JP2004063056A - Signal reproducing method and device, signal recording method and device, code column forming method, recording medium, signal superposing method and system, and signal superposition encoding method and system - Google Patents

Abstract

<P>PROBLEM TO BE SOLVED: To make reproduction of high-quality contents possible by acquiring a small quantity of additional data while making trial listening of contents, such as music, possible, and to enhance a willingness to buy for users who are satisfied merely with the contents of low quality for trial listening. <P>SOLUTION: Low tone quality portions which are first frame portions arrayed with frames FB being first code columns formed with part of the code columns obtained by encoding original signals in frame units as dummy data and message addition portions which are second frame portions arrayed with frames FA of the code columns encoded with the signals formed by compositing another signals, for example, message signals, such as "This is trial listening data.", with the original signals are alternately arranged at prescribed intervals in such a manner that narrow-band reproduction is performed for trial listening purposes. <P>COPYRIGHT: (C)2004,JPO

Description

[0001]
TECHNICAL FIELD OF THE INVENTION
The present invention relates to a signal reproducing method and apparatus, a signal recording method and apparatus, a code string generating method, a recording medium, a signal superimposing method and apparatus, and a signal superimposing encoding method and apparatus. Signal encoding method, and as a result, if a trial viewer decides to purchase, a signal reproducing method and apparatus, which enables high quality reproduction or recording by adding a small amount of information data, The present invention relates to a recording method and apparatus, a code string generation method, a recording medium, a signal superposition method and apparatus, and a signal superposition encoding method and apparatus.
[0002]
[Prior art]
For example, there is known a content (software) distribution method in which a signal such as sound is encrypted and broadcasted, or recorded on a recording medium, and only a person who purchases a decryption key is permitted to view the decryption key. ing.
[0003]
As an encryption method, for example, an initial value of a random number sequence is given as a key signal to a bit sequence of a PCM audio signal, and an exclusive OR of the generated 0/1 random number sequence and the PCM bit sequence is calculated. There is known a method of transmitting a bit string transmitted or recording it on a recording medium. By using this method, only a person who has obtained the key signal can correctly reproduce the sound signal, and a person who has not obtained the key signal can reproduce only noise. Needless to say, a more complicated method such as a so-called DES (Data Encryption Standard) can be used as the encryption method. The contents of the DES standard are disclosed in the documents “Federation Information Processing Standards Publication 46, Specifications for the DATA ENCRYPTION STANDARD, 1977, January 15”.
[0004]
On the other hand, a method of compressing and broadcasting an audio signal or recording it on a recording medium is widespread, and a recording medium such as a magneto-optical disk capable of recording coded audio or audio signals is widely used. ing.
[0005]
There are various methods for high-efficiency encoding of signals such as audio or voice. For example, a non-blocking frequency band in which an audio signal or the like on a time axis is divided into a plurality of frequency bands and encoded without being blocked. Band division coding (sub-band coding: SBC), which is a division method, or conversion of a signal on the time axis into a signal on the frequency axis (spectral conversion) to divide the signal into a plurality of frequency bands, and for each band A block frequency band division scheme to be coded, so-called transform coding, and the like can be given. Further, a high-efficiency coding method combining the above-described band division coding and transform coding is also considered. In this case, for example, after performing band division by the above band division coding, The spectrum of the signal for each band is converted into a signal on the frequency axis, and encoding is performed for each of the spectrum-converted bands.
[0006]
Here, as the above-mentioned filter, for example, there is a QMF filter, and this QMF filter is described in the literature "1976, RE. Crochiere, Digital coding of speech in subbands, Bell System. Tech. J. Vol. 8, 1976 ". In addition, a document "ICASSP 83, BOSTON, Polyphase Quadrature filters-A new subband coding technique, Joseph H. Rothweiler" describes a filter dividing method of equal bandwidth.
[0007]
In addition, as the above-described spectral transform, for example, an input audio signal is divided into blocks in a predetermined unit time (frame), and a discrete Fourier transform (DFT), a discrete cosine transform (DCT), a modified DCT (MDCT), and the like are provided for each block. There is a spectrum transformation that transforms the time axis to the frequency axis by performing. For the MDCT, see the document "ICASSP, 1987, Subband / Transform Coding Using Filter Bank Designs Based on Time Domain Aliasing Cancer, J.P.y.n.R.B.B.I.N. Have been.
[0008]
When the above-described DFT or DCT is used as a method of converting a waveform signal into a spectrum, M independent real number data can be obtained by performing conversion using a time block including M samples. In order to reduce the connection distortion between the time blocks, the blocks adjacent to each other are usually overlapped by M1 samples each. Therefore, in DFT or DCT, M samples are used for (M-M1) samples on average. Is quantized and encoded.
[0009]
On the other hand, when the above-described MDCT is used as a method for converting into a spectrum, independent real number data is obtained from 2M samples overlapping each other with time on both sides, so that the average is obtained. Then, in MDCT, M real number data is quantized and encoded for M samples. In the decoding device, the waveform signal can be reconstructed by adding the waveform elements obtained by performing the inverse transform in each block from the code obtained by using the MDCT while causing them to interfere with each other. .
[0010]
In general, by increasing the time block for the conversion, the frequency resolution of the spectrum is increased and the energy is concentrated on specific spectral components. Therefore, by using a MDCT in which a transform is performed with a long block length by overlapping the neighboring blocks by half each and the number of obtained spectral signals does not increase with respect to the number of original time samples, DFT or It is possible to perform more efficient coding than when DCT is used. In addition, by providing a sufficiently long overlap between adjacent blocks, distortion between blocks of a waveform signal can be reduced.
[0011]
By quantizing the signal divided for each band by the filter or the spectrum conversion in this way, it is possible to control the band in which the quantization noise is generated, and to use the characteristics such as a masking effect to achieve a higher auditory efficiency. Encoding can be performed. Further, if the normalization is performed for each band, for example, with the maximum value of the absolute value of the signal component in that band before performing the quantization, more efficient encoding can be performed.
[0012]
As a frequency division width when quantizing each frequency component divided into frequency bands, for example, band division is performed in consideration of human auditory characteristics. That is, an audio signal may be divided into a plurality of (for example, 25 bands) bands with a bandwidth that is generally referred to as a critical band (critical band) such that the higher the band, the wider the bandwidth. When encoding data for each band at this time, predetermined bits are allocated to each band, or encoding is performed by adaptive bit allocation (bit allocation) for each band. For example, when encoding the coefficient data obtained by the MDCT processing by the bit allocation, adaptively assigning bits to the MDCT coefficient data for each band obtained by the MDCT processing for each block. The encoding will be performed with numbers.
[0013]
The following two methods are known as such a bit allocation method. That is, first, in the documents "Adaptive Transform Coding of Speech Signals, R. Zelinski and P. Noll, IEEE Transactions of Acoustics, Speech, and Sigma, each of the sig- nals, Species, and Sigma. Bit allocation is performed based on the magnitude of the signal. In this method, the quantization noise spectrum becomes flat and the noise energy becomes the minimum, but the actual feeling of noise is not optimal because the masking effect is not utilized in terms of auditory sense. In addition, the document "ICASSP 1980, The critical band coder --- digital encoding of the perceptual requirements of the audition, each system is used for each type of noise, and the M.A. A method of obtaining a ratio and performing fixed bit allocation is described. However, in this method, even when the characteristic is measured with a sine wave input, the characteristic value is not so good because the bit allocation is fixed.
[0014]
In order to solve these problems, all bits available for bit allocation are divided into a fixed bit allocation pattern predetermined for each small block and a bit allocation depending on the signal size of each block. There has been proposed a high-efficiency coding apparatus that is divided and used, and the division ratio depends on a signal related to an input signal, and the smoother the spectrum of the signal, the larger the division ratio into the fixed bit allocation pattern.
[0015]
According to this method, when energy is concentrated on a specific spectrum, such as a sine wave input, by assigning many bits to a block including the spectrum, the overall signal-to-noise characteristic can be significantly improved. it can. In general, human hearing is extremely sensitive to signals having steep spectral components. Therefore, using such a method to improve the signal-to-noise characteristic simply increases the numerical value obtained by measurement. In addition, it is effective in improving sound quality in terms of hearing.
[0016]
Numerous other ways of assigning bits have been proposed, and furthermore, models of auditory perception have been refined, and if the capabilities of the encoding device are increased, encoding will be perceptually more efficient. . In these methods, it is general to obtain a real bit allocation reference value that realizes the signal-to-noise characteristic obtained by calculation as faithfully as possible, and set an integer value approximating the reference value to the allocated bit number. .
[0017]
Also, in the specification and drawings of Japanese Patent Application No. 5-152865 or WO 94/28633 previously proposed by the present inventors, from the spectral signal, a tone component that is particularly important for hearing, that is, around a specific frequency. There has been proposed a method of separating a signal component in which energy is concentrated and encoding the signal component separately from other spectral components, thereby achieving high compression of an audio signal or the like with almost no deterioration in hearing. It is possible to code efficiently at a rate.
[0018]
In constructing an actual code string, first, quantization accuracy information and normalization coefficient information are encoded with a predetermined number of bits for each band in which normalization and quantization are performed, and then normalized and quantized. The encoded spectral signal may be encoded. Also,
ISO / IEC 11172-3: 1993 (E), 1993
Describes a high-efficiency coding scheme in which the number of bits representing quantization accuracy information is set differently depending on the band, and the higher the band, the smaller the number of bits representing quantization accuracy information becomes. It has been standardized.
[0019]
Instead of directly encoding the quantization accuracy information, a method of determining the quantization accuracy information from the normalization coefficient information in the decoding device is also known. However, in this method, the normalization is performed when a standard is set. Since the relationship between the coefficient information and the quantization accuracy information is determined, it will not be possible to introduce quantization accuracy control based on a more advanced auditory model in the future. In addition, when there is a range in the compression ratio to be realized, it is necessary to determine the relationship between the normalization coefficient information and the quantization accuracy information for each compression ratio.
[0020]
The quantized spectral signal is converted into a variable as described, for example, in the document "DA Huffman: A Method for Construction of Minimal Redundancy Codes, Proc. IRE, 40, p. 1098 (1952)". There is also known a method of encoding more efficiently by encoding using a long code.
