JP5157931B2 - Information embedding device for acoustic signal and position detecting device using acoustic signal - Google Patents

Abstract

<P>PROBLEM TO BE SOLVED: To provide an apparatus for embedding information in a sound signal and an apparatus for detecting a position using the sound signal, which specifies a detailed position in two dimensions by using sounds emitted from the smallest possible number of speakers. <P>SOLUTION: Nine kinds of IDs composed of bit strings of a prescribed length are prepared. The sound signal, produced by varying the strength of a prescribed frequency component, based on a bit value of the ID11 and a difference in the bit values between the ID11 and the ID31, and the ID11 and the ID13, is outputted from an FL speaker. The same processing is performed for other channels and the sound signals produced by varying the strength of the prescribed frequency component are outputted from an FR speaker, a BL speaker and a BR speaker. In each microphone installed in a sphere surrounded by the four speakers, an ID33 is extracted from the ID11, and the corresponding position is detected from the relation in correspondence between the ID and the position which is recorded beforehand. <P>COPYRIGHT: (C)2010,JPO&amp;INPIT

Description

  The present invention relates to a technique for detecting a position using information embedded in an acoustic signal.

  In recent years, GPS technology has been adopted for mobile phones and the like, and it has become possible to detect the position of a terminal. However, since position detection using GPS has a low resolution, there is a problem that it is not sufficient for specifying an indoor position.

  On the other hand, the applicant changes the ratio of the frequency components of the acoustic signal according to the bit value of the attribute information so that it can be applied to a music attribute information providing service that can know the title of the music being played. Thus, a technique for embedding attribute information (additional information) has been proposed (see Patent Document 1).

In addition, the applicant solves the problem that in the invention described in Patent Document 1, when two speakers are separated by a wide hall or the like, the sound from both channels cannot be extracted near the center. Therefore, a technique for embedding and extracting information using the principle of sound pulse fractionation has been proposed (see Patent Document 2).
JP 2006-323246 A JP 2008-256948 A

  As described above, since position detection using GPS has low resolution, there is a problem that it is not sufficient to specify a detailed position indoors. By using the techniques of Patent Documents 1 and 2 above, if information for specifying the position is embedded in the speakers emitted from each, it can be used for position detection. There is a problem of having to install a speaker. In addition, there is a problem that not only position detection in one direction but also position detection in two dimensions cannot be performed.

  Therefore, the present invention uses an information embedding device for an acoustic signal capable of specifying a detailed position in two dimensions using sounds emitted from as few speakers as possible and position detection using the acoustic signal. It is an object to provide an apparatus.

In order to solve the above-described problem, in the present invention, when the acoustic signal is reproduced by at least four speakers with respect to an acoustic signal composed of a time-series sample sequence of at least four channels, N ² types of identification information that are different from each other are extracted at a total of n ² positions in the middle (n ² -5) position (n is an odd number of 5 or more ) of 4 locations and 4 locations, and the central location of 4 speakers. An apparatus for embedding n ² types of identification information different from each other in an inaudible state as a combination of n types of front / rear identification information and n types of left / right identification information. An acoustic frame reading means for reading a sample for each channel and creating an acoustic frame, and one of the odd numbered and even numbered acoustic frames of each read channel A type and the other B type, frequency conversion is performed on the A type acoustic frame using a first window function, and frequency conversion is performed on the B type acoustic frame using a second window function. Frequency conversion means for obtaining a first window spectrum that is a spectrum corresponding to the first window function, a second window spectrum that is a spectrum corresponding to the second window function, and a predetermined frequency from each of the generated window spectra Two spectrum sets that do not overlap each other in the frequency range are extracted so that the frequency width of the higher frequency spectrum set becomes larger , and each spectrum set is equally divided into ½ frequency widths, and each window spectrum is divided. 1D1, 1D2, 1U1, 1U in order from the lowest frequency of the spectrum sets extracted from the first window spectrum. And then, said spectral set extracted from the second window spectrum in order frequency is low, when the 2D1,2D2,2U1,2U2, based on the bit values constituting the original longitudinal or lateral identification information to be embedded into the channel One of the product of the intensity value of 1D1 and the intensity value of 2D2 and the product of the intensity value of 1D2 and the intensity value of 2D1 is larger than a predetermined level than the other, and simultaneously the intensity value of 1U1 and 2U2 and the product of the intensity values, which one of the product of the intensity values of the intensity values and 2U1 of 1U2 changes beyond a predetermined degree than the size Kunar so the other hand, when using a front-rear identification information, wherein In the range that does not reverse the magnitude relationship of intensity values, the degree to which the intensity value is changed is set to (n + 1) / 2 levels, and is extracted almost at the center with other speakers. When the bit value of the front and rear identification information to be performed is the same as the bit value of the original front and rear identification information, the maximum value is obtained, and before and after the original front and rear identification information is extracted at a position adjacent to the position. When the bit value of the identification information is different from the bit value of the original front-and-rear identification information, it is set to a minimum degree, and when using the left-right identification information, in a range that does not reverse the magnitude relationship of the intensity values, The degree to which the intensity value is changed is set to (n + 1) / 2 stages, and the bit value of the left / right identification information to be extracted at the approximate center between the other speakers is the same as the bit value of the original left / right identification information. In some cases, the bit value of the left / right identification information to be extracted at the position adjacent to the position where the original left / right identification information is extracted is When the bit value of the original left / right identification information is different, the frequency component changing means is set to the minimum degree, and the frequency spectrum is inversely transformed for each window spectrum including the changed spectrum set to be modified. Provided is an information embedding device for an acoustic signal, comprising frequency inverse transform means for generating an acoustic frame and modified acoustic frame output means for sequentially outputting the generated modified acoustic frames.

In the present invention, the acoustic signal is reproduced by at least four speakers from the acoustic signal in which n ² types of identification information are embedded in a state incapable of being heard in advance. A device for detecting the position of a total of n ² locations, that is, an intermediate (n ² -5) location and a central location of 4 speakers, and a positional relationship in which a relative position based on each speaker is associated with the identification information And a positional relationship storage means for storing the sound signal and monaural input of the sound signal at a predetermined position in the space where the sound signal is reproduced, digitizing the predetermined section, and obtaining an acoustic frame composed of a predetermined number of samples Sound frame acquisition means for performing the operation, and the odd-numbered and even-numbered ones of the read sound frames are A type and the other is B type. Then, frequency conversion is performed using the first window function, frequency conversion is performed using the second window function for the B type acoustic frame, and the first window spectrum which is a spectrum corresponding to the first window function The frequency conversion means for obtaining a second window spectrum that is a spectrum corresponding to the second window function, and two spectrum sets that do not overlap each other in a predetermined frequency range from each of the generated window spectra, have a higher frequency. The spectrum set is extracted so that the frequency width of the spectrum set becomes larger, each spectrum set is equally divided into ½ frequency widths, and the intensity values of the four spectrum sets extracted from the first window spectrum are low in frequency. E1, E3, E5, E7 are calculated in order, and the intensity values of the four spectrum sets extracted from the second window spectrum are E2, E4, 6 and E8, embedded bits based on the relationship between the product of E1 and E4 and the product of E2 and E3, and the product of E5 and E8 and the product of E6 and E7 Encoding means for extracting a value, and converting the extracted bit value in word units according to a predetermined rule to extract n kinds of front and rear identification information and left and right identification information in time series, an identification information extraction means for extracting as n ² kinds of identification information, by using the identification information obtained by said identification information extracting means, obtains position information by referring to the positional relationship storage means, the acquired position information A position detection device using an acoustic signal having a position output means for outputting the signal is provided.

According to the present invention, a total of n ² types of original identification information for each of at least four channels, (n ² -5) types of identification information for intermediate detection, and one type of identification information for the center of the four speakers Are prepared, and when the bit values corresponding to each of the four channels are embedded, the intensity level of the frequency component to be changed in a predetermined frequency range of each channel is stepwise according to the bit value of the identification information for intermediate detection. Since the sound is changed, the sound acquired by the microphones installed at predetermined intervals between the speakers is greatly affected by the sound of any speaker in the predetermined frequency range. ^2-4 ) It is possible to extract the identification information of the species, and it is possible to specify the relative position in the range surrounded by the four speakers by the extracted information.

Hereinafter, embodiments of the present invention will be described in detail with reference to the drawings.
(1. Basic concept of the present invention)
(1.1. One-dimensional position detection)
In describing the basic concept of the present invention, one-dimensional position detection that is the basis of the present invention will be described. FIG. 1A is a diagram illustrating extraction of identification information using an acoustic signal in the related art (techniques disclosed in Patent Documents 1 and 2, etc.). In the example of FIG. 1A, ID1 is embedded as identification information in the L channel acoustic signal, and ID2 is embedded as identification information in the R channel acoustic signal. When such an acoustic signal is reproduced, the extraction device extracts ID1 and ID2 from the sound captured from the microphone near the L speaker and the sound captured from the microphone near the R speaker, respectively. However, the extraction device cannot correctly extract the ID from the sound captured from the microphones that are approximately equidistant from the L speaker and the R speaker. This is because the sound from the L speaker and the sound from the R speaker are mixed in a microphone that is substantially equidistant from the L speaker and the R speaker. With the technique disclosed in Patent Document 2, it is possible to extract either ID1 or ID2 near the center, but it is not possible to extract an ID different from ID1 and ID2.

  The applicant proposed in Japanese Patent Application No. 2008-24450 a technique for extracting an ID different from the vicinity of the left and right speakers near the center of the left and right speakers. In the invention of Japanese Patent Application No. 2008-244502, the extraction device extracts ID1 and ID2 from the sound taken in from the microphone near the L speaker and the sound taken in from the microphone near the R speaker, respectively. A separate ID3 is extracted from the sound captured from the microphones that are approximately equidistant from the speaker and the R speaker. That is, the sound from the L speaker in which ID1 is embedded and the sound from the R speaker in which ID2 is embedded are simultaneously captured to extract ID3.

  Furthermore, in Japanese Patent Application No. 2008-312741, as shown in FIG. 1B, from the sound taken in from the microphone near the L speaker and the sound taken in from the microphone near the R speaker, each of the extraction devices is ID1. , ID5 is extracted, and separate ID3 is extracted from the sound captured by the microphones that are approximately equidistant (center) from the L speaker and the R speaker. A separate ID2 is extracted from the sound captured by the above step, and a separate ID4 is extracted from the sound captured by a microphone that is approximately equidistant from the microphone near the R speaker and the center microphone. That is, the sound from the L speaker in which ID1 is embedded and the sound from the R speaker in which ID5 is embedded are simultaneously captured to extract ID2, ID3, and ID4.

  The concept of Japanese Patent Application Nos. 2008-244502 and 2008-312741 will be described in more detail with reference to FIG. In FIG. 2, “0” and “1” indicate embedded bit values, and rectangles including “0” and “1” indicate acoustic frames in which bit values are embedded. The acoustic frame will be described in detail later, but indicates one section of a digital acoustic signal having a predetermined length. In practice, as will be described later, one bit is embedded in one acoustic frame, but this is not simplified in FIG. In FIG. 2, the horizontal direction in the drawing indicates the time length, and the vertical direction in the drawing indicates the intensity of the frequency component in the predetermined frequency range. FIG. 2 (a) shows the invention described in Japanese Patent Application No. 2008-24450. In Japanese Patent Application No. 2008-244502, one type of identification information is extracted at the center, and a total of three types of identification information is extracted. In the example of FIG. 2A, codes “01 (hexadecimal notation)” and “EF (hexadecimal notation)” are embedded as identification information for the L speaker and the R speaker, respectively. According to the bit string “01001011” constituting the code “4B (hexadecimal notation)” that is the power identification information, the intensity of the frequency component in the predetermined frequency range is changed and recorded. Thereby, when the sound acquired by the central microphone is analyzed, the code “4B” is extracted as the identification information.

  In Japanese Patent Application No. 2008-312741, the process further proceeds from Japanese Patent Application No. 2008-244501, and one type of identification information is extracted at the center, one or more pieces of identification information between the L speaker and the center, and one between the R speaker and the center. A total of n types of identification information are extracted. In the example of FIG. 2B, one piece of identification information is extracted between the middle of the L speaker and the middle of the R speaker, and a total of five types of identification information are extracted.

  In the example of FIG. 2B, codes “04 (hexadecimal notation)” and “DF (hexadecimal notation)” are embedded as identification information for the L speaker and the R speaker, respectively. At this time, each bit string “00000101” “00010111” constituting the codes “05 (hexadecimal notation)” “17 (hexadecimal notation)” “5F (hexadecimal notation)”, which is identification information to be extracted at three intermediate positions. According to “01011111”, the intensity of the frequency component in the predetermined frequency range is changed and recorded. As can be seen from the bit string rectangle corresponding to the first-stage code “04” and the bit-string rectangle corresponding to the fifth-stage code “DF” in FIG. The intensity of the frequency component in the predetermined frequency range for the speaker and the R speaker is changed in three stages of large, medium, and small. As a result, the predetermined frequency components of the sound acquired by the three microphones in the middle are affected, and when the predetermined frequency components of the acquired sound are analyzed, the codes “05”, “17”, and “5F” are respectively identified as identification information. Will be extracted as

  In FIG. 2B, an arrow extending in the vertical direction of the drawing from each bit of the first-stage bit string and the fifth-stage bit string indicates a range in which the bit affects. For example, since the leftmost bit “0” of the first-stage bit string has the strength “high”, it affects even a relatively far microphone, and the bit of the leftmost bit of the second to fourth-stage bit string. The value is “0”. Conversely, the leftmost bit “1” of the fifth-stage bit string has a strength of “small”, and does not affect the microphone at a relatively close position.

  For the second bit from the left end, the bit value “0” is “high” in the first bit string, which is the same as in the left bit, but the bit value “1” in the fifth bit string. Strength is “medium”. For this reason, in the fourth stage, the influence from the fifth stage is superior to the influence from the first stage, and the second bit value from the left is “1”. For the third bit from the left end, the bit values “0” and “high” are both “1” and “5” in the first and fifth bit strings. It becomes.

  In Japanese Patent Application No. 2008-312741, in order to change the intensity of the frequency component acquired by the intermediate microphone according to the intensity of the predetermined frequency range in the first stage and the fifth stage, different bit values are set in the middle microphone. It will not be detected. For example, in the case of the leftmost bit, the bit value “0” in the first stage affects up to the fourth stage, so that the bit values in the second and third stages in the middle become “1”. No, the first bit to the fourth row all have the same bit value “0”. When the second to eighth bit values from the left are viewed in the vertical direction (vertical direction in the drawing), there is only one difference in bit values between adjacent stages, and there is a difference between the two positions. Absent.

(1.2. Two-dimensional position detection)
Next, two-dimensional position detection, which is a feature of the present invention, will be described. FIG. 21A is a diagram illustrating a state in which different identification information is extracted from microphones arranged in a two-dimensional direction. In the example of FIG. 21A, ID11 is embedded as identification information in the FL channel acoustic signal, ID12 is embedded as identification information in the FR channel acoustic signal, ID21 is embedded as identification information in the BL channel acoustic signal, and the BR channel acoustic signal is embedded in the BR channel acoustic signal. ID 22 is embedded as identification information. When such an acoustic signal is reproduced, it is extracted from the sound captured from the microphone near the FL speaker, the sound captured from the microphone near the FR speaker, the sound captured from the microphone near the BL speaker, and the sound captured from the microphone near the BR speaker. The device extracts ID11, ID12, ID21, and ID22, respectively. Further, a separate ID 13 is extracted from the sound captured from the microphones of approximately equal distance from the FL speaker and the FR speaker, and a separate ID 31 is extracted from the sound captured from the microphone of approximately equal distance from the FL speaker and the BL speaker. A separate ID 32 is extracted from the sound captured from the microphones at approximately the same distance from the BR speaker, and a separate ID 23 is extracted from the sound captured from the microphones at the approximately equal distance from the BL speaker and the BR speaker. Then, a separate ID 33 is extracted from the sound captured from the central microphone (almost equidistant from the four speakers). That is, ID13, ID31, ID32, ID23, and ID33 are extracted by simultaneously capturing sounds from a plurality of speakers.

