JP2007121626A - Network connection device and network connection system - Google Patents

Abstract

<P>PROBLEM TO BE SOLVED: To provide a network connection device and a network connection system that can provide address information such as a URL etc., so as not to disturb a content main body and easily acquire address information of a terminal device such as a mobile phone. <P>SOLUTION: After a sound generated by a sound signal reproducing device 400 is acquired as a digital sound signal through a microphone 310, an address information extraction unit 320 extracts embedded address information from the sound signal and a network access unit 330 uses the extracted address information to access the corresponding site 601. <P>COPYRIGHT: (C)2007,JPO&INPIT

Description

  The present invention extracts information such as URLs from audio signals of broadcast programs and CD / DVD packages, uses a mobile phone to access a web site related to a predetermined content, extracts detailed information, or answers a questionnaire The present invention relates to a contactless Internet connection technology.

Conventionally, a Web site is accessed by inputting a URL into a PC or a mobile phone. In recent years, URLs are not directly input, but URLs are embedded with QR codes or image digital watermarks on printed matter, or QR codes are embedded as still images in frames of TV moving images, and URLs are extracted with a mobile phone camera. A technology for guiding to a designated web site has been realized (see, for example, Patent Documents 1 to 3).
JP 2000-267966 A JP 2004-94398 A JP 2005-176287 A

  However, in the above conventional method, when extracting a QR code or digital watermark on an image with a mobile phone, it is necessary to manually adjust the camera finder to the area where the code is embedded, which takes time and effort. There's a problem. In addition, when the villa is a medium, if it takes time to prepare for reading, there is a problem that the screen on which the code is displayed is switched, and it is easy to miss a reading opportunity. In addition, in the case of QR codes, the layout is conspicuous, and in the case of moving image media in particular, there is a problem that the area occupied by the code is larger than the ratio of the screen size, which obstructs the layout.

  Therefore, the present invention can provide address information such as a URL so as not to interfere with the content body, and can easily acquire address information in a terminal device such as a mobile phone. It is an object to provide an apparatus and a network connection system.

  In order to solve the above-described problem, in the present invention, a sound emitted from a speaker is acquired as an acoustic signal, information embedded in an inaudible state in advance is extracted from the acoustic signal, and a network is extracted using the extracted information. As an apparatus for accessing the predetermined site above, an acoustic frame acquisition means for digitizing a predetermined section of the acoustic signal and acquiring an acoustic frame composed of a predetermined number of samples, and a frequency for the acoustic frame Using frequency conversion means for performing conversion to obtain a spectrum for each acoustic frame, and using the generated spectrum, at least two sets of spectrum sets are extracted, and a total value of spectrum intensities is calculated for each spectrum set. , A coding means for outputting a predetermined code based on a ratio between the spectrum sets of the total value, and the output An address information extracting means for converting the information array corresponding to the encoded code according to a predetermined rule and extracting address information; and a network access means for accessing a predetermined site on the network using the address information. A network connection device using a characteristic acoustic signal is provided.

  According to the present invention, the address information embedded in the audio signal is extracted, and the extracted address information is used to access the site on the network. Therefore, the URL is provided so as not to interfere with the content body. In addition, it is possible to easily obtain address information in a terminal device such as a mobile phone.

Hereinafter, embodiments of the present invention will be described in detail with reference to the drawings.
(1. First embodiment)
First, a first embodiment of the present invention will be described. FIG. 1 is a configuration diagram of a network connection apparatus according to the first embodiment of the present invention. In FIG. 1, 300 is a network connection device, 310 is a microphone, 320 is an address information extraction unit, 321 is an embedded information extraction unit, 322 is an address information conversion unit, 330 is a network access unit, 400 is an acoustic signal reproduction device, 500 Is a network, and 601 is a site.

  The network connection device 300 according to the present invention includes a microphone 310, an address information extraction unit 320, and a network access unit 330. The network connection device 300 acquires the sound reproduced and emitted by the acoustic signal reproduction device 400 as an acoustic signal. It has a function of accessing the site 601 on the network 500 using the address information extracted from the acoustic signal. The network connection device 300 is realized by mounting software that executes processing unique to the present invention on a mobile phone having a network connection function.

  The microphone 310 is a microphone that can detect a low-frequency component. As long as the low frequency component can be detected, it may be monaural omnidirectional or stereo directional. Even if it is stereo-directional, only one channel needs to be used. In particular, when using an acoustic signal in which information is embedded by an embedding device to be described later, information can be extracted even if a microphone with a general accuracy is used instead of a highly accurate one.

  The address information extraction unit 320 includes an embedded information extraction unit 321 and an address information conversion unit 322, and has a function of extracting address information from the acoustic signal acquired by the microphone 310. The embedded information extraction unit 321 has a function of extracting information embedded in the acoustic signal. The address information conversion unit 322 has a function of converting the extracted information into address information by referring to the address table held by itself.

  The network access unit 330 has a function of accessing the site 601 corresponding to the address information via the network 500 using the address information extracted by the address information extraction unit 320. The network access unit 330 is realized by a Web browser that acquires address information and accesses a site on the network.

  The acoustic signal reproduction device 400 has a function of reproducing an acoustic signal embedded in a state where information is inaudible and generating sound from a speaker, such as a so-called general-purpose television receiver, radio receiver, audio player, etc. Can be realized. The network 500 is a computer network such as the Internet that performs data communication according to a predetermined protocol. The site 601 is a Web site on a computer that can be accessed from a PC, a mobile phone, or the like via the network 500.

  Next, the processing operation of the apparatus shown in FIG. 1 will be described. FIG. 2 is a flowchart showing the processing operation of the network connection apparatus according to the first embodiment of the present invention. First, when the user knows that the address information is embedded in the music being played, the network connection device is instructed to start. For example, when the network connection device is realized by a mobile terminal such as a mobile phone, this can be executed by operating a predetermined button. When the instruction is input, the network connection device takes in the music flowing from the microphone 310, and the address information extraction unit 320 records it and inputs it as a digital sound signal (S801). Specifically, a process of digitizing audio input from the microphone 310 that is an omnidirectional microphone (or one channel of the directional microphone) by an A / D converter is performed.

  Subsequently, the embedded information extraction unit 321 extracts embedded additional information from the acquired acoustic signal (S802). Details of the extraction processing in the embedded information extraction unit 321 will be described later.

  Next, the address information conversion unit 322 converts the extracted additional information into address information (S803). The address information conversion unit 322 includes a conversion table in which the additional information and the address information are associated with each other. The address information conversion unit 322 refers to the conversion table with the extracted additional information and performs a process of converting into the address information.

  Subsequently, the network access unit 330 accesses the corresponding site using the address information obtained by the conversion (S804). For example, when the obtained address information indicates the site 601, the site 601 can be accessed and information held by the site 601 can be acquired.

(2. Second Embodiment)
Next, a second embodiment of the present invention will be described. FIG. 3 is a configuration diagram of a network connection apparatus according to the second embodiment of the present invention. In FIG. 3, those having the same functions as those of the first embodiment are denoted by the same reference numerals and description thereof is omitted. In FIG. 3, 340 is a content acquisition unit, and 602 is a second site.

  The content acquisition unit 340 has a function of acquiring content when the access destination site has downloadable content. The content acquisition unit 340 has a function of passing the acoustic signal to the embedded information extraction unit 321 when the acquired content includes an acoustic signal.

