JP2010078728A - Device for embedding information in sound signal and device for extracting information from sound signal - Google Patents

Abstract

<P>PROBLEM TO BE SOLVED: To provide a device for embedding information in a sound signal and a device for extracting information from the sound signal, which extract separate information from information which is separately embedded in each speaker, by utilizing sounds emitted by a plurality of speakers. <P>SOLUTION: In the case where "1010010011111001"(a)" is used as additional information of an L channel, and "0111101110001100"(c) as an R channel, intensity in a predetermined frequency range is relatively changed between the R and L channels, when each bit is embedded. For example, one bit (from left) is "1" at an L-side, and "0" at an R-side, but a signal level of the L-side is relatively "weak", and a signal level of the R-side is relatively "strong", and thereby, an influence of the R-side whose signal is stronger becomes large around a center of right and left speakers, and " 0" is detected as a result of analysis. By processing each bit in the same way, additional information of "0111011010011001"(b) is extracted around the center of both right and left speakers. <P>COPYRIGHT: (C)2010,JPO&INPIT

Description

  The present invention relates to the protection of music copyright (monitoring illegal copying) in the field of package music for viewing for consumer and business use using CDs and DVDs, and the field of broadcasting and network music distribution distributed for commercial purposes by broadcasters and the like. ) And music attribute information (music title search service) field, museum, event information service service field linked to exhibition explanation narration, URL and other information extracted from audio signals of broadcast programs and CD / DVD packages In addition, the present invention relates to a non-contact Internet gateway service field in which a mobile phone is used to access a web site related to a predetermined content and extract detailed information or answer a questionnaire.

  Conventionally, as a service for providing music attribute information that allows you to know the title of music that is playing, you can query the broadcast station for the date and time of the broadcasted music, and record music fragments that are played on mobile phones. In addition, there is a service that matches the melody registered in the database. The present applicant has further developed this and proposed a method of embedding attribute information (additional information) by changing the ratio of frequency components of an acoustic signal according to the bit value of the attribute information (Patent Document 1). reference).

In addition, in the invention described in Patent Document 1, the present applicant has a problem that when two speakers are separated by a wide hall or the like, it cannot be extracted near the center where the sound from both channels is mixed. In order to solve this problem, 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 Japanese Patent Application No. 2007-98970

  With the technique of the above-mentioned Patent Document 2, it is possible to extract information even in the vicinity of the center of the two speakers. This is information that can be detected from the sound from one of the speakers. There wasn't. On the other hand, there is a demand for extracting additional information in the vicinity of each of the two speakers and detecting other information in the vicinity of the center of the two speakers.

  Accordingly, the present invention provides an information embedding device for an acoustic signal that can extract information separate from information embedded in sound emitted from each speaker alone, using sound emitted from a plurality of speakers, and An object is to provide an apparatus for extracting information from an acoustic signal.

  In order to solve the above-described problem, in the present invention, when a stereo sound signal composed of a two-channel sample sequence in time series is reproduced by left and right speakers, the vicinity of the left and right speakers and the center Is a device that embeds three different types of additional information in an inaudible state so that three different types of additional information can be extracted at the three extraction positions of the stereo sound signal. A sample is read for each channel, an acoustic frame reading means for creating an acoustic frame L and an acoustic frame R, and a frequency conversion using a predetermined window function for the read two-channel acoustic frame L and acoustic frame R To obtain a window spectrum L for the acoustic frame L and a window spectrum R for the acoustic frame R, respectively. Corresponding to the window spectrum L based on the bit array 1 of the first additional information 1 to be embedded, by extracting a plurality of spectrum sets in a predetermined frequency range from the generated window spectrum and the wave number conversion means The ratio of intensity values between a plurality of spectrum sets is changed by a predetermined modification ratio L, and the modification ratio L is determined based on the difference from the bit array 3 of the third additional information 3 corresponding to the bit array 1 Based on the bit array 2 of the second additional information 2 to be embedded, the intensity value ratio between the plurality of spectrum sets corresponding to the window spectrum R is changed by a predetermined modification ratio R, and A frequency component changing means for determining a modification ratio R based on a difference from the bit array 3 corresponding to the bit array 2; Frequency inverse transform means for generating a modified acoustic frame L and a modified acoustic frame R by performing frequency inverse transformation on each window spectrum including the generated spectrum set, and the generated modified acoustic frame L and modified acoustic frame R Are provided for embedding information in an acoustic signal having a modified acoustic frame output means for sequentially outputting.

  Further, in the present invention, the stereo sound signal is reproduced by the left and right speakers from the stereo sound signal in which the three types of additional information are embedded in a state incapable of being heard in advance, and is extracted at three positions in the vicinity of the left and right speakers and in the center. A device for extracting each of the three types of additional information, wherein the stereo sound signal is monaurally input at a predetermined position in a space where the stereo sound signal is reproduced in stereo, and a predetermined number of samples are digitized. An acoustic frame acquisition means for acquiring an acoustic frame, and among the read acoustic frames, odd-numbered and even-numbered one is A type and the other is B type. Frequency conversion is performed using a one window function, frequency conversion is performed using a second window function for the B type acoustic frame, and the first window function is performed. Frequency conversion means for obtaining a first window spectrum that is a corresponding spectrum and a second window spectrum that is a spectrum corresponding to the second window function, and two pieces that do not overlap each other in a predetermined frequency range from the generated window spectra Are extracted so that the frequency width of the higher frequency spectrum set is larger, and each spectrum set is equally divided into ½ frequency widths, and the four extracted from the first window spectrum are extracted. The intensity values of the spectrum set are calculated as E1, E3, E5, and E7 in ascending order of frequency, and the intensity values of the four spectrum sets extracted from the second window spectrum are set as E2, E4, E6, and E8 in ascending order of frequency. Calculated based on the magnitude relationship between the product of E1 and E4 and the product of E2 and E3, and the magnitude relationship between the product of E5 and E8 and the product of E6 and E7. And an audio signal having encoding means for extracting the embedded bit value, and additional information extraction means for converting the extracted bit value in a word unit according to a predetermined rule and extracting the three types of additional information. An apparatus for extracting information from a computer is provided.

