JP4867765B2 - Information embedding device for 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, capable of extracting embedded information even near the center of both speakers, in case of stereophonic reproduction, while degradation of reproduction quality is prevented, even in case of monophonic reproduction. <P>SOLUTION: When additional information is embedded in the sound signal, sound frames of odd numbers from a predetermined period of the sound signal are obtained as A type, and sound frames of even numbers are obtained as B type. Intensity of spectrum sets of high frequency side and a low frequency side of a predetermined frequency range F1 to F2 of the sound frame is changed so as to be the same on right and left channels, and so as to be inverted on A and B types. <P>COPYRIGHT: (C)2009,JPO&amp;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 Literature). 1 and 2).

  In the inventions described in Patent Documents 1 and 2, data is embedded in a 2-channel stereo sound signal while moving corresponding signal components between the left and right channels, and at the time of extraction, the L side of two speakers being reproduced in stereo Since the microphones are placed close to each other and monaural input is used for extraction, when two speakers are separated by a wide hall etc., data cannot be physically extracted if they are close to the R-side speakers. In some cases, the L side and the R side are reversed, or there are places where it is difficult to determine which side is the L side, and the extraction operation may be confused.

  Since the L-side signal and the R-side signal are in a complementary relationship, the R-side signal has been modified based on the embedded data. However, in order to facilitate misalignment correction, etc. On the L side, a signal component in a predetermined region is cut compared to the R side, and the amount is added to the R side, so that the design is basically difficult to extract from the R side signal. In addition, when the operation mode in which only the L-side signal is reproduced in monaural is used, there is a problem that noise is conspicuous because the signal degradation is biased to the L side.

In order to solve such a problem, the present applicant has proposed a technique capable of extracting information from either sound of a two-channel stereo sound signal (patent). Reference 3).
JP 2006-235359 A JP 2006-323246 A Japanese Patent Application No. 2007-53324

  However, the invention described in Patent Document 3 has a problem in that it cannot be extracted near the center where sounds from both channels are mixed.

  Therefore, according to the present invention, when stereo playback is performed, it is possible to extract embedded information even in the vicinity of the center of both speakers, and prevent deterioration in playback quality even when monaural playback is performed. It is an object of the present invention to provide an information embedding device for an acoustic signal and an information extracting device for the acoustic signal.

  In order to solve the above-described problem, the present invention is an apparatus for embedding additional information in an inaudible state with respect to an acoustic signal composed of a time-series sample sequence, and a predetermined number of samples are embedded from the acoustic signal. The acoustic frame reading means to be read, and among the read acoustic frames, odd-numbered and even-numbered one is A type and the other is B type, and the first window function is used for frequency conversion for the A type acoustic frame. The frequency conversion is performed on the B type acoustic frame using a second window function, and the first window spectrum corresponding to the first window function and the spectrum corresponding to the second window function are used. Frequency conversion means for obtaining a certain second window spectrum, and extracting two spectrum sets that do not overlap each other in a predetermined frequency range from each of the generated window spectra Based on the bit arrangement to be embedded, the intensity of the spectrum set extracted from the first window spectrum and the second window spectrum is based on the bit arrangement to be embedded. Frequency component changing means for changing the intensity of each spectrum set so as to reverse in the second window spectrum, and frequency conversion is performed on each window spectrum including the changed spectrum set to generate a modified acoustic frame. There is provided a device for embedding information with respect to an acoustic signal, comprising frequency inverse transforming means for performing, and modified acoustic frame output means for sequentially outputting the generated modified acoustic frames.

  Further, the present invention is an apparatus for extracting the additional information from an acoustic signal in which additional information is embedded in an inaudible state in advance, and is configured by digitizing a predetermined section of the acoustic signal and including a predetermined number of samples. An acoustic frame acquisition means for acquiring an acoustic frame, and among the read acoustic frames, an odd-numbered and even-numbered one is an A type and the other is a B type, and a first window function for the A type acoustic frame The first window spectrum, which is a spectrum corresponding to the first window function, and the second window function, the frequency conversion is performed using the second window function for the B type acoustic frame. Frequency conversion means for obtaining a second window spectrum that is a spectrum corresponding to the Two spectrum sets are extracted, and among the spectrum sets extracted from the first window spectrum and the second window spectrum, a group on the low frequency side of the first window spectrum and a high frequency side of the second window spectrum, and a high frequency of the first window spectrum And the extraction means for calculating the spectral intensity of each group divided into a group on the low frequency side of the side and the second window spectrum, and extracting the embedded bit value based on the calculated spectral intensity, and the extraction There is provided an apparatus for extracting information from an acoustic signal having additional information extraction means for converting the bit value obtained in units of words according to a predetermined rule and extracting additional information.

  According to the present invention, when embedding additional information in an acoustic signal, a predetermined frequency range in each acoustic frame is divided into a high frequency side and a low frequency side, and the magnitude relationship between these intensities is an odd numbered acoustic frame and an even numbered acoustic frame. Since the intensity of the spectrum set is changed so that it is reversed at Can be extracted, and even when monaural reproduction is performed, deterioration of reproduction quality can be prevented.

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 the present invention, the principle of sound wave segregation, which is a human psychoacoustic characteristic, is used. The syllable segregation is an illusion that humans interpolate and hear sounds as if two tracks of high and low are flowing continuously against a pattern in which high and low sounds alternate in time series. It is.

