Encoding apparatus and method, recording medium, and decoding apparatus and method

BACKGROUND OF THE INVENTION

1. Field of the Invention

The present invention relates to an encoding apparatus and method, adapted to encode a second code string which can be encoded with a higher efficiency than that with which a first code string can be encoded.

2. Description of the Related Art

The technique to record information to a recording medium capable of recording an encoded audio or speech signal, such as a magneto-optical disc or the like, is widely used. For a highly efficient coding of an audio or speech signal, there have been proposed various methods such as the subband coding method (SBC) in which an audio signal or the like on a time base is divided into a plurality of frequency bands without blocking, and the so-called transform coding method in which a signal on the time base is transformed to a one on the frequency base (spectrum transform), divided into a plurality of frequency bands and then the signal in each of the frequency bands is encoded. Also, a high efficiency coding method has also been proposed which is a combination of the SBC method and transform coding method. In this third one, for example, after an audio or speech signal is divided into a plurality of frequency bands by the SBC method, the signal in each frequency band is spectrum-transformed to a signal on the frequency base, and the signal is encoded in each spectrum-transformed frequency band. The QMF filter for example is used in this coding method. The QMF filter is defined in R. E. Crochiere: Digital Coding of Speech in Subbands, Bell Syst. Tech. Journal, Vol. 55, No. 8, 1976”. Also, the method for equal-bandwidth division by filter is defined in “Joseph H. Rothweiler: Polyphase Quadrature Filters—A New subband Cording Technique, ICASSP 83, BOSTON”.

In an example of the above-mentioned spectrum, an input audio signal is blocked at predetermined unit times (encoding frames), and each of the blocks is subjected to the discrete Fourier transform (DFT), discrete cosine transform (DCT) or modified discrete cosine transform (MDCT) to transform a time base to a frequency base. The MDCT is described in “J. P. Princen and A. B. Bradley, Univ. of Surrey Royal Melbourne Insit. of Tech.: Subband/Transform Coding Using Filter Bank Designs Based on Time Domain Aliasing Cancellation, ICASSP, 1987”.

When the above-mentioned DFT or DCT is used for of a waveform signal to a spectrum, with a time block consisting of M samples will yield a number M of independent real data. Normally, a time block is arranged to overlap Ml samples thereof its neighboring blocks each to suppress the distortion of the connection between time blocks. Therefore, in the DFT and DCT, signal will be encoded by quantizing on average M real data for a number (M−M

1

) of samples.

When the MDCT is used as the method for of a waveform signal to a spectrum, M independent real data can be obtained from 2M samples arranged to overlap M ones thereof its neighboring blocks each. Therefore, in the MDCT, signal is encoded by quantizing on average M real data for the M samples. In a decoder, waveform elements obtained from a code resulted from the MDCT by inverse transform in each block are added together while being made to interfere with each other, thereby permitting to reconstruct the waveform signal.

Generally, by increasing the length of the time block, the frequency separation of the spectrum is increased and energy is concentrated on a specific spectrum component. Therefore, by transforming a waveform signal to a spectrum with an increased block length obtained by overlapping a time block a half thereof its neighboring time blocks each and using the MDCT in which the number of spectrum signals obtained will not increase relative to the number of original time samples, it will be possible to enable a coding whose efficiency is higher than that attainable with the DFT or DCT.

By quantizing a signal divided into plurality of frequency bands by the filtering or spectrum as in the above, it is possible to control any frequency band where quantization noise occurs and encode an audio signal with a higher efficiency in the auditory sense using a property such as the masking effect. Also, by normalizing, for each of the frequency bands, the audio signal with a maximum absolute value of a signal component in the frequency band before effecting the quantization, a further higher efficiency of the coding can be attained.

The width of frequency division for quantization of each frequency component resulted from a frequency band division is selected with the auditory characteristic of the human being for example taken in consideration. That is, an audio signal is divided into a plurality of frequency bands (25 bands for example) in such a bandwidth as will be larger as its frequency band is higher, which is generally called “critical band”, as the case may be. Also, at this time, data in each band is encoded by a bit distribution to each band or with an adaptive bit allocation to each band. For example, when a coefficient data obtained using the MDCT is encoded with the above bit allocation, an MDCT coefficient data in each band, obtained using the MDCT at each block, will be encoded with an adaptively allocated number of bits. The of the adaptive bit allocation information can be determined so as to be previously included in a code string, whereby the sound quality can be improved by improving the coding method even after determining a format for decoding. The known bit allocation techniques include the following two:

One of them is disclosed in “R. Zelinski and P. Noll: Adaptive Transform Coding of Speech Signals, IEEE Transactions of Acoustics, Speech, and Signal Processing, Vol. ASSP-25, No. 4, August 1977”. This technique is such that the bit allocation is made based on the size of a signal in each frequency band. With this technique, the quantization noise spectrum can be flat an the noise energy be minimum, but since no masking effect is used, the actual noise will not feel auditorily optimum.

The other one is disclosed in “M. A. Kransner, MIT: The Critical Band Coder—Digital encoding of the perceptual requirements of the auditory system, ICASSP, 1980”. This technique is such that the auditory masking is used to acquire a necessary signal-to-noise ratio for each frequency band, thus making a fixed bit allocation. With this technique, however, since the bit allocation is a fixed one, the signal characteristic will not be so good even when it is measured on a sine wave input.

To solve the above problem, there has been proposed a high efficiency encoder in which all bits usable for the bit allocation are divided for a fixed bit allocation pattern predetermined for each small block and for a bit distribution dependent upon a signal size of each block at a ratio dependent upon a signal related with an input signal and whose number of bits for the fixed bit allocation pattern is larger as the spectrum of the signal is smoother.

With the above method adopted in the encoder, the entire signal-to-noise ratio can considerably be improved by allocating more bits to a block including a specific spectrum to which energy is concentrated, such as a sine wave input. Generally, since the human ears are extremely sensitive to a signal having a steep spectrum component, the above method can be used to improve the signal-to-noise ratio, which does not only improve a measured value but also can effectively improve the sound quality.

The bit allocation methods include many other ones as well. The auditory model is further elaborated to enable a higher-efficiency coding if the encoder could. Generally, in these methods, a reference for the real bit allocation to realize a computed signal-to-noise ratio with a highest possible fidelity is determined and an integral value approximate to the computed value is taken as a number of allocated bits.