  According to the present invention, a total of three types of additional information, ie, first and second additional information for two channels and third additional information for center detection are prepared, and bit values corresponding to the two channels are embedded. At this time, since the degree of intensity of the frequency component to be changed in the predetermined frequency range is changed according to the bit value of the additional information for center detection, the sound acquired with the microphones installed near the center of both the left and right speakers is The third additional information can be extracted, and the sound emitted from each speaker alone can be extracted using the sound emitted from a plurality of speakers. It is possible to extract information that is separate from the embedded information.

Hereinafter, embodiments of the present invention will be described in detail with reference to the drawings.
(1. Basic concept of the present invention)
First, the basic concept of the present invention will be described in comparison with the prior art. FIG. 1A is a diagram illustrating conventional extraction of additional information using an acoustic signal. In the example of FIG. 1A, ID1 is embedded as additional information in the L channel acoustic signal, and ID2 is embedded as additional 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 speaker L and the sound captured from the microphone near the speaker R, respectively. However, the extraction device cannot correctly extract the ID from the sound taken from the microphones that are approximately equidistant from the speakers L and R. This is because the sound from the speaker L and the sound from the speaker R are mixed in the microphones that are substantially equidistant from the speakers L and R.

  In the present invention, as shown in FIG. 1B, the extraction device extracts ID1 and ID2 from the sound taken in from the microphone near the speaker L and the sound taken in from the microphone near the speaker R, respectively, as in the prior art. Separate IDs 3 are extracted from the sound taken from the microphones that are approximately equidistant from the speakers L and R. That is, the sound from the speaker L in which ID1 is embedded and the sound from the speaker R in which ID2 is embedded are simultaneously captured to extract ID3.

  Next, the concept of the present invention will be described in more detail. For example, let us consider a case where additional information as shown in FIG. 2 is embedded in the sound signals of the L channel and the R channel, the sound signal is reproduced, and sound is emitted from each speaker. In the example of FIG. 2, 16 bits of “1010010011111001” as shown in FIG. 2A are added as L channel additional information, and 16 bits of “0111101110001100” are shown as R channel additional information shown in FIG. Is used. When embedding each bit, the frequency intensity in a predetermined frequency range is relatively changed between the left and right channels, so that “0111011010011001” as shown in FIG. Bit additional information is extracted.

  The state of FIG. 2 is described for each bit. In the first bit (from the left), the L side is “1” and the R side is “0”, but the L side signal level is relatively “weak”. Since the side signal level is relatively “strong”, the influence of the signal on the R side having a strong signal increases near the middle between the left and right speakers, and “0” is detected as a result of the analysis. Similarly, the second bit is also influenced by the R side and detects “1”. In the third bit, the signal level is “strong” on both the L side and the R side. In such a case, the same bit is always embedded in the L side and the R side. Therefore, the predetermined signal state reaches the amplified state from the left and right speakers, and “1” that is the same as the L side and the R side is detected near the middle of the left and right speakers.

(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 additional 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 additional 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 additional information extracted from the additional 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 means 60 includes an acoustic signal storage unit 61 that stores a stereo acoustic signal to be embedded as additional information, an additional information storage unit 62 that is configured as a bit string and stores additional information embedded in the stereo acoustic signal, and additional 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 additional information from the additional information storage unit 62, referring to the code conversion table for each word of the additional information, and creating a corresponding bit array.

  The additional information is information that should be added to the sound information and embedded, and includes attribute information such as a title and artist name, and other information other than the attribute information. In the present embodiment, Nw and Nh are set to Nw = 7 and Nh = 16, respectively. In the present embodiment, since the ASCII code is adopted as the code format of the additional information, Nw = 7 in the additional information, and 7 bits are one word. The bit array created by the bit array creating means 70 is 16 bits, and after the bit array is created, this is 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 sound frame reading means 10 reads a predetermined number N of samples as one sound frame from each of the left and right channels of the stereo sound signal stored in the sound 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. Therefore, the acoustic frame reading means 10 sequentially reads 4096 samples for each of the left channel and the right channel as acoustic frames.

  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.

[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.