  For example, as shown in FIG. 14 (a), three low sounds, low tone 1, low sound 3, low sound 5, and three high sounds, high sound 2, high sound 4, and high sound 6, are low sound 1, high sound 2, and low sound. Consider a case in which 3, 3 high sounds, 5 low sounds, and 6 high sounds are played in this order. Low pitch 1, low pitch 3, low pitch 5 and high pitch 2, high pitch 4, high pitch 6 are about 1 octave apart, and the low and high sounds are not played at the same time, but the time intervals are almost continuous. Shall. In this case, as shown in FIG. 14 (b), it sounds to humans that a portion where low and high sounds are not played is interpolated and played. That is, although the actual performance is a single melody, as shown in FIG. 14 (b), a human being can play a high tone 1 ', a high tone 3', a high tone 5 ', a low tone 2', a low tone 4 ', and a low tone 6'. Sounds like it is interpolated. For example, the high sound 3 ′ is heard as an average component of the high sound 2 and the high sound 4 so that the high sound 2 and the high sound 4 are continuously connected. Further, the high pitch 1 'and the low pitch 6' at the end also sound like components corresponding to the low tone 1 and the high tone 6 with components close to the adjacent high tone 2 and low tone 5, respectively. However, an electroacoustic apparatus such as a microphone acquires the sound shown in FIG. 14A as it is. The present invention utilizes such a property. Note that the interpolated sound is not necessarily interpolated to the same level as the sound played back and forth, but roughly speaking, about 50% of the sound played back and forth is interpolated. Sounds like

  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.

(2.1. Configuration of embedded device)
Next, an information embedding device for an acoustic signal according to the present invention will be described. FIG. 1 is a functional block diagram showing the configuration of an information embedding device for an acoustic signal according to the present invention. In FIG. 1, 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. The apparatus shown in FIG. 1 can deal with both a stereo sound signal and a monaural sound signal. Here, a case where processing is performed on a stereo sound signal will be described.

  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 with additional information, an additional information storage unit 62 that is configured as a bit array and stores additional information embedded in the stereo acoustic signal, and an additional information It has a modified acoustic signal storage unit 63 for storing the modified acoustic signal after information is embedded, and stores various information necessary for other processing. The bit array creation means 70 extracts the additional information from the additional information storage unit 62, adds a 1-bit parity bit to each word of the additional information, and further adds a 4-bit check code bit according to a predetermined rule. It has a function to create an added 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, since the ASCII code is adopted as the code format of the additional information, 7 bits are set as one word in the additional information, and the bit array generated by the bit array generating means 70 is a check code after adding a parity bit. The 12 bits subjected to the additional processing are defined as one word. Each component shown in FIG. 1 is actually realized by installing a dedicated program in 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. 1 will be described. The sound frame reading means 10 reads a predetermined number 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 of samples of one sound frame read by the sound 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 conversion unit 20 performs frequency conversion on the sound frame read by the sound frame reading unit 10 to obtain a spectrum of the sound 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 the present invention, not only to prevent the generation of pseudo-harmonic components but also to create a plurality of states for embedding information from one acoustic frame, a plurality of window functions are prepared, Then, Fourier transform is performed using each window function to obtain a plurality of spectra. In this embodiment, as the plurality of window functions, a first window function W (1, i) and a second window function W (2, i) as shown in FIGS. Easy to recognize on the extraction side. The first window function W (1, i) is for use with an A type acoustic frame, and has 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).

  In the present invention, sound frames are 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] to [Equation 2]. In FIG. 2, 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. 2A and 2B, the vertical axis indicates the amplitude value (level) of the signal. 2A and 2B, 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 / 4, W (1, i) = 0.5−0.5 cos (4πi / N)
When N / 4 <i ≦ 11N / 16, W (1, i) = 1.0
When 11N / 16 <i ≦ 13N / 16, W (1, i) = 0.5−0.5 cos (8π (i−9N / 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 ≦ 3N / 4, W (2, i) = 1.0
When i> 3N / 4, W (2, i) = 0.5−0.5 cos (4π (i−N / 2) / N)

  As is clear from FIG. 2 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 conversion means 20 performs Fourier transform on the B type sound frame, the left channel signal Xl (i) and the right channel signal Xr (i) (i = 0,..., N−1). The window function W (2, i) is used to perform processing according to the following [Equation 4], and the real part Al (2, j) and imaginary part Bl (2, j) of the conversion data corresponding to the left channel ), 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.

  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, the bit arrangement is read bit by bit, and 1-bit information is embedded in one acoustic frame. 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.

  In this embodiment, the component of the change target frequency band of the acoustic frame is 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. 3 shows a state of a predetermined frequency component of each channel 1 sound frame of A type and B type according to the present embodiment. In each acoustic frame shown in FIG. 3, the horizontal axis indicates the time direction, and the vertical axis indicates the frequency direction. A shaded portion indicates a portion where a frequency component exists, and the darker the shade, the stronger the component strength.

  In FIG. 3, the frequency region is divided into four in the frequency direction of the vertical axis, but the second and third regions from the top, that is, the frequency band from F1 to F2 is the frequency band to be changed. 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. 3 (a), for the change target frequency band of the L channel A type acoustic frame, the spectrum on the high frequency side is represented by L1U, the spectrum on the low frequency side is represented by L1D, and the A type acoustic frame of the R channel is represented. For the frequency band to be changed, the spectrum on the high frequency side is represented by R1U, and the spectrum on the low frequency side is represented by R1D. Also, as shown in FIG. 3B, for the change target frequency band of the L-channel B type acoustic frame, the spectrum on the high frequency side is represented by L2U, the spectrum on the low frequency side is represented by L2D, and the B channel type of the R channel is represented. For the frequency band to be changed of the acoustic frame, the spectrum on the high frequency side is represented by R2U, and the spectrum on the low frequency side is represented by R2D.

  In the present embodiment, when the code 1 is embedded, as shown in FIGS. 3C and 3E, the components of L1D, R1D, L2U, and R2U are changed to relatively strong states, and L1U, R1U, L2D, and R2D. The component of is changed to a relatively weak state. This state is referred to as “state 1”. When embedding code 2, as shown in FIGS. 3D and 3F, the components of L1U, R1U, L2D, and R2D are changed to a relatively strong state, and the components of L1D, R1D, L2U, and R2U are relatively Change to a weak state. This state is referred to as “state 2”.