For example, the Application of the present invention has proposed an encoding method in which a signal component having an auditorily important tone component, namely, a signal component having an energy concentrated around a predetermined frequency thereof, is separated from a spectrum signal and encoded separately from the other spectrum component. Thus, this method allows to encode an audio signal or the like efficiently with a high compression rate with little auditory deterioration.

To form an actual code string, it suffices to first encode quantizing precision information and normalizing coefficient information with a predetermined number of bits for each frequency band in which the normalization and quantization are effected, and then encode the normalized and quantized signals. Also, in the ISO/IEC 11172-3: 1998 (E), 1993, a high efficiency coding method is defined in which the number of bits indicating quantizing precision information varies from one frequency band to another in such a manner that as the frequency is higher, the number of bits indicating quantizing precision information will be smaller.

It has also been proposed to determine quantizing precision information based on normalizing coefficient information for example in a decoder instead of directly encoding the quantizing precision information. In this method, however, since the relation between the normalizing efficient information and quantizing precision information will be determined when a format is set, so it is not possible to introduce the control of the precision of quantization based on a further advanced auditory model which will be available in future if any. Also, when a compression rate to be realized ranges wide, it is necessary to determine the relation between the normalizing coefficient information and quantizing precision information for each compression rate.

Also, there is known an encoding method in which a quantized spectrum signal is encoded using a variable-length code defined in “D. A. Huffman: A Method for Construction of Minimum Redundancy Codes, Proc. I. R. E, 40, p. 1098 (1952)” for example with a higher efficiency.

As in the above, techniques for a higher-efficiency coding have been developed one after another. By employing a format incorporating a newly developed technique, it is possible to record for a longer time, and also record an audio signal having a higher sound quality for the same length of recording time.

However, if players capable of playing back only signals recorded in a predetermined format (will be referred to as “first format” hereinafter) prevail (this player will be referred to as “first format-conforming player” hereinafter), the first format-conforming players will not be able to read a recording medium in which signals are recorded in a format using a higher-efficiency coding method (this format will be referred to as “second format” hereinafter). More specifically, even if the recording medium has a flag indicating a format when the first format is determined, the first format-conforming player adapted to read a signal with no disregard for the flag signal will read signals from the recording medium taking that all signals in the recording medium have been recorded in the first format. Therefore, all the first format-conforming players will not recognize that signals in the recording medium have been recorded in the second format if applicable.

Thus, if the first format-conforming player plays back a signal recorded in the second format taking that the signal has been recorded in the first format, a terrible noise will possibly occur.

To avoid the above, the Applicant of the present invention has also applied for patent an improved method for recording data in a so-called TOC area, in which when a music piece is recorded by the second format codec, the first format-conforming player will actually play back a warning message recorded in nay other area than the TOC area by the first format codec.

However, the above method proposed by the Applicant needs that an ambient spare area in the TOC area in the first format and is not advantageous in that the playback by a second format-conforming player is complicated.

OBJECT AND SUMMARY OF THE INVENTION

It is therefore an object of the present invention to overcome the above-mentioned drawbacks of the prior art by providing an encoding apparatus and method, which needs no ambient spare area in the TOC area and in which the playback by a second format-conforming player is not complicated.

The above object can be attained by providing an encoder including according to the present invention:

a first encoding means for generating a first code string by encoding a warning message signal or silent signal;

a second encoding means for generating, when the first encoding means is encoding a silent signal, a second code string by encoding an input signal; and

means for generating a synthetic code string by combining the first and second code strings together.

Also the above object can be attained by providing an encoding method including according to the present invention:

a first encoding step of generating a first code string by encoding a warning message signal or silent signal;

a second encoding step of generating, when the first encoding means is encoding a silent signal, a second code string by encoding an input signal; and

a step of generating a synthetic code string by combining the first and second code strings together.

Also the above object can be attained by providing a recording medium for recording a synthetic signal generated by combining a first code string and second code string, in which the first code string is generated by encoding a warning message or silent signal while the second code string is generated by encoding an input signal when the first code string is a silent signal encoded.

Also the above object can be attained by providing a decoder including according to the present invention:

means for receiving a code string synthesized by combining a code string encoded by a first encoding means and a code string encoded by a second encoding means;

means for detecting a predetermined bit pattern in the first code string; and

means for decoding the second code string;

the second code string decoding means providing a predetermined sound when the predetermined bit pattern has not been detected by the bit pattern detecting means.

Also the above object can be attained by providing a decoding method including, according to the present invention, steps of:

receiving a code string synthesized by combining a code string encoded by a first encoding and a code string encoded by a second encoding;

means for detecting a predetermined bit pattern in the first code string; and

means for decoding the second code string;

at the second code string decoding step, there being provided a predetermined sound when the predetermined bit pattern has not been detected at the bit pattern detecting step.

Also the above object can be attained by providing a decoder including according to the present invention:

means for receiving a code string synthesized by recording, in a predetermined-length encoding frame, a first code string from the top of the encoding frame and a second code string from the bottom of the encoding frame; and

means for decoding the second code string recorded from the bottom of the encoding frame.

Also the above object can be attained by providing a decoding method including, according to the present invention, steps of:

receiving a code string synthesized by recording, in a predetermined-length encoding frame, a first code string from the top of the encoding frame and a second code string from the bottom of the encoding frame; and

decoding the second code string recorded from the bottom of the encoding frame.

These objects and other objects, features and advantages of the present intention will become more apparent from the following detailed description of the preferred embodiments of the present invention when taken in conjunction with the accompanying drawings.