  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. 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 larger the ratio between the two, the higher the accuracy at the time of extraction. However, the ratio of interpolation becomes 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 whether the bit detected at the center is emitted from the left or right speaker. Therefore, in this embodiment, in a specific process described later, coefficients α ₁ = 0.8 and α ₂ = 0.66 for setting a strong state, and coefficients β ₁ = 0. 2, β ₂ = 0.33. 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, the frequency component changing means 30 corrects the coefficients α ₁ , α ₂ , β ₁ , and β ₂ by executing the processing according to the following [Equation 8] according to the value of the intensity ratio γ to obtain the coefficient Obtain α and β. In the following [Equation 8], α _k means α ₁ or α ₂ . If the bit value to be embedded in the channel and the bit value of the intermediate code are the same, α ₁ is used as α _k , and if the bit value to be embedded in the channel and the bit value of the intermediate code are different, α _k is set to α ₂ Is used.

[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 11]
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 11]. Using the real part Ar ′ (1, j) of the right channel, imaginary part Br ′ (1, j), etc., according to the following [Equation 12], X ′ '(I) and Xr' (i) are calculated. For the frequency component not modified in the above [Equation 11], 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 12]
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 [Formula 12], 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 12] indicate 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 12], 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 11]. , Using the real part Ar ′ (2, j) and imaginary part Br ′ (2, j) of the right channel, the processing according to the following [Equation 13] is performed, and Xl ′ (i), Xr ′ (i ) Is calculated. For frequency components that are not modified in the above [Equation 11], in the following [Equation 13], 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 13]
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 13], 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 Nw-bit registration ranks and Nh-bit Hamming codes 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 6 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 (S601). Specifically, the 7-bit 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 7-bit code KF is set to 1, 16-bit code The initial value of HF is 2. Subsequently, i = 0 is initially set (S602). Next, the Hamming distance between the 16-bit code HF and another 16-bit code already registered in the i-th code conversion table is calculated (S603).

  When the calculated Hamming distance is less than 6, the value of HF is updated by 1 (S604), and then the process returns to S602 and the next 16-bit code HF is processed. On the other hand, if the calculated Hamming distance is 6 or more, the value of i is incremented by 1 and updated (S605). If i is less than KF-1, the process returns to S603 to calculate the hamming distance from the 16th code registered in the code conversion table. If i becomes KF-1 or more after S605, the 16-bit code HF is registered at the position of the 7-bit code KF in the code conversion table, and the values of KF and HF are respectively incremented by one and updated. (S606). If KF is less than 65536, the process returns to S602 to perform processing for the next 7-bit code KF. If KF is 65536 or more, 16-bit codes HF corresponding to all 7-bit codes KF have been registered, and thus the code conversion table creation process ends.

  In this way, a code conversion table as shown in FIG. 6 is created. As shown in FIG. 6, 16-bit codes are registered in the code conversion table in association with values 0 to 127 that can be taken by 7-bit codes. As can be seen from the binary representation of the 16-bit code, all the 16-bit codes have a Hamming distance of 6 or more. In the example of FIG. 6, for convenience of explanation, a decimal notation of a 7-bit code and a decimal notation and a binary notation of a 16-bit code are shown. Code bit strings are registered in association with each other.

  When the code conversion table as shown in FIG. 6 is prepared, next, three types of additional information are created, and these are associated with each other to create an additional information correspondence table. In the present embodiment, the additional information is created by an ASCII code, and among these, 62 uppercase letters, lowercase letters, and numbers are adopted. These 62 characters correspond to 62 out of 127 7-bit codes in the code conversion table of FIG. 6 according to the ASCII code. All three types of additional information need to be selected from the 62, but are characteristic of the present invention. Since the third additional information must be the same as at least one bit of the first and second additional information, a device is required for creation.

  Here, the first additional information and the second additional information in a state where 62 7-bit codes corresponding to 62 uppercase letters, lowercase letters, and numbers of the ASCII code are converted into 16-bit codes by the code conversion table of FIG. FIG. 8 shows a correspondence rule that allows the third additional information to be established. In FIG. 8, four possible third additional information are defined for each combination of one first additional information and second additional information. For example, in FIG. 1-NO. 4, the first additional information is a combination of a number “0” and the second additional information is a lowercase letter “m”. However, the third additional information is a combination of the first additional information, the first additional information in the 16-bit bit string. 2 indicates that the number “4”, the number “9”, the lowercase letter “d”, and the lowercase letter “i” can be taken depending on which bit value of the additional information is used. In FIG. 8, each bit is indicated by “1” and “2”. Based on the additional information correspondence rule as shown in FIG. 8, the creator creates first to third additional information.

  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 the processing for one word of additional information of one channel. Therefore, in the case of two channels, the same processing is executed in parallel on both channels.

  In FIG. 9, first, the bit array creation means 70 creates a corresponding bit array by referring to the code conversion table for each word of the additional information extracted from the additional information storage unit 62 (S101). Specifically, first, from the additional information storage unit 62, the first additional information for the left (L) channel or the second additional information for the right (R) channel is read in one word (7 bits) at the center. The third additional information is extracted by one word (7 bits), and a 16-bit bit array composed of corresponding Hamming codes is created with reference to the code conversion table shown in FIG.

  Each 16 bits is read into a register in a computer used as an information embedding device for an acoustic signal. Thus, in the additional information storage unit 62, one word is 7 bits, but at the time of embedding processing, processing for one word in the additional information is performed with this 16-bit array.