  In the present embodiment, information is embedded by changing the frequency components of the A-type and B-type acoustic frames to two states as shown in FIGS. 3C, 3E, 3D, and 3F. ing. Since there are two states, this corresponds to an information amount of 1 bit.

  In the present embodiment, the change target frequency bands F1 to F2 are set to either “1.7 kHz to 3.4 kHz” or “3.4 kHz to 6.8 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. Therefore, the lower limit is set to 1.7 kHz, which is one octave lower than the upper limit of 3.4 kHz. Further, when a device other than a mobile phone is used as a receiving terminal, a frequency region higher than 3.4 kHz can be used as long as it is below the upper limit of audible frequency (22 kHz), but in a high sound region exceeding 10 kHz, compression / modulation is possible. The lower limit is set to 3.4 kHz, which is the upper limit of the telephone line bandwidth, and the upper limit is set to 6.8 kHz, which is one octave higher than 3.4 kHz. It was. Note that the values “1.7 kHz”, “3.4 kHz”, and “6.8 kHz” are representative values, and are not necessarily accurate values, and may be slightly deviated from them. In the present specification, “1.7 kHz to 3.4 kHz” is referred to as “telephone high frequency band”, and “3.4 kHz to 6.8 kHz” is referred to as “super telephone frequency band”. Further, since the upper limit of the telephone line band is around 3.4 kHz as described above, the above “telephone high frequency band” and “super telephone frequency band” are slightly lower than the upper limit of the telephone line band in the audible frequency range. This corresponds to a predetermined frequency band on the high sound side.

  In the example shown in FIG. 3, the change to the relatively strong state and the weak state has been described. However, 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 portion to be interpolated 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 set to 70% and the weak side is set to 30%, the sound of the portion to be interpolated is almost the same as the sound played by the original acoustic signal before 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 intensity ratio of the relatively strong spectrum set and the relatively weak spectrum set as 70% and 30%. In order to realize this, in this embodiment, in a specific process described later, a coefficient α = 0.7 for setting a strong state and a coefficient β = 0.3 for setting a weak state are set. However, 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 the processing according to the following [Equation 5] to obtain the intensity ratio γ of the spectrum set to be changed to the strong state with respect to the spectrum set to be changed to the weak state. calculate.

[Formula 5]
E1d = Σ _{j = m,..., M + G−1} {Al (1, j) ² + Bl (1, j) ² }
E2d = Σ _{j = m,..., M + G−1} {Al (2, j) ² + Bl (2, j) ² }
E1u = Σ _{j = m + G,..., M + 2G-1} {Al (1, j) ² + Bl (1, j) ² }
E2u = Σ _{j = m + G,..., M + 2G-1} {Al (2, j) ² + Bl (2, j) ² }
When the embedded data has a value 1, γ = (E1d + E2u) / (E1u + E2d)
When the embedded data is 2, γ = (E1u + E2d) / (E1d + E2u)

  In the above [Equation 5], m is the number of the lower limit component of the change target frequency band, and m + 2G is the number of the upper limit component of the change target frequency band. For example, when 1.7 kHz to 3.4 kHz is set as the change target frequency band, m = 160 and m + 2G = 320. Therefore, the width G of one frequency region is 80.

  Further, according to the value of the intensity ratio γ, the frequency component changing unit 30 corrects the coefficients α and β by executing the processing according to the following [Equation 6] to obtain the coefficients α ′ and β ′. obtain.

[Formula 6]
When γ <1.0, α ′ = α · γ ^−1/2 , β ′ = β · γ ^1/2
When γ ≧ 1.0, α ′ = α · γ ^1/2 , β ′ = β · γ ^−1/2

  If γ ≧ 1.0, it may be set not to perform correction. Further, when the information to be embedded is “value 1”, the frequency component changing unit 30 executes the processing according to the following [Equation 7] to change the frequency component state to “state 1”, that is, 3 (c) Change to the state shown in (e).

[Formula 7]
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

  When the information to be embedded is “value 2”, by executing the processing according to the following [Equation 8], the state of the frequency component is changed to “state 2”, that is, FIGS. Change to the state shown.

[Formula 8]
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) · 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 + 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) · 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

  When the coefficients α ′ and β ′ are obtained by executing the processing according to the above [Expression 4], the coefficients α ′ and β are replaced with the coefficients α and β in the above [Expression 7] and [Expression 8]. 'Is used.

  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 performs the real part Al ′ (1,1) of the left channel of the spectrum obtained by any of the above [Formula 7] and [Formula 8]. j), imaginary part Bl ′ (1, j), etc., real part Ar ′ (1, j), etc. of the right channel, imaginary part Br ′ (1, j), etc. The process according to this is performed and Xl '(i) and Xr' (i) are calculated. For frequency components that are not modified in the above [Equation 7] and [Equation 8], the original frequency component Al (1, j) or the like is used as Al ′ (1, j) or the like.

[Formula 9]
Xl' (i) = 1 / N · {Σ j Al' (1, j) · cos (2πij / N) -Σ j Bl' (1, j) · sin (2πij / N)} + Xlp (i + N / 2 )
Xr' (i) = 1 / N · {Σ j Ar' (1, j) · cos (2πij / N) -Σ j Br' (1, j) · sin (2πij / N)} + Xrp (i + N / 2 )

In the above [Formula 9], 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 9] are the data Xlp (i) of the modified acoustic frame modified immediately before, When Xrp (i) exists, the addition is performed in consideration of the overlap of N / 2 samples on the time axis. By the above [Equation 9], 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 the real part Al ′ (2, j) and imaginary part of the left channel of the spectrum obtained by any one of the above [Formula 7] and [Formula 8]. Using Bl ′ (2, j), the real part Ar ′ (2, j) of the right channel, and the imaginary part Br ′ (2, j), processing according to the following [Equation 10] is performed, and Xl ′ ( i) and Xr ′ (i) are calculated. For frequency components that are not modified in the above [Formula 7] and [Formula 8], in the following [Formula 10], Al ′ (2, j), Bl ′ (2, j), Ar ′ (2, The original values Al (2, j), Bl (2, j), Ar (2, j), and Br (2, j) are used as j) and Br ′ (2, j).