[0021]
[Problems to be solved by the invention]
By the way, a signal such as a sound encoded by the above-described method is encrypted and distributed, or recorded on a recording medium, and only the person who purchased the key is permitted to view the software content. Distribution methods are known. As an encryption method, for example, an initial value of a random number sequence is given as a key signal to a bit sequence of an acoustic signal of PCM (Pulse Code Modulation) or to a bit sequence of an encoded signal, and the generated 0 There has been known a method of transmitting a bit string obtained by performing an exclusive OR operation of a random number sequence of / 1 and the above-described bit string or recording the bit string on a recording medium. By using this method, only a person who has obtained the key signal can correctly reproduce the sound signal, and a person who has not obtained the key signal can reproduce only noise.
[0022]
However, in these scramble methods, when there is no key or when reproduced by ordinary reproducing means, when reproduced, it causes noise, and it is not possible to grasp the contents of the software. For this reason, for example, a disk in which music is recorded with relatively low sound quality is distributed, and a person who listens to the music can purchase a key only for the one that he / she likes and reproduce it with high sound quality. It could not be used for purposes such as making it possible to purchase a new disc recorded with high sound quality after listening to the software.
[0023]
Also, conventionally, when encrypting a signal that has been subjected to high-efficiency encoding, it has been difficult to provide a code sequence that is meaningful to ordinary reproducing means and not to reduce its compression efficiency. That is, as described above, when scrambling is performed on a code sequence formed by applying a high-efficiency code, not only noise is generated even when the code sequence is reproduced, but also the code sequence generated by scrambling is changed to the original sequence. If the standard does not conform to the high efficiency code standard, the reproducing means may not operate at all. Conversely, if the PCM signal is scrambled and then subjected to high-efficiency coding, for example, if the amount of information is reduced by utilizing the characteristics of hearing, when the high-efficiency coding is released, Since a signal obtained by scrambling a PCM signal cannot be reproduced, it is difficult to correctly descramble the scramble. For this reason, it is necessary to select a compression method that can descramble the scramble correctly even if the efficiency is reduced.
[0024]
On the other hand, according to the technique described in Japanese Patent Application Laid-Open No. Hei 10-135944 previously proposed by the present inventors, for example, a music signal converted into a spectrum signal and coded, There is disclosed an audio encoding system that allows only a narrowband signal to be previewed without a key if only a key is encrypted. That is, in this method, for example, the high-frequency side is encrypted, the high-frequency side bit allocation information and the like are replaced with dummy data, and the high-frequency true bit allocation information is recorded in a position where a normal decoder ignores. are doing. If this method is adopted, for example, it is possible to enjoy only the favorite music with high sound quality as a result of the trial listening.
[0025]
By the way, in the technology described in the above-mentioned Japanese Patent Application Laid-Open No. 10-135944, since the security depends only on encryption, if the encryption is decrypted, a fee cannot be collected, There is a risk that high quality music can be heard.
[0026]
On the other hand, we provide low-quality listening data that does not include some data, such as the high-frequency side, for free or at a low price, and obtain only a small amount of high-quality data for your favorite music. Although a method that can improve the sound quality has been considered, in such a method, if the listener is satisfied with the relatively low-quality sound quality of the trial listening, data for improving the quality is purchased. There is a problem that there is a risk that it will not be cut.
[0027]
The present invention has been proposed in view of the above-described situation, and it is possible to perform high-quality signal reproduction only by obtaining a relatively small amount of additional data while performing trial viewing. A signal reproducing method and apparatus, a signal recording method and apparatus, a code string generating method, a recording medium, and a signal superimposing method that can prevent a user from being satisfied with only low-quality data that can be viewed. And an apparatus, and a signal superposition encoding method and apparatus.
[0028]
[Means for Solving the Problems]
The present invention has been proposed in consideration of the above-described problems, and encodes music or the like obtained by repeatedly superimposing a short message such as “data for trial listening” on content such as music, and listens to the encoded music. When the user decides to purchase the song, the code string of the portion where the message is superimposed is replaced with a code string of music-only data that does not include the message. Further, the signal level at the time of superimposing the messages is set to an appropriate level so as to solve the problems such as masking of the message, occurrence of overflow, and occurrence of signal switching noise.
[0029]
Here, the applicant has previously replaced part of the code string with dummy data in PCT / JP02 / 01106 and PCT / JP02 / 01105 (both not disclosed) so that trial viewing is possible. Distribute the data (test viewing data) and let the player freely play the relatively low-quality playback audio and images with a narrow bandwidth. As a result, if the contents are liked and the test viewer decides to purchase, dummy data It proposes a technology that receives the correct data (high quality data) that replaces and can enjoy high-quality playback. In this case, the music signal is converted into spectral coefficients for each block of a predetermined length, divided into tonal components and other components, and the tonal components are normalized and requantized for each tonal component. The other components are normalized and requantized for each predetermined band, and when the spectrum signal obtained by requantization is encoded from the low frequency side using a variable length code, a tone component at or above a predetermined frequency is used. And normalizing coefficients of other components are replaced with dummy data, and among the variable-length coded spectral components, a method of replacing low-frequency side spectral components at or above the predetermined frequency with dummy data is included. I have. As described above, an additional data sequence for replacing the dummy data is obtained with respect to the trial data sequence including the dummy data, and by integrating both, a music signal with a wide band can be reproduced. In such a technique, further, for a part of a frame sequence, music or the like obtained by repeatedly superimposing the short messages as described above is encoded, and when a user who listens to the music decides to purchase the song, The code string of the part where the message is superimposed is replaced with a code string of music-only data that does not include the message.
[0030]
That is, in the signal reproducing method and apparatus according to the present invention, when reproducing the code string obtained by encoding the original signal in frame units, the first frame portion and another signal are combined with the original signal. And a second frame portion composed of a code sequence obtained by encoding the input signal, and replacing the second frame portion with a code sequence obtained by encoding the corresponding frame portion of the original signal. One frame part and the replaced code string or the second frame part are decoded.
[0031]
Here, as the signal reproducing method and apparatus according to the present invention, when reproducing a code string obtained by encoding an original signal in frame units, a part of the code string is converted from a code string that is set as dummy data. A first code sequence having a first frame portion composed of a first frame portion and a second frame portion composed of a code sequence obtained by encoding a signal obtained by combining another signal with the original signal is input to the second frame portion. A part is replaced with a code string obtained by coding a corresponding frame part of the original signal, at least a part of the first code string is complemented with a second code string, and the replaced code string and the complement are replaced. And decoding the code string or the first code string.
[0032]
In the signal recording method and apparatus according to the present invention, when recording a code sequence obtained by encoding an original signal in frame units, a first frame portion and another signal are combined with the original signal. Inputting a code sequence having a second frame portion composed of a code sequence obtained by encoding the obtained signal, and replacing the second frame portion with a code sequence obtained by encoding the corresponding frame portion of the original signal. And
[0033]
Also, in the code string generation method according to the present invention, in the code string generation method for generating a code string obtained by encoding an original signal on a frame basis, the first frame portion and another signal may be included in the original signal. And generating a code sequence including a second frame portion including a code sequence obtained by coding the synthesized signal.
[0034]
Further, in the recording medium according to the present invention, in a recording medium in which a code string obtained by encoding an original signal in frame units is recorded, a first frame portion and another signal are combined with the original signal. A code sequence having a second frame portion composed of a code sequence obtained by encoding a signal is recorded.
[0035]
Here, the first frame portion may include a first code string in which a part of the code string of the original signal is dummy data. The other signal may be a message signal, the code sequence obtained by encoding the original signal replacing the second frame portion may be encrypted, and the second frame portion may be encrypted. Is given at predetermined intervals.
[0036]
In the encoding, the input signal is spectrum-transformed, band-divided, and a code string of a predetermined format including quantization accuracy information, normalization coefficient information, and spectrum coefficient information for each band is generated. The dummy data may include data corresponding to a part of the spectrum coefficient information.
[0037]
Furthermore, when the music signal and the message signal are superimposed as they are, if the level of the music signal is high, the message signal is masked by the music signal, and there is a problem that the message signal cannot be clearly heard. When two high music signals are simply added together, the level range may overflow.In the case of overflow, the method of replacing the level with the maximum or minimum value of the level may cause noise such as a bursty noise. In consideration of the problem that the message may be superimposed, in the present invention, the message is superimposed by lowering the level of the original audio signal and superimposing the message signal at the position where the message signal is superimposed. Masks and overflows The problem is solved by making the level change smoothly rather than making the level change abruptly, thereby preventing the occurrence of noise such as a click which occurs when the level changes abruptly. .
[0038]
That is, the signal superimposition method and apparatus according to the present invention, when superimposing a message signal on at least a part of the first signal, at least a part of the part superimposing the message signal on the first signal, For a first combining coefficient R1 for the first signal and a second combining coefficient R2 for the message signal,
| R1 | <| R2 |
Is satisfied.
[0039]
Also, the signal superposition encoding method and apparatus according to the present invention superimposes a message signal on at least a part of the first signal and, when encoding the superimposed signal, adds a message signal to the first signal. Is superimposed on at least a part of the first combined coefficient R1 for the first signal and the second combined coefficient R2 for the message signal.
| R1 | <| R2 |
Is satisfied.
[0040]
BEST MODE FOR CARRYING OUT THE INVENTION
First, prior to describing an embodiment according to the present invention, an optical disc recording / reproducing apparatus as a general compressed data recording / reproducing apparatus for describing the embodiment of the present invention will be described with reference to the drawings.
[0041]
FIG. 1 is a block diagram illustrating an example of an optical disk recording / reproducing apparatus. In the apparatus shown in FIG. 1, a magneto-optical disk 1 driven by a spindle motor 51 is used as a recording medium. When recording data on the magneto-optical disk 1, for example, a so-called magnetic field modulation recording is performed by applying a modulation magnetic field corresponding to the recording data with the magnetic head 54 while irradiating the laser light with the optical head 53. The data is recorded along the recording track. At the time of reproduction, a recording track of the magneto-optical disk 1 is traced by a laser beam by the optical head 53 to perform magneto-optical reproduction.