  In the present invention, the number of microphones is increased to extract more identification information. In the example of FIG. 21B, the number of microphones is increased to 25, and 25 types of identification information are extracted.

  The concept of the present invention will be described in more detail with reference to FIG. In FIG. 22, “0” and “1” indicate embedded bit values, and rectangles including “0” and “1” indicate acoustic frames in which bit values are embedded. The acoustic frame will be described in detail later, but indicates one section of a digital acoustic signal having a predetermined length. In practice, as will be described later, one bit is embedded in one acoustic frame, but this is not simplified in FIG. The length in the left-right direction of the rectangle including “0” and “1” shown in FIG. 22 indicates the time length. Of the rectangles including “0” and “1”, in the four sets close to the speaker, the intensity of the frequency component in the predetermined frequency range is expressed by the length in the vertical direction of the drawing.

  FIG. 22 shows a case where nine positions are detected. In this case, one type of identification information is extracted at the center, and a total of nine types of identification information is extracted. The identification information shown in FIG. 22 has a two-word configuration of one word and four bits, with the first half of the word specifying the front-rear direction and the latter half of the word specifying the left-right direction. In the example of FIG. 22, the codes “01 (hexadecimal notation)”, “0F (hexadecimal notation)”, “E1 (hexadecimal notation) are used as identification information for the FL speaker, the FR speaker, the BL speaker, and the BR speaker, respectively. Notation) ”and“ EF (hexadecimal notation) ”are embedded. At this time, codes “41 (hexadecimal notation)”, “0B (hexadecimal notation)”, “EB (hexadecimal notation)”, and “4F (hexadecimal notation)”, which are identification information to be extracted in the middle, are configured. In accordance with the bit string to be recorded, the intensity of the frequency component in the predetermined frequency range is changed and recorded.

  FIG. 23 shows a case where 25 positions are detected. In this case, one type of identification information is extracted at the center, and a total of 25 types of identification information are extracted. In the example of FIG. 23, the codes “0A (hexadecimal notation)”, “05 (hexadecimal notation)”, “FA (hexadecimal notation) are used as identification information for the FL speaker, the FR speaker, the BL speaker, and the BR speaker, respectively. Notation) ”and“ F5 (hexadecimal notation) ”are embedded. At this time, codes “1A (hexadecimal notation)” “3A (hexadecimal notation)” “7A (hexadecimal notation)” “0B (hexadecimal notation)” “09” are identification information to be extracted at the three intermediate positions. (Hexadecimal notation) "" 0D (hexadecimal notation) "" FB (hexadecimal notation) "" F9 (hexadecimal notation) "" FD (hexadecimal notation) "" 15 (hexadecimal notation) "" 35 (16 In accordance with each bit string constituting “75 (hexadecimal notation)”, the intensity of the frequency component in a predetermined frequency range is changed and recorded. The bit string rectangles corresponding to the codes “0A (hexadecimal notation)”, “05 (hexadecimal notation)”, “FA (hexadecimal notation)”, and “F5 (hexadecimal notation)” near the four speakers in FIG. As can be seen, in the present invention, the intensity of frequency components in a predetermined frequency range for the FL speaker, the FR speaker, the BL speaker, and the BR speaker is changed in three stages of large, medium, and small. Thereby, the predetermined frequency component of the sound acquired by each intermediate microphone is affected, and different identification information is extracted when the predetermined frequency component of the acquired sound is analyzed.

  In FIG. 23, the influence of the strength when embedding each bit on the detection position is the same as that described with reference to FIG. For example, since the leftmost bit “0” in the bit string near the FL speaker (first stage) has the strength “high”, it affects the microphone at a relatively far position, and the second to fourth stage bit strings. The bit value of the leftmost bit of “0” is “0”. Conversely, the leftmost bit “1” in the bit string near the BL speaker (fifth stage) has a strength of “small” and therefore does not affect the microphone at a relatively close position.

  For the second bit from the left end, the bit value “0” is the intensity “high” in the bit string near the FL speaker (first stage), which is the same as the case of the left end bit. The bit value “1” is the intensity “medium” in the bit string of the (stage). For this reason, in the fourth stage, the influence from the fifth stage is superior to the influence from the first stage, and the second bit value from the left is “1”.

  As in the case described with reference to FIG. 2B, in the example of FIG. 23, the intensity of the frequency component acquired by the intermediate microphone is changed according to the intensity of the predetermined frequency range in the first stage and the fifth stage. Therefore, different bit values are not detected in the midway microphone. For example, in the case of the leftmost bit, the bit value “0” in the first stage affects up to the fourth stage, so that the bit values in the second and third stages in the middle become “1”. No, the first bit to the fourth row all have the same bit value “0”. When the second to fourth bit values from the left are viewed in the vertical direction (vertical direction in the drawing), there is only one difference in bit values between adjacent stages, and there is a difference between the two positions. Absent.

  In the example of FIG. 23, the first half 4 bits of the first word are the same in the left-right direction, and the second half 4 bits of the second word are the same in the front-rear direction. The second word of the identification information extracted by the microphone arranged in the middle of the left and right is affected by the intensity value when each bit of the second word of the identification information near the left and right speakers is embedded. The state of this influence is the same as in the case of before and after and in the example of FIG.

  Next, how to change the strength when embedding codes of four channels will be described. When extracting nine types of identification information as shown in FIGS. 21A and 22, it is necessary to change the intensity in two stages, and 25 types as shown in FIGS. 21B and 23. When the identification information is extracted, it is necessary to change the intensity into three levels of “large”, “medium”, and “small”. Here, the case where the intensity is “large”, “medium”, and “small” will be described. FIG. 20 is a diagram illustrating a state of a predetermined frequency component in a pair of A-type sound frames and B-type sound frames (identified by odd-numbered or even-numbered). In the present invention, as will be described later, a predetermined number of samples obtained by sampling an acoustic signal are processed as one acoustic frame. In the case of 4-channel audio signals, the same processing is performed for FL-ch (front left channel), FR-ch (front right channel), BL-ch (rear left channel), and BR-ch (rear right channel). However, in FIG. 20, only the FL-ch (front left channel) is shown as a representative. In addition, left and right 1 and 2 indicate whether the acoustic frame is of type A or B, U and D indicate relative high and low frequency components in the entire frequency band to be changed, respectively, The difference between the low frequency side and the high frequency side in each of the high frequency and the low frequency is shown. Therefore, in the example of FIG. 20, two acoustic frames are shown. For example, “1U2” indicates a higher frequency side of the high frequency side of the A type acoustic frame. In addition, in the A-type and B-type acoustic frames, half of the samples are actually set to overlap, but in FIG. 20, they are shown in an independent form for convenience of explanation. In FIG. 20, the size of each frequency component character such as “1U2” indicates the relative strength.

  In the present invention, processing is performed so that the magnitude relationship between the component strengths on the high frequency side and the low frequency side of the odd-numbered acoustic frame and the magnitude relationship between the component strengths on the high frequency side and the low frequency side of the even-numbered acoustic frame are reversed. . The bit value to be embedded can be changed depending on whether the high-frequency side of the odd-numbered or even-numbered acoustic frame is increased. For example, in FIG. 20A, if the bit value “0” is embedded, the bit values “1” are embedded when 1U2 and 2U1 are increased and 1U1 and 2U2 are decreased. It becomes a state. The strengths “large”, “medium”, and “small” described with reference to FIG. 2 correspond to FIGS. 20A, 20B, and 20C, respectively. That is, the greater the strength is, the greater the difference between the higher strength side (1U1 and 2U2 in the example of FIG. 20) and the lower strength side (1U2 and 2U1 in the example of FIG. 20), and the strength is “small”. The difference between the higher strength side and the lower strength side becomes smaller. When extracting nine types of identification information, it is only necessary to use the strengths “large” and “small”.

(2.1. Configuration of embedded device)
Next, an information embedding device for an acoustic signal according to the present invention will be described. FIG. 3 is a functional block diagram showing a configuration of an information embedding device for an acoustic signal according to the present invention. In FIG. 3, 10 is an acoustic frame reading means, 20 is a frequency converting means, 30 is a frequency component changing means, 40 is a frequency inverse converting means, 50 is a modified acoustic frame output means, 60 is a storage means, and 61 is an acoustic signal storage section. , 62 is an identification information storage unit, 63 is a modified acoustic signal storage unit, and 70 is a bit array creation means. Note that the apparatus shown in FIG. 3 is compatible with stereo sound signals.

  The sound frame reading means 10 has a function of reading a predetermined number of samples as one frame from each channel of the original stereo sound signal to be embedded with identification information. The frequency conversion means 20 has a function of generating a spectrum by frequency-converting the frame of the acoustic signal read by the acoustic frame reading means 10 by Fourier transformation or the like. The frequency component changing unit 30 extracts a plurality of spectrum sets corresponding to a predetermined frequency range from the generated spectrum, and based on the bit arrangement created by the bit arrangement creating unit 70 from the identification information extracted from the identification information storage unit 62. The function of changing the state of the spectrum set is provided. The frequency inverse transform means 40 has a function of generating a modified acoustic frame by performing frequency inverse transform on a plurality of spectra including the changed spectrum set. The modified sound frame output means 50 has a function of sequentially outputting the generated modified sound frames.

  The storage unit 60 includes an acoustic signal storage unit 61 that stores a stereo acoustic signal to be embedded with identification information, an identification information storage unit 62 that is configured as a bit string and stores identification information embedded in the stereo acoustic signal, and identification information. It has a modified acoustic signal storage unit 63 for storing the modified acoustic signal after embedding, and stores various information necessary for other processing. The bit array creation means 70 has a function of extracting the identification information from the identification information storage unit 62 and creating a corresponding bit array by referring to the code conversion table for each word of the identification information.

  The identification information embedded in the acoustic information is information that can be uniquely identified by distinguishing each position from other positions in order to specify the position. In the present embodiment, in order to improve the accuracy of extracting identification information on the detection side, Nw-bit identification information is converted into Nh bits, and then processing for embedding each Nh-bit is performed. Although specific numerical values of Nw and Nh can be set as appropriate, in this embodiment, Nw and Nh are set to Nw = 8 and Nh = 16, respectively. In the present embodiment, the bit array created by the bit array creating means 70 is 16 bits, and after the bit array is created, the 16 bits are processed as one word. Each component shown in FIG. 3 is actually realized by mounting a dedicated program on hardware such as a computer and its peripheral devices. That is, the computer executes the contents of each means according to a dedicated program.

(2.2. Processing operation of embedded device)
Next, the processing operation of the information embedding device for the acoustic signal shown in FIG. 3 will be described. The acoustic frame reading means 10 reads a predetermined number N of samples as one acoustic frame from each of the front left, front right, rear left, and rear right channels of the stereo acoustic signal stored in the acoustic signal storage unit 61. The number N of samples of one acoustic frame read by the acoustic frame reading means 10 can be set as appropriate, but is desirably about 4096 samples when the sampling frequency is 44.1 kHz. Accordingly, the acoustic frame reading means 10 sequentially reads 4096 samples as acoustic frames for each of the front left, front right, rear left, and rear right channels.

  There are A type and B type as acoustic frames. In the A-type acoustic frame and the B-type acoustic frame, the next sample after the last sample of the preceding acoustic frame of the same type is set as the first sample. The A-type and B-type sound frames are set by overlapping a predetermined number (2048 in this embodiment) of samples. For example, if the A type acoustic frame is A1, A2, A3... From the top and the B type acoustic frame is B1, B2, B3... From the top, A1 is samples 1 to 4096, A2 is samples 4097 to 8192, A3. Is samples 8193-12288, B1 is samples 2049-6144, B2 is samples 6145-10240, and B3 is samples 10241-14336. Since the A type and the B type are relative, either one may be first. That is, contrary to the above, A1 is samples 2049 to 6144, A2 is samples 6145 to 10240, A3 is samples 10241 to 14336, B1 is samples 1 to 4096, B2 is samples 4097 to 8192, and B3 is samples 8193 to 12288. May be.

  The frequency converting means 20 performs frequency conversion on the acoustic frame after amplitude conversion, and obtains the spectrum of the acoustic frame. Specifically, frequency conversion is performed using a window function. As frequency conversion, Fourier transform, wavelet transform, and other various known methods can be used. In the present embodiment, a case where Fourier transform is used will be described as an example.

  In general, when Fourier transform is performed on a predetermined signal, it is necessary to divide the signal into predetermined lengths. In this case, if Fourier transform is performed on a signal of a predetermined length as it is, a pseudo-harmonic wave is generated. Ingredients are generated. Therefore, in general, when performing Fourier transform, a signal value is changed using a window function called a Hanning window, and then Fourier transform is performed on the changed value.

  In this embodiment, the window function is used, but the window function to be used is divided into the A type acoustic frame and the B type acoustic frame. In the present embodiment, the first window function W (1, i) and the second window function W (2, i) as shown in FIGS. 4A and 4B are prepared so that the extraction side can easily recognize them. I made it. The first window function W (1, i) is for use with an A type acoustic frame and takes a maximum value of 1 at the position of a predetermined sample number i as shown in FIG. In the rear part, the minimum value is set to 0. Which sample number takes the maximum value depends on the design of the window function W (1, i), but in this embodiment, it is defined by [Equation 1] described later. The Fourier transform for the A type acoustic frame is performed on the product of the window function W (1, i).

  The second window function W (2, i) is for use with a B-type acoustic frame, and has a maximum value at the position of a predetermined sample number i as shown in FIG. 1 is set, and the front portion is set to have a minimum value of 0. Which sample number takes the maximum value depends on the design of the window function W (2, i), but in this embodiment, it is defined by [Expression 2] described later. The Fourier transform for the B type acoustic frame is performed on the product of the window function W (2, i).

  As described above, in the present embodiment, the sound frame is read in duplicate. That is, a predetermined number of samples are redundantly read in the odd-numbered sound frames and the even-numbered sound frames. As described above, the window function used is different between the odd frame and the even frame, but since either the odd frame or the even frame is simply the difference between the odd frame and the even frame, either process may be performed on either. . Therefore, in this specification, one of the odd-numbered frame and the even-numbered frame is referred to as an A-type frame, and the other is referred to as a B-type frame. In the present embodiment, an odd frame is described as an A type frame and an even frame is described as a B type frame. Conversely, an even frame may be an A type frame and an odd frame may be a B type frame.

  In the present embodiment, the window functions W (1, i) and W (2, i) are defined by the following [Equation 1] and [Equation 2]. In FIG. 4, the horizontal axis is the time axis (i). As described later, i is a serial number assigned to N samples in each acoustic frame, and is proportional to time t. 4A and 4B, the vertical axis indicates the amplitude value (level) of the signal. 4A and 4B, the vertical axis indicates the values of the window functions W (1, i) and W (2, i), and the maximum values of W (1, i) and W (2, i). Are all 1.

[Formula 1]
When i ≦ N / 8, W (1, i) = 0.0
When N / 8 <i ≦ 3N / 8, W (1, i) = 0.5−0.5 cos (4π (i−N / 8) / N)
When 3N / 8 <i ≦ 11N / 16, W (1, i) = 1.0
When 11N / 16 <i ≦ 13N / 16, W (1, i) = 0.5 + 0.5 cos (8π (i-11N / 16) / N)
When i> 13N / 16, W (1, i) = 0.0

[Formula 2]
When i ≦ 3N / 16, W (2, i) = 0.0
When 3N / 16 <i ≦ 5N / 16, W (2, i) = 0.5−0.5 cos (8π (i−3N / 16) / N)
When 5N / 16 <i ≦ 5N / 8, W (2, i) = 1.0
When 5N / 8 <i ≦ 7N / 8, W (2, i) = 0.5 + 0.5 cos (4π (i−5N / 8) / N)
When i> 7N / 8, W (2, i) = 0.0

  As is clear from FIG. 4 and [Formula 1] and [Formula 2], the window functions W (1, i) and W (2, i) are asymmetrical to each other. This is for facilitating identification between the two on the extraction side described later.

