JP2006323246A - Embedding device for information into sound signal, extraction device of information from sound signal, and sound signal reproducing apparatus - Google Patents

Abstract

<P>PROBLEM TO BE SOLVED: To provide an embedding device for information into sound signal, capable of coping with either monaural or stereophonic sound signals and of exactly extracting the embedded additive information from the sound signal, an extraction device for the information from the sound signal, and a sound signal reproducing apparatus. <P>SOLUTION: A plurality of window functions are previously prepared and the sound signals are read in a sound frame unit of a prescribed magnitude. For each sound frame, frequency conversion is performed by using a plurality of the window functions. The relations of the low-frequency components of the front parts to the rear parts of the sound frames are changed, according to the binary of the bit trains to be embedded; and thereafter, these frequency bands are combined with the high-frequency components. In the case of the stereophonic sound signals, the sound signals are embedded into the L-channel signals but the processing to add the components deleted for the purpose of the embedding processing to the R-channel is performed to prevent the deterioration of the signals. <P>COPYRIGHT: (C)2007,JPO&INPIT

Description

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

  As a service to provide music attribute information that allows you to know the titles of music that has been played recently, you can query the broadcast station for the date and time of the broadcast music, and record music fragments that are being played on mobile phones. Services that collate with melodies registered in the database have been put into practical use (see, for example, Patent Documents 1 and 2).

  In the inventions described in Patent Documents 1 and 2, since the recorded music fragments are checked against the melodies registered in the database, the processing load increases as the number of songs registered in the database increases, and similar melodies are mistaken. The possibility of judging increases. Therefore, a method of embedding music attribute information such as a song name and artist information as an inaudible digital watermark in an acoustic signal has also been proposed (see, for example, Patent Documents 3 to 6).

In the methods described in Patent Documents 1 to 6, the amount of information that can be embedded is small, the sound quality is deteriorated to some extent, watermark information is lost by various signal processing, and watermark detection is difficult for analog copies. There is a problem. Therefore, the present applicant changes the ratio of the components of the two low frequency bands in the low frequency region of the acoustic signal according to the bit value of the attribute information, so that both the monaural and stereo acoustic signals are A method that can embed attribute information (additional information) has been proposed (see Patent Document 7).
JP 2002-259421 A Japanese Patent Laid-Open No. 2003-157087 JP-A-11-145840 JP-A-11-219172 Japanese Patent No. 3321767 JP 2003-99077 A Japanese Patent Application No. 2005-58824

  However, in the method described in Patent Document 7, the low frequency band used for embedding additional information is often not included in the general microphone sensitivity range, and the embedded information is correctly identified. There is a problem that can not be.

  In such a case, it is desirable to narrow the range of the low frequency band used for embedding the additional information and set the lower low frequency band to a position included in the general microphone sensitivity range. In this case, the two low frequency bands are close to each other, causing mutual interference in the process of embedding and extracting, and there is still a problem that the embedded information cannot be correctly identified.

  Therefore, the present invention is compatible with both monaural and stereo acoustic signals, and can embed additional information correctly, and can embed information in an acoustic signal, and extract information from the acoustic signal. It is an object to provide a device and an acoustic signal reproduction device.

  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. A spectrum corresponding to the first window function by performing frequency conversion on the acoustic frame using a first window function, a second window function, and a third window function, and an acoustic frame reading means for reading as an acoustic frame. Frequency conversion means for generating a first window spectrum, a second window spectrum that is a spectrum corresponding to the second window function, a third window spectrum that is a spectrum corresponding to the third window function, and each of the generated windows A spectrum set corresponding to a predetermined low frequency band is extracted from the spectrum, and the extracted 3 is extracted based on the value of the information array of the additional information to be embedded. Low frequency component changing means for changing the ratio of the spectral intensity of the first window function at the head and the third window function at the rear of the first set of window functions and deleting the component of the second window function at the center; Frequency inverse transform means for generating a modified acoustic frame by performing frequency inverse transform on each window spectrum including the changed spectrum intensity, and modified acoustic frame output means for sequentially outputting the generated modified acoustic frames An information embedding apparatus for an acoustic signal is provided.

  According to the present invention, the acoustic signal is divided into predetermined sections, and the ratio of the spectral intensity of the leading portion and the rear portion of the low frequency component in the predetermined section is changed based on the value of the information array of the additional information to be embedded. Therefore, it is possible to deal with both monaural and stereo sound signals, and it is possible to accurately extract the embedded additional information.

Hereinafter, embodiments of the present invention will be described in detail with reference to the drawings.
(1. Basic concept of the present invention)
First, the basic concept of the present invention will be described. As described above, in the patent document 7, the applicant of the present invention changes either the monaural or the stereo by changing the ratio of two low frequency band components in the low frequency region of the acoustic signal according to the bit value of the attribute information. The technique which can embed additional information also in the acoustic signal is proposed. As described above, the present invention is the same in that the low frequency component is changed according to the information to be embedded.

  However, in the method shown in Patent Document 7, the low frequency component is further divided into two frequency bands. Therefore, in the lower frequency band, it is out of the general microphone sensitivity range, and is acquired on the extraction side. You may not be able to. Therefore, in the present invention, the low frequency component is not divided into finer frequency bands, but instead is divided in the time direction to embed information.

  As will be described later, in both Patent Document 7 and the present invention, the acoustic signal is divided into acoustic frames composed of a predetermined number of samples, and the processing is performed. Accordingly, the information on the front (head) or rear of the acoustic frame is changed.

(2. Information embedding device for acoustic signals)
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 low 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. , 62 is an additional information storage unit, 63 is a modified acoustic signal storage unit, and 70 is an additional information reading 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 frame 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 low frequency component changing means 30 extracts three sets of spectrum sets corresponding to three predetermined frequency ranges from the generated frame spectrum, and based on the additional information extracted from the additional information storage unit 62, the spectrum of the low frequency intensity data It has a function to change the ratio (ratio) between sets. The frequency reverse conversion means 40 has a function of generating a modified acoustic frame by performing frequency reverse conversion on a plurality of frame spectra including the changed low frequency intensity data. 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 additional information reading means 70 has a function of extracting additional information from the additional information storage unit 62. 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. 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.

(3. Processing operation of the embedding device)
Next, the processing operation of the information embedding device for the acoustic signal shown in FIG. 1 will be described with reference to the flowchart of FIG. Here, a case where processing is performed on a stereo sound signal having two channels of L (left) and R (right) as sound signals will be described. FIG. 2 corresponds to processing of 1 byte of additional information. First, the additional information reading means 70 reads additional information from the additional information storage unit 62 in units of 1 byte (S101). Specifically, one byte is read into a register in a computer used as an information embedding device for an acoustic signal. Subsequently, the mode is set to the separation mode (S102). There are two types of modes: separation mode and bit mode. The delimiter mode indicates a mode for performing processing at a delimiter in units of 1 byte, and the bit mode indicates a mode for performing processing based on the value of each bit of 1 byte. When one byte is read from the additional information storage unit 62, the separator mode is always set immediately after that.

  Subsequently, the acoustic frame reading means 10 reads a predetermined number of samples as one acoustic frame from each of the left and right channels of the stereo acoustic signal stored in the acoustic signal storage unit 61 (S104). 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.

  Subsequently, the frequency conversion means 20 performs frequency conversion on each read sound frame to obtain a frame spectrum that is a spectrum of the sound frame (S105). Specifically, each acoustic frame is performed using three window functions. 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.

  Here, a case where general Fourier transform is performed will be described. 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, the delimiter part is discontinuous. become. 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.

  Here, FIG. 15 shows how the signal waveform changes when general Fourier transform is performed. In FIG. 15, the horizontal axis is the time axis (t), and FIGS. 15 (a) to 15 (c) all correspond to each other. In FIGS. 15A and 15C, the vertical axis indicates the amplitude value (level) of the signal. In FIG. 15B, the vertical axis indicates the value of the window function W (t). Note that W (t) = 0.5−0.5 · cos (2πt / T), and the maximum value of W (t) is 1.

  In the case of a general Fourier transform, a signal having a predetermined length as shown in FIG. 15A is multiplied by a window function W (t) as shown in FIG. The signal is converted into Then, the Fourier transform is executed with a signal having a waveform as shown in FIG.

  In the present invention, in order to create a plurality of states for embedding information from one acoustic frame, not to remove the discontinuity after Fourier transform, a plurality of window functions are prepared, Then, Fourier transform is performed using each window function to obtain a plurality of spectra. Here, FIG. 16 shows a signal waveform of one acoustic frame and a plurality of window functions W (1, i), W (2, i), and W (3, i). In FIG. 16, 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. 16A, 16E, 16F, and 16G, the vertical axis indicates the amplitude value (level) of the signal. 16B to 16D, the vertical axis indicates the values of the window functions W (1, i), W (2, i), W (3, i), and W (1, i), W The maximum values of (2, i) and W (3, i) are both 1.