  Next, the processing operation of the apparatus shown in FIG. 3 will be described. FIG. 4 is a flowchart showing the processing operation of the network connection apparatus according to the second embodiment of the present invention. Also in FIG. 4, the same processes as those in the first embodiment are denoted by the same reference numerals and description thereof is omitted. Therefore, from S801 to S804, the same processing as in the first embodiment is performed. In the second embodiment, after accessing the site 601, if there is downloadable content in the site 601, the content acquisition unit 340 acquires the content (S805).

  The content acquisition unit 340 determines whether the acquired content includes an acoustic signal (S806). When the acquired content includes an acoustic signal, the content acquisition unit 340 passes the acoustic signal to the embedded information extraction unit 321. When receiving the acoustic signal from the content acquisition unit 340, the embedded information extraction unit 321 extracts the embedded additional information in the same manner as the acoustic signal acquired through the microphone 310 (S802). Then, after the address information conversion unit 322 converts the address information (S803), the network access unit 330 accesses the corresponding site using the address information (S804). For example, when the obtained address information indicates the second site 602, it is possible to access the second site 602 and acquire the information held by the second site 602.

  In this way, as long as the downloaded content includes the sound signal and the access information is extracted from the sound signal, the processes of S802 to S806 are repeated, and different sites are accessed one after another. In S806, if the downloaded content does not include an audio signal, new processing is not performed, and the information on the connection destination site is still displayed on the screen of the network connection device.

(3. Third embodiment)
Next, a network connection system according to the third embodiment of the present invention will be described. FIG. 5 is a configuration diagram of a network connection system according to the third embodiment of the present invention. In FIG. 5, components having the same functions as those of the network connection devices according to the first and second embodiments are denoted by the same reference numerals and description thereof is omitted. In FIG. 5, 301 is a terminal device, 320a is an address information extraction server, and 350 is an acoustic signal acquisition unit.

  The network connection system shown in FIG. 5 basically includes the functions of the network connection device 300 according to the first embodiment shown in FIG. 3 distributed in the terminal device 301 and the address information extraction server 320a. It has become. Therefore, the address information extraction server 320a has an embedded information extraction unit 321 and an address information conversion unit 322, similar to the address information extraction unit 320 shown in FIG. It has a function of extracting address information from the received acoustic signal. However, the address information extraction server 320a has a network access function to communicate with the terminal device 301.

  The acoustic signal acquisition unit 350 has a function of digitizing the acoustic signal acquired by the microphone 310. In the first embodiment shown in FIG. 3, this function is performed by the address information extraction unit 320.

  Next, the processing operation of the system shown in FIG. 5 will be described. FIG. 6 is a flowchart showing the processing operation of the network connection system according to the third embodiment of the present invention. Also in FIG. 6, the same processes as those in the first embodiment are denoted by the same reference numerals and the description thereof is omitted. Therefore, each process of S801, S802, S803, to S804 is performed in the same manner as in the first embodiment. In the third embodiment, after the terminal device 350 acquires an acoustic signal (S801), the network access unit transmits the acoustic signal to the address information extraction server 320a (S807). When the address information extraction server 320a receives the acoustic signal, the embedded information extraction unit 321 extracts the embedded additional information from the acquired acoustic signal (S802). Next, the address information conversion unit 322 converts the extracted additional information into address information (S803).

  Subsequently, the address information extraction server 320a transmits the address information obtained by the conversion to the terminal device 301 that is the transmission source (S808). When the terminal device 301 receives the address information, the network access unit 330 accesses the corresponding site using the received address information (S804).

  In the first to third embodiments, the address information extraction unit 320 includes the embedded information extraction unit 321 and the address information conversion unit 322, and extracts additional information embedded in the acoustic signal. The additional information is converted into address information, but the address information is embedded in the acoustic signal, and the address information extraction unit 320 has the same function as the embedded information extraction unit 321 and directly extracts the address information. You may make it do. However, since address information such as URL has a relatively large number of characters, it is preferable to embed shorter information as additional information in order to shorten the recording time, and the address information conversion unit 322 converts it into address information. . For example, a URL requires several tens of bytes, but if 4 to 5 bytes of characters are embedded as additional information, it is possible to record an acoustic signal of a portion where the additional information is embedded in a short time.

(4. Embedding additional information in acoustic signals)
Next, embedding of additional information in an acoustic signal will be described. The acoustic signal in which this additional information is embedded is reproduced by the acoustic signal reproduction device 400. FIG. 7 is a functional block diagram showing a configuration of an information embedding device for an acoustic signal. In FIG. 7, 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. Note that the apparatus shown in FIG. 7 can handle 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 unit 30 extracts three sets of spectrum sets corresponding to three 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. Note that the additional information is information other than the acoustic signal itself to be embedded as an additional to the acoustic signal, and whether the content is address information for accessing a computer on the network, such as an IP address or a URL, Or it is ID for converting into address information. In this embodiment, a case where an ID is used will be described. Each component shown in FIG. 7 is actually realized by mounting a dedicated program on hardware such as a computer and its peripheral devices. That is, the computer executes the contents of each means according to a dedicated program.

  Next, the processing operation of the information embedding device for the acoustic signal shown in FIG. 7 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. 8 corresponds to processing for 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 three types of modes: separation mode, bit mode, and continuous identification 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. In the continuous identification mode, when an acoustic frame with a low low-frequency component signal level appears, if the delimiter information is embedded, it is a new one starting from the beginning or continued because it was interrupted. A mode for recording information for identifying whether or not there is present is shown.

  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. That is, contrary to the above, A1 is samples 2049 to 6144, A2 is samples 6145 to 10240, A3 is samples 10241 to 14336, B1 is samples 1 to 4096, B2 is samples 4097 to 8192, and B3 is samples 8193 to 12288. May be. In the example of FIG. 8, 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 S109, which will be described later, similarly, frequency conversion is performed on the A type acoustic frame read in S108 by using a window function.

  Here, a window function used for Fourier transform in the present embodiment will be described. 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. In the present embodiment, for A type acoustic frames, window functions W (1, i), W (2, i) as shown in FIGS. 9B to 9D are used instead of the Hanning window function. ), W (3, i) is used for Fourier transform. This makes it easier for the extraction side to recognize the embedded information. The window function W (1, i) is for extracting the front part of the acoustic frame, and takes a maximum value 1 at the position of a predetermined sample number i in the front part as shown in FIG. In the rear part, the minimum value is set to 0. Which sample number takes the maximum value depends on the design of the window function W (1, i), but in this embodiment, it is defined by [Equation 1] described later. By multiplying by the window function W (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 component is deleted, and this becomes a Fourier transform target.

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

  The window function W (3, i) is for extracting the rear part of the acoustic frame. As shown in FIG. 9D, the window function W (3, i) takes a minimum value 0 in the front part and takes a predetermined sample in the rear part. The maximum value 1 is set at the position of the number i. Which sample number takes the maximum value depends on the design of the window function W (3, i), but in this embodiment, it is defined by [Equation 3] described later. By multiplying the window function W (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 becomes a Fourier transform target. Thus, after extracting the front part, the central part, and the rear part, the Fourier transform is executed, so that spectra corresponding to the front part, the central part, and the rear part are obtained. In order to embed a bit value in one acoustic frame, the bit value is originally divided into two parts, a front part and a rear part. However, on the extraction side, it is not always possible to read a signal synchronously. Therefore, in order to clearly distinguish the front part from the rear part, in this embodiment, the signal component at the center part is always deleted at the time of embedding, and the front part and the rear part are separated in time (however, at the time of extraction) Only need to analyze the front and rear, and ignore the middle). The greatest feature of the window function used in the present embodiment is that the window function W (1, i) and the window function W (3, i) are asymmetrical. For this reason, in particular, bit inversion hardly occurs.