  In the present invention, since odd frames and even frames are redundantly read by a predetermined number of samples, after embedding information and then restoring to an acoustic signal, the odd frame multiplied by the window function and the window function are multiplied. When overlapping samples of even frames are added, it is necessary to return almost to the original value. For this reason, when the window functions W (1, i) and W (2, i) are added in the overlapping portion of the odd frame and the even frame, it is defined to be a fixed value 1 for all sections.

  When the frequency conversion means 20 performs Fourier transform on the A type sound frame, the left channel signal Xl (i) and the right channel signal Xr (i) (i = 0,..., N−1). The window function W (1, i) is used to perform processing according to the following [Equation 3], and real part Al (1, j) and imaginary part Bl (1, j) of the conversion data corresponding to the left channel are performed. ), Real part Ar (1, j) and imaginary part Br (1, j) of the conversion data corresponding to the right channel are obtained. In the following description, the description using the left channel signal Xl (i) is common to the front left channel signal and the rear left channel signal, and the description using the right channel signal Xr (i) This is common to the right channel signal and the rear right channel signal.

[Formula 3]
Al (1, j) = Σi _{= 0,..., N-1} W (1, i) .Xl (i) .cos (2πij / N)
Bl (1, j) = Σi _{= 0,..., N-1} W (1, i) · Xl (i) · sin (2πij / N)
Ar (1, j) = Σi _{= 0,..., N-1} W (1, i) .Xr (i) .cos (2πij / N)
Br (1, j) = Σ _{i = 0,..., N-1} W (1, i) · Xr (i) · sin (2πij / N)

  When the frequency converting means 20 performs Fourier transform on the B type sound frame, the left channel signal Xl (i + N / 2), the right channel signal Xr (i + N / 2) (i = 0,..., N−1) ) Is processed using the window function W (2, i) according to the following [Equation 4], and the real part Al (2, j) and imaginary part Bl of the conversion data corresponding to the left channel are performed. (2, j), real part Ar (2, j) and imaginary part Br (2, j) of the conversion data corresponding to the right channel are obtained.

[Formula 4]
Al (2, j) = Σi _{= 0,..., N-1} W (2, i) · Xl (i + N / 2) · cos (2πij / N)
Bl (2, j) = Σi _{= 0,..., N-1} W (2, i) .Xl (i + N / 2) .sin (2πij / N)
Ar (2, j) = Σi _{= 0,..., N-1} W (2, i) .Xr (i + N / 2) .cos (2πij / N)
Br (2, j) = Σi _{= 0,..., N-1} W (2, i) .Xr (i + N / 2) .sin (2πij / N)

  In the above [Expression 3] and [Expression 4], i is a serial number assigned to N samples in each acoustic frame, and takes an integer value of i = 0, 1, 2,... N−1. Further, j is a serial number assigned in order from the smallest value of the frequency value, and takes an integer value of j = 0, 1, 2,... N / 2-1 like i. When the sampling frequency is 44.1 kHz and N = 4096, if the value of j is different by one, the frequency will be different by 10.8 Hz.

  By executing the processing according to the above [Equation 3] and [Equation 4], a spectrum corresponding to each window function of each acoustic frame is obtained. Subsequently, the frequency component changing unit 30 extracts a spectrum set in a predetermined frequency range from the generated spectrum. In the present embodiment, a range between F1 and F2 is extracted.

  After performing the frequency conversion, the frequency conversion means 20 performs conversion so that the average value of each spectrum set becomes a set value. A so-called amplitude conversion process is performed. The purpose of this amplitude conversion is to reduce the level difference for each frame so that appropriate information can be embedded. Therefore, the set value Zo can be set as appropriate.

  The frequency conversion means 20 first calculates the conversion magnification when performing the amplitude conversion on the obtained spectrum set. The conversion magnification is calculated by dividing the set value by an average effective value that is an average of effective intensity values of a spectrum set in a predetermined frequency range. Specifically, conversion magnifications Zl (1) and Zl (2) for spectrum sets of the L-channel A-type acoustic frame, the B-type acoustic frame, the R-channel A-type acoustic frame, and the B-type acoustic frame. , Zr (1), Zr (2) are calculated by processing according to the following [Equation 5]. When the target sound frame is close to silence and the square sum of the denominator is less than a predetermined value, the conversion magnification is set to 1.0 and the amplitude conversion is not performed. By performing the amplitude conversion, all the sound frames are in the state of amplitude conversion so that the average effective value of the spectrum set of each sound frame becomes the set value regardless of the signal intensity in the original state. Will be embedded.

[Formula 5]
Zl (1) = Zo / [Σ _{j = m,..., M−1} {Al (1, j) ² + Bl (1, j) ² }] ^1/2
Zl (2) = Zo / [Σ _{j = m,..., M−1} {Al (2, j) ² + Bl (2, j) ² }] ^1/2
Zr (1) = Zo / [Σ _{j = m,..., M−1} {Ar (1, j) ² + Br (1, j) ² }] ^1/2
Zr (2) = Zo / [Σ _{j = m,..., M−1} {Ar (2, j) ² + Br (2, j) ² }] ^1/2

  In the above [Equation 5], m and M are the lower limit and upper limit of the frequency band to be changed, and Zo = M−m. In the present embodiment, Zo = 288.

  Further, Zl (1) for each element of Al (1, j) and Bl (1, j) in the range of j = m,..., M−1 (corresponding to frequencies F1,..., F2). ), Multiply each element of Al (2, j) and Bl (2, j) by Zl (2), and multiply each element of Ar (1, j) and Br (1, j) The amplitude is converted by multiplying Zr (1) and multiplying each element of Ar (2, j) and Br (2, j) by Zr (2). In the following description, Al (1, j), Bl (1, j), Al (2, j), Bl (2, j), Ar (1, j), Br (1, j), Ar (2, j) and Br (2, j) are values obtained by performing these amplitude conversions.

  The frequency component changing unit 30 performs a process of changing the ratio of the predetermined frequency component for the A type sound frame according to the bit arrangement created by the bit arrangement creating unit 70. In the present invention, a bit array is read one bit or two bits at a time, and 1-bit or 2-bit information is embedded in a pair of acoustic frames of A type and B type. There are two 1-bit values to be embedded: “0” and “1”. In the present embodiment, these are defined as value 1 and value 2. These can be expressed as code 1 and code 2 in that two types of codes can be embedded. At this time, any one of “0” and “1” may be defined as a value 1 and a value 2 (reference numerals 1 and 2). This is because it is sufficient that one bit embedded on the extraction side can be specified on the extraction side. Therefore, this definition must match between the embedding side and the extraction side.

  Various methods of changing the frequency component are conceivable, but in this embodiment, the principle of sound pulse condensation, which is a human auditory psychological characteristic, is used. The syllable segregation is an illusion that humans interpolate and hear sounds as if two tracks of high and low are flowing continuously against a pattern in which high and low sounds alternate in time series. It is.

  Specifically, when the frequency component of a predetermined frequency range in an acoustic frame composed of a predetermined number of samples extracted from an acoustic signal is changed by the embedding device, the strength and weakness generate a sound pulse fraction. Change to As a result, although it does not sound as if the sound is interrupted by a human, the extraction device can recognize the clear change.

  In the present embodiment, using the principle of sound pulse segregation, the components of the change target frequency band of the acoustic frame are changed to two states, and 1-bit information is embedded. As an information embedding method using the principle of sound pulse segregation, the method shown in the above-mentioned Patent Document 2 can be adopted. However, in this embodiment, for the purpose of improving the extraction sensitivity and accuracy. In order to secure a large frequency region for embedding the identification information, the principle of sound pulse concentration double functions in two bands, high and low, in the frequency region for embedding the identification information. In addition, since the two high and low bands function the sound pulse separation function independently, both frequency widths do not necessarily have to be the same (however, each band is further divided into two in the frequency direction, but divided into two). The width of the area is the same). Here, the state of change of the predetermined frequency component of the acoustic frame before and after the embedding process will be described. FIG. 5 shows a state of predetermined frequency components of the A type and B type L channel 1 sound frames according to the present embodiment. The R channel is omitted because it is the same as the L channel. In each acoustic frame shown in FIG. 5, the horizontal axis indicates the time direction, and the vertical axis indicates the frequency direction.

  In FIG. 5, the frequency region is divided into six in the frequency direction of the vertical axis, but the change target frequency band is the second to fifth regions from the top, that is, between F1 and F2. The uppermost part, that is, the frequency exceeding F2, and the lowermost part, that is, less than F1, are frequency bands not to be changed. That is, in the present embodiment, the intensity of the spectrum set is changed by setting the frequency F1 to F2 as a predetermined frequency range. As shown in FIG. 5A, regarding the change target frequency band of the A type acoustic frame, the spectrum set is expressed by 1D1, 1D2, 1U1, and 1U2 in order of increasing frequency. Further, as shown in FIG. 5 (b), for the change target frequency band of the B-type acoustic frame, the spectrum set is represented by 2D1, 2D2, 2U1, and 2U2 in order of increasing frequency.

  When the code 1 is embedded by the 1-bit embedding method, as shown in FIGS. 5C and 5E, the product of the strengths of 1D1 and 2D2 and the product of the strengths of 1U1 and 2U2 are changed to a relatively strong state. The product of the strengths of 1D2 and 2D1 is changed to a relatively weak state. This state is referred to as “state 1”. When embedding the code 2, as shown in FIGS. 5D and 5F, the product of the intensity of 1D2 and 2D1 is changed to a relatively strong state, and the product of the intensity of 1U2 and 2U1 is changed to a relatively strong state. The product of the intensity of 1U1 and 2U2 is changed to a relatively weak state. This state is referred to as “state 2”. The shaded portions indicate that the ones having the same concentration are a spectrum set that is a set for obtaining a product. The darker shaded color indicates a group that is changed to a relatively strong state.

  In the 1-bit embedding method, information is embedded by changing the frequency components of the A-type and B-type sound frames in two states as shown in FIGS. 5C, 5E, 5D, and 5F. Is going. Since there are two states, this corresponds to an information amount of 1 bit. As shown in FIGS. 5C to 5F, in the case of the 1-bit embedding method, the upper frequency band and the lower frequency band of the change target frequency band have the same pattern. In the case of the 2-bit embedding method, by independently changing the upper frequency band and the lower frequency band, it is possible to represent 1 bit in the upper frequency band and 1 bit in the lower frequency band.

  In the present embodiment, the change target frequency bands F1 to F2 are set to “0.34 kHz to 3.4 kHz”. This is due to the following reasons. That is, when a mobile phone having a high degree of spread as voice communication is used as a receiving terminal, the upper limit needs to be 3.4 kHz which is the upper limit of the telephone line band and the mobile phone. This is because the low-pass filter mounted inside the mobile phone corresponds to 3.4 kHz or less in accordance with the telephone exchange. Also, the lower limit of the upper frequency band is set to 1.7 kHz, which is one octave lower than the upper limit of 3.4 kHz. The upper limit of the lower frequency band is set to 1.7 kHz, similar to the lower limit of the upper frequency band, and the lower limit of the lower frequency band needs to be 0.3 kHz or more, which is the lower limit of the telephone line band and the mobile phone. This is because the high-pass filter mounted inside the mobile phone corresponds to 0.3 kHz or more according to the telephone exchange. Therefore, the upper limit of 1.7 kHz is set to 0.34 kHz, which is slightly lower by 2 octaves. The frequency range of the lower frequency band is slightly narrower than the frequency range of the upper frequency band, but the intensity of the distributed signal components is about 4 times on average, so the sensitivity of the lower frequency band is greater than that of the upper frequency band. Can be estimated. The values “0.34 kHz”, “1.7 kHz”, and “3.4 kHz” are representative values, and are not necessarily accurate values, and may be slightly deviated from them.

In the example illustrated in FIG. 5, the change to the relatively strong state and the weak state has been described, but the degree of the strength can be set according to the situation. As will be described below, the greater the difference between the two, the higher the accuracy at the time of extraction. However, the ratio of interpolation is incomplete, and noise due to discontinuous components is heard during reproduction. On the other hand, as the ratio between the two is equal, the reproduction quality approaches the original sound, but the embedded bits cannot be extracted, and the reproduction quality and extraction accuracy are in a trade-off relationship. For example, when the strong side is set to 100% and the weak side is set to 0%, the sound of the part to be interpolated due to the pulse division is 50% of the sound that was played in the original acoustic signal before the change as shown in FIG. % Has been confirmed. Therefore, if the strong side is 70% and the weak side is 30%, the sound of the part to be interpolated is almost the same as the sound that was played in the original sound signal before making the change as shown in FIG. It has been confirmed that this ratio is the limit that can maintain the extraction accuracy. For this reason, it is preferable to set the ratio of the intensity of the relatively strong spectrum set and the relatively weak spectrum set to about 70% and 30%. However, in the present invention, it is necessary to change the strength depending on from which speaker the bit detected in the middle is emitted. Therefore, in this embodiment, in a specific process described later, coefficients α ₁ = 0.66, α ₂ = 0.72, α ₃ = 0.78, and a weak state are set for setting a strong state. The coefficients β ₁ = 0.34, β ₂ = 0.28, and β ₃ = 0.22. Note that α ₃ and β ₃ are necessary when the intensity is set at three levels in order to extract 25 types of identification information. In order to extract 9 types of identification information, the intensity is 2 This is not necessary when setting the stage. On the other hand, in order to extract more identification information, when the intensity is set to more stages, more α _k (k = 1, 2,..., (N + 1) / 2) Setting is required. Further, when the intensity of the spectrum set to be changed to a strong state is originally small, it is necessary to correct the coefficients α and β. For this reason, the frequency component changing means 30 first executes processing according to the following [Equation 6] to thereby obtain the intensities E _1D1 , E _2D1 , E _1D2 , E _2D2 , E _1U1 , E _{2U1 of} each spectrum set. , to calculate the E _1U2, E _2U2.

[Formula 6]
E _1D1 = Σ _{j = m,..., M + G−1} {Al (1, j) ² + Bl (1, j) ² }
E _2D1 = Σ _{j = m,..., M + G−1} {Al (2, j) ² + Bl (2, j) ² }
E _1D2 = Σ _{j = m + G,..., M + 2G−1} {Al (1, j) ² + Bl (1, j) ² }
E _2D2 = Σ _{j = m + G,..., M + 2G−1} {Al (2, j) ² + Bl (2, j) ² }
E _1U1 = Σ _{j = m + 2G,..., M + 2G + Gu-1} {Al (1, j) ² + Bl (1, j) ² }
E _2U1 = Σ _{j = m + 2G,..., M + 2G + Gu-1} {Al (2, j) ² + Bl (2, j) ² }
E _1U2 = Σ _{j = m + 2G + Gu,..., M + 2G + 2Gu−1} {Al (1, j) ² + Bl (1, j) ² }
E _2U2 = Σ _{j = m + 2G + Gu,..., M + 2G + 2Gu−1} {Al (2, j) ² + B1 (2, j) ² }

  In the above [Equation 6], m is the number of the lower limit component of the frequency band to be changed, G is the width of the two divided regions of the lower frequency band in the frequency band to be changed, and Gu is the upper frequency band in the frequency band to be changed. M + 2G + 2Gu is the number of the upper limit component of the frequency band to be changed. For example, when 0.34 kHz to 3.4 kHz is set as the change target frequency band, m = 32 and m + 2G + 2Gu = 320 (= M). Therefore, the width G (= (M / 2−m) / 2) = 64 of the lower frequency band, and the width Gu (= (M−M / 2) / 2) of the upper frequency band. = 80.

  Further, the frequency component changing unit 30 uses the calculated intensity of each spectrum set to calculate the intensity ratio γ of the spectrum set to be changed to a strong state with respect to the spectrum set to be changed to the weak state. Specifically, the intensity ratio γ is calculated according to the following [Equation 7].