  Next, the frequency component changing unit 30 performs a process of reading one bit from Nh (= 16) bits held in the register (S102). 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] and [Formula 8] is executed. Then, the conversion ratios α and β are determined (S105). Specifically, it is determined whether the original channel bit values of the bit values and the intermediate code is either the same read, if the same with alpha ₁ as alpha _k, if different with alpha ₂ as alpha _k Then, α and β are determined by executing the processing according to the above [Equation 8]. Using this determined conversion ratio, the frequency component changing unit 30 executes the processing according to the above [Equation 11] according to the value 1 and the value 2 received from the bit array creating unit 70, and changes the frequency band to be changed. The state of the component 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 11]. , J) and the like, the imaginary part Br ′ (1, j) and the like are used to perform processing according to the above [Equation 12] 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 transform 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 11] 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. Processing according to [Formula 13] 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. 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 to perform processing for reading the next word of the additional information. . When the processing is completed for all the words of the additional information, the processing returns to the first word of the additional 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 the 1-word 7-bit ASCII code, which is the original additional information, is converted and prepared as a 16-bit bit array, and processing is performed for one word of additional information. In the present invention, as long as there is an agreement with the extraction side, it is possible to record one word of additional information in units of other number of bits.

  Of the modified acoustic signal obtained as described above, for the portion in which the additional information is embedded, the components of the change target frequency band have 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 extraction device)
Next, an apparatus for extracting information from an acoustic signal according to the present invention will be described. FIG. 14 is a block diagram showing an embodiment of an apparatus for extracting information from 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 additional information. Extraction means, 170 is an acoustic frame holding means, and 180 is a conversion table creation 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. Also, when information is embedded in the apparatus shown in FIG. 3, the information is reproduced in stereo. However, when additional information embedded in the L channel is extracted, a microphone is installed near the L speaker, and R When extracting additional information embedded in a channel, a microphone is installed in the vicinity of the R speaker, and when extracting third additional information using information embedded in both, an intermediate between the L speaker and the R speaker. Install a microphone in 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 to capture 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 multiple phases, even if the original acoustic signal is stereo, 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 14] 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 14]
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 [Formula 15] to obtain a real part A (2, j, p) and an imaginary part B (2, j, p) of the converted data. ing.

[Formula 15]
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 16]. 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 16]
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 17].

[Formula 17]
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 17], the fifth formula from the top (second from the bottom) is the B type in which 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 17], a negative value is obtained. If this happens, set it to 0.

  In the above [Equation 17], 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 18].

[Formula 18]
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 17] and [Equation 18], 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 in which 1 bit is embedded 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 19].

[Formula 19]
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 19], by replacing E1 (p) to E8 (p) with E1 ′ (p) to E8 ′ (p), respectively, the candidate code B is substituted. 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. 20] is executed, the candidate code B1 is provisionally determined and the code determination parameter C1 is calculated.

[Formula 20]
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 a process according to the following [Equation 21], the candidate code B2 is provisionally determined and the code determination parameter C2 is calculated.

[Formula 21]
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 20] and [Equation 21], by replacing E1 (p) to E8 (p) 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 additional information extraction unit 160 extracts the binary array output by the code output unit 150 in units of Nh bits, refers to the code inverse conversion table, converts it to the registration rank of Nw bits, and further, a predetermined rule It has a function to extract as additional information having meaning after conversion by the above. The acoustic frame holding means 170 is a buffer memory capable of holding two consecutive reference frames (a total of four reference frames for each channel) for each of the A type and B type for each channel. Similarly to the above-described conversion table creation device, the conversion table creation means 180 has Nh (> Nw) bits with a Hamming distance of at least 4 or more with respect to all the 2 Nw power registration ranks that Nw bits can take. By assigning a Hamming code, it has a function of creating a code conversion table in which an Nw bit registration order and an Nh bit Hamming code are associated with each other. 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 more easily achieve the object of the present invention, it is desirable to use a portable terminal device such as a cellular phone as hardware. Note that in this specification, even a portable terminal device such as a cellular phone has an arithmetic processing function as a kind of computer.

(3.2. Processing operation of extraction device)
Next, the processing operation of the apparatus for extracting information from the acoustic signal shown in FIG. 14 will be described. When the extraction device is activated, first, the conversion table creation means 180 creates a code conversion table. The code conversion table is created by the conversion table creating means 180 by the process according to the flowchart of FIG. 7 as in the case of the conversion table creating apparatus, and the code conversion table as shown in FIG. 6 is obtained.

  Next, the processing operation of the extraction device after the code conversion table creation process will be described. The extraction apparatus according to the present invention can be set not to perform error correction when an error is detected by a check code, or can be set to perform 1-bit error correction. Hereafter, the processing operation of the extraction apparatus in a setting in which error correction is not performed will be described according to the flowchart of FIG. First, the apparatus initializes the phase determination table S (p), the phase determination log, the phase determination flag, and the bit counter (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. For the bit counter, 0 is set as an initial value.

  In this way, when the code conversion table is created and the initial value is set, if the user wants to know the attribute information such as the song name of the music that is playing, first, the extraction device On the other hand, an activation instruction as an extraction device is given. For example, this can be executed by operating a predetermined button when the extraction device is realized by a mobile terminal such as a mobile phone. When an instruction is input to the extraction device, the acoustic signal input unit 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 extraction 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 the additional information is embedded.