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

  In the present embodiment, the sound frame is read in duplicate. That is, a predetermined number of samples are redundantly read in the odd-numbered sound frames and the even-numbered sound frames. As 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. According to the example of FIG. 8, odd frames are B type frames and even frames are A type frames. Conversely, even frames may be B type frames and odd frames may be A type frames.

  In the present embodiment, the window functions W (1, i) to W (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 window functions W (1, i), W (2, i), W (3, i), and W (4, i). The maximum values of (1, i), W (2, i), W (3, i), and W (4, i) are all 1.

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

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

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

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

  As is clear from FIG. 14 and [Formula 1] to [Formula 4], the window functions W (1, i) and W (3, i) have asymmetric shapes. This is for facilitating identification between the two on the extraction side described later. The window functions W (1, i), W (2, i), and W (3, i) have a maximum value of 1 when i is a predetermined value, and i takes other values. , I is a window function that monotonically increases or decreases according to the value of i, and therefore, when the window functions W (1, i) and W (3, i) are determined, the window function W (2, i ) Is inevitably determined. For this reason, the window function W (2, i) has a left-right asymmetric shape.

  In this embodiment, since odd frames and even frames are read in duplicate by predetermined samples, after embedding information, when restoring to an acoustic signal, an odd frame multiplied by a window function and a window function are When overlapping samples of even frames multiplied are added, it must be returned to the original value. Therefore, the shape of the window function W (4, i) is inevitably determined according to the values of the window functions W (1, i), W (2, i), and W (3, i). That is, when the window functions W (1, i), W (2, i), W (3, i), and W (4, i) are added in the overlapping portion of the odd and even frames, the fixed value 1 for the entire section is obtained. Is defined to be The above is the outline of the window function used in this embodiment.

  Returning to S105, description will be made. When performing the Fourier transform in S105, specifically, the window function W (4, 4) is applied to the left channel signal Xl (i) and the right channel signal Xr (i) (i = 0,..., N−1). i) is used to perform processing according to the following [Equation 5] and correspond 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. Real part Ar (4, j) and imaginary part Br (4, j) of the conversion data are obtained.

[Formula 5]
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 5], 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, processing according to the following [Equation 6] is executed for the window 4 component.

  By executing the processing according to the above [Equation 5], a frame spectrum in which the signal component of each acoustic frame is represented by a spectrum corresponding to the frequency is obtained. Subsequently, the low frequency component changing unit 30 extracts a spectrum set of three predetermined frequency ranges from the generated frame spectrum. Human hearing is known to be less sensitive to directionality for low frequency components up to about 200-300 Hz (Corona Corp., issued October 30, 1990, "Sound Engineering Course 1. Basic Acoustics"). Engineering, Acoustical Society of Japan ”p.247 (see FIGS. 9 and 26). Therefore, in this embodiment, the low frequency component is about 200 Hz or less. In the vicinity of a frequency of 200 Hz, j corresponds to 20, so the real part Al (4, j), the imaginary part Bl (4, j), the real part Ar (4, j) calculated by the above [Equation 5], Among the imaginary parts Br (4, j), those with j ≦ 20 (= M) are extracted.

[Formula 6]
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 one of the above [Equation 6], the real part Ar ′ ( 4, j) and imaginary part Br ′ (4, j) are used to perform processing according to the following [Equation 7] to calculate Xl ′ (i) and Xr ′ (i). For frequency components that are not modified in the above [Equation 6], in the following [Equation 7], 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 7]
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 7], 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. 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 B type sound frame and the A 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, each acoustic frame is performed using three window functions of window functions W (1, i), W (2, i), and W (3, i). 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 performing Fourier transform in S109, specifically, three window functions W (1) for the left channel signal Xl (i) and the right channel signal Xr (i) (i = 0,..., N−1). , I), W (2, i), and W (3, i), the processing according to the following [Equation 8] is performed, and the real part Al (1, j) of the conversion data corresponding to the left channel , Al (2, j), Al (3, j), imaginary part Bl (1, j), Bl (2, j), Bl (3, j), real part Ar () of the conversion data corresponding to the right channel 1, j), Ar (2, j), Ar (3, j), imaginary part Br (1, j), Br (2, j), Br (3, j) are obtained. Note that the window functions W (1, i), W (2, i), and W (3, i) are functions whose values increase near the front (front), near the center, and near the rear of the acoustic frame, respectively. ing.

[Formula 8]
Al (1, j) = Σi _{= 0,..., N-1} W (1, i) .Xl (i) .cos (2πij / N)
Bl (1, j) = Σi _{= 0,..., N-1} W (1, i) · Xl (i) · sin (2πij / N)
Al (2, j) = Σi _{= 0,..., N-1} W (2, i) · Xl (i) · cos (2πij / N)
Bl (2, j) = Σi _{= 0,..., N-1} W (2, i) · Xl (i) · sin (2πij / N)
Al (3, j) = Σi _{= 0,..., N-1} W (3, i) · Xl (i) · cos (2πij / N)
Bl (3, j) = Σi _{= 0,..., N-1} W (3, i) .Xl (i) .sin (2πij / N)
Ar (1, j) = Σi _{= 0,..., N-1} W (1, i) .Xr (i) .cos (2πij / N)
Br (1, j) = Σ _{i = 0,..., N-1} W (1, i) · Xr (i) · sin (2πij / N)
Ar (2, j) = Σi _{= 0,..., N-1} W (2, i) .Xr (i) .cos (2πij / N)
Br (2, j) = Σi _{= 0,..., N-1} W (2, i) .Xr (i) .sin (2πij / N)
Ar (3, j) = Σi _{= 0,..., N-1} W (3, i) · Xr (i) · cos (2πij / N)
Br (3, j) = Σi _{= 0,..., N-1} W (3, i) · Xr (i) · sin (2πij / N)

  As in [Formula 5], in [Formula 8], 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 8], a frame spectrum in which the signal component of each acoustic frame is represented 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. 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 200 Hz or less here as well. In the vicinity of a frequency of 200 Hz, j corresponds to 20, so the real part Al (1, j), Al (2, j), Al (3, j), and imaginary part Bl ( 1, j), Bl (2, j), Bl (3, j), real part Ar (1, j), Ar (2, j), Ar (3, j), imaginary part Br (1, j) , Br (2, j), Br (3, j), j ≦ 20 (= M) are extracted.

Subsequently, the low frequency component changing unit 30 extracts the real part Al (1, j), Al (3, j), the imaginary part Bl (1, j), Bl (3, j), and the right channel of the extracted left channel. of the real part Ar (1, j), Ar (3, j), the imaginary part Br (1, j), using the Br (3, j), the following [equation 9], sum E _1, to calculate the total value E _2.