[0042]
The optical head 53 includes, for example, a laser light source such as a laser diode, a collimator lens, an objective lens, an optical component such as a polarizing beam splitter, a cylindrical lens, and a photodetector having a light receiving unit having a predetermined pattern. The optical head 53 is provided at a position facing the magnetic head 54 via the magneto-optical disk 1. When data is recorded on the magneto-optical disk 1, the magnetic head 54 is driven by a recording-system head drive circuit 66, which will be described later, to apply a modulation magnetic field in accordance with the recording data. By irradiating the track with laser light, thermomagnetic recording is performed by a magnetic field modulation method. The optical head 53 detects the reflected light of the laser beam applied to the target track, detects a focus error by, for example, a so-called astigmatism method, and detects a tracking error by, for example, a so-called push-pull method. When reproducing data from the magneto-optical disk 1, the optical head 53 detects the focus error and the tracking error, and at the same time, detects the difference in the polarization angle (Kerr rotation angle) of the reflected light of the laser light from the target track to reproduce the data. Generate a signal.
[0043]
The output of the optical head 53 is supplied to an RF circuit 55. The RF circuit 55 extracts the focus error signal and the tracking error signal from the output of the optical head 53 and supplies the same to the servo control circuit 56, and also binarizes the reproduction signal and supplies it to a reproduction system decoder 71 described later. .
[0044]
The servo control circuit 56 includes, for example, a focus servo control circuit, a tracking servo control circuit, a spindle motor servo control circuit, a thread servo control circuit, and the like. The focus servo control circuit performs focus control of the optical system of the optical head 53 so that the focus error signal becomes zero. Further, the tracking servo control circuit performs tracking control of the optical system of the optical head 53 so that the tracking error signal becomes zero. Further, the spindle motor servo control circuit controls the spindle motor 51 so as to rotate the magneto-optical disk 1 at a predetermined rotation speed (for example, a constant linear speed). Further, the thread servo control circuit moves the optical head 53 and the magnetic head 54 to target track positions of the magneto-optical disk 1 specified by the system controller 57. The servo control circuit 56 that performs such various control operations sends information indicating the operation state of each unit controlled by the servo control circuit 56 to the system controller 57.
[0045]
A key input operation unit 58 and a display unit 59 are connected to the system controller 57. The system controller 57 controls a recording system and a reproduction system based on operation input information based on operation input information from the key input operation unit 58. In addition, the system controller 57 performs the above-described recording traced by the optical head 53 and the magnetic head 54 on the basis of the address information in the unit of sector reproduced from the recording track of the magneto-optical disk 1 by the header time and the Q data of the subcode. Manages recording and playback positions on a track. Further, the system controller 57 controls the display section 59 to display the reproduction time based on the data compression ratio of the compressed data recording / reproducing apparatus and the reproduction position information on the recording track.
[0046]
This reproduction time display is based on the reciprocal of the data compression ratio (for example, 1/1) with respect to address information (absolute time information) in sector units reproduced from a recording track of the magneto-optical disk 1 by so-called header time or so-called subcode Q data. In the case of 4-compression, the actual time information is obtained by multiplying by 4), and this is displayed on the display unit 59. At the time of recording, for example, when absolute time information is recorded in advance on a recording track of a magneto-optical disk or the like (preformatted), the preformatted absolute time information is read and the data compression ratio is determined. By multiplying by the reciprocal, it is possible to display the current position with the actual recording time.
[0047]
Next, in the recording system of the optical disk recording / reproducing apparatus shown in FIG. _IN Is supplied to an A / D converter 62 via a low-pass filter 61, and the A / D converter 62 _IN Is quantized. The digital audio signal obtained from the A / D converter 62 is supplied to an ATC (Adaptive Transform Coding) encoder 63. Also, the digital audio input signal D from the input terminal 67 _IN Is supplied to the ATC encoder 63 via the digital input interface circuit 68. The ATC encoder 63 receives the input signal A _IN Is subjected to bit compression (data compression) processing according to a predetermined data compression rate on digital audio PCM data having a predetermined transfer rate, which is quantized by the A / D converter 62, and is output from the ATC encoder 63. The compressed data (ATC data) is supplied to a memory (RAM) 64. For example, a case where the data compression rate is 1/8 will be described. Here, the data transfer rate is a data transfer rate (75 sectors / second) of a so-called CD-DA format which is a standard digital audio CD format. This is reduced to 1/8 (9.375 sectors / sec).
[0048]
Next, the memory (RAM) 64 is controlled by the system controller 57 to write and read data, temporarily stores the ATC data supplied from the ATC encoder 63, and records it on the disk as needed. As a buffer memory. That is, for example, when the data compression rate is 1/8, the compressed audio data supplied from the ATC encoder 63 has a data transfer rate of a standard CD-DA format data transfer rate (75 sectors / second). The compressed data is reduced to 1/8, that is, 9.375 sectors / sec. As described above, it is sufficient for this compressed data (ATC data) to record one sector per eight sectors, but such recording every eight sectors is practically impossible. I try to record.
[0049]
This recording is performed at intervals of the same data transfer rate (75 sectors / second) as in the standard CD-DA format by using a cluster consisting of a plurality of predetermined sectors (for example, 32 sectors + several sectors) as a recording unit through a pause period. It is done on a regular basis. That is, in the memory 64, ATC audio data having a data compression rate of 1/8 continuously written at a low transfer rate of 9.375 (= 75/8) sectors / sec according to the bit compression rate is used as recording data. As a burst at the transfer rate of 75 sectors / sec. The overall data transfer rate of the read and recorded data, including the recording pause period, is as low as 9.375 sectors / sec. The instantaneous data transfer rate in the above is the standard 75 sectors / second. Therefore, when the disk rotation speed is the same as the standard CD-DA format (constant linear speed), the same recording density and recording pattern as in the CD-DA format are recorded.
[0050]
The ATC audio data, that is, the recording data, which is burst-read from the memory 64 at the (instantaneous) transfer rate of 75 sectors / second is supplied to the encoder 65. Here, in a data string supplied from the memory 64 to the encoder 65, a unit continuously recorded in one recording is a cluster including a plurality of sectors (for example, 32 sectors) and a cluster connection arranged before and after the cluster. For several sectors. The cluster connection sector is set to be longer than the interleave length of the encoder 65, so that even if interleaved, data of other clusters is not affected.
[0051]
The encoder 65 performs encoding processing for error correction (parity addition and interleaving processing), EFM encoding processing, and the like on the recording data supplied in bursts from the memory 64 as described above. The recording data that has been subjected to the encoding process by the encoder 65 is supplied to the magnetic head drive circuit 66. The magnetic head drive circuit 66 is connected to the magnetic head 54 and drives the magnetic head 54 so as to apply a modulation magnetic field corresponding to the recording data to the magneto-optical disk 1.
[0052]
Further, the system controller 57 controls the memory 64 as described above so that the recording data read from the memory 64 in a burst by the memory control is continuously recorded on the recording track of the magneto-optical disk 1. The recording position is controlled. The recording position is controlled by controlling the recording position of the recording data read in a burst from the memory 64 by the system controller 57 and transmitting a control signal designating the recording position on the recording track of the magneto-optical disk 1 to a servo control circuit. 56.
[0053]
Next, a reproducing system of the optical disk recording / reproducing apparatus shown in FIG. 1 will be described. This reproducing system is for reproducing the recorded data continuously recorded on the recording tracks of the magneto-optical disk 1 by the above-mentioned recording system. The decoder 71 is provided with a reproduced output obtained by tracing at step (1) and binarized by the RF circuit 55 and supplied. In this case, not only a magneto-optical disc but also a read-only optical disc such as a so-called CD (Compact Disc) and a so-called CD-R type optical disc can be read.
[0054]
The decoder 71 corresponds to the encoder 65 in the above-described recording system, and performs the above-described decoding processing for error correction, EFM decoding processing, and the like on the reproduced output binarized by the RF circuit 55. Then, the ATC audio data having the data compression rate of 1/8 is reproduced at a transfer rate of 75 sectors / sec, which is higher than the normal transfer rate. The reproduction data obtained by the decoder 71 is supplied to a memory (RAM) 72.
[0055]
In the memory (RAM) 72, the writing and reading of data are controlled by the system controller 57, and the reproduced data supplied from the decoder 71 at a transfer rate of 75 sectors / second is written in burst at the transfer rate of 75 sectors / second. It is. Also, in the memory 72, the reproduction data written in a burst at the transfer rate of 75 sectors / second is continuously read at a transfer rate of 9.375 sectors / second corresponding to a data compression rate of 1/8. .
[0056]
The system controller 57 performs memory control such that the reproduced data is written into the memory 72 at a transfer rate of 75 sectors / second, and the reproduced data is continuously read from the memory 72 at the transfer rate of 9.375 sectors / second. . Further, the system controller 57 performs the above-described memory control for the memory 72, and continuously reproduces the reproduction data written in the memory 72 in a burst manner by the memory control from the recording track of the magneto-optical disk 1. Control the playback position. The reproduction position is controlled by controlling the reproduction position of the reproduction data continuously read from the memory 72 by the system controller 57 and transmitting a control signal designating the reproduction position on the recording track of the magneto-optical disk 1 or the optical disk 1. This is performed by supplying the signal to the servo control circuit 56.
[0057]
ATC audio data obtained as reproduction data continuously read from the memory 72 at a transfer rate of 9.375 sectors / sec is supplied to the ATC decoder 73. The ATC decoder 73 corresponds to the ATC encoder 63 of the recording system, and reproduces 16-bit digital audio data by, for example, expanding the ATC data by 8 times (bit expansion). The digital audio data from the ATC decoder 73 is supplied to a D / A converter 74.
[0058]
The D / A converter 74 converts the digital audio data supplied from the ATC decoder 73 into an analog signal and outputs the analog audio output signal A _OUT To form The analog audio signal A obtained by the D / A converter 74 _OUT Is output from an output terminal 76 via a low-pass filter 75.
[0059]
Next, the high-efficiency compression encoding of a signal will be described in detail. That is to say, refer to FIG. 2 et seq. For a technique for performing high-efficiency encoding of an input digital signal such as an audio PCM signal using band division coding (SBC), adaptive conversion coding (ATC), and adaptive bit allocation. It will be explained while doing so.
[0060]
FIG. 2 is a block diagram showing a specific example of an audio waveform signal encoding device used for describing the embodiment of the present invention. In this example, an input signal waveform 101 is converted into a signal 102 of a signal frequency component by a conversion unit 1101, each component is coded by a signal component coding unit 1102, and a code sequence 104 is generated by a code sequence generation unit 1103. Is generated.