  Here, details of the 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 14] and [Equation 15] is executed, and the real part A (1, j, p), imaginary part B (1, j, p), real part A of the converted data is 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 (2, j, p), B (2, j, p), A (2, j, p), B (2, j, p), the above [Expression 16] 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 [Equation 17] to [Equation 19]. 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 used by using the first and second expressions of [Expression 17]. ), E1 (p), E2 (p), E3 (p), E4 (p) are calculated using the first to fourth formulas of [Formula 18]. 19], 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 17] to obtain basic intensity values E ′ (1, j, p), E ′ (2, j, p). Is calculated. Then, E1 ′ (p), E2 ′ (p), E3 ′ (p), and E4 ′ (p) are calculated using the fifth to eighth formulas of [Formula 18]. 19] to calculate the candidate code B ′ and the correction code determination parameter C ′.

  Further, 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 22]
When C> C ′, S (p) ← S (p) + C
When C ≦ C ′, S (p) ← S (p) + C ′

  Returning to the flowchart of FIG. 16, 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.

  If the bit counter is 16 or more, the conversion table is referred to convert the 16-bit bit array stored in the buffer into a 7-bit bit array (S206). Details of the decoding process will be described with reference to the flowchart of FIG.

  The additional information extraction unit 160 first performs initialization processing (S701). Specifically, a 7-bit code KF and a minimum hamming distance HD are defined, and a process for initializing KF = 0 and HD = 2 is performed. Subsequently, i = 0 is initially set (S702). Next, a 16-bit bit array stored in the buffer is set to a 16-bit code HF, and a Hamming distance hd with a 16-bit code corresponding to the i-th code conversion table is calculated. If hd <HD, KF = i , HD is updated to hd (S703). Then, the value of i is incremented by 1 (S704), and if the value of i is less than 128, the process of S703 is repeated. The additional information extraction means 160 calculates the hamming distance hd for all 128 hamming codes recorded in the code conversion table shown in FIG. 6, and the minimum hamming distance HD and 7 bits when the hamming distance is minimum. A reference code KF is obtained.

  When the minimum hamming distance HD is obtained, the process returns to FIG. 15 to determine whether the obtained minimum hamming distance HD is 0 or 1 or more (S207). If it is determined in S207 that the minimum Hamming distance is 1 or more, the process returns to S201 to perform processing for extracting the next reference frame.

  When a 7-bit bit array is obtained, 1 bit is added to the 7 bits and output (S208). Specifically, a bit “0” is added to 7 bits to output as one word in the ASCII code. When the Hamming distance is 1 or more, there is a high possibility that the 16 bits used for collation are shifted from the word in the additional information. In this case, the first 1 bit is discarded, and processing for obtaining a new 1 bit is performed by the processing from S201 to S204.

  If the minimum hamming distance is determined to be 0, there is a high possibility that the portion is a word delimiter, so if it is really a delimiter, then if it is extracted 16 bits at a time, all will be extracted accurately in units of words. It can be performed. On the other hand, even if the minimum hamming distance is determined to be 0, it is a coincidence and may not be a word break. In such a case, there is a high possibility of nonconformity at the next inspection, and the correct division can be accurately determined after repeating several times. In S208, when 1 bit is added to 7 bits and output, the bit counter is initialized to 0 (S209). Then, the process returns to S201 to perform processing for extracting the next reference frame.

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

  In the process of S208, the additional information extraction unit 160 first converts the value output by the code output unit 150 according to a predetermined rule and extracts it as meaningful additional information. As the predetermined rule, various rules can be adopted as long as the information intended by the person who embeds the information can be recognized by the recipient, but in this embodiment, the ASCII code is adopted. is doing. That is, the additional information extraction unit 160 recognizes the bit value array obtained from the code determined by the code determination parameter calculation unit 140 and output from the code output unit 150 in units of 1 byte (8 bits), and recognizes this as ASCII. Recognizes character information according to the code. The character information thus obtained is displayed and output on a screen of a display device (not shown).

  Therefore, if the embedding device embeds attribute information such as the song title and artist in the sound signal as character information, the user wants to know the song title and artist by hearing that the music is playing. If a predetermined operation is performed on the mobile terminal that functions as the extraction device, attribute information such as a song title and an artist is displayed as character information on the screen of the mobile terminal.

  In the above example, the processing operation of the extraction apparatus in a setting in which error correction is not performed has been described according to the flowchart of FIG. When error correction is not performed, it is possible to output one word of additional information only when no error occurs.

  Next, the processing operation of the extraction apparatus in a setting for correcting 1-bit error will be described with reference to the flowchart of FIG. 19 includes a portion that performs the same processing as in FIG. Therefore, parts that perform the same processing as in FIG. Also in the example of FIG. 19, first, initialization processing is performed (S200). In this initialization process, as in the example of FIG. 15, the phase determination table S (p), the phase determination log, the phase determination flag, and the bit counter are initialized, and further, the automatic correction mode is set to OFF.

  Subsequently, as in FIG. 15, after extracting an acoustic frame composed of a predetermined number of samples as a reference frame (S201), the phase of each read acoustic frame is specified, and the embedded information is determined. The corresponding code is output (S202). Note that the processing of S202 is as shown in FIGS.

  When the phase is determined and the code is output, 1 bit corresponding to the output code value is stored in the buffer (S203), and the bit counter is incremented by "1" (S204). Then, 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 reference frame.