[Formula 9]
E ₁ = Σj _{= 1,..., M-3} {Al (1, j) ² + Bl (1, j) ² + Ar (1, j) ² + Br (1, j) ² }
E ₂ = Σ _{j = 1,..., M-3} {Al (3, j) ² + Bl (3, j) ² + Ar (3, j) ² + Br (3, j) ² }

E ₁ calculated by the above [Equation 9] represents the sum of the component intensities of the spectrum set near the front of the acoustic frame, and E ₂ represents the sum of the component intensities of the spectrum set near the rear of the acoustic frame. Subsequently, it is determined whether or not the combined values E ₁ and E ₂ are equal to or higher than the level lower limit value Lev. The level lower limit value Lev is set to 0.5 when the maximum amplitude value of the acoustic signals Xl (i) and Xr (i) is normalized to 1 and M = 20. This value of Lev = 0.5 is a level at which the resistance to analog conversion can be maintained empirically, and when there are few low frequency components, it will be reduced as appropriate. In this case, the detection accuracy also decreases due to analog conversion. Will do.

The reason why it is determined whether or not the combined values E ₁ and E ₂ are equal to or greater than the level lower limit value Lev is that if the signal strength is small, even if the signal is changed, the change cannot be detected on the extraction side. is there. In this embodiment, when the bit value that can take the first value (for example, “1”) and the second value (for example, “0”) is “1”, the window three components (low by the third window function). When the bit value is “0” in each component of the frequency spectrum, it is embedded in the window 1 component (each component of the low frequency spectrum by the first window function). Therefore, when the bit value to be embedded is “1”, the sum value E ₁ is less than the lower limit value Lev, and when the bit value to be embedded is “0”, the sum value E ₂ is less than the level lower limit value Lev. In order to process again from the first bit without recording according to the bit value of the access information, the reading position is returned to the first bit, and the mode is set to the delimiter mode (S110). On the other hand, when the bit value to be embedded is “1” and the total value E ₁ is greater than or equal to the level lower limit value Lev, or when the bit value to be embedded is “0” and the total value E ₂ is greater than or equal to the level lower limit value Lev, the mode is determined. It will be.

  When the mode is the separation mode, the low-frequency component changing unit 30 performs processing for equalizing the low-frequency components of the window 1 component and the window 3 component (including the case where all are 0) in the left (L) channel signal. It performs (S112). Specifically, according to the following [Equation 10], processing for setting both L side to 0 is executed. In this case, the window 1 component and the window 3 component of the right (R) channel signal are not necessarily equal.

[Formula 10]
For j = 1 to M,
Al ′ (1, j) = 0
Bl ′ (1, j) = 0
Al ′ (3, j) = 0
Bl ′ (3, j) = 0
In the case of stereo, the following corresponding to the right signal is also calculated: E (1, j) = {Al (1, j) ² + Bl (1, j) ² + Ar (1, j) ² + Br (1, j) ² } ^{1 / 2}
Ar ′ (1, j) = Ar (1, j) · E (1, j) / {Ar (1, j) ² + Br (1, j) ² } ^1/2
Br ′ (1, j) = Br (1, j) · E (1, j) / {Ar (1, j) ² + Br (1, j) ² } ^1/2
E (3, j) = {Al (3, j) ² + B1 (3, j) ² + Ar (3, j) ² + Br (3, j) ² } ^1/2
Ar ′ (3, j) = Ar (3, j) · E (3, j) / {Ar (3, j) ² + Br (3, j) ² } ^1/2
Br ′ (3, j) = Br (3, j) · E (3, j) / {Ar (3, j) ² + Br (1, j) ² } ^1/2

  By executing the processing according to the above [Equation 10], the low frequency component of the frame spectrum of the left channel is the same when both the window 1 component and the window 3 component are “0”. The pattern in which the window 1 component and the window 3 component are equal is information indicating the head position (separation) of the additional information. In the above [Equation 10], Al '(j) = Bl' (j) = 0 is set for both the window 1 component and the window 3 component. Therefore, if the value is sufficiently small, it is not always necessary to set it to zero. Further, the window 1 component and the window 3 component do not necessarily have to be the same, and it is sufficient that the difference is small. In this sense, the term “equal” is used here.

  On the other hand, when the mode is the bit mode or the continuous identification mode, the low frequency component changing means 30 determines the window 1 component of the left channel signal and the bit value of the bit array of the additional information extracted from the additional information storage unit 62. Processing for changing the ratio of the spectral intensity of the window 3 component to a state where the window 1 component is dominant or the window 3 component is dominant (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, the window 1 component is obtained by executing the processing according to any of the following [Formula 11] and [Formula 12] according to the bit values that can take the first value and the second value. The magnitude relationship between the spectral intensity of the window 3 and the spectral intensity of the window 3 component is changed, and processing is performed to change the window 1 component to dominant or the window 3 component to dominant. For example, when the first value is 1 and the second value is 0, when the bit value is 1, the processing according to the following [Equation 11] is performed on the window 1 component.

[Formula 11]
For j = 1 to M, Al ′ (1, j) = 0
Bl ′ (1, j) = 0
In the case of stereo, the following corresponding to the right signal is also calculated: E (1, j) = {Al (1, j) ² + Bl (1, j) ² + Ar (1, j) ² + Br (1, j) ² } ^{1 / 2}
Ar ′ (1, j) = Ar (1, j) · E (1, j) / {Ar (1, j) ² + Br (1, j) ² } ^1/2
Br ′ (1, j) = Br (1, j) · E (1, j) / {Ar (1, j) ² + Br (1, j) ² } ^1/2

  In this case, the processing according to the following [Equation 12] is executed for the three components of the window.

[Formula 12]
For three components j = M−2, M−1 and M, Al ′ (3, j) = 0
Bl ′ (3, j) = 0
In the case of stereo, the following corresponding to the right signal is also calculated. E (3, j) = {Al (3, j) ² + Bl (3, j) ² + Ar (3, j) ² + Br (3, j) ² } ^{1 / 2}
Ar ′ (3, j) = Ar (3, j) · E (3, j) / {Ar (3, j) ² + Br (3, j) ² } ^1/2
Br ′ (3, j) = Br (3, j) · E (3, j) / {Ar (3, j) ² + Br (3, j) ² } ^1/2
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 j = 1 to M-3, Ar ′ (3, j) = 0
Br ′ (3, j) = 0
Al ′ (3, j) = Al (3, j) · E (3, j) / {Al (3, j) ² + Bl (3, j) ² } ^1/2
Bl ′ (3, j) = B1 (3, j) · E (3, j) / {Al (3, j) ² + Bl (3, j) ² } ^1/2

  As a result of performing the processing according to the above [Formula 11] and [Formula 12], a value of “0” is obtained when j = M−2, M−1, and M of the three components of the window, but signals other than the predetermined value are obtained. Ingredients will be present. Therefore, in this case, the ratio of the spectral intensity is changed so that the window 3 component is dominant. Subsequently, when the bit value is 0, the processing according to the following [Equation 13] is executed for the three window components.

[Formula 13]
For each component of j = 1 to M, Al ′ (3, j) = 0
Bl ′ (3, j) = 0
In the case of stereo, the following corresponding to the right signal is also calculated. E (3, j) = {Al (3, j) ² + Bl (3, j) ² + Ar (3, j) ² + Br (3, j) ² } ^{1 / 2}
Ar ′ (3, j) = Ar (3, j) · E (3, j) / {Ar (3, j) ² + Br (3, j) ² } ^1/2
Br ′ (3, j) = Br (3, j) · E (3, j) / {Ar (3, j) ² + Br (3, j) ² } ^1/2

  In this case, the processing according to the following [Equation 14] is executed for the window 1 component.