[Formula 7]
If the embedded data is a value _{1, γ = (E 1D1 ·} E 1U1 · E 2D2 · E 2U2) / (E 1D2 · E 1U2 · E 2D1 · E 2U1)
If the embedded data is a value _{2, γ = (E 1D2 ·} E 1U2 · E 2D1 · E 2U1) / (E 1D1 · E 1U1 · E 2D2 · E 2U2)

Further, according to the value of the intensity ratio γ, the frequency component changing means 30 executes the processing according to the following [Equation 8], whereby the coefficients α ₁ , α ₂ , α ₃ , β ₁ , β ₂ , β ₃ is corrected to obtain coefficients α and β. In the following [Equation 8], α _k means any one of α ₁ , α ₂ , and α ₃ .

[Formula 8]
When 0.01 <γ <1.0, α = α _k · γ ^−1/4 , β = β _k · γ ^1/4
When γ ≦ 0.01, α = 10.0 · α _k , β = 0.1 · β _k
When γ ≧ 1.0, correction is not performed (α = α _k , β = β _k )

  Further, the frequency component changing means 30 is provided with real parts Al (1, j), Ar (1, j), Al (2, j), Ar (2, j) in the continuous A type acoustic frame and B type acoustic frame. , Bl (1, j), Br (1, j), Bl (2, j), Br (2, j) as frequency domain parameters, the lower limit m (= 32) to the upper limit M (= 320) , M to M / 2 are divided into two regions having a width G (= (M / 2−m) / 2), and m + 2G to M are divided into width Gu (= (M−M / 2) / It is divided into two areas having 2), and each is modified according to the bit value to be embedded. As an example, when the bit value to be embedded is “value 1”, the state of the frequency component is changed to “state 1”, that is, FIG. Change to the state shown in e).

[Formula 9]
For each component of j = m to m + G−1, E (1, j) = {Al (1, j) ² + Bl (1, j) ² + Ar (1, j) ² + Br (1, j) ² } ^1/2
Al ′ (1, j) = Al (1, j) · E (1, j) · α / {Al (1, j) ² + Bl (1, j) ² } ^1/2
Bl ′ (1, j) = Bl (1, j) · E (1, j) · α / {Al (1, j) ² + Bl (1, j) ² } ^1/2
Ar ′ (1, j) = Ar (1, j) · E (1, j) · α / {Ar (1, j) ² + Br (1, j) ² } ^1/2
Br ′ (1, j) = Br (1, j) · E (1, j) · α / {Ar (1, j) ² + Br (1, j) ² } ^1/2
E (2, j) = {Al (2, j) ² + Bl (2, j) ² + Ar (2, j) ² + Br (2, j) ² } ^1/2
Al ′ (2, j) = Al (2, j) · E (2, j) · β / {Al (2, j) ² + Bl (2, j) ² } ^1/2
Bl ′ (2, j) = Bl (2, j) · E (2, j) · β / {Al (2, j) ² + Bl (2, j) ² } ^1/2
Ar ′ (2, j) = Ar (2, j) · E (2, j) · β / {Ar (2, j) ² + Br (2, j) ² } ^1/2
Br ′ (2, j) = Br (2, j) · E (2, j) · β / {Ar (2, j) ² + Br (2, j) ² } ^1/2
For each component of j = m + G to m + 2G-1, E (1, j) = {Al (1, j) ² + Bl (1, j) ² + Ar (1, j) ² + Br (1, j) ² } ^1/2
Al ′ (1, j) = Al (1, j) · E (1, j) · β / {Al (1, j) ² + Bl (1, j) ² } ^1/2
Bl ′ (1, j) = B1 (1, j) · E (1, j) · β / {Al (1, j) ² + B1 (1, j) ² } ^1/2
Ar ′ (1, j) = Ar (1, j) · E (1, j) · β / {Ar (1, j) ² + Br (1, j) ² } ^1/2
Br ′ (1, j) = Br (1, j) · E (1, j) · β / {Ar (1, j) ² + Br (1, j) ² } ^1/2
E (2, j) = {Al (2, j) ² + Bl (2, j) ² + Ar (2, j) ² + Br (2, j) ² } ^1/2
Al ′ (2, j) = Al (2, j) · E (2, j) · α / {Al (2, j) ² + Bl (2, j) ² } ^1/2
Bl ′ (2, j) = Bl (2, j) · E (2, j) · α / {Al (2, j) ² + Bl (2, j) ² } ^1/2
Ar ′ (2, j) = Ar (2, j) · E (2, j) · α / {Ar (2, j) ² + Br (2, j) ² } ^1/2
Br ′ (2, j) = Br (2, j) · E (2, j) · α / {Ar (2, j) ² + Br (2, j) ² } ^1/2
For each component of j = m + 2G to m + 2G + Gu−1, E (1, j) = {Al (1, j) ² + Bl (1, j) ² + Ar (1, j) ² + Br (1, j) ² } ^1/2
Al ′ (1, j) = Al (1, j) · E (1, j) · α / {Al (1, j) ² + Bl (1, j) ² } ^1/2
Bl ′ (1, j) = Bl (1, j) · E (1, j) · α / {Al (1, j) ² + Bl (1, j) ² } ^1/2
Ar ′ (1, j) = Ar (1, j) · E (1, j) · α / {Ar (1, j) ² + Br (1, j) ² } ^1/2
Br ′ (1, j) = Br (1, j) · E (1, j) · α / {Ar (1, j) ² + Br (1, j) ² } ^1/2
E (2, j) = {Al (2, j) ² + Bl (2, j) ² + Ar (2, j) ² + Br (2, j) ² } ^1/2
Al ′ (2, j) = Al (2, j) · E (2, j) · β / {Al (2, j) ² + Bl (2, j) ² } ^1/2
Bl ′ (2, j) = Bl (2, j) · E (2, j) · β / {Al (2, j) ² + Bl (2, j) ² } ^1/2
Ar ′ (2, j) = Ar (2, j) · E (2, j) · β / {Ar (2, j) ² + Br (2, j) ² } ^1/2
Br ′ (2, j) = Br (2, j) · E (2, j) · β / {Ar (2, j) ² + Br (2, j) ² } ^1/2
For each component of j = m + 2G + Gu to M−1, E (1, j) = {Al (1, j) ² + Bl (1, j) ² + Ar (1, j) ² + Br (1, j) ² } ^1/2
Al ′ (1, j) = Al (1, j) · E (1, j) · β / {Al (1, j) ² + Bl (1, j) ² } ^1/2
Bl ′ (1, j) = B1 (1, j) · E (1, j) · β / {Al (1, j) ² + B1 (1, j) ² } ^1/2
Ar ′ (1, j) = Ar (1, j) · E (1, j) · β / {Ar (1, j) ² + Br (1, j) ² } ^1/2
Br ′ (1, j) = Br (1, j) · E (1, j) · β / {Ar (1, j) ² + Br (1, j) ² } ^1/2
E (2, j) = {Al (2, j) ² + Bl (2, j) ² + Ar (2, j) ² + Br (2, j) ² } ^1/2
Al ′ (2, j) = Al (2, j) · E (2, j) · α / {Al (2, j) ² + Bl (2, j) ² } ^1/2
Bl ′ (2, j) = Bl (2, j) · E (2, j) · α / {Al (2, j) ² + Bl (2, j) ² } ^1/2
Ar ′ (2, j) = Ar (2, j) · E (2, j) · α / {Ar (2, j) ² + Br (2, j) ² } ^1/2
Br ′ (2, j) = Br (2, j) · E (2, j) · α / {Ar (2, j) ² + Br (2, j) ² } ^1/2

  In this case, the intensity pattern of the spectrum set unit is the same on the high frequency side and the low frequency side. As described above, the frequency inverse transform means 40 performs the process of obtaining the modified sound frame by performing the frequency inverse transform on the frame spectrum in which the state of the frequency component is changed. Naturally, the inverse frequency conversion needs to correspond to the method executed by the frequency conversion means 20. In the present embodiment, since the frequency transform unit 20 performs the Fourier transform, the frequency inverse transform unit 40 executes the Fourier inverse transform.

  Specifically, for the A type sound frame, the frequency inverse transform means 40 uses the imaginary part Bl such as the real part Al ′ (1, j) of the left channel of the spectrum obtained by the above [Equation 9]. Using the real part Ar ′ (1, j) of the right channel, imaginary part Br ′ (1, j), etc., according to the following [Equation 10], X ′ '(I) and Xr' (i) are calculated. For frequency components that are not modified in the above [Equation 9], Al (1, j) or the like that is the original frequency component is used as Al ′ (1, j) or the like. In calculating the inverse frequency transform, for Al ′ (1, j) and Bl ′ (1, j), Zl (1) in [Formula 5] is replaced with Ar ′ (1, j) and Br ′ (1 , J), it is necessary to simultaneously perform inverse amplitude transformation by dividing Zr (1) in [Formula 5].

[Formula 10]
Xl' (i) = 1 / N · {Σ j Al' (1, j) · cos (2πij / N) / Zl (1) -Σ j Bl' (1, j) · sin (2πij / N) / Zl (1)} + Xlp (i + N / 2)
Xr' (i) = 1 / N · {Σ j Ar' (1, j) · cos (2πij / N) / Zr (1) -Σ j Br' (1, j) · sin (2πij / N) / Zr (1)} + Xrp (i + N / 2)

In the above [Expression 10], Σ _{j = 0,} ... _{, N−1} is shown as Σ _{j in} order to prevent the expression from becoming complicated. The terms “+ Xlp (i + N / 2)” in the first equation and “+ Xrp (i + N / 2)” in the second equation in the above [Equation 10] are the data Xlp (i) of the modified acoustic frame modified immediately before, When Xrp (i) exists, the addition is performed in consideration of the overlap of N / 2 samples on the time axis. By the above [Equation 10], each sample Xl ′ (i) of the left channel and each sample Xr ′ (i) of the right channel of the A type modified acoustic frame are obtained.

  For the B type acoustic frame, the frequency inverse transform means 40 performs real part Al ′ (2, j) and imaginary part Bl ′ (2, j) of the left channel of the spectrum obtained by the above [Equation 9]. , Using the real part Ar ′ (2, j) and imaginary part Br ′ (2, j) of the right channel, the processing according to the following [Equation 11] is performed, and Xl ′ (i), Xr ′ (i ) Is calculated. For frequency components that are not modified in the above [Equation 9], in the following [Equation 11], Al ′ (2, j), Bl ′ (2, j), Ar ′ (2, j), Br The original values Al (2, j), Bl (2, j), Ar (2, j), and Br (2, j) are used as ′ (2, j). In calculating the frequency inverse transform, for Al ′ (2, j) and Bl ′ (2, j), Zl (2) in [Equation 5] is replaced with Ar ′ (2, j) and Br ′ (2 , J), it is necessary to simultaneously perform inverse amplitude transformation by dividing Zr (2) in [Equation 5].

[Formula 11]
Xl' (i + N / 2) = 1 / N · {Σ j Al' (2, j) · cos (2πij / N) / Zl (2) -Σ j Bl' (2, j) · sin (2πij / N ) / Zl (2)} + Xlp (i + N)
Xr' (i + N / 2) = 1 / N · {Σ j Ar' (2, j) · cos (2πij / N) / Zr (2) -Σ j Br' (2, j) · sin (2πij / N ) / Zr (2)} + Xrp (i + N)

  By the above [Equation 11], each sample Xl ′ (i) of the left channel and each sample Xr ′ (i) of the right channel of the B type modified acoustic frame are obtained.

  The modified sound frame output unit 50 sequentially outputs the A type modified sound frame and the B type modified sound frame obtained by the processing of the frequency inverse transform unit 40 to an output file.

  Next, the overall flow of processing of the information embedding device for the acoustic signal shown in FIG. 3 will be described. First, a code conversion table in which an Nw-bit reference code and an Nh-bit Hamming code are associated with each other as shown in FIG. 6 is prepared in advance. This code conversion table may be created in any way as long as the Hamming distance between each Hamming code is 3 or more. In this embodiment, it is created by causing a computer to function as a conversion table creation device by executing a dedicated program. FIG. 7 shows a flowchart of code conversion table creation by the conversion table creation device.

  The conversion table creation device first performs initialization processing (S611). Specifically, the 8-bit reference code “0” is associated with the 16-bit code “1” and registered in the i (= 0) -th code conversion table, and the initial value of the 8-bit reference code KF is set to 1, 16 The initial value of the bit code HF is 2. Subsequently, i = 0 is initially set (S612). Next, the Hamming distance between the 16-bit code HF and its 15 cyclic codes and the other 16-bit code already registered in the code conversion table and its 15 cyclic codes is calculated (S613). ). At this stage, 16 ways × 16 ways and 256 ways of Hamming distances are calculated.

  If at least one of the calculated hamming distances is less than 3, the value of HF is updated by 1 (S614), and then the process returns to S612 and the process for the next 16-bit code HF is performed. On the other hand, if all the calculated Hamming distances are 3 or more, the value of i is incremented by 1 and updated (S615). If i is less than KF-1, the process returns to S613 to calculate the Hamming distance between the i-th registered 16-bit code in the code conversion table and its 15 cyclic codes. If i becomes KF-1 or more after S615, the 16-bit code HF is registered at the position of the 8-bit code KF in the code conversion table, and the values of KF and HF are respectively incremented by 1 and updated. (S616). If HF is less than 65536, the process returns to S612 to perform the process for the next 8-bit code KF. If the HF is 65536 or more, all 16-bit codes HF that can be expressed in a 16-bit range have been registered, and the code conversion table creation process ends.

  In this way, the code conversion table shown in FIG. 6 is created. The code conversion table of FIG. 6 is obtained by extracting only the corresponding part with the Hamming code that does not consider the cyclic code. In this code conversion table, an 8-bit reference code is defined so that 16 cyclic codes each have a Hamming distance of 3 or more within a range of 0 to 65535 in which a 16-bit Hamming code can be defined. Thus, as the reference code, a value of 0 to 131 slightly more than 7 bits can be expressed.

  When the code conversion table as shown in FIG. 6 is prepared, next, 25 types of identification information are determined. In the present embodiment, the identification information has a two-word configuration, and 25 types of identification information are determined by determining that one word in the first half is five types and one word in the second half is five types. The first half and the second half are performed by selecting one of the reference codes 0 to 131 shown in the code conversion table. As described with reference to FIG. 2B, due to the nature of the present invention, each bit of the second, third, and fourth codes is the same as at least one of the first and fifth codes. It must be devised to create it. As such a combination, there are two examples as shown in FIG.

  In the first code to the fifth code, the same bit value must be continuous between adjacent codes, or all must be the same bit value. If the 16-bit bit string satisfying this is limited to ASCII alphanumeric characters (all 62 characters), there are only two combinations shown in FIG. Therefore, the creator selects one of the combinations shown in FIG. 8 and creates the first code to the fifth code. In the example of FIG. 8, the reference code is shown with an English character converted to ASCII alphanumeric. One word is formed using any one of the combinations shown in FIG. 8 as the first code to the fifth code, and the second word selects a code completely different from the first word. For example, the first word Another word is formed by the bit inversion codes of the first code to the fifth code, and 25 types of identification information having a 2-word configuration are stored in the identification information storage unit 62. Since the two types of code sets in FIG. 8 are partially the same, they cannot be used for the first word and the second word.

  When the preparation of the identification information is completed, processing by the embedding device shown in FIG. 3 becomes possible. Next, the processing of the embedding device shown in FIG. 3 will be described according to the flowchart of FIG. Each component which comprises the apparatus shown in FIG. 3 performs the process according to FIG. 9 in cooperation. FIG. 9 corresponds to processing of one word of identification information of one channel. Therefore, in the case of four channels, almost the same processing is executed in parallel in each channel. In the present invention, the processing for reading the next one word is executed after repeatedly executing one word for a set number of times Na. For example, when Na = 3 is set, the same bit string for three words is repeatedly embedded.