  The window function W (1, i) is for extracting the front part of the acoustic frame, and takes a maximum value 1 at the position of a predetermined sample number i in the front part as shown in FIG. In the rear part, the minimum value is set to 0. Which sample number has the maximum value depends on the design of the window function W (1, i). By multiplying the window function W (1, i), the signal waveform of the acoustic frame as shown in FIG. 16A has a signal component remaining in the front part as shown in FIG. The component is deleted, and this becomes a Fourier transform target. Further, the window function W (2, i) is for extracting the central part of the acoustic frame. As shown in FIG. 16C, the window function W (2, i) has a maximum value at the position of the predetermined sample number i in the central part. 1 is set, and the front and rear portions are set to have a minimum value of 0. By multiplying the window function W (2, i), the signal waveform of the acoustic frame as shown in FIG. 16A has a signal component remaining at the center as shown in FIG. The rear signal component is deleted, and this is subjected to Fourier transform. The window function W (3, i) is for extracting the rear part of the acoustic frame. As shown in FIG. 16D, the window function W (3, i) takes a minimum value of 0 at the front part, and a predetermined sample at the rear part. The maximum value 1 is set at the position of the number i. By multiplying the window function W (3, i), the signal waveform of the acoustic frame as shown in FIG. 16 (a) is deleted from the front signal component as shown in FIG. 16 (g). The signal component remains and becomes a Fourier transform target. Thus, after extracting the front part, the central part, and the rear part, the Fourier transform is executed, so that spectra corresponding to the front part, the central part, and the rear part are obtained. In order to embed a bit value in one acoustic frame, the bit value is originally divided into two parts, a front part and a rear part. However, on the extraction side, it is not always possible to read a signal synchronously. Therefore, in order to clearly distinguish the front portion and the rear portion, the present invention also extracts the signal at the center portion.

  When performing Fourier transform in S105, specifically, three window functions W (1) for the left channel signal xl (i) and the right channel signal xr (i) (i = 0,..., N−1). , I), W (2, i), and W (3, i), the processing according to the following [Equation 1] is performed, and the real part Al (1, j) of the conversion data corresponding to the left channel , Al (2, j), Al (3, j), imaginary part Bl (1, j), Bl (2, j), Bl (3, j), real part Ar () of the conversion data corresponding to the right channel 1, j), Ar (2, j), Ar (3, j), imaginary part Br (1, j), Br (2, j), Br (3, j) are obtained. Note that the window functions W (1, i), W (2, i), and W (3, i) are functions whose values increase near the front (front), near the center, and near the rear of the acoustic frame, respectively. ing.

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

  In [Expression 1], 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−1 similarly to 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 1], a frame spectrum in which the signal component of each acoustic frame is expressed by a spectrum corresponding to the frequency is obtained. Subsequently, the low frequency component changing unit 30 extracts a spectrum set of three predetermined frequency ranges from the generated frame spectrum. Human hearing is known to be less sensitive to directionality for low frequency components up to about 200-300 Hz (Corona Corp., issued October 30, 1990, "Sound Engineering Course 1. Basic Acoustics"). Engineering, Acoustical Society of Japan ”p.247 (see FIGS. 9 and 26). Therefore, in this embodiment, the low frequency component is about 200 Hz or less. In the vicinity of a frequency of 200 Hz, j corresponds to 20, so the real part Al (1, j), Al (2, j), Al (3, j), and imaginary part Bl ( 1, j), Al (2, j), Al (3, j), real part Ar (1, j), Ar (2, j), Ar (3, j), imaginary part Br (1, j) , Br (2, j), Br (3, j), j ≦ 20 are extracted.

Subsequently, the low frequency component changing means 30 extracts the extracted real part Al (1, j) and the like (the same applies to other real parts and imaginary parts including (2, j) (3, j)), and the imaginary part. Among the real parts Ar (1, j), etc., the imaginary parts Br (1, j), etc., among the real parts Al (1, j), etc., the imaginary parts Bl (1, j), etc. ) And the like, the total value E ₁ and the total value E ₂ are calculated by the following [Equation 2].

[Formula 2]
E ₁ = Σ _{j = 1,..., M-3} {Al (1, j) ² + Bl (1, j) ² }
E ₂ = Σ _{j = 1,..., M-3} {Al (3, j) ² + Bl (3, j) ² }

E ₁ calculated by the above [Equation 2] indicates the sum of the component intensities of the spectrum set near the front of the acoustic frame, and E ₂ indicates the sum of the component intensities of the spectrum set near the rear of the acoustic frame. Subsequently, it is determined whether or not the combined values E ₁ and E ₂ are equal to or higher than the level lower limit value Lev. The level lower limit value Lev is set to 0.5 when the maximum amplitude value of the acoustic signals xl (i) and xr (i) is normalized to 1 and M = 20. This value of Lev = 0.5 is a level at which the resistance to analog conversion can be maintained empirically, and when there are few low frequency components, it will be reduced as appropriate. In this case, the detection accuracy also decreases due to analog conversion. Will do.

The reason why it is determined whether or not the combined values E ₁ and E ₂ are equal to or greater than the level lower limit value Lev is that if the signal strength is small, even if the signal is changed, the change cannot be detected on the extraction side. is there. In this embodiment, when the bit value that can take the first value (for example, “1”) and the second value (for example, “0”) is “1”, the bit value is “0” for the three components of the window. In the case of "," the window 1 component is embedded. Therefore, when the bit value to be embedded is “1”, the sum value E ₁ is less than the lower limit value Lev, and when the bit value to be embedded is “0”, the sum value E ₂ is less than the level lower limit value Lev. In order to process again from the first bit without recording according to the bit value of the additional information, the reading position is returned to the first bit, and the mode is set to the delimiter mode (S106). On the other hand, when the bit value to be embedded is “1” and the total value E ₁ is greater than or equal to the level lower limit value Lev, or when the bit value to be embedded is “0” and the total value E ₂ is greater than or equal to the level lower limit value Lev, the mode is determined. It will be.

  When the mode is the separation mode, the low-frequency component changing unit 30 performs a process for equalizing the window 1 component and the window 3 component (including the case where all are 0) in the left (L) channel signal (S108). . Specifically, according to the following [Equation 3], processing for setting both L side to 0 is executed. In this case, the window 1 component and the window 3 component of the right (R) channel signal are not necessarily equal.

[Formula 3]
For j = 1 to M,
Al ′ (1, j) = 0
Bl ′ (1, j) = 0
Al ′ (3, j) = 0
Bl ′ (3, j) = 0
In the case of stereo, the following corresponding to the right signal is also calculated: E (1, j) = {Al (1, j) ² + Bl (1, j) ² + Ar (1, j) ² + Br (1, j) ² } ^{1 / 2}
Ar ′ (1, j) = Ar (1, j) · E (1, j) / {Ar (1, j) ² + Br (1, j) ² } ^1/2
Br ′ (1, j) = Br (1, j) · E (1, j) / {Ar (1, j) ² + Br (1, j) ² } ^1/2
E (3, j) = {Al (3, j) ² + B1 (3, j) ² + Ar (3, j) ² + Br (3, j) ² } ^1/2
Ar ′ (3, j) = Ar (3, j) · E (3, j) / {Ar (3, j) ² + Br (3, j) ² } ^1/2
Br ′ (3, j) = Br (3, j) · E (3, j) / {Ar (3, j) ² + Br (1, j) ² } ^1/2

  By executing the processing according to the above [Equation 3], the low frequency component of the frame spectrum of the left channel is the same when both the window 1 component and the window 3 component are “0”. The pattern in which the window 1 component and the window 3 component are equal is information indicating the head position (separation) of the additional information. In the above [Equation 3], Al ′ (j) = Bl ′ (j) = 0 is set for both the window 1 component and the window 3 component, but the purpose is to make it possible to recognize the separation on the extraction side. Therefore, if the value is sufficiently small, it is not always necessary to set it to zero. Further, the window 1 component and the window 3 component do not necessarily have to be the same, and it is sufficient that the difference is small. In this sense, the term “equal” is used here.

  On the other hand, when the mode is the bit mode, the low frequency component changing unit 30 determines whether the window 1 component and the window 3 component of the left channel signal are in accordance with the bit value of the bit array of the additional information extracted from the additional information storage unit 62. A process of changing the ratio of the spectrum intensity to a state where the window 1 component is dominant or the window 3 component is dominant (S107). Here, “dominant” indicates that the spectral intensity in the spectrum set of one window component is larger than the spectral intensity in the spectrum set of the other window component. Therefore, in S107, the window 1 component is obtained by executing the processing according to any of the following [Equation 4] and [Equation 5] according to the bit value that can take the first value and the second value. The magnitude relationship between the spectral intensity of the window 3 and the spectral intensity of the window 3 component is changed, and processing is performed to change the window 1 component to dominant or the window 3 component to dominant. For example, when the first value is 1 and the second value is 0, when the bit value is 1, the process according to the following [Equation 4] is executed for the window 1 component.

[Formula 4]
For j = 1 to M, Al ′ (1, j) = 0
Bl ′ (1, j) = 0
In the case of stereo, the following corresponding to the right signal is also calculated: E (1, j) = {Al (1, j) ² + Bl (1, j) ² + Ar (1, j) ² + Br (1, j) ² } ^{1 / 2}
Ar ′ (1, j) = Ar (1, j) · E (1, j) / {Ar (1, j) ² + Br (1, j) ² } ^1/2
Br ′ (1, j) = Br (1, j) · E (1, j) / {Ar (1, j) ² + Br (1, j) ² } ^1/2

  In this case, the processing according to the following [Equation 5] is executed for the three components of the window.

[Formula 5]
For three components j = M−2, M−1 and M, Al ′ (3, j) = 0
Bl ′ (3, j) = 0
In the case of stereo, the following corresponding to the right signal is also calculated. E (3, j) = {Al (3, j) ² + Bl (3, j) ² + Ar (3, j) ² + Br (3, j) ² } ^{1 / 2}
Ar ′ (3, j) = Ar (3, j) · E (3, j) / {Ar (3, j) ² + Br (3, j) ² } ^1/2
Br ′ (3, j) = Br (3, j) · E (3, j) / {Ar (3, j) ² + Br (3, j) ² } ^1/2

  When the bit value is 0, the processing according to the following [Equation 6] is executed for the three window components.