[Formula 14]
For three components j = M−2, M−1 and M, Al ′ (1, j) = 0
Bl ′ (1, j) = 0
In the case of stereo, the following corresponding to the right signal is also calculated: E (1, j) = {Al (1, j) ² + Bl (1, j) ² + Ar (1, j) ² + Br (1, j) ² } ^{1 / 2}
Ar ′ (1, j) = Ar (1, j) · E (1, j) / {Ar (1, j) ² + Br (1, j) ² } ^1/2
Br ′ (1, j) = Br (1, j) · E (1, j) / {Ar (1, j) ² + Br (1, j) ² } ^1/2
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 j = 1 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 13] and [Equation 14], a value of “0” is obtained when j = M−2, M−1, and M of one component of the window, but signals other than the predetermined value are obtained in other cases. Ingredients will be present. Therefore, in this case, the ratio of the spectrum intensity is changed so that the window 1 component is dominant.

  By executing the processing according to any one of [Formula 11] and [Formula 12], or [Formula 13] and [Formula 14], the left channel signal is determined according to each bit value of the bit array of the additional information. Thus, the pattern is changed so that the window 1 component is dominant or the window 3 component is dominant. In S111, in the case of the continuous identification mode, if it is new, the distribution between the low frequency component window 1 component and the window 3 component is changed to a state in which the window 3 component predominates according to [Equation 11], and continues. In this case, according to [Equation 12], the distribution of the low frequency component between the window 1 component and the window 3 component is changed to a state in which the window 1 component is dominant.

  In 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 processing based on [Equation 10] in the case of the separation mode in S112, and [Equation 11], [Equation 12] or [Equation 13] in the bit mode or the continuous identification mode. The processing based on Equation 14] is performed in S111.

  In either case of S111 or S112, next, the low frequency component changing means 30 deletes the window two components (the components of the low frequency spectrum by the second window function) (S113). Specifically, the processing according to the following [Formula 15] is executed for the two components of the window.

[Formula 15]
For each component of j = 1 to M, Al ′ (2, j) = 0
Bl ′ (2, j) = 0
In the case of stereo, the following corresponding to the right signal is also calculated: E (2, j) = {Al (2, j) ² + Bl (2, j) ² + Ar (2, j) ² + Br (2, j) ² } ^{1 / 2}
Ar ′ (2, j) = Ar (2, j) · E (2, j) / {Ar (2, j) ² + Br (2, j) ² } ^1/2
Br ′ (2, j) = Br (2, j) · E (2, j) / {Ar (2, j) ² + Br (2, j) ² } ^1/2

  Next, the frequency inverse transform means 40 performs a 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 to S113 (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 10] to [Formula 15], right Using the real part Ar ′ (1, j) of the channel, the imaginary part Br ′ (1, j), etc., processing according to the following [Equation 16] is performed, and Xl ′ (i), Xr ′ (i ) Is calculated. For the frequency components not modified in the above [Equation 10] to [Equation 15], Al (1, j) or the like that is the original frequency component is used as Al ′ (1, j) or the like.

[Formula 16]
Xl' (i) = 1 / N · {Σ j Al' (1, j) · cos (2πij / N) -Σ j Bl' (1, j) · sin (2πij / N)} + 1 / N · { Σ _j Al ′ (2, j) · cos (2πij / N) −Σ _j Bl ′ (2, j) · sin (2πij / N)} + 1 / N · {Σ _j Al ′ (3, j) · cos (2πij / N) −Σ _j Bl ′ (3, j) · sin (2πij / N)} + Xlp (i + N / 2)
Xr' (i) = 1 / N · {Σ j Ar' (1, j) · cos (2πij / N) -Σ j Br' (1, j) · sin (2πij / N)} + 1 / N · { _{Σ j Ar' (2, j)} · cos (2πij / N) -Σ j Br' (2, j) · sin (2πij / N)} + 1 / N · {Σ j Ar' (3, j) · cos _{(2πij / N) -Σ j Br'} (3, j) · sin (2πij / N)} + Xrp (i + N / 2)

In the above [Expression 16], 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 16] are the data Xlp (i) of the modified acoustic frame modified immediately before, When Xrp (i) exists, the addition is performed in consideration of the overlap of N / 2 samples on the time axis. By the above [Equation 16], each sample Xl ′ (i) of the left channel and each sample Xr ′ (i) of the right channel of the A type modified acoustic frame are obtained. 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 continuous identification mode (S117), and then the acoustic frame reading means 10 A type acoustic frame is read (S104). On the other hand, when the mode is the bit mode or the continuous identification mode, after setting the mode to the bit mode (S118), the low frequency component changing means 30 reads the next bit in the bit array of the access 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 access information is read. When the processing is completed for all the words of the access information, the process returns to the first word of the access 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.

A change in the left channel signal by the above processing will be described with reference to FIG. In FIG. 10, the horizontal direction in the drawing is the time axis and is proportional to the number of samples. In addition, a large number of rectangles in the figure indicate the window 1 component and the window 3 component of the modified acoustic frame. The horizontal width of the rectangle indicating the window component indicates the number of samples, and the vertical width indicates the intensity. However, in FIG. 10, neither the horizontal width nor the vertical width is accurately shown, and a strong signal is applied to the head portion corresponding to the window 1 component. It indicates whether there is a component and whether there is a strong signal component in the rear portion corresponding to the window 3 component. FIG. 10A shows a good signal when there is no acoustic frame in which the total values E ₁ and E ₂ calculated by the above [Equation 9] are less than the level lower limit value Lev, that is, for embedding additional information. The case is shown. FIG. 10B shows a signal that is not good for embedding additional information when there is an acoustic frame in which the combined values E ₁ and E ₂ calculated by the above [Equation 9] are less than the level lower limit value Lev. The case is shown.

  For example, it is assumed that as additional information, a 2-word bit array in which the first word is “11011100” and the second word is “11000001” is embedded. First, at the beginning of each word, the window 1 component and the window 3 component are set to be equal as information indicating a break. This is obtained as a result of executing the processing according to the above [Equation 10] in S112, which is set to the separation mode in S102. Subsequently, before the processing corresponding to each bit of the additional information is performed, information indicating whether the information is new or continued is recorded.

  In the present embodiment, even if there is an acoustic frame that is less than the level lower limit value Lev, since the bit processed at that time is valid and is continuously performed from there, whether the bit is new or continued. It is necessary to record some information. Therefore, after recording the information indicating the break, information indicating whether it is new or continued is recorded. Specifically, the mode determination is performed in the separation mode state (S116), the continuous identification mode is set (S117), and the process returns to the extraction of the B type sound frame without reading the bits of the additional information (S104). ). Then, after frequency conversion (S109), if new, the processing according to [Equation 11] makes the distribution between the window 1 component and the window 3 component, which are low frequency components, the window 3 component dominant. Change (S111).

  After recording the information indicating whether it is new or continued in this way, the mode is determined in the continuous identification mode (S116), so the bit mode is set (S118) and the first bit is read from the register ( S103), the B type sound frame is extracted (S104). In the example of FIG. 10A, since there is no acoustic frame that is less than the level lower limit value Lev, one word is continuously processed in S111. This is because the loop from S103 to S118 was repeated continuously 8 times (in the case of 1 word = 1 byte), and during that time, it was not passed through S110, S112, and S117 because it was less than the level lower limit value Lev. Is shown. As shown in FIG. 10, when the bit value of the additional information is 1, a low frequency component exists in the window 3 component, and when the bit value of the additional information is 0, the low frequency component exists in the window 1 component. . As can be seen from the above [Formula 11] to [Formula 14], in this case, the low frequency component of the other window component is zero.