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

  According to the above [Equation 10], each sample Xl ′ (i) of the left channel and each sample Xr ′ (i) of the right channel of the B type modified sound 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. 1 will be described with reference to the flowchart of FIG. Each component which comprises the apparatus shown in FIG. 1 cooperates, and performs the process according to FIG. FIG. 4 corresponds to processing of one word of additional information. Although one word can be set to any number of bits, as described above, in this embodiment, it is set to substantially 7 bits of the ASCII code.

  In FIG. 4, first, the bit array creation means 70 performs a predetermined process on the additional information extracted from the additional information storage unit 62 to create a new bit array (S101). Specifically, first, the additional information storage unit 62 is extracted in units of 1 word (7 bits), and 5 bits of the check code are added thereto to obtain 12 bits.

  Specifically, as shown in FIG. 5A, a bit array is created by adding five check codes P1 to P5 to 7 bits D1 to D7 extracted from the additional information. Each check code P1 to P5 is defined as shown in FIG. Among these, the check code P1 is a parity code, and the check codes P2 to P5 are 4-bit Hamming codes. Specifically, the check code P1 is calculated as an exclusive OR of the bits D1 to D7. The check code P2 is calculated as an exclusive OR of the bits D1, D2, D3, and D7. The check code P3 is calculated as an exclusive OR of the bits D1, D4, D5, D7, and P1. The check code P4 is calculated as an exclusive OR of the bits D2, D4, D6, D7, and P1. The check code P5 is calculated as an exclusive OR of the bits D3, D5, D6, and P1. Accordingly, the bit array creation means 70 creates a 12-bit bit array as shown in FIG. 5A in S101 according to the definition shown in FIG. 5B.

  These 12 bits are 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 12-bit array.

  Next, the frequency component changing unit 30 performs a process of reading 1 bit from 12 bits held in the register (S102). Subsequently, the acoustic frame reading means 10 reads a predetermined number of samples as one A-type acoustic frame from each of the left and right channels of the stereo acoustic signal stored in the acoustic signal storage unit 61 (S103). Subsequently, the frequency converting means 20 and the frequency component changing means 30 perform a process of changing the state of the frequency component of the A type acoustic frame according to the read bit value (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 (1, i). Then, the frequency component changing unit 30 executes processing according to the above [Formula 7] and [Formula 8] according to the value 1 and the value 2 received from the bit array creating unit 70, and the state of the component of the frequency band to be changed Is changed to either “state 1” or “state 2”.

  Next, the frequency inverse transform means 40 performs a process of performing frequency inverse transform on the spectrum whose intensity of each spectrum set has been changed by the process of S104 to obtain a modified acoustic frame (S105). As a matter of course, this frequency inverse transform needs to correspond to the method executed by the frequency converter 20 in S104. 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 channel Al ′ (1, j), etc., the imaginary part Bl ′ (1, j), etc. of the left channel of the spectrum obtained by any of the above [Equation 7] and [Equation 8], the right channel. Using the real part Ar ′ (1, j), etc., the imaginary part Br ′ (1, j), etc., the processing according to the above [Equation 9] is performed, and Xl ′ (i), Xr ′ (i) calculate. The modified sound frame output means 50 sequentially outputs the obtained modified sound frames to an output file.

  Subsequently, the acoustic frame reading means 10 reads a predetermined number of samples as one B-type acoustic frame from each of the left and right channels of the stereo acoustic signal stored in the acoustic signal storage unit 61 (S103). Next, the frequency converting unit 20 and the frequency component changing unit 30 perform processing for changing the state of the frequency component of the B-type acoustic frame according to the read bit value (S107). 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 4] is performed using the window function W (2, i). Then, the frequency component changing unit 30 executes processing according to the above [Formula 7] and [Formula 8] according to the value 1 and the value 2 received from the bit array creating unit 70, and the state of the component of the frequency band to be changed Is changed to either “state 1” or “state 2”.

  Next, the frequency inverse transform means 40 performs the process of obtaining the modified sound frame by performing the frequency inverse transform on the spectrum in which the intensity of each spectrum set is changed by the process of S107 (S108). Specifically, the real channel Al ′ (2, j), etc., the imaginary part Bl ′ (2, j), etc. of the left channel of the spectrum obtained by any one of the above [Equation 7] and [Equation 8], the right channel. Using the real part Ar ′ (2, j), etc., the imaginary part Br ′ (2, j), etc., the processing according to the above [Equation 10] is performed, and Xl ′ (i), Xr ′ (i) calculate.

  In the flowchart of FIG. 4, the state of the A type sound frame is changed in S103 to S105, and the state of the B type sound frame is changed in S106 to S108.

  The modified sound frame output means 50 sequentially outputs the obtained modified sound frames to an output file. When the processing for one acoustic frame for each of the A type and B type is finished for each channel, 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, S106), the process ends. When the processing corresponding to each bit of the bit arrangement (12 bits) for one word read in S101 is completed, the process returns from S102 to S101, and the process of reading the next word of the additional information and creating the bit arrangement is performed. Will do. 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, the case has been described in which the additional information is 7 bits per word, the check code is added to 12 bits, and processing is performed for 1 word of additional information. However, as long as there is an agreement with the extraction side, the additional information Can be recorded in an arbitrary number of bits.

  Of the left channel of the modified acoustic signal obtained as described above, with respect to the portion where the additional information is embedded, the component of the change target frequency band has only two distributions of the state 1 and the state 2 described above. become. 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. Further, as is apparent from the processing of [Formula 7] and [Formula 8], the components of the frequency band to be changed are changed at the same rate in the left channel and the right channel. Therefore, even at a position equidistant from both speakers, the components in the frequency band to be changed are in a relationship to be amplified without being canceled out, and information can be easily extracted.