BRIEF DESCRIPTION OF THE DRAWINGS

FIG. 1

is a block diagram of a preferred embodiment of the encoder according to the present invention;

FIG. 2

is a block diagram of a first conventional encoder to encode an input signal based on a first coding method;

FIG. 3

is a block diagram of a transform block forming the first conventional encoder;

FIG. 4

is a block diagram of a signal component encode block forming the first conventional encoder;

FIG. 5

explains a first coding method which is adopted in the first conventional encoder shown in

FIG. 2

;

FIG. 6

shows in detail a code string which will be when a signal encoded by the first encoder is recorded into a recording medium;

FIG. 7

explains a code string of a music piece formed from a sequence of encoding frames generated by the first conventional encoder, and TOC area;

FIG. 8

is a block diagram of a signal component encode block forming together with the transform block the second codec encode block shown in

FIG. 1

;

FIG. 9

explains a spectrum the signal component encode block shown in

FIG. 8

is to encode;

FIG. 10

shows in detail a code string which will be when a signal encoded by the second coding method is recorded into the recording medium;

FIG. 11

explains a first method adopted in the encoder shown in

FIG. 1

;

FIG. 12

explains a second method adopted in the encoder shown in

FIG. 1

;

FIG. 13

shows in detail a recording the synthetic code string shown in

FIGS. 11 and 12

into the recording medium;

FIG. 14

shows in detail an encoding frame data consisting of code strings generated by an encoder according to another embodiment of the present invention;

FIG. 15

explains another code string recording method implemented using the code string shown in

FIG. 14

;

FIG. 16

is a block diagram of a encoder to generate another code string shown in

FIG. 15

;

FIG. 17

is a block diagram of a decoder to read an acoustic signal from a recording medium having recorded therein the code string shown in

FIG. 13

;

FIG. 18

is a flow chart of operations effected in playback of an acoustic signal by a selective silencer forming a part of the decoder in

FIG. 17

;

FIG. 19

is a block diagram of a conventional decoder corresponding to the encoder shown in

FIG. 2

;

FIG. 20

is a block diagram of an inverse transform block forming a part of the conventional decoder shown in

FIG. 19

;

FIG. 21

is a block diagram of a signal component decode block forming a part of the conventional decoder shown in

FIG. 19

;

FIG. 22

is a block diagram of essential parts of the decoder for decoding a tone component separated and encoded by the encoder shown in

FIG. 8

;

FIG. 23

is a block diagram of a recorder and/or player to which the conventional encoder and decoder, or the encoder and decoder according to the present invention can be applied;

FIG. 24

is a block diagram of an information processor in which the encoding method according to the present invention is employed; and

FIG. 25

is a flow chart of operations effected in execution of a coding program by the information processor in FIG.

21

.

DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENTS

Referring first to

FIG. 1

, there is illustrated in the form of block diagram the preferred embodiment of the encoder according to the present invention. For a silent playback without generation of a noise even when a first format-conforming player reads a recording medium having recorded therein a second code string conforming to a second format based on a second coding method which will further be described and having been encoded with a higher efficiency than a first code string conforming to a first format based on a first coding method which will further be described later, the encoder shown in

FIG. 1

prevents the user from considering the recording medium to have no sound recorded therein, which would be possible because of the silent playback. Note that the first format is an existing old format while the second format is a new format upper-compatible with the first format.

Therefore, as shown in

FIG. 1

, the encoder includes a first codec-based warning message generator

151

to generate a warning message by the first codec, a first codec-based silent fixed pattern generator

152

to generate a first codec-based silent fixed pattern, a second codec encode block

154

to encode an input signal

153

by a second codec, and a code string generator

155

to generate a synthetic code using string

156

by combining outputs from the above components in an encoding frame having a predetermined length being a unit of encoding. The encoder includes a controller

150

to control the above encoder components as well.

Note that the “codec” generally means “code-decode” but it will be used herein in each of the encoding and decoding methods to mean intra-codec encoding and intra-codec decoding, respectively.

The encoder builds a music piece from a warning message part and music piece part, each formed from a plurality of the above encoding frames. In this encoder, the first codec-based warning message generator

151

is controlled by the controller

150

to generate a first codec-based warning message “this music piece has been recorded by second codec” which will be recorded in the leading part of each music piece, and sends it to the code string generator

155

. Also, under the control of the controller

150

, the first codec-based silent fixed pattern generator

152

generates a first codec-based silent fixed pattern which will be recorded in the top part of the encoding frame of the music piece part, and sends it to the code string generator

155

. The second codec encode block

154

encodes a PCM input signal

153

of a music piece by the second codec, and sends it to the code string generator

155

. The code string generator

155

combines the warning message, silent fixed pattern and second codec-encoded data for each encoding frame to generate a synthetic code string

156

.

The first codec is originally a kind of high-efficiency coding for compression. The first codec encodes an input signal such as audio PCM signal or the like with a high efficiency using the subband coding (SBC), adaptive transform coding (ATC) and adaptive bit allocation.

Referring now to

FIG. 2

, there is illustrated in the form of a block diagram a first conventional encoder to encode an input signal based on the first codec. The signal supplied at an input terminal

40

is transformed by a transformer

41

to signal frequency components, and each of the components is encoded by a signal component encode block

42

. A code string generator

43

generates a code string which will be delivered at an output terminal

44

.

Referring now to

FIG. 3

, there is illustrated in the form of a block diagram the transformer

41

forming the first conventional encoder. As shown, in the transformer

41

in the first conventional encoder, a signal divided by a subband filter

46

into two frequency bands is transformed by forward spectrum transformers

47

and

48

such as MDCT to spectrum signal components in the respective frequency bands. The bandwidth of the spectrum signal components from the forward spectrum transformers

47

and

48

is a half of the bandwidth of the input signal, namely, it is halved. Of course, the transformer

41

may be any other one selected from many transformers. For example, the input signal may be transformed by the MDCT directly to spectrum signal components. Otherwise, it may be transformed by the DFT or DCT in place of the MDCT to spectrum signal components. Also it is possible to divide the input signal by the so-called subband filter into frequency band components. In this embodiment, however, it will be convenient to transform an input signal to frequency components by the spectrum transform by which it is made possible to obtain many frequency components with a relatively small number of operations.

Referring now to

FIG. 4

, there is illustrated in the form of a block diagram the signal component encode block

42

in FIG.

2

. As shown, each signal component supplied from an input terminal

51

is normalized by a normalizer

52

for each predetermined frequency band, and then quantized by a quantizer

54

based on a quantizing precision data calculated by a quantizing precision determination block

53

. The quantizer

54

provides quantized signal components and normalizing coefficient information and quantizing precision information. These outputs are delivered at an output terminal

55

.

Referring now to

FIG. 5

, there is illustrated a first conventional coding method adopted in the first conventional encoder shown in FIG.