  In the example of FIG. 10B, as a result of the processing according to the above [Equation 9], there is an acoustic frame that is less than the level lower limit value Lev. In this case, the above [Equation 10] passes through S110 and S112. As a result of executing the processing according to the above, the window 1 component and the window 3 component are set to be equal. In this case, since the separation mode is set in S110, information indicating whether it is new or continued is recorded via S117. In the example of FIG. 10B, when embedding “11011100” of the first word, an acoustic frame less than the level lower limit value Lev appears when 1 bit of “1” of the first bit is first processed. Therefore, after the information indicating the break is recorded, the information indicating the continuation is recorded, and the processing is continued from “1” of the second bit. Then, since “1011” from the second bit to the fifth bit has been processed, an acoustic frame less than the level lower limit value Lev has appeared, so after recording the information indicating the break, the information indicating the continuation is recorded. Then, processing is continued from “1” of the sixth bit.

  In the example of FIG. 10, the case where one word of the additional information is 1 byte has been described. However, in order to record information indicating whether it is new or continued, one word of the additional information is recorded in an arbitrary number of bits. It is possible.

  Of the left channel of the modified acoustic signal obtained as described above, for the portion where the additional information is embedded, the low frequency component is equal to the window 1 component and the window 3 component, or the window 1 There are only three distributions: the component predominates or the window three component predominates. However, since the high frequency component remains the original acoustic signal, it has various distributions based on the setting of the producer. Further, as shown in the above example, when a stereo sound signal is used, the low frequency component changed in the left channel is apparent from the processing of [Expression 10] to [Expression 15]. 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.

  Therefore, the sound signal in which the additional information is embedded as described above is reproduced by the sound signal reproduction device 400, and the sound emitted as the sound from the speaker can be heard as the normal sound signal, but the sound is acquired. In the network connection device 300, additional information can be extracted.

(5. Details of embedded information extraction unit 321)
Next, details of the embedded information extraction unit 321 included in the network connection device 300 and the address information extraction server 320a will be described. FIG. 11 is a diagram illustrating details of the embedded information extraction unit 321. In FIG. 11, 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, 160 is an additional information extraction means, and 170 is an acoustic frame. Holding means.

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 a value E _C1 of each low frequency intensity data for each of the window 1 component and the window 3 component. E _C2 is calculated based on the following [Equation 17], and the total value 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 , the predetermined state is obtained. Has the function to judge. The following [Equation 17] is obtained by deleting the right channel component in the above [Equation 9], and does not refer to the right channel component at the time of extraction.

[Formula 17]
E _C1 = Σ _{j = 1,..., M-3} {Al (1, j) ² + Bl (1, j) ² }
E _C2 = Σ _{j = 1,..., M-3} {Al (3, j) ² + Bl (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.

Next, the processing operation of the embedded information extraction unit 321 shown in FIG. 11 will be described with reference to the flowchart of FIG. First, in this apparatus, the average code levels HL1 and HL2 and the phase determination table are initialized. These will be described. The average code levels HL1 and HL2 are low frequencies calculated by the above [Equation 17] for an acoustic frame (hereinafter referred to as an effective frame) that is determined to have a binary value corresponding to a bit value embedded therein. 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.

  As described above, in a state where the initial value is set, the reference frame acquisition unit 110 extracts an audio frame composed of a predetermined number of samples as the reference frame from the audio signal digitized by the address information extraction unit 320. (S201). Specifically, the reference frame is extracted and read into the acoustic frame holding unit 170. The number of samples of one sound frame read as the reference frame by the reference frame acquisition unit 110 needs to be the same as that set by the sound frame reading unit 10 shown in FIG. Therefore, in the present embodiment, the reference frame acquisition unit 110 sequentially reads 4096 samples 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 embedded information extraction unit 321 can be divided into a reference frame that is set adjacently without interruption from the beginning, and a phase change frame in which the phase has been 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.

  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 the same as the process in the frequency conversion means 20 shown in FIG. However, since only the left channel is used for extraction, the processing according to [Formula 8] is performed, and the imaginary part Bl (1) such as the real part Al (1, j) of the conversion data corresponding to the left channel is performed. , J) etc.

By the processing in the frequency conversion means 130, a frame spectrum expressed by a spectrum that is a component corresponding to the frequency is obtained. Subsequently, the code determination parameter calculation unit 140 calculates the average code levels HL1 and HL2 (S402). Specifically, v1 which is the integrated value E _C1 of the sound frames for which the past window 1 component is determined to be dominant is represented by the number of sound frames for which the past window 1 component is determined to be dominant. HL1 is calculated by dividing by a certain n1, and v2 that is an integrated value of the total value E _C2 for the sound frame for which the past window 3 component is determined to be dominant is determined to be the state where the past window 3 component is dominant. HL2 is calculated by dividing by n2, which is the number of sound frames that have been performed. Therefore, the average code levels HL1 and HL2 are average values of the sum values of the low-frequency intensity data of the acoustic frames that have been determined to have a dominant window component corresponding to the past.

Furthermore, the code determination parameter calculation unit 140 extracts each low frequency intensity data in a predetermined frequency range from the generated frame spectrum. The frequency range to be extracted needs to correspond to the embedding device. Therefore, here, low frequency intensity data having a frequency of about 200 Hz or less is extracted, and the real part Al (j) and imaginary part of the left channel calculated by the above [Equation 8] as in the case of the embedding device. Of Bl (j), those with j ≦ 20 are extracted. The code determination parameter calculating means 140, by executing the processing according to the above [Equation 17], the window 1 component of the sum E _C1, calculating a sum value E _C2 of the window 3 components. The embedded information extraction unit 321 uses this 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 window 1 component of the sum E _C1, window 3 components sum E _C2 of whether each is below a predetermined value (S404). Specifically, 1/20 of the average code levels HL1 and HL2 are set as predetermined values, respectively. When the sum value E _C1 is equal to or less than 1/20 of the average code level HL1 and the sum value E _C2 is equal to or less than 1/20 of the average code level HL2, the code determination parameter calculation unit 140 uses the delimiter information. It is determined that there is (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 18] (S405), and the comparison result is obtained. Output the corresponding code.

[Formula 18]
When E _C2 > (predetermined value) and E _C2 / E _C1 > 2, the window 3 component is dominant. When E _C1 > (predetermined value) and E _C1 / E _C2 > 2, the window 1 component is dominant. Otherwise, both window components are equal

The code determination parameter calculation unit 140 outputs a ternary code according to the determination result for each acoustic frame. That is, when it is determined that the window 3 component is dominant, the first bit value (for example, “1”) is output (S406). When it is determined that the window 1 component is dominant, the second bit value is output. The bit value (eg, “0”) is output (S407), and if it is determined that the two window components are equal, a code indicating delimiter information is output (S408). In S405, if it is determined that the window 3 component is in a dominant state, E _C1 is HL1 or higher. If it is determined that the window 1 component is in a dominant state, is E _C2 higher than HL2. If these conditions are not satisfied, a code indicating delimiter information is output (S408).