(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. 6 is a block diagram showing an embodiment of an apparatus for extracting information from an acoustic signal according to the present invention. In FIG. 6, 100 is an acoustic signal input unit, 110 is a reference frame acquisition unit, 120 is a phase change frame setting unit, 130 is a frequency conversion unit, 140 is a code determination parameter calculation unit, 150 is a code output unit, and 160 is additional information. Extraction means 170 is an acoustic frame holding means.

  The acoustic signal input unit 100 has a function of acquiring and inputting a flowing sound as a digital acoustic signal. In reality, it is realized by a microphone and an A / D converter. Any microphone can be used, regardless of whether it is monaural omnidirectional or stereo directional, as long as the component in the frequency band to be changed can be detected. Even if it is stereo-directional, only one channel needs to be used. In addition, when information is embedded by the apparatus shown in FIG. 1, the information is reproduced in stereo. However, the sound from either the left or right speaker may be input, or the sound from both speakers may be mixed and input. There is no restriction on the position of the microphone. Of course, when the signal is reproduced in monaural or when information is embedded in monaural unlike the above, the microphone may be directed to a single speaker to be reproduced. 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). 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 the reference frame and a predetermined sample at a time. The frequency conversion means 130 has the same function as the frequency conversion means 20 shown in FIG.

The code determination parameter calculation unit 140 extracts a spectrum set corresponding to a predetermined frequency range from the generated spectrum, and calculates intensity values E1d ′, E2d ′, E1u ′, and E2u ′ of each spectrum set as follows (Formula 12). Calculate based on In addition, the code determination parameters C1d, C2d, C1u, and C2u are calculated using the intensity values of the components corresponding to the respective elements j, and based on the magnitude relationship between the code determination parameters C1d, C2d, C1u, and C2u, It has the function to judge that it is in a state. 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 values E1d ′, E2d ′, E1u are set for a certain acoustic frame. When calculating ', E2u' and sign determination parameters C1d, C2d, C1u, C2u, it is necessary to remove the reverberation component of the immediately preceding acoustic frame. Each component corresponding to the B type acoustic frame one frame before the respective components Al (1, j, p) and Bl (1, j, p) of the A type acoustic frame is represented by Al ₋₁ (2, j, p), Bl ₋₁ (2, j, p), one frame corresponding to each component Al (2, j, p), Bl (2, j, p) of the B type acoustic frame The sound frame before the minute becomes the A-type sound frame before correction, and the intensities E (1, j, p) and E (2, j, p) from which the reverberation component is removed are expressed by the following [Formula 11 ] Is calculated.

[Formula 11]
E (1, j, p) = Al (1, j, p) ² + Bl (1, j, p) ² −q {Al ₋₁ (2, j, p) ² + Bl ₋₁ (2, j, p ² }
E (2, j, p) = Al (2, j, p) ² + Bl (2, j, p) ² −q {Al (1, j, p) ² + Bl (1, j, p) ² }

  Of the two formulas in [Formula 11], the above formula removes the reverberation component from the B-type acoustic frame in which the N / 2 samples in the first half overlap when focusing on a certain A-type acoustic frame. Is to do. The following equation is for removing a reverberation component from an A type acoustic frame in which the N / 2 samples in the first half overlap when focusing on a certain B type acoustic frame. In the above [Expression 11], q is a coefficient indicating the magnitude of the reverberation component. As a result of experiments, q = 0.06 when N = 4096, and q = 0 when N = 2048. .12, q = 0.24 when N = 1024, and q = 0.48 when N = 2048. P is a phase number and takes an integer value of 0 to 5. Since only one channel signal is used at the time of extraction, it is not necessary to distinguish between the L channel and the R channel. However, in [Formula 11], the same symbols Al and Bl as those of the L channel are used. Therefore, Ar and Br may be used instead of Al and Bl.

  Then, the intensity values E1d ′, E2d ′, E1u ′, and E2u ′ of each spectrum set are calculated using the calculated reverberation components E (1, j, p) and E (2, j, p) as follows: ] Based on the above.

[Formula 12]
E1d ′ = Σ _{j = m,..., M + G−1} E (1, j, p) · F (j−m) · (aj) ²
E2d ′ = Σ _{j = m,..., M + G−1} E (2, j, p) · F (j−m) · (aj) ²
E1u ′ = Σ _{j = m + G,..., M + 2G-1} E (1, j, p) .F (jm-G). (Aj) ²
E2u '= [Sigma] _{j = m + G, ..., m + 2G-1} E (2, j, p) .F (jm-G). (Aj) ²

In the above [Equation 12], (aj) ² and F (j−m) are functions used to correct the spectral elements corresponding to each j. Among these, (aj) ² is a correction function generally expressed by (af) ^b and shows a case where b = 2.0. Since j is a value proportional to the frequency f, in [Equation 12], j is used instead of the frequency f. Here, the relationship between the frequency of the acoustic signal and the energy distribution is shown in FIG. As shown in FIG. 7, the energy distribution of the acoustic signal tends to decrease in proportion to the square of the frequency f. For this reason, when it is determined whether or not information is embedded in the high frequency component, it is more difficult to make an erroneous determination if the signal intensity is corrected to be larger than that of the low frequency component. Therefore, in the present invention, the intensity values E1d ′, E2d ′, E1u ′, E2u ′ corrected by the correction function (af) ^b are obtained. Moreover, a is a real number constant of 1 or less, and in this embodiment, a = 0.178 · (512 / N). F (j−m) is a frequency direction window function defined by the following [Equation 13]. Although these functions act on each spectral element, in calculating the intensity of the spectrum set, the above [Equation 12] calculates the total sum of the intensities of each spectral element, so that it acts on the sum. Yes.