2

. The spectrum signal has been provided from the transformer

41

shown in FIG.

3

. In

FIG. 5

, the absolute value of the spectrum signal from the MDCT is transformed to a level (dB). The input signal is transformed to 64 spectrum signals each for a predetermined time block (encoding frame). The spectrum signals are grouped in 8 bands from U

1

to U

8

(each will be referred to as “encoding unit” hereinafter), and they are normalized and quantized for each encoding unit. By varying the quantizing precision for each encoding unit depending upon how the frequency components are distributed, the deterioration of sound quality can be minimized for an auditorily high efficiency of encoding. If any spectrum signal in the encoding unit has not to be encoded actually, the encoding unit may be allocated zero bit to make silent the signal in the frequency band corresponding to the encoding unit.

Referring now to

FIG. 6

, there is illustrated in detail a code string which will be when a signal encoded by the first encode block is recorded into a recording medium. In this example, each of the encoding frames F

0

, F

1

, . . . has disposed at the top thereon a fixed-length header

80

in which a sync signal

81

and a number of encoding units

82

are recorded. In the code string, the header

80

is followed by quantizing precision data

83

for the number of encoding units

82

, and the quantizing precision data

83

is followed by normalizing coefficient data

84

for the number of encoding units

82

. Normalized and quantized spectrum coefficient data

85

follows the normalizing coefficient data

84

. In case each of the encoding frames F

0

, F

1

, . . . has a fixed length, a blank area

86

may be provided following the spectrum coefficient data

85

.

Referring now to

FIG. 7

, there is illustrated a code string of a music piece formed from a sequence of encoding frames F

0

, F

1

, . . . generated by the first conventional encoder, and a TOC area

201

. The code string and TOC area

201

are recorded in a recording medium. As shown in

FIG. 7

, a signal recording area

202

includes areas

202

1

,

202

2

and

203

2

. Each of the areas

202

1

to

202

3

has recorded therein a code string of a music piece formed from the sequence of encoding frames F

0

, F

1

, . . . The TOC area

201

has recorded therein information on which portion each music piece starts at or similar information, which makes it possible to know where the leading end and trailing end of each music piece exist. More specifically, the TOC area

201

has recorded therein a first music piece information address A

1

, second music piece information address A

2

, third music piece information address A

3

, . . . The first music piece information address A

1

includes a first music piece start address A

1

S, music piece end address A

1

E, music piece encoding mode M

1

and reserved information R

1

recorded in the area

202

1

. Similarly, the second music piece information address A

2

includes a second music piece start address A

2

S, music piece end address A

2

E, music piece encoding mode M

2

and reserved information R

2

recorded in the area

202

2

. Note that the music piece encoding mode is for example the compress coding mode such as ATC.

The first coding method having been described in the foregoing can further be improved in efficiency of coding. For example, a relatively small code length is assigned to ones of the quantized spectrum signals that appear frequently while a relative large code length is assigned to ones of the quantized spectrum signals that appear less frequently, thereby permitting to improve the efficiency of coding. Also, when the transform block length is increased, sub information such as quantizing precision information and normalizing coefficient information can relatively be reduced in amount and the frequency resolution can be raised, so that the quantizing precision on the frequency base can be controlled more elaborately. The efficiency of coding can thus be improved.

Moreover, the Applicant of the present invention has also applied for patent an encoding method in which a signal component having a special auditory importance, that is, a signal component having energy concentrated around a predetermined frequency thereof, is separated from a spectrum signal and it is encoded separately from other spectrum components. This method permits to encode an audio signal efficiently at a high compression rate with little auditory deterioration. It should be noted that this embodiment adopts this encoding method as the second codec.

The second codec encode block

151

shown in

FIG. 1

is supplied with a PCM input signal via an input terminal

130

and generates, using the second codec, a second codec-based code string. It should be noted however that he second codec encode block

154

has the functions of both the transformer

41

and signal component encode block

42

shown in FIG.

2

.

The signal component encode block

42

forming along with the transformer

41

the second codec encode block

154

in

FIG. 1

is constructed as shown in FIG.

8

. As shown, the output of the transformer

41

shown in

FIG. 2

is supplied to a tone component separator

91

via an input terminal

90

. The tone component separator

91

separates the transformed output of the transformer

41

into a tone component and non-tone component and supplies them to a tone component encode block

92

and non-tone component encode block

93

, respectively. The tone component encode block

92

and non-tone component encode block

93

are constructed similarly to the encode block shown in FIG.

4

and encode the tone component and non-tone component, respectively. The tone component encode block

92

encodes position data of the tone component as well.

The spectrum to be encoded by the signal component encode block

42

will be described below with reference to FIG.

9

. Also in

FIG. 9

, the absolute spectrum value of the MDCT is transformed to a level (dB). An input signal is transformed to sixty four spectrum signals for each predetermined time block (encoding frame). The 64 spectrum signals are grouped into eight encoding units from U

1

to U

8

, and normalized and quantized for each encoding unit. Note that although the description is made herein concerning the 64 spectrum signals for the simplicity of the illustration and explanation, 128 pieces of spectrum data can be provided if the transform length is set double that in the example shown in FIG.

5

. The difference from that in

FIG. 5

is that a high-level one is separated as a tone component Ti from the spectrum signals and encoded. For example, for three tone components T

1

, T

2

and T

3

, their respective position data P

1

, P

2

and P

3

are also required. However, spectrum signals from which the tone components T

1

, T

2

and T

3

have been extracted can be quantized with less bits. This method can conveniently be adopted for a signal including a special spectrum signal to which energy is concentrated, thereby permitting to attain a high efficiency of encoding.

Referring now to

FIG. 10

, there is illustrated in detail a specific example of a code string which will be when a signal encoded by the second coding method is recorded into a recording medium. In this example, a tone code string

110

is recorded between a header

121

and quantizing precision data

124

in a code string

120

generated by the second coding method to separate tone components from each other. The code string

120

generated by the second coding method is a one having recorded therein a second format header

121

including a sync signal

122

, number of encoding units

123

, etc., the second header

121

being followed by the tone code string

110

, quantizing precision data

124

, normalizing coefficient data

125

, spectrum coefficient data

126

, etc. in this order. The tone code string

110

has first recorded therein a number of tone components

111

, the latter being followed by data on each tone component

112

0

, more specifically, position data

113

, quantizing precision data

114

, normalizing coefficient data

115

and spectrum coefficient data

116

. Further in this example, the length of transform block to be transformed to spectrum signals is set double that in the example based on the first coding method shown in

FIG. 6

to raise the frequency resolution, and in addition, a variable-length code is introduced to record, in the encoding frames F

0

, F

1

, . . . , of the same number of bytes as that in the example in

FIG. 6

, a code string of an acoustic signal having a length two times larger than that in the example in FIG.