[0061]
FIG. 3 shows a specific example of the converter 1101 shown in FIG. 2. A signal divided into two bands by a band division filter is converted into spectral signal components 221 and 222 by forward spectrum converters 1211 and 1212 such as MDCT in each band. Has been converted to. The signal 201 in FIG. 3 corresponds to the signal 101 in FIG. 2, and the signals 221 and 222 in FIG. 3 correspond to the signal 102 in FIG. 3, the bandwidths of the signals 211 and 212 are の of the bandwidth of the signal 201, and are decimated to １ ／ of the bandwidth of the signal 201. Various conversion means other than this specific example are conceivable. For example, an input signal may be directly converted into a spectrum signal by MDCT, or DFT (discrete Fourier transform) or DCT (discrete cosine transform) instead of MDCT. May be converted by Although it is possible to divide the signal into band components by a so-called band division filter, it is convenient to adopt a method of converting the frequency components into frequency components by the above-described spectrum conversion in which a large number of frequency components are obtained with a relatively small amount of calculation.
[0062]
FIG. 4 shows a specific example of the signal component encoding unit 1102 in FIG. 2. The input signal 301 is normalized by a normalizing unit 1301 for each predetermined band (signal 302), and then the quantization precision is determined. Based on the quantization accuracy information 303 calculated by the means 1302, the data is quantized by the quantization means 1303 and extracted as a signal 304. The signal 301 in FIG. 4 corresponds to the signal 102 in FIG. 2, and the signal 304 in FIG. 4 corresponds to the signal 103 in FIG. 2. Here, the signal 304 includes a quantized signal component and a normalization coefficient. Information and quantization accuracy information are also included.
[0063]
FIG. 5 is a block diagram illustrating a specific example of a decoding device that outputs an audio signal from a code sequence generated by the coding device illustrated in FIG. In this specific example, the code 402 of each signal component is extracted from the code sequence 401 by the code sequence decomposing means 1401, and the signal component 403 is restored from the code 402 by the signal component decoding means 1402, and then the inverse transform means 1403 As a result, an acoustic waveform signal 404 is output.
[0064]
FIG. 6 shows a specific example of the inverse conversion means 1403 in FIG. 5, which corresponds to the specific example of the conversion means in FIG. 3, and the signal 511 of each band obtained by the inverse spectrum conversion means 1501 and 1502. , 512 are synthesized by the band synthesis filter 1511. The signals 501 and 502 in FIG. 6 correspond to the signal 403 in FIG. 5, and the signal 521 in FIG. 6 corresponds to the signal 404 in FIG.
[0065]
FIG. 7 shows a specific example of the signal component decoding means 1402 in FIG. 5. The signal 551 in FIG. 7 corresponds to the signal 402 in FIG. 5, and the signal 553 in FIG. 7 corresponds to the signal 403 in FIG. After the spectrum signal 551 is inversely quantized by the inverse quantization means 1551 (signal 552), the spectrum signal 551 is inversely normalized by the inverse normalization means 1552 and is extracted as a signal 553.
[0066]
FIG. 8 is a diagram for describing an encoding method conventionally performed in the encoding device shown in FIG. In the example of this figure, the spectrum signal is obtained by the conversion means of FIG. 3, and FIG. 8 shows the absolute value of the spectrum of the MDCT with the level converted to dB. The input signal is converted into, for example, 64 spectral signals for each predetermined time block, which are collectively normalized and quantized into, for example, eight bands b1 to b8 (hereinafter referred to as encoding units). Is performed. By changing the quantization accuracy for each coding unit depending on the distribution of frequency components, it is possible to perform audio-efficient coding that minimizes deterioration of sound quality.
[0067]
FIG. 9 shows a specific example in a case where the signal encoded as described above is recorded on a recording medium. In this specific example, a fixed-length header including a synchronization signal SC is provided at the beginning of each frame, and the number UN of coding units is also recorded here. Following the header, quantization accuracy information QN is recorded by the number of coding units, and thereafter, normalization accuracy information NP is recorded by the number of coding units. The normalized and quantized spectral coefficient information SP is recorded thereafter, but when the frame length is fixed, an empty area may be formed after the spectral coefficient information SP. The example of this figure is obtained by encoding the spectrum signal of FIG. 8, and the quantization accuracy information QN is illustrated from, for example, 6 bits of the lowest-band coding unit to 2 bits of the highest-band coding unit, for example. The normalization coefficient information NP is assigned as shown from a value of, for example, 46 for the lowest band coding unit to a value of, for example, 22 for the highest band coding unit. As the normalization coefficient information NP, for example, a value proportional to the dB value is used.
[0068]
Compared with the method described above, it is possible to further increase the coding efficiency. For example, among the quantized spectrum signals, a relatively short code length is assigned to a high-frequency signal, and a relatively long code length is assigned to a low-frequency signal, thereby improving coding efficiency. be able to. Also, for example, by increasing the transform block length, the amount of sub-information such as quantization accuracy information and normalization coefficient information can be relatively reduced, and the frequency resolution is increased, so that the quantization accuracy can be more finely adjusted on the frequency axis. , The coding efficiency can be improved.
[0069]
Furthermore, in the specification and drawings of Japanese Patent Application No. 5-152865 or WO 94/28633 previously proposed by the inventors of the present invention, a tone component that is particularly important for hearing from a spectral signal, that is, a frequency component around a specific frequency, There has been proposed a method of separating a signal component whose energy is concentrated into a signal and encoding the signal component separately from other spectral components, whereby an audio signal or the like is hardly deteriorated in auditory sense. It is possible to encode efficiently at a compression ratio.
[0070]
FIG. 10 is a diagram for explaining a method of performing encoding using such a method. Codes obtained by separating a signal having a particularly high level from a spectral signal as tone components, for example, tone components Tn1 to Tn3. It shows a state of becoming. For each of the tone components Tn1 to Tn3, its position information, for example, position data Pos1 to Pos3 is also required, but the spectrum signal after extracting the tone components Tn1 to Tn3 can be quantized with a small number of bits. Therefore, if such a method is applied to a signal whose energy is concentrated in a specific spectrum signal, particularly efficient encoding can be performed.
[0071]
FIG. 11 shows the configuration of the signal component encoding unit 1102 of FIG. 2 in the case of separating and encoding the tone component as described above. The output signal 102 (signal 601 in FIG. 11) of the conversion means 1101 in FIG. 2 is separated into a tone component (signal 602) and a non-tone component (signal 603) by the tone component separation means 1601, Encoding means 1602 and non-tone component encoding means 1603, and are extracted as signals 604 and 605, respectively. The tone component encoding unit 1602 and the non-tone component encoding unit 1603 have the same configuration as in FIG. 4, but the tone component encoding unit 1602 also encodes the position information of the tone component.
[0072]
Similarly, FIG. 12 shows the configuration of the signal component decoding means 1402 of FIG. 5 in the case of decoding the coded component by separating the tone component as described above. 12 corresponds to signal 604 in FIG. 11, and signal 702 in FIG. 12 corresponds to signal 605 in FIG. The signal 701 is decoded by the tone component decoding unit 1701 and sent to the spectrum signal synthesis unit 1703 as a signal 703. The signal 702 is decoded by the non-tone component decoding unit 1702 and sent to the spectrum signal synthesis unit 1703 as a signal 704. Spectral signal combining means 1703 combines the tone component (signal 703) and the non-tone component (signal 704) and outputs as signal 705.
[0073]
FIG. 13 shows a specific example in a case where the signal encoded as described above is recorded on a recording medium. In this specific example, tone components are separated and encoded, and the code sequence is recorded in a portion between the header part and the quantization accuracy information QN. In the tone component sequence, first, tone component number information TN is recorded, and then data of each tone component is recorded. The tone component data includes position information P, quantization accuracy information QN, normalization coefficient information NP, and spectrum coefficient information SP. Further, in this specific example, the frequency resolution is increased by making the conversion block length for converting into a spectrum signal twice as large as that of the specific example of FIG. 9, and by introducing a variable length code, the specific example of FIG. In comparison, a code string of an acoustic signal corresponding to twice the length is recorded in frames having the same number of bytes.
[0074]
The above description describes the technology prior to the description of the embodiment of the present invention. However, in the embodiment of the present invention, for example, when applied to audio, a relatively low-quality audio signal is The audio signal of high quality can be heard by purchasing a relatively small amount of additional data, etc., and the amount of additional data can be reduced. In order to reduce the number, a part of the additional data is embedded in the audio signal for trial listening as it is or encrypted.
[0075]
That is, in the base technology of the embodiment of the present invention, for example, as shown in FIG. 9, the dummy quantization accuracy in the quantization accuracy information As data, data indicating 0-bit allocation is encoded for the four encoding units on the high frequency side, and four encodings on the high frequency side are used as dummy normalization coefficient data in the normalization coefficient information NP. In the unit, the minimum value of the normalization coefficient information 0 is encoded (in this specific example, the normalization coefficient takes a value proportional to the dB value). As described above, by setting the quantization accuracy information on the high frequency side to 0, the region Neg of the data coefficient information ignored at the time of the trial listening, in fact, the spectral coefficient information of the region Neg in FIG. 14 is ignored. Is reproduced by an ordinary reproducing apparatus, narrow-band data having a spectrum as shown in FIG. 15 is reproduced. This can be used as data for trial listening. Also, by encoding the dummy coefficient data for the normalization coefficient information, it becomes more difficult to estimate the quantization accuracy information and illegally perform high-quality reproduction.
[0076]
In the above example, both the quantization accuracy information and the normalization coefficient information are replaced with dummy data, but only one of them may be replaced with dummy data. When only the quantization accuracy information is the dummy data of 0-bit data, narrow-band data having a spectrum as shown in FIG. 15 is reproduced. On the other hand, when only the normalization coefficient information is dummy data having a value of 0, the spectrum has a spectrum as shown in FIG. 16 and the spectrum on the high frequency side does not become exactly 0, From the viewpoint of audibility, it is substantially the same as 0, and in the embodiment of the present invention, this case is also referred to as a narrowband signal.
