JP4760540B2 - Information embedding device for acoustic signals - Google Patents

Abstract

<P>PROBLEM TO BE SOLVED: To provide a device for embedding information into an acoustic signal, having improved accuracy at extraction time and increasing a volume of information which can be embedded. <P>SOLUTION: When the acoustic signal is divided into predetermined sections, and a rate of spectrum intensity of a head part and a tail part of a low frequency component of the predetermined section is changed based on a value of information arrangement of additional information to be embedded, frequency conversion is performed after a window function corresponding to the header part or a window function corresponding to the tail part, and a window function corresponding to the center part are combined. Thereby, accuracy at extraction time is improved by suppressing the occurrence of a quasi harmonics component. <P>COPYRIGHT: (C)2008,JPO&amp;INPIT

Description

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

  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 divides the low frequency region of the acoustic signal by a plurality of window functions, and based on the value of the information array of the attribute information (additional information) to be embedded, the spectrum of the head portion and the rear portion of the low frequency component When changing the intensity ratio, a method has been proposed in which additional information can be embedded by extracting the leading portion and the trailing portion of the low-frequency component using mutually asymmetric window functions (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. 2005-308980

  In the method described in Patent Document 7, a method is used in which information is embedded by dividing the window function into two in time series, and in order to clarify the separation of the two window components, an elongated window is provided in the center portion. Three window function processes were performed to remove the window component at the center. The original purpose of the window function is to suppress the generation of pseudo-harmonic components based on the clipping process when the original signal is partially clipped and subjected to signal processing such as Fourier transform. However, when the window function is divided as described in Patent Document 7, a pseudo-harmonic component based on the division is generated. In particular, an extra pseudo signal component is added to the elongated window at the center, and the quality of the embedded acoustic signal is deteriorated. In addition, the extraction accuracy of embedded additional information was reduced.

  Also, in the technique described in Patent Document 7, even if there is a frame that cannot be embedded because the low frequency component is small, is it the beginning of one word in order to enable efficient embedding? A flag bit indicating whether it was in the middle was embedded. For this reason, there is a problem that the overhead increases when divided into units of 2 bits or less.

  In the method described in Patent Document 7, a DC component remaining in a lower frequency region than in a region where data is embedded wraps around the embedded region during extraction, leading to a decrease in signal determination accuracy.

  Therefore, an object of the present invention is to provide an information embedding device for an acoustic signal that can increase the accuracy of extraction and increase the amount of information that can be embedded.

In order to solve the above-described problem, the present invention is an apparatus for embedding additional information in an inaudible state in an acoustic signal composed of a time-series sample sequence, and a predetermined number N of samples are obtained from the acoustic signal. And an acoustic frame reading means for reading two types of A-type acoustic frames and B-type acoustic frames that are shifted by a time corresponding to a predetermined number of samples less than N, and a first basic for the A-type acoustic frames. Frequency conversion is performed using a window function, a second basic window function, and a third basic window function, respectively, and the first basic window function kw (1, i) or the third basic window function is determined based on the value of additional information to be embedded. A first composite window spectrum that is a spectrum corresponding to the first composite window function W (1, i) is generated using the window function kw (3, i) as the first composite window function W (1, i). In both cases, the second basic window function kw (2, i) and the third basic window function kw (3, i) or the second basic window function kw (1, i) combined with the second synthetic window function W ( 2, i) is generated, a second synthesized window spectrum is generated, frequency conversion is performed on the B-type acoustic frame using a fourth basic window function, and the second basic window function corresponds to the fourth basic window function. A frequency conversion means for generating a fourth basic window spectrum, which is a spectrum, and a low frequency spectrum corresponding to a predetermined low frequency band from each of the generated window spectra, and an information array of the additional information to be embedded Based on the value, the ratio of the low frequency spectrum intensity by the first synthetic window function and the second synthetic window function is changed among the extracted low frequency spectra, and the low frequency by the fourth basic window function is changed. Low frequency component changing means for removing each component of the wave spectrum, frequency inverse converting means for generating a modified acoustic frame by performing frequency inverse conversion on each window spectrum including the changed low frequency spectrum, and Modified acoustic frame output means for sequentially outputting the generated modified acoustic frames, wherein the first basic window function, the second basic window function, the third basic window function, and the fourth basic window function are added. Then, the entire interval fixed value is set to 1, and the first basic window function and the second basic window function, and the second basic window function and the third basic window function are both at the same time. Are set so that there is a non-zero value at the same time, and the first and third basic window functions are set so that when one has a non-zero value, the other is always zero. by Tei Rutotomoni, each Provided is an information embedding device for an acoustic signal set so that both sides of a window function have an asymmetric cosine function.

  According to the present invention, the acoustic signal is divided into predetermined sections, and the ratio of the spectral intensity of the first 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. Since the frequency conversion is performed after synthesizing the window function corresponding to the head part or the window function corresponding to the rear part and the window function corresponding to the center part, generation of the pseudo-harmonic component can be suppressed, There is an effect that the accuracy at the time of extraction can be increased.

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 present applicant uses a plurality of window functions to divide the low frequency region of the acoustic signal into three in time series, and the ratio of the components of the two low frequency bands among them. A method has been proposed in which additional information can be embedded by changing the value according to the bit value of attribute information, or the delimiter and exception processing. The present invention is the same in that the three window functions used in Patent Document 7 are used.

  Here, the basic window function used in the present invention will be described with reference to FIG. In the present invention, basic window functions kw (1, i), kw (2, i), and kw (3, i) as shown in FIGS. 9B to 9D are prepared. The basic window function kw (1, i) is for extracting the front part of the acoustic frame, and has a maximum value of 1 at the position of a predetermined sample number i in the front part as shown in FIG. 9B. In the rear part, the minimum value is set to 0. Which sample number has the maximum value depends on the design of the basic window function kw (1, i), but in this embodiment, it is defined by [Equation 1] described later. By multiplying by the basic window function kw (1, i), the signal waveform of the acoustic frame as shown in FIG. 9A has a signal component remaining in the front part as shown in FIG. The signal component is deleted, and this becomes a Fourier transform target.

  Further, the basic window function kw (2, i) is for extracting the central portion of the acoustic frame. As shown in FIG. 9C, the basic window function kw (2, i) is maximum at a predetermined sample number i in the central portion. The value 1 is set, and the front part and the rear part are set to take the minimum value 0. Which sample number takes the maximum value depends on the design of the basic window function kw (2, i), but in this embodiment, it is defined by [Equation 2] described later. By multiplying the basic window function kw (2, i), the signal waveform of the acoustic frame as shown in FIG. 9 (a) has a signal component remaining at the center as shown in FIG. 9 (g). And the rear signal component are removed, and this is subjected to Fourier transform.