  If the bit counter is 16 or more, the conversion table is referred to convert the 16-bit bit array stored in the buffer into a 7-bit bit array (S206). Subsequently, it is determined whether the obtained Hamming distance is 0 or 1 or more (S207). If it is determined in S207 that the hamming distance is 0, immediately before the bit string output processing in S208, processing for setting the automatic correction mode to ON is performed (S210).

  If it is determined in S207 that the Hamming distance is 1 or more, it is further determined whether the Hamming distance is 1 or 2 (S211). As a result of the determination, if it is determined that the hamming distance is 2 or more, after performing the process of setting the automatic setting mode to OFF (S212), the process returns to S201, and the process of extracting the next reference frame is performed.

  If the hamming distance is determined to be 1 as a result of the determination in S211, it is confirmed whether the automatic correction mode is ON or OFF (S213). If the automatic correction mode is OFF, the process returns to S201 to perform processing for extracting the next reference frame.

  When the automatic correction mode is ON, the additional information extraction unit 160 adds 1 bit to the 7-bit reference code KF obtained by the processing of FIG. 18 and outputs it (S208). Then, the bit counter is initialized to 0 (S209).

  As shown in the example of FIG. 19, in the case of setting for 1-bit error correction, even if a 1-bit error occurs, it is possible to output 1 word of additional information.

(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 first 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. Note that, 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.

(4. When the number of bits in the bit array Nh and the Hamming distance are changed)
In the above embodiment, when the bit array is created, the created bit array is 16 bits (Nh bits) and the Hamming distance is 6. However, the bit number Nh of the bit array and the Hamming distance are appropriately changed. Is possible. In general, if the bit number Nh of the bit array is increased, the coding efficiency is deteriorated, but the number of error bits that can be corrected is increased. Therefore, when using an extraction device with a relatively high microphone sensitivity and arithmetic processing capability, a bit array with a small number of bits and a hamming distance is created, and an extraction device with a relatively low microphone sensitivity and arithmetic processing capability is used. When used, a bit array having a large number of bits and a hamming distance is created.

  Here, an example in which the bit number Nh of the bit arrangement is 12 and the Hamming distance is 4 will be described. Even when the number of bits of the bit array is 12 and the Hamming distance is 4, the code conversion table creation by the conversion table creation device is basically performed according to the flowchart shown in FIG. However, HF is a 12-bit code, and after calculating the Hamming distance between the 12-bit code HF and another 12-bit code already registered in the code conversion table in S603, whether to proceed to S604 or S605 The threshold is determined by whether the Hamming distance is less than 4 or more than 4 and is different from the case of the number of bits Nh = 16 and the Hamming distance 6.

  The code conversion table created in this way is shown in FIG. As shown in FIG. 20, in the code conversion table, 12-bit codes are registered in association with values 0 to 127 that can be taken by the 7-bit code. As can be seen from the binary representation of the 12-bit code, all the 12-bit codes have a hamming distance of 4 or more. In the example of FIG. 20, for convenience of explanation, a decimal representation of a 7-bit code and a decimal representation and a binary representation of a 12-bit code are shown. Code bit strings are registered in association with each other.

  After the code conversion table creation process, the embedding process is performed according to the flowchart of FIG. At this time, processing for 12 bits is performed as one word.

  At the time of extraction, the conversion table creation means 180 first creates a code conversion table. After the code conversion table is created, additional information is extracted by the processing according to FIGS. In the case of handling a 12-bit code, in S205 in FIGS. 15 and 19, a determination is made based on whether the bit counter is 11 or less or 12 or more. Also, in S701 of FIG. 18, initialization is performed with the minimum hamming distance HD = 4, and in S703, HF is defined as a 12-bit code. Further, the process proceeds to S212 in FIG. 19 when the minimum hamming distance is 4 or more.

(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. The processing as shown in FIG. 5 is executed to update the values of Xl (i) and Xr (i).

[Formula 24]
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 [Equation 24] 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 extraction device for extracting information from the acoustic signal on the extraction side is the same as that in FIG. 14, and the processing operation is shown in the flowcharts of FIGS. It is the same as followed.

(6. When creating a sign reverse conversion table during extraction)
In the above embodiment, the extraction apparatus obtains the corresponding 7-bit reference code KF by executing the processing shown in FIG. 18 in a state where Nh (12 or 16) bits are stored in the buffer. However, in this case, since the processing shown in FIG. 18 is executed in units of one word, the processing load increases. In order to reduce such a processing load, it is desirable to create a code reverse conversion table for converting the obtained Nh bits into Nw bits. Here, a case where such a code reverse conversion table is used will be described.

  In this case, the conversion table creation means 180 of the extraction apparatus has a function of creating a sign reverse conversion table in addition to the function of the conversion table creation apparatus, and the additional information extraction means 160 performs the process according to FIG. Instead of the function, a function to refer to the code reverse conversion table is provided. Specifically, the additional information extraction unit 160 refers to the code inverse conversion table instead of executing the processing according to FIG. 18 in S206 of FIGS.

  In this case, the conversion table creation means 180 records the Hamming distance to the Hamming code to be converted for all Hamming codes of Nh (> Nw) bits, and for Hamming codes whose Hamming distance is a predetermined value or less. Thus, it has a function of creating a code reverse conversion table in which the registration order of the corresponding Nw bits is associated.

  When the extraction device is activated, first, the conversion table creation unit 180 creates a code reverse conversion table. The code reverse conversion table is created by the conversion table creating means 180 first by performing the same process as the conversion table creating apparatus to create the code conversion table, and then using this code conversion table.