[0077]
Regarding which of the quantization accuracy information and the normalization coefficient information is to be used as dummy data, there is a difference in the risk that these true values are estimated and reproduced in high quality. When both the quantization accuracy information and the normalization coefficient information are dummy data, it is safest because there is no data for estimating their true values. If only the quantization accuracy information is dummy data, for example, if the original bit allocation algorithm is to obtain the quantization accuracy information based on the normalization coefficient, the quantization accuracy information is used as a clue to the normalization coefficient information. The risk is relatively high because there is a risk that On the other hand, since it is relatively difficult to obtain the normalization coefficient information from the quantization accuracy information, the method of using only the normalization coefficient information as dummy data is compared with the method of using only the quantization accuracy information as dummy data. The risk is lower. The quantization accuracy information or the normalization coefficient information may be selectively used as dummy data depending on the band.
[0078]
In addition, a part of the spectrum coefficient information may be replaced with zero dummy data. In particular, since the spectrum in the middle band has an important meaning in terms of sound quality, this part may be replaced with dummy data of 0, and the middle and high band part may be replaced with dummy quantization accuracy information or dummy normalization coefficient information. The dummy data does not necessarily need to be replaced with 0, and may be replaced with an arbitrary code that is shorter than a code representing a true numeric value when performing variable-length coding. In this case, the band to be replaced with the dummy quantization accuracy information or the dummy normalization coefficient information covers the band in which a part of the spectrum coefficient information is replaced with the dummy data, so that the narrow band reproduction is performed correctly. In particular, when a variable length code is used to encode the spectrum coefficient information, some information in the middle band is lost, so that all data in the higher band cannot be decoded.
[0079]
In any case, it is more difficult to guess relatively large data that enters the signal content than to decrypt a relatively short key length used in normal encryption. It can be said that the risk of fraudulently violating the rights of persons is reduced. Also, even if dummy data is guessed for a certain song, unlike the case where the encryption algorithm decryption method is known, there is no danger that damage will spread to other songs. It can be said that the security is higher than when encryption is performed.
[0080]
However, in the above method, there is a risk that the user is satisfied with the narrowband signal for trial viewing and that the user does not need to purchase data for high quality. Therefore, in the embodiment of the present invention, the band is limited, and a signal different from the content to be viewed, for example, a short message signal such as "Trial data" in the case of an audio signal is partially overlapped. . Specifically, for example, a signal obtained by repeating a short message signal such as “Trial data” every 20 seconds and superimposing it on a music signal is used for trial listening. To remove this message, the signal is superimposed. The music is sold only by replacing the true music signal (original signal) with a code string obtained by encoding only the frame in which the music is recorded. Here, for example, assuming that the message "Preview data" is reproduced for 2 seconds and is repeated at 20-second intervals, the data amount for replacing this message portion is 1/1 / of the quality-enhanced data. The size is about 10. Therefore, the time required for the user to obtain the quality-enhanced data is, in addition to the above-described data amount of the wideband, the quality-enhanced data for obtaining the data amount for replacing the message portion. Only about one-tenth of the acquisition time is required, and the whole time is about one-tenth or less of the time to acquire high-quality data. As will be described later, a code string having a portion obtained by coding a wideband signal not including dummy data and a portion obtained by coding a signal obtained by combining another signal such as the message with the wideband signal is used. It may be used.
[0081]
Here, FIG. 17 shows an example of a preview file used in the embodiment of the present invention, in which the above-mentioned message signal "Trial data" is superimposed on music at predetermined intervals. The encoded signal is embedded in the code sequence. The hatched portion in FIG. 17 is a portion (second frame portion) in which the frame FA in which the message signal is superimposed on the music described above is arranged (second frame portion). This is a portion (first frame portion) corresponding to the first code string in which low-quality frames FB as data are arranged. Note that the first frame portion may be composed of a code string that does not include dummy data and can be reproduced with high sound quality, or a frame with high sound quality and a frame with low sound quality may be mixed.
[0082]
FIG. 18 shows a portion of a rewrite frame data sequence for replacing a code sequence of a frame in which a message signal is superimposed on a music signal with a code sequence of only a music signal in the preview file of FIG. It is a specific example of a code sequence of a quality-enhanced file including a portion of a frame data sequence for quality improvement including true data for replacing dummy data of a portion. A rewrite start frame ID and a rewrite end frame ID are coded for a portion to which each message is added (a portion where the frame FA in FIG. 17 is arranged), and subsequently, a code string (only a real music signal is coded) Rewriting frame data strings). Subsequently, a frame data sequence (the second code) for enhancing the low sound quality portion (the portion where the frames FB in FIG. 17 are arranged, corresponding to the first code sequence for trial viewing) is provided. (Corresponding to a row). FIG. 19 shows an example of one frame of the high-quality sound frame data sequence. In the specific example of FIG. 18, the rewrite start frame ID and the rewrite end frame ID are encoded. However, if the portion where the message is to be superimposed is determined from the beginning, the information is not encoded. May be.
[0083]
FIG. 19 shows an example of a true code string for one frame of the above-described high-quality sound frame data string, and is used to change the information of the Nth frame shown in FIG. 14 to the information shown in FIG. As a result, in the code string containing the dummy data (portion where the frames FB are arranged in FIG. 17), the reproduced sound having the spectrum shown in FIG. 15 is changed to the reproduced sound having the spectrum shown in FIG. become. Here, it is assumed that part of the spectrum coefficient information of the Neg portion in FIG. 14 has been replaced with dummy data.
[0084]
Here, in the above-described embodiment, the first frame portion is the first code string for trial viewing including the dummy data. However, the first frame portion has high sound quality (broadband) without the dummy data. A code string may be used. In this case, the high-quality sound frame data string shown in FIG. 19 is unnecessary. In the first frame portion, the first code string for trial viewing including dummy data and a code string of high sound quality not including dummy data may be mixed.
[0085]
As described above, when a high-quality (broadband) code string not including the dummy data is used for at least a part of the first frame portion, there is an advantage that the sound quality of the wideband data can be confirmed.
[0086]
Next, FIG. 20 is a block diagram showing an example of a reproducing apparatus used in the embodiment of the present invention, in which the conventional decoding means of FIG. 5 is improved. In the reproducing apparatus of FIG. 20, a code string of a signal on which a message is superimposed (a part where frames FA are arranged) and a code string (a part of which is replaced with dummy data) as shown in FIG. A code string 801 in which a portion where the frames FB are arranged, corresponding to the first code string of low sound quality) is alternately arranged at a predetermined interval. The input code sequence 801 is, for example, a sequence of frames FA on which the above messages are superimposed for 2 seconds and a sequence of frames FB of low sound quality continuing for 18 seconds alternately (total of 20 frames). Code sequence that appears in a second cycle). The period can be set arbitrarily and may not be periodic.
[0087]
Such an input code string 801 is first sent to the code string rewriting means 1801. On the other hand, the control unit 1804 receives a code sequence (a code sequence for improving quality) 805 in the format shown in FIG. The control means 1804 controls the code string rewriting means 1801 to convert the code string of the signal in the input code string 801 on which the message is superimposed (the part where the frame FA of FIG. 17 is arranged) of FIG. FIG. 18 shows the dummy data portion of the code sequence (the portion where the frames FB in FIG. 17 are arranged) in which the part of the input code sequence 801 is replaced with dummy data. And the result is sent to the signal component decoding means 1802 as data 802. The signal component decoding means 1802 decodes the data 802 into spectrum data 803, and the inverse conversion means 1803 converts the data into time series data 804, and does not include a message. To play. However, under the control of the control unit 1804, the input code string 801 is decoded and reproduced as it is without rewriting the code string.
[0088]
FIG. 21 is a block diagram illustrating an example of a recording apparatus used in the embodiment of the present invention. In FIG. 21, the input code sequence 821 is composed of a portion in which the frames FA are arranged as shown in FIG. 17 and a portion in which the frames FB are arranged alternately, as in the case of the input code sequence 801 in FIG. The code sequence is sent to the code sequence rewriting unit 1821. The code string 824 input to the control means 1823 is a code string for improving the quality of the format shown in FIG. 18, like the code string 805 input to the control means 1804 in FIG. The control means 1823 controls the code string rewriting means 1821 to convert the code string of the signal in the input code string 821 on which the message is superimposed (the part of the frame FA in FIG. 17) into the rewriting frame in FIG. The dummy data part of the code string (the part of the frame FB in FIG. 17) in which the input code string 821 is partially rewritten with the dummy data is replaced with the frame data for high sound quality in FIG. The code string 822 as a result is sent to the recording unit 1822. The recording unit 1822 records the code string 823 on a recording medium. Here, the recording medium on which the code string 823 is recorded may be the recording medium on which the code string 821 was originally recorded. Further, under the control of the control unit 1823, the input code string 821 is recorded as it is on another recording medium without rewriting the code string.
[0089]
FIG. 22 is a flowchart illustrating an example of a case where reproduction is performed using software as an embodiment of the present invention.
[0090]
First, in step S11, a variable (frame number) J is initialized (J = 1), and in the next step S12, the J-th frame, which is the current processing target frame in the input code string, overlaps the message signal. It is determined whether or not the received signal is a frame with a message (frame FA in FIG. 17). If the determination is NO, the process proceeds to step S15a described later, and if the determination is YES, the process proceeds to the next step S13. In step S13, it is determined whether or not to play only music (reproduction of music not including a message). If YES, the process proceeds to the next step S14, and if NO, the process proceeds to step S15a described later. In step S14, the contents of the code string of the frame with the message (frame FA in FIG. 17) are rewritten to those of the music signal only (frame data for rewriting in FIG. 18), and the process proceeds to the next step S15a. In step S15a, it is determined whether or not high-quality sound reproduction is to be performed on the J-th frame, which is the current frame in the input code string, that is, the frame of the low-quality sound part in which the current frame is partially replaced with dummy data. (Frame FB in FIG. 17), and it is determined whether or not to increase the bandwidth (improve the sound quality). In the case of YES for performing high-quality sound reproduction, the code sequence shown in FIG. 19 for the corresponding frame in the high-quality frame data sequence in FIG. 18 is received, and the process proceeds to step S15b. In the case of NO, Proceed to step S15c. In step S15b, the dummy data portion is rewritten with true data, and the process proceeds to step S15c. In step S15c, signal components are decoded to construct a spectrum signal, and the process proceeds to step S16, where the signal is inversely converted into a time-series audio signal, and the process proceeds to step S17. In step S17, it is checked whether or not the frame is the last frame. If YES, the process is terminated. If NO, the value of the frame number J is increased by 1 in step S18, and the process returns to step S12 to return to the above process. repeat. In this way, the user can listen to low-quality music with a message, or listen to high-quality music without extra messages by paying a price, or selectively reproduce an audio signal. . In this case, even a user who can be satisfied with music with low sound quality to some extent can not be satisfied with the addition of a message in many cases, and can motivate the purchase of high quality data.