  The basic window function kw (3, i) is for extracting the rear part of the acoustic frame. As shown in FIG. 9 (d), the front part has a minimum value of 0 and a predetermined value in the rear part. The maximum value 1 is set at the position of sample number i. Which sample number takes the maximum value depends on the design of the basic window function kw (3, i), but in this embodiment, it is defined by [Equation 3] described later. By multiplying the basic window function kw (3, i), the signal waveform of the acoustic frame as shown in FIG. 9A is removed from the front signal component as shown in FIG. The signal component remains, and this becomes a Fourier transform target.

  In the present invention, acoustic frames are read in duplicate, and for odd frames (or even frames), the basic window function kw (1, i), the basic window function kw (2, i), and the basic window function kw (3 , I) and the basic window function kw (4, i) as shown in FIG. 9E is used for even frames (or odd frames).

  In the present invention, sound frames are read in duplicate. That is, a predetermined number of samples are redundantly read in the odd-numbered sound frames and the even-numbered sound frames. As mentioned above, the window function used is different between odd frames and even frames, but because odd frames and even frames are simply the difference between odd and even, which one is processed for which? Also good. Therefore, in this specification, one of the odd-numbered frame and the even-numbered frame is referred to as an A-type frame, and the other is referred to as a B-type frame. In the present embodiment, the even frame is described as an A type frame and the odd frame is described as a B type frame. Conversely, the odd frame may be an A type frame and the even frame may be a B type frame.

  In the present embodiment, the basic window function kw (1, i) to the basic window function kw (4, i) are defined by the following [Equation 1] to [Equation 4]. In FIG. 9, 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. 9A, 9F, 9G, 9H, and 9I, the vertical axis indicates the amplitude value (level) of the signal. 9B to 9E, the vertical axis indicates the values of the basic window functions kw (1, i), kw (2, i), kw (3, i), kw (4, i), The maximum values of kw (1, i), kw (2, i), kw (3, i), and kw (4, i) are all 1.

[Formula 1]
When i ≦ 3N / 8, kw (1, i) = 0.5−0.5 cos (8πi / (3N))
When 3N / 8 <i ≦ N / 2, kw (1, i) = 0.5−0.5 cos (8π (i−N / 4) / N)
When i> N / 2, kw (1, i) = 0.0

[Formula 2]
When i ≦ 3N / 8, kw (2, i) = 0.0
When 3N / 8 <i ≦ N / 2, kw (2, i) = 0.5−0.5 cos (8π (i−3N / 8) / N)
When N / 2 <i ≦ 3N / 4, kw (2, i) = 0.5−0.5 cos (4π (i−N / 4) / N)
When i> 3N / 4, kw (2, i) = 0.0

[Formula 3]
When i ≦ N / 2, kw (3, i) = 0.0
When i> N / 2, kw (3, i) = 0.5−0.5 cos (4π (i−N / 2) / N)

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

  As is clear from FIG. 9 and [Formula 1] to [Formula 4], the basic window functions kw (1, i) and kw (3, i) have asymmetric shapes. This is for facilitating identification between the two on the extraction side described later. The basic window functions kw (1, i), kw (2, i), kw (3, i) take a maximum value of 1 when i is a predetermined value, and i takes other values. Is obtained by dividing the basic window function that monotonically increases or decreases according to the value of i. Therefore, when the basic window functions kw (1, i) and kw (3, i) are determined, the basic window function kw (2, i) is inevitably determined. For this reason, the basic window function kw (2, i) has an asymmetric shape.

  In the present invention, since odd frames and even frames are redundantly read by a predetermined number of samples, after embedding information and then restoring to an acoustic signal, the odd frame multiplied by the window function and the window function are multiplied. When overlapping samples of even frames are added, it is necessary to return almost to the original value. Therefore, the shape of the basic window function kw (4, i) is inevitably determined according to the values of the basic window functions kw (1, i), kw (2, i), and kw (3, i). That is, when the basic window functions kw (1, i), kw (2, i), kw (3, i), kw (4, i) are added in the overlapping portion of the odd frame and the even frame, the fixed value for the entire section is obtained. It is defined to be 1.

  In the present invention, the above basic window functions are synthesized, and synthesized window functions W (1, i) and W (2, i) are generated according to the following [Equation 5], and Fourier is obtained using these synthesized window functions. Perform the conversion.

[Formula 5]
When embedding the first value (for example, “0”) and leaving the low frequency component in the front part W (1, i) = kw (1, i)
W (2, i) = kw (2, i) + kw (3, i)
When embedding the second value (for example, “1”) and leaving a low frequency component at the rear part W (1, i) = kw (3, i)
W (2, i) = kw (1, i) + kw (2, i)

  In the above [Equation 5], the basic window function that leaves the low frequency component is not synthesized, and the basic window function that removes the low frequency component is synthesized with the basic window function kw (2, i). . In the above [Equation 5], when not embedding a bit value, such as when embedding information indicating a delimiter of one word, a low frequency component is not left in either the front part or the rear part. It may be set.

(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 two sets of spectrum sets corresponding to two predetermined frequency ranges from the generated frame spectrum for the A type sound frame, and based on the additional information extracted from the additional information storage unit 62. The ratio (ratio) between the spectrum sets of the low frequency intensity data is changed, and for the B type sound frame, the low frequency intensity data in the predetermined frequency range of the generated frame spectrum is set to “0”. is doing. 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 one word of additional information. One word can be set to an arbitrary number of bits, but is usually set to 1 byte. First, the additional information reading means 70 reads additional information from the additional information storage unit 62 in units of one word (S101). Specifically, one word 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 in a delimiter in units of one word, and the bit mode indicates a mode for performing processing based on the value of each bit of one word. When one word is read from the additional information storage unit 62, the delimiter 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. In the present invention, there are A type and B type as acoustic frames. In the A-type acoustic frame and the B-type acoustic frame, the next sample after the last sample of the preceding acoustic frame of the same type is set as the first sample. The A-type and B-type sound frames are set by overlapping a predetermined number (2048 in this embodiment) of samples. For example, if the A type acoustic frame is A1, A2, A3... From the top and the B type acoustic frame is B1, B2, B3... From the top, A1 is samples 1 to 4096, A2 is samples 4097 to 8192, A3. Is samples 8193-12288, B1 is samples 2049-6144, B2 is samples 6145-10240, and B3 is samples 10241-14336. Since the A type and the B type are relative, either one may be first. In other words, A1 is samples 2049 to 6144, A2 is samples 6145 to 10240, A3 is samples 10241 to 14336, B1 is samples 1 to 4096, B2 is samples 4097 to 8192, and B3 is samples 8193 to 12288. May be. In the example of FIG. 2, the case where the B type sound frame is set first is shown.