6

.

The embodiment of the encoder according to the present invention shown in

FIG. 1

is intended to prevent a terrible noise from occurring when a recording medium having information recorded in the code string shown in

FIG. 10

is played in a player capable of reading only a recording medium having information recorded in the code string shown in

FIG. 6

, and also prevents the user from considering the recording medium to have no sound recorded therein, which would be possible because of the silent playback.

First, for a silent playback by prevention of a noise from occurring, the encoder shown in

FIG. 1

uses the first coding method to record, as shown in

FIG. 11

, a silent signal in the first format, and the second coding method to record the second code string in a blank area in the second format enabling a high efficiency, thereby implementing a long recording time. More specifically, the first format header

80

and zero bit-allocated quantizing precision data

83

are generated by a first codec-based silent fixed pattern generator

152

. Namely, when the quantizing precision data

83

is allocated zero, no bit may be allocated to the spectrum coefficient data

85

in FIG.

6

. Thus, the normalizing coefficient data

84

shown in

FIG. 11

is followed by a blank area

87

. A second code string generated by the second coding method is embedded in the blank area

87

. Thus, a relatively wide recording area can be assured for the second coding method, and even if the second code string is played back by the first format-conforming player, no noise will occur.

Further, there is a method by which a further wide recording area can be assured for the second coding method while preventing noise from occurring when the second code string is played back by the first formed-conforming player, thereby permitting to implement a higher sound quality. This method is shown in FIG.

12

. As shown, the quantizing precision data

83

of all the encoding units, defined by the number of encoding units

82

written in the first format header

80

, is set zero while the code string

120

generated by the second coding method is recorded in a blank area

88

immediately after the quantizing precision data

83

. More specifically, 4 bytes is allocated to the first format header

80

, a total of 10 bytes (80 bits) for 20 encoding units, in which one quantizing precision can be expressed with 4 bits, is allocated to the quantizing precision data

83

, and 198 bytes is allocated to the blank area

88

. Thus 212 bytes can be allocated to one encoding frame. Actually, different values will be set for the first format-conforming normalizing coefficient data but since the quantizing precision data are set all to zero, so it will be interpreted that all the spectrum data are zero for the first coding method. Eventually, when the code string data shown in

FIG. 12

is played back by the first format-conforming player, no sound is played back and thus no terrible noise will take place. With the number of encoding units being set to a minimum one allowable by the first format, a wide recording area can be assured for the second codec and the top position of the second codec can be fixed.

The first format-conforming player can play back a music piece part consisting of the plurality of encoding formats formed from the synthetic code strings shown in

FIGS. 11 and 12

silently with no noise.

Further, the above encoder records, by the first codec, the warning message “this music piece has been recorded by second codec” in the leading part of each music piece as having previously been described, to avoid the user's confusion.

FIG. 13

shows a code string encoded by the encoder. In this example, a warning message encoded by the first codec is recorded in a part before each music piece part (warning message part)

300

, and then a first codec-based silent fixed pattern

302

and data encoded by the second codec and recorded are recorded in each encoding frame

303

of the music piece part

301

.

Thus, following the message “this music piece has been recorded by second codec”, the first format-conforming player makes a silent playback, thus preventing the user of the first format-conforming player from being confused.

On the other hand, when the first codec-based silent fixed pattern is recorded, the second format-conforming player decodes the second codec-based code string. Also, when the first codec-based silent fixed pattern is not recorded, the second format-conforming player will make a silent playback. More specifically, the second format-conforming player will read a recording medium having recorded therein a code string shown in

FIG. 14

to make a brief silent playback at the stop of a music piece, and then play back a music piece encoded by the second codec. This second format-conforming player will further be described later.

FIG. 14

shows in detail encoding frame data consisting of code strings generated by the encoder according to another embodiment of the present invention. In this embodiment, since in each of the coding frames F

0

, F

1

, . . . , the second codec-based code strings are recorded in an opposite order to that in which the first codec-based code strings are recorded, each of the codec-based code strings can be read out independently. Since the silent data in both the first and second codec-based code strings can be made compact in size, the first codec-based sound signal code string and second codec-based silent signal code string, and a second code-based silent signal code string and second code-based silent signal code string, are recorded dually, the sound quality of the sound signal can be assured to be sufficiently high. In this embodiment, the second format-conforming player should always only decode each encoding frame from its trailing end. Thus, since the first codec-based code string may not be checked to see if it has a silent fixed patter, the operation may conveniently be simplified. Note that by setting the quantized precision data

83

all to zero, the normalizing coefficient data

84

and spectrum coefficient data

85

may be partially added to the recording area of the second codec.

FIG. 15

shows another code string recording method implemented by the use of the code string shown in FIG.

14

. When playing back an encoding frame

306

of a warning message part

305

, the first format-conforming player will play back a warning message “this music piece has been recorded by the second codec” recorded by the first codec. Thereafter, in a music piece part

308

, a first codec-based silent fixed pattern

310

in an encoding frame

309

is read and silently played back. On the contrary, since the second format-conforming player will play back a second codec-based code string by decoding each encoding frame from its trailing end, the first codec-based silent fixed pattern may not be checked.

FIG. 16

shows the construction of the encoder to generate the other code. This encoder is different from the encoder in

FIG. 1

in that it is provided with a second codec-based silent signal generator

157

. That is, when recording a second codec-based code string whose recorded order is opposite to that of the first codec-based code string in each frame, the encoder shown in

FIG. 16

will generate a second codec-based silent fixed pattern

307

as shown in

FIG. 15

by the second codec-based silent signal generator

157

.

Next, the embodiment of the decoder according to the present invention will be described. Referring now to

FIG. 17

, there is illustrated in the form of a block diagram a decoder to read an acoustic signal from a recording medium having recorded therein the code string shown in FIG.