  In FIG. 9, the bit array creation means 70 first creates a corresponding bit array by referring to the code conversion table for each word of the identification information extracted from the identification information storage unit 62 (S101). Specifically, first, from the identification information storage unit 62, the first code for the front left (FL) channel, the front right (FR) channel or the rear left (BL) channel, the fifth for the rear right (BR) channel. The code is extracted by 1 word (8 bits), and the second to fourth codes for intermediate reading are extracted by 1 word (8 bits). The code conversion table shown in FIG. A 16-bit bit array is created. In S101, different processing is performed for the first word and the second word of the identification information. In the case of the first word, it is as described above. However, in the case of the second word, from the identification information storage unit 62, the first code or the front right (front right (FL) channel, rear left (BL) channel ( The 5th code for the FR) channel and the rear right (BR) channel is extracted by 1 word (8 bits), and the 2nd to 4th codes for intermediate reading are extracted by 1 word (8 bits) and shown in FIG. With reference to the code conversion table, a 16-bit bit array composed of the corresponding Hamming code is inverted and created.

  Then, each 16 bits (48 bits per channel because of one word for each of the three codes) is read into a register in a computer used as an information embedding device for an acoustic signal. In this way, in the identification information storage unit 62, one word is 8 bits, but at the time of embedding processing, processing for one word in the identification information is performed with this 16-bit array.

  Next, the frequency component changing means 30 performs a process of reading 1 bit from Nh (= 16) bits of each identification information held in the register (S102). The processing in S102 is also different for the first word and the second word. For the first word, when processing is performed for the FL channel and the FR channel, each bit of the first to third codes is read, and when processing is performed for the BL channel and the BR channel, the third to third channels are read. One bit of each of the 5 codes is read. For the second word, when processing is performed for the FL channel and BL channel, each bit of the first to third codes is read, and when processing is performed for the FR channel and BR channel, the third to third channels are read. One bit of each of the 5 codes is read.

  Subsequently, the acoustic frame reading means 10 reads a predetermined number of samples from the left and right channels of the stereo acoustic signal stored in the acoustic signal storage unit 61 as one A-type acoustic frame, and the frequency converting means 20 Conversion is performed, and amplitude conversion is performed on the obtained frame spectrum (S103). Specifically, first, frequency conversion is performed on the read sound frame to obtain a frame spectrum that is a spectrum of the sound frame. That is, for each acoustic frame, processing according to the above [Equation 2] is performed using the window function W (1, i). Then, the process according to the above [Equation 5] is executed to calculate Zl (1) and Zr (1) and perform amplitude conversion. Similarly, the acoustic frame reading means 10 reads a predetermined number of samples as one B-type acoustic frame from the left and right channels of the stereo acoustic signal stored in the acoustic signal storage unit 61, and the frequency conversion means 20 Conversion is performed, and amplitude conversion is performed on the obtained frame spectrum (S104). Specifically, first, frequency conversion is performed on the read sound frame to obtain a frame spectrum that is a spectrum of the sound frame. That is, for each acoustic frame, processing according to the above [Equation 3] is performed using the window function W (2, i). Then, the process according to the above [Equation 5] is executed to calculate Zl (2) and Zr (2) and perform amplitude conversion.

Subsequently, when the frequency component changing unit 30 changes the state of the frequency component of the A type sound frame and the B type sound frame according to the read bit value, the processing according to [Formula 6] to [Formula 8] is performed. A process is executed to determine the conversion ratios α and β (S105). Specifically, the code of the read original channel (if the first word, the FL channel, the FR channel, the first code, the BL channel, BR channel, the fifth code, the second word If there is, the bit value of the first code in the case of the FL channel and the BL channel, the fifth code in the case of the FR channel and the BR channel) and the intermediate code (if the first word, the FL channel and the FR channel The second code for the BL channel and the BR channel, the fourth code for the second word, the second code for the FL channel and the BL channel, the fourth code for the FR channel and the BR channel. Whether the bit value of the central code (third code) is the same. If the bit value of the original channel code, the bit value of the intermediate code, and the bit value of the central code are all the same, α ₃ is used as α _k , and the bit value of the intermediate code and the bit of the central code are used. When the values are different, α ₂ is used as α _k , and when the bit value of the original channel code is different from the bit value of the intermediate code, α ₁ is used as α _k , and the processing according to the above [Equation 8] To determine α and β. Using the determined conversion ratios α and β, the frequency component changing unit 30 executes the process according to the above [Equation 9] according to the value 1 and the value 2 received from the bit array creating unit 70, and changes them. The state of the component of the target frequency band is changed to either “state 1” or “state 2” (S106).

  Next, the frequency inverse transforming unit 40 performs a process of obtaining a modified acoustic frame by performing an amplitude inverse transform and a frequency inverse transform on the spectrum in which the intensity of each spectrum set corresponding to the A type acoustic frame is changed by the process of S106. Perform (S107). This inverse amplitude transform is performed by multiplying the spectrum by the inverse of Zl (1) and Zr (1) calculated by [Equation 5]. This inverse frequency transform is naturally performed by the frequency transforming means 20 in S103. It is necessary to support this method. In the present embodiment, since the frequency transform unit 20 performs the inverse Fourier transform, the frequency inverse transform unit 40 performs the inverse Fourier transform. Specifically, the real part Ar ′ (1) of the right channel, such as the real part Al ′ (1, j) of the left channel, the imaginary part Bl ′ (1, j), etc. of the spectrum obtained by the above [Equation 9]. , J) and the like, the imaginary part Br ′ (1, j) and the like are used to perform processing according to the above [Equation 10] to calculate Xl ′ (i) and Xr ′ (i). The modified sound frame output means 50 sequentially outputs the obtained modified sound frames to an output file.

  Similarly, the frequency inverse transform unit 40 performs a process of obtaining a modified acoustic frame by performing an amplitude inverse transform and a frequency inverse transform on the spectrum in which the intensity of each spectrum set corresponding to the B type acoustic frame is changed by the process of S106. Perform (S108). Specifically, the inverse amplitude transformation is performed by multiplying the spectrum by the reciprocal of Zl (2) and Zr (2) calculated by [Equation 5], and the actual left channel of the spectrum obtained by [Equation 9] is obtained. Using the part Al ′ (2, j) etc., the imaginary part Bl ′ (2, j) etc., the real part Ar ′ (2, j) of the right channel, the imaginary part Br ′ (2, j) etc. A process according to [Formula 11] is performed to calculate Xl ′ (i) and Xr ′ (i).

  The modified sound frame output means 50 sequentially outputs the obtained modified sound frames to an output file. When the processing for the two A-type and B-type sound frames is finished for each channel in this way, the frequency component changing means 30 reads the next 1 bit in the bit array (S102). The above processing is executed over all samples of both channels of the acoustic signal. That is, a predetermined number of samples are read as sound frames, and when there are no more sound frames to be read from the sound signal (S103, S104), the process ends. Note that when the processing corresponding to each bit of the bit arrangement (Nh = 16 bits) for one word read in S101 is completed, the processing returns from S102 to S101, and the processing for reading the next word of the identification information is performed. . When the process is completed for all the words of the identification information, the process returns to the first word of the identification information. As a result, all modified acoustic frames that have been processed for all acoustic frames are recorded in the output file and obtained as modified acoustic signals. The obtained modified acoustic signal is output to and stored in the modified acoustic signal storage unit 63 in the storage unit 60.

  In the present embodiment, a case has been described in which 8 bits of 1 word that is the original identification information are converted into a 16-bit bit array and processing for 1 word of the identification information is performed. As long as there is an agreement, it is possible to record one word of identification information in units of other number of bits.

  Of the modified acoustic signal obtained as described above, for the portion in which the identification information is embedded, the component of the frequency band to be changed has only two distributions of state 1 and state 2. However, since the components other than the component of the frequency band to be changed remain as the original acoustic signals, various distributions are made based on the setting of the producer.

  The embedding apparatus has been described above. Here, embedding and extraction using amplitude conversion performed in this embodiment will be described with reference to FIGS. 10 to 13. 10 and 11 show the embedding process and extraction process when amplitude conversion is not used for comparison, and FIGS. 12 and 13 show the embedding process and extraction process using amplitude conversion in the present embodiment. In these explanatory diagrams, the signal waveform of the acoustic signal assuming that 3-bit information is embedded is shown for the odd and even three pairs of acoustic frames, and the acoustic frames are shown in a state where they do not overlap each other for convenience of explanation. ing. In each figure, the waveform at the left end shows a normal acoustic signal waveform in the time dimension, the horizontal direction is the time axis, the time advances toward the right, and the vertical direction is intensity (amplitude). The waveform at the center or right end represents the sum of spectral components in a predetermined frequency range after frequency conversion as a signal waveform, and is a signal waveform after virtually passing through a bandpass filter. The vertical direction is similarly strong, and is originally significantly lower than the left end, but here, for convenience of explanation, it is enlarged and expressed to the same extent as the left end. Although the levels of the two types of waveforms are originally different, they are represented at the same level here. Actually, there are high frequency components and low frequency components outside the predetermined frequency range as the data after frequency conversion, but these are not shown on the assumption that they are not modified, and are embedded according to FIG. The two signal waveforms are limited to the two frequency band components for performing the above.

  FIG. 10 (a) shows the original sound signal. The first pair of sound frames has a substantially constant intensity throughout, and the center pair of sound frames has a smaller odd frame intensity and an even frame intensity. The last pair of sound frames has a large odd-numbered frame and a small even-numbered frame. FIG. 10 (b) shows the result of frequency conversion performed on such an acoustic signal, and the component to be embedded is divided into two parts, as shown in FIG. ) Are expressed separately as two waveforms (b-1) and (b-2). The waveform shape after frequency conversion differs for each frequency and is not necessarily similar to that shown in FIG. 10A, but here it is assumed that it is the same as before frequency conversion. 10B-1 and 10B-2, by embedding 3-bit data [0, 0, 0] based on the method described in FIG. 10 (c-1) and (c-2) are obtained. The first pair can express a pattern as shown in FIG. 5, but the other two pairs of upper and lower ones are required to construct a magnitude relationship in the opposite direction with respect to the original stage. It turns out that an appropriate pattern has not been constructed. FIG. 10 (d) shows the result of frequency inverse transformation performed on the result of such modification. In general, the external shape (referred to as an envelope) of an acoustic signal waveform is determined by low-frequency components having a large energy distribution, which are included in signal components outside a predetermined frequency range in this application, and are modified in the stage of FIG. 10 (c). Therefore, FIG. 10D after frequency inverse transformation has a shape similar to FIG. 10A before frequency transformation.

  FIG. 11 shows how extraction processing is performed on the result of such embedding. 11 (a) and 11 (b) correspond to FIGS. 10 (d) and 10 (c), respectively. In FIG. 11B, when performing bit determination, since the pattern assumed in FIG. 5 is not formed except for the first pair, it is difficult to determine a correct bit.

  Next, a method in which amplitude conversion is introduced in the present embodiment will be described with reference to FIGS. FIG. 12 (a) shows the same original acoustic signal as FIG. 10 (a), and FIG. 12 (b) obtained by frequency conversion in the same manner is the same as FIG. 10 (b). Here, FIG. 12C shows the result of amplitude conversion performed in units of six frames. In the case of FIG. 12, since the amplitude in each frame is flat in FIG. 12 (a), the whole is flat in FIG. 12 (c). Since the intra-frame variation is followed even in the stage of FIG. 12C, it does not normally become completely flat as shown in FIG. (Actually, since FIG. 12 (c) is frequency dimension data, the frequency dimension data is not flat and fluctuates.) Also, the conversion magnification is set for each frame, and the upper and lower two frequency components are set. Since the conversion is performed at the same magnification, there is usually a significant difference between the upper and lower sides (however, the time axis direction is relatively uniform). On the other hand, when 3-bit data [0, 0, 0] is embedded, FIGS. 12D-1 and 12D-2 are obtained. Since FIG. 12C, which is the original stage, has a flat waveform, it can be seen that an ideal pattern as shown in FIG. 5 can be easily constructed in all frames. Next, FIG. 12E shows the result of inverse amplitude transformation performed by multiplying the inverse of the magnification set for each frame. At this stage, a pattern similar to that shown in FIG. 10C is often generated, but any shape can be used. Finally, when frequency inverse transform is performed, FIG. 12F is obtained, and similarly, the shape is similar to that of FIG. 12A of the original signal waveform.

  FIG. 13 shows how extraction processing is performed on the result of such embedding. FIGS. 13 (a), (b) and (c) correspond to FIGS. 12 (f), (e) and (d), respectively. Although the waveform shape after frequency conversion in FIG. 13B is basically different from that in FIG. 12B, the calculated amplitude conversion magnification is a similar value, and the amplitude conversion is performed at substantially the same magnification. 13 (c) is obtained. If bit determination is performed at the stage of FIG. 13C, an ideal pattern as shown in FIG. 5 is formed in all frames, so that correct bits can be determined.

  As described above, in the present embodiment in which the amplitude conversion is introduced, since the signal component to be embedded is converted and embedded so as to be flat in the time axis direction as illustrated in FIG. The probability of making an unnatural change that completely reverses the magnitude relationship between the component intensity on the low frequency side and the low frequency side is reduced, and the extraction accuracy on the extraction side can be increased while maintaining the quality.

(3.1. Configuration of position detection device)
Next, a position detection apparatus using an acoustic signal according to the present invention will be described. FIG. 14 is a block diagram showing an embodiment of a position detecting device using an acoustic signal according to the present invention. In FIG. 14, 100 is an acoustic signal input means, 110 is a reference frame acquisition means, 120 is a phase change frame setting means, 130 is a frequency conversion means, 140 is a code determination parameter calculation means, 150 is a code output means, and 160 is identification information. Extraction means, 170 an acoustic frame holding means, 180 a cyclic code table creation means, 190 a positional relationship storage means, and 200 a position output means.

  The acoustic signal input unit 100 has a function of acquiring and inputting a flowing sound as a digital acoustic signal. In reality, it is realized by a microphone and an A / D converter. Any microphone can be used, regardless of whether it is monaural omnidirectional or stereo directional, as long as the component in the frequency band to be changed can be detected. Even if it is stereo-directional, only one channel needs to be used. In addition, when information is embedded with the apparatus shown in FIG. 3, the sound signal is reproduced in stereo, but when identifying information embedded in the FL channel is extracted, a microphone is installed near the FL speaker. When extracting identification information embedded in the FR channel, a microphone is installed near the FR speaker, and when extracting identification information embedded in the BL channel, a microphone is installed near the BL speaker, and the BR channel is extracted. When extracting identification information embedded in the speaker, a microphone is installed in the vicinity of the BR speaker, and when extracting identification information using a plurality of information embedded in each channel, the FL speaker, the FR speaker, and the BL speaker In the range surrounded by BR speakers, the microphones are separated by approximately four equal intervals between adjacent speakers. To install the phone. This microphone is not highly accurate, and information can be extracted using a microphone with general accuracy. The reference frame acquisition unit 110 has a function of reading an audio frame composed of a predetermined number of samples as a reference frame from the input digital monaural audio signal (or one channel of a stereo audio signal). As the reference frame, A type and B type are set as in the case of embedding. The phase change frame setting means 120 has a function of setting, as a phase change frame, an acoustic frame whose phase has been changed by moving each of the A type and B type reference frames and predetermined samples.

  The frequency conversion unit 130 basically has the same function as the frequency conversion unit 20 shown in FIG. However, since the timing for capturing the sound does not start from the beginning of the acoustic signal, in order to identify the correct phase, amplitude conversion and frequency conversion are performed with a plurality of phases, even if the original acoustic signal is four channels. The difference is that one channel is used.

  When performing the Fourier transform on the A type acoustic frame, the frequency converting unit 130 applies the window function W to the signal X (i−N / 2 + pN / 6) (i = 0,..., N−1). (1, i) is used to perform processing according to the following [Equation 12] to obtain the real part A (1, j, p) and imaginary part B (1, j, p) of the converted data. Have. p is a phase number and takes an integer value of 0 to 5.