  Subsequently, the frequency conversion means 20 performs frequency conversion on the read B-type sound frame to obtain a frame spectrum that is a spectrum of the sound frame (S105). Specifically, frequency conversion is performed using the window function for the B type acoustic frame read in S104. As frequency conversion, Fourier transform, wavelet transform, and other various known methods can be used. In the present embodiment, a case where Fourier transform is used will be described as an example.

  In general, when Fourier transform is performed on a predetermined signal, it is necessary to divide the signal into a predetermined length. It becomes discontinuous. 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. When performing Fourier transform in S105, specifically, the window function W (4, i) is applied to the left channel signal Xl (i) and the right channel signal Xr (i) (i = 0,..., N−1). ), The processing according to the following [Equation 6] is performed, and the conversion corresponding to the real part Al (4, j), the imaginary part Bl (4, j) and the right channel of the conversion data corresponding to the left channel is performed. The real part Ar (4, j) and the imaginary part Br (4, j) of the data are obtained. Note that window function W (4, i) = basic window function kw (4, i).

[Formula 6]
Al (4, j) = Σi _{= 0,..., N-1} W (4, i) · Xl (i) · cos (2πij / N)
Bl (4, j) = Σi _{= 0,..., N-1} W (4, i) .Xl (i) .sin (2πij / N)
Ar (4, j) = Σi _{= 0,..., N-1} W (4, i) .Xr (i) .cos (2πij / N)
Br (4, j) = Σi _{= 0,..., N-1} W (4, i) · Xr (i) · sin (2πij / N)

  In [Expression 6], 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. Next, the low frequency component changing means 30 removes the window 4 components (the components of the low frequency spectrum by the fourth window function) (S106). Specifically, the processing according to the following [Formula 7] is executed for the window 4 component.

  By executing the processing according to the above [Equation 6], 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 300 Hz or less. In the vicinity of a frequency of 300 Hz, j corresponds to 29, so the real part Al (4, j), the imaginary part Bl (4, j), the real part Ar (4, j) calculated by the above [Equation 6], Among the imaginary parts Br (4, j), those with j ≦ 29 (= M) are extracted.

[Formula 7]
For each component of j = 1 to M, Al ′ (4, j) = 0
Bl ′ (4, j) = 0
In the case of stereo, the following corresponding to the right signal is also calculated. E (4, j) = {Al (4, j) ² + B1 (4, j) ² + Ar (4, j) ² + Br (4, j) ² } ^{1 / 2}
Ar ′ (4, j) = Ar (4, j) · E (4, j) / {Ar (4, j) ² + Br (4, j) ² } ^1/2
Br ′ (4, j) = Br (4, j) · E (4, j) / {Ar (4, j) ² + Br (4, j) ² } ^1/2

  Next, the frequency inverse transform means 40 performs a process of obtaining a modified acoustic frame by performing frequency inverse transform on the frame spectrum from which the window four components have been removed by the process of S106 (S107). 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 Fourier transform, the frequency inverse transform unit 40 executes the Fourier inverse transform. Specifically, the real part Al ′ (4, j), the imaginary part Bl ′ (4, j) of the left channel of the spectrum obtained by any of the above [Equation 7], the real part Ar ′ ( 4, j) and imaginary part Br ′ (4, j) are used to perform processing according to the following [Equation 8] to calculate Xl ′ (i) and Xr ′ (i). For frequency components that are not modified in the above [Expression 7], in the following [Expression 8], Al ′ (4, j), Bl ′ (4, j), Ar ′ (4, j), Br The original values Al (4, j), Bl (4, j), Ar (4, j), and Br (4, j) are used as ′ (4, j).

[Formula 8]
Xl' (i) = 1 / N · {Σ j Al' (4, j) · cos (2πij / N) -Σ j Bl' (4, j) · sin (2πij / N)} + Xlp (i + N / 2 )
Xr' (i) = 1 / N · {Σ j Ar' (4, j) · cos (2πij / N) -Σ j Br' (4, j) · sin (2πij / N)} + Xrp (i + N / 2 )

  By the above [Equation 8], each sample Xl ′ (i) of the left channel and each sample Xr ′ (i) of the right channel of the B type modified acoustic frame are obtained. The modified sound frame output means 50 sequentially outputs the obtained modified sound frames to an output file. Subsequently, the acoustic frame reading means 10 reads a predetermined number of samples as A-type acoustic frames from the left and right channels of the stereo acoustic signal stored in the acoustic signal storage unit 61 (S108). As described above, the A type sound frame and the B type sound frame are overlapped by 2048 samples. Therefore, in S108, the acoustic frame reading means 10 moves the 2048 samples from the position where the B-type acoustic frame was read in S104, and reads the acoustic frame.

  Subsequently, the frequency conversion means 20 performs frequency conversion on the read A-type sound frame to obtain a frame spectrum that is a spectrum of the sound frame (S109). Specifically, for each acoustic frame, two synthetic window functions of window functions W (1, i) and W (2, i) are used. The composite window functions W (1, i) and W (2, i) are used to determine whether the bit value to be embedded is the first value (“0”) or the second value (“1”). Based on the above [Equation 5], the composite window function is set, and the frequency conversion means 20 calculates it. As the frequency conversion, Fourier transform, wavelet transform, and other various known methods can be used. In the present embodiment, Fourier transform is used as in the case of S105.

  When the Fourier transform is performed in S109, specifically, two composite window functions W (() for the left channel signal Xl (i) and the right channel signal Xr (i) (i = 0,..., N−1). 1, i) and W (2, i) are used to perform processing according to the following [Equation 9], and real parts Al (1, j), Al (2, j) of the conversion data corresponding to the left channel ), Imaginary part Bl (1, j), Bl (2, j), real part Ar (1, j), Ar (2, j) of the conversion data corresponding to the right channel, imaginary part Br (1, j) , Br (2, j). From the contents shown in [Formula 5], the first composite window function W (1, i) is a function that increases in the vicinity of the front portion of the acoustic frame when the first value is to be embedded. On the contrary, when the second value is to be embedded, it is a function that increases in the vicinity of the rear portion of the acoustic frame.

