Data processing: speech signal processing – linguistics – language – Audio signal bandwidth compression or expansion
Reexamination Certificate
2000-08-04
2004-09-28
Dorvil, Richemond (Department: 2654)
Data processing: speech signal processing, linguistics, language
Audio signal bandwidth compression or expansion
C704S200100, C704S229000
Reexamination Certificate
active
06799164
ABSTRACT:
BACKGROUND OF THE INVENTION
1. Field of the Invention
The present invention relates to a digital acoustic signal coding apparatus, a method of coding a digital acoustic signal, and a recording medium for recording a program of coding the digital acoustic signal, in particular, the compression/coding of a digital acoustic signal utilized in, for instance, DVD recording/reproducing or in a digital broadcast, etc.
2. Discussion of the Background
The background arts are discussed with the main focus being on the compression of an acoustic signal.
At present, in the digital audio field, MP3 is a very popular coding technique MP3 is an abbreviation for an acoustic signal compression coding method called “MPEG-1 Audio Layer III”. By employing MP3, digital audio such as data used for a CD, can be compressed to the extent of 1/11 without deteriorating the sound quality. Because of the convenience of compressing a large volume of acoustic data to a compact size that can be transmitted in a short time, MP3 is becoming popular for transmissions on the internet. At present, reproducing apparatuses suitable for use with MP3 are being introduced by several manufacturing companies, and some music distributing businesses are being operated using it.
On the other hand, even in the field of digital broadcasting, in accordance with the development of digitalization, the adoption of sound signal (acoustic signal) compressing technology has become common. At present, the method of MPEG-2 Audio BC is employed in CS broadcasting. Furthermore, the method of MPEG-2 Audio AAC is scheduled to be employed in BS and digital broadcasting of the wave on the ground, both to be started in 2000 or in subsequent years.
The above-mentioned matters relate to technology belonging to international standard of acoustic signal compression all called “MPEG Audio”. In addition to MPEG Audio, for instance, acoustic signal compressing methods such as Dolby Digital (AC-3) and ATRAC, are used for DVD and MD.
As stated above, the use of compression/coding technology for digital audio has become more and more familiar day by day. The fundamental technology of acoustic signal compressing and the recent trends thereof are described hereinafter.
In acoustic signal compressing, acoustic signals are largely classified into “voice sound” and “musical sound” categories. Here, voice sound signifies the human voice and musical sound signifies not only the human voice, but also any general acoustic signal including music, life sound, natural sound, etc. The reason why sound has to be classified is that the object and utilized technology of the coding differs for each.
In voice sound coding, the human voice signal of low sampling rate of about 8-16 KHz is compressed for use with a low bit rate such as over a telephone circuit. On the other hand, in musical sound coding an acoustic signal of a high sampling rate of about 32-96 KHz is compressed keeping the sound quality as high as possible. In the former method, deterioration of sound quality cannot be avoided compared with the original sound, while, in the latter method, sound compression which fundamentally does not degrade the sound can be accomplished. Both of MP3 and AAC are included in the latter coding (musical sound coding).
The compressing of digital information is classified into two methods; reversible compression and non-reversible compression. In the former method, the original signal can be faithfully reproduced at the time of decoding. However, in the latter method, the distortion of the signal generally occurs. For performing acoustic signal compression coding, both of those methods are suitably combined. First, the reversible compression method is described.
Here, Huffman code employed also in MPEG Audio as the representative reversible compression method is described. Huffman coding is a method in which short code and long code are respectively allocated to a large frequency value and a small frequency value in accordance with an appearance frequency of the original signal value, and the signal is compressed such that the entire code value is made as small as possible. A code which is not of constant length is called a variable-length code, while a code of constant length for all values is called a fixed-length code. The original signal of acoustic compression is a fixed-length code represented by a number of bits of the respective constant digital sample values (16 bits in the case of CD).
FIG. 21
shows an example of such a fixed-length code and a Huffman code, and
FIG. 28
shows an example of allocating such a code to an actual numerical value row utilizing the above-mentioned two codes. As shown in
FIG. 21
, in order to discriminate six sorts of different original signal value with the fixed-length code, it is necessary to allocate at least a 3 bit code to the respective values.
On the other hand, as is apparent from the numerical value row as shown in
FIG. 28
, if the appearance frequency of “2” is largest (e.g., 7 times) and the appearance frequencies of“1” and “5” are smallest (e.g., once), here, regarding the Huffman code shown in
FIG. 21
, a 2-bit code is allocated to “2” and a 4-bit code is allocated to “1” and “5”. Regarding the other remaining values, a code of a length corresponding to the respective appearance frequencies is allocated thereto.
An important property of a Huffman code is that the original signal row can be decoded to have one meaning. In the example of
FIG. 21
, if the Huffman code row is “00110”, the original signal row is “20”. Since there is a one-meaning property for decoding, Huffman coding is reversible.
For reference, an example of a code not capable of decoding to have one meaning is also shown in FIG.
21
. In this example, when the code row “000001” is received, it is impossible to distinguish which of the meanings of the original signal (“25”, “13”, or “223”) was intended. Moreover, since a method of constructing a code capable of decoding with one meaning has been already shown, the description thereof is omitted here.
Now, in the case of allocating a fixed-length code, such as shown in
FIG. 21
, to a numerical value row, such as shown in (a) of
FIG. 28
, the code row becomes the one shown in (b) of
FIG. 28
, and the entire code amount turns out to be 3×20=60 bits. On the other hand, in the case of allocating a Huffman code, as also shown in
FIG. 21
, to a numerical value row shown in (a) of
FIG. 28
, the code row becomes one such as shown in (c) of
FIG. 28
, and the entire code amount turns out to be smaller (46 bits). In such a way, the entire code amount is further reduced in the case of allocating a Huffman code as compared with the case of a fixed-length code. Namely, when Huffman code is employed, the original signal value can be faithfully reproduced with a smaller code amount as compared with a fixed-length code. However, there is a limitation as to the compression factor, e.g., almost 77% as to the upper limit. Accordingly, it is impossible to expect a high compression factor, e.g., 1/11 in such a situation as mentioned above. Therefore, the technology of non-reversible compression is inevitably required. The basic quantization technology therefor is described hereinafter.
The use of quantization signifies the use of a method of classifying the level of the original signal value into plural steps and causing the values representing the respective levels to correspond to a restoring value (decoded) value. The above-mentioned method is described with reference to the example of FIG.
22
.
Here, it is assumed that the original signal value is distributed as an integer between 0 and 60. When the value is converted to the fixed-length code as it is with a binary number, the respective value has to be expressed with bits. In this example, the original signal value is quantized to 6 levels and caused to correspond to the respective restored (decoded) values as shown in FIG.
22
.
At the time of coding, the original signal value is divided by “10” and the decimal fraction part is removed (cut down). The
Dorvil Richemond
Storm Donald L.
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