  The code reverse conversion table records the minimum hamming distance to each hamming code registered in the code conversion table for all Nh-bit hamming codes, and for the hamming codes whose hamming distance is a predetermined value or less. Is associated with the registration order of Nw bits as the inverse transform destination (registration order in which the Hamming code having the Hamming distance is registered in the code conversion table), so when Nh = 12, HF = It is created for each Hamming code from 0 to 4095. The processing when the value of HF is specified is the same as that shown in FIG. In S704, when the value of i is 128 or more, and the hamming distance HD is 0 or 1, the 7-bit code KF and the minimum hamming distance HD are associated with the hamming code HF and the code inverse conversion table is used. Register with.

  By executing the processing shown in FIG. 18 for all of HF = 0 to 4095, a code reverse conversion table as shown in FIG. 21 is created. As shown in FIG. 21, in the code reverse conversion table, the values 0 to 4095 that can be taken by the 12-bit code HF are registered in association with the Hamming distance, and the Hamming distance is 1 or less. Corresponding 7-bit code KF is registered. In the example of FIG. 21, for convenience of explanation, a decimal representation of a 7-bit code and a decimal representation and a binary representation of a 12-bit code are shown. Code bit strings are registered in association with each other.

  Further, in the case of Nh = 16, by executing the process shown in FIG. 18 for all of HF = 0 to 65535, a code reverse conversion table as shown in FIG. 22 is created.

  Then, when performing the processing of S206 in FIG. 15, if Nh = 12, the corresponding 12-bit code KF and the Hamming distance are obtained by referring to FIG. 21 with the obtained 12 bits, and Nh When = 16, the corresponding 16-bit code KF and the Hamming distance are obtained with reference to FIG. 22 using the obtained 16 bits.

(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. Application to position detection)
The extraction apparatus according to the present invention can detect the position of the microphone by using extraction of additional information. For example, if the third additional information can be correctly extracted, it can be detected that the microphone is located at approximately the same distance from the L speaker and the R speaker. If the positions of the L speaker and the R speaker are specified, the absolute position of the microphone from which the third additional information has been correctly extracted can be specified.

(9. Other)
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-described embodiment, the case where the device for extracting information from the acoustic signal is realized by a single mobile terminal device such as a mobile phone has been described as an example. However, the device may be realized in cooperation with other computers. 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 portable terminal device includes an acoustic signal input unit 100, a reference frame acquisition unit 110, a phase change frame setting unit 120, an additional information extraction unit 160, and a conversion table creation unit 180. The frequency conversion unit 130, the code determination parameter calculation unit 140, the code output means 150, and the sound frame holding means 170 are provided in a dedicated computer 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.

  In the above embodiment, a case where a two-channel stereo sound signal is used has been described as an example. However, a one-channel monaural sound signal may be used. In this case, the process performed on either the L channel or the R channel may be executed.

  Further, the change of the predetermined frequency component by the frequency component changing means 30 is executed according to FIG. 5 and [Equation 11], but various changes of the frequency component according to the bit value to be embedded can be used. 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 explanatory drawing of the basic concept of this invention. It is a figure which shows the relationship of 3 types of additional information. 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 a 7 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 corresponding | compatible rule of 1st additional information, 2nd additional information, and 3rd additional information. 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. 1 is a functional block diagram of an apparatus for extracting information from an acoustic signal according to the present invention. It is a flowchart which shows the process outline | summary in the setting which does not perform error correction of the apparatus shown in FIG. 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 flowchart which shows the process for obtaining the corresponding 7-bit code KF and the minimum hamming distance HD when the 16-bit code HF is specified. 15 is a flowchart showing an outline of processing in a setting for performing 1-bit error correction in the apparatus shown in FIG. 14. It is a figure which shows an example of the code conversion table of 7 bit code and 12 bit code. It is a figure which shows an example of the code reverse conversion table of a 7 bit code and a 12 bit code. It is a figure which shows an example of the code reverse conversion table of a 7 bit code and a 16 bit code.

Explanation of symbols

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 ... additional 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 ... additional information extraction means 170 ... acoustic frame holding means 180 ... conversion table creation means

Claims (11)