[Formula 9]
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)
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)

  As in [Formula 6] above, in [Formula 9], i is a serial number assigned to N samples in each acoustic frame, and an integer value of i = 0, 1, 2,... Take. 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 9], 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 two sets of spectra in a predetermined frequency range from the generated frame spectrum. As described above, the human auditory sense makes it difficult to detect the directionality of the low frequency component up to about 200 to 300 Hz, so the low frequency component is set to about 300 Hz or less. In the vicinity of a frequency of 300 Hz, j corresponds to 29, so the real part Al (1, j), Al (2, j), imaginary part Bl (1, j), Bl ( 2, j), real part Ar (1, j), Ar (2, j), imaginary part Br (1, j), Br (2, j), where j ≦ 29 (= M) is extracted Will do.

Subsequently, the low-frequency component changing unit 30 performs real part Al (1, j), imaginary part Bl (1, j) of the left channel, real part of the right channel on the first composite window function W (1, i) side. By using the part Ar (1, j) and the imaginary part Br (1, j), the total value E ₁ is calculated by the following [Equation 10].

[Formula 10]
E ₁ = Σ _{j = m,..., M-3} {Al (1, j) ² + Bl (1, j) ² + Ar (1, j) ² + Br (1, j) ² }

E ₁ calculated by the above [Equation 10] indicates the sum of the component intensities of the spectrum set in the vicinity of the front part or the rear part of the acoustic frame. Subsequently, it is determined whether or not the total value E ₁ is 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 = 29 is set. 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 total value E ₁ is equal to or higher 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. Further, in the present embodiment, regardless of the bit value to be embedded, it is embedded in the front and rear components (each component of the low frequency spectrum by the first composite window function W (1, i)). Here, each component of the low-frequency spectrum by the first composite window function W (1, i) is referred to as the front and rear component, as shown in the above [Equation 5], the first composite window function W (1, The actual condition i) is a first basic window function having a non-zero value in the front part or a third basic window function having a non-zero value in the rear part, and either one is replaced depending on the bit value of the additional information to be embedded. Because it is a thing. Therefore, in actuality, in the case of the first value (“0”), the bit value is the second value (“1”) for each component of the low frequency spectrum (basic window 1 component) by the first basic window function. In this case, the low frequency spectrum component (basic window three components) by the third basic window function is embedded. Therefore, when the total value E ₁ is less than the lower limit value Lev, the recording according to the bit value of the additional information is not performed, and the processing based on [Equation 11] is performed in the same manner as in the delimiter mode, and Therefore, the reading position is maintained as it is and the mode is set to the separation mode (S110). On the other hand, when the total value E ₁ is 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 (including the case where all of the low frequency components of the front and rear components and the window 2 component are all 0) in the left (L) channel signal. It performs (S112). Specifically, according to the following [Equation 11], processing for setting both L side to 0 is executed. In this case, the front and rear components of the right (R) channel signal and the window 2 component are not necessarily equal.

[Formula 11]
For j = 1 to M,
Al ′ (1, j) = 0
Bl ′ (1, j) = 0
Al ′ (2, j) = 0
Bl ′ (2, 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 (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

By executing the processing according to the above [Equation 11], the low-frequency component of the left channel frame spectrum is the same for both the front and rear components and the window 2 component of “0”. The pattern in which the front and rear components and the window 2 components are equal is information indicating the head position (separation) of the additional information or the sum value E ₁ is less than the lower limit value Lev. Information indicating that recording was not performed. In the above [Equation 11], Al ′ (j) = Bl ′ (j) = 0 is set for both the front and rear components and the two components of the window, 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 front and rear components and the window 2 components 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 means 30 is dominant in the front and rear components of the left channel signal regardless of the bit value of the bit array of the additional information extracted from the additional information storage unit 62. A process of changing to a state to become is performed (S111). 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 S111, for the component after Fourier transform using the composite window function calculated according to [Formula 5] according to the bit value that can take the first value and the second value, the following [Formula 12] and [Formula 13] are executed to change the spectral intensity of the front and rear components to a dominant state.

[Formula 12]
For each component of j = 0 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

  In this case, the processing according to the following [Equation 13] is executed for the front and rear components.

[Formula 13]
For each component of j = 0 to m−1, 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
Further, in the case of stereo, processing for moving the right channel component to the left channel component is performed as follows in order to emphasize the remaining component.
For each component of j = m to M-3, Ar ′ (1, j) = 0
Br ′ (1, j) = 0
Al ′ (1, j) = Al (1, j) · E (1, j) / {Al (1, j) ² + Bl (1, j) ² } ^1/2
Bl ′ (1, j) = Bl (1, j) · E (1, j) / {Al (1, j) ² + Bl (1, j) ² } ^1/2

  As a result of performing the processing according to the above [Equation 12] and [Equation 13], when j = 0,..., M−1 and j = M−2, M−1, M of the front and rear components, the value is “0”. However, there are other signal components that are equal to or greater than a predetermined value. Therefore, in this case, the ratio of the spectral intensity is changed so that the front and rear components are dominant. In the present embodiment, m = 10.

  Here, the situation of the low frequency component by the processing according to the above [Formula 12] and [Formula 13] is shown in FIG. In FIG. 3, the vertical direction is the frequency direction, and j = 2047 corresponds to 22.05 kHz. FIG. 3A shows the conventional system shown in Patent Document 7, and FIG. 3B shows the system according to the present invention. FIG. 3B shows a case where m = 10 and M = 29 in the above [Equation 13]. The shaded shade conceptually indicates the intensity of the signal component. In FIG. 3, the vertical direction is the frequency direction, and j = 2047 corresponds to 22.05 kHz. The conventional method and the present invention are the same in that the original sound is maintained without changing the high frequency component of j = 30 (about 300 Hz) or more. Also, the component of j = 27 to 29 is moved from the left (L) signal to the right (R) signal as in the conventional method.

  The present invention is different from the conventional method in that the conventional method sets j = 0 to maintain the original sound, moves the components of j = 1 to 29 in principle from the right (R) signal to the left (L) signal, and j = 1 to 26. In the present invention, j = 1 to 26 is set as an embedding area by selectively moving from the right (R) signal to the left (L) signal in accordance with the additional information. = 0 to 29 components are moved from the right (R) signal to the left (L) signal in principle, and the components of j = 10 to 26 are selectively changed from the right (R) signal to the left (in accordance with the additional information). L) By moving to a signal, j = 10 to 26 is set as an embedded region. The embedding region expresses the strength of the dominant window component, the corresponding R channel window component on the opposite side is moved and added, and the non-dominant window component moves in the reverse direction. And added to the window component of the R channel.