[Formula 12]
A (1, j, p) = Σi _{= 0,..., N-1} W (1, i) .X (i-N / 2 + p.N / 6) .cos (2πij / N)
B (1, j, p) = Σi _{= 0,..., N-1} W (1, i) .X (i-N / 2 + p.N / 6) .sin (2πij / N)

  When the Fourier transform is performed on the B type acoustic frame, the frequency converting unit 130 applies the window function W (2) to the signal X (i + p · N / 6) (i = 0,..., N−1). , I) is used to perform processing according to the following [Equation 13] to obtain the real part A (2, j, p) and imaginary part B (2, j, p) of the converted data. ing.

[Formula 13]
A (2, j, p) = Σ _{i = 0,..., N-1} W (2, i) · X (i + p · N / 6) · cos (2πij / N)
B (2, j, p) = Σi _{= 0,..., N-1} W (2, i) .X (i + p.N / 6) .sin (2πij / N)

The frequency conversion unit 130 performs amplitude conversion in the same manner as the frequency conversion unit 20. In performing amplitude conversion, first, conversion magnification is calculated. The conversion magnification is calculated by dividing the set value by the average effective value that is the average of the effective intensity values of the spectrum set in the predetermined frequency range. Although the set value can be determined as appropriate, it is necessary to set the same value as in the case of amplitude conversion at the time of embedding. Therefore, in the present embodiment, the set value Zo needs to be 288 (= M−m). Specifically, the conversion magnifications Z (1, p) and Z (2, p) for the A type acoustic frame and the B type acoustic frame, the A type acoustic frame immediately before these, and the B type acoustic frame. The conversion magnifications Z ₋₁ (1, p) and Z ₋₁ (2, p) are calculated by processing according to the following [Equation 14]. By performing the amplitude conversion, all the sound frames are extracted in a state where the amplitude is converted so that the average effective value of each sound frame becomes a set value regardless of the signal strength in the original state. Will be done. In addition, p is a phase number and takes an integer value of 0-5.

[Formula 14]
Z (1, p) = Zo / [Σ _{j = m,..., M−1} {A (1, j, p) ² + B (1, j, p) ² }] ^1/2
Z (2, p) = Zo / [Σ _{j = m,..., M−1} {A (2, j, p) ² + B (2, j, p) ² }] ^1/2
Z ₋₁ (1, p) = Zo / [Σ _{j = m,..., M−1} {A ₋₁ (1, j, p) ² + B ₋₁ (1, j, p) ² }] ^1/2
Z ₋₁ (2, p) = Zo / [Σ _{j = m,..., M−1} {A ₋₁ (2, j, p) ² + B ₋₁ (2, j, p) ² }] ^1/2

j = m,..., M-1 (corresponding to frequencies F1,..., F2) and p = 0,. . . , 5, each element of A (1, j, p) and B (1, j, p) is multiplied by Z (1, p) and A (2, j, p) and B ( 2, j, p) is multiplied by Z (2, p), and for each element of A _-1 (1, j, p) and B _-1 (1, j, p) Multiply by Z _-1 (1, p) and multiply each element of A _-1 (2, j, p) and B _-1 (2, j, p) by Z _-1 (2, p) Thus, amplitude conversion is performed. In the following description, A (1, j, p), B (1, j, p), A (2, j, p), B (2, j, p), A ₋₁ (1, j, p) , B ₋₁ (1, j, p), A ₋₁ (2, j, p), and B ₋₁ (2, j, p) are values obtained by performing these amplitude conversions.

  The code determination parameter calculation unit 140 extracts a spectrum set corresponding to a predetermined frequency range from the generated spectrum, calculates an intensity value of each spectrum set, and calculates a code determination parameter using the intensity value. Based on the magnitude relationship of the code determination parameters, a function for determining a predetermined state is provided. As described above, in the present embodiment, since the A type acoustic frame and the B type acoustic frame are set to overlap each other by N / 2 samples, the intensity value and the code determination parameter are calculated for a certain acoustic frame. In this case, it is necessary to consider the reverberation component due to the immediately preceding acoustic frame. However, since the reverberation component is calculated, the accurate one is not always calculated. If the calculated reverberation component is removed, the extraction accuracy may be lowered due to warping. Therefore, in this embodiment, the intensity values E1, E2, E3, E4 and the sign determination parameter C when the reverberation component is not removed, and the intensity values E1 ′, E2 ′, E3 ′, E4 ′ when removed, and A correction code determination parameter C ′ is calculated, and a state that seems to be optimal is determined using these parameters.

Each component corresponding to the B type acoustic frame one frame before the respective components A (1, j, p) and B (1, j, p) of the A type acoustic frame is represented by A ₋₁ (2, j, p), B ₋₁ (2, j, p), one frame corresponding to each component A (2, j, p), B (2, j, p) of the B type acoustic frame The sound frame before the minute is the A-type sound frame before correction. The code determination parameter calculation unit 140 first calculates the basic intensity values E (1, j, p), E (2, j, p), and reverberation components of each spectrum set before calculating the intensity values of each spectrum set. The basic intensity values E ′ (1, j, p) and E ′ (2, j, p) from which the above is removed are calculated by processing according to the following [Equation 15].

[Formula 15]
E (1, j, p) = A (1, j, p) ² + B (1, j, p) ²
E (2, j, p) = A (2, j, p) ² + B (2, j, p) ²
E ₋₁ (1, j, p) = A ₋₁ (1, j, p) ² + B ₋₁ (1, j, p) ²
E ₋₁ (2, j, p) = A ₋₁ (2, j, p) ² + B ₋₁ (2, j, p) ²
E ′ (1, j, p) = E (1, j, p) −q · E ₋₁ (2, j, p)
E ′ (2, j, p) = E (2, j, p) −q · E (1, j, p)

  Of the six formulas in [Formula 15], the fifth formula from the top (second from the bottom) is the B type in which the N / 2 samples in the first half overlap when focusing on a certain type A acoustic frame. This is for removing the reverberation component from the acoustic frame. In addition, the sixth expression from the top (first from the bottom) removes the reverberation component from the A-type sound frame in which the N / 2 samples in the first half overlap when focusing on a certain B-type sound frame. Is for. E ′ (1, j, p) ≧ 0, E ′ (2, j, p) ≧ 0, and as a result of processing according to the fifth and sixth equations from the top of [Equation 15], If this happens, set it to 0.

  In the above [Expression 15], q is a coefficient indicating the magnitude of the reverberation component, but this coefficient q has a value less than 1, and as a result of experiment, when N = 4096, q = 0.06, When N = 2048, q = 0.12. When N = 1024, q = 0.24. When N = 512, q = 0.48 is optimal.

  Then, the intensity values E1, E2, E3, E4, E5, E6, E7, E8 of each spectrum set when the reverberation component is not removed, and the intensity values E1 ′, E2 ′, E3 ′, E4 ′, E5 when removed. ′, E6 ′, E7 ′, E8 ′ are calculated from the calculated basic intensity values E (1, j, p), E (2, j, p), E ′ (1, j, p), E ′ (2, j, p) is used to calculate based on the following [Equation 16].

[Formula 16]
E1 (p) = Σ _{j = m,..., M + G-1} E (1, j, p)
E2 (p) = Σ _{j = m, ..., m + G-1} E (2, j, p)
E3 (p) = Σ _{j = m + G,..., M + 2G-1} E (1, j, p)
E4 (p) = Σ _{j = m + G,..., M + 2G-1} E (2, j, p)
E5 (p) = Σ _{j = m + 2G,..., M + 2G + Gu-1} E (1, j, p)
E6 (p) = Σ _{j = m + 2G,..., M + 2G + Gu-1} E (2, j, p)
E7 (p) = Σ _{j = m + 2G + Gu, ..., m + 2G + 2Gu-1} E (1, j, p)
E8 (p) = Σ _{j = m + 2G + Gu, ..., m + 2G + 2Gu-1} E (2, j, p)
E1 ′ (p) = Σ _{j = m,..., M + G-1} E ′ (1, j, p)
E2 ′ (p) = Σ _{j = m,..., M + G-1} E ′ (2, j, p)
E3 ′ (p) = Σ _{j = m + G,..., M + 2G-1} E ′ (1, j, p)
E4 ′ (p) = Σ _{j = m + G,..., M + 2G-1} E ′ (2, j, p)
E5 ′ (p) = Σ _{j = m + 2G,...,} M _{+ 2G + Gu−1} E ′ (1, j, p)
E6 ′ (p) = Σ _{j = m + 2G,...,} M _{+ 2G + Gu−1} E ′ (2, j, p)
E7 ′ (p) = Σ _{j = m + 2G + Gu,..., M + 2G + 2Gu−1 E ′} (1, j, p)
E8 ′ (p) = Σ _{j = m + 2G + Gu,..., M + 2G + 2Gu−1 E ′} (2, j, p)

  Eventually, the intensity values E1, E2, E3, E4, E5, E6, E7, and E8 of each spectrum set are calculated by [Equation 15] and [Equation 16], and the type corresponding to each spectrum set is determined. Intensity values E1 ′, E2 ′, E3 ′, E4 ′, E5 ′, E6 ′, E7 ′, E8 ′ are calculated by subtracting the value obtained by multiplying the intensity of the spectrum set in the immediately preceding acoustic frame by q. become.

  Also, the code determination parameter calculation unit 140 calculates the code determination parameter C using the intensity values E1, E2, E3, E4, E5, E6, E7, and E8 calculated without removing the reverberation component. In the case of a method of embedding 1 bit in one acoustic frame, a candidate code B is provisionally determined and a code determination parameter C is calculated by executing processing according to the following [Equation 17].

[Formula 17]
1) In the case of E1 (p) · E5 (p)> E2 (p) · E6 (p) and E4 (p) · E8 (p)> E3 (p) · E7 (p)
C = E1 (p) .E5 (p). {E4 (p) .E8 (p) -E2 (p) .E6 (p) .E3 (p) .E7 (p)} / {E1 (p). E5 (p), E4 (p), E8 (p) + E2 (p), E6 (p), E3 (p), E7 (p)}
2) When E2 (p) · E6 (p)> E1 (p) · E5 (p) and E3 (p) · E7 (p)> E4 (p) · E8 (p)
C = {E2 (p) .E6 (p) .E3 (p) .E7 (p) -E1 (p) .E5 (p) .E4 (p) .E8 (p)} / {E1 (p). E5 (p), E4 (p), E8 (p) + E2 (p), E6 (p), E3 (p), E7 (p)}
3) In the case of E1 (p), E5 (p), E4 (p), E8 (p)> E2 (p), E6 (p), E3 (p), E7 (p)
C = {E1 (p) .E5 (p) .E4 (p) .E8 (p) -E2 (p) .E6 (p) .E3 (p) .E7 (p)} / {E1 (p). E5 (p), E4 (p), E8 (p) + E2 (p), E6 (p), E3 (p), E7 (p)}
4) In cases other than the above 1) to 3) B = 1 and provisional determination,
C = {E2 (p) .E6 (p) .E3 (p) .E7 (p) -E1 (p) .E5 (p) .E4 (p) .E8 (p)} / {E1 (p). E5 (p), E4 (p), E8 (p) + E2 (p), E6 (p), E3 (p), E7 (p)}

  In addition, the code determination parameter calculation unit 140 performs correction code determination using the intensity values E1 ′, E2 ′, E3 ′, E4 ′, E5 ′, E6 ′, E7 ′, and E8 ′ calculated by removing the reverberation component. The parameter C ′ is calculated. In the case of a method of embedding 1 bit in one acoustic frame, in the above [Equation 17], E1 (p) to E8 (p) are replaced with E1 ′ (p) to E8 ′ (p), respectively, thereby replacing the candidate code B. The candidate code B ′ is tentatively determined, and the correction code determination parameter C ′ is calculated instead of the code determination parameter C.

  On the other hand, in the case of a method of embedding 2 bits in one acoustic frame, the code determination parameter calculation unit 140 assumes that the candidate codes are B1 and B2 and the code determination parameters are C1 and C2, respectively. 18] is executed, the candidate code B1 is provisionally determined and the code determination parameter C1 is calculated.

[Formula 18]
1) When E1 (p)> E2 (p) and E4 (p)> E3 (p) B1 = 0 and provisional determination
C1 = {E1 (p) · E4 (p) −E2 (p) · E3 (p)} / {E1 (p) · E4 (p) + E2 (p) · E3 (p)}
2) When E2 (p)> E1 (p) and E3 (p)> E4 (p)
C1 = {E2 (p) · E3 (p) −E1 (p) · E4 (p)} / {E1 (p) · E4 (p) + E2 (p) · E3 (p)}
3) In the case of E1 (p) · E4 (p)> E2 (p) · E3 (p) B1 = 0 is temporarily determined,
C1 = {E1 (p) · E4 (p) −E2 (p) · E3 (p)} / {E1 (p) · E4 (p) + E2 (p) · E3 (p)}
4) In cases other than the above 1) to 3) B1 = 1 and provisional determination,
C1 = {E2 (p) · E3 (p) −E1 (p) · E4 (p)} / {E1 (p) · E4 (p) + E2 (p) · E3 (p)}

  Further, by executing the processing according to the following [Equation 19], the candidate code B2 is provisionally determined and the code determination parameter C2 is calculated.

[Formula 19]
1) When E5 (p)> E6 (p) and E8 (p)> E7 (p) B2 = 0 and provisional determination
C2 = {E5 (p) .E8 (p) -E6 (p) .E7 (p)} / {E5 (p) .E8 (p) + E6 (p) .E7 (p)}
2) When E6 (p)> E5 (p) and E7 (p)> E8 (p)
C2 = {E6 (p) .E7 (p) -E5 (p) .E8 (p)} / {E5 (p) .E8 (p) + E6 (p) .E7 (p)}
3) In the case of E5 (p) · E8 (p)> E6 (p) · E7 (p) B2 = 0 is temporarily determined,
C2 = {E5 (p) .E8 (p) -E6 (p) .E7 (p)} / {E5 (p) .E8 (p) + E6 (p) .E7 (p)}
4) In cases other than the above 1) to 3) B2 = 1 and provisional determination,
C2 = {E6 (p) .E7 (p) -E5 (p) .E8 (p)} / {E5 (p) .E8 (p) + E6 (p) .E7 (p)}

  In addition, the code determination parameter calculation unit 140 performs correction code determination using the intensity values E1 ′, E2 ′, E3 ′, E4 ′, E5 ′, E6 ′, E7 ′, and E8 ′ calculated by removing the reverberation component. Parameters C1 ′ and C2 ′ are calculated. In the case of the method of embedding 2 bits in one acoustic frame, in the above [Equation 18] and [Equation 19], E1 (p) to E8 (p) are replaced with E1 ′ (p) to E8 ′ (p), respectively. The candidate codes B1 ′ and B2 ′ are provisionally determined instead of the codes B1 and B2, and correction code determination parameters C1 ′ and C2 ′ are calculated instead of the code determination parameters C1 and C2.

  The code output means 150 determines what is determined to be the optimum phase from the acoustic frames (reference frame and phase change frame) corresponding to one reference frame, and selects a code corresponding to the state of the acoustic frame. It has a function to output. The code determination parameter calculation unit 140 and the code output unit 150 constitute an encoding unit. The identification information extraction unit 160 has a function of extracting the binary array output by the code output unit 150 in units of Nh bits, and converting it into an Nw-bit reference code by referring to the cyclic code table. . The acoustic frame holding means 170 is a buffer memory capable of holding two consecutive reference frames for each of the A type and B type. Similar to the conversion table creation device, the cyclic code table creation means 180 has Nh (> Nw) bits whose hamming distance is at least 3 or more with respect to 2 Nw power reference codes that can be taken by Nw bits. By assigning a Hamming code, it has a function of creating a code conversion table and a cyclic code table in which an Nw-bit reference code and an Nh-bit Hamming code are associated with each other. The positional relationship storage unit 190 stores a positional relationship table in which a relative position with reference to the left and right speakers is associated with a reference code. The position output means 200 refers to the positional relationship table in the positional relationship storage means 190 using the reference code obtained by the identification information extraction means 160, and acquires and outputs position information. For specific output, a display device such as a liquid crystal display is used.