[Formula 6]
For each component of j = 1 to M, Al ′ (3, j) = 0
Bl ′ (3, j) = 0
In the case of stereo, the following corresponding to the right signal is also calculated. E (3, j) = {Al (3, j) ² + Bl (3, j) ² + Ar (3, j) ² + Br (3, j) ² } ^{1 / 2}
Ar ′ (3, j) = Ar (3, j) · E (3, j) / {Ar (3, j) ² + Br (3, j) ² } ^1/2
Br ′ (3, j) = Br (3, j) · E (3, j) / {Ar (3, j) ² + Br (3, j) ² } ^1/2

  In this case, the processing according to the following [Equation 7] is executed for the window 1 component.

[Formula 7]
For three components j = M−2, M−1 and M, Al ′ (1, j) = 0
Bl ′ (1, j) = 0
In the case of stereo, the following corresponding to the right signal is also calculated: E (1, j) = {Al (1, j) ² + Bl (1, j) ² + Ar (1, j) ² + Br (1, j) ² } ^{1 / 2}
Ar ′ (1, j) = Ar (1, j) · E (1, j) / {Ar (1, j) ² + Br (1, j) ² } ^1/2
Br ′ (1, j) = Br (1, j) · E (1, j) / {Ar (1, j) ² + Br (1, j) ² } ^1/2

  As a result of performing the processing according to [Formula 6] and [Formula 7], a value of “0” is obtained at j = M−2, M−1, and M of one component of the window, but signals other than the predetermined value are obtained at other times. Ingredients will be present. Therefore, in this case, the ratio of the spectrum intensity is changed so that the window 1 component is dominant.

  By executing the processing according to any one of [Equation 4] and [Equation 5], or [Equation 6] and [Equation 7], according to each bit value of the bit array of the additional information, the left channel signal Thus, the pattern is changed so that the window 1 component is dominant or the window 3 component is dominant.

  In this case, there is always a portion where the signal component is “0” between the high frequency band and the low frequency band, thereby preventing the signal components of the high frequency band and the low frequency band from being mixed. Eventually, the low frequency component changing means 30 performs the processing based on [Equation 3] in the separation mode in S108, and in the bit mode, it changes to [Equation 4] [Equation 5] or [Equation 6] [Equation 7] The process based on this is performed in S107.

  In either case of S107 and S108, the low-frequency component changing unit 30 then deletes the two window components (S109). Specifically, the processing according to the following [Formula 8] is executed for the two components of the window.

[Formula 8]
For each component of j = 1 to M, Al ′ (2, j) = 0
Bl ′ (2, j) = 0
In the case of stereo, the following corresponding to the right signal is also calculated: E (2, j) = {Al (2, j) ² + Bl (2, j) ² + Ar (2, j) ² + Br (2, j) ² } ^{1 / 2}
Ar ′ (2, j) = Ar (2, j) · E (2, j) / {Ar (2, j) ² + Br (2, j) ² } ^1/2
Br ′ (2, j) = Br (2, j) · E (2, j) / {Ar (2, j) ² + Br (2, j) ² } ^1/2

  Next, the frequency inverse transform means 40 performs the process of obtaining the modified acoustic frame by performing the frequency inverse transform on the frame spectrum in which the ratio between the spectrum sets of the window components is changed by the processes of S107 to S109 (S110). As a matter of course, this frequency inverse transform needs to correspond to the method executed by the frequency transform unit 20 in S105. In the present embodiment, since the frequency transform unit 20 performs the inverse Fourier transform, the frequency inverse transform unit 40 performs the inverse Fourier transform. Specifically, the real part 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 3] to [Equation 8], the right Using the real part Ar ′ (1, j) of the channel, the imaginary part Br ′ (1, j), etc., processing according to the following [Equation 9] is performed, and xl ′ (i), xr ′ (i ) Is calculated. For frequency components not processed in the above [Formula 3] to [Formula 8], Al ′ (1, j) etc., Bl ′ (1, j) etc., Ar ′ (1, j) etc., Br As ′ (1, j) etc., Al (1, j) etc., Bl (1, j) etc., Ar (1, j) etc., Br (1, j) etc. are used.

[Formula 9]
xl' (i) = 1 / N · {Σ j Al' (1, j) · cos (2πij / N) -Σ j Bl' (1, j) · sin (2πij / N)} + 1 / N · { Σ _j Al ′ (2, j) · cos (2πij / N) −Σ _j Bl ′ (2, j) · sin (2πij / N)} + 1 / N · {Σ _j Al ′ (3, j) · cos (2πij / N) −Σ _j Bl ′ (3, 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)} + 1 / N · { _{Σ j Ar' (2, j)} · cos (2πij / N) -Σ j Br' (2, j) · sin (2πij / N)} + 1 / N · {Σ j Ar' (3, j) · cos _{(2πij / N) -Σ j Br'} (3, 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 overlapping of N / 2 samples on the time axis. With the above [Equation 9], each sample xl ′ (i) of the left channel and each sample xr ′ (i) of the right channel of the modified acoustic frame are obtained. The modified sound frame output means 50 sequentially outputs the obtained modified sound frames to the output file (S111). When the processing for one acoustic frame is completed in this way, after the mode is set to the bit mode (S112), the additional information reading means 70 reads the next bit in the bit array of the additional information (S103). 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 (S104), the process ends. When the processing corresponding to each bit of the 1-byte data read in S101 is completed, the process returns from S103 to S101, and the next byte of the additional information is read. When the processing is completed for all the bytes of the additional information, the processing returns to the first byte 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.

The change of the left channel signal by the above processing will be described with reference to FIG. In FIG. 3, the horizontal direction in the drawing is the time axis and is proportional to the number of samples. In addition, a large number of rectangles in the figure indicate the window 1 component and the window 3 component of the modified acoustic frame. The horizontal width of the rectangle indicating the window component indicates the number of samples, and the vertical width indicates the strength. However, in FIG. 3, neither the horizontal width nor the vertical width is accurately shown, and a strong signal is applied to the leading portion corresponding to the window 1 component. It indicates whether there is a component and whether there is a strong signal component in the rear portion corresponding to the window 3 component. FIG. 3A shows a good signal when there is no acoustic frame in which the total values E ₁ and E ₂ calculated by the above [Equation 2] are less than the level lower limit value Lev, that is, for embedding additional information. The case is shown. FIG. 3B shows a signal that is not good for embedding additional information when there is an acoustic frame in which the total values E ₁ and E ₂ calculated by the above [Equation 2] are less than the level lower limit value Lev. The case is shown.

For example, it is assumed that a 2-byte bit array in which the first byte is “11011100” and the second byte is “11000001” is embedded as additional information. First, at the head of each byte, the first low frequency band and the second low frequency band are set to be equal as information indicating a delimiter. This is obtained as a result of executing the processing in accordance with the above [Equation 3] in S108 by setting the separation mode in S102. Further, in the example of FIG. 3A, since there is no acoustic frame in which the sum values E ₁ and E ₂ are less than the level lower limit value Lev, 1 byte is continuously processed by S107. This indicates that the loop from S103 to S112 was repeated 8 times in succession, and during that time, it did not pass through S106 and S108 because it was less than the level lower limit value. As shown in FIG. 3, when the bit value of the additional information is 1, a low frequency component exists in the window 3 component, and when the bit value of the additional information is 0, the low frequency component exists in the window 1 component. . As can be seen from the above [Formula 4] to [Formula 7], in this case, the low frequency component of the other window component is zero.

In the example of FIG. 3B, there are acoustic frames in which E ₁ and E ₂ corresponding to the bit values to be embedded are less than the level lower limit value Lev. In this case, the above [Formula 3 is obtained via S106 and S108. ] As a result of executing the processing according to the above, the window 1 component and the window 3 component are set to be equal. In this case, since the reading position is returned to the first bit in S106, the same bit is read again. In the example of FIG. 3B, when “11011100” in the first byte is embedded, an acoustic frame less than the level lower limit value Lev appears at the time when the first two bits of “11” are processed, and the second time “11011”. ”Indicates that an audio frame less than the level lower limit value Lev appeared at the time of 5 bit processing, and finally 8 bit processing could be performed at the third time.

  However, when the processing according to FIG. 2 is performed, as shown in FIG. 3B, every time an acoustic frame having a low signal level of low frequency components appears, it is repeated from the beginning in units of 1 byte. Since it will be processed, efficiency is bad. Therefore, instead of FIG. 2, processing as shown in FIG. 4 may be performed.

  The processing shown in FIG. 4 differs greatly from the processing shown in FIG. 2 in that it has a continuous identification mode in addition to the separation mode and the bit mode, and has three modes, and an acoustic frame with a low signal level of low frequency components has appeared. Even in this case, the process is continued without returning to the top. Therefore, in the continuous identification mode, information for identifying whether the next bit is a new one starting from the beginning or continued because it is interrupted is recorded. In the process of FIG. 4, the additional information is processed in units of words instead of units of bytes. This is because, as shown in S106 of FIG. 2, there is no process for forcibly returning the reading position to the beginning, and it is easy to process in units other than bytes. The number of bits in one word can be set freely and can be set to 1 byte.