13

. In the decoder, a code string decomposer

136

sends to a first codec-based dummy string inspector

137

a silent fixed pattern portion of a code string shown in

FIG. 13

, supplied via an input terminal

135

, corresponding to the first format header

80

and first codec-based quantizing precision data

83

, whose position and length in the encoding frame are fixed, while sending to a second codec decode block

138

other second codec-based code string portion of the code string. The first codec-based dummy string inspector

137

will check whether the received code string has a silent fixed pattern consisting of a first format header and zero bit-allocated quantizing precision data. If it is determined that the code string received by the first codec-based dummy string inspector

137

has the silent fixed pattern, a selective silencer

139

will provide an acoustic signal provided from the second codec decode block

138

. When it is determined that the received code string has not the silent fixed pattern, the code string is taken as an invalid one and a silent playback is done.

Referring now to

FIG. 18

, there is shown a flow chart of operations effected when the selective silencer

139

plays back an acoustic signal based on the result of the inspection by the first codec-based dummy string inspector

137

as in the above. At step S

21

, it is judged whether the first codec-based part is the silent fixed pattern. If the result of the judgment is NO, the operation goes to step S

22

where silent data is provided as an output. On the contrary, if the judgment result is YES, the operation goes to step S

23

where a decoded data generated by decoding the second codec-based data is provided as an output.

The conventional decoder corresponding to the encoder shown in

FIG. 2

is provided to generate an acoustic signal from the code string generated by the encoder in FIG.

2

. As shown in

FIG. 19

, it supplies a code string provided at an input terminal

60

to a code string decomposer

61

which in turn will extract a code of each signal component. Then, after each signal component is restored from the code by a signal component decode block

62

, an inverse transform block

63

provides an acoustic waveform signal as an output.

Referring now to

FIG. 20

, there is illustrated in the form of a block diagram the inverse transform block

63

forming the conventional decoder shown in FIG.

19

. The transform block

63

corresponds to the specific example of the transform block shown in

FIG. 3. A

signal component supplied from input terminals

65

and

66

is transformed by inverse spectrum transform blocks

67

and

68

to signals of various frequency bands. These signals are combined by a band synthesis filter

69

and then delivered at an output terminal

70

.

Referring now to

FIG. 21

, there is illustrated in the form of a block diagram the signal component decode block

62

forming the decoder in FIG.

19

. An output signal from the code string decomposer

61

is supplied to a dequantizer

72

via an input terminal

71

where it will in turn be dequantized, and then it is de-normalized by a de-normalizer

73

to a spectrum signal which is delivered at an output terminal

74

.

FIG. 22

is a block diagram of the essential parts of the decoder to decode a signal whose tone component has been separated and encoded by the encoder shown in FIG.

8

. The decoder itself is constructed similarly to that shown in FIG.

19

. The signal component decode block

62

in

FIG. 16

is constructed as in FIG.

22

. Namely, a tone component in a code string decomposed by the code string decomposer

61

is supplied from an input terminal

96

to a tone component decode block

98

while a non-tone component is supplied from an input terminal

97

to a non-tone component decode block

99

. The tone component decode block

98

and non-tone component decode block

99

decode the tone and non-tone components, respectively, and supply their outputs to a spectrum signal synthesizer

100

. A synthetic spectrum signal generated by the spectrum signal synthesizer

100

is delivered at an output terminal

101

.

The encoder shown in FIG.

2

and decoder shown in

FIG. 19

are employed in a recorder and/or player shown in

FIG. 23

for example. The recorder and/or player is intended to write a first code string encoded by the first encoder and conforming to the first format to a recording medium and also read only that first code string. Thus, since the recorder and/or player will read a second code string conforming to the second format and supplied from the second encoder from a recording medium as a code string encoded by the first encode block, a terrible noise will take place. To avoid this, data in a code string shown in

FIG. 13

or

15

, encoded by the encoder according to the present invention, will be effectively written to or read from such a recorder and/or player.

First, the construction of the recorder and/or player will be described below:

A recording medium used in this recorder and/or player is a magneto-optical disc

1

driven to rotate by a spindle motor

11

. For write of data to the magneto-optical disc

1

, a modulated field corresponding to the to-be-written data is applied to the disc

1

by a magnetic head

14

while a laser light is being irradiated to the disc

1

from an optical head

13

. That is, a magnetic field modulated recording is effected to write the data to the magneto-optical disc

1

along the recording track thereon. Also, to read data from the magneto-optical disc

1

, the recording track on the disc

1

is traced with a laser light by the optical head

13

to magneto-optically read the data from the disc

1

.

The optical head

13

includes for example a laser source such as a laser diode or the like, optical parts such as a collimator lens, objective lens, polarizing beam splitter, cylindrical lens, etc., a photodetector having a predetermined pattern of photosensors, etc. The optical head

13

is provided opposite to the magnetic head

14

with the magneto-optical disc

1

placed between them. For writing data to the magneto-optical disc

1

, a head drive circuit

26

in a recording system which will further be described later drives the magnetic head

14

to apply a modulated magnetic field corresponding to the to-be-written data while driving the optical head

14

to irradiate a laser light to a destination track on the magneto-optical disc

1

, thereby effecting a thermoelectric recording by the magnetic field modulating method. Also, the optical head

13

detects a return light of the laser light irradiated to the destination track to detect a focus error by the so-called astigmatic method for example and also a tracking error by the so-called pushpull method for example. To rad data from the magneto-optical disc

1

, the optical head

13

detects the focus error and tracking error while detecting a difference in the polarized angle (Kerr rotation angle) of the return light of the laser light from the destination track to generate a reading signal.

The output of the optical head

13

is supplied to an RF circuit

15

. The RF circuit

15

extracts the focus error signal and tracking error signal from the output of the optical head

13

and supplies them to a servo control circuit

16

while binarizing the reading signal and supplying it to a decoder

31

in a playback system which will further be described later.

