difference between the input digital audio 
signal x(n,i) and the decoded digital audio signal y2(n,i) from the 
decoder 20B is calculated therein. The difference signal e2(n,i) from the 
subtractor 35 may be represented as: 
EQU e2(n,i)=x(n,i)-y2(n,i) Eq. (6) 
wherein n and i have the same meanings as previously defined. 
Therefore, it should be appreciated that the power density spectrum E2(k,i) 
for the difference signal e2(n,i) can be obtained by windowing the 
difference signal e2(n,i) with the hanning window h(n) as is done for the 
difference signal e1(n,i) in Eq.(4). Said power density spectrum E2(k,i) 
of the difference signal e2(n,i) for the ith frame may be obtained as: 
##EQU4## 
wherein N, n, k, and i have the same meanings as previously defined, with 
w2(n,i)=e2(n,i).multidot.h(n). 
In the meanwhile, the masking threshold estimation unit 40 is adapted to 
receive the input digital audio signal x(n,i) of the ith frame and to 
estimate the masking threshold thereof. The masking threshold estimation 
unit 40 includes a power density spectrum estimator 41 and a masking 
threshold estimator 42. The power density spectrum estimator 41 is 
substantially identical to the power density spectrum estimator included 
in the first or second power density spectrum estimation unit excepting 
that the power density spectrum X(k,i) of the input digital audio signal 
x(n,i) for the ith frame is calculated therein. The power density spectrum 
X(k,i) of the input digital audio signal x(n,i) for the ith frame may be 
obtained as: 
##EQU5## 
wherein N, n, k, and i have the same meanings as previously defined, with 
w(n,i)=x(n,i).multidot.h(n). 
The power density spectrum of the input digital audio signal, X(k,i), 
estimated at the power density spectrum estimator 41 is then provided to 
the masking threshold estimator 42 which serves to estimate a masking 
threshold depending on the power density spectrum of the input digital 
audio signal. 
The masking threshold represents an audible limit closely reflecting the 
human auditory perception, which is a sum of the intrinsic audible limit 
or threshold of a sound and an increment caused by the presence of 
another(masking) contemporary sound in the frequency domain, as described 
in an article, which is incorporated herein by reference, entitled "Coding 
of Moving Pictures and Associated Audio", ISO/IEC/JTC1/SC29/WG11 NO501 
MPEG 93(Jul., 1993), wherein the so-called Psychoacoustic Models I and II 
are discussed for the calculation of the masking threshold. In a preferred 
embodiment of the present invention, Psychoacoustic Model I is 
advantageously employed in the masking threshold estimator 42. 
The power density spectrums E1(k,i) and E2(k,i) and the masking threshold 
M(k,i) are simultaneously provided to the perceptual spectrum distance 
estimator 50 which is adapted to derive first and second perceptual 
spectrum distances PSD1(i) and PSD2(i) representative of the audio 
distortions for the power density spectrums E1(k,i) and E2(k,i) as 
perceived by the human auditory faculty with the masking effect taken into 
consideration. The first perceptual spectrum distance PSD1(i) for the 
power density spectrum E1(k,i) from the power density spectrum estimator 
32 may be represented as: 
##EQU6## 
wherein k and i are the same as previously defined. 
Similarly, the second perceptual spectrum distance PSD2(i) for the power 
density spectrum E2(k,i) from the power density spectrum estimator 36 may 
be defined as: 
##EQU7## 
wherein k and i are the same as previously defined. 
As can be seen from Eqs.(9) and (10), the perceptual spectrum distance for 
the ith frame is estimated by the power density spectrum of the difference 
signal which exceeds the masking threshold. The first and second 
perceptual spectrum distances PSD1(i) and PSD2(i) are applied to the 
comparator 60 which serves to generate a selection signal identifying a 
least distorted digital audio signal among the two encoded digital audio 
signals from the encoders, e.g., 10A and 10B, by comparing their 
perceptual spectrum distances. The selection signal from the comparator 60 
is then provided to the selector 70 and the formatting circuit 80. 
In response to the selection signal from the comparator 60, the selector 70 
selects the least distorted digital audio signal among the encoded digital 
audio signals from the encoders to thereby provide the selected audio 
signal to the formatting circuit 80. 
At the formatting circuit 80, the selection signal from the comparator 60 
and the selected audio signal from the selector 70 formatted and 
transmitted to a transmitter (not shown) for the transmission thereof. 
While the present invention has been shown and described with reference to 
the particular embodiments, it will be apparent to those skilled in the 
art that many changes and modifications may be made without departing from 
the spirit and scope of the invention as defined in the appended claim.