  When it is determined that the window 3 component is dominant and the first bit value is output (S406), or when it is determined that the window 1 component is dominant and the second bit value is output (S407). Further, the phase determination table S (p) is updated according to the following [Equation 19] (S409).

[Formula 19]
When the three components of the window are dominant, S (p) ← S (p) + E _C1 / E _C2
When the window 1 component is 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 process according to the above [Equation 9] 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 parameter of the average code level is updated (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. The number of frames n1 and n2 is updated by adding 1 to n1 and n2, respectively. Subsequently, the mode is determined (S204). Two modes, a delimited mode and a bit output mode, are prepared. If it is in the bit output mode, the bit value is stored in the buffer (S209). Subsequently, the bit counter is counted up (S210). On the other hand, if the result of determination in S204 is that the mode is separation mode, it is further determined whether the extracted code means new or continuation (S205). As a result, if it is new, it means that one word has been completed immediately before that, and the additional information extracting means 160 outputs the data for one word recorded in the buffer (S206). Then, the bit counter is initialized to 0 (S207). Further, the mode is set to the bit output mode (S208). In S205, if it is determined to continue, the value should be output to the bit in the buffer, so only the processing for setting the bit output mode is performed. If a code corresponding to the delimiter information is extracted in S202, the mode is set to the delimiter mode in order to extract new or continued information from the next sound frame (S211). By executing the process shown in FIG. 12 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 delimits a code indicating that the window 3 component and the window 1 component are equal among the ternary codes output by the code determination parameter calculation unit 140. Then, the next code is the head, and a bit array is created by associating a code indicating that the window 3 component is dominant and the window 1 component is dominant with the bit value. Subsequently, this bit arrangement is converted according to a predetermined rule and extracted as meaningful additional information. As the predetermined rule, various rules can be applied as long as the information intended by the person who embeds the information can be recognized by the person who has received it. As a rule. That is, the access information extraction means 160 recognizes the code determined by the code determination parameter calculation means 140 and output from the code output means 150 in units of 1 byte (8 bits), and character information according to the set code system. To recognize the access information. In this embodiment, the URL is recognized.

  Therefore, if the embedded device embeds the URL of the web site related to the song name or artist of the song as character information in the acoustic signal, the user can hear the music flowing and listen to the song name or When you want to know detailed information related to the artist, if you perform a specified operation on your mobile terminal that functions as an extraction device, the URL of the web site related to the song or artist on your mobile terminal Can be acquired.

Here, advantages of using window function window functions W (1, i), W (2, i), W (3, i), and W (4, i) as shown in the present embodiment will be described. FIG. 14 is a diagram showing the state of low-frequency components immediately before and after embedding when using a window function that does not have the characteristics of this embodiment, and FIG. It is a figure which shows the state of a frequency component. Comparing FIG. 14B and FIG. 15B, which are low-frequency component states after embedding, in this embodiment, the A-type sound frame has a value in a wider range than in other cases. I understand that. Therefore, the sum values E _C1 and E _C2 calculated using [Equation 17] in S402 are increased, and the probability that a bit value is extracted in S406 and S407 is increased.

  In the above processing, in order to accurately extract additional information in the embedded information extraction unit 321, processing for correcting the phase, processing for correcting the intensity balance between the window 3 component and the window 1 component, and an invalid frame The process which correct | amends the lower limit threshold value for judging this is performed. Next, supplementary explanation will be given for these three correction processes.

(5.2. 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.

(5.3. About lower limit threshold correction processing)
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.

(6. Monaural sound signal)
In the above embodiment, the case where both the embedding device and the embedding information extraction unit 321 embed additional information in the left channel signal of a stereo sound signal having left and right channels has been described as an example. Additional information may be embedded in. This is because the present invention is not related to the left and right characteristics. Further, when processing is performed on a monaural sound signal having only one channel, the processing performed on the left channel signal is performed in the above embodiment. Since the present invention embeds and extracts additional information from one channel signal, it can be similarly performed for a monaural sound signal or a stereo sound signal.

  FIG. 16 is a conceptual diagram in the case where additional information is embedded in the stereo sound signal and the monaural sound signal according to the present invention. FIG. 16A shows a stereo sound signal, and FIG. 16B shows a monaural sound signal. In the example of FIG. 16, the low frequency component for one acoustic frame is represented by a waveform, and the bit value “0” is embedded as an example.

  In the case of a stereo sound signal, embedding is performed on the left channel (L-ch) signal. As shown in FIG. 16A, after frequency conversion, signal separation is performed, and bit embedding processing is further performed. Specifically, bit embedding is performed as a result of the processing of [Equation 10] to [Equation 15]. Here, as described above, when “0” is embedded, [Equation 13] and [Equation 14] are used. Therefore, after bit embedding processing, the signal component in the low frequency band is 0 (represented by no waveform in the figure) near the center of the acoustic frame (corresponding to window 2) and near the rear (corresponding to window 3). Become. At this time, as is clear from the contents of [Equation 13] and [Equation 14], the signal component from which the left channel signal is deleted is added to the right channel (R-ch) signal. Therefore, as shown in the lower part of FIG. 16A, the low frequency component of the right channel signal is increased. After bit embedding processing, signal synthesis including high frequency components is performed, and then frequency inverse conversion is performed to obtain a modified acoustic signal. On the other hand, as is apparent from the last three equations of [Equation 11], the right channel (R-ch) signal component corresponding to the remaining signal component of the left channel signal is added to the left channel signal. . Therefore, as shown in the upper part of FIG. 16A, the low frequency component corresponding to the window 1 of the left channel signal becomes large.

  In the case of a monaural sound signal, processing is performed as shown in FIG. 16B. As can be seen from comparison with the upper part of FIG. 16A, processing similar to that for the left channel of a stereo sound signal is performed. Become.

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

  In this case, the information embedding process in the embedding apparatus shown in FIG. 7 is performed according to the flowchart of FIG. The flowchart of FIG. 17 differs from the flowchart of FIG. 8 in that the low frequency component changing unit 30 does not determine the level in the frequency conversion process in S709, and there is no setting process for the separation mode corresponding to S110. Is a point. This is because the processing according to FIG. 17 forcibly embeds information even if the signal level is low. Therefore, it is determined whether there is a portion with a low signal level where information cannot be embedded. This is because it is not necessary to set the mode.

Therefore, as a process for setting one of the window 1 component and the window 3 component in S710 to be in an advantageous state, first, the fixed value V calculated according to the following [Equation 20] is used as the intensity of the low frequency component. It is set in place of the total values E ₁ and E ₂ .

[Formula 20]
V = {0.5 · Lev / (M−3)} ^1/2

  When the first value is 1 and the second value is 0, when the bit value is 1, the processing according to the above [Formula 11] and [Formula 12] is executed, and then the following [Formula 21 ] Is executed.

[Formula 21]
For the three components of the window, Al ′ (3, j) = Al (3, j) · V / {Ar (3, j) ² + Br (3, j) ² } ^1/2
Bl ′ (3, j) = Bl (3, j) · V / {Ar (3, j) ² + Br (3, j) ² } ^1/2

  When the bit value is 0, the process according to the above [Formula 13] and [Formula 14] is executed, and then the process according to the following [Formula 22] is executed.