The servo control circuit

16

consists of, for example, a focus servo control circuit, tracking servo control circuit, spindle motor servo control circuit, sled servo control circuit, etc. The focus servo control circuit controls the focus of the optical system of the optical head

13

so that the focus error signal will be zero. The tracking servo control circuit controls the tracking of the optical system of the optical head

13

for the tracking error signal to become zero. Further, the spindle motor servo control circuit controls the spindle motor

11

to rotate the magneto-optical disc

1

at a predetermined speed (at a constant linear velocity, for example). Further, the sled servo control circuit moves the optical head

13

and magnetic head

14

to a destination track position on the magneto-optical disc

1

, designated by a system controller

17

. The servo control circuit

16

providing such control operations sends information indicative of the operating status of each of the components controlled thereby to the system controller

17

.

The system controller

17

has a key input control unit

18

and display unit

19

connected thereto. The system controller

17

is supplied with operation input information from the key input control unit

18

to control the recording and playback systems according to the information. Also the system controller

17

manages the write position and read position on the recording track, traced by the optical head

13

and magnetic head

14

, respectively, based on address information in sectors, read as a header time and sub-code Q data from the recording track on the magneto-optical disc

1

. Moreover the system controller

17

controls the display unit

19

to display a read time based on the data compression rate of the recorder and/or player and information on the read position on the recording track.

For the read time, an actual time information is determined by multiplying the address information in sectors (absolute time information) read as the so-called header time and so-called sub-code Q data read from the recording track on the magneto-optical disc

1

by the reciprocal of the data compression rate (for example, “4” when the compression rate is ¼), and it is displayed on the display unit

19

. Note that also during data write, in case an absolute time information is previously recorded in the recording track on the magneto-optical disc (preformatted) for example, the preformatted absolute time information is read and multiplied by the data compression rate, whereby the present position can be displayed as an actual write time.

Next, in the recording system of the disc recorder/player, an analog audio input signal AIN from an input terminal

20

is supplied to an A/D converter

22

via a lowpass filter

21

, and it is quantized by the A/D converter

22

. A digital audio signal from the A/D converter

22

is supplied to an ATC (adaptive transform coding) encoder

23

being a specific example of the encoder shown in

FIG. 2. A

digital audio input signal DIN from an input terminal

27

is also supplied to the ATC encoder

23

via a digital input interface circuit

28

. The ATC encoder

23

subjects a digital audio PCM data to be transferred at a predetermined rate, generated by quantizing the input signal AIN by the A/D converter

22

, to a bit compression (data compression) based on a predetermined data compression rate. The compressed data (ATC data) from the ATC encoder

23

is supplied to a memory

24

. Concerning a data compression rate being ⅛ for example, the data transfer rate is reduced to ⅛ (9.375 sectors/sec) of the data transfer rate (75 sectors/sec) of data in the standard CD-DA format.

The memory

24

is used as a buffer memory to and from which data write and read are controlled by the system controller

17

to provisionally store the ATC data supplied from the ATC encoder

23

and write data to the disc as necessary. More specifically, when the data compression rate is ⅛ for example, compressed audio data supplied from the ATC encoder

23

is transferred at a rate reduced to ⅛ (9.375 sectors/sec) of the transfer rate (75 sectors/sec) of data in the standard CD-DA format. The compressed audio data is continuously written into the memory

24

. The compressed data (ATC data) can be written in every 8 sectors. However, since such data write in every 8 sectors is almost impossible in practice, data write is made in successive sectors as will be described later.

The data write is made at a burst at the same transfer rate (75 sectors/sec) as that of data in the standard CD-DA format taking as a recording unit a cluster of a predetermined plurality of sectors (32 sectors+a few sectors, for example) with a pause between sectors. More specifically, ATC audio data written successively at a rate as slow as 9.375 (=75/8) sectors/sec corresponding to the bit compression rate and compressed at a rate of ⅛ is read, as data to be written to the disc, from the memory

24

at a burst at the transfer rate of 75 sectors/sec. The read data to be written to the disc is transferred at a rate as slow as 9.375 sectors/sec including the write pause, while the rate of momentary data transfer within a time of the writing operation effected at a burst is the standard 75 sectors/sec. Therefore, when the disc rotating speed is the same as the transfer rate of data in the standard CD-DA format (constant linear velocity), data will be written at the same recording density and in the same storage pattern as those of data in the CD-DA format.

The ATC data, or data to be written to the magneto-optical disc, having continuously been read out from the memory

24

at a burst at the transfer rate (momentary rate) of 75 sectors/sec, is supplied to an encoder

25

. In data supplied from the memory

24

to the encoder

25

, the unit continuously written per write operation includes a cluster containing a plurality of sectors (e.g., 32 sectors) and a few sectors disposed before and after the cluster to connect clusters to each other. The cluster connecting sectors are set longer than the interleave length in the encoder

25

and not to influence the data in the other clusters when interleft between the clusters.

The encoder

25

subjects the to-be-written data supplied at a burst from the memory

24

as in the above to an encoding process for error correction (parity addition and interleaving), EFM encoding process, etc. The to-be-written data encoded by the encoder

25

is supplied to a magnetic head drive circuit

26

. The magnetic head drive circuit

26

has the magnetic head

14

connected thereto, and drives the magnetic head

14

to apply a modulated magnetic field corresponding to the to-be-written data to the magneto-optical disc

1

.

The system controller

17

provides the above-mentioned control of the memory

24

and also controls the write position in such a manner that the to-be-written data read at a bust from the memory

24

under the above control is continuously written to the recording tack on the magneto-optical disc

1

. The write position control is effected by the system controller

17

managing the write position for the to-be-written data read at a burst from the memory

24

and supplying the servo control circuit

16

with a control signal designating the write position on the recording track on the magneto-optical disc

1

.

Next, the playback system will be described. The playback system is destined to read data continuously written on the recording track on the magneto-optical disc

1

by the aforementioned recording system. It includes a decoder

31

which is supplied with a read output acquired by tracing the recording track on the magneto-optical disc

1

with a laser light from the optical head

13

and then binarized by the RF circuit

15

. At this time, it is possible to read not only the magneto-optical disc but a read-only optical disc similar to a compact disc.

The decoder

31

is provided correspondingly to the encoder

25

included in the aforementioned recording system. It subjects the read output binarized by the RF circuit

15

to the above-mentioned decoding process for error correction and EFM decoding process to play back the ATC audio data having been compressed at a rate of ⅛ at the transfer rate of 75 sectors/sec faster than the normal transfer rate. The read data provided from the decoder

31

is supplied to a memory

32

.