[Formula 13]
When 0 ≦ j <G / 8, F (j) = 0.5j ² / (G / 8) ²
When G / 8 ≦ j <G / 4, F (j) = 1.0−0.5 · (j−G / 4) ² / (G / 8) ²
When G / 4 ≦ j <3G / 4, F (j) = 1.0
When 3G / 4 ≦ j <7G / 8, F (j) = 1.0−0.5 · (j−3G / 4) ² / (G / 8) ²
When 7G / 8 ≦ j <G, F (j) = 0.5 (G−j) ² / (G / 8) ²

In the above [Equation 13], G represents the number of components of the spectrum set in the predetermined frequency band, and G = (M−m) / 2. Here, a graph of the frequency direction window function F (j) and the correction function (aj) ² is shown in FIG. In the above [Equation 12], the frequency direction window function F (j) and the correction function (aj) ² are multiplied by G spectrum elements each having j ranging from m to m + G−1 and m + G to m + 2G−1. Will be. A more detailed graph of the frequency direction window function F (j) is shown in FIG. As can be seen from a comparison between FIG. 9 and the above [Equation 13], the frequency direction window function F (j) is divided into five according to the value of j, particularly when j <G / 8 and j ≧ 7G /. When the value is 8, the value decreases. For this reason, when the frequency direction window function F (j) extracts a spectrum set by dividing a predetermined frequency band into a plurality of frequency directions, the spectrum set on the low frequency component side and the spectrum set on the high frequency component side are clearly defined. Help to distinguish.

  In addition, the sign determination parameter calculation unit 140 includes the number of elements satisfying E (1, j, p)> E (2, j, p) among G elements of j = m,..., M + G−1. Is calculated as a code determination parameter C1d (p), and G-C1d (p) is calculated as a code determination parameter C2d (p). In addition, the number of elements satisfying E (1, j, p)> E (2, j, p) among G elements of j = m + G,..., M + 2G−1 is the code determination parameter C1u (p). And G-C1u (p) is calculated as the code determination parameter C2u (p).

  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 has a function of converting the binary array output by the code output unit 150 according to a predetermined rule and extracting it as meaningful additional information. The acoustic frame holding means 170 is a buffer memory that can hold two consecutive reference frames. Each component shown in FIG. 6 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 even a portable terminal device such as a cellular phone is a type of computer having an arithmetic processing function.

(3.2. Processing operation of extraction device)
Next, the processing operation of the apparatus for extracting information from the acoustic signal shown in FIG. 6 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 initial value is set and the user wants to know the attribute information such as the song name of the music that is flowing, first, the extraction device Instruct startup. 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 acoustic frame read as the reference frame by the reference frame acquisition unit 110 needs to be the same as that set by the acoustic 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 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). This process is the same as the process in the frequency conversion means 20 shown in FIG. However, since only the left channel is used for extraction, the processing according to [Formula 3] is performed, and the imaginary part Bl (1) such as the real part Al (1, j) of the conversion data corresponding to the left channel is performed. , J) etc.

  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, the code determination parameter calculation unit 140 calculates the code determination parameters C1d (p), C2d (p), C1u (p), and C2u (p) as described above, and then determines the code determination parameters C1d (p). , C2d (p), C1u (p), and C2u (p), what is the state of the component of the frequency band to be changed, that is, what value is embedded as a 1-bit value? Processing to determine whether or not it has been performed is performed (S402). Specifically, first, the following determination processing is executed, and as a result, it is determined that the corresponding state is obtained, and the corresponding value is output.

When C1d (p) + C2u (p)> C2d (p) + C1u (p), it is determined that the state is “1”, and a value of 1 is output.
In cases other than the above, it is determined that the state is “state 2”, and the value 2 is output.

  Regarding the processing of S401 and S402, actually, after performing frequency conversion on one set of A type sound frame and B type sound frame, intensity E (1, j, p) and E (2, j, p) are calculated, and further, C1d (p), C2d (p), C1u (p), and C2u (p) are calculated.

  The code determination parameter calculation means 140 sets either value 1 or value 2 in the candidate code table B (p) according to the determination result for each acoustic frame (S403). Specifically, when the value is 1, B (p) = 0, and when the value is 2, B (p) = 1.

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

[Formula 14]
If it is determined that the status is "1" and the value 1 is output,
S (p) ← S (p) + E _C1 + E _C4 −E _C2 −E _C3
If it is determined that the status is "2" and the value 2 is output,
S (p) ← S (p) + E _C2 + E _C3 −E _C1 −E _C4

  Returning to the flowchart of FIG. 11, 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 (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 11 or less or 12 or more (S205). If the bit counter is 11 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 12 or more, a check code is calculated from the previous 7 bits in the bit array stored in the buffer (S206). Specifically, the calculation is performed according to the definition shown in FIG. As a result, five check codes are obtained. Subsequently, the calculated check code is compared with the last 5 bits in the bit array (S207). Specifically, the five check codes calculated in S206 and the subsequent 5 bits are collated in order. As a result, if all five pieces match, it is judged to be complete conformity, and in other cases it is judged to be nonconformity. If it is determined in S207 that it is nonconforming, the process returns to S201 to perform processing for extracting the next reference frame.