The processing in FIG. 4 has many parts in common with the processing in FIG. First, the additional information reading means 70 reads additional information from the additional information storage unit 62 in units of one word (S201). Thereafter, the mode is set to the separation mode (S202), and the acoustic frame reading means 10 reads the acoustic frames from the left and right channels of the stereo acoustic signal stored in the acoustic signal storage unit 61 (S204), and the frequency converting means 20 However, for each read acoustic frame, frequency conversion is performed according to [Equation 1] to obtain a frame spectrum (S205), and the low frequency component changing means 30 extracts the real part Al (1, j), etc. Using the imaginary part Bl (1, j), the real part Ar (1, j), etc., the imaginary part Br (1, j), etc., the combined values E ₁ and E ₂ are calculated by [Equation 2], The process is the same until it is determined whether or not the summed values E ₁ and E ₂ are equal to or higher than the level lower limit value Lev. However, in the process of FIG. 4, when the summed values E ₁ and E ₂ are less than the level lower limit value Lev, the mode is simply set to the separation mode (S206). On the other hand, when the combined values E ₁ and E ₂ are equal to or higher than the level lower limit value Lev, the mode is determined.

  When the mode is the separation mode, the low frequency component changing unit 30 performs processing for equalizing the L-side window 1 component and the window 3 component in accordance with the above [Equation 3] (S208). On the other hand, when the mode is the bit mode, the low frequency component changing means 30 according to the bit value of the bit array of the additional information extracted from the additional information storage unit 62 according to the above [Equation 4] to [Equation 7], Processing for changing the ratio between the spectral intensity window 1 component and the window 3 component is performed (S207). Further, in S207, unlike S107 in FIG. 2, processing is performed even in the continuous identification mode. In the continuation identification mode, when it is new, the distribution between the low frequency component window 1 component and the window 3 component is changed to a dominant state according to [Formula 4], and when it is continuation, [Formula 4] 5], the distribution of the low frequency component between the window 1 component and the window 3 component is changed to a state in which the window 1 component is dominant.

  In either case of S207 and S208, the low-frequency component changing unit 30 then deletes the two window components (S209). Specifically, the process according to the above [Formula 8] is executed for the three components of the window.

  Next, the frequency inverse transform unit 40 performs a process of performing frequency inverse transform on the frame spectrum in which the ratio between the spectrum sets is changed according to [Equation 9] to obtain a modified acoustic frame (S210), and the modified acoustic frame output unit 50 Sequentially outputs the obtained modified acoustic frames to an output file (S211). When the processing for one acoustic frame is completed in this way, the mode is determined (S212). If the mode is the separation mode, the mode is set to the continuous identification mode (S213), and then the acoustic frame reading means 10 A frame is read (S204). On the other hand, when the mode is the bit mode or the continuous identification mode, after setting the mode to the bit mode (S214), the low frequency component changing means 30 reads the next bit in the bit array of the additional information (S203). Then, all the sound frames are processed, and when there are no more sound frames to be read from the sound signal (S204), the process ends.

A state of the change of the left channel signal by the processing according to FIG. 4 will be described with reference to FIG. In FIG. 5, as in FIG. 3, the horizontal direction is the time axis, and a large number of rectangles in the figure indicate the window 1 component and the window 3 component of the modified acoustic frame. FIG. 5A shows a case where there is no acoustic frame in which the sum values E ₁ and E ₂ calculated by the above [Equation 2] are equal to or higher than the level lower limit value Lev, as in FIG. 3A. FIG. 5 (b) shows a case where there is an acoustic frame in which the total values E ₁ and E ₂ calculated by the above [Formula 2] are less than the level lower limit value Lev, as in FIG. 3 (b). Yes.

  For comparison with the processing of FIG. 2, the additional information to be embedded is also a 2-byte bit array in which the first byte is “11011100” and the second byte is “11000001”, as in FIG. As in FIG. 3, the window 1 component and the window 3 component are set in an equal state at the head of each byte as information indicating a break. This is obtained as a result of executing the processing according to the above [Equation 3] in S208 by setting the separation mode in S202. Subsequently, before the processing corresponding to each bit of the additional information is performed, information indicating whether the information is new or continued is recorded. According to the process shown in FIG. 2, when there is an acoustic frame that is less than the level lower limit value Lev, the bit to be processed is returned to the beginning of each byte and processed, so all are new, but as shown in FIG. In the processing, even if there is an acoustic frame that is less than the level lower limit value Lev, the bit processed at that time is valid and continues from there, so whether the bit is new or continued It is necessary to record the information. Therefore, after recording the information indicating the break, information indicating whether it is new or continued is recorded. Specifically, the mode determination is performed in the separation mode state (S212), the continuous identification mode is set (S213), and the acoustic frame is extracted without reading the bits of the additional information (S204). After the frequency conversion (S205), if it is new, the distribution according to [Equation 4] changes the distribution of the low frequency component between the window 1 component and the window 3 component to the dominant state of the window 3 component. (S207).

  After recording the information indicating whether it is new or continuation in this way, the mode determination is performed in the continuation identification mode (S212), so the bit mode is set (S214), and the first bit is read from the register ( (S203), an acoustic frame is extracted (S204). In the example of FIG. 5A, since there is no acoustic frame that is less than the level lower limit value Lev, one byte is continuously processed in S207. This indicates that the loop from S203 to S214 was repeated eight times in succession, and during that time, it was assumed that the level was lower than the lower limit value Lev, and that it did not pass through S206, S208, and S213.

  In the example of FIG. 5B, as a result of the processing according to the above [Equation 2], there is an acoustic frame that is less than the level lower limit value Lev. In this case, the above [Equation 3] passes through S206 and S208. As a result of executing the processing according to the above, the window 1 component and the window 3 component are set to be equal. In this case, since the separation mode is set in S206, information indicating whether it is new or continued is recorded via S212. In the example of FIG. 5B, when embedding “11011100” of the first byte, an acoustic frame less than the level lower limit value Lev appears when 1 bit of “1” of the first bit is initially processed. Therefore, after the information indicating the break is recorded, the information indicating the continuation is recorded, and the processing is continued from “1” of the second bit. Then, since “1011” from the second bit to the fifth bit has been processed, an acoustic frame less than the level lower limit value Lev has appeared, so after recording the information indicating the break, the information indicating the continuation is recorded. Then, processing is continued from “1” of the sixth bit.

  In the example of FIG. 5, the case where the additional information is recorded in units of bytes has been described for comparison with FIG. 3, but the process shown in FIG. 4 records information indicating whether it is new or continued. Additional information can be recorded in an arbitrary number of bits.

  In the above example, information indicating a delimiter is inserted in a fixed-length byte unit or a variable-length word unit, but it is also possible to insert information indicating a delimiter in bit units. In this case, when the acoustic frame reading means 10 extracts the acoustic frame, it extracts an overlapping acoustic frame that overlaps the preceding and following acoustic frames, performs frequency conversion on the overlapping acoustic frame according to [Equation 1], and In accordance with [Equation 3], processing for equalizing the window 1 component and the window 3 component is performed. The overlapping sound frames are set so that half of the samples overlap with the preceding and following sound frames. For example, when the preceding sound frame is sample number 1 to 4096 and the subsequent sound frame is sample number 4097 to 8192, the overlapping sound frame set between this is sample number 2049 to 6144. Similarly, for all sections of the acoustic signal, a duplicate acoustic frame is read and processing for equalizing the window 1 component and the window 3 component is performed.

  As described above, when the overlapping acoustic frame is set and the processing for equalizing the window 1 component and the window 3 component is performed, the window 1 component and the window 3 component are equalized in order to reflect this in the modified acoustic signal. It is necessary to perform a process of performing inverse frequency conversion on the duplicated frame spectrum after processing to obtain a modified duplicated acoustic frame and further connecting to the acoustic frame.

  Of the left channel of the modified acoustic signal obtained as described above, for the portion where the additional information is embedded, the low frequency component is equal to the window 1 component and the window 3 component, or the window 1 There are only three distributions: the component predominates or the window three component predominates. However, since the high frequency component remains the original acoustic signal, it has various distributions based on the setting of the producer. Further, as shown in the above example, when a stereo sound signal is used, the low frequency component changed in the left channel is apparent from the processing of [Expression 3] to [Expression 8]. In addition, it is always added to the low frequency component of the right channel. Therefore, since the right channel supplements the deleted component in the left channel, there is no signal degradation when viewed as both channels as a whole. Human auditory senses directionality with respect to high-frequency components, but it is difficult to sense directionality with respect to low-frequency components. Therefore, even if the low frequency component is biased to one side, it will be heard as if it is a normal acoustic signal for the listener.

(4. Device for extracting information from acoustic signals)
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, as long as it can detect a low-frequency component, whether it is monaural omnidirectional or stereo directional. Even if it is stereo-directional, only one channel needs to be used. Further, when information is embedded by the apparatus shown in FIG. 1, the information can be extracted even if a microphone with a general accuracy is used instead of a high 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. Sign determination parameter calculating means 140 extracts the respective low frequency intensity data corresponding the generated frame spectrum below a predetermined frequency, a window 1 component, sum E ₁ of each low-frequency intensity data for each window 3 components, E ₂ is calculated, and the combined values E ₁ and E ₂ are used as code determination parameters. Based on the ratio of the code determination parameters E ₁ and E ₂ , a function of determining a predetermined state is provided.