Three types of stereo sound signals composed of time-series two-channel sample strings, which are different from each other at three extraction positions in the vicinity of the left and right speakers and when the stereo sound signals are reproduced by the left and right speakers. A device that embeds three different types of additional information in an inaudible state so that the additional information can be extracted,
A sound frame reading means for reading a predetermined number of samples for each channel from the stereo sound signal and creating sound frames L and R;
A frequency conversion is performed on the read two-channel sound frame L and sound frame R using a predetermined window function to obtain a window spectrum L for the sound frame L and a window spectrum R for the sound frame R, respectively. Frequency conversion means;
A plurality of spectrum sets in a predetermined frequency range are extracted from each of the generated window spectra, and between a plurality of spectrum sets corresponding to the window spectrum L based on the bit array 1 of the first additional information 1 to be embedded. The intensity value ratio is changed by a predetermined modification ratio L, and the modification ratio L is determined based on a difference from the bit array 3 of the third additional information 3 corresponding to the bit array 1, Based on the bit arrangement 2 of the second additional information 2 to be embedded, the ratio of intensity values between a plurality of spectrum sets corresponding to the window spectrum R is changed by a predetermined modification ratio R, and the modification ratio R is A frequency component changing unit configured to determine based on a difference from the bit array 3 corresponding to the bit array 2;
Frequency inverse transform means for performing frequency inverse transform on each window spectrum including the modified spectrum set to generate the modified acoustic frame L and the modified acoustic frame R;
Modified acoustic frame output means for sequentially outputting the generated modified acoustic frame L and modified acoustic frame R;
An information embedding device for an acoustic signal, comprising: In claim 1,
The frequency conversion means sets the odd-numbered and even-numbered acoustic frames L and R as the A type and the other as the B type, and the A type acoustic frame L or the acoustic frame R. Frequency conversion using the first window function, frequency conversion using the second window function for the B-type acoustic frame or acoustic frame R, and a spectrum corresponding to the first window function. Obtaining a first window spectrum, a second window spectrum which is a spectrum corresponding to the second window function;
The frequency component changing means extracts two spectrum sets that do not overlap each other in a predetermined frequency range from the generated window spectrum so that the frequency width of the higher frequency spectrum set is larger, The spectrum set is equally divided into ½ frequency widths, and the four spectrum sets for each window spectrum are extracted from the first window spectrum, and the spectrum sets are designated as 1D1, 1D2, 1U1, 1U2 in ascending order of frequency. When the spectrum set extracted from the second window spectrum is 2D1, 2D2, 2U1, 2U2 in order of increasing frequency, the intensity value of 1D1 and the intensity of 2D2 based on the bit array 1 or the bit array 2 to be embedded One of the product of the value and the product of the intensity value of 1D2 and the intensity value of 2D1 is the modification ratio from the other. Alternatively, the modification ratio R is greater than or equal to, and at the same time, one of the product of the intensity value of 1U1 and the intensity value of 2U2 and the product of the intensity value of 1U2 and the intensity value of 2U1 is greater than the modification ratio L or the An apparatus for embedding information in an acoustic signal, wherein the intensity of each spectrum set is changed so as to be larger than the modification ratio R. In claim 1 or claim 2,
The frequency component changing means sets any one of two levels Vh and Vl of the modification ratio L and the modification ratio R, and when the bit array 3 corresponding to the bit array 1 is different from each other, When Vl is set as the modification ratio L and the bit arrays 3 corresponding to the bit array 1 are the same bits, Vh is set as the modification ratio L and the bit arrays 3 corresponding to the bit array 2 are different from each other. In the case of bits, Vl is set as the modification ratio R, and Vh is set as the modification ratio R when the bit arrays 3 corresponding to the bit array 2 are the same bit. An information embedding device for acoustic signals. In claim 2,
The frequency component changing means, based on the state of 1 bit in the bit array 1 or the bit array 2, a product of an intensity value of 1D1, an intensity value of 1U1, an intensity value of 2D2, and an intensity value of 2U2; Each spectrum set so that any one of the product of the 1D2 intensity value, the 1U2 intensity value, the 2D1 intensity value, and the 2U1 intensity value is greater than the predetermined modification ratio L or the modification ratio R over the other. A device for embedding information in an acoustic signal, characterized in that the intensity of the sound signal is changed. In any one of Claims 2-4,
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 a strength multiplied by a coefficient β smaller than the coefficient α. When Vh is set as the modification ratio L or modification ratio R, the coefficient α is increased by a predetermined ratio and the coefficient β is determined by a predetermined ratio. When Vl is set as the modification ratio L or modification ratio R, the coefficient α is decreased by a predetermined ratio and the coefficient β is increased by a predetermined ratio. Information embedding device for the acoustic signal to be. In claim 5,
The frequency component changing means divides the intensity product of the group that changes the intensity product to be relatively large by the product of the intensity of the group that changes the intensity product to be relatively small. When the value γ is smaller than 1, the coefficient α ′ obtained by dividing the coefficient α by the square root of γ and the coefficient β ′ obtained by multiplying the coefficient β by the square root of γ are used in place of the coefficients α and β. An information embedding device for an acoustic signal, characterized in that: In any one of Claims 1-6,
For each channel, the acoustic frame reading means uses the A type acoustic frame L and the B type acoustic frame L, and the A type acoustic frame R and the B type acoustic frame R for a predetermined number of samples. Are duplicated and read,
The modified acoustic frame output means outputs the generated modified acoustic frame L and modified acoustic frame R connected to the preceding modified acoustic frame L and modified acoustic frame R, respectively, for output. An information embedding device. In any one of Claims 1-7,
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 in any one of Claims 1-8. The stereo sound signal is reproduced by the left and right speakers from the stereo sound signal in which the three kinds of additional information are embedded in a state in which it cannot be heard in advance, and the three kinds of addition are respectively performed at three extraction positions near and in the center of the left and right speakers. An apparatus for extracting information,
Sound frame acquisition means for acquiring a sound frame composed of a predetermined number of samples by monophonically inputting the sound signal at a predetermined position in a space where the stereo sound signal is reproduced in stereo, digitizing a predetermined section;
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;
Additional information extraction means for converting the extracted bit values in units of words according to a predetermined rule and extracting the three types of additional information;
An apparatus for extracting information from an acoustic signal, comprising:   A program for causing a computer to function as an apparatus for extracting information from an acoustic signal according to claim 10.

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