  By executing the processing according to the above [Equation 12] and [Equation 13], the front and rear components of the left channel signal are changed to a dominant pattern regardless of the bit values of the bit array of the additional information. It will be.

  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 11] in the separation mode in S112, and the processing based on [Equation 11] or [Equation 12] and [Equation 13] in the bit mode. Will be done.

  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 each window component is changed by the processes of S111 and S112 (S114). 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 [Formula 11] to [Formula 13], 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 14] is performed, and Xl ′ (i), Xr ′ (i ) Is calculated. For the frequency components that are not modified in the above [Formula 11] to [Formula 13], Al (1, j) or the like that is the original frequency component is used as Al ′ (1, j) or the like.

[Formula 14]
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)} + 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)} + Xrp (i + N / 2)

In the above [Expression 14], 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 14] are the data Xlp (i) of the modified acoustic frame modified immediately before, When Xrp (i) exists, the addition is performed in consideration of the overlap of N / 2 samples on the time axis. According to the above [Equation 14], each sample Xl ′ (i) of the left channel and each sample Xr ′ (i) of the right channel of the type A modified acoustic frame are obtained. The modified sound frame output means 50 sequentially outputs the obtained modified sound frames to an output file. When the processing for one acoustic frame is completed in this way, the mode is determined (S116). If the mode is the separation mode, the mode is set to the bit mode (S117), and then the acoustic frame reading means 10 is the B type. An acoustic frame is read (S104). On the other hand, when the mode is the bit mode, the low frequency component changing means 30 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 1-word data read in S101 is completed, the process returns from S103 to S101, and the next word of the additional information is read. When the processing is completed for all the words of the additional information, the processing returns to the first word of the additional information. As a result, all modified acoustic frames that have been processed for all acoustic frames are recorded in the output file and obtained as modified acoustic signals. The obtained modified acoustic signal is output to and stored in the modified acoustic signal storage unit 63 in the storage unit 60.

  The state of the change in the left channel signal by the above processing will be described in comparison with Patent Document 7 above. FIG. 10 shows a bit embedding procedure by the method described in Patent Document 7. In FIG. 10, the horizontal direction in the drawing is the time axis direction. In addition, a large number of rectangles in the figure indicate the presence (not deleted) of the basic window 1 component and the basic window 3 component of the modified acoustic frame. As described above, in the method described in Patent Document 7, if there is a frame that cannot be embedded because the low-frequency component is small, it is determined whether it is the beginning (new) or halfway (continuation) of one word. The flag bit shown was embedded. In the figure, “section” indicates a partition, “new” indicates new, “join” indicates continuation, and “1” and “0” indicate bit values of additional information. In the example of FIG. 10, a case is shown in which additional information of three consecutive words of 8 bits, 6 bits, and 12 bits is embedded. As shown in FIG. 10, after the information indicating the delimiter is first embedded, the information indicating new is embedded, and subsequently the information indicating the bit value is embedded. When there is no frame that cannot be embedded because the low frequency component is small, 8 bits for one word are continuously embedded, and then information indicating a delimiter and information indicating new are embedded again, The information indicating the bit value is embedded. When there is a frame that cannot be embedded because the low-frequency component is small, information indicating a delimiter is embedded after the first bit as in the third word shown in FIG. Next, information indicating continuation is embedded, and then embedded from the second bit.

On the other hand, in the present invention, when there is a frame that cannot be embedded because the low frequency component is small, the information indicating the delimiter is embedded, but the flag bit is not embedded, and the bit value of the subsequent additional information is continuously displayed. Perform the embedding process. FIG. 11 shows a bit embedding procedure according to the present invention. In the example of FIG. 11, a case is shown where additional information of four consecutive words having a fixed length of 8 bits is embedded. As shown in FIG. 11, after the information indicating the break is first embedded, the information indicating the bit value is embedded. When there is no frame that cannot be embedded because the low-frequency component is small, 8 bits for one word are continuously embedded, and then information indicating a delimiter is embedded again, followed by a bit value. Information is embedded. If there is a frame that cannot be embedded because the low-frequency component is small, information indicating a delimiter is embedded after the second bit as in the third word shown in FIG. Is embedded from the third bit without being embedded. In the present embodiment, when there is a frame that cannot be embedded because the low frequency component is small, there is an acoustic frame in which the total value E ₁ calculated by the above [Equation 10] is less than the level lower limit value Lev. Shows when to do.

  10 and 11 show an example of the processing in Patent Document 7 and the present invention. As shown in FIG. 10, the technique of Patent Document 7 can embed only 26 bits using 36 frames. On the other hand, in the method of the present invention, only 30 bits can be embedded using 36 frames, and the embedding efficiency is increased.

  In the present embodiment, the case where one word of additional information is 1 byte has been described. However, as long as there is an agreement with the extraction side, one word of additional information can be recorded in an arbitrary number of bits. .

  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 front component and the rear component, or the front component Therefore, there are only three distributions of whether the rear component is dominant. 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 11] to [Expression 13]. 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. 4 is a block diagram showing an embodiment of an apparatus for extracting information from an acoustic signal according to the present invention. In FIG. 4, 100 is an acoustic signal input means, 110 is a reference frame acquisition means, 120 is a phase change frame setting means, 130 is a frequency conversion means, 140 is a code determination parameter calculation means, 150 is a code output means, and 160 is additional information. Extraction means 170 is an acoustic frame holding means.

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. The code determination parameter calculation means 140 extracts each low frequency intensity data corresponding to a predetermined frequency or less from the generated frame spectrum, and adds up the value E of each low frequency intensity data for each of the basic window 1 component and the basic window 3 component. _C1 and E _C2 are calculated based on the following [Equation 15], and the combined values E _C1 and E _C2 are used as code determination parameters. Based on the ratio of the code determination parameters E _C1 and E _C2 , It has a function to judge that there is. The following [Formula 15] is obtained by deleting the right channel component in the above [Formula 10] because the right channel component is not referred to during extraction.