  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 additional information extraction unit 160 has a function of converting the ternary array, which is a set of codes 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 achieve the object of the present invention more easily, it is desirable to use a portable terminal device as hardware.

(5. 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 with reference to the flowchart of FIG. First, in this apparatus, the average code levels HL1 and HL2 and the phase determination table are initialized. These will be described. The average code levels HL1 and HL2 are the low frequencies calculated by the above [Equation 2] for an acoustic frame (hereinafter referred to as an effective frame) that is determined to have embedded binary values corresponding to bit values. sum E _1, the average value of E ₂ components, namely, which is given by the average value of the sum E _1, E ₂ in the past valid frame, the initial value is, the level limit is also used in the above embedding device The value Lev is set. 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.

  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 (S301). Specifically, the reference frame is extracted and read into the acoustic frame holding unit 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 as reference frames. The acoustic frame holding means 170 can store two reference frames as described above, and when a new reference frame is read, the old reference frame is discarded. Therefore, the sound frame holding means 170 always stores two reference frames (continuous 8192 samples).

  The acoustic frame processed by the embedding device can be divided into a reference frame that is set adjacently without interruption from the head, and a phase change frame in which the phase is changed. For the reference frame, after setting the first reference frame from sample number 1 to sample number 4096, the next reference frame is sample number 4097 to sample number 8192, and the next reference frame is sample number 8193 to sample number 12288. It is set without interruption. 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 determine embedded information from each read sound frame and output a corresponding code (S302). The format of the information to be output is a binary format corresponding to the bit value on the embedding side and a ternary format of values input as delimiters.

  Details of the code determination process in step S302 will be described with reference to the flowchart of FIG. First, the frequency conversion means 130 performs frequency conversion on each read sound frame to obtain a frame 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 the above [Equation 1] 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. , 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 average code levels HL1 and HL2 (S402). Specifically, v1 past window 1 component is an integrated value of the sum E ₁ for acoustic frame is determined dominant state, the number of sound frames past the window 1 component is determined to dominant state by dividing in a certain n1 calculates HL1, the v2 past window 3 component is an integrated value of the sum E ₂ for acoustic frame is determined dominant state, it determines the last window 3 component dominant state HL2 is calculated by dividing by n2, which is the number of sound frames that have been performed. Therefore, the average code levels HL1 and HL2 are average values of the sum values of the low-frequency intensity data of the acoustic frames that have been determined to have a dominant window component corresponding to the past.

Furthermore, the code determination parameter calculation unit 140 extracts each low frequency intensity data in a predetermined frequency range from the generated frame spectrum. The frequency range to be extracted needs to correspond to the embedding device. Therefore, here, low frequency intensity data having a frequency of about 200 Hz or less is extracted, and the real part Al (j) and imaginary part of the left channel calculated by the above [Equation 1] as in the case of the embedding device. Of Bl (j), those with j ≦ 20 are extracted. The code determination parameter calculating means 140, by executing the processing according to the above [Equation 2], the total value E ₁ of the window 1 component, calculates a total value E ₂ of the window 3 components. In the extraction device, this is used as a code determination parameter.

  Subsequently, the code determination parameter calculation unit 140 initializes the candidate code table (S403). The candidate code table records a phase number of 0 to 5 that specifies one reference frame and five phase change frames, and a ternary code obtained from the states of the six acoustic frames.

Subsequently, the code determination parameter calculation unit 140 determines whether or not the sum value E _{1 of} the window ₁ component and the sum value E ₂ of the window 3 component are each equal to or less than a predetermined value (S404). Specifically, 1/20 of the average code levels HL1 and HL2 are set as predetermined values, respectively. And the sum E ₁ is less than 1/20 of the average code level HL1, and if the total value E ₂ is less than one twentieth of the average code level HL2, sign determination parameter calculating means 140, with delimiter information It is determined that there is (S408).

On the other hand, when either one of the total values E ₁ and E ₂ is larger than the predetermined value, the code determination parameter calculation means 140 performs the comparison determination of the calculated code determination parameters E ₁ and E ₂ as follows: ] (S405), and outputs a code corresponding to the comparison result.

[Formula 10]
When E ₂ / E ₁ > 2, the window 3 component is dominant. When E ₁ / E ₂ > 2, the window 1 component is dominant. E ₂ / E ₁ ≦ 2 and E ₁ / E ₂ ≦ 2 If both window components are equal

The code determination parameter calculation unit 140 outputs a ternary code according to the determination result for each acoustic frame. That is, when it is determined that the window 3 component is dominant, the first bit value (for example, “1”) is output (S406). When it is determined that the window 1 component is dominant, the second bit value is output. The bit value (eg, “0”) is output (S407), and if it is determined that the two window components are equal, a code indicating delimiter information is output (S408). In S405, if it is determined that the window 3 component is in a dominant state, E ₁ is HL1 or higher. If it is determined that the window 1 component is in a dominant state, is E ₂ HL2 or higher? If these conditions are not satisfied, a code indicating delimiter information is output (S408).

  When it is determined that the window 3 component is dominant and the first bit value is output (S406), or when it is determined that the window 1 component is dominant and the second bit value is output (S407). Further, the phase determination table S (p) is updated according to the following [Equation 11] (S409).

[Formula 11]
When the three components of the window are dominant, S (p) ← S (p) + E ₁ / E ₂
When the window 1 component is dominant, S (p) ← S (p) + E ₂ / E ₁

Subsequently, the code determination parameter calculation unit 140 saves the candidate for the optimum phase in the candidate code table (S410). Specifically, the value of the phase number p that maximizes the value of S (p) recorded in the phase determination table, one of the three values determined in S406 to S408, and the sound frame The values of E ₁ and E ₂ corresponding to the low frequency components calculated by executing the processing according to the above [Equation 2] are stored in the candidate code table as candidates for the optimum phase.

  Subsequently, it is determined whether or not the processing corresponding to all the phase numbers p has been completed (S411). This determines whether all phase change frames have been processed for a certain reference frame. In the present 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 that has been 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 p recorded in the candidate storage table is the optimum phase, and the code recorded in the candidate storage table is output. (S412).

Returning to the flowchart of FIG. If the code corresponding to the bit value is output as a result of the processing in S302, the parameter of the average code level is updated (S303). Specifically, the integrated values v ₁ and v ₂ are added to the integrated values v ₁ and v 2 that are the numerators for calculating the average code levels HL 1 and HL 2, respectively, and the integrated values v ₁ and v 2 are updated, and the number of frames serving as the denominator. The number of frames n1 and n2 is updated by adding 1 to n1 and n2, respectively. Subsequently, the bit value corresponding to the output code is stored in the buffer (S304). Subsequently, the bit counter is counted up (S305). Then, it is determined whether the bit counter is 8 bits or more (S306). As a result, when the bit counter is 8 bits or more, since the bit value for 1 byte is stored in the buffer, the additional information extraction unit 160 outputs the data for 1 byte in the buffer. (S307).

  Then, the bit counter is initialized to 0 (S308). By executing the processing shown in FIG. 7 for each acoustic frame, additional information is extracted. If it is determined in S301 that all sound frames have been extracted, the process ends.

  In the processing of S307, the additional information extraction unit 160 first delimits a code indicating that the window 3 component and the window 1 component are equal among the ternary codes output by the code determination parameter calculation unit 140. Then, the next code is the head, and a bit array is created by associating a code indicating that the window 3 component is dominant and the window 1 component is dominant with the bit value. Subsequently, this bit arrangement is converted according to a predetermined rule and extracted as meaningful additional information. As the predetermined rule, various rules can be applied as long as the information intended by the person who embeds the information can be recognized by the person who has received it. As a rule. In other words, the additional information extraction unit 160 recognizes the code output from the code output unit 150 as determined by the code determination parameter calculation unit 140 in units of 1 byte (8 bits), and character information according to the set code system. Recognize 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 processing, in order to accurately extract additional information by the extraction device, the processing for correcting the phase, the processing for correcting the balance of the intensity of the window 3 component and the window 1 component, and the invalid frame are determined. The process of correcting the lower limit threshold is performed. Next, supplementary explanation will be given for these three correction processes.

(6. 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. This phase correction process is performed centering on the processes in S403, S409, S410, S411, and S412.

(7. Lower limit correction process)
When the signal level is low, the magnitude of the intensity of the window component cannot be determined, and erroneous determination is often made on the extraction side. Therefore, it is determined that frames whose sum values E ₁ and E ₂ are equal to or less than a predetermined threshold are invalid frames. The threshold at this time is determined as an integrated value of low frequency intensities for past effective frames. Correction processing is performed using. By varying the threshold value in this way, it is possible to accurately determine whether the frame is invalid or valid even if the signal level varies. This lower threshold correction process is performed centering on the processes in S402 and S303.

  The flowchart in FIG. 7 corresponds to the additional information recorded on the embedding side in byte units. When additional information is recorded in units of words on the embedding side, processing according to the flowchart of FIG. 9 is performed. First, as in S301 of FIG. 7, the reference frame acquisition unit 110 reads a predetermined number of samples as one reference frame from each channel of the stereo audio signal input from the audio signal input unit 100 (S501).

  Subsequently, as in S301 of FIG. 7, the frequency conversion unit 130 and the code determination parameter calculation unit 140 specify the acoustic frame having the optimum phase from the reference frame or the phase change frame using each read reference frame, A code corresponding to the information embedded in the sound frame is output (S502). The details of the processing in S502 are as shown in FIG. 8 as in S302.