  If it is determined in S207 that it is completely compatible, the additional information extraction unit 160 adds 1 bit to the previous 7 bits and outputs the result (S208). In the case of perfect match in S207, there is a high possibility that the first bit in the buffer is the first bit of the word in the additional information. Therefore, by adding bit “0” to the 7 bits from the head, it is output as one word in the ASCII code. In the case of nonconformity, there is a high possibility that the previous 7 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 it is judged to be a perfect match, there is a high possibility that the part is a word delimiter, so if it is really a delimiter, then if it is extracted 12 bits at a time, all will be extracted accurately in units of words. Can do. On the other hand, even if it is determined to be a perfect match, it may be a coincidence and not actually 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 the previous 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 processing shown in FIG. 10 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 the attribute information such as the song title or artist in the sound signal as the character information, the user wants to know the song title or artist by listening to the music being played. 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 performing 1-bit error correction will be described with reference to the flowchart of FIG. 13 includes a portion that performs the same processing as in FIG. Therefore, portions that perform the same processing as in FIG. Also in the example of FIG. 13, first, initialization processing is performed (S200). In this initialization process, as in the example of FIG. 10, 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. 10, after extracting an acoustic frame composed of a predetermined number of samples as a reference frame (S201), after identifying the phase of each read acoustic frame, 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 11 or less or 12 or more (S205). If the bit counter is 11 or less, the process returns to S201 to perform processing for extracting the next reference frame.

  If the bit counter is 12 or more, a check code is calculated from the previous 7 bits in the bit array stored in the buffer (S206). Subsequently, the calculated check code is compared with the subsequent 5 bits (S207). If it is determined in S207 that the matching is complete, processing for setting the automatic correction mode to ON is performed immediately before the bit string output processing in S208 (S210).

  If it is determined in S207 that it is nonconforming, it is determined whether or not an error bit can be specified (S211). Specifically, when P1 is rejected and P2 and P3, P2 and P4, P2 and P5, P3 and P4, P3 and P5, and any two bits of P4 and P5 are rejected, D1, D2, It is specified as a one-bit error of D3, D4, D5, and D6. If P1 is unsuccessful, and any 3 bits of P2, P3, and P4, and P3, P4, and P5 are unsuccessful, they are identified as 1-bit errors of D7 and P1, respectively. If P1 fails and any one of P2 to P5 fails, each is identified as its own 1-bit error. If the error bit cannot be specified as a result of the determination, the process of setting the automatic setting mode to OFF is performed (S212), and the process returns to S201 to perform the process of extracting the next reference frame.

  If the error bit can be identified as a result of the determination in S211, it is determined 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.

  If the automatic correction mode is ON, 1 bit in the previous 7 bits is inverted and corrected (S214). At this time, the bit to be inverted is the error bit specified in S211. Subsequently, a bit “0” is added to the previous 7 bits of the in-buffer bits after inversion correction and output (S208).

  As shown in the example of FIG. 13, 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 top acoustic frame is originally a sample of sample numbers 1 to 4096, but six pieces composed of 4096 samples starting from sample numbers 1, 683, 1366, 2049, 2732, and 3413 Are processed, and a code corresponding to the optimum acoustic frame is output. As described with reference to the flowchart of FIG. 11, in the present embodiment, when the same phase continues a predetermined number of times, the processing is performed after that phase is determined.

(4. Method of reliably embedding information even if the signal component is small)
As described so far, in the present invention, the signal component of the frequency band to be changed is always changed regardless of the state of the original signal component, but the magnitude of the component is the signal of both channels. There is no greater than strength. For this reason, if the original signal component does not exist or is too small, the signal component necessary for extraction cannot be obtained, and information cannot be appropriately extracted, resulting in an extraction error. Therefore, hereinafter, a method for enabling signal embedding so that information can be appropriately extracted even when the original signal component is small will be described.

  Conventionally, in order to appropriately extract information when the original signal component is small, as shown in Patent Document 3, by giving a predetermined fixed value as the intensity of the component in the spectrum set, It was possible to embed information. However, this method has a problem that the reproduction quality is deteriorated. Therefore, in this embodiment, 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. 1 has a function of adding or subtracting the same value in units of a predetermined number of samples within a predetermined amplitude range after reading a predetermined number of samples as one acoustic frame. Yes. In this embodiment, with respect to the left channel signal Xl (i) and the right channel signal Xr (i) (i = 0,..., N−1), an amplitude range of −32 to +32 (Xl (i), Xr ( i) is a value between −32768 to +32767), and a uniform random number H is generated, and the same value H is used between K (K = 5 in the present embodiment) samples as follows: The processing as shown in FIG. 5 is executed to update the values of Xl (i) and Xr (i).

[Formula 15]
Xl (i) ← Xl (i) + H
Xr (i) ← Xr (i) -H

  That is, white noise is generated in one acoustic frame by executing the process according to the above [Formula 15] over N samples. The white noise generation process is performed immediately after the acoustic frame extraction process of S103 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. 6, and the processing operation is shown in the flowcharts of FIGS. It is the same as followed.

(5. 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 the acoustic signal input unit 100, the reference frame acquisition unit 110, the phase change frame setting unit 120, and the additional information extraction unit 160, and the frequency conversion unit 130, the code determination parameter calculation unit 140, and the code output unit 150. The sound frame holding means 170 is provided in a dedicated computer so that necessary information is communicated between the two. 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.

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 flowchart which shows the process outline | summary of the apparatus shown in FIG. It is a figure which shows the mode of preparation of a bit arrangement | sequence. 1 is a functional block diagram of an apparatus for extracting information from an acoustic signal according to the present invention. It is a figure which shows the relationship between the frequency of an acoustic signal, and energy distribution. 7 is a flowchart illustrating an outline of processing in a setting in which error correction is not performed in the apparatus illustrated in FIG. 6. It is a flowchart which shows the detail of the phase determination and code | symbol output of S202 of FIG. It is a flowchart which shows the detail of the code | symbol determination process of S302 of FIG. 7 is a flowchart showing an outline of processing in a setting for performing 1-bit error correction in the apparatus shown in FIG. 6. It is a figure which shows the frequency direction window function F (j) used in one Embodiment of this invention, and a correction function. It is a figure which shows the detail of the frequency direction window function F (j) used by one Embodiment of this invention. It is explanatory drawing of the principle of the sound pulse fractionation which is a human auditory psychological characteristic.