[Formula 15]
E _C1 = Σ _{j = m,..., M-3} {kAl (1, j) ² + kBl (1, j) ² }
E _C2 = Σ _{j = m,..., M-3} {kAl (3, j) ² + kBl (3, j) ² }

  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. 4 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. 4 will be described with reference to the flowchart of FIG. First, in this apparatus, the average code levels HL1 and HL2, the phase determination table S (p), and the non-code counter Nn are initialized (S200). These will be described. The average code levels HL1 and HL2 are low frequencies calculated by the above [Formula 15] for an acoustic frame (hereinafter referred to as an effective frame) that is determined to have embedded a binary value corresponding to a bit value. sum the average value of E _C1, E _C2 components, i.e., which is given by the average value of the sum E _C1, E _C2 in previous 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. The non-sign counter Nn is a counter of the number of frames that are determined to have a low signal level and non-sign (same as information indicating a break), and is set to Nn = 0 in the initial state.

  In this way, when the initial value is set and the user wants to know the attribute information such as the song name of the music that is flowing, first, the extraction device Instruct startup. For example, this can be executed by operating a predetermined button when the extraction device is realized by a mobile terminal such as a mobile phone. When an instruction is input to the extraction device, the acoustic signal input unit 100 records the flowing music, digitizes it, and inputs it as a digital acoustic signal. More specifically, the audio input from the omnidirectional microphone (or one channel of the directional microphone) is digitized by the A / D converter.

  Subsequently, the reference frame acquisition unit 110 extracts an acoustic frame including a predetermined number of samples from the acoustic signal input from the acoustic signal input unit 100 as a reference frame (S201). Specifically, 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 (S202). 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.

  Here, details of the code determination process in step S202 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 similar to the process in the frequency conversion means 20 shown in FIG. 1, but it is not necessary to refer to the second basic window function component in the extraction determination, and therefore, it is not a composite window function but the first first of both ends. And a third basic window function. Since only the left channel is used for extraction, processing according to the following [Equation 16] is performed, and real parts kAl (1, j) and kAl (3, j) of the conversion data corresponding to the left channel are performed. ), Imaginary parts kBl (1, j), kBl (3, j) are obtained.

[Formula 16]
kAl (1, j) = Σi _{= 0,..., N-1} kw (1, i) · Xl (i) · cos (2πij / N)
kBl (1, j) = Σi _{= 0,..., N-1} kw (1, i) · Xl (i) · sin (2πij / N)
kAl (3, j) = Σi _{= 0,..., N-1} kw (3, i) · Xl (i) · cos (2πij / N)
kBl (3, j) = Σi _{= 0,..., N-1} kw (3, i) · Xl (i) · sin (2πij / N)

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 which is an integrated value of the total value E _C1 for the sound frame for which the past basic window 1 component is determined to be dominant is used as the sound frame for which the past basic window 1 component is determined to be dominant. HL1 is calculated by dividing by n1, which is a number, and v2 which is the integrated value E _C2 of the sound frames for which the past basic window three components are determined to be dominant is dominant for the past basic window three components. HL2 is calculated by dividing by n2, which is the number of sound frames determined to be in a bad state. 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 100 Hz or more and about 300 Hz or less is extracted, and the real part Al (j of the left channel calculated by the above [Equation 9] is extracted as in the case of the embedding device. ), 10 ≦ j ≦ 29 are extracted from the imaginary part Bl (j). The code determination parameter calculating means 140, by executing the processing according to the above [Equation 15], the basic window 1 component of the sum E _C1, calculating a sum value E _C2 of the basic 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 calculating means 140 makes a determination basic window 1 component of the sum E _C1, the sum E _C2 of the basic window 3 components of whether each is below a predetermined value (S404). Specifically, values obtained by multiplying average code levels HL1 and HL2 by 0.001 are set as predetermined values. When the total value E _C1 is equal to or less than the value obtained by multiplying the average code level HL1 by 0.001 and the total value E _C2 is equal to or less than the value obtained by multiplying the average code level HL2 by 0.001, the code determination parameter calculation means 140 is determined to be delimiter information (S408).

On the other hand, the code determination parameter calculation means 140 performs the comparison determination with the predetermined values of the calculated code determination parameters E _C1 and E _C2 and the mutual comparison determination according to the following [Equation 17] (S405), and the comparison result is obtained. Output the corresponding code.

[Formula 17]
When E _C2 > (predetermined value) and E _C2 / E _C1 > 2, the basic window 3 component is dominant. When E _C1 > (predetermined value) and E _C1 / E _C2 > 2, the basic window 1 component is dominant. In other cases, 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 basic window 1 component is in a dominant state, a first bit value (for example, “0”) is output (S406). When it is determined that the basic window 3 component is in a dominant state, A second bit value (for example, “1”) is output (S407), and if both basic window components are determined to be equal, a code indicating delimiter information is output (S408). In S405, if it is determined that the basic window 1 component is in a dominant state, E _C1 is HL1 or higher, or if it is determined that the basic window 3 component is in a dominant state, E _C2 is HL2 or higher. If these conditions are not satisfied, a code indicating delimiter information is output (S408). Here, the code determined as “separation information” includes an embedding error indicating that the bit value was not recorded in the frame because the total value E ₁ described in paragraph 0048 is less than the lower limit value Lev. It is necessary to keep in mind that it is not necessarily a word delimiter. Whether or not the delimiter information indicates a true word delimiter is determined by whether or not the bit counter has reached 8 or more as described below.

  When it is determined that the basic window 1 component is in the dominant state and the first bit value is output (S406), or when it is determined that the basic window 3 component is in the dominant state and the second bit value is output ( In step S407, the phase determination table S (p) is updated in accordance with the following [Equation 18] (S409).

[Formula 18]
S (p) ← S (p) + E _C1 / E _C2 when one component of the basic window is dominant
When the three basic window components are dominant, S (p) ← S (p) + E _C2 / E _C1

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 _C1 and E _C2 corresponding to the low frequency component calculated by executing the processing according to the above [Equation 10] 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 process in S202, the average code level parameter is updated and the non-code counter is initialized (S203). Specifically, the sum values E _C1 and E _C2 are added to the accumulated values v1 and v2 which are numerators when calculating the average code levels HL1 and HL2, respectively, and the accumulated values v1 and v2 are updated, and the number of frames serving as the denominator. This is done by adding 1 to n1 and n2, respectively, and updating the number of frames n1 and n2. In the initialization process of the non-sign counter, the non-sign counter Nn = 0 is set as in the process in the initialization process of S200. Next, the output bit value is stored in a buffer (S209). Subsequently, the bit counter is counted up (S210).