[0091]
FIG. 23 is an example of a flowchart showing a procedure for performing recording using software in the recording method used in the embodiment of the present invention. In FIG. 23, in step S21, a variable (frame number) J is initialized (J = 1), and in the next step S22, the J-th frame which is the current frame is the frame with the message (the frame in FIG. 17). FA) is determined. When the determination is NO, the process proceeds to step S25a described later, and when the determination is YES, the process proceeds to the next step S23. In step S23, it is determined whether or not to play only music that does not include a message. If YES, the process proceeds to the next step S24, and if NO, the process proceeds to step S25a described later. In step S24, the content of the code string of the frame with the message is rewritten to a music signal only, and the process proceeds to the next step S25a. In step S25a, it is determined whether or not high-quality sound reproduction is to be performed, that is, whether or not the current frame is a frame of the low-quality sound part (the frame FB in FIG. 17) in which the above-mentioned part is replaced with dummy data and the band is broadened. Is determined. In the case of YES, the process proceeds to step S25b, in which the code sequence shown in FIG. 19 for the corresponding frame in the high-quality sound frame data sequence in FIG. 18 is received. In the case of NO, the process proceeds to step S25c. In step S25b, the dummy data portion is further rewritten with true data, and the process proceeds to step S25c. In step S25c, the code string of the frame is recorded, and the process proceeds to step S26. In step S26, it is checked whether or not the frame is the last frame. If YES, the process is terminated. If NO, the process proceeds to step S27 to increase the value of the frame number J by 1, and returns to step S22. The above processing is repeated. In this way, the recording of the audio signal is selectively performed, such as recording the low-quality music with the message, or recording only the high-quality music without the extra message by paying the price. Can be. In this case, even a user who can be satisfied with music with low sound quality to some extent can not be satisfied with the addition of a message in many cases, and can motivate the purchase of high quality data.
[0092]
In the embodiment of the present invention, it is desirable to enhance the security by encrypting a code string for replacing a music signal with a message signal as shown in FIG. 18 with only a music signal.
[0093]
In the above-described embodiment, the first frame portion is the first code string for trial viewing including the dummy data. However, a high-quality (wideband) code not including the dummy data is used. A sequence may be used, or a first code sequence for trial viewing including dummy data and a high-quality code sequence not including dummy data may be mixed.
[0094]
Next, when synthesizing the music signal and the message signal in the above-described embodiment, for example, when the music signal and the message signal are simply added together, if the music signal has a high level, the message signal is masked by the music signal. And the message signal cannot be clearly heard. Furthermore, simply adding together two high-level music signals may cause the level range to overflow, and in the event of an overflow, replacing the maximum or minimum level value. However, there is a problem that noise, such as a wobbling noise, may occur.
[0095]
Therefore, in the embodiment of the present invention, the above-described problem is solved by superimposing the message signal at a position where the message signal is superimposed, for example, by lowering the level of the original audio signal. Further, the level is not changed suddenly but is changed smoothly, thereby preventing generation of noise such as a click which occurs when the level changes rapidly.
[0096]
That is, generally, when a message signal is superimposed on at least a part of a first signal (for example, a music signal), at least a part of the part that superimposes the message signal on the first signal has the first signal. For a first synthesis coefficient R1 for the signal and a second synthesis coefficient R2 for the message signal,
| R1 | <| R2 |
Seeking to satisfy the relationship. Further, in the portion where the message signal is superimposed on the first signal, the second synthesis coefficient R2 at the boundary position with the portion including only the first signal continuously changes so as to have a value near 0. Coefficient value. Specifically, for example, the second synthesis coefficient R2 is gradually increased from 0 from the message signal superimposition start point, and the second synthesis coefficient R2 is gradually increased toward the message signal superimposition end point. To be reduced to zero. Further, the first combined coefficient R1 for the first signal and the second combined coefficient R2 for the message signal are both set to values equal to or greater than 0, and the sum of the combined coefficients R1 and R2 is set to 1 (R1 + R2 = 1).
[0097]
FIG. 24 is an explanatory diagram for describing an embodiment of a method of superimposing a message signal on a music signal described above. In this embodiment, a coefficient R1 to be multiplied by a music signal and a message signal The multiplication coefficient R2 takes a value of 0 or more, and is set so that the sum (R1 + R2) of both becomes exactly 1. As a result, first, no overflow occurs due to the combination of the two signals. In this embodiment, the coefficient of the message signal is larger than the coefficient of the music signal in the part where the actual message is contained. If the size is set close to, the message signal is not masked by the music signal and cannot be heard. Furthermore, in this embodiment, the coefficient R1 to be multiplied by the music signal is gradually and gradually (continuously) from 1.0 to 0.25 between the portion other than the message signal superimposition portion and the message signal superimposition portion. ) Since it is changing, it is possible to prevent the generation of noise such as a click which occurs when the level changes abruptly.
[0098]
In the example of FIG. 24, in the actual message section Tm within the message superimposition section Tmx, the music signal coefficient R1 is a constant 0.25, and the message signal coefficient R2 is a constant 0.75. In the message section Tm, since the music signal coefficient R1 is smaller than the message signal coefficient R2 (R1 <R2), the message signal is not masked by the music signal and cannot be heard. Start point t of message superimposition section Tmx _s , The music signal coefficient R1 is gradually and continuously changed from 1.0 to 0.25, and the message signal coefficient R2 is changed from 0.0 to 0.75. It is changing gradually and continuously. Similarly, from the end of the actual message section Tm to the end point t of the message superimposition section Tmx _e , The coefficient R1 for the music signal gradually changes continuously from 0.25 to 1.0, and the coefficient R2 for the message signal changes gradually continuously from 0.75 to 0.0. ing. This prevents the occurrence of signal switching noise and the like.
[0099]
The coefficients used for synthesizing these signals may be obtained from a calculation formula at the time of synthesizing, or coefficient values calculated in advance may be stored in a memory and read out at the time of synthesizing. You may do it. In this embodiment, it is assumed that a silence signal is included as a message signal in a section other than the actual message section Tm in the message superimposition section Tmx.
[0100]
Further, the respective synthesis coefficients R1 and R2 change with time as described above. Specifically, for example, using the sample position K in the message superimposition section of the digital signal, R1 (K), R2 (K), which are simply represented by R1 and R2 for ease of explanation, and R1 (K) and R2 (K) It shall be expressed as follows.
[0101]
Next, FIG. 25 shows a schematic configuration of an example of an embodiment of the signal superposition apparatus of the present invention. In FIG. 25, a music signal 841 and a message signal 842 are input to a signal combining unit 1841, and these signals are combined under the control of the control unit 1843 in a message superimposition section. For this combination, the control means 1843 reads out the coefficient 844 (coefficient R2) to be multiplied by the message signal from the coefficient storage means 1842, and subtracts the coefficient R1 to be multiplied by 1.0 from the coefficient R1 to be multiplied by the music signal by 1.0. (R1 = 1.0-R2), and based on these coefficients, the control signal 845 causes the signal combining means 1841 to perform signal combining in the message superimposition section. From the signal synthesizing unit 1841, a signal 843 in which a message signal is superimposed on a music signal is extracted. Thus, if the user listens to the PCM signal and purchases it if he / she likes it, only the sample signal in the message signal superimposed section is replaced with the original music signal, and the message signal that hinders music appreciation is quickly erased. be able to.
[0102]
Next, FIG. 26 is a block diagram illustrating a schematic configuration of an example of an embodiment of a signal superposition encoding device that superimposes and encodes a message signal on a music signal. The signal superposition coding apparatus shown in FIG. 26 is obtained by adding coding means to the signal superposition apparatus shown in FIG.
[0103]
In FIG. 26, a music signal 861 and a message signal 862 are input to a signal synthesizing unit 1861, and these signals are synthesized under the control of the control unit 1863 in a message superimposition section. The control unit 1863 reads out the coefficient 864 (coefficient R2) to be multiplied by the message signal from the coefficient storage unit 1862 and subtracts the coefficient R1 by which the music signal is multiplied from 1.0 by the coefficient R2 by which the message signal is multiplied from 1.0. (R1 = 1.0−R2), and based on these coefficients, the control signal 865 causes the signal combining means 1861 to perform signal combining in the message superimposed portion. An output 863 from the signal synthesizing unit 1861 is sent to, for example, an encoding unit 1864 as shown in FIG. 2 to generate a code string 866. As a result, if the user decodes this code string and listens to it and purchases it if he likes it, the user can simply replace only the coded frame including the message signal superimposed section with the original music signal, which can quickly hinder music appreciation. Can be turned off.
[0104]
Next, FIG. 27 is a flowchart for explaining a signal superposition method used in the embodiment of the present invention. In FIG. 27, first, in step S31, the value of the variable J representing the sample number is set to 1 at the same time as reading the first sample. Next, in step S32, whether this sample is a sample with a message signal is determined. Determine whether As a criterion for this determination, for example, assuming that a message of 44.1 kHz in monaural and a message of 2 seconds every 20 seconds is put, the above variable J is divided by (44100 × 20), and the remainder is (44100 × 3). If the value is less than (44100 × 5), it may be determined that the sample is in the message superimposition section. Otherwise, it may be determined that the sample is not the first sample in the message superimposition section. Further, as another determination criterion, for example, the sample numbers of the start position and the end position of the message superimposition section are stored in advance in the storage unit, and the determination is made based on whether the value of the variable J is within this range. It may be.