  If a code corresponding to the bit value is extracted as a result of the processing in S502, the parameter of either the average code level HL1 or HL2 is updated as in S303 (S503). Subsequently, the mode is determined (S504). Two modes, a delimited mode and a bit output mode, are prepared. If it is in the bit output mode, the bit value is stored in the buffer (S509). Subsequently, the bit counter is counted up (S510). On the other hand, if the result of determination in S504 is that the mode is the separation mode, it is further determined whether the extracted code means new or continuation (S505). As a result, if it is new, it means that one word has been completed immediately before, and the additional information extracting means 160 outputs the data for one word recorded in the buffer (S506). Then, the bit counter is initialized to 0 (S507). Further, the mode is set to the bit output mode (S508). If it is determined in S505 that the operation is to be continued, the value should be output to the bit in the buffer, so only the processing for setting the bit output mode is performed. If a code corresponding to the delimiter information is extracted in S502, the mode is set to the delimiter mode in order to extract new or continued information from the next sound frame (S511). By executing the processing shown in FIG. 9 for each reference frame, additional information is extracted. If it is determined in S501 that all the reference frames have been extracted, the process ends.

(8. Sound signal reproduction device)
Next, the acoustic signal reproducing apparatus according to the present invention will be described. FIG. 10 is a block diagram showing an embodiment of an acoustic signal reproduction device according to the present invention. In FIG. 10, reference numeral 200 denotes an acoustic frame reading unit, 210 denotes an additional information extraction / display unit, 240 denotes a reproduction frame input unit, 250 denotes a reproduction frame storage unit, 260 denotes a sound device driver, 261 denotes a sound device, and 262 denotes a timer. The sound signal reproducing apparatus according to the present invention can reproduce either a monaural sound signal or a stereo sound signal, but the apparatus shown in FIG. 10 will be described as reproducing a stereo sound signal.

  Similar to the acoustic frame reading means 10 shown in FIG. 1, the acoustic frame reading means 200 has a function to frame and read a predetermined number of samples from each channel of the stereo acoustic signal. Each sound frame read by the sound frame reading means 200 is processed in two systems of the additional information extraction / display means 210 and the reproduction frame input means 240. More specifically, the additional information extraction / display unit 210 includes the phase change frame setting unit 120, the frequency conversion unit 130, the code determination parameter calculation unit 140, the code output unit 150, and the additional information extraction unit of the extraction apparatus illustrated in FIG. The means 160, the sound frame holding means 170, and the means for displaying and outputting the additional information extracted by the additional information extracting means 160 are configured.

  The reproduction frame input unit 240 has a function of inputting each sound frame read by the sound frame reading unit 200 to the reproduction frame storage unit 250. However, the reproduction frame input unit 240 not only simply inputs an audio frame, but also has a function of controlling the input of an audio frame when the reproduction frame storage unit 250 is not empty, as will be described later. . The reproduction frame accumulating means 250 has a plurality of buffer memories for accumulating the sound frames, and the sound frames accumulated in these buffer memories are input into the FIFO (first in first out) system, that is, first. It has a function to process information in a way that goes out first. That is, the playback frame storage unit 250 has a function of storing the sound frames input from the playback frame input unit 240 in the input order and passing them to the sound device driver 260 in that order. The sound device driver 260 has a function of driving the sound device 261 to reproduce sound frames, and the sound device 261 has a function of D / A converting the sound frames that are digital data and reproducing them as sound. Have. That is, the sound device driver 260 and the sound device 261 function as sound frame reproduction means. The timer 262 is a timer used for timing the reproduction of the acoustic signal by the sound device and the reproduction of the acoustic signal of the external device, and is shared with a timer that performs time management in the computer.

  In the acoustic signal reproduction apparatus shown in FIG. 10, as described above, there are two systems, namely, a system for extracting and displaying additional information in the additional information extraction / display means 210 and a system for reproducing sound by the reproduction frame input means 240 to the sound device driver 260. A system exists. When processing a monaural sound signal, since there is only one channel in the sound signal and additional information is embedded in that channel, both systems process the signal of one channel. On the other hand, when processing a stereo sound signal, the sound reproduction system processes the signal of two channels and performs stereo reproduction. However, the additional information extraction system only processes the signal of one channel in which the additional information is embedded. Will be processed.

(9. Processing operation of playback device)
Next, the processing operation of the acoustic signal reproducing device shown in FIG. 10 will be described. When the recording device on which the modified acoustic signal in which the additional information is embedded by the embedding device shown in FIG. 1 is recorded is reproduced on the reproducing device, first, the acoustic frame reading means 200 starts from each channel of the stereo acoustic signal read from the recording medium. Each of a predetermined number of samples is read as one acoustic frame. The number of samples of one sound frame read by the sound frame reading means 200 needs to be the same as that set by the sound frame reading means 10 shown in FIG. Therefore, in the present embodiment, the acoustic frame reading means 200 sequentially reads 4096 samples for each of the left channel and the right channel as acoustic frames. Further, when the sound frame reading means 10 shown in FIG. 1 reads the sound frame so that the sample is overlapped with the sound frame adjacent to the sound frame, the sound frame reading means 200 is also adjacent to the sound frame. Read so that the sound frame and the sample overlap. In this case, the number of samples to be duplicated is the same as the number of samples duplicated by the acoustic frame reading means 10 and is 2048 samples, which is half of the 4096 samples constituting one acoustic frame. Even if the acoustic frame is read in duplicate, only the non-overlapping acoustic frame is passed to the additional information extraction / display unit 210. Further, the sound frame passed to the additional information extraction / display unit 210 is only for the left channel in which the additional information is embedded.

  Subsequently, the additional information extraction / display unit 210 performs the same processing as that of the extraction apparatus shown in FIG. 6 using each read sound frame as a reference frame, extracts the additional information, and displays and outputs it on the screen.

  On the other hand, the sound frame read by the sound frame reading means 200 is accumulated in the reproduction frame accumulation means 250 by the reproduction frame input means 240. If at least one frame is stored in the playback frame storage unit 250, the playback process by the sound device driver 260 is started. This playback process is performed by the sound frame reading unit 200, the additional information extraction display unit 210, and the playback frame input unit 240. Processing time is overwhelmingly longer than processing. (In other words, the additional information extraction and display unit 210 is designed to be overwhelmingly faster than the reproduction process.) Therefore, in the present embodiment, the reproduction frame accumulation unit 250 can accumulate up to four frames. As shown in FIG. 10, the sound frame reading means 200 continuously reads the next sound frame, and the playback frame input means 240 can store up to four sound frames in the playback frame storage means 250. Yes. When four frames are accumulated in the reproduction frame accumulation unit 250, the sound frame reading unit 200 and the reproduction frame input unit 240 are in a standby state until a space is generated in the reproduction frame accumulation unit 250. Accordingly, four frames are always stored in the playback frame storage unit 250, and the sound device driver 260 plays back the first acoustic frame among the acoustic frames stored in the playback frame storage unit 250. Specifically, the sound device 261 D / A converts the sound frame data and outputs it to the speaker. The acoustic frame that has been acoustically reproduced is deleted from the reproduction frame storage means 250.

  When the sound frame is deleted and there is room in the reproduction frame storage means 250, the sound frame is input from the reproduction frame input means 240 to the reproduction frame storage means 250. As a result, the reproduction frame storage means 250 stores the maximum capacity again. In reality, the read sound frame is input into the reproduction frame accumulating unit 250 when the CPU functions as the reproduction frame input unit 240. The reproduction frame input unit 240 not only simply inputs the audio frame to the reproduction frame storage unit 250 but also if the reproduction frame storage unit 250 has no free space, the reproduction frame storage unit 250 displays the audio frame reading unit 200 and the additional information extraction / display unit 210. On the other hand, a message for interrupting the process is sent to control the input of the sound frame to the reproduction frame storage means 250.

  On the other hand, the sound device 261 sequentially reproduces the first acoustic frame among the acoustic frames accumulated in the reproduction frame accumulation unit 250. At this time, each time the sound device 261 finishes the sound reproduction of one sound frame, the sound device 261 allows the sound frame reading means 200, the additional information extraction / display means 210, and the reproduction frame input means 240 to execute each process. Send.

  Here, the outline of the processing of the sound playback side in the playback apparatus, that is, the playback frame input means 240, the playback frame storage means 250, and the sound device driver 260 is organized and shown in the flowchart of FIG. First, the playback frame input means 240 searches for a free buffer memory in the playback frame storage means 250 (S601). If there is no vacant buffer memory, a message for interrupting the process is sent to the sound frame reading means 200 to wait for reception of a reproduction end message from the sound device 261 (S602). If there is a playback end message from the sound device driver 261, the playback end buffer is set as an empty buffer by deleting from the buffer memory storing the sound frame that has been played back (S603). Since the playback end message from the sound device 261 is simultaneously transmitted to the sound frame reading means 200, the additional information extraction / display means 210, and the playback frame input means 240, the sound frame reading means 200 and the playback frame input means 240 perform processing. The process is resumed (S604). Subsequently, the sound frame is stored in an empty buffer memory (S605). On the other hand, the sound device 261 always searches the buffer memory in the reproduction frame storage means 250 (S606), and when there is an acoustic frame, reproduces the acoustic frame (S607). Waiting for the reproduction of one acoustic frame (S608), when the reproduction is completed, a reproduction end message is transmitted to the acoustic frame reading means 200 and the reproduction frame input means 240 (S609).