  On the other hand, if a code corresponding to the delimiter information is output as a result of the process in S202, the non-code counter is incremented (S211). Specifically, 1 is added to the value of the unsigned counter Nn. If the value of the non-sign counter Nn is greater than or equal to a predetermined value, the process returns to S200 and initialization processing is performed. When the value of the non-sign counter Nn is less than a predetermined value, it is determined whether the bit counter is 8 or more. If it is less than 8, the code corresponding to the delimiter information is not a word delimiter but an embedding error. The current bit counter value is maintained, and the process returns to S201 to continue the code extraction process. If the bit counter is 8 or more, it can be determined that the code corresponding to the delimiter information indicates a word delimiter, so that the additional information extraction unit 160 outputs the data for one word recorded in the buffer (S206). . Then, the bit counter is initialized to 0 (S207). By executing the processing shown in FIG. 5 for each reference frame, additional information is extracted. If it is determined in S201 that all reference frames have been extracted, the process ends.

  In the process of S206, the additional information extraction unit 160 first generates a code indicating that the basic window 3 component and the basic window 3 component are equal among the ternary codes output by the code determination parameter calculation unit 140. Assuming that the next code is the head as the delimiter position, a bit array is created by associating the code indicating that the basic window 3 component is dominant and the code indicating that the basic window 3 component is dominant with the bit value. To do. Subsequently, when the code indicating the delimiter is extracted, if the bit counter of the bit array is less than 8, the code indicating the delimiter is determined to indicate an embedding error, and the current bit counter value is maintained and the bit is maintained. Continue creating the array. If the bit counter of the bit array just exceeds 8 or 8, it is determined that the code indicating the delimiter indicates the delimiter of the word (1 byte), and the last 8 bits of the bit array are converted according to a predetermined rule. And extracted as meaningful additional information. Here, the case of exceeding 8 bits is likely to occur immediately after starting the data extraction process, and occurs when the bit counter is initialized by misjudging the code indicating the delimiter as the head. 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.

(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 the frames whose sum values E _C1 and E _C2 are equal to or less than the predetermined threshold are invalid frames. The threshold at this time is set as the integrated value of the low frequency intensity for the 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 limit threshold correction process is performed centering on the processes in S402 and S203.

(8. Sound signal reproduction device)
Next, the acoustic signal reproducing apparatus according to the present invention will be described. FIG. 7 is a block diagram showing an embodiment of an acoustic signal reproduction device according to the present invention. In FIG. 7, reference numeral 200 denotes an audio frame reading means, 210 denotes additional information extraction / display means, 240 denotes a reproduction frame input means, 250 denotes a reproduction frame storage means, 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. 7 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 a phase change frame setting unit 120, a frequency conversion unit 130, a code determination parameter calculation unit 140, a code output unit 150, and an additional information extraction unit in 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 reproducing apparatus shown in FIG. 7, 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. 7 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. 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 device shown in FIG. 4 using each of the read sound frames 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. 7, the sound frame reading means 200 continuously reads the next sound frame so that the reproduction frame input means 240 can store up to four sound frames in the reproduction 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 acoustic signal reproduction device shown in FIG. 7 performs the acoustic reproduction process on the acoustic signal regardless of whether or not the additional information is embedded in the acoustic signal. 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 in the right channel by the processing of [Formula 11] to [Formula 13]. 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.

(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 front and rear components, and if the front and rear components are less than a predetermined size, information is embedded. I can't do it. Therefore, a method for enabling signal embedding even when the front and rear components are equal to or smaller than a predetermined size will be described below.

In this case, the information embedding process in the embedding apparatus shown in FIG. 1 is performed according to the flowchart of FIG. 2, but the total value E ₁ is not less than the level lower limit value Lev. The frequency component changing means 30 does not determine the level, and there is no setting process for the separation mode corresponding to S110. This is because in this process, information is forcibly embedded even if the signal level is low, so it is determined whether there is a portion with a low signal level where information cannot be embedded, and the separation mode is set. It is not necessary.

Therefore, the process of setting the front and rear components dominant state in S111, first, a fixed value V, which is calculated according to the following [Equation 19], as the intensity of the low-frequency component, instead of the sum E ₁ Set.

[Formula 19]
V = {0.5 · Lev / (M−2−m)} ^1/2

  Then, in any case of the bit value, after executing the processing according to the above [Equation 12] and [Equation 13], the processing according to the following [Equation 20] is executed.

[Formula 20]
For each component of j = m to M-3, 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

  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. 4, 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 in FIG. 7, and the processing operation is the same as that according to the flowchart of FIG.

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. It is a figure which shows the condition of the change of the low frequency component by the process of this invention. 1 is a functional block diagram of an apparatus for extracting information from an acoustic signal according to the present invention. 5 is a flowchart showing an outline of processing of the apparatus shown in FIG. 4. It is a flowchart which shows the detail of the code | symbol determination process of S202 of 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 figure which shows the basic window function used by this invention. It is a figure which shows the procedure of the bit embedding by the conventional method. It is a figure which shows the procedure of the bit embedding 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 (7)