[Formula 22]
For one component of window, Al ′ (1, j) = Al (1, j) · V / {Ar (1, j) ² + Br (1, j) ² } ^1/2
Bl ′ (1, j) = Bl (1, j) · V / {Ar (1, j) ² + Br (1, j) ² } ^1/2

  After performing the above processing in S710, the processing after the window two component deletion processing (S712) is performed in the same manner as the processing after S113 shown in FIG.

  As described above, even when information is embedded when the frequency component is small, the configuration of the extraction device for extracting information from the acoustic signal on the extraction side is the same as that in FIG. 11, and the processing operation is the flowchart of FIG. Is the same as

  FIG. 18 shows a conceptual diagram when embedding additional information when the signal component is small with respect to the stereo sound signal and the monaural sound signal. FIG. 18A shows a stereo sound signal, and FIG. 18B shows a monaural sound signal. In the example of FIG. 18, as in the case of FIG. 16, a low-frequency component for one acoustic frame is represented by a waveform and a bit value “0” is embedded as an example.

  18 is different from the case of FIG. 16 in that the original signal component is small. As in the example shown in FIG. 18A, in the stage after the signal separation, even if the values of the window 1 component and the window 3 component are small, the bits are processed in accordance with the processing of [Expression 20] to [Expression 22]. By embedding, it has the same signal component as in FIG.

  In the case of a monaural sound signal, processing is performed as shown in FIG. 18B, but as can be seen from the upper part of FIG. 18A, processing similar to that for the left channel of a stereo sound signal is performed. Become.

It is a block diagram of the network connection apparatus which concerns on the 1st Embodiment of this invention. It is a flowchart which shows the processing operation of the network connection apparatus which concerns on the 1st Embodiment of this invention. It is a block diagram of the network connection apparatus which concerns on the 2nd Embodiment of this invention. It is a flowchart which shows the processing operation of the network connection apparatus which concerns on the 2nd Embodiment of this invention. It is a block diagram of the network connection system which concerns on the 3rd Embodiment of this invention. It is a flowchart which shows the processing operation of the network connection system which concerns on the 3rd Embodiment of this invention. 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 window function used by this invention. FIG. 9 shows how the low frequency component changes due to the processing according to FIG. 8. 5 is a functional block diagram showing details of an embedded information extraction unit 321. FIG. 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 conceptual diagram which shows the low frequency part of the acoustic signal before and behind embedding when not using the window function of this embodiment. It is a conceptual diagram which shows the low frequency part of the acoustic signal before and behind embedding by this embodiment. It is a conceptual diagram of the embedding process of the additional information by this invention. FIG. 8 is a flowchart showing an outline of processing in the apparatus shown in FIG. 7 when information can be embedded even if the original signal component is small. It is a conceptual diagram of an additional information embedding process when the original signal component is small.

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 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 300 ... Network connection device 310 ... Microphone 320 .. Address information extraction unit 320a ... Address information server 321 ... Embedded information extraction unit 322 ... Address information conversion Unit 330 ... Network access unit 340 ... Content acquisition unit 350 ... Acoustic signal acquisition unit 400 ... Acoustic signal reproduction device 500 ... Network 601 ... Site 602 ... Second site

Claims (7)

A device that obtains sound emitted from a speaker as an acoustic signal, extracts information embedded in an inaudible state in advance from the acoustic signal, and accesses a predetermined site on the network using the extracted information. And
An acoustic frame acquisition means for digitizing a predetermined section of the acoustic signal and acquiring an acoustic frame composed of a predetermined number of samples;
Frequency conversion means for performing frequency conversion on the acoustic frame to obtain a spectrum for each acoustic frame;
Using the generated spectrum, at least two sets of spectrum sets are extracted, a sum value of spectrum intensities is calculated for each spectrum set, and a predetermined code is calculated based on a ratio of the sum values between the spectrum sets. Encoding means for outputting
Address information extraction means for converting the information array corresponding to the output code according to a predetermined rule and extracting address information;
Network access means for accessing a predetermined site on the network using the address information;
A network connection device using an acoustic signal. In claim 1,
The frequency conversion means performs frequency conversion on the acoustic frame using a first window function and a third window function, respectively, and a first window spectrum and a third window which are spectra corresponding to the first window function. A third window spectrum that is a spectrum corresponding to the function,
The encoding means extracts each spectrum set corresponding to a predetermined low frequency band from each generated window spectrum, calculates a sum value of spectrum intensities for each spectrum set, and between the spectrum sets of the sum value A network connection device using an acoustic signal, which outputs a predetermined code based on the ratio of. In claim 1 or claim 2,
Furthermore, it has content acquisition means for acquiring content including an acoustic signal from the accessed site,
The address information extracting means extracts second address information from an acoustic signal included in the acquired content;
The network connection device using an acoustic signal, wherein the network access means accesses the second site on the network using the second address information. In any one of Claims 1-3,
The network connection apparatus, wherein the acoustic frame acquisition means, the frequency conversion means, the encoding means, the address information extraction means, and the network access means are provided inside a single mobile phone.   In any one of Claims 1 to 4, the address information extraction unit extracts an information array corresponding to the output code as additional information, and then converts it into address information using a conversion table. A network connection device for obtaining address information. The terminal device and the server computer on the network cooperate with each other, acquire the sound emitted from the speaker as an acoustic signal, extract information embedded in an inaudible state in advance from the acoustic signal, and use the extracted information A system for accessing a predetermined site on a network,
The terminal device has transfer means for transferring the acquired acoustic signal to the server computer, and network access means for accessing a predetermined site on the network using address information received from the server computer,
The server computer digitizes a predetermined section of the acoustic signal received from the terminal device to acquire an acoustic frame composed of a predetermined number of samples; and first to the acoustic frame Frequency conversion is performed using each of the window function and the third window function to generate a first window spectrum that is a spectrum corresponding to the first window function and a third window spectrum that is a spectrum corresponding to the third window function. A spectrum conversion unit and a spectrum set corresponding to a predetermined low frequency band are extracted from each of the generated window spectra, and a sum of spectrum intensities is calculated for each spectrum set. Based on the ratio, the encoding means for outputting a predetermined code and an information array corresponding to the output code according to a predetermined rule Networked system using an acoustic signal, characterized in that the address information extraction means for extracting the transformation to the address information, is the extracted address information having address information transmitting means for transmitting to the terminal device Ri. A device that embeds address information for accessing a predetermined site on a network in an inaudible state for 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;
The A type acoustic frame is subjected to frequency conversion using a first window function, a second window function, and a third window function, respectively, and a first window spectrum, which is a spectrum corresponding to the first window function, A second window spectrum that is a spectrum corresponding to a two-window function and a third window spectrum that is a spectrum corresponding to the third window function are generated, and a fourth window function is used for the B-type acoustic frame. Frequency conversion means for performing frequency conversion and generating a fourth window spectrum that is a spectrum corresponding to the fourth window function;
A low frequency spectrum corresponding to a predetermined low frequency band is extracted from each of the generated window spectrums, 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 window function and the rear third window function and removing each component of the low frequency spectrum by the second window function and the fourth 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,
When the first window function, the second window function, the third window function, and the fourth window function are added, the whole section is set to a fixed value of 1,
The first window function and the second window function, and the second window function and the third window function are set so that there are locations where both have non-zero values at the same time,
The first window function and the third window function are set so that when one has a non-zero value, the other is always 0, and both sides of each window function have an asymmetric cosine function. A device for embedding address information for an acoustic signal, characterized in that:

Images