  The sound signal reproduction device shown in FIG. 10 performs sound reproduction processing on the sound signal in the same manner regardless of whether or not the additional information is embedded. Therefore, the sound signal in which the additional information is not embedded is also reproduced as it is. Of the acoustic signal, the portion in which the additional information of the left channel is embedded is modified. However, since the high frequency component remains the original acoustic signal, it has various distributions based on the setting of the producer. As described above, when a stereo sound signal is used, the low frequency component changed in the left channel is always changed to the low frequency component of the right channel by the processing of [Equation 3] to [Equation 8]. Since it is added, there is no signal degradation when viewed as both channels as a whole. Therefore, it can be heard as normal sound signals.

(10. In the case of monaural sound signal)
In the above embodiment, the case where the additional information is embedded in the left channel signal of the stereo sound signal having the left and right channels has been described as an example in any of the embedding device, the extraction device, and the playback device. Additional information may be embedded. This is because the present invention is not related to the left and right characteristics. Further, when processing is performed on a monaural sound signal having only one channel, the processing performed on the left channel signal is performed in the above embodiment. Since the present invention embeds and extracts additional information from one channel signal, it can be similarly performed for a monaural sound signal or a stereo sound signal.

  FIG. 12 is a conceptual diagram when embedding additional information according to the present invention in a stereo sound signal and a monaural sound signal. FIG. 12A shows a stereo sound signal, and FIG. 12B shows a monaural sound signal. In the example of FIG. 12, the low frequency component for one acoustic frame is represented by a waveform, and the bit value “0” is embedded as an example.

  In the case of a stereo sound signal, embedding is performed on the left channel (L-ch) signal. As shown in FIG. 12A, after frequency conversion, signal separation is performed, and bit embedding processing is performed. Specifically, bit embedding is performed as a result of the processing of [Formula 3] to [Formula 8]. Here, as described above, when “0” is embedded, [Equation 6] and [Equation 7] are used. Therefore, after bit embedding processing, the signal component in the low frequency band is 0 (represented by no waveform in the figure) near the center of the acoustic frame (corresponding to window 2) and near the rear (corresponding to window 3). Become. At this time, as is clear from the contents of [Equation 6] and [Equation 7], the signal component from which the left channel signal is deleted is added to the right channel (R-ch) signal. Therefore, as shown in the lower part of FIG. 12A, the low-frequency component of the right channel signal is increased. After bit embedding processing, signal synthesis including high frequency components is performed, and then frequency inverse conversion is performed to obtain a modified acoustic signal.

  In the case of a monaural sound signal, processing is performed as shown in FIG. 12B, but as can be seen from the upper part of FIG. 12A, processing similar to the left channel of the stereo sound signal is performed. Become.

(11. Technique that enables information to be embedded even if the signal component is small)
In the processing described so far, it is necessary that a signal component having a predetermined size or more exists in the window 1 component and the window 3 component, and both the window 1 component and the window 3 component are not more than a predetermined size. In this case, information cannot be embedded. Therefore, hereinafter, a method for enabling signal embedding even when the window 1 component and the window 3 component are both equal to or smaller than a predetermined size will be described.

  In this case, the information embedding process in the embedding apparatus shown in FIG. 1 is performed according to the flowchart of FIG. The flowchart of FIG. 13 is different from the flowchart of FIG. 2 in that the low frequency component changing unit 30 does not determine the level in the frequency conversion process in S705, and there is no setting process for the separation mode corresponding to S206. Is a point. This is because the processing according to FIG. 13 forcibly embeds information even if the signal level is low. Therefore, it is determined whether there is a portion with a low signal level where information cannot be embedded. This is because it is not necessary to set the mode.

Therefore, as a process of setting one of the window 1 component and the window 3 component in S706 to be dominant, first, the fixed value V calculated according to the following [Equation 12] is used as the intensity of the 0th low frequency band. , Instead of the above summed values E ₁ and E ₂ .

[Formula 12]
V = {0.5 · Lev / (M−3)} ^1/2

  When the first value is 1 and the second value is 0, when the bit value is 1, the processing according to the above [Equation 4] and [Equation 5] is performed, and then the following [Equation 13 ] Is executed.

[Formula 13]
For the three components of the window, Al ′ (3, j) = Al (3, j) · V / {Ar (3, j) ² + Br (3, j) ² } ^1/2
Bl ′ (3, j) = Bl (3, j) · V / {Ar (3, j) ² + Br (3, j) ² } ^1/2

  When the bit value is 0, the processing according to the above [Equation 6] and [Equation 7] is executed, and then the processing according to the following [Equation 14] is executed.

[Formula 14]
For one component of window, Al ′ (1, j) = Al (1, j) · V / {Ar (1, j) ² + Br (1, j) ² } ^1/2
Bl ′ (1, j) = Bl (1, j) · V / {Ar (1, j) ² + Br (1, j) ² } ^1/2

  After performing the above processing in S706, the processing after the window two component deletion processing (S708) is performed in the same manner as the processing after S209 shown in FIG.

  As described above, even when information is embedded when the frequency component is small, 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 the flowchart in FIG. Is the same as Further, the configuration of the acoustic signal reproducing apparatus is the same as that of FIG. 10, and the processing operation is the same as that according to the flowchart of FIG.

  FIG. 14 shows a conceptual diagram when the additional information is embedded when the signal component is small with respect to the stereo sound signal and the monaural sound signal. FIG. 14A shows the case of a stereo sound signal, and FIG. 14B shows the case of a monaural sound signal. In the example of FIG. 14, as in the case of FIG. 12, a low frequency component for one acoustic frame is represented by a waveform and a bit value “0” is embedded as an example.

  In FIG. 14, the difference from the case of FIG. 12 is that the original signal component is small. As in the example shown in FIG. 14A, in the stage after the signal separation, even if the values of the window 1 component and the window 3 component are small, the bits are processed according to the processing of [Expression 12] to [Expression 14]. By embedding, the same signal component as in FIG. 12 is obtained.

  In the case of a monaural sound signal, processing is performed as shown in FIG. 14B. As can be seen from the upper part of FIG. 14A, processing similar to that for the left channel of a stereo sound signal is performed. Become.

It is a functional block diagram of an information embedding device for an acoustic signal. It is a flowchart which shows the process outline | summary of the apparatus shown in FIG. FIG. 3 shows how a low-frequency component is changed by processing according to FIG. 2. It is a flowchart which shows the process outline | summary by the other method of the apparatus shown in FIG. FIG. 5 shows how the low frequency component changes due to the processing according to FIG. 4. FIG. 1 is a functional block diagram of an apparatus for extracting information from an acoustic signal according to the present invention. It is a flowchart which shows the process outline | summary of the apparatus shown in FIG. It is a flowchart which shows the detail of the code | symbol determination process of S302 of FIG. It is a flowchart which shows the process outline | summary by the other method of the apparatus shown in FIG. It is a functional block diagram of the acoustic signal reproducing device according to the present invention. It is a flowchart which shows the process outline | summary of the apparatus shown in FIG. It is a conceptual diagram of the embedding process of the additional information by this invention. FIG. 2 is a flowchart showing an outline of processing when information can be embedded even if the original signal component is small in the apparatus shown in FIG. 1. It is a conceptual diagram of an additional information embedding process when the original signal component is small. It is a figure which shows the mode of the change of the signal waveform in the case of performing general Fourier transform. It is a figure which shows the window function used by this invention.

Explanation of symbols

DESCRIPTION OF SYMBOLS 10 ... Sound frame reading means 20 ... Frequency conversion means 30 ... Low frequency component change means 40 ... Frequency reverse conversion means 50 ... Modified sound frame output means 60 ... Storage means 61 ... -Acoustic signal storage unit 62 ... Additional information storage unit 63 ... Modified acoustic signal storage unit 70 ... Additional information reading means 100 ... Acoustic signal input means 110 ... Reference frame acquisition means 120 ... Phase change frame setting means 130: Frequency conversion means 140 ... Code determination parameter calculation means 150 ... Code output means 160 ... Additional information extraction means 170 ... Acoustic frame holding means 200 ... Acoustic frame Reading means 210... Additional information extraction and display means 240... Playback frame input means 250... Playback frame storage means 260. Device driver 261 ... Sound device 262 ... Timer