A device for embedding additional information in an inaudible state with respect to an acoustic signal composed of a time-series sample sequence,
An acoustic frame reading means for reading two types of A-type acoustic frames and B-type acoustic frames that are shifted by a time corresponding to a predetermined number of samples less than N from the acoustic signal, with a predetermined number of N samples as acoustic frames;
For the A type acoustic frame, frequency conversion is performed using the first basic window function, the second basic window function, and the third basic window function, respectively, and the first basic window is based on the value of the additional information to be embedded. The function kw (1, i) or the third basic window function kw (3, i) is defined as a first synthesized window function W (1, i), and the spectrum corresponding to the first synthesized window function W (1, i) A first synthetic window spectrum is generated, and the second basic window function kw (2, i) and the third basic window function kw (3, i) or the first basic window function kw (1, i) are obtained. Generating a second synthesized window spectrum that is a spectrum corresponding to the synthesized second synthesized window function W (2, i), performing frequency conversion on the B type acoustic frame using a fourth basic window function, A spectrum corresponding to the fourth basic window function. And frequency conversion means for generating a fourth basic window spectrum,
From each of the generated window spectra, a low frequency spectrum corresponding to a predetermined low frequency band is extracted, and 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 low frequency spectrum intensity by the first synthetic window function and the second synthetic window function and removing each component of the low frequency spectrum by the fourth basic window function;
Frequency inverse transform means for performing frequency inverse transform on each window spectrum including the modified low frequency spectrum to generate a modified acoustic frame;
Modified sound frame output means for sequentially outputting the generated modified sound frames,
The first basic window function, the second basic window function, the third basic window function, the fourth basic window function, adding, has been set to be all the sections fixed value 1,
The first basic window function and the second basic window function, and the second basic window function and the third basic window function are set so that there are places where both have non-zero values at the same time,
The first basic window function third basic window function, one other is always set to be 0 when has a non-zero value Tei Rutotomoni, to have a bilateral asymmetric cosine function for each window function information embedding device for the acoustic signal, characterized in that the one set in. In claim 1,
The low frequency component changing means sets a predetermined fixed value V as an intensity of a low frequency band and uses the fixed value V to give a predetermined intensity to the low frequency component by the first composite window function. Accordingly, the information embedding apparatus for the acoustic signal is characterized in that the ratio of the low frequency spectrum intensity by the first synthetic window function and the second synthetic window function is changed. In claim 1 or claim 2,
The information embedding apparatus for an acoustic signal, wherein the low frequency component changing means sets a predetermined low frequency band in a low frequency region of 300 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 two kinds of sound frames corresponding to each channel, and the frequency conversion means performs frequency conversion on the A type sound frame of each channel and based on the value of additional information to be embedded. The first synthesized window spectrum and the second synthesized window spectrum for each channel are generated, the frequency conversion is performed on the B type acoustic frame of each channel, and the fourth basic window spectrum is generated.
The low frequency component changing means is configured to perform two sets of spectrum sets corresponding to a predetermined low frequency band extracted from the window spectrum of one channel.
Based on the value of the information array of the additional information to be embedded, the ratio of the low frequency spectrum intensity by the first synthesized window function and the second synthesized window function is changed, and the component of the fourth basic window function is deleted, Changing the ratio between the spectral sets of the spectral intensity of the other channel to supplement the components removed by the changes made in the one channel;
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. In claim 4,
In changing the ratio of the low frequency spectrum intensity by the first synthesis window function and the second synthesis window function, the component by the second synthesis window function is deleted in the one channel to supplement the deleted component. Add to the corresponding component of the other channel,
For the component by the first synthesis window function that has not been deleted in the one channel, the corresponding component in the other channel is deleted, and the deleted component is added to the corresponding component in the one channel. An apparatus for embedding information in an acoustic signal. A method of embedding additional information in an inaudible state for an acoustic signal composed of a time-series sample sequence,
An acoustic frame reading step of reading two types of A type acoustic frames and B type acoustic frames that are shifted by a time corresponding to a predetermined number of samples less than N from the acoustic signal as a predetermined number N of samples as acoustic frames;
For the A type acoustic frame, frequency conversion is performed using the first basic window function, the second basic window function, and the third basic window function, respectively, and the first basic window is based on the value of the additional information to be embedded. The function kw (1, i) or the third basic window function kw (3, i) is defined as a first synthesized window function W (1, i), and the spectrum corresponding to the first synthesized window function W (1, i) A first synthetic window spectrum is generated, and the second basic window function kw (2, i) and the third basic window function kw (3, i) or the first basic window function kw (1, i) are obtained. Generating a second synthesized window spectrum that is a spectrum corresponding to the synthesized second synthesized window function W (2, i);
A frequency conversion step of performing a frequency conversion on the B-type acoustic frame using a fourth basic window function to generate a fourth basic window spectrum that is a spectrum corresponding to the fourth basic window function;
From each of the generated window spectra, a low frequency spectrum corresponding to a predetermined low frequency band is extracted, and based on the value of the information array of the additional information to be embedded, A low frequency component changing step of changing a ratio of the low frequency spectrum intensity by the first synthetic window function and the second synthetic window function and removing each component of the low frequency spectrum by the fourth basic window function;
A frequency inverse transform stage that performs frequency inverse transform on each window spectrum including the modified low frequency spectrum to generate a modified acoustic frame;
Has a modified acoustic frame output stage for sequentially outputting the generated modified acoustic frame,
The first basic window function, the second basic window function, the third basic window function, the fourth basic window function, adding, has been set to be all the sections fixed value 1,
The first basic window function and the second basic window function, and the second basic window function and the third basic window function are set so that there are places where both have non-zero values at the same time,
The first basic window function and the third basic window function has a case with one of non-zero and the other is always set to be 0 Tei Rutotomoni, both sides are asymmetrical cosine function of each basic window functions embedding method of the information for the acoustic signal, characterized in that the set is intended to. 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 two types of A-type acoustic frames and B-type acoustic frames which are shifted from the acoustic signal by a time corresponding to a predetermined number of samples less than N, with a predetermined number N of samples as acoustic frames;
For the A type acoustic frame, frequency conversion is performed using the first basic window function, the second basic window function, and the third basic window function, respectively, and the first basic window is based on the value of the additional information to be embedded. The function kw (1, i) or the third basic window function kw (3, i) is defined as a first synthesized window function W (1, i), and the spectrum corresponding to the first synthesized window function W (1, i) A first synthetic window spectrum is generated, and the second basic window function kw (2, i) and the third basic window function kw (3, i) or the first basic window function kw (1, i) are obtained. Generating a second synthesized window spectrum that is a spectrum corresponding to the synthesized second synthesized window function W (2, i);
Frequency conversion means for performing frequency conversion on the B type acoustic frame using a fourth basic window function to generate a fourth basic window spectrum that is a spectrum corresponding to the fourth basic window function;
From each of the generated window spectra, a low frequency spectrum corresponding to a predetermined low frequency band is extracted, and based on the value of the information array of the additional information to be embedded, A low frequency component changing means for changing a ratio of the low frequency spectrum intensity by the first synthetic window function and the second synthetic window function and removing each component of the low frequency spectrum by the fourth basic window function;
Frequency inverse transform means for performing a frequency inverse transform on each window spectrum including the modified low frequency spectrum to generate a modified acoustic frame;
Causing the computer to function as modified acoustic frame output means for sequentially outputting the generated modified acoustic frames;
The first basic window function, the second basic window function, the third basic window function, the fourth basic window function, adding, has been set to be all the sections fixed value 1,
The first basic window function and the second basic window function, and the second basic window function and the third basic window function are set so that there are places where both have non-zero values at the same time,
The first basic window function and the third basic window function has a case with one of non-zero and the other is always set to be 0 Tei Rutotomoni, both sides are asymmetrical cosine function of each basic window functions A program characterized by being set as follows .

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