[0105]
If Yes (the sample with the message signal) is determined in step S32, the process proceeds to step S33 to determine whether the sample is the first sample in the message signal superimposition section. As a criterion for this determination, for example, assuming that a message of 44.1 kHz sampling in monaural and a message of 2 seconds every 20 seconds are put, the above variable J is divided by (44100 × 20), and the remainder is (44100 × 3). ), It may be determined that the frame is the first sample of the message superimposed section, and otherwise, it may be determined that the frame is only music. As another determination criterion, for example, a sample number of the start position of the message superimposition section may be stored in advance in the storage unit, and the determination may be made based on whether or not the value of the variable J is equal to this sample number.
[0106]
If Yes (the first sample in the message signal superimposition section) in step S33, the process proceeds to step S34, where the value of the variable K representing the sample number in the message superimposition section is set to 1, and the process proceeds to step S36. For example, the process proceeds to step S35 to increase the value of the variable K by 1 (K = K + 1), and then proceeds to step S36. In step S36, the message signal coefficient R2 (K) for the current sample is read from the coefficient storage means, and the flow advances to step S37. In step S37, a music signal coefficient R1 (K) for the current sample is obtained by subtracting R2 (K) from 1.0 (R1 (K) = 1.0-R2 (K)), and the process proceeds to step S38. move on.
[0107]
In step S38, the values of R2 (K) and R1 (K) are sent to the signal synthesizing means, and both the message signal and the music signal are synthesized using these coefficients R2 (K) and R1 (K). , And the process proceeds to step S39. On the other hand, if it is determined in step S32 that this sample is not a sample in the message superimposition section, the flow advances to step S41 to instruct the signal synthesizing unit to output a PCM signal containing only a music signal. Then, the process proceeds to step S39. In step S39, it is determined whether or not the next sample can be read. If No, the process ends. If Yes, the process proceeds to step S40 to read the next sample and set the value of the variable J to 1 by one. Increase (J = J + 1), return to step S32, and repeat the above processing.
[0108]
According to the embodiments of the signal superposition method and apparatus according to the present invention described above, or the signal superposition encoding method and apparatus, in a portion where a message is superimposed on music, a message signal is masked and becomes inaudible, Noise does not occur, and it is possible to provide a more comfortable listening sound in determining whether to obtain information necessary for high-quality playback after checking the contents of the software, and it is possible to provide a smoother listening sound Software can be distributed. In particular, it is possible to smoothly motivate a user who is satisfied even if the bandwidth becomes narrower to purchase high quality data.
[0109]
In the above description, the case where the same message signal is repeated and used has been described. Of course, the content of the message signal may be different depending on the location of the superimposed portion. Further, the arrangement interval of the message signal, the length of the message superimposition section, and the like may or may not be constant. Further, the method of the present invention is applicable to a case where a message is superimposed over the entire music signal.
[0110]
Further, in the above-described embodiment, a case has been described in which an encoded bit stream is recorded on a recording medium. However, the method of the present invention can be applied to a case where a bit stream is transmitted. For example, only those who have obtained a code string that replaces an audio signal with a broadcast message with a normal music signal can reproduce only the music signal, and the content is sufficient for other listeners. It can be understood, but it is also possible to play music with extra messages.
[0111]
Further, in the above-described embodiment, the case where an audio signal is used has been described as an example. However, the present invention can be applied to an image signal. That is, for example, an image with a message may be superimposed on a part of an image signal, and high quality may be achieved by a code string that replaces a part of the message signal. Also, when an image signal having a high luminance of a message character is superimposed on a part of the original image signal, the luminance of the original image signal is reduced and the signal of the message character is superimposed. Can be prevented from becoming difficult to see.
[0112]
Still further, the first frame portion in the embodiment is the first code string for trial viewing including the dummy data, but a high-quality (wideband) code string not including the dummy data. Alternatively, the first code string for trial viewing including dummy data and a high-quality code string not including dummy data may be mixed.
[0113]
【The invention's effect】
According to the present invention, when reproducing or recording a code string obtained by encoding an original signal in frame units, a first frame portion and a signal obtained by combining another signal with the original signal are encoded. A code sequence having a second frame portion composed of a converted code sequence is input, and the second frame portion is replaced with a code sequence obtained by encoding the corresponding frame portion of the original signal, thereby obtaining software contents. , It is possible to determine whether or not information necessary for high-quality reproduction should be obtained, and it is possible to distribute software more smoothly.
[0114]
Further, according to the present invention, when reproducing or recording a code string obtained by encoding the original signal in frame units, a part of the code string is composed of a first code string which is dummy data. A code sequence having a first frame portion and a second frame portion formed of a code sequence obtained by encoding a signal obtained by combining the original signal with another signal is input, and the second frame portion is converted to the original signal. By replacing the corresponding frame part with an encoded code string and complementing at least a part of the first code string with a second code string, the contents of the software are confirmed and high-quality reproduction is performed. It is possible to determine whether necessary information should be obtained, and it is possible to distribute software more smoothly, especially for users who are satisfied even if the bandwidth becomes narrower. Purchasing data It is possible to motivate.
[0115]
Further, according to the present invention, when a message signal is superimposed on at least a part of the first signal, a portion where the message signal is superimposed on the first signal is subjected to a first synthesis with respect to the first signal. For the coefficient R1 and the second synthesis coefficient R2 for the message signal,
| R1 | <| R2 |
Satisfies the relationship, the message signal is not masked in the portion where the message signal is superimposed on the first signal such as music, so that it becomes difficult to hear. Further, regarding the above coefficients, both values are set to 0 or more,
R1 + R2 = 1
Satisfies the relationship, it is possible to prevent the occurrence of noise in the portion where the message signal is superimposed on the first signal such as music.
[Brief description of the drawings]
FIG. 1 is a block diagram showing a schematic configuration of an example of an optical disk recording / reproducing apparatus provided for describing an embodiment of the present invention.
FIG. 2 is a block diagram illustrating a schematic configuration of an example of an encoding device used for describing an embodiment of the present invention.
FIG. 3 is a block diagram illustrating a specific example of a conversion unit of the encoding device of FIG. 2;
FIG. 4 is a block diagram showing a specific example of a signal component encoding unit of the encoding device in FIG. 2;
FIG. 5 is a block diagram illustrating a schematic configuration of an example of a decoding device used for describing an embodiment of the present invention.
6 is a block diagram showing a specific example of an inverse transform unit of the decoding device in FIG.
FIG. 7 is a block diagram illustrating a specific example of a signal component decoding unit of the decoding device in FIG. 5;
FIG. 8 is a diagram for describing an encoding method used for describing an embodiment of the present invention.
FIG. 9 is a diagram for explaining an example of a code string obtained by an encoding method used for describing an embodiment of the present invention.
FIG. 10 is a diagram for explaining another example of an encoding method used for describing an embodiment of the present invention.
FIG. 11 is a block diagram illustrating an example of a signal component encoding unit for implementing the encoding method described with reference to FIG. 10;
FIG. 12 is a block diagram illustrating an example of a signal component decoding unit used in a decoding device for decoding a code string obtained by the coding method described with reference to FIG. 10;
FIG. 13 is a diagram illustrating an example of a code string obtained by the coding method described with reference to FIG. 10;
FIG. 14 is a diagram illustrating an example of a code string obtained by an encoding method used in the base technology of the embodiment of the present invention.
FIG. 15 is a diagram illustrating an example of a spectrum of a reproduced signal when a code string obtained by the encoding method described with reference to FIG. 14 is reproduced.
FIG. 16 is a diagram illustrating an example of a spectrum of a reproduced signal when a code string obtained by another example of the encoding method described with reference to FIG. 14 is reproduced.
FIG. 17 is a diagram illustrating an example of a code string obtained by the coding method used in the embodiment of the present invention.
FIG. 18 is a diagram showing an example of a code sequence for improving the quality of a code sequence obtained by the encoding method used in the embodiment of the present invention.
FIG. 19 is a diagram showing an example of information for replacing dummy data of a frame of a low sound quality part in a code string obtained by the coding method used in the embodiment according to the present invention.
FIG. 20 is a block diagram illustrating an example of a schematic configuration of a playback device used in the embodiment of the present invention.
FIG. 21 is a block diagram illustrating an example of a schematic configuration of a recording apparatus used in an embodiment of the present invention.
FIG. 22 is a flowchart for describing a reproducing method used in the embodiment of the present invention.
FIG. 23 is a flowchart illustrating a recording method used in the embodiment of the present invention.
FIG. 24 is an explanatory diagram for describing a method of superimposing a message signal on a music signal according to an embodiment of the present invention.
FIG. 25 is a block diagram illustrating a schematic configuration of an example of an embodiment of a signal superimposing device that superimposes a message signal on a music signal.
FIG. 26 is a block diagram illustrating a schematic configuration of an example of an embodiment of a signal superimposition encoding apparatus that superimposes and encodes a message signal on a music signal.
FIG. 27 is a flowchart illustrating a signal superposition method used in the embodiment of the present invention.