The memory

32

is controlled by the system controller

17

concerning the data write and read. The read data supplied at the transfer rate of 75 sectors/sec from the decoder

31

is written into the memory

32

at a burst at the transfer rate of 75 sectors/sec. Also, from the memory

32

, the read data written once into the memory

32

at the transfer rate of 75 sectors/sec is continuously read out at the transfer rate of 9.375 sectors/sec corresponding to the data compression rate of ⅛.

The system controller

17

writes the read data into the memory

32

at the transfer rate of 75 sectors/sec, and controls the memory

32

for continuous read of the read data from the memory

32

at the transfer rate of 9.375 sectors/sec. Also, the system controller

17

provides the above-mentioned control of the memory

32

and also controls the read position in such a manner that the read data written at a bust into the memory

32

under the above control is continuously read from the recording tack on the magneto-optical disc

1

. The read position control is effected by the system controller

17

managing the read position for the read data written at a burst into the memory

32

and supplying the servo control circuit

16

with a control signal designating the read position on the recording track on the magneto-optical disc or optical disc

1

.

The ATC audio data provided as the data continuously read from the memory

32

at the transfer rate of 9.375 sectors/sec is supplied to an ATC decoder

33

that is the decoder shown in FIG.

5

. The ATC decoder

33

is provided correspondingly to the ATC encoder

23

in the recording system. It plays back 16-bit digital audio data by expanding (bit expansion) 8 times for example. Digital audio data from the ATC decoder

33

is supplied to a D/A converter

34

.

The D/A converter

34

converts the digital audio data supplied from the ATC decoder

33

to an analog signal to generate an analog audio signal AOUT. The analog audio signal AOUT provided from the D/A converter

34

is delivered at an output terminal

36

via a lowpass filter

35

.

By having the recorder and/or player constructed and operative as having been described in the foregoing play a magneto-optical disc having recorded therein the code strings shown in

FIGS. 13 and 15

, noise can be prevented from taking place. This is because the ATC decoder

33

in the playback system of the recorder and/or player recognizes as a silent data the second one, generated by the second coding method, of the code strings shown in

FIGS. 13 and 15

. Also, when such a magneto-optical disc is played by a player capable of reading only the first codec-based data, the warning message will be played back, whereby it is possible to prevent the user from considering the recording medium to have no sound recorded therein, which would be possible because of the silent playback.

Also, the ATC decoder

33

included in the playback system of the recorder and/or player has the function of the decoder shown in FIG.

17

. For example, when it is determined by reading the TOC area for example that the magneto-optical disc having recorded therein the code strings shown in

FIGS. 13 and 15

is loaded in the recorder and/or player, it is possible to provide an acoustic signal by the above-mentioned operations. When the code string is judged to be invalid as the second code string, silent playback can be done.

Further, the ATC encoder

23

provided in the recording system of the recorder and/or player has the function of the encoder shown in

FIG. 1

, the recorder and/or player can generate the code strings shown in

FIGS. 13 and 15

by encoding at the time of reading, and also read them.

Another embodiment of the encoding method according to the present invention will be illustrated and described. The information processor executes a program based on the encoding method. It records in an internal recording medium thereof or downloads via a removable recording medium such as a floppy disc an encoding program to which the encoding method is applied, and executes the encoding program by a CPU included therein. Namely, the information processor functions as the aforementioned encoder.

The information processor is generally indicated with a reference

300

. It will be described in detail with reference to FIG.

24

. It has a CPU (central processing unit)

320

having connected thereto via a bus

340

a ROM

310

, RAM

330

, communications interface (I/F)

380

, driver

370

and an HDD

350

. The driver

370

drives a removable recording medium

360

such as a PC card, CD-ROM or floppy disc (FD).

The ROM

310

has stored therein an IPL (initial program loading) program and the like. According to the IPL program stored in the ROM

310

, the CPU

320

executes an OS (operating system) program stored in the HDD

350

, and further executes a data exchange program stored in the HDD

350

for example under the control of the OS program. The RAM

330

stores provisionally programs and data necessary for the operations of the CPU

320

. The communications interface

380

is provided for communications with external devices.

The encoding program is taken out from the HDD

350

for example by the CPU

320

and executed in the RAM

330

as a work area by the CPU

320

which will effect the operations shown in the flow chart in FIG.

25

.

As shown in

FIG. 25

, it is made sure at step S

1

if a portion being processed is a warning message portion. When the check result is YES, a first codec-based warning message code string is generated at step S

2

. If the check result is NO, a first codec-based silent fixed pattern is generated at step S

3

. Then, at step S

4

, a second codec-based code string is generated, and at step S

5

, a synthetic code string is generated from both the first codec-based and second codec-based code strings.

Since the information processor executes the encoding program, it functions like the encoder with no dedicated hardware. That is, when a recording medium having recorded therein data conforming to the second format based on the second coding method, whose encoding efficiency is higher than the first format based on the first coding method, is played in a first format-conforming player silently without any noise, the warning message will be played back from the top portion of the data. Thus it is possible to prevent the user from considering the recording medium to have no sound recorded therein, which would be possible because of the silent playback.

As apparent from the foregoing description, according to the present invention, a user going to play back a signal encoded by the second format-conforming codec using a first format-conforming player, can be given a warning message while being allowed to simply control the second format-conforming playback.

Number	Name	Date	Kind
4783654	Ichikawa	Nov 1988	A
5388105	Takagi et al.	Feb 1995	A
5528572	Koizumi et al.	Jun 1996	A
5570339	Nagano	Oct 1996	A
5640161	K.O. et al.	Jun 1997	A
6147826	Hashimura	Nov 2000	A

Number	Date	Country
065898	Jun 1995	EP
0 878 798	Nov 1998	EP
2150734	Jul 1985	GB

Encoding apparatus and method, recording medium, and decoding apparatus and method

Information

Patent Number

Date Filed

Date Issued

Inventors

Original Assignees

Examiners

Agents

CPC

US Classifications

Field of Search

US

International Classifications

Abstract

Description

Claims

Priority Claims (1)

US Referenced Citations (6)

Foreign Referenced Citations (3)

Non-Patent Literature Citations (1)