  Each component shown in FIG. 14 is actually realized by mounting a dedicated program on hardware such as a small computer having an information processing function and its peripheral devices. In particular, in order to achieve the object of the present invention more easily, it is desirable to use as a hardware a portable terminal device having an arithmetic processing function (function as a computer) such as a mobile phone or a PDA (Personal Digital Assistant). . A portable terminal device having a function as a position detection device is carried and moved between the left and right speakers, so that it is surrounded by four speakers based on the sound acquired by the microphone included in the portable terminal device. The relative position within the range can be detected.

(3.2. Processing operation of position detection device)
Next, the processing operation of the position detection apparatus using the acoustic signal shown in FIG. 14 will be described according to the flowchart of FIG. When the position detection device is activated, first, in this device, the phase determination table S (p), the phase determination log, the phase determination flag, the total value accumulation table Te (1, p) to Te (8, p), and the bit counter are set. Initialization is performed (S200). The phase determination table S (p) is a table for determining the phase, and p takes an integer value of 0 to 5. The initial value is set to S (p) = 0. The phase determination log records the determined phase, that is, the phase number p for each set of one reference frame and five phase change frames, and 0 is set in the initial state. The phase determination flag is a flag indicating whether or not the phase is fixed, and is set to Off in the initial state. The total value accumulation tables Te (1, p) to Te (8, p) include six types of total values E1 to E4 for the past (byte period × Nh frame) and total values E1 ′ to E4 ′ after reverberation correction. This is stored for each phase, and is set to Te (1, p) to Te (8, p) = 0 in the initial state. For the bit counter, 0 is set as an initial value.

  Subsequently, the cyclic code table creating unit 180 creates a cyclic code table (S210). The cyclic code table is a code conversion table shown in FIG. 6 in which a cyclic bit pattern in which a bit string of a 16-bit Huffman code after conversion corresponding to a conversion source 7-bit reference code is moved bit by bit is recorded. is there. The cyclic code table creating means 180 first creates a cyclic code table after creating the code conversion table as shown in FIG. 6 by the processing according to the flowchart of FIG. Specifically, the cyclic code table creating unit 180 uses a cyclic bit pattern obtained by moving a bit string of a 16-bit Hamming code associated with an 8-bit reference code bit by bit in the code conversion table as shown in FIG. Created and recorded in association with 8-bit reference code. For example, since the 16-bit Hamming code “0000, 0000, 0000, 0001” is associated with the 8-bit reference code “0” in FIG. 6 as a normal bit pattern, the cyclic code table creating unit 180 has 1 Bit-shifted 16-bit code “0000, 0000, 0000,0010” (decimal notation “2”), 2-bit shifted 16-bit code “0000, 0000, 0000, 0100” (decimal notation “4”),... 15 bit patterns up to 16-bit code “1000, 0000, 0000, 0000” (decimal notation “32768”) shifted by 15 bits are generated as cyclic bit patterns. An example of the cyclic code table created in this way is shown in FIG. When the bit number Nh of the bit arrangement is 16 and the Hamming distance is 3, the bit pattern is collated using the cyclic code table shown in FIG.

  In the cyclic code table of FIG. 16, all are shown in decimal notation. For example, for the 8-bit reference code “0” in the first row, the regular 16-bit Hamming code “1” and 15 cyclic bit patterns “2” “4” “8” “16” “32” “ 64 ”“ 128 ”“ 256 ”“ 512 ”“ 1024 ”“ 2048 ”“ 4096 ”“ 8192 ”“ 16384 ”“ 32768 ”are recorded in association with each other.

  In this manner, with the initial value set and the cyclic code table created, the mobile terminal is instructed to start as a position detection device. For example, this can be executed by operating a predetermined button when the position detection device is realized by a mobile terminal such as a mobile phone. In the position detection device, when an instruction is input, the acoustic signal input means 100 records the flowing music, digitizes it, and inputs it as a digital acoustic signal. More specifically, the audio input from the omnidirectional microphone (or one channel of the directional microphone) is digitized by the A / D converter.

  Subsequently, the reference frame acquisition unit 110 extracts an acoustic frame including a predetermined number of samples from the acoustic signal input from the acoustic signal input unit 100 as a reference frame (S201). Specifically, reference frames for A type and B type are extracted and read into the acoustic frame holding means 170. The number of samples of one sound frame read as the reference frame by the reference frame acquisition unit 110 needs to be the same as that set by the sound frame reading unit 10 shown in FIG. Therefore, in the present embodiment, the reference frame acquisition unit 110 sequentially reads 4096 samples for each of the A type and B type as reference frames. The acoustic frame holding unit 170 can store two reference frames of A type and B type for each channel, that is, 2.5N samples. When a new reference frame is read, the old reference frame is discarded. It is supposed to be. Therefore, the sound frame holding means 170 always stores four reference frames (continuous 10240 samples).

  The acoustic frame processed by the position detection device can be divided into a reference frame that is set adjacently without interruption from the beginning, and a phase change frame in which the phase is changed. As for the reference frame, when the A type sound frame and the B type sound frame overlap each other by 2048 samples, after setting sample number 1 to sample number 4096 as the first reference frame, the next reference frame is Sample number 2049 to sample number 6144, the next reference frame is set without interruption, such as sample number 4097 to sample number 8192, and the next reference frame is set to sample number 6145 to sample number 10240. Then, for each reference frame, five phase change frames moved by 1/6 frame (about 683 samples) are set. For example, for the first reference frame, five phase change frames configured by 4096 samples starting from sample numbers 683, 1366, 2049, 2732, and 3413 are set. Subsequently, the frequency conversion unit 130 and the code determination parameter calculation unit 140 specify the phase of each read sound frame, determine embedded information, and output a corresponding code (S202). The format of the information to be output has two formats of value 1 and value 2 corresponding to the case where identification information is embedded.

  Here, details of phase determination and code determination in step S202 will be described with reference to the flowchart of FIG. First, it is confirmed whether the phase determination flag is On or Off (S301). When the phase determination flag is On, the phase determination process (S303 to S309) is not performed, and only the code determination process is performed (S302). However, since the phase is not fixed in the initial state and the phase determination flag is Off, the candidate code table B (p) is initialized (S303). Candidate code table B (p) records one reference frame and five phase change frames, a phase number of p = 0 to 5 and a binary code obtained from the states of these six acoustic frames. To do.

  Subsequently, the code determination parameter calculation unit 140 performs a code determination process (S302). Details of the code determination process are shown in FIG. First, the frequency conversion means 130 performs frequency conversion on each read sound frame to obtain each window spectrum (S401). Specifically, the processing according to the above [Equation 12] and [Equation 13] is executed, and the real part A (1, j, p), the imaginary part B (1, j, p), and the real part A of the converted data are executed. (2, j, p) and imaginary part B (2, j, p) are obtained.

  Regarding the processing of S401, in practice, after frequency conversion is performed on the A type sound frame, the frequency conversion is performed on the B type sound frame after being shifted by N / 2 samples. For these conversion data A (1, j, p), B (1, j, p), A (2, j, p), B (2, j, p), the above [Formula 14] is followed. The process is executed to perform amplitude conversion (S402).

  By the processing in the frequency conversion means 130, a frame spectrum expressed by a spectrum that is a component corresponding to the frequency is obtained. Subsequently, after the code determination parameter calculation unit 140 calculates the code determination parameters C and C ′ as described above, the code determination parameters C and C ′ are used to determine the state of the component of the change target frequency band. In other words, a process of determining what value is embedded as a 1-bit value is performed (S403). Specifically, the code determination parameters C and C ′ are calculated by executing the processing according to the above [Formula 15] to [Formula 17]. Then, comparing these two, if C> C ′, the candidate code B is set in the candidate code table B (p), and if C ≦ C ′, the candidate code B ′ is set in the candidate code table B (p). Output to.

  As a specific processing procedure of S403, first, the basic intensity values E (1, j, p), E (2, j, p) are utilized by using the first and second expressions of [Formula 15]. ), E1 (p), E2 (p), E3 (p), E4 (p) are calculated using the first to fourth formulas of [Formula 16]. 17], the candidate code B and the code determination parameter C are calculated. Subsequently, reverberation correction processing is performed using the third to sixth expressions of [Expression 15] to obtain basic intensity values E ′ (1, j, p), E ′ (2, j, p). Is calculated. Then, E1 ′ (p), E2 ′ (p), E3 ′ (p), E4 ′ (p) are calculated using the fifth to eighth expressions of the above [Expression 16], and the above [Expression 16]. 17] is performed to calculate the candidate code B ′ and the correction code determination parameter C ′.

  As a result of the above determination, when either value 1 or value 2 is output to the candidate code table B (p) in the phase p, the phase determination table S (p) Update is performed (S404).

[Formula 20]
When C> C ′, S (p) ← S (p) + C
When C ≦ C ′, S (p) ← S (p) + C ′

  Returning to the flowchart of FIG. 17, the code determination parameter calculation unit 140 stores the code temporarily determined in the code determination process (S302) in the phase p in the candidate code table B (p) (S304).

  Subsequently, it is determined whether or not the processing corresponding to all the phase numbers p has been completed (S305). This determines whether all phase change frames have been processed for a certain reference frame. In this embodiment, since p takes a value from 0 to 5, if the processing is not performed six times, an acoustic frame having a different phase is set by shifting a predetermined number of samples from the acoustic frame being processed. Return and repeat the process. The case where p = 0 is a reference frame, and the case where p = 1 to 5 is a phase change frame. When the processing corresponding to all the phase numbers p is completed, it is determined that the phase corresponding to the phase number pmax having the maximum value in the phase determination table S (p) is the optimum phase, and the candidate code table B (p The code B (pmax) recorded in () is output (S306).

  Subsequently, the phase determination log is updated (S307). The phase determination log records the determined phase, that is, the phase number p for each set of one reference frame and five phase change frames. Then, referring to the phase determination log, it is determined whether or not the phase has been the same a predetermined number of times in the past (S308). In the present embodiment, this number is 10 times. If the phase has been the same a predetermined number of times in the past, the phase determination flag is set to On (S309). As a result, when the same phase continues for a predetermined number of times, the optimum phase is likely to be pmax. Therefore, the phase determination process (S303 to S309) is not performed, and the code determination process is performed only for the phase number p = pmax. (S302) is performed.

  Returning to the flowchart of FIG. When the phase is determined and the code is output, 1 bit corresponding to the output code value is stored in the buffer (S203). Next, the bit counter is incremented by “1” (S204).

  Next, it is determined whether the bit counter is 15 or less or 16 or more (S205). If the bit counter is 15 or less, the process returns to S201 to perform processing for extracting the next A type and B type reference frames.

In the following, the bit counter to determine whether it is either 16 or 15 or less (S205). If the bit counter is 15 or less, the process returns to S201 to perform processing for extracting the next A type and B type reference frames.

If bi Ttokaunta is 16 or more, for matching the bit pattern in the cyclic code table and saved 16-bit bit sequence in the buffer (S206). If the matching bit pattern exists as a 16-bit Hamming code or a cyclic bit pattern as a result of the collation, it is determined that there is a matching pattern, and the corresponding 8 bits of the reference code are extracted and output (S208). When the 16-bit bit string extracted from the acoustic frame is one of the cyclic bit patterns, these bit patterns should be embedded at the time of embedding, and therefore the position where the bit pattern of the corresponding 16-bit Hamming code is shifted Therefore, the 8-bit reference code is extracted. If the matching bit pattern does not exist in either the 16-bit Hamming code or the cyclic bit pattern as a result of the verification in S206, it is determined that there is no matching pattern, and the process returns to S201 to extract the next reference frame. Do.

  If the 8 bits of the reference code are extracted and output in S208, the bit counter is initialized to 0 (S209). Then, the process returns to S201, and processing for extracting from the next reference frame is performed. In the present invention, the same word is embedded by overlapping Na times at the time of embedding, but normally (Na-1) times of the same word is extracted unless 16 bits are accidentally extracted from the first bit at the time of embedding. (Since one word is extracted across different words before and after, a normal error occurs as a result of collation in S206). In this case, since only one word is actually required as information, the 8-bit reference code for the following (Na-2) words is skipped. Of course, depending on the situation, only an 8-bit reference code of (Na-2) words or less may be read, or it may not be read at all.

  By executing the process shown in FIG. 15 for each reference frame, one word of identification information is extracted. If it is determined in S201 that all reference frames have been extracted, the process ends.

  In the processing of S208, the identification information extraction unit 160 first extracts the binary array output by the code output unit 150 in units of Nh bits, and refers to the cyclic code table to obtain the Nw-bit reference code. Convert. If the process is performed correctly, this reference code is the same as any one of the first to fifth codes. Then, using the identification information obtained by the identification information extraction unit 160, the position output unit 200 refers to the positional relationship table in the positional relationship storage unit 190 to acquire and output the positional information. An example of the positional relationship table in the positional relationship storage means 190 is shown in FIG. The correspondence relationship between the detected position and the identification information is recorded in the positional relationship table. As the identification information, a two-word configuration in which each word is one of the first code to the fifth code is recorded in advance. By referring to this positional relationship table with the extracted identification information, information on the detected position can be obtained, so that the position output means 200 outputs the detected position. Specifically, contents such as “near the front left speaker” shown as the detection position in FIG. 19 are displayed and output on the screen of a display device (not shown).

(3.3. About phase correction processing)
As described above, at the time of extraction, it is not always possible to read an acoustic signal corresponding to the acoustic frame embedded at the time of embedding. Therefore, the phase of the acoustic frame is shifted and read in a plurality of ways (six in this embodiment), the optimum phase is determined, and a code corresponding to the acoustic frame specified by the phase is output. Yes. For example, in the case of reading in six ways, the top acoustic frame is originally a sample of sample numbers 1 to 4096, but six pieces composed of 4096 samples starting from sample numbers 1, 683, 1366, 2049, 2732, and 3413 Are processed, and a code corresponding to the optimum acoustic frame is output. As described with reference to the flowchart of FIG. 16, in the present embodiment, when the same phase continues for a predetermined number of times, the processing is performed after that phase is determined.

(Example corresponding to 4.5.1ch surround playback)
In the above example, the case where a 4-channel sound signal is reproduced from four speakers has been described. However, the present invention can deal with more channels. Here, as an example, a case where 5.1ch surround reproduction is supported will be described. In order to support 5.1ch surround reproduction, in addition to the four speakers as described above, an FC speaker that is a central speaker and an LF speaker that is a speaker for emphasizing bass are prepared. Then, the frequency component is changed as described above for the low frequency components of the FC channel acoustic signal to be output from the FC speaker and the LF channel acoustic signal to be output from the LF speaker.

  Here, application of the present invention in the case of 5.1ch surround reproduction will be described in comparison with FIG. 22 in the case of four channels. FIG. 24 is a diagram showing the relationship between six speakers in 5.1ch surround playback and identification information detected at each position. Nine bit strings (9 types of identification information) surrounded by the FL speaker, the FR speaker, the BL speaker, and the BR speaker are the same as those shown in FIG. 22, but in FIG. 24, the intensity of the predetermined frequency component is reflected. Not done. In FIG. 24, a bit string in which the intensity of a predetermined frequency component is reflected in identification information embedded in sound emitted from each speaker. For FL speakers, FC speakers, LF speakers, and FR speakers, a BL speaker, The BR speaker is shown at the bottom of the drawing.

  Even in the case of 5.1ch surround reproduction, the predetermined frequency components of the sound emitted from the FL speaker, FR speaker, BL speaker, and BR speaker are exactly the same as in the case of 4 channels. In this case, both the FC speaker and the LF speaker emit sound in which identification information “00001011” is embedded in a state where the intensity of the predetermined frequency component is reduced. Compared to FIG. 22, even when sound is emitted from the FC speaker and LF speaker, the extraction accuracy of identification information based on the previous four speakers is not affected. When 5.1ch surround playback is made to correspond to 25 types of identification information, in the example shown in FIG. 22, the FC speaker is arranged between the FL speaker and the FR speaker, and the LF speaker is matched to the space of the room. It is arranged at an arbitrary position, and both emit sound in which identification information “00000101” is embedded. In FIG. 24, the LF speaker is arranged between the FL speaker and the FR speaker. However, since the LF speaker (deep bass) has no directivity, it may be arranged anywhere in 5.1ch surround playback. It is a specification.