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,
From the acoustic signal, acoustic frame reading means for reading a predetermined number of samples as an acoustic frame;
The first window function, the second window function, and the second window function, which are spectrums corresponding to the first window function, by performing frequency conversion on the acoustic frame using a first window function, a second window function, and a third window function, respectively. Frequency conversion means for generating a second window spectrum that is a spectrum corresponding to, and a third window spectrum that is a spectrum corresponding to the third window function;
A spectrum set corresponding to a predetermined low frequency band is extracted from each of the generated window spectra, and a head of the extracted three sets of spectrum sets is based on the value of the information array of the additional information to be embedded. Low frequency component changing means for changing the ratio of the spectral intensity of the first window function and the rear third window function and deleting the component of the center second window function;
Frequency inverse transform means for performing frequency inverse transform on each window spectrum including the changed spectrum intensity 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 low frequency component changing means sets a predetermined fixed value V as the intensity of the low frequency band, and gives the predetermined intensity to the first window component and the third window component using the fixed value V. Thus, the information embedding device for the acoustic signal is characterized in that the ratio of the spectral intensity of the first window component and the third window component is changed. In claim 1 or claim 2,
The acoustic frame reading means reads the preceding acoustic frame and a predetermined number of samples in an overlapping manner, multiplies the entire read acoustic frame by a predetermined window function and passes it to the frequency conversion means,
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 1-3,
The apparatus for embedding information in an acoustic signal, wherein the low frequency component changing means sets a predetermined low frequency band in a low frequency region of 200 Hz or less. In claim 1,
The acoustic signal is a stereo acoustic signal composed of a time-series sample sequence of two left and right channels,
The sound frame reading means reads sound frames corresponding to the respective channels, and the frequency converting means performs frequency conversion on the sound frames of the respective channels, and the first window spectrum and the second window spectrum for each channel. , Generating a third window spectrum,
The low frequency component changing means is configured to determine the first information based on the value of the information array of the additional information to be embedded with respect to three sets of spectrum sets corresponding to a predetermined low frequency band extracted from the window spectrum of one channel. Change the ratio of the spectral intensities of the one window function and the third window function and delete the component of the second window function to supplement the component removed by the change made in the one channel. Changing the proportion of the spectral intensity between the spectral sets;
The frequency inverse transform means generates a modified acoustic frame by performing frequency inverse transform on the frame spectrum including the changed spectrum intensity for each channel, and the modified acoustic frame output means is generated for each channel. A device for embedding information in an acoustic signal, wherein the modified acoustic frames are sequentially output. A method of embedding additional information in an inaudible state for an acoustic signal composed of a time-series sample sequence,
From the acoustic signal, an acoustic frame reading stage for reading a predetermined number of samples as an acoustic frame;
The first window function, the second window function, and the second window function, which are spectrums corresponding to the first window function, by performing frequency conversion on the acoustic frame using a first window function, a second window function, and a third window function, respectively. A frequency conversion step of generating a second window spectrum that is a spectrum corresponding to, a third window spectrum that is a spectrum corresponding to the third window function;
A spectrum set corresponding to a predetermined low frequency band is extracted from each of the generated window spectra, and a head of the extracted three sets of spectrum sets is based on the value of the information array of the additional information to be embedded. Changing the ratio of the spectral intensities of the first window function and the rear third window function, and deleting the component of the center second window function;
A frequency inverse transform step of performing a frequency inverse transform on each window spectrum including the changed spectrum intensity to generate a modified acoustic frame;
A modified sound frame output step of sequentially outputting the generated modified sound frames;
A method for embedding information in an acoustic signal, comprising: A program that causes a computer to function as an apparatus 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 as an acoustic frame from the acoustic signal,
The first window function, the second window function, and the second window function, which are spectrums corresponding to the first window function, by performing frequency conversion on the acoustic frame using a first window function, a second window function, and a third window function, respectively. Frequency conversion means for generating a second window spectrum that is a spectrum corresponding to the third window function, a third window spectrum that is a spectrum corresponding to the third window function,
A spectrum set corresponding to a predetermined low frequency band is extracted from each of the generated window spectra, and a head of the extracted three sets of spectrum sets is based on the value of the information array of the additional information to be embedded. Low frequency component changing means for changing the ratio of the spectral intensity of the first window function and the rear third window function and deleting the component of the center second window function,
Frequency inverse transform means for performing a frequency inverse transform on each window spectrum including the changed spectrum intensity to generate a modified acoustic frame;
Modified acoustic frame output means for sequentially outputting the generated modified acoustic frames;
Program to function as. An apparatus for extracting additional information embedded in an inaudible state in advance from an acoustic signal,
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;
Frequency conversion is performed on the acoustic frame using a first window function and a third window function, respectively, and a first window spectrum and a spectrum corresponding to the third window function are spectra corresponding to the first window function. A frequency converting means for generating a third window spectrum;
A spectrum set corresponding to a predetermined low frequency band is extracted from each of the generated window spectra, a total value of spectrum intensities is calculated for each spectrum set, and based on a ratio between the spectrum sets of the total value Encoding means for outputting a predetermined code;
Additional information extracting means for extracting the additional information by converting the information array corresponding to the output code according to a predetermined rule;
An apparatus for extracting information from an acoustic signal, comprising: In claim 8,
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 method for extracting additional information embedded in an inaudible state in advance from an acoustic signal,
An acoustic frame acquisition step of digitizing a predetermined section of the acoustic signal to acquire an acoustic frame composed of a predetermined number of samples;
Frequency conversion is performed on the acoustic frame using a first window function and a third window function, respectively, and a first window spectrum and a spectrum corresponding to the third window function are spectra corresponding to the first window function. A frequency conversion stage for generating a third window spectrum;
A spectrum set corresponding to a predetermined low frequency band is extracted from each of the generated window spectra, a total value of spectrum intensities is calculated for each spectrum set, and based on a ratio between the spectrum sets of the total value An encoding stage for outputting a predetermined code;
An additional information extraction step of extracting the additional information by converting the information array corresponding to the output code according to a predetermined rule;
A method for extracting information from an acoustic signal. A program that causes a computer to function as an apparatus that extracts additional information embedded in an inaudible state in advance from an acoustic signal,
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;
Frequency conversion is performed on the acoustic frame using a first window function and a third window function, respectively, and a first window spectrum and a spectrum corresponding to the third window function are spectra corresponding to the first window function. A frequency converting means for generating a third window spectrum;
A spectrum set corresponding to a predetermined low frequency band is extracted from each of the generated window spectra, a total value of spectrum intensities is calculated for each spectrum set, and based on a ratio between the spectrum sets of the total value Encoding means for outputting a predetermined code;
Additional information extracting means for extracting the additional information by converting the information array corresponding to the output code according to a predetermined rule;
Program to function as. A device that reproduces sound while extracting additional information embedded in an inaudible state in advance from an acoustic signal,
From the acoustic signal, acoustic frame reading means for reading a predetermined number of samples as an acoustic frame;
The read sound frame is subjected to frequency conversion using a first window function and a third window function, respectively, and corresponds to a first window spectrum and a third window function which are spectra corresponding to the first window function. A frequency converting means for generating a third window spectrum which is a spectrum;
A spectrum set corresponding to a predetermined low frequency band is extracted from each of the generated window spectra, a total value of spectrum intensities is calculated for each spectrum set, and based on a ratio between the spectrum sets of the total value Encoding means for outputting a predetermined code;
Additional information display means for converting the information array corresponding to the outputted code into additional information for display output;
Reproduction frame storage means for storing two or more of the acoustic frames;
Reproduction frame input means for inputting the sound frame read by the sound frame reading means to the reproduction frame storage means;
A new sound frame can be inserted by sound-reproducing the first input sound frame among the sound frames existing in the playback frame storage means and deleting the sound frame from the playback frame storage means after the end of playback. Acoustic frame reproduction means for providing a room for the reproduction frame accumulation means and, when there is a next accumulated acoustic frame, for acoustic reproduction of the next acoustic frame in succession to the first input acoustic frame; ,
An acoustic signal reproducing device comprising: A method of reproducing sound while extracting additional information embedded in an inaudible state in advance from an acoustic signal,
From the acoustic signal, an acoustic frame reading stage for reading a predetermined number of samples as an acoustic frame;
The read sound frame is subjected to frequency conversion using a first window function and a third window function, respectively, and corresponds to a first window spectrum and a third window function which are spectra corresponding to the first window function. A frequency conversion stage for generating a third window spectrum which is a spectrum;
A spectrum set corresponding to a predetermined low frequency band is extracted from each of the generated window spectra, a total value of spectrum intensities is calculated for each spectrum set, and based on a ratio between the spectrum sets of the total value An encoding stage for outputting a predetermined code;
An additional information display step of converting the information array corresponding to the outputted code into additional information and displaying it;
A reproduction frame input step of inputting the sound frame read in the sound frame reading step into a reproduction frame storage means for storing two or more sound frames;
A new sound frame can be inserted by sound-reproducing the first input sound frame among the sound frames existing in the playback frame storage means and deleting the sound frame from the playback frame storage means after the end of playback. An acoustic frame reproduction step of providing a room for the reproduction frame storage means and, when there is a next stored acoustic frame, acoustically reproducing the next acoustic frame in succession to the first input acoustic frame; ,
An acoustic signal reproducing method comprising: A program for causing a computer to function as an apparatus for reproducing sound while extracting additional information embedded in an inaudible state in advance from an acoustic signal,
An acoustic frame reading means for reading a predetermined number of samples as an acoustic frame from the acoustic signal,
The read sound frame is subjected to frequency conversion using a first window function and a third window function, respectively, and corresponds to a first window spectrum and a third window function which are spectra corresponding to the first window function. A frequency converting means for generating a third window spectrum which is a spectrum;
A spectrum set corresponding to a predetermined low frequency band is extracted from each of the generated window spectra, a total value of spectrum intensities is calculated for each spectrum set, and based on a ratio between the spectrum sets of the total value Encoding means for outputting a predetermined code;
Additional information display means for converting the information array corresponding to the outputted code into additional information for display output;
Reproduction frame storage means for storing two or more of the acoustic frames;
Reproduction frame input means for inputting the audio frame read by the audio frame reading means to the reproduction frame storage means;
A new sound frame can be inserted by sound-reproducing the first input sound frame among the sound frames existing in the playback frame storage means and deleting the sound frame from the playback frame storage means after the end of playback. An acoustic frame reproducing means for acoustically reproducing the next acoustic frame in succession to the first input acoustic frame when there is a room for the acoustic frame stored next, and a room is provided in the reproduced frame storage means;
Program to function as.
An acoustic signal reproducing device comprising:

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