[Explanation of symbols]
1801, 1821 code string rewriting means, 1802 signal component decoding means, 1803 inverse conversion means, 1804, 1823, 1843, 1863 control means, 1822 recording means, 1841, 1861 signal synthesis means, 1842, 1862 coefficient storage means, 1864 encoding means

Claims (49)

In a signal reproducing method for reproducing a code string obtained by encoding an original signal in frame units,
A code sequence inputting step of inputting a code sequence having a first frame portion and a second frame portion formed of a code sequence obtained by encoding a signal obtained by combining another signal with the original signal;
A replacement step of replacing the second frame portion with a code string obtained by encoding a corresponding frame portion of the original signal;
A decoding step of decoding the first frame portion and the code string or the second frame portion replaced by the replacement step. The first frame portion includes a first code sequence in which a part of the code sequence of the original signal is dummy data,
A complementing step of supplementing at least a part of the dummy data of the first code string by using a second code string;
2. The decoding step, wherein the code string or the first code string complemented by the complementing step and the code string or the second frame replaced by the replacing step are decoded. The signal reproducing method described in the above. 2. The signal reproducing method according to claim 1, wherein the other signal is a message signal. 2. The signal reproducing method according to claim 1, wherein a code string obtained by coding the original signal replacing the second frame portion is encrypted. 2. The signal reproducing method according to claim 1, wherein the second frame portion is provided at predetermined intervals. In the encoding, the input signal is spectrally transformed and divided into bands, and a code string of a predetermined format including quantization accuracy information, normalization coefficient information, and spectrum coefficient information for each band is generated,
3. The signal reproducing method according to claim 2, wherein the dummy data includes data corresponding to a part of the spectrum coefficient information. In a signal reproducing apparatus that reproduces a code string obtained by encoding an original signal in frame units,
Code string input means for inputting a code string having a first frame part and a second frame part consisting of a code string obtained by coding a signal obtained by combining the original signal with another signal;
Code string rewriting means for replacing the second frame part with a code string obtained by encoding the corresponding frame part of the original signal;
A signal reproducing device, comprising: decoding means for decoding the first frame part and the code string or the second frame part replaced by the replacement means. The first frame portion includes a first code sequence in which a part of the code sequence of the original signal is dummy data,
The code string rewriting means replaces the second frame part with a code string obtained by encoding a corresponding frame part of the original signal, and replaces at least a part of the dummy data of the first code string with a second code part. Complement using a code string,
8. The signal reproducing apparatus according to claim 7, wherein the decoding unit decodes the complemented code sequence or the first code sequence and the replaced code sequence or the second frame. The signal reproducing device according to claim 7, wherein the other signal is a message signal. 8. The signal reproducing apparatus according to claim 7, wherein a code string obtained by coding the original signal replacing the second frame portion is encrypted. 8. The signal reproducing apparatus according to claim 7, wherein the second frame portion is provided at predetermined intervals. In the encoding, the input signal is spectrally transformed and divided into bands, and a code string of a predetermined format including quantization accuracy information, normalization coefficient information, and spectrum coefficient information for each band is generated,
9. The signal reproducing apparatus according to claim 8, wherein a part of the dummy data of the first code string includes a part of the spectrum coefficient information. A signal recording method for recording a code string obtained by encoding an original signal in frame units,
A code sequence inputting step of inputting a code sequence having a first frame portion and a second frame portion formed of a code sequence obtained by encoding a signal obtained by combining another signal with the original signal;
Replacing the second frame portion with a code string obtained by encoding the corresponding frame portion of the original signal. The first frame portion includes a first code sequence in which a part of the code sequence of the original signal is dummy data,
14. The signal recording method according to claim 13, further comprising a complementing step of supplementing at least a part of the dummy data of the first code string by using a second code string. 14. The signal recording method according to claim 13, wherein the other signal is a message signal. 14. The signal recording method according to claim 13, wherein a code string obtained by coding the original signal replacing the second frame portion is encrypted. 14. The signal recording method according to claim 13, wherein the second frame portion is provided at predetermined intervals. In the encoding, the input signal is spectrally transformed and divided into bands, and a code string of a predetermined format including quantization accuracy information, normalization coefficient information, and spectrum coefficient information for each band is generated,
15. The signal recording method according to claim 14, wherein the dummy data includes data corresponding to a part of the spectrum coefficient information. In a signal recording device that records a code string obtained by encoding the original signal in frame units,
Code string input means for inputting a code string having a first frame part and a second frame part consisting of a code string obtained by coding a signal obtained by combining the original signal with another signal;
A signal sequence rewriting means for replacing the second frame portion with a code sequence obtained by encoding the corresponding frame portion of the original signal. The first frame portion includes a first code sequence in which a part of the code sequence of the original signal is dummy data,
The code string rewriting means replaces the second frame part with a code string obtained by coding a corresponding frame part of the original signal, and replaces at least a part of the dummy data of the first code string with a second code part. 20. The signal recording apparatus according to claim 19, wherein complementation is performed using a code string. 20. The signal recording apparatus according to claim 19, wherein the other signal is a message signal. 20. The signal recording apparatus according to claim 19, wherein a code string obtained by encoding the original signal replacing the second frame portion is encrypted. 20. The signal recording apparatus according to claim 19, wherein the second frame portion is provided at predetermined intervals. In the encoding, the input signal is spectrally transformed and divided into bands, and a code string of a predetermined format including quantization accuracy information, normalization coefficient information, and spectrum coefficient information for each band is generated,
21. The signal recording apparatus according to claim 20, wherein a part of the dummy data of the first code string includes a part of the spectrum coefficient information. In a code string generation method for generating a code string obtained by encoding an original signal in frame units,
A code sequence generating step of generating a code sequence having a first frame portion and a second frame portion formed of a code sequence obtained by coding a signal obtained by combining the original signal with another signal. Code string generation method. The first frame portion includes a first code sequence in which a part of the code sequence of the original signal is dummy data,
The code string generating step includes:
Encoding step of generating a code string of a predetermined format by encoding the original signal,
A first code string generating step of generating a first code string by rewriting a part of the code string of the predetermined format into dummy data;
26. A code string generation method according to claim 25, further comprising: a second code string generation step of generating a second code string for complementing at least a part of the first code string. 26. The method according to claim 25, wherein the other signal is a message signal. 26. The method according to claim 25, wherein the second frame portion is provided at predetermined intervals. In the encoding, the input signal is spectrally transformed and divided into bands, and a code string of a predetermined format including quantization accuracy information, normalization coefficient information, and spectrum coefficient information for each band is generated,
27. The method according to claim 26, wherein the dummy data includes data corresponding to a part of the spectrum coefficient information. In the second frame portion, a first synthesis coefficient R1 for the original signal and a second synthesis coefficient R2 for the other signal are:
| R1 | <| R2 |
26. The method according to claim 25, wherein the following relationship is satisfied. In at least a part of the second frame portion, a coefficient value that changes continuously so that the second combined coefficient R2 at a boundary position with the first frame portion becomes a value near 0 is taken. 31. The method according to claim 30, wherein: The first combined coefficient R1 and the second combined coefficient R2 are both set to values equal to or greater than 0,
R1 + R2 = 1
31. The method according to claim 30, wherein the following relationship is satisfied. In a recording medium in which a code string obtained by encoding an original signal in frame units is recorded,
A recording medium characterized by recording a code sequence having a first frame portion and a second frame portion made up of a code sequence obtained by encoding a signal obtained by combining the original signal with another signal. 34. The recording medium according to claim 33, wherein the first frame portion comprises a first code string in which a part of the code string of the original signal is dummy data. The recording medium according to claim 33, wherein the other signal is a message signal. The recording medium according to claim 33, wherein the second frame portion is provided at predetermined intervals. In the encoding, the input signal is spectrally transformed and divided into bands, and a code string of a predetermined format including quantization accuracy information, normalization coefficient information, and spectrum coefficient information for each band is generated,
The recording medium according to claim 34, wherein the dummy data includes data corresponding to a part of the spectrum coefficient information. In a signal superimposition method for superimposing a message signal on at least a part of a first signal,
In at least a part of a portion where a message signal is superimposed on the first signal, a first synthesis coefficient R1 for the first signal and a second synthesis coefficient R2 for the message signal
| R1 | <| R2 |
Satisfies the following relationship: In a portion where a message signal is superimposed on the first signal, the second combination coefficient R2 at a boundary position between the first signal and the portion including only the first signal continuously changes so as to have a value near 0. 39. The signal superposition method according to claim 38, wherein the method is a numerical value. The first combined coefficient R1 for the first signal and the second combined coefficient R2 for the message signal are both set to values equal to or greater than 0,
R1 + R2 = 1
39. The signal superposition method according to claim 38, wherein the following relationship is satisfied. Signal synthesizing means to which a first signal and a message signal are input, and for superimposing a message signal on at least a part of the first signal;
Coefficient storage means or coefficient generation means for storing at least one of a first synthesis coefficient R1 for the first signal and a second synthesis coefficient R2 for the message signal;
Control means for controlling the synthesis of the first signal and the message signal by the signal synthesis means based on the synthesis coefficient from the coefficient storage means or the coefficient generation means,
The control means may include, for at least a part of a portion where a message signal is superimposed on the first signal, for the first combined coefficient R1 and the second combined coefficient R2,
| R1 | <| R2 |
A signal superposition apparatus that controls signal synthesis so as to satisfy the following relationship. In a portion where a message signal is superimposed on the first signal, the second combination coefficient R2 at a boundary position between the first signal and the portion including only the first signal continuously changes so as to have a value near 0. The signal superposition apparatus according to claim 41, wherein the signal superposition apparatus takes a numerical value. The first combined coefficient R1 for the first signal and the second combined coefficient R2 for the message signal are both set to values equal to or greater than 0,
R1 + R2 = 1
42. The signal superposition apparatus according to claim 41, wherein the following relationship is satisfied. In a signal superposition encoding method for superimposing a message signal on at least a part of a first signal and encoding the superimposed signal,
In at least a part of a portion where a message signal is superimposed on the first signal, a first synthesis coefficient R1 for the first signal and a second synthesis coefficient R2 for the message signal
| R1 | <| R2 |
Satisfies the following relationship: In a portion where a message signal is superimposed on the first signal, the second combination coefficient R2 at a boundary position between the first signal and the portion including only the first signal continuously changes so as to have a value near 0. The signal superposition encoding method according to claim 44, wherein a numerical value is taken. The first combined coefficient R1 for the first signal and the second combined coefficient R2 for the message signal are both set to values equal to or greater than 0,
R1 + R2 = 1
The signal superposition encoding method according to claim 44, wherein the following relationship is satisfied. Signal synthesizing means to which a first signal and a message signal are input, and for superimposing a message signal on at least a part of the first signal;
Coefficient storage means or coefficient generation means for storing at least one of a first synthesis coefficient R1 for the first signal and a second synthesis coefficient R2 for the message signal;
Control means for controlling the synthesis of the first signal and the message signal by the signal synthesis means based on the synthesis coefficient from the coefficient storage means or the coefficient generation means;
Encoding means for encoding the output signal from the signal synthesizing means,
The control means may include, for at least a part of a portion where a message signal is superimposed on the first signal, for the first combined coefficient R1 and the second combined coefficient R2,
| R1 | <| R2 |
Characterized in that signal synthesis is controlled so as to satisfy the following relationship: In a portion where a message signal is superimposed on the first signal, the second combination coefficient R2 at a boundary position between the first signal and the portion including only the first signal continuously changes so as to have a value near 0. 48. The signal superposition encoding apparatus according to claim 47, wherein the apparatus takes a numerical value. The first combined coefficient R1 for the first signal and the second combined coefficient R2 for the message signal are both set to values equal to or greater than 0,
R1 + R2 = 1
The signal convolutional encoding device according to claim 47, wherein the following relationship is satisfied.

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