(5. Method of embedding information more reliably when the signal component is small)
As described so far, according to the present invention, information can be embedded even if the original signal component has a portion close to silence. Of course, this is sufficient, but in the present invention, it is possible to add a process for embedding information more reliably. Specifically, before performing the frequency conversion, weak white noise that is difficult to hear is superimposed on the acoustic signal.

  In this case, the acoustic frame reading means 10 shown in FIG. 3 has a function of adding or subtracting the same value in units of a predetermined number of samples within a predetermined amplitude range after reading a predetermined number of samples as one acoustic frame. Yes. In this embodiment, with respect to the left channel signal Xl (i) and the right channel signal Xr (i) (i = 0,..., N−1), an amplitude range of −32 to +32 (Xl (i), Xr ( i) is a value between −32768 to +32767), and a uniform random number H is generated, and the same value H is used between K (K = 5 in the present embodiment) samples as follows: The process shown in FIG. 6 is executed to update the values of Xl (i) and Xr (i).

[Formula 21]
Xl (i) ← Xl (i) + H
Xr (i) ← Xr (i) -H

  That is, white noise is generated in one acoustic frame by executing the processing according to the above [Equation 21] over N samples. The white noise generation process is performed immediately after the acoustic frame extraction process of S103 and S104 in the flowchart of FIG.

  As described above, even when white noise is generated, the configuration of the position detection device using the acoustic signal on the extraction side is the same as that shown in FIG. 14, and the processing operation is shown in FIGS. It is the same as that according to the flowchart.

(6. Expansion of resolution)
In the above-described embodiment, the component intensities in a predetermined frequency range are based on three types of codes in one direction in each channel (one of which overlaps with other channels, so there are five types of codes in one direction as a whole). Is changed into three stages, so that the relative positions in a total of 25 locations including the vicinity of the left and right speakers in the two-dimensional range are specified. In addition, each channel has two levels of component intensity in a predetermined frequency range based on two types of codes per direction (of which one overlaps with other channels, so there are three types of codes in one direction as a whole). It has also been explained that the relative positions in a total of nine locations including the vicinity of the left and right speakers can be specified in the two-dimensional range by changing to. In the present invention, in order to specify the relative positions in more places, the component strength in the predetermined frequency range may be changed to more stages in each channel based on more codes. For example, by changing the component intensity in a predetermined frequency range in four steps based on four types of codes for each channel in each channel, the relative positions at a total of 49 locations including the vicinity of each speaker in the two-dimensional range are specified. be able to. Further, by changing the component intensity of the predetermined frequency range in five steps based on five types of codes for each direction in each channel, the relative positions at a total of 81 locations including the vicinity of each speaker in the two-dimensional range are specified. be able to. In general, by changing the component intensity of a predetermined frequency range to (n + 1) / 2 stages based on (n + 1) / 2 types of codes in each channel, a total of n ² including the vicinity of each speaker in a two-dimensional range is obtained. The relative position in the place can be specified. By specifying the relative positions of many places, the resolution of position detection increases.

(7. Modified examples of frequency range)
In this application, in order to improve the extraction accuracy, a method of expanding the embedding area in the entire telephone band is adopted. However, in order to improve the reproduction quality, it is also possible to make the width of the lower frequency band to be changed narrower than the above embodiment. is there. Specifically, the lower limit F1 of the change target frequency band is set to 0.85 kHz. In this case, the upper frequency band is 1.7 kHz to 3.4 kHz, and the lower frequency band is 0.85 kHz to 1.7 kHz. However, as a result of the experiment, there is no significant difference in quality compared to the method in which the embedded area is set over the entire telephone band, but rather the disadvantage of a decrease in extraction accuracy is greater, and as a result, superiority is seen. There wasn't. In addition, when setting the embedding area in the entire telephone band, the applicant narrows the lower frequency band to 0.85 kHz to 1.7 kHz instead of the method shown in the above embodiment, and further separates the frequency band below it. Experiments were also conducted on a method of installing 0.34 kHz to 0.85 kHz in triplicate, but as a result of comparison, it is more effective to suppress noise by combining the lower frequency bands into one of 0.34 kHz to 1.7 kHz, and the sensitivity is also high. It was found that it was magnified about twice. The bass part has a large signal energy component, so it seems that the sensitivity was significantly improved by a slight enlargement. In the end, the method that divides the frequency range into four frequency regions and makes the upper and lower frequency bands close to each other to form a double sound pulse concentration can suppress the noise most for human hearing. all right.

(8. Others)
As mentioned above, although it limited about the suitable embodiment of the present invention, the present invention is not limited to the above-mentioned embodiment, and various modifications are possible. For example, in the above embodiment, the number of samples of one acoustic frame is N = 4096, but N = 2048, 1024, 512, etc. may be set. As a result, the number of sound frames per same time is doubled, quadrupled, and quadrupled, and 2 to 8 times of information can be embedded as a whole.

  In the above embodiment, the position detection device using an acoustic signal has been described as an example of a mobile terminal device such as a mobile phone. However, the position detection device may be realized in cooperation with another computer. good. Specifically, the portable terminal device and the dedicated computer are connected so as to be capable of wireless communication, and among the components of the acoustic signal input unit 100 to the acoustic frame holding unit 170, those having a large calculation load are processed by the dedicated computer. For example, the acoustic signal input unit 100, the reference frame acquisition unit 110, the phase change frame setting unit 120, the identification information extraction unit 160, the cyclic code table creation unit 180, the positional relationship storage unit 190, and the position output unit 200 are included in the portable terminal device. The dedicated computer is provided with the frequency conversion unit 130, the code determination parameter calculation unit 140, the code output unit 150, and the sound frame holding unit 170 so that necessary information can be communicated between them. This makes it possible to perform high-speed processing even when the processing performance of the portable terminal device is low.

  Further, in the above-described embodiment, a case where four-channel stereo sound signals having different contents of the front left channel, the front right channel, the rear left channel, and the rear right channel is used as an acoustic signal in which identification information is embedded has been described as an example. However, the original acoustic signal itself may be the same for all four channels. In other words, monophonic audio signals in which different identification information is embedded (exactly the same audio signals except for a predetermined frequency range) may be reproduced from the FL speaker, the FR speaker, the BL speaker, and the BR speaker.

  Further, the change of the predetermined frequency component by the frequency component changing means 30 is executed according to FIG. 5 and [Equation 9], but various known methods are used to change the frequency component according to the bit value to be embedded. For example, it is possible to use a technique as shown in Patent Document 1. In this case, on the extraction side, the extraction is naturally performed by a method corresponding to the embedding method.

It is a figure which shows the basic concept of a one-dimensional position detection. It is the figure explaining the concept of the one-dimensional position detection in detail. It is a functional block diagram of an information embedding device for an acoustic signal. It is a figure which shows the time direction window function used by this invention. It is a figure which shows the state of a change of the component of the change object frequency band in one Embodiment of this invention. It is a figure which shows an example of the code conversion table of an 8-bit code and a 16-bit code. It is a flowchart which shows the process outline | summary of code conversion table preparation. It is a figure which shows the example of a combination of a 1st code | symbol-a 5th code | symbol. It is a flowchart which shows the process outline | summary of the apparatus shown in FIG. It is a figure for demonstrating the embedding process which does not use amplitude conversion. It is a figure for demonstrating the extraction process which does not use amplitude conversion. It is a figure for demonstrating the embedding process using amplitude conversion. It is a figure for demonstrating the extraction process using amplitude conversion. It is a functional block diagram of the position detection apparatus using the acoustic signal which concerns on this invention. It is a flowchart which shows the process outline | summary of the apparatus shown in FIG. It is a figure which shows an example of the cyclic code table used by this invention. 16 is a flowchart showing details of phase determination and code output in S202 of FIG. It is a flowchart which shows the detail of the code | symbol determination process of S302 of FIG. It is a figure which shows an example of the positional relationship table which recorded the relationship between a detection position and extraction identification information. It is a figure explaining how a predetermined frequency component is changed in order to make intensity "large" "medium" "small" when embedding identification information. It is explanatory drawing of the basic concept of this invention. It is the figure explaining the concept of this invention in detail. It is detailed explanatory drawing of the concept of this invention at the time of increasing a detection position. It is a detailed explanatory view of the concept of the present invention in the case of 5.1ch surround playback.

DESCRIPTION OF SYMBOLS 10 ... Acoustic frame reading means 20 ... Frequency conversion means 30 ... Frequency component change means 40 ... Frequency reverse conversion means 50 ... Modified acoustic frame output means 60 ... Storage means 61 ... Acoustic signal storage unit 62 ... identification information storage unit 63 ... modified acoustic signal storage unit 70 ... bit array creation means 100 ... acoustic signal input means 110 ... reference frame acquisition means 120 ... phase Change frame setting means 130 ... frequency conversion means 140 ... code determination parameter calculation means 150 ... code output means 160 ... identification information extraction means 170 ... acoustic frame holding means 180 ... cyclic code table Creation means 190 ... Position relation storage means 200 ... Position output means

Claims (8)

When the acoustic signal is reproduced by at least four speakers with respect to an acoustic signal composed of a time-series sample sequence of at least four channels, four locations in the vicinity of the respective speakers (n ²⁾ -5) point (n is 5 or more odd number), and so different from n ² kinds of identification information is extractable from each other at the location of the four central one location a total of n ² places speakers, different n ² or to each other A device that embeds the identification information of n types of front and rear identification information and n types of right and left identification information in an inaudible state,
From the acoustic signal, an acoustic frame reading means for reading a predetermined number of samples for each channel and creating an acoustic frame;
Among the read sound frames of each channel , odd-numbered and even-numbered one is A type and the other is B type, and frequency conversion is performed on the A type sound frame using a first window function, A frequency conversion is performed on the B-type acoustic frame using a second window function, and a first window spectrum that is a spectrum corresponding to the first window function and a second spectrum that corresponds to the second window function. A frequency conversion means for obtaining a window spectrum;
Two spectrum sets that do not overlap each other in a predetermined frequency range are extracted from the generated window spectra so that the frequency width of the higher frequency spectrum set becomes larger , and each spectrum set is reduced to 1/2. The spectrum set obtained by equally dividing the frequency spectrum into four spectrum sets for each window spectrum and extracting the spectrum set from the first window spectrum in order of increasing frequency is 1D1, 1D2, 1U1, 1U2, and from the second window spectrum. When the extracted spectrum set is 2D1, 2D2, 2U1, 2U2 in order of frequency , based on the bit values that constitute the original front / rear or left / right identification information to be embedded in the channel, the intensity value of 1D1 and 2D2 One of the product of the intensity value, the intensity value of 1D2 and the intensity value of 2D1 is predetermined from the other. Becomes larger than the degree, at the same time the product of the intensity values of the intensity values and 2U2 of 1U1, either the product of the intensity values of the intensity values and 2U1 of 1U2 is the size Kunar so than a predetermined degree than the other To change,
When using the front-rear identification information, the degree to which the intensity value is further changed is set to (n + 1) / 2 steps within the range in which the magnitude relationship of the intensity values is not reversed, and is extracted approximately at the center with other speakers. When the bit value of the power front and rear identification information is the same as the bit value of the original front and rear identification information, the front and rear identification should be extracted at a position adjacent to the position where the original front and rear identification information is extracted. When the bit value of the information is different from the bit value of the original front and rear identification information, it is set to a minimum degree,
When using the left / right identification information, the degree to which the intensity value is further changed is set to (n + 1) / 2 steps within the range in which the magnitude relationship of the intensity values is not reversed, and is extracted approximately at the center with other speakers. Left and right identification to be extracted at a position adjacent to the position where the original left and right identification information is extracted so that the bit value of the right and left identification information is the same as the bit value of the original left and right identification information When the bit value of the information is different from the bit value of the original left and right identification information, the frequency component changing means configured to be a minimum degree;
Frequency inverse transform means for performing frequency inverse transform on each window spectrum including the modified spectrum set to generate a modified acoustic frame;
Modified acoustic frame output means for sequentially outputting the generated modified acoustic frames;
An information embedding device for an acoustic signal, comprising: Oite to claim 1,
The frequency component changing means includes a product of an intensity value of 1D1 and an intensity value of 2D2, a product of an intensity value of 1D2 and an intensity value of 2D1, a product of an intensity value of 1U1 and an intensity value of 2U2, and 1U2. Of the product of the intensity value of 2 and the intensity value of 2U1, the spectrum set on the side to be changed to be larger is multiplied by the coefficient α, and the spectrum set on the side to be changed to be smaller. Is an intensity multiplied by a coefficient β smaller than the coefficient α, and the coefficient α and the coefficient β are determined according to the degree of (n + 1) / 2 stages in which the intensity value is changed. An information embedding device for an acoustic signal. In claim 2 ,
The frequency component changing means is the product of the intensity of the spectrum set that changes the intensity product to be relatively large, and the product of the intensity of the spectrum set that changes the intensity product to be relatively small. When the divided value γ is smaller than 1, a coefficient obtained by dividing the coefficient α by the square root of γ and a coefficient obtained by multiplying the coefficient β by the square root of γ are used instead of the coefficients α and β. An information embedding device for an acoustic signal. In any one of claims 1 to 3,
The acoustic frame reading means reads the A type acoustic frame and the B type acoustic frame by overlapping a predetermined number of samples for each channel,
The apparatus for embedding information in an acoustic signal, wherein the modified acoustic frame output means outputs the generated modified acoustic frame by connecting it to a preceding modified acoustic frame. In any one of claims 1 to 4,
The apparatus for embedding information in an acoustic signal, wherein the frequency component changing means sets the predetermined frequency range as 0.34 kHz or more and 3.4 kHz or less. The program for functioning a computer as an information embedding apparatus with respect to the acoustic signal as described in any one of Claims 1-5 . The acoustic signal is reproduced by at least four speakers from an acoustic signal in which n ² types of identification information are embedded in a state incapable of being heard in advance, and four locations near each speaker (n ² −5) ) And a device for detecting a total of n ² positions of one center of the four speakers,
A positional relationship storage means for storing a positional relationship in which a relative position based on each speaker is associated with the identification information;
An acoustic frame acquisition means for monophonically inputting an acoustic signal at a predetermined position in a space where the acoustic signal is reproduced, digitizing a predetermined section, and acquiring an acoustic frame composed of a predetermined number of samples;
Of the read acoustic frames, odd-numbered and even-numbered ones are A type and the other is B type, and frequency conversion is performed on the A type acoustic frames using a first window function, and the B type Frequency conversion is performed on the acoustic frame using the second window function to obtain a first window spectrum that is a spectrum corresponding to the first window function and a second window spectrum that is a spectrum corresponding to the second window function. Frequency conversion means;
Two spectrum sets that do not overlap each other in a predetermined frequency range are extracted from the generated window spectra so that the frequency width of the higher frequency spectrum set becomes larger, and each spectrum set is reduced to 1/2. Equally divided into frequency widths, intensity values of four spectrum sets extracted from the first window spectrum are calculated as E1, E3, E5, and E7 in order of increasing frequency, and the four spectrum sets extracted from the second window spectrum The intensity values of the spectrum set are calculated as E2, E4, E6, and E8 in ascending frequency, and the magnitude relationship between the product of E1 and E4 and the product of E2 and E3, and the product of E5 and E8, and E6 and E7 Encoding means for extracting the embedded bit value based on the magnitude relationship with the product;
The extracted bit values are converted in word units according to a predetermined rule, and n kinds of front and rear identification information and left and right identification information are extracted in time series, and these are combined and extracted as the n ² kinds of identification information. Identification information extraction means;
Using the identification information obtained by the identification information extracting means, referring to the positional relationship storage means, obtaining position information, and outputting the obtained position information;
A position detection device using an acoustic signal characterized by comprising: The program for functioning a computer as a position detection apparatus using the acoustic signal of Claim 7 .

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