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 ... sound frame holding means

Claims (14)

A device for embedding additional information in an inaudible state with respect to an acoustic signal composed of a time-series sample sequence,
An acoustic frame reading means for reading a predetermined number of samples from the acoustic signal;
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 each of the generated window spectra, and the intensities of the spectrum sets extracted from the first window spectrum and the second window spectrum are based on the bit arrangement to be embedded, Frequency component changing means for changing the intensity of each spectrum set so that the magnitude relationship between the intensity of the spectrum set on the high frequency side and the spectrum set on the low frequency side is reversed in the first window spectrum and the second window spectrum;
Frequency inverse transform means for performing frequency inverse transform on each window spectrum including the modified spectrum set to generate a modified acoustic frame;
Modified acoustic frame output means for sequentially outputting the generated modified acoustic frames;
An information embedding device for an acoustic signal, comprising: In claim 1,
The acoustic signal is a stereo signal of two or more channels;
The acoustic signal is characterized in that the acoustic frame reading means, the frequency converting means, the frequency component changing means, the frequency inverse converting means, and the modified acoustic frame output means execute processing for each channel. An information embedding device. In claim 1,
The frequency component changing means is 1D, 2U, 1U when the high-frequency spectrum set extracted from the first window spectrum and the second window spectrum is 1U, 2U, and the low-frequency spectrum set is 1D, 2D. And 2D are groups, and the intensity of a group whose intensity is changed relatively large is multiplied by a coefficient α, and the intensity of the spectrum set of a group whose intensity is changed relatively small is a coefficient. A device for embedding information in an acoustic signal, characterized in that the intensity is multiplied by a coefficient β smaller than α. In claim 3,
The frequency component changing unit is configured to change the coefficient when the ratio of the sum of the intensities of the groups that change the intensity relatively large to the sum of the intensities of the groups that change the intensity relatively small is smaller than a predetermined value. A device for embedding information in an acoustic signal, wherein a coefficient α ′ having a larger α and a coefficient β ′ having a smaller coefficient β are used in place of the coefficients α and β. In any one of Claims 1-4,
A bit array creating means for creating the bit array by adding a check code for detecting an error in the bit string in the additional information;
The apparatus for embedding information in an acoustic signal, wherein the frequency component changing means changes the intensity of the spectrum set based on the bit arrangement. In any one of Claims 1-5,
The acoustic frame reading means reads the A type acoustic frame and the B type acoustic frame by overlapping a predetermined number of samples,
The apparatus for embedding information in an acoustic signal, wherein the modified acoustic frame output means outputs the generated modified acoustic frame by connecting it with a preceding modified acoustic frame. In any one of Claims 1-6,
The frequency component changing means sets the predetermined frequency range as 1.7 kHz or more and 3.4 kHz or less, or sets the predetermined frequency range as 3.4 kHz or more and 6.8 kHz or less. An information embedding device. In any one of Claims 1-7,
The acoustic frame reading means generates white noise with extremely low amplitude that is difficult to hear during reproduction with the same number of samples as the acoustic frame, and adds the white noise to the acoustic frame. An information embedding device.   The program for functioning a computer as an information embedding apparatus with respect to the acoustic signal in any one of Claims 1-8. An apparatus for extracting the additional information from an acoustic signal in which the additional information is embedded in a state incapable of being heard in advance,
An acoustic frame acquisition means for digitizing a predetermined section of the acoustic signal and acquiring an acoustic frame composed of a predetermined number of samples;
Of the read acoustic frames, odd-numbered and even-numbered ones are A type and the other is B type, and frequency conversion is performed on the A type acoustic frames using a first window function, and the B type Frequency conversion is performed on the acoustic frame using the second window function to obtain a first window spectrum that is a spectrum corresponding to the first window function and a second window spectrum that is a spectrum corresponding to the second window function. Frequency conversion means;
Two spectrum sets that do not overlap each other in a predetermined frequency range are extracted from each of the generated window spectra. Among the spectrum sets extracted from the first window spectrum and the second window spectrum, the low frequency side of the first window spectrum and Dividing the high frequency side group of the second window spectrum, the high frequency side of the first window spectrum and the low frequency side group of the second window spectrum to calculate the spectral intensity of each group, based on the calculated spectral intensity, An encoding means for extracting the embedded bit value;
Additional information extracting means for converting the extracted bit value in units of words according to a predetermined rule and extracting additional information;
An apparatus for extracting information from an acoustic signal, comprising: In claim 10,
The encoding means is characterized in that the spectral intensity of each group is calculated by subtracting a value obtained by multiplying a corresponding frequency component of the immediately preceding acoustic frame by a coefficient q having a value of less than 1. An information embedding device for an acoustic signal. In claim 10 or claim 11,
When the set of extracted bit values becomes a bit array having a predetermined number of bits, the additional information extracting means uses a predetermined number of bits from the back of the bit array as a check code, and checks the bit array. The code calculated based on the bits other than the code is compared with the check code, and when the determination result is in a predetermined state, some bits other than the check code in the bit array are inverted, and the bit An apparatus for extracting information from an acoustic signal, wherein the additional information is extracted by outputting all bits other than the check code in the array as additional information. In any one of Claims 10-12,
The acoustic frame acquisition means changes a phase by moving a reference frame and a predetermined sample from a reference frame acquisition means for acquiring an acoustic frame composed of a predetermined number of samples as a reference frame from the acoustic signal. It is constituted by phase change frame setting means for setting a plurality of set sound frames as phase change frames,
The encoding means is based on the code determination parameter calculation means for calculating a code determination parameter based on the extracted spectrum set, and on the code determination parameter calculated in a past in-phase acoustic frame having a different reference frame, It is determined that one of the reference frame and the plurality of phase change frames has an optimal phase, and a predetermined code is determined based on the code determination parameter determined for the acoustic frame having the optimal phase. An apparatus for extracting information from an acoustic signal, characterized by comprising code output means for outputting.   A program for causing a computer to function as an apparatus for extracting information from an acoustic signal according to any one of claims